diff options
Diffstat (limited to 'sound')
245 files changed, 18727 insertions, 4878 deletions
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a2104671f51d..5dcf88bed9b7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1242,6 +1242,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par return -EINVAL; return 0; } +EXPORT_SYMBOL(snd_pcm_hw_constraint_mask64); /** * snd_pcm_hw_constraint_integer - apply an integer constraint to an interval diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index ec4536c8d8d4..dafcf82139e2 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -932,7 +932,7 @@ int snd_hda_bus_new(struct snd_card *card, } EXPORT_SYMBOL_GPL(snd_hda_bus_new); -#ifdef CONFIG_SND_HDA_GENERIC +#if IS_ENABLED(CONFIG_SND_HDA_GENERIC) #define is_generic_config(codec) \ (codec->modelname && !strcmp(codec->modelname, "generic")) #else @@ -1339,23 +1339,15 @@ get_hda_cvt_setup(struct hda_codec *codec, hda_nid_t nid) /* * Dynamic symbol binding for the codec parsers */ -#ifdef MODULE -#define load_parser_sym(sym) ((int (*)(struct hda_codec *))symbol_request(sym)) -#define unload_parser_addr(addr) symbol_put_addr(addr) -#else -#define load_parser_sym(sym) (sym) -#define unload_parser_addr(addr) do {} while (0) -#endif #define load_parser(codec, sym) \ - ((codec)->parser = load_parser_sym(sym)) + ((codec)->parser = (int (*)(struct hda_codec *))symbol_request(sym)) static void unload_parser(struct hda_codec *codec) { - if (codec->parser) { - unload_parser_addr(codec->parser); - codec->parser = NULL; - } + if (codec->parser) + symbol_put_addr(codec->parser); + codec->parser = NULL; } /* @@ -1570,7 +1562,7 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec) EXPORT_SYMBOL_GPL(snd_hda_codec_update_widgets); -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) /* if all audio out widgets are digital, let's assume the codec as a HDMI/DP */ static bool is_likely_hdmi_codec(struct hda_codec *codec) { @@ -1620,12 +1612,20 @@ int snd_hda_codec_configure(struct hda_codec *codec) patch = codec->preset->patch; if (!patch) { unload_parser(codec); /* to be sure */ - if (is_likely_hdmi_codec(codec)) + if (is_likely_hdmi_codec(codec)) { +#if IS_MODULE(CONFIG_SND_HDA_CODEC_HDMI) patch = load_parser(codec, snd_hda_parse_hdmi_codec); -#ifdef CONFIG_SND_HDA_GENERIC - if (!patch) +#elif IS_BUILTIN(CONFIG_SND_HDA_CODEC_HDMI) + patch = snd_hda_parse_hdmi_codec; +#endif + } + if (!patch) { +#if IS_MODULE(CONFIG_SND_HDA_GENERIC) patch = load_parser(codec, snd_hda_parse_generic_codec); +#elif IS_BUILTIN(CONFIG_SND_HDA_GENERIC) + patch = snd_hda_parse_generic_codec; #endif + } if (!patch) { printk(KERN_ERR "hda-codec: No codec parser is available\n"); return -ENODEV; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8321a97d5c05..d9a09bdd09db 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3269,7 +3269,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, mutex_unlock(&codec->control_mutex); snd_hda_codec_flush_cache(codec); /* flush the updates */ if (err >= 0 && spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return err; } @@ -3390,7 +3390,7 @@ static int cap_single_sw_put(struct snd_kcontrol *kcontrol, return ret; if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, ucontrol); + spec->cap_sync_hook(codec, kcontrol, ucontrol); return ret; } @@ -3795,7 +3795,7 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx, return 0; snd_hda_activate_path(codec, path, true, false); if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); path_power_down_sync(codec, old_path); return 1; } @@ -5270,7 +5270,7 @@ static void init_input_src(struct hda_codec *codec) } if (spec->cap_sync_hook) - spec->cap_sync_hook(codec, NULL); + spec->cap_sync_hook(codec, NULL, NULL); } /* set right pin controls for digital I/O */ diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 07f767231c9f..c908afbe4d94 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -274,6 +274,7 @@ struct hda_gen_spec { void (*init_hook)(struct hda_codec *codec); void (*automute_hook)(struct hda_codec *codec); void (*cap_sync_hook)(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); /* PCM hooks */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index fa2879a21a50..e354ab1ec20f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -198,7 +198,7 @@ MODULE_DESCRIPTION("Intel HDA driver"); #endif #if defined(CONFIG_PM) && defined(CONFIG_VGA_SWITCHEROO) -#ifdef CONFIG_SND_HDA_CODEC_HDMI +#if IS_ENABLED(CONFIG_SND_HDA_CODEC_HDMI) #define SUPPORT_VGA_SWITCHEROO #endif #endif diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 7a426ed491f2..8ed0bcc01386 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -244,6 +244,19 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec, } } +/* Toshiba Satellite L40 implements EAPD in a standard way unlike others */ +static void ad1986a_fixup_eapd(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + codec->inv_eapd = 0; + spec->gen.keep_eapd_on = 1; + spec->eapd_nid = 0x1b; + } +} + enum { AD1986A_FIXUP_INV_JACK_DETECT, AD1986A_FIXUP_ULTRA, @@ -251,6 +264,7 @@ enum { AD1986A_FIXUP_3STACK, AD1986A_FIXUP_LAPTOP, AD1986A_FIXUP_LAPTOP_IMIC, + AD1986A_FIXUP_EAPD, }; static const struct hda_fixup ad1986a_fixups[] = { @@ -311,6 +325,10 @@ static const struct hda_fixup ad1986a_fixups[] = { .chained_before = 1, .chain_id = AD1986A_FIXUP_LAPTOP, }, + [AD1986A_FIXUP_EAPD] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1986a_fixup_eapd, + }, }; static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { @@ -318,6 +336,7 @@ static const struct snd_pci_quirk ad1986a_fixup_tbl[] = { SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40", AD1986A_FIXUP_EAPD), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP), SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA), @@ -472,6 +491,8 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec) static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + static hda_nid_t conn_0c[] = { 0x08 }; + static hda_nid_t conn_0d[] = { 0x09 }; int err; err = alloc_ad_spec(codec); @@ -479,8 +500,14 @@ static int patch_ad1983(struct hda_codec *codec) return err; spec = codec->spec; + spec->gen.mixer_nid = 0x0e; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + + /* limit the loopback routes not to confuse the parser */ + snd_hda_override_conn_list(codec, 0x0c, ARRAY_SIZE(conn_0c), conn_0c); + snd_hda_override_conn_list(codec, 0x0d, ARRAY_SIZE(conn_0d), conn_0d); + err = ad198x_parse_auto_config(codec, false); if (err < 0) goto error; @@ -999,6 +1026,9 @@ static void ad1884_fixup_thinkpad(struct hda_codec *codec, spec->gen.keep_eapd_on = 1; spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook; spec->eapd_nid = 0x12; + /* Analog PC Beeper - allow firmware/ACPI beeps */ + spec->beep_amp = HDA_COMPOSE_AMP_VAL(0x20, 3, 3, HDA_INPUT); + spec->gen.beep_nid = 0; /* no digital beep */ } } @@ -1065,6 +1095,7 @@ static int patch_ad1884(struct hda_codec *codec) spec = codec->spec; spec->gen.mixer_nid = 0x20; + spec->gen.mixer_merge_nid = 0x21; spec->gen.beep_nid = 0x10; set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 54d14793725a..46ecdbb9053f 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2662,60 +2662,6 @@ static bool dspload_wait_loaded(struct hda_codec *codec) } /* - * PCM stuffs - */ -static void ca0132_setup_stream(struct hda_codec *codec, hda_nid_t nid, - u32 stream_tag, - int channel_id, int format) -{ - unsigned int oldval, newval; - - if (!nid) - return; - - snd_printdd( - "ca0132_setup_stream: NID=0x%x, stream=0x%x, " - "channel=%d, format=0x%x\n", - nid, stream_tag, channel_id, format); - - /* update the format-id if changed */ - oldval = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_STREAM_FORMAT, - 0); - if (oldval != format) { - msleep(20); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_STREAM_FORMAT, - format); - } - - oldval = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - newval = (stream_tag << 4) | channel_id; - if (oldval != newval) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CHANNEL_STREAMID, - newval); - } -} - -static void ca0132_cleanup_stream(struct hda_codec *codec, hda_nid_t nid) -{ - unsigned int val; - - if (!nid) - return; - - snd_printdd(KERN_INFO "ca0132_cleanup_stream: NID=0x%x\n", nid); - - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONV, 0); - if (!val) - return; - - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, 0); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID, 0); -} - -/* * PCM callbacks */ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, @@ -2726,7 +2672,7 @@ static int ca0132_playback_pcm_prepare(struct hda_pcm_stream *hinfo, { struct ca0132_spec *spec = codec->spec; - ca0132_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, spec->dacs[0], stream_tag, 0, format); return 0; } @@ -2745,7 +2691,7 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) msleep(50); - ca0132_cleanup_stream(codec, spec->dacs[0]); + snd_hda_codec_cleanup_stream(codec, spec->dacs[0]); return 0; } @@ -2822,10 +2768,8 @@ static int ca0132_capture_pcm_prepare(struct hda_pcm_stream *hinfo, unsigned int format, struct snd_pcm_substream *substream) { - struct ca0132_spec *spec = codec->spec; - - ca0132_setup_stream(codec, spec->adcs[substream->number], - stream_tag, 0, format); + snd_hda_codec_setup_stream(codec, hinfo->nid, + stream_tag, 0, format); return 0; } @@ -2839,7 +2783,7 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, if (spec->dsp_state == DSP_DOWNLOADING) return 0; - ca0132_cleanup_stream(codec, hinfo->nid); + snd_hda_codec_cleanup_stream(codec, hinfo->nid); return 0; } @@ -4742,6 +4686,8 @@ static int patch_ca0132(struct hda_codec *codec) return err; codec->patch_ops = ca0132_patch_ops; + codec->pcm_format_first = 1; + codec->no_sticky_stream = 1; return 0; } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4e0ec146553d..bcf91bea3317 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3291,7 +3291,8 @@ static void cxt_update_headset_mode(struct hda_codec *codec) } static void cxt_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { cxt_update_headset_mode(codec); } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 56a8f1876603..8d0a84436674 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -708,7 +708,8 @@ static void alc_inv_dmic_sync(struct hda_codec *codec, bool force) } static void alc_inv_dmic_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_inv_dmic_sync(codec, false); } @@ -1821,6 +1822,7 @@ enum { ALC889_FIXUP_IMAC91_VREF, ALC889_FIXUP_MBA11_VREF, ALC889_FIXUP_MBA21_VREF, + ALC889_FIXUP_MP11_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -2190,6 +2192,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC889_FIXUP_MBP_VREF, }, + [ALC889_FIXUP_MP11_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba11_vref, + .chained = true, + .chain_id = ALC885_FIXUP_MACPRO_GPIO, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2253,7 +2261,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_FIXUP_MACPRO_GPIO), + SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC889_FIXUP_MP11_VREF), SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC889_FIXUP_MBP_VREF), @@ -3211,7 +3219,8 @@ static void alc269_fixup_hp_gpio_mute_hook(void *private_data, int enabled) /* turn on/off mic-mute LED per capture hook */ static void alc269_fixup_hp_gpio_mic_mute_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct alc_spec *spec = codec->spec; unsigned int oldval = spec->gpio_led; @@ -3521,7 +3530,8 @@ static void alc_update_headset_mode(struct hda_codec *codec) } static void alc_update_headset_mode_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { alc_update_headset_mode(codec); } @@ -3606,6 +3616,19 @@ static void alc_fixup_auto_mute_via_amp(struct hda_codec *codec, } } +static void alc_no_shutup(struct hda_codec *codec) +{ +} + +static void alc_fixup_no_shutup(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + struct alc_spec *spec = codec->spec; + spec->shutup = alc_no_shutup; + } +} + static void alc_fixup_headset_mode_alc668(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3834,6 +3857,7 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC269_FIXUP_NO_SHUTUP, ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, @@ -4010,6 +4034,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, }, + [ALC269_FIXUP_NO_SHUTUP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_no_shutup, + }, [ALC269_FIXUP_LENOVO_DOCK] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4243,6 +4271,7 @@ static const struct hda_fixup alc269_fixups[] = { }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x0283, "Acer TravelMate 8371", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700), @@ -4298,7 +4327,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0651, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0652, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0653, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0657, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4307,6 +4338,54 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), + /* ALC282 */ + SND_PCI_QUIRK(0x103c, 0x220f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2213, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2266, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2267, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2268, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2269, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22a0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22bf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c1, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cd, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22ce, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22cf, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22d0, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + /* ALC290 */ + SND_PCI_QUIRK(0x103c, 0x2260, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2261, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2262, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2263, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2264, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2265, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x227f, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2280, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2281, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2282, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x2289, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228a, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228c, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228d, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x228e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c5, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c6, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c8, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c3, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), + SND_PCI_QUIRK(0x103c, 0x22c4, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -4322,6 +4401,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), + SND_PCI_QUIRK(0x104d, 0x90b5, "Sony VAIO Pro 11", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), @@ -4343,6 +4423,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), @@ -5096,12 +5177,13 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x060a, "Dell XPS 13", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0623, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_AUTO_MUTE), - SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x064e, "Dell", ALC668_FIXUP_AUTO_MUTE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6998cf29b9bc..3bc29c9b2529 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -83,6 +83,7 @@ enum { STAC_DELL_M6_BOTH, STAC_DELL_EQ, STAC_ALIENWARE_M17X, + STAC_92HD89XX_HP_FRONT_JACK, STAC_92HD73XX_MODELS }; @@ -97,6 +98,7 @@ enum { STAC_92HD83XXX_HP_LED, STAC_92HD83XXX_HP_INV_LED, STAC_92HD83XXX_HP_MIC_LED, + STAC_HP_LED_GPIO10, STAC_92HD83XXX_HEADSET_JACK, STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, @@ -194,7 +196,7 @@ struct sigmatel_spec { int default_polarity; unsigned int mic_mute_led_gpio; /* capture mute LED GPIO */ - bool mic_mute_led_on; /* current mic mute state */ + unsigned int mic_enabled; /* current mic mute state (bitmask) */ /* stream */ unsigned int stream_delay; @@ -324,19 +326,26 @@ static void stac_gpio_set(struct hda_codec *codec, unsigned int mask, /* hook for controlling mic-mute LED GPIO */ static void stac_capture_led_hook(struct hda_codec *codec, - struct snd_ctl_elem_value *ucontrol) + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) { struct sigmatel_spec *spec = codec->spec; - bool mute; + unsigned int mask; + bool cur_mute, prev_mute; - if (!ucontrol) + if (!kcontrol || !ucontrol) return; - mute = !(ucontrol->value.integer.value[0] || - ucontrol->value.integer.value[1]); - if (spec->mic_mute_led_on != mute) { - spec->mic_mute_led_on = mute; - if (mute) + mask = 1U << snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + prev_mute = !spec->mic_enabled; + if (ucontrol->value.integer.value[0] || + ucontrol->value.integer.value[1]) + spec->mic_enabled |= mask; + else + spec->mic_enabled &= ~mask; + cur_mute = !spec->mic_enabled; + if (cur_mute != prev_mute) { + if (cur_mute) spec->gpio_data |= spec->mic_mute_led_gpio; else spec->gpio_data &= ~spec->mic_mute_led_gpio; @@ -1788,6 +1797,12 @@ static const struct hda_pintbl intel_dg45id_pin_configs[] = { {} }; +static const struct hda_pintbl stac92hd89xx_hp_front_jack_pin_configs[] = { + { 0x0a, 0x02214030 }, + { 0x0b, 0x02A19010 }, + {} +}; + static void stac92hd73xx_fixup_ref(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -1906,6 +1921,10 @@ static const struct hda_fixup stac92hd73xx_fixups[] = { [STAC_92HD73XX_NO_JD] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd73xx_fixup_no_jd, + }, + [STAC_92HD89XX_HP_FRONT_JACK] = { + .type = HDA_FIXUP_PINS, + .v.pins = stac92hd89xx_hp_front_jack_pin_configs, } }; @@ -1966,6 +1985,8 @@ static const struct snd_pci_quirk stac92hd73xx_fixup_tbl[] = { "Alienware M17x", STAC_ALIENWARE_M17X), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0490, "Alienware M17x R3", STAC_DELL_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x2b17, + "unknown HP", STAC_92HD89XX_HP_FRONT_JACK), {} /* terminator */ }; @@ -2110,6 +2131,17 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, } } +static void stac92hd83xxx_fixup_hp_led_gpio10(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct sigmatel_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->gpio_led = 0x10; /* GPIO4 */ + spec->default_polarity = 0; + } +} + static void stac92hd83xxx_fixup_headset_jack(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -2604,6 +2636,12 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP, }, + [STAC_HP_LED_GPIO10] = { + .type = HDA_FIXUP_FUNC, + .v.func = stac92hd83xxx_fixup_hp_led_gpio10, + .chained = true, + .chain_id = STAC_92HD83XXX_HP, + }, [STAC_92HD83XXX_HEADSET_JACK] = { .type = HDA_FIXUP_FUNC, .v.func = stac92hd83xxx_fixup_headset_jack, @@ -2682,6 +2720,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1888, "HP Envy Spectre", STAC_HP_ENVY_BASS), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1899, + "HP Folio 13", STAC_HP_LED_GPIO10), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18df, "HP Folio", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x18F8, @@ -4462,7 +4502,7 @@ static void stac_setup_gpio(struct hda_codec *codec) if (spec->mic_mute_led_gpio) { spec->gpio_mask |= spec->mic_mute_led_gpio; spec->gpio_dir |= spec->mic_mute_led_gpio; - spec->mic_mute_led_on = true; + spec->mic_enabled = 0; spec->gpio_data |= spec->mic_mute_led_gpio; spec->gen.cap_sync_hook = stac_capture_led_hook; diff --git a/sound/pci/hda/thinkpad_helper.c b/sound/pci/hda/thinkpad_helper.c index 5799fbc24c28..8fe3b8c18ed4 100644 --- a/sound/pci/hda/thinkpad_helper.c +++ b/sound/pci/hda/thinkpad_helper.c @@ -39,6 +39,7 @@ static void update_tpacpi_mute_led(void *private_data, int enabled) } static void update_tpacpi_micmute_led(struct hda_codec *codec, + struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { if (!ucontrol || !led_set_func) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index d62ce483a443..0060b31cc3f3 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -50,6 +50,7 @@ source "sound/soc/pxa/Kconfig" source "sound/soc/samsung/Kconfig" source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" +source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 62a1822e77bf..5f1df02984f8 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -27,6 +27,7 @@ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += samsung/ obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ +obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index e634eb78ed03..4789619a52d8 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -58,6 +58,6 @@ config SND_AT91_SOC_AFEB9260 depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC select SND_ATMEL_SOC_PDC select SND_ATMEL_SOC_SSC - select SND_SOC_TLV320AIC23 + select SND_SOC_TLV320AIC23_I2C help Say Y here to support sound on AFEB9260 board. diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 1ead3c977a51..de433cfd044c 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -341,6 +341,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, { int id = dai->id; struct atmel_ssc_info *ssc_p = &ssc_info[id]; + struct ssc_device *ssc = ssc_p->ssc; struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; @@ -466,7 +467,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(RCMR_START, start_event) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) - | SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_PIN : SSC_CKS_CLOCK); rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) @@ -481,7 +483,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TCMR_START, start_event) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) - | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_CLOCK : SSC_CKS_PIN); tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(TFMR_FSDEN, 0) @@ -550,7 +553,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(RCMR_START, SSC_START_RISING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) - | SSC_BF(RCMR_CKS, SSC_CKS_PIN); + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_PIN : SSC_CKS_CLOCK); rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) @@ -565,7 +569,8 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TCMR_START, SSC_START_RISING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) - | SSC_BF(TCMR_CKS, SSC_CKS_PIN); + | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? + SSC_CKS_CLOCK : SSC_CKS_PIN); tfmr = SSC_BF(TFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(TFMR_FSDEN, 0) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index f15bff1548f8..174bd546c08b 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -155,25 +155,14 @@ static int at91sam9g20ek_wm8731_init(struct snd_soc_pcm_runtime *rtd) return ret; } - /* Add specific widgets */ - snd_soc_dapm_new_controls(dapm, at91sam9g20ek_dapm_widgets, - ARRAY_SIZE(at91sam9g20ek_dapm_widgets)); - /* Set up specific audio path interconnects */ - snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon)); - /* not connected */ snd_soc_dapm_nc_pin(dapm, "RLINEIN"); snd_soc_dapm_nc_pin(dapm, "LLINEIN"); -#ifdef ENABLE_MIC_INPUT - snd_soc_dapm_enable_pin(dapm, "Int Mic"); -#else - snd_soc_dapm_nc_pin(dapm, "Int Mic"); +#ifndef ENABLE_MIC_INPUT + snd_soc_dapm_nc_pin(&rtd->card->dapm, "Int Mic"); #endif - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); - return 0; } @@ -194,6 +183,11 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .dai_link = &at91sam9g20ek_dai, .num_links = 1, .set_bias_level = at91sam9g20ek_set_bias_level, + + .dapm_widgets = at91sam9g20ek_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), }; static int at91sam9g20ek_audio_probe(struct platform_device *pdev) diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..6347d5910138 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -11,20 +11,20 @@ config SND_BF5XX_I2S config SND_BF5XX_SOC_SSM2602 tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x - select SND_SOC_SSM2602 + select SND_SOC_SSM2602_SPI if SPI_MASTER + select SND_SOC_SSM2602_I2C if I2C help Say Y if you want to add support for the Analog Devices SSM2602 Audio Codec Add-On Card. config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && I2C select SND_BF5XX_SOC_I2S select SND_SOC_ADAU1701 - select I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ board connected to one of the Blackfin evaluation boards like the @@ -45,9 +45,10 @@ config SND_SOC_BFIN_EVAL_ADAU1373 config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S - select SND_SOC_ADAV80X + select SND_SOC_ADAV801 if SPI_MASTER + select SND_SOC_ADAV803 if I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or EVAL-ADAV803 board connected to one of the Blackfin evaluation boards @@ -58,7 +59,7 @@ config SND_SOC_BFIN_EVAL_ADAV80X config SND_BF5XX_SOC_AD1836 tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SPI_MASTER select SND_BF5XX_SOC_I2S select SND_SOC_AD1836 help @@ -66,9 +67,10 @@ config SND_BF5XX_SOC_AD1836 config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S + depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI select SND_BF5XX_SOC_I2S - select SND_SOC_AD193X + select SND_SOC_AD193X_I2C if I2C + select SND_SOC_AD193X_SPI if SPI_MASTER help Say Y if you want to add support for AD193X codec on Blackfin. This driver supports AD1936, AD1937, AD1938 and AD1939. diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 06f938deda15..5477c5475923 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -1,6 +1,6 @@ config SND_EP93XX_SOC tristate "SoC Audio support for the Cirrus Logic EP93xx series" - depends on (ARCH_EP93XX || COMPILE_TEST) && SND_SOC + depends on ARCH_EP93XX || COMPILE_TEST select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to @@ -18,7 +18,7 @@ config SND_EP93XX_SOC_SNAPPERCL15 tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23 + select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for I2S audio on the Bluewater Systems Snapper CL15 module. diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index 29238a7476dd..5b68b106cfc2 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -65,18 +65,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MICIN", NULL, "Mic Jack"}, }; -static int snappercl15_tlv320aic23_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, tlv320aic23_dapm_widgets, - ARRAY_SIZE(tlv320aic23_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; -} - static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", @@ -84,7 +72,6 @@ static struct snd_soc_dai_link snappercl15_dai = { .codec_dai_name = "tlv320aic23-hifi", .codec_name = "tlv320aic23-codec.0-001a", .platform_name = "ep93xx-i2s", - .init = snappercl15_tlv320aic23_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | SND_SOC_DAIFMT_CBS_CFS, .ops = &snappercl15_ops, @@ -95,6 +82,11 @@ static struct snd_soc_card snd_soc_snappercl15 = { .owner = THIS_MODULE, .dai_link = &snappercl15_dai, .num_links = 1, + + .dapm_widgets = tlv320aic23_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int snappercl15_probe(struct platform_device *pdev) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 75d0ad5d2dcb..b07e17160f94 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -448,38 +448,38 @@ static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"}; static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"}; -static const struct soc_enum pm860x_hs1_opamp_enum = - SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs1_opamp_enum, + PM860X_HS1_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_hs2_opamp_enum = - SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs2_opamp_enum, + PM860X_HS2_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_hs1_pa_enum = - SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs1_pa_enum, + PM860X_HS1_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_hs2_pa_enum = - SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs2_pa_enum, + PM860X_HS2_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_lo1_opamp_enum = - SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo1_opamp_enum, + PM860X_LO1_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_lo2_opamp_enum = - SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo2_opamp_enum, + PM860X_LO2_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_lo1_pa_enum = - SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo1_pa_enum, + PM860X_LO1_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_lo2_pa_enum = - SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo2_pa_enum, + PM860X_LO2_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_spk_pa_enum = - SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_spk_pa_enum, + PM860X_EAR_CTRL_1, 5, pm860x_pa_texts); -static const struct soc_enum pm860x_ear_pa_enum = - SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_ear_pa_enum, + PM860X_EAR_CTRL_2, 0, pm860x_pa_texts); -static const struct soc_enum pm860x_spk_ear_opamp_enum = - SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_spk_ear_opamp_enum, + PM860X_EAR_CTRL_1, 3, pm860x_opamp_texts); static const struct snd_kcontrol_new pm860x_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2, @@ -561,8 +561,8 @@ static const char *aif1_text[] = { "PCM L", "PCM R", }; -static const struct soc_enum aif1_enum = - SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text); +static SOC_ENUM_SINGLE_DECL(aif1_enum, + PM860X_PCM_IFACE_3, 6, aif1_text); static const struct snd_kcontrol_new aif1_mux = SOC_DAPM_ENUM("PCM Mux", aif1_enum); @@ -572,8 +572,8 @@ static const char *i2s_din_text[] = { "DIN", "DIN1", }; -static const struct soc_enum i2s_din_enum = - SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text); +static SOC_ENUM_SINGLE_DECL(i2s_din_enum, + PM860X_I2S_IFACE_3, 1, i2s_din_text); static const struct snd_kcontrol_new i2s_din_mux = SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum); @@ -583,8 +583,8 @@ static const char *i2s_mic_text[] = { "Ex PA", "ADC", }; -static const struct soc_enum i2s_mic_enum = - SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text); +static SOC_ENUM_SINGLE_DECL(i2s_mic_enum, + PM860X_I2S_IFACE_3, 4, i2s_mic_text); static const struct snd_kcontrol_new i2s_mic_mux = SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum); @@ -594,8 +594,8 @@ static const char *adcl_text[] = { "ADCR", "ADCL", }; -static const struct soc_enum adcl_enum = - SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text); +static SOC_ENUM_SINGLE_DECL(adcl_enum, + PM860X_PCM_IFACE_3, 4, adcl_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM("ADC Left Mux", adcl_enum); @@ -605,8 +605,8 @@ static const char *adcr_text[] = { "ADCL", "ADCR", }; -static const struct soc_enum adcr_enum = - SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text); +static SOC_ENUM_SINGLE_DECL(adcr_enum, + PM860X_PCM_IFACE_3, 2, adcr_text); static const struct snd_kcontrol_new adcr_mux = SOC_DAPM_ENUM("ADC Right Mux", adcr_enum); @@ -616,8 +616,8 @@ static const char *adcr_ec_text[] = { "ADCR", "EC", }; -static const struct soc_enum adcr_ec_enum = - SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text); +static SOC_ENUM_SINGLE_DECL(adcr_ec_enum, + PM860X_ADC_EN_2, 3, adcr_ec_text); static const struct snd_kcontrol_new adcr_ec_mux = SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum); @@ -627,8 +627,8 @@ static const char *ec_text[] = { "Left", "Right", "Left + Right", }; -static const struct soc_enum ec_enum = - SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text); +static SOC_ENUM_SINGLE_DECL(ec_enum, + PM860X_EC_PATH, 1, ec_text); static const struct snd_kcontrol_new ec_mux = SOC_DAPM_ENUM("EC Mux", ec_enum); @@ -638,36 +638,36 @@ static const char *dac_text[] = { }; /* DAC Headset 1 Mux / Mux10 */ -static const struct soc_enum dac_hs1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_hs1_enum, + PM860X_ANA_INPUT_SEL_1, 0, dac_text); static const struct snd_kcontrol_new dac_hs1_mux = SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum); /* DAC Headset 2 Mux / Mux11 */ -static const struct soc_enum dac_hs2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_hs2_enum, + PM860X_ANA_INPUT_SEL_1, 2, dac_text); static const struct snd_kcontrol_new dac_hs2_mux = SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum); /* DAC Lineout 1 Mux / Mux12 */ -static const struct soc_enum dac_lo1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_lo1_enum, + PM860X_ANA_INPUT_SEL_1, 4, dac_text); static const struct snd_kcontrol_new dac_lo1_mux = SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum); /* DAC Lineout 2 Mux / Mux13 */ -static const struct soc_enum dac_lo2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_lo2_enum, + PM860X_ANA_INPUT_SEL_1, 6, dac_text); static const struct snd_kcontrol_new dac_lo2_mux = SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum); /* DAC Spearker Earphone Mux / Mux14 */ -static const struct soc_enum dac_spk_ear_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_spk_ear_enum, + PM860X_ANA_INPUT_SEL_2, 0, dac_text); static const struct snd_kcontrol_new dac_spk_ear_mux = SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum); @@ -677,29 +677,29 @@ static const char *in_text[] = { "Digital", "Analog", }; -static const struct soc_enum hs1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text); +static SOC_ENUM_SINGLE_DECL(hs1_enum, + PM860X_ANA_TO_ANA, 0, in_text); static const struct snd_kcontrol_new hs1_mux = SOC_DAPM_ENUM("Headset1 Mux", hs1_enum); /* Headset 2 Mux / Mux16 */ -static const struct soc_enum hs2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text); +static SOC_ENUM_SINGLE_DECL(hs2_enum, + PM860X_ANA_TO_ANA, 1, in_text); static const struct snd_kcontrol_new hs2_mux = SOC_DAPM_ENUM("Headset2 Mux", hs2_enum); /* Lineout 1 Mux / Mux17 */ -static const struct soc_enum lo1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text); +static SOC_ENUM_SINGLE_DECL(lo1_enum, + PM860X_ANA_TO_ANA, 2, in_text); static const struct snd_kcontrol_new lo1_mux = SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum); /* Lineout 2 Mux / Mux18 */ -static const struct soc_enum lo2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text); +static SOC_ENUM_SINGLE_DECL(lo2_enum, + PM860X_ANA_TO_ANA, 3, in_text); static const struct snd_kcontrol_new lo2_mux = SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum); @@ -709,8 +709,8 @@ static const char *spk_text[] = { "Earpiece", "Speaker", }; -static const struct soc_enum spk_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text); +static SOC_ENUM_SINGLE_DECL(spk_enum, + PM860X_ANA_TO_ANA, 6, spk_text); static const struct snd_kcontrol_new spk_demux = SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum); @@ -720,8 +720,8 @@ static const char *mic_text[] = { "Mic 1", "Mic 2", }; -static const struct soc_enum mic_enum = - SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text); +static SOC_ENUM_SINGLE_DECL(mic_enum, + PM860X_ADC_ANA_4, 4, mic_text); static const struct snd_kcontrol_new mic_mux = SOC_DAPM_ENUM("MIC Mux", mic_enum); @@ -1327,7 +1327,9 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; - codec->control_data = pm860x->regmap; + ret = snd_soc_codec_set_cache_io(codec, pm860x->regmap); + if (ret) + return ret; for (i = 0; i < 4; i++) { ret = request_threaded_irq(pm860x->irq[i], NULL, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..f0e840137887 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -8,6 +8,8 @@ config SND_SOC_I2C_AND_SPI default y if I2C=y default y if SPI_MASTER=y +menu "CODEC drivers" + config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST @@ -16,15 +18,20 @@ config SND_SOC_ALL_CODECS select SND_SOC_AB8500_CODEC if ABX500_CORE select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER - select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI + select SND_SOC_AD193X_SPI if SPI_MASTER + select SND_SOC_AD193X_I2C if I2C select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAV801 if SPI_MASTER + select SND_SOC_ADAV803 if I2C + select SND_SOC_ADAU1977_SPI if SPI_MASTER + select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4554 select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C @@ -37,6 +44,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -59,20 +67,26 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 + select SND_SOC_PCM512x_I2C if I2C + select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2518 if I2C - select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_SSM2602_SPI if SPI_MASTER + select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C - select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC23_I2C if I2C + select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C @@ -182,6 +196,14 @@ config SND_SOC_AD1836 config SND_SOC_AD193X tristate +config SND_SOC_AD193X_SPI + tristate + select SND_SOC_AD193X + +config SND_SOC_AD193X_I2C + tristate + select SND_SOC_AD193X + config SND_SOC_AD1980 tristate @@ -189,41 +211,66 @@ config SND_SOC_AD73311 tristate config SND_SOC_ADAU1701 + tristate "Analog Devices ADAU1701 CODEC" + depends on I2C select SND_SOC_SIGMADSP - tristate config SND_SOC_ADAU1373 tristate +config SND_SOC_ADAU1977 + tristate + +config SND_SOC_ADAU1977_SPI + tristate + select SND_SOC_ADAU1977 + select REGMAP_SPI + +config SND_SOC_ADAU1977_I2C + tristate + select SND_SOC_ADAU1977 + select REGMAP_I2C + config SND_SOC_ADAV80X tristate +config SND_SOC_ADAV801 + tristate + select SND_SOC_ADAV80X + +config SND_SOC_ADAV803 + tristate + select SND_SOC_ADAV80X + config SND_SOC_ADS117X tristate config SND_SOC_AK4104 - tristate + tristate "AKM AK4104 CODEC" + depends on SPI_MASTER config SND_SOC_AK4535 tristate config SND_SOC_AK4554 - tristate + tristate "AKM AK4554 CODEC" config SND_SOC_AK4641 tristate config SND_SOC_AK4642 - tristate + tristate "AKM AK4642 CODEC" + depends on I2C config SND_SOC_AK4671 tristate config SND_SOC_AK5386 - tristate + tristate "AKM AK5638 CODEC" config SND_SOC_ALC5623 tristate + config SND_SOC_ALC5632 tristate @@ -234,14 +281,17 @@ config SND_SOC_CS42L51 tristate config SND_SOC_CS42L52 - tristate + tristate "Cirrus Logic CS42L52 CODEC" + depends on I2C config SND_SOC_CS42L73 - tristate + tristate "Cirrus Logic CS42L73 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 - tristate + tristate "Cirrus Logic CS4270 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function @@ -252,8 +302,18 @@ config SND_SOC_CS4270_VD33_ERRATA depends on SND_SOC_CS4270 config SND_SOC_CS4271 + tristate "Cirrus Logic CS4271 CODEC" + depends on SND_SOC_I2C_AND_SPI + +config SND_SOC_CS42XX8 tristate +config SND_SOC_CS42XX8_I2C + tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42XX8 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY @@ -283,6 +343,9 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate "HDMI stub CODEC" + config SND_SOC_ISABELLE tristate @@ -301,18 +364,32 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_HDMI_CODEC - tristate - config SND_SOC_PCM1681 - tristate + tristate "Texas Instruments PCM1681 CODEC" + depends on I2C config SND_SOC_PCM1792A - tristate + tristate "Texas Instruments PCM1792A CODEC" + depends on SPI_MASTER config SND_SOC_PCM3008 tristate +config SND_SOC_PCM512x + tristate + +config SND_SOC_PCM512x_I2C + tristate "Texas Instruments PCM512x CODECs - I2C" + depends on I2C + select SND_SOC_PCM512x + select REGMAP_I2C + +config SND_SOC_PCM512x_SPI + tristate "Texas Instruments PCM512x CODECs - SPI" + depends on SPI_MASTER + select SND_SOC_PCM512x + select REGMAP_SPI + config SND_SOC_RT5631 tristate @@ -321,7 +398,8 @@ config SND_SOC_RT5640 #Freescale sgtl5000 codec config SND_SOC_SGTL5000 - tristate + tristate "Freescale SGTL5000 CODEC" + depends on I2C config SND_SOC_SI476X tristate @@ -330,11 +408,15 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIRF_AUDIO_CODEC + tristate "SiRF SoC internal audio codec" + select REGMAP_MMIO + config SND_SOC_SN95031 tristate config SND_SOC_SPDIF - tristate + tristate "S/PDIF CODEC" config SND_SOC_SSM2518 tristate @@ -342,6 +424,14 @@ config SND_SOC_SSM2518 config SND_SOC_SSM2602 tristate +config SND_SOC_SSM2602_SPI + select SND_SOC_SSM2602 + tristate + +config SND_SOC_SSM2602_I2C + select SND_SOC_SSM2602 + tristate + config SND_SOC_STA32X tristate @@ -352,20 +442,33 @@ config SND_SOC_STAC9766 tristate config SND_SOC_TAS5086 - tristate + tristate "Texas Instruments TAS5086 speaker amplifier" + depends on I2C config SND_SOC_TLV320AIC23 tristate +config SND_SOC_TLV320AIC23_I2C + tristate + select SND_SOC_TLV320AIC23 + +config SND_SOC_TLV320AIC23_SPI + tristate + select SND_SOC_TLV320AIC23 + config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X - tristate + tristate "Texas Instruments TLV320AIC3x CODECs" + depends on I2C config SND_SOC_TLV320DAC33 tristate @@ -414,55 +517,69 @@ config SND_SOC_WM8400 tristate config SND_SOC_WM8510 - tristate + tristate "Wolfson Microelectronics WM8510 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8523 - tristate + tristate "Wolfson Microelectronics WM8523 DAC" + depends on I2C config SND_SOC_WM8580 - tristate + tristate "Wolfson Microelectronics WM8523 CODEC" + depends on I2C config SND_SOC_WM8711 - tristate + tristate "Wolfson Microelectronics WM8711 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8727 tristate config SND_SOC_WM8728 - tristate + tristate "Wolfson Microelectronics WM8728 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8731 - tristate + tristate "Wolfson Microelectronics WM8731 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8737 - tristate + tristate "Wolfson Microelectronics WM8737 ADC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate + tristate "Wolfson Microelectronics WM8737 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 - tristate + tristate "Wolfson Microelectronics WM8750 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8753 - tristate + tristate "Wolfson Microelectronics WM8753 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8770 - tristate + tristate "Wolfson Microelectronics WM8770 CODEC" + depends on SPI_MASTER config SND_SOC_WM8776 - tristate + tristate "Wolfson Microelectronics WM8776 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8900 tristate config SND_SOC_WM8903 - tristate + tristate "Wolfson Microelectronics WM8903 CODEC" + depends on I2C config SND_SOC_WM8904 tristate @@ -480,7 +597,8 @@ config SND_SOC_WM8961 tristate config SND_SOC_WM8962 - tristate + tristate "Wolfson Microelectronics WM8962 CODEC" + depends on I2C config SND_SOC_WM8971 tristate @@ -553,4 +671,7 @@ config SND_SOC_ML26124 tristate config SND_SOC_TPA6130A2 - tristate + tristate "Texas Instruments TPA6130A2 headphone amplifier" + depends on I2C + +endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..3c4d275d064b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,11 +3,18 @@ snd-soc-ab8500-codec-objs := ab8500-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o +snd-soc-ad193x-spi-objs := ad193x-spi.o +snd-soc-ad193x-i2c-objs := ad193x-i2c.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau1977-objs := adau1977.o +snd-soc-adau1977-spi-objs := adau1977-spi.o +snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adav80x-objs := adav80x.o +snd-soc-adav801-objs := adav801.o +snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -23,6 +30,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42xx8-objs := cs42xx8.o +snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -46,6 +55,9 @@ snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm512x-objs := pcm512x.o +snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o +snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -53,19 +65,25 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-si476x-objs := si476x.o +snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-ssm2602-spi-objs := ssm2602-spi.o +snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o +snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o +snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o snd-soc-tlv320aic26-objs := tlv320aic26.o -snd-soc-tlv320aic3x-objs := tlv320aic3x.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o +snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -134,11 +152,18 @@ obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o +obj-$(CONFIG_SND_SOC_AD193X_SPI) += snd-soc-ad193x-spi.o +obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o +obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o +obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o +obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o +obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o +obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -156,6 +181,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o +obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o @@ -179,6 +206,9 @@ obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o +obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o +obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o @@ -188,14 +218,19 @@ obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o +obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o +obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o +obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o -obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 77f459868579..685998dd086e 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -40,8 +40,8 @@ struct ad1836_priv { */ static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; -static const struct soc_enum ad1836_deemp_enum = - SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); +static SOC_ENUM_SINGLE_DECL(ad1836_deemp_enum, + AD1836_DAC_CTRL1, 8, ad1836_deemp); #define AD1836_DAC_VOLUME(x) \ SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \ diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c new file mode 100644 index 000000000000..df3a1a415825 --- /dev/null +++ b/sound/soc/codecs/ad193x-i2c.c @@ -0,0 +1,54 @@ +/* + * AD1936/AD1937 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "ad193x.h" + +static const struct i2c_device_id ad193x_id[] = { + { "ad1936", 0 }, + { "ad1937", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ad193x_id); + +static int ad193x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = ad193x_regmap_config; + config.val_bits = 8; + config.reg_bits = 8; + + return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); +} + +static int ad193x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ad193x_i2c_driver = { + .driver = { + .name = "ad193x", + }, + .probe = ad193x_i2c_probe, + .remove = ad193x_i2c_remove, + .id_table = ad193x_id, +}; +module_i2c_driver(ad193x_i2c_driver); + +MODULE_DESCRIPTION("ASoC AD1936/AD1937 audio CODEC driver"); +MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c new file mode 100644 index 000000000000..390cef9b9dc2 --- /dev/null +++ b/sound/soc/codecs/ad193x-spi.c @@ -0,0 +1,48 @@ +/* + * AD1938/AD1939 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "ad193x.h" + +static int ad193x_spi_probe(struct spi_device *spi) +{ + struct regmap_config config; + + config = ad193x_regmap_config; + config.val_bits = 8; + config.reg_bits = 16; + config.read_flag_mask = 0x09; + config.write_flag_mask = 0x08; + + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); +} + +static int ad193x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver ad193x_spi_driver = { + .driver = { + .name = "ad193x", + .owner = THIS_MODULE, + }, + .probe = ad193x_spi_probe, + .remove = ad193x_spi_remove, +}; +module_spi_driver(ad193x_spi_driver); + +MODULE_DESCRIPTION("ASoC AD1938/AD1939 audio CODEC driver"); +MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 5a42dca535b7..6844d0b2af68 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -6,12 +6,10 @@ * Licensed under the GPL-2 or later. */ -#include <linux/init.h> #include <linux/module.h> #include <linux/kernel.h> #include <linux/device.h> -#include <linux/i2c.h> -#include <linux/spi/spi.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -19,6 +17,7 @@ #include <sound/initval.h> #include <sound/soc.h> #include <sound/tlv.h> + #include "ad193x.h" /* codec private data */ @@ -32,8 +31,8 @@ struct ad193x_priv { */ static const char * const ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; -static const struct soc_enum ad193x_deemp_enum = - SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp); +static SOC_ENUM_SINGLE_DECL(ad193x_deemp_enum, AD193X_DAC_CTRL2, 1, + ad193x_deemp); static const DECLARE_TLV_DB_MINMAX(adau193x_tlv, -9563, 0); @@ -320,17 +319,9 @@ static struct snd_soc_dai_driver ad193x_dai = { .ops = &ad193x_dai_ops, }; -static int ad193x_probe(struct snd_soc_codec *codec) +static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ad193x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } /* default setting for ad193x */ @@ -348,11 +339,11 @@ static int ad193x_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); - return ret; + return 0; } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { - .probe = ad193x_probe, + .probe = ad193x_codec_probe, .controls = ad193x_snd_controls, .num_controls = ARRAY_SIZE(ad193x_snd_controls), .dapm_widgets = ad193x_dapm_widgets, @@ -366,140 +357,31 @@ static bool adau193x_reg_volatile(struct device *dev, unsigned int reg) return false; } -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config ad193x_spi_regmap_config = { - .val_bits = 8, - .reg_bits = 16, - .read_flag_mask = 0x09, - .write_flag_mask = 0x08, - +const struct regmap_config ad193x_regmap_config = { .max_register = AD193X_NUM_REGS - 1, .volatile_reg = adau193x_reg_volatile, }; +EXPORT_SYMBOL_GPL(ad193x_regmap_config); -static int ad193x_spi_probe(struct spi_device *spi) +int ad193x_probe(struct device *dev, struct regmap *regmap) { struct ad193x_priv *ad193x; - ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv), - GFP_KERNEL); - if (ad193x == NULL) - return -ENOMEM; - - ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config); - if (IS_ERR(ad193x->regmap)) - return PTR_ERR(ad193x->regmap); - - spi_set_drvdata(spi, ad193x); - - return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, - &ad193x_dai, 1); -} - -static int ad193x_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver ad193x_spi_driver = { - .driver = { - .name = "ad193x", - .owner = THIS_MODULE, - }, - .probe = ad193x_spi_probe, - .remove = ad193x_spi_remove, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) - -static const struct regmap_config ad193x_i2c_regmap_config = { - .val_bits = 8, - .reg_bits = 8, - - .max_register = AD193X_NUM_REGS - 1, - .volatile_reg = adau193x_reg_volatile, -}; - -static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, ad193x_id); - -static int ad193x_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - struct ad193x_priv *ad193x; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv), - GFP_KERNEL); + ad193x = devm_kzalloc(dev, sizeof(*ad193x), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config); - if (IS_ERR(ad193x->regmap)) - return PTR_ERR(ad193x->regmap); - - i2c_set_clientdata(client, ad193x); - - return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, - &ad193x_dai, 1); -} - -static int ad193x_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static struct i2c_driver ad193x_i2c_driver = { - .driver = { - .name = "ad193x", - }, - .probe = ad193x_i2c_probe, - .remove = ad193x_i2c_remove, - .id_table = ad193x_id, -}; -#endif - -static int __init ad193x_modinit(void) -{ - int ret; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ad193x_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n", - ret); - } -#endif - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&ad193x_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register AD193X SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(ad193x_modinit); + ad193x->regmap = regmap; -static void __exit ad193x_modexit(void) -{ -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&ad193x_spi_driver); -#endif + dev_set_drvdata(dev, ad193x); -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ad193x_i2c_driver); -#endif + return snd_soc_register_codec(dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } -module_exit(ad193x_modexit); +EXPORT_SYMBOL_GPL(ad193x_probe); MODULE_DESCRIPTION("ASoC ad193x driver"); MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 473388049992..ab9a998f15be 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -9,6 +9,13 @@ #ifndef __AD193X_H__ #define __AD193X_H__ +#include <linux/regmap.h> + +struct device; + +extern const struct regmap_config ad193x_regmap_config; +int ad193x_probe(struct device *dev, struct regmap *regmap); + #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 #define AD193X_PLL_INPUT_MASK 0x6 diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 7257a8885f42..34d965a4a040 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -57,8 +57,8 @@ static const u16 ad1980_reg[] = { static const char *ad1980_rec_sel[] = {"Mic", "CD", "NC", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; -static const struct soc_enum ad1980_cap_src = - SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 7, ad1980_rec_sel); +static SOC_ENUM_DOUBLE_DECL(ad1980_cap_src, + AC97_REC_SEL, 8, 0, ad1980_rec_sel); static const struct snd_kcontrol_new ad1980_snd_ac97_controls[] = { SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1), diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index eb836ed5271f..877f5737bb6b 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -345,15 +345,15 @@ static const char *adau1373_fdsp_sel_text[] = { "Channel 5", }; -static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text); static const char *adau1373_hpf_cutoff_text[] = { @@ -362,7 +362,7 @@ static const char *adau1373_hpf_cutoff_text[] = { "800Hz", }; -static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text); static const char *adau1373_bass_lpf_cutoff_text[] = { @@ -388,14 +388,14 @@ static const unsigned int adau1373_bass_tlv[] = { 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), }; -static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text); -static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, +static SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text, adau1373_bass_clip_level_values); -static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text); static const char *adau1373_3d_level_text[] = { @@ -409,9 +409,9 @@ static const char *adau1373_3d_cutoff_text[] = { "0.16875 fs", "0.27083 fs" }; -static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text); static const unsigned int adau1373_3d_tlv[] = { @@ -427,11 +427,11 @@ static const char *adau1373_lr_mux_text[] = { "Stereo", }; -static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text); static const struct snd_kcontrol_new adau1373_controls[] = { @@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = { "DMIC1", }; -static const struct soc_enum adau1373_decimator_enum = - SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adau1373_decimator_enum, + adau1373_decimator_text); static const struct snd_kcontrol_new adau1373_decimator_mux = SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); @@ -1376,15 +1376,8 @@ static int adau1373_probe(struct snd_soc_codec *codec) struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; - int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } - if (pdata) { if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) return -EINVAL; diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c new file mode 100644 index 000000000000..9700e8c838c9 --- /dev/null +++ b/sound/soc/codecs/adau1977-i2c.c @@ -0,0 +1,59 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "adau1977.h" + +static int adau1977_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 8; + + return adau1977_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + id->driver_data, NULL); +} + +static int adau1977_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id adau1977_i2c_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); + +static struct i2c_driver adau1977_i2c_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_i2c_probe, + .remove = adau1977_i2c_remove, + .id_table = adau1977_i2c_ids, +}; +module_i2c_driver(adau1977_i2c_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c new file mode 100644 index 000000000000..b05cf5da3a94 --- /dev/null +++ b/sound/soc/codecs/adau1977-spi.c @@ -0,0 +1,76 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include "adau1977.h" + +static void adau1977_spi_switch_mode(struct device *dev) +{ + struct spi_device *spi = to_spi_device(dev); + + /* + * To get the device into SPI mode CLATCH has to be pulled low three + * times. Do this by issuing three dummy reads. + */ + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); +} + +static int adau1977_spi_probe(struct spi_device *spi) +{ + const struct spi_device_id *id = spi_get_device_id(spi); + struct regmap_config config; + + if (!id) + return -EINVAL; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 16; + config.read_flag_mask = 0x1; + + return adau1977_probe(&spi->dev, + devm_regmap_init_spi(spi, &config), + id->driver_data, adau1977_spi_switch_mode); +} + +static int adau1977_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id adau1977_spi_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); + +static struct spi_driver adau1977_spi_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_spi_probe, + .remove = adau1977_spi_remove, + .id_table = adau1977_spi_ids, +}; +module_spi_driver(adau1977_spi_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c new file mode 100644 index 000000000000..fd55da7cb9d4 --- /dev/null +++ b/sound/soc/codecs/adau1977.c @@ -0,0 +1,1018 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2. + */ + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/gpio/consumer.h> +#include <linux/i2c.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_data/adau1977.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/slab.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "adau1977.h" + +#define ADAU1977_REG_POWER 0x00 +#define ADAU1977_REG_PLL 0x01 +#define ADAU1977_REG_BOOST 0x02 +#define ADAU1977_REG_MICBIAS 0x03 +#define ADAU1977_REG_BLOCK_POWER_SAI 0x04 +#define ADAU1977_REG_SAI_CTRL0 0x05 +#define ADAU1977_REG_SAI_CTRL1 0x06 +#define ADAU1977_REG_CMAP12 0x07 +#define ADAU1977_REG_CMAP34 0x08 +#define ADAU1977_REG_SAI_OVERTEMP 0x09 +#define ADAU1977_REG_POST_ADC_GAIN(x) (0x0a + (x)) +#define ADAU1977_REG_MISC_CONTROL 0x0e +#define ADAU1977_REG_DIAG_CONTROL 0x10 +#define ADAU1977_REG_STATUS(x) (0x11 + (x)) +#define ADAU1977_REG_DIAG_IRQ1 0x15 +#define ADAU1977_REG_DIAG_IRQ2 0x16 +#define ADAU1977_REG_ADJUST1 0x17 +#define ADAU1977_REG_ADJUST2 0x18 +#define ADAU1977_REG_ADC_CLIP 0x19 +#define ADAU1977_REG_DC_HPF_CAL 0x1a + +#define ADAU1977_POWER_RESET BIT(7) +#define ADAU1977_POWER_PWUP BIT(0) + +#define ADAU1977_PLL_CLK_S BIT(4) +#define ADAU1977_PLL_MCS_MASK 0x7 + +#define ADAU1977_MICBIAS_MB_VOLTS_MASK 0xf0 +#define ADAU1977_MICBIAS_MB_VOLTS_OFFSET 4 + +#define ADAU1977_BLOCK_POWER_SAI_LR_POL BIT(7) +#define ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE BIT(6) +#define ADAU1977_BLOCK_POWER_SAI_LDO_EN BIT(5) + +#define ADAU1977_SAI_CTRL0_FMT_MASK (0x3 << 6) +#define ADAU1977_SAI_CTRL0_FMT_I2S (0x0 << 6) +#define ADAU1977_SAI_CTRL0_FMT_LJ (0x1 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_24BIT (0x2 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_16BIT (0x3 << 6) + +#define ADAU1977_SAI_CTRL0_SAI_MASK (0x7 << 3) +#define ADAU1977_SAI_CTRL0_SAI_I2S (0x0 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_2 (0x1 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_4 (0x2 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_8 (0x3 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_16 (0x4 << 3) + +#define ADAU1977_SAI_CTRL0_FS_MASK (0x7) +#define ADAU1977_SAI_CTRL0_FS_8000_12000 (0x0) +#define ADAU1977_SAI_CTRL0_FS_16000_24000 (0x1) +#define ADAU1977_SAI_CTRL0_FS_32000_48000 (0x2) +#define ADAU1977_SAI_CTRL0_FS_64000_96000 (0x3) +#define ADAU1977_SAI_CTRL0_FS_128000_192000 (0x4) + +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK (0x3 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_32 (0x0 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_24 (0x1 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_16 (0x2 << 5) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT (0x0 << 4) +#define ADAU1977_SAI_CTRL1_LRCLK_PULSE BIT(3) +#define ADAU1977_SAI_CTRL1_MSB BIT(2) +#define ADAU1977_SAI_CTRL1_BCLKRATE_16 (0x1 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_32 (0x0 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_MASK (0x1 << 1) +#define ADAU1977_SAI_CTRL1_MASTER BIT(0) + +#define ADAU1977_SAI_OVERTEMP_DRV_C(x) BIT(4 + (x)) +#define ADAU1977_SAI_OVERTEMP_DRV_HIZ BIT(3) + +#define ADAU1977_MISC_CONTROL_SUM_MODE_MASK (0x3 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_1CH (0x2 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_2CH (0x1 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_4CH (0x0 << 6) +#define ADAU1977_MISC_CONTROL_MMUTE BIT(4) +#define ADAU1977_MISC_CONTROL_DC_CAL BIT(0) + +#define ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET 4 +#define ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET 0 + +struct adau1977 { + struct regmap *regmap; + bool right_j; + unsigned int sysclk; + enum adau1977_sysclk_src sysclk_src; + struct gpio_desc *reset_gpio; + enum adau1977_type type; + + struct regulator *avdd_reg; + struct regulator *dvdd_reg; + + struct snd_pcm_hw_constraint_list constraints; + + struct device *dev; + void (*switch_mode)(struct device *dev); + + unsigned int max_master_fs; + unsigned int slot_width; + bool enabled; + bool master; +}; + +static const struct reg_default adau1977_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x41 }, + { 0x02, 0x4a }, + { 0x03, 0x7d }, + { 0x04, 0x3d }, + { 0x05, 0x02 }, + { 0x06, 0x00 }, + { 0x07, 0x10 }, + { 0x08, 0x32 }, + { 0x09, 0xf0 }, + { 0x0a, 0xa0 }, + { 0x0b, 0xa0 }, + { 0x0c, 0xa0 }, + { 0x0d, 0xa0 }, + { 0x0e, 0x02 }, + { 0x10, 0x0f }, + { 0x15, 0x20 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x1a, 0x00 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(adau1977_adc_gain, -3562, 6000); + +static const struct snd_soc_dapm_widget adau1977_micbias_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU1977_REG_MICBIAS, + 3, 0, NULL, 0) +}; + +static const struct snd_soc_dapm_widget adau1977_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("Vref", ADAU1977_REG_BLOCK_POWER_SAI, + 4, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC1", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 0, 0), + SND_SOC_DAPM_ADC("ADC2", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 1, 0), + SND_SOC_DAPM_ADC("ADC3", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 2, 0), + SND_SOC_DAPM_ADC("ADC4", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 3, 0), + + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("AIN3"), + SND_SOC_DAPM_INPUT("AIN4"), + + SND_SOC_DAPM_OUTPUT("VREF"), +}; + +static const struct snd_soc_dapm_route adau1977_dapm_routes[] = { + { "ADC1", NULL, "AIN1" }, + { "ADC2", NULL, "AIN2" }, + { "ADC3", NULL, "AIN3" }, + { "ADC4", NULL, "AIN4" }, + + { "ADC1", NULL, "Vref" }, + { "ADC2", NULL, "Vref" }, + { "ADC3", NULL, "Vref" }, + { "ADC4", NULL, "Vref" }, + + { "VREF", NULL, "Vref" }, +}; + +#define ADAU1977_VOLUME(x) \ + SOC_SINGLE_TLV("ADC" #x " Capture Volume", \ + ADAU1977_REG_POST_ADC_GAIN((x) - 1), \ + 0, 255, 1, adau1977_adc_gain) + +#define ADAU1977_HPF_SWITCH(x) \ + SOC_SINGLE("ADC" #x " Highpass-Filter Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0) + +#define ADAU1977_DC_SUB_SWITCH(x) \ + SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0) + +static const struct snd_kcontrol_new adau1977_snd_controls[] = { + ADAU1977_VOLUME(1), + ADAU1977_VOLUME(2), + ADAU1977_VOLUME(3), + ADAU1977_VOLUME(4), + + ADAU1977_HPF_SWITCH(1), + ADAU1977_HPF_SWITCH(2), + ADAU1977_HPF_SWITCH(3), + ADAU1977_HPF_SWITCH(4), + + ADAU1977_DC_SUB_SWITCH(1), + ADAU1977_DC_SUB_SWITCH(2), + ADAU1977_DC_SUB_SWITCH(3), + ADAU1977_DC_SUB_SWITCH(4), +}; + +static int adau1977_reset(struct adau1977 *adau1977) +{ + int ret; + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_RESET); + regcache_cache_bypass(adau1977->regmap, false); + if (ret) + return ret; + + return ret; +} + +/* + * Returns the appropriate setting for ths FS field in the CTRL0 register + * depending on the rate. + */ +static int adau1977_lookup_fs(unsigned int rate) +{ + if (rate >= 8000 && rate <= 12000) + return ADAU1977_SAI_CTRL0_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + return ADAU1977_SAI_CTRL0_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + return ADAU1977_SAI_CTRL0_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + return ADAU1977_SAI_CTRL0_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + return ADAU1977_SAI_CTRL0_FS_128000_192000; + else + return -EINVAL; +} + +static int adau1977_lookup_mcs(struct adau1977 *adau1977, unsigned int rate, + unsigned int fs) +{ + unsigned int mcs; + + /* + * rate = sysclk / (512 * mcs_lut[mcs]) * 2**fs + * => mcs_lut[mcs] = sysclk / (512 * rate) * 2**fs + * => mcs_lut[mcs] = sysclk / ((512 / 2**fs) * rate) + */ + + rate *= 512 >> fs; + + if (adau1977->sysclk % rate != 0) + return -EINVAL; + + mcs = adau1977->sysclk / rate; + + /* The factors configured by MCS are 1, 2, 3, 4, 6 */ + if (mcs < 1 || mcs > 6 || mcs == 5) + return -EINVAL; + + mcs = mcs - 1; + if (mcs == 5) + mcs = 4; + + return mcs; +} + +static int adau1977_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int slot_width; + unsigned int ctrl0, ctrl0_mask; + unsigned int ctrl1; + int mcs, fs; + int ret; + + fs = adau1977_lookup_fs(rate); + if (fs < 0) + return fs; + + if (adau1977->sysclk_src == ADAU1977_SYSCLK_SRC_MCLK) { + mcs = adau1977_lookup_mcs(adau1977, rate, fs); + if (mcs < 0) + return mcs; + } else { + mcs = 0; + } + + ctrl0_mask = ADAU1977_SAI_CTRL0_FS_MASK; + ctrl0 = fs; + + if (adau1977->right_j) { + switch (params_width(params)) { + case 16: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_16BIT; + break; + case 24: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl0_mask |= ADAU1977_SAI_CTRL0_FMT_MASK; + } + + if (adau1977->master) { + switch (params_width(params)) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT; + slot_width = 16; + break; + case 24: + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT; + slot_width = 32; + break; + default: + return -EINVAL; + } + + /* In TDM mode there is a fixed slot width */ + if (adau1977->slot_width) + slot_width = adau1977->slot_width; + + if (slot_width == 16) + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_16; + else + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_32; + + ret = regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK | + ADAU1977_SAI_CTRL1_BCLKRATE_MASK, + ctrl1); + if (ret < 0) + return ret; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ctrl0_mask, ctrl0); + if (ret < 0) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_MCS_MASK, mcs); +} + +static int adau1977_power_disable(struct adau1977 *adau1977) +{ + int ret = 0; + + if (!adau1977->enabled) + return 0; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, 0); + if (ret) + return ret; + + regcache_mark_dirty(adau1977->regmap); + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 0); + + regcache_cache_only(adau1977->regmap, true); + + regulator_disable(adau1977->avdd_reg); + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); + + adau1977->enabled = false; + + return 0; +} + +static int adau1977_power_enable(struct adau1977 *adau1977) +{ + unsigned int val; + int ret = 0; + + if (adau1977->enabled) + return 0; + + ret = regulator_enable(adau1977->avdd_reg); + if (ret) + return ret; + + if (adau1977->dvdd_reg) { + ret = regulator_enable(adau1977->dvdd_reg); + if (ret) + goto err_disable_avdd; + } + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 1); + + regcache_cache_only(adau1977->regmap, false); + + if (adau1977->switch_mode) + adau1977->switch_mode(adau1977->dev); + + ret = adau1977_reset(adau1977); + if (ret) + goto err_disable_dvdd; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, ADAU1977_POWER_PWUP); + if (ret) + goto err_disable_dvdd; + + ret = regcache_sync(adau1977->regmap); + if (ret) + goto err_disable_dvdd; + + /* + * The PLL register is not affected by the software reset. It is + * possible that the value of the register was changed to the + * default value while we were in cache only mode. In this case + * regcache_sync will skip over it and we have to manually sync + * it. + */ + ret = regmap_read(adau1977->regmap, ADAU1977_REG_PLL, &val); + if (ret) + goto err_disable_dvdd; + + if (val == 0x41) { + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_PLL, + 0x41); + if (ret) + goto err_disable_dvdd; + regcache_cache_bypass(adau1977->regmap, false); + } + + adau1977->enabled = true; + + return ret; + +err_disable_dvdd: + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); +err_disable_avdd: + regulator_disable(adau1977->avdd_reg); + return ret; +} + +static int adau1977_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = adau1977_power_enable(adau1977); + break; + case SND_SOC_BIAS_OFF: + ret = adau1977_power_disable(adau1977); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1, drv; + unsigned int slot[4]; + unsigned int i; + int ret; + + if (slots == 0) { + /* 0 = No fixed slot width */ + adau1977->slot_width = 0; + adau1977->max_master_fs = 192000; + return regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL0, ADAU1977_SAI_CTRL0_SAI_MASK, + ADAU1977_SAI_CTRL0_SAI_I2S); + } + + if (rx_mask == 0 || tx_mask != 0) + return -EINVAL; + + drv = 0; + for (i = 0; i < 4; i++) { + slot[i] = __ffs(rx_mask); + drv |= ADAU1977_SAI_OVERTEMP_DRV_C(i); + rx_mask &= ~(1 << slot[i]); + if (slot[i] >= slots) + return -EINVAL; + if (rx_mask == 0) + break; + } + + if (rx_mask != 0) + return -EINVAL; + + switch (width) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_16; + break; + case 24: + /* We can only generate 16 bit or 32 bit wide slots */ + if (adau1977->master) + return -EINVAL; + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_24; + break; + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 2: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_2; + break; + case 4: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_4; + break; + case 8: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_8; + break; + case 16: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_C(0) | + ADAU1977_SAI_OVERTEMP_DRV_C(1) | + ADAU1977_SAI_OVERTEMP_DRV_C(2) | + ADAU1977_SAI_OVERTEMP_DRV_C(3), drv); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP12, + (slot[1] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[0] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP34, + (slot[3] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[2] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_SAI_MASK, ctrl0); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK, ctrl1); + if (ret) + return ret; + + adau1977->slot_width = width; + + /* In master mode the maximum bitclock is 24.576 MHz */ + adau1977->max_master_fs = min(192000, 24576000 / width / slots); + + return 0; +} + +static int adau1977_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = ADAU1977_MISC_CONTROL_MMUTE; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MISC_CONTROL, + ADAU1977_MISC_CONTROL_MMUTE, val); +} + +static int adau1977_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0 = 0, ctrl1 = 0, block_power = 0; + bool invert_lrclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + adau1977->master = false; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ctrl1 |= ADAU1977_SAI_CTRL1_MASTER; + adau1977->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = true; + break; + default: + return -EINVAL; + } + + adau1977->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + adau1977->right_j = true; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = false; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) + block_power |= ADAU1977_BLOCK_POWER_SAI_LR_POL; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + ADAU1977_BLOCK_POWER_SAI_LR_POL | + ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE, block_power); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_FMT_MASK, + ctrl0); + if (ret) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_MASTER | ADAU1977_SAI_CTRL1_LRCLK_PULSE, + ctrl1); +} + +static int adau1977_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + u64 formats = 0; + + if (adau1977->slot_width == 16) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE; + else if (adau1977->right_j || adau1977->slot_width == 24) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &adau1977->constraints); + + if (adau1977->master) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 8000, adau1977->max_master_fs); + + if (formats != 0) + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, formats); + + return 0; +} + +static int adau1977_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (tristate) + val = ADAU1977_SAI_OVERTEMP_DRV_HIZ; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_HIZ, val); +} + +static const struct snd_soc_dai_ops adau1977_dai_ops = { + .startup = adau1977_startup, + .hw_params = adau1977_hw_params, + .mute_stream = adau1977_mute, + .set_fmt = adau1977_set_dai_fmt, + .set_tdm_slot = adau1977_set_tdm_slot, + .set_tristate = adau1977_set_tristate, +}; + +static struct snd_soc_dai_driver adau1977_dai = { + .name = "adau1977-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, + .ops = &adau1977_dai_ops, +}; + +static const unsigned int adau1977_rates[] = { + 8000, 16000, 32000, 64000, 128000, + 11025, 22050, 44100, 88200, 172400, + 12000, 24000, 48000, 96000, 192000, +}; + +#define ADAU1977_RATE_CONSTRAINT_MASK_32000 0x001f +#define ADAU1977_RATE_CONSTRAINT_MASK_44100 0x03e0 +#define ADAU1977_RATE_CONSTRAINT_MASK_48000 0x7c00 +/* All rates >= 32000 */ +#define ADAU1977_RATE_CONSTRAINT_MASK_LRCLK 0x739c + +static bool adau1977_check_sysclk(unsigned int mclk, unsigned int base_freq) +{ + unsigned int mcs; + + if (mclk % (base_freq * 128) != 0) + return false; + + mcs = mclk / (128 * base_freq); + if (mcs < 1 || mcs > 6 || mcs == 5) + return false; + + return true; +} + +static int adau1977_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = 0; + unsigned int clk_src; + unsigned int ret; + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + if (clk_id != ADAU1977_SYSCLK) + return -EINVAL; + + switch (source) { + case ADAU1977_SYSCLK_SRC_MCLK: + clk_src = 0; + break; + case ADAU1977_SYSCLK_SRC_LRCLK: + clk_src = ADAU1977_PLL_CLK_S; + break; + default: + return -EINVAL; + } + + if (freq != 0 && source == ADAU1977_SYSCLK_SRC_MCLK) { + if (freq < 4000000 || freq > 36864000) + return -EINVAL; + + if (adau1977_check_sysclk(freq, 32000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_32000; + if (adau1977_check_sysclk(freq, 44100)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_44100; + if (adau1977_check_sysclk(freq, 48000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_48000; + + if (mask == 0) + return -EINVAL; + } else if (source == ADAU1977_SYSCLK_SRC_LRCLK) { + mask = ADAU1977_RATE_CONSTRAINT_MASK_LRCLK; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_CLK_S, clk_src); + if (ret) + return ret; + + adau1977->constraints.mask = mask; + adau1977->sysclk_src = source; + adau1977->sysclk = freq; + + return 0; +} + +static int adau1977_codec_probe(struct snd_soc_codec *codec) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (adau1977->type) { + case ADAU1977: + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau1977_micbias_dapm_widgets, + ARRAY_SIZE(adau1977_micbias_dapm_widgets)); + if (ret < 0) + return ret; + break; + default: + break; + } + + return 0; +} + +static struct snd_soc_codec_driver adau1977_codec_driver = { + .probe = adau1977_codec_probe, + .set_bias_level = adau1977_set_bias_level, + .set_sysclk = adau1977_set_sysclk, + .idle_bias_off = true, + + .controls = adau1977_snd_controls, + .num_controls = ARRAY_SIZE(adau1977_snd_controls), + .dapm_widgets = adau1977_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1977_dapm_widgets), + .dapm_routes = adau1977_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes), +}; + +static int adau1977_setup_micbias(struct adau1977 *adau1977) +{ + struct adau1977_platform_data *pdata = adau1977->dev->platform_data; + unsigned int micbias; + + if (pdata) { + micbias = pdata->micbias; + if (micbias > ADAU1977_MICBIAS_9V0) + return -EINVAL; + + } else { + micbias = ADAU1977_MICBIAS_8V5; + } + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MICBIAS, + ADAU1977_MICBIAS_MB_VOLTS_MASK, + micbias << ADAU1977_MICBIAS_MB_VOLTS_OFFSET); +} + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)) +{ + unsigned int power_off_mask; + struct adau1977 *adau1977; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau1977 = devm_kzalloc(dev, sizeof(*adau1977), GFP_KERNEL); + if (adau1977 == NULL) + return -ENOMEM; + + adau1977->dev = dev; + adau1977->type = type; + adau1977->regmap = regmap; + adau1977->switch_mode = switch_mode; + adau1977->max_master_fs = 192000; + + adau1977->constraints.list = adau1977_rates; + adau1977->constraints.count = ARRAY_SIZE(adau1977_rates); + + adau1977->avdd_reg = devm_regulator_get(dev, "AVDD"); + if (IS_ERR(adau1977->avdd_reg)) + return PTR_ERR(adau1977->avdd_reg); + + adau1977->dvdd_reg = devm_regulator_get_optional(dev, "DVDD"); + if (IS_ERR(adau1977->dvdd_reg)) { + if (PTR_ERR(adau1977->dvdd_reg) != -ENODEV) + return PTR_ERR(adau1977->dvdd_reg); + adau1977->dvdd_reg = NULL; + } + + adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); + if (IS_ERR(adau1977->reset_gpio)) { + ret = PTR_ERR(adau1977->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return PTR_ERR(adau1977->reset_gpio); + adau1977->reset_gpio = NULL; + } + + dev_set_drvdata(dev, adau1977); + + if (adau1977->reset_gpio) { + ret = gpiod_direction_output(adau1977->reset_gpio, 0); + if (ret) + return ret; + ndelay(100); + } + + ret = adau1977_power_enable(adau1977); + if (ret) + return ret; + + if (type == ADAU1977) { + ret = adau1977_setup_micbias(adau1977); + if (ret) + goto err_poweroff; + } + + if (adau1977->dvdd_reg) + power_off_mask = ~0; + else + power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + power_off_mask, 0x00); + if (ret) + goto err_poweroff; + + ret = adau1977_power_disable(adau1977); + if (ret) + return ret; + + return snd_soc_register_codec(dev, &adau1977_codec_driver, + &adau1977_dai, 1); + +err_poweroff: + adau1977_power_disable(adau1977); + return ret; + +} +EXPORT_SYMBOL_GPL(adau1977_probe); + +static bool adau1977_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1977_REG_STATUS(0): + case ADAU1977_REG_STATUS(1): + case ADAU1977_REG_STATUS(2): + case ADAU1977_REG_STATUS(3): + case ADAU1977_REG_ADC_CLIP: + return true; + } + + return false; +} + +const struct regmap_config adau1977_regmap_config = { + .max_register = ADAU1977_REG_DC_HPF_CAL, + .volatile_reg = adau1977_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1977_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1977_reg_defaults), +}; +EXPORT_SYMBOL_GPL(adau1977_regmap_config); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.h b/sound/soc/codecs/adau1977.h new file mode 100644 index 000000000000..95e714345a86 --- /dev/null +++ b/sound/soc/codecs/adau1977.h @@ -0,0 +1,37 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars@metafoo.de> + * + * Licensed under the GPL-2. + */ + +#ifndef __SOUND_SOC_CODECS_ADAU1977_H__ +#define __SOUND_SOC_CODECS_ADAU1977_H__ + +#include <linux/regmap.h> + +struct device; + +enum adau1977_type { + ADAU1977, + ADAU1978, + ADAU1979, +}; + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)); + +extern const struct regmap_config adau1977_regmap_config; + +enum adau1977_clk_id { + ADAU1977_SYSCLK, +}; + +enum adau1977_sysclk_src { + ADAU1977_SYSCLK_SRC_MCLK, + ADAU1977_SYSCLK_SRC_LRCLK, +}; + +#endif diff --git a/sound/soc/codecs/adav801.c b/sound/soc/codecs/adav801.c new file mode 100644 index 000000000000..790fce33ab10 --- /dev/null +++ b/sound/soc/codecs/adav801.c @@ -0,0 +1,53 @@ +/* + * ADAV801 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "adav80x.h" + +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + +static int adav80x_spi_probe(struct spi_device *spi) +{ + struct regmap_config config; + + config = adav80x_regmap_config; + config.read_flag_mask = 0x01; + + return adav80x_bus_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); +} + +static int adav80x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, +}; +module_spi_driver(adav80x_spi_driver); + +MODULE_DESCRIPTION("ASoC ADAV801 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li@analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c new file mode 100644 index 000000000000..66d9fce34e62 --- /dev/null +++ b/sound/soc/codecs/adav803.c @@ -0,0 +1,50 @@ +/* + * ADAV803 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "adav80x.h" + +static const struct i2c_device_id adav803_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav803_id); + +static int adav803_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_regmap_config)); +} + +static int adav803_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver adav803_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav803_probe, + .remove = adav803_remove, + .id_table = adav803_id, +}; +module_i2c_driver(adav803_driver); + +MODULE_DESCRIPTION("ASoC ADAV803 driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li@analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a7c461..5062e34ee8dc 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -8,17 +8,15 @@ * Licensed under the GPL-2 or later. */ -#include <linux/init.h> #include <linux/module.h> #include <linux/kernel.h> -#include <linux/i2c.h> -#include <linux/spi/spi.h> +#include <linux/regmap.h> #include <linux/slab.h> -#include <sound/core.h> + #include <sound/pcm.h> #include <sound/pcm_params.h> -#include <sound/tlv.h> #include <sound/soc.h> +#include <sound/tlv.h> #include "adav80x.h" @@ -541,6 +539,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; if (dir == SND_SOC_CLOCK_IN) { switch (clk_id) { @@ -573,7 +572,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, iclk_ctrl2); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync(dapm); } } else { unsigned int mask; @@ -600,17 +599,21 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, adav80x->sysclk_pd[clk_id] = false; } + snd_soc_dapm_mutex_lock(dapm); + if (adav80x->sysclk_pd[0]) - snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL1"); else - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL1"); if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2]) - snd_soc_dapm_disable_pin(&codec->dapm, "PLL2"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2"); else - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } return 0; @@ -722,7 +725,7 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - if (!codec->active || !adav80x->rate) + if (!snd_soc_codec_is_active(codec) || !adav80x->rate) return 0; return snd_pcm_hw_constraint_minmax(substream->runtime, @@ -735,7 +738,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - if (!codec->active) + if (!snd_soc_codec_is_active(codec)) adav80x->rate = 0; } @@ -798,15 +801,8 @@ static struct snd_soc_dai_driver adav80x_dais[] = { static int adav80x_probe(struct snd_soc_codec *codec) { - int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } - /* Force PLLs on for SYSCLK output */ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); @@ -864,39 +860,26 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) +int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; - int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); - adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + adav80x = devm_kzalloc(dev, sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; - dev_set_drvdata(dev, adav80x); adav80x->regmap = regmap; - ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + return snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); - if (ret) - kfree(adav80x); - - return ret; -} - -static int adav80x_bus_remove(struct device *dev) -{ - snd_soc_unregister_codec(dev); - kfree(dev_get_drvdata(dev)); - return 0; } +EXPORT_SYMBOL_GPL(adav80x_bus_probe); -#if defined(CONFIG_SPI_MASTER) -static const struct regmap_config adav80x_spi_regmap_config = { +const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, @@ -908,105 +891,7 @@ static const struct regmap_config adav80x_spi_regmap_config = { .reg_defaults = adav80x_reg_defaults, .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), }; - -static const struct spi_device_id adav80x_spi_id[] = { - { "adav801", 0 }, - { } -}; -MODULE_DEVICE_TABLE(spi, adav80x_spi_id); - -static int adav80x_spi_probe(struct spi_device *spi) -{ - return adav80x_bus_probe(&spi->dev, - devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); -} - -static int adav80x_spi_remove(struct spi_device *spi) -{ - return adav80x_bus_remove(&spi->dev); -} - -static struct spi_driver adav80x_spi_driver = { - .driver = { - .name = "adav801", - .owner = THIS_MODULE, - }, - .probe = adav80x_spi_probe, - .remove = adav80x_spi_remove, - .id_table = adav80x_spi_id, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -static const struct regmap_config adav80x_i2c_regmap_config = { - .val_bits = 8, - .pad_bits = 1, - .reg_bits = 7, - - .max_register = ADAV80X_PLL_OUTE, - - .cache_type = REGCACHE_RBTREE, - .reg_defaults = adav80x_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), -}; - -static const struct i2c_device_id adav80x_i2c_id[] = { - { "adav803", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); - -static int adav80x_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - return adav80x_bus_probe(&client->dev, - devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); -} - -static int adav80x_i2c_remove(struct i2c_client *client) -{ - return adav80x_bus_remove(&client->dev); -} - -static struct i2c_driver adav80x_i2c_driver = { - .driver = { - .name = "adav803", - .owner = THIS_MODULE, - }, - .probe = adav80x_i2c_probe, - .remove = adav80x_i2c_remove, - .id_table = adav80x_i2c_id, -}; -#endif - -static int __init adav80x_init(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&adav80x_i2c_driver); - if (ret) - return ret; -#endif - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&adav80x_spi_driver); -#endif - - return ret; -} -module_init(adav80x_init); - -static void __exit adav80x_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&adav80x_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&adav80x_spi_driver); -#endif -} -module_exit(adav80x_exit); +EXPORT_SYMBOL_GPL(adav80x_regmap_config); MODULE_DESCRIPTION("ASoC ADAV80x driver"); MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>"); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h index adb0fc76d4e3..8a1d7c09dca5 100644 --- a/sound/soc/codecs/adav80x.h +++ b/sound/soc/codecs/adav80x.h @@ -9,6 +9,13 @@ #ifndef _ADAV80X_H #define _ADAV80X_H +#include <linux/regmap.h> + +struct device; + +extern const struct regmap_config adav80x_regmap_config; +int adav80x_bus_probe(struct device *dev, struct regmap *regmap); + enum adav80x_pll_src { ADAV80X_PLL_SRC_XIN, ADAV80X_PLL_SRC_XTAL, diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b4819dcd4f4d..10adf25d4c14 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -174,8 +174,6 @@ static int ak4104_probe(struct snd_soc_codec *codec) struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ak4104->regmap; - /* set power-up and non-reset bits */ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 684fe910669f..30e297890fec 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -388,15 +388,6 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { - struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ak4535->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 94cbe508dd37..868c0e2da1ec 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -113,14 +113,14 @@ static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0); static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0); -static const struct soc_enum ak4641_mono_out_enum = - SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out); -static const struct soc_enum ak4641_hp_out_enum = - SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out); -static const struct soc_enum ak4641_mic_select_enum = - SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select); -static const struct soc_enum ak4641_mic_or_dac_enum = - SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac); +static SOC_ENUM_SINGLE_DECL(ak4641_mono_out_enum, + AK4641_SIG1, 6, ak4641_mono_out); +static SOC_ENUM_SINGLE_DECL(ak4641_hp_out_enum, + AK4641_MODE2, 2, ak4641_hp_out); +static SOC_ENUM_SINGLE_DECL(ak4641_mic_select_enum, + AK4641_MIC, 1, ak4641_mic_select); +static SOC_ENUM_SINGLE_DECL(ak4641_mic_or_dac_enum, + AK4641_BTIF, 4, ak4641_mic_or_dac); static const struct snd_kcontrol_new ak4641_snd_controls[] = { SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum), @@ -519,14 +519,6 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f646c6e90c6..92655cc189ae 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -465,14 +465,6 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 25bdf6ad4a54..998fa0c5a0b9 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -15,6 +15,7 @@ #include <linux/init.h> #include <linux/i2c.h> #include <linux/delay.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/soc.h> #include <sound/initval.h> @@ -23,104 +24,99 @@ #include "ak4671.h" -/* codec private data */ -struct ak4671_priv { - enum snd_soc_control_type control_type; -}; - /* ak4671 register cache & default register settings */ -static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { - 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ - 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ - 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ - 0x02, /* AK4671_FORMAT_SELECT (0x03) */ - 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ - 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ - 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ - 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ - 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ - 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ - 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ - 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ - 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ - 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ - 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ - 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ - 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ - 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ - 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ - 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ - 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ - 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ - 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ - 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ - 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ - 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ - 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ - 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ - 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ - 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ - 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ - 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ - 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ - 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ - 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ - 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ - 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ - 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ - 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ - 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ - 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ - 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ - 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ - 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ - 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ - 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ - 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ - 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ - 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ - 0x00, /* this register not used */ - 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ - 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ - 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ - 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ - 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ - 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ - 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ - 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ - 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ - 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ - 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ - 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ - 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ - 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ - 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ - 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ - 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ - 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ - 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ - 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ - 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ - 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ - 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ - 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ - 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ - 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ - 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ - 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ - 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ - 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ - 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ - 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ - 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ - 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ - 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ - 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ - 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ - 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ - 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ - 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ - 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +static const struct reg_default ak4671_reg_defaults[] = { + { 0x00, 0x00 }, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + { 0x01, 0xf6 }, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + { 0x02, 0x00 }, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + { 0x03, 0x02 }, /* AK4671_FORMAT_SELECT (0x03) */ + { 0x04, 0x00 }, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + { 0x05, 0x55 }, /* AK4671_MIC_AMP_GAIN (0x05) */ + { 0x06, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + { 0x07, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + { 0x08, 0xb5 }, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + { 0x09, 0x00 }, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + { 0x0a, 0x00 }, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + { 0x0b, 0x00 }, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + { 0x0c, 0x00 }, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + { 0x0d, 0x00 }, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + { 0x0e, 0x00 }, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + { 0x0f, 0x00 }, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + { 0x10, 0x00 }, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + { 0x11, 0x80 }, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + { 0x12, 0x91 }, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + { 0x13, 0x91 }, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + { 0x14, 0xe1 }, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + { 0x15, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + { 0x16, 0x00 }, /* AK4671_ALC_TIMER_SELECT (0x16) */ + { 0x17, 0x00 }, /* AK4671_ALC_MODE_CONTROL (0x17) */ + { 0x18, 0x02 }, /* AK4671_MODE_CONTROL1 (0x18) */ + { 0x19, 0x01 }, /* AK4671_MODE_CONTROL2 (0x19) */ + { 0x1a, 0x18 }, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + { 0x1b, 0x18 }, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + { 0x1c, 0x00 }, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + { 0x1d, 0x02 }, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + { 0x1e, 0x00 }, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + { 0x1f, 0x00 }, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + { 0x20, 0x00 }, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + { 0x21, 0x00 }, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + { 0x22, 0x00 }, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + { 0x23, 0x00 }, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + { 0x24, 0x00 }, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + { 0x25, 0x00 }, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + { 0x26, 0x00 }, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + { 0x27, 0x00 }, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + { 0x28, 0xa9 }, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + { 0x29, 0x1f }, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + { 0x2a, 0xad }, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + { 0x2b, 0x20 }, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + { 0x2c, 0x00 }, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + { 0x2d, 0x00 }, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + { 0x2e, 0x00 }, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + { 0x2f, 0x00 }, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + { 0x30, 0x00 }, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + + { 0x32, 0x00 }, /* AK4671_E1_COEFFICIENT0 (0x32) */ + { 0x33, 0x00 }, /* AK4671_E1_COEFFICIENT1 (0x33) */ + { 0x34, 0x00 }, /* AK4671_E1_COEFFICIENT2 (0x34) */ + { 0x35, 0x00 }, /* AK4671_E1_COEFFICIENT3 (0x35) */ + { 0x36, 0x00 }, /* AK4671_E1_COEFFICIENT4 (0x36) */ + { 0x37, 0x00 }, /* AK4671_E1_COEFFICIENT5 (0x37) */ + { 0x38, 0x00 }, /* AK4671_E2_COEFFICIENT0 (0x38) */ + { 0x39, 0x00 }, /* AK4671_E2_COEFFICIENT1 (0x39) */ + { 0x3a, 0x00 }, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + { 0x3b, 0x00 }, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + { 0x3c, 0x00 }, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + { 0x3d, 0x00 }, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + { 0x3e, 0x00 }, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + { 0x3f, 0x00 }, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + { 0x40, 0x00 }, /* AK4671_E3_COEFFICIENT2 (0x40) */ + { 0x41, 0x00 }, /* AK4671_E3_COEFFICIENT3 (0x41) */ + { 0x42, 0x00 }, /* AK4671_E3_COEFFICIENT4 (0x42) */ + { 0x43, 0x00 }, /* AK4671_E3_COEFFICIENT5 (0x43) */ + { 0x44, 0x00 }, /* AK4671_E4_COEFFICIENT0 (0x44) */ + { 0x45, 0x00 }, /* AK4671_E4_COEFFICIENT1 (0x45) */ + { 0x46, 0x00 }, /* AK4671_E4_COEFFICIENT2 (0x46) */ + { 0x47, 0x00 }, /* AK4671_E4_COEFFICIENT3 (0x47) */ + { 0x48, 0x00 }, /* AK4671_E4_COEFFICIENT4 (0x48) */ + { 0x49, 0x00 }, /* AK4671_E4_COEFFICIENT5 (0x49) */ + { 0x4a, 0x00 }, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + { 0x4b, 0x00 }, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + { 0x4c, 0x00 }, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + { 0x4d, 0x00 }, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + { 0x4e, 0x00 }, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + { 0x4f, 0x00 }, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + { 0x50, 0x88 }, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + { 0x51, 0x88 }, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + { 0x52, 0x08 }, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + { 0x53, 0x00 }, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + { 0x54, 0x00 }, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + { 0x55, 0x00 }, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + { 0x56, 0x18 }, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + { 0x57, 0x18 }, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + { 0x58, 0x00 }, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + { 0x59, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + { 0x5a, 0x00 }, /* AK4671_SAR_ADC_CONTROL (0x5a) */ }; /* @@ -241,19 +237,17 @@ static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { /* Input MUXs */ static const char *ak4671_lin_mux_texts[] = {"LIN1", "LIN2", "LIN3", "LIN4"}; -static const struct soc_enum ak4671_lin_mux_enum = - SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, - ARRAY_SIZE(ak4671_lin_mux_texts), - ak4671_lin_mux_texts); +static SOC_ENUM_SINGLE_DECL(ak4671_lin_mux_enum, + AK4671_MIC_SIGNAL_SELECT, 0, + ak4671_lin_mux_texts); static const struct snd_kcontrol_new ak4671_lin_mux_control = SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); static const char *ak4671_rin_mux_texts[] = {"RIN1", "RIN2", "RIN3", "RIN4"}; -static const struct soc_enum ak4671_rin_mux_enum = - SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, - ARRAY_SIZE(ak4671_rin_mux_texts), - ak4671_rin_mux_texts); +static SOC_ENUM_SINGLE_DECL(ak4671_rin_mux_enum, + AK4671_MIC_SIGNAL_SELECT, 2, + ak4671_rin_mux_texts); static const struct snd_kcontrol_new ak4671_rin_mux_control = SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); @@ -619,21 +613,7 @@ static struct snd_soc_dai_driver ak4671_dai = { static int ak4671_probe(struct snd_soc_codec *codec) { - struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec); - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - snd_soc_add_codec_controls(codec, ak4671_snd_controls, - ARRAY_SIZE(ak4671_snd_controls)); - - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return ret; + return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int ak4671_remove(struct snd_soc_codec *codec) @@ -646,28 +626,36 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .probe = ak4671_probe, .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, - .reg_cache_size = AK4671_CACHEREGNUM, - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4671_reg, + .controls = ak4671_snd_controls, + .num_controls = ARRAY_SIZE(ak4671_snd_controls), .dapm_widgets = ak4671_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4671_dapm_widgets), .dapm_routes = ak4671_intercon, .num_dapm_routes = ARRAY_SIZE(ak4671_intercon), }; +static const struct regmap_config ak4671_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4671_SAR_ADC_CONTROL, + .reg_defaults = ak4671_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4671_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int ak4671_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct ak4671_priv *ak4671; + struct regmap *regmap; int ret; - ak4671 = devm_kzalloc(&client->dev, sizeof(struct ak4671_priv), - GFP_KERNEL); - if (ak4671 == NULL) - return -ENOMEM; - - i2c_set_clientdata(client, ak4671); - ak4671->control_type = SND_SOC_I2C; + regmap = devm_regmap_init_i2c(client, &ak4671_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ak4671, &ak4671_dai, 1); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h index 61cb7ab7552c..394a34d3f50a 100644 --- a/sound/soc/codecs/ak4671.h +++ b/sound/soc/codecs/ak4671.h @@ -105,8 +105,6 @@ #define AK4671_DIGITAL_MIXING_CONTROL2 0x59 #define AK4671_SAR_ADC_CONTROL 0x5a -#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) - /* Bitfield Definitions */ /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d3036283482a..09f7e773bafb 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -21,6 +21,7 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -38,26 +39,13 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u8 id; unsigned int sysclk; - u16 reg_cache[ALC5623_VENDOR_ID2+2]; unsigned int add_ctrl; unsigned int jack_det_ctrl; }; -static void alc5623_fill_cache(struct snd_soc_codec *codec) -{ - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; - - /* not really efficient ... */ - codec->cache_bypass = 1; - for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = snd_soc_read(codec, i); - codec->cache_bypass = 0; -} - static inline int alc5623_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, ALC5623_RESET, 0); @@ -228,32 +216,37 @@ static const char *alc5623_aux_out_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; /* auxout output mux */ -static const struct soc_enum alc5623_aux_out_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 6, + alc5623_aux_out_input_sel); static const struct snd_kcontrol_new alc5623_auxout_mux_controls = SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); /* speaker output mux */ -static const struct soc_enum alc5623_spkout_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 10, + alc5623_spkout_input_sel); static const struct snd_kcontrol_new alc5623_spkout_mux_controls = SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); /* headphone left output mux */ -static const struct soc_enum alc5623_hpl_out_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 9, + alc5623_hpl_out_input_sel); static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); /* headphone right output mux */ -static const struct soc_enum alc5623_hpr_out_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 8, + alc5623_hpr_out_input_sel); static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); /* speaker output N select */ -static const struct soc_enum alc5623_spk_n_sour_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum, + ALC5623_OUTPUT_MIXER_CTRL, 14, + alc5623_spk_n_sour_sel); static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); @@ -338,8 +331,9 @@ SND_SOC_DAPM_VMID("Vmid"), }; static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; -static const struct soc_enum alc5623_amp_enum = - SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); +static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum, + ALC5623_OUTPUT_MIXER_CTRL, 13, + alc5623_amp_names); static const struct snd_kcontrol_new alc5623_amp_mux_controls = SOC_DAPM_ENUM("Route", alc5623_amp_enum); @@ -869,18 +863,28 @@ static struct snd_soc_dai_driver alc5623_dai = { static int alc5623_suspend(struct snd_soc_codec *codec) { + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(alc5623->regmap, true); + return 0; } static int alc5623_resume(struct snd_soc_codec *codec) { - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int ret; /* Sync reg_cache with the hardware */ - for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) - snd_soc_write(codec, i, cache[i]); + regcache_cache_only(alc5623->regmap, false); + ret = regcache_sync(alc5623->regmap); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync register cache: %d\n", + ret); + regcache_cache_only(alc5623->regmap, true); + return ret; + } alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -900,14 +904,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - alc5623_reset(codec); - alc5623_fill_cache(codec); /* power on device */ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -980,9 +977,15 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, - .reg_cache_size = ALC5623_VENDOR_ID2+2, - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, +}; + +static const struct regmap_config alc5623_regmap = { + .reg_bits = 8, + .val_bits = 16, + .reg_stride = 2, + + .max_register = ALC5623_VENDOR_ID2, + .cache_type = REGCACHE_RBTREE, }; /* @@ -996,19 +999,32 @@ static int alc5623_i2c_probe(struct i2c_client *client, { struct alc5623_platform_data *pdata; struct alc5623_priv *alc5623; - int ret, vid1, vid2; + unsigned int vid1, vid2; + int ret; - vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); - if (vid1 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); + if (alc5623 == NULL) + return -ENOMEM; + + alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap); + if (IS_ERR(alc5623->regmap)) { + ret = PTR_ERR(alc5623->regmap); + dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret); + return ret; + } + + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); + return ret; } vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); - vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); - if (vid2 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); + return ret; } if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { @@ -1021,11 +1037,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), - GFP_KERNEL); - if (alc5623 == NULL) - return -ENOMEM; - pdata = client->dev.platform_data; if (pdata) { alc5623->add_ctrl = pdata->add_ctrl; @@ -1048,7 +1059,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5623); - alc5623->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index fb001c56cf8d..ec071a6306ef 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -293,51 +293,59 @@ static const char * const alc5632_i2s_out_sel[] = { "ADC LR", "Voice Stereo Digital"}; /* auxout output mux */ -static const struct soc_enum alc5632_aux_out_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_aux_out_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 6, + alc5632_aux_out_input_sel); static const struct snd_kcontrol_new alc5632_auxout_mux_controls = SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); /* speaker output mux */ -static const struct soc_enum alc5632_spkout_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_spkout_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 10, + alc5632_spkout_input_sel); static const struct snd_kcontrol_new alc5632_spkout_mux_controls = SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); /* headphone left output mux */ -static const struct soc_enum alc5632_hpl_out_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_hpl_out_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 9, + alc5632_hpl_out_input_sel); static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); /* headphone right output mux */ -static const struct soc_enum alc5632_hpr_out_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_hpr_out_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 8, + alc5632_hpr_out_input_sel); static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); /* speaker output N select */ -static const struct soc_enum alc5632_spk_n_sour_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_spk_n_sour_enum, + ALC5632_OUTPUT_MIXER_CTRL, 14, + alc5632_spk_n_sour_sel); static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); /* speaker amplifier */ static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; -static const struct soc_enum alc5632_amp_enum = - SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static SOC_ENUM_SINGLE_DECL(alc5632_amp_enum, + ALC5632_OUTPUT_MIXER_CTRL, 13, + alc5632_amp_names); static const struct snd_kcontrol_new alc5632_amp_mux_controls = SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); /* ADC output select */ -static const struct soc_enum alc5632_adcr_func_enum = - SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_adcr_func_enum, + ALC5632_DAC_FUNC_SELECT, 5, + alc5632_adcr_func_sel); static const struct snd_kcontrol_new alc5632_adcr_func_controls = SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum); /* I2S out select */ -static const struct soc_enum alc5632_i2s_out_enum = - SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_i2s_out_enum, + ALC5632_I2S_OUT_CTL, 5, + alc5632_i2s_out_sel); static const struct snd_kcontrol_new alc5632_i2s_out_controls = SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum); @@ -1055,14 +1063,6 @@ static int alc5632_probe(struct snd_soc_codec *codec) struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = alc5632->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e4295fee8f13..29e198f57d4c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -53,6 +53,14 @@ #define ARIZONA_AIF_RX_ENABLES 0x1A #define ARIZONA_AIF_FORCE_WRITE 0x1B +#define ARIZONA_FLL_VCO_CORNER 141900000 +#define ARIZONA_FLL_MAX_FREF 13500000 +#define ARIZONA_FLL_MIN_FVCO 90000000 +#define ARIZONA_FLL_MAX_FRATIO 16 +#define ARIZONA_FLL_MAX_REFDIV 8 +#define ARIZONA_FLL_MIN_OUTDIV 2 +#define ARIZONA_FLL_MAX_OUTDIV 7 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -542,67 +550,76 @@ static const char *arizona_vol_ramp_text[] = { "15ms/6dB", "30ms/6dB", }; -const struct soc_enum arizona_in_vd_ramp = - SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, - ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_in_vd_ramp, + ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VD_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_in_vd_ramp); -const struct soc_enum arizona_in_vi_ramp = - SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, - ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_in_vi_ramp, + ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VI_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_in_vi_ramp); -const struct soc_enum arizona_out_vd_ramp = - SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, - ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_out_vd_ramp, + ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VD_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_out_vd_ramp); -const struct soc_enum arizona_out_vi_ramp = - SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, - ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_out_vi_ramp, + ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VI_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_out_vi_ramp); static const char *arizona_lhpf_mode_text[] = { "Low-pass", "High-pass" }; -const struct soc_enum arizona_lhpf1_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf1_mode, + ARIZONA_HPLPF1_1, + ARIZONA_LHPF1_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); -const struct soc_enum arizona_lhpf2_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf2_mode, + ARIZONA_HPLPF2_1, + ARIZONA_LHPF2_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); -const struct soc_enum arizona_lhpf3_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf3_mode, + ARIZONA_HPLPF3_1, + ARIZONA_LHPF3_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); -const struct soc_enum arizona_lhpf4_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf4_mode, + ARIZONA_HPLPF4_1, + ARIZONA_LHPF4_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); static const char *arizona_ng_hold_text[] = { "30ms", "120ms", "250ms", "500ms", }; -const struct soc_enum arizona_ng_hold = - SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, - 4, arizona_ng_hold_text); +SOC_ENUM_SINGLE_DECL(arizona_ng_hold, + ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_HOLD_SHIFT, + arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); static const char * const arizona_in_hpf_cut_text[] = { "2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz" }; -const struct soc_enum arizona_in_hpf_cut_enum = - SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT, - ARRAY_SIZE(arizona_in_hpf_cut_text), - arizona_in_hpf_cut_text); +SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, + ARIZONA_HPF_CONTROL, + ARIZONA_IN_HPF_CUT_SHIFT, + arizona_in_hpf_cut_text); EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { @@ -1377,74 +1394,147 @@ struct arizona_fll_cfg { int gain; }; -static int arizona_calc_fll(struct arizona_fll *fll, - struct arizona_fll_cfg *cfg, - unsigned int Fref, - unsigned int Fout) +static int arizona_validate_fll(struct arizona_fll *fll, + unsigned int Fref, + unsigned int Fout) { - unsigned int target, div, gcd_fll; - int i, ratio; + unsigned int Fvco_min; + + if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } - arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + Fvco_min = ARIZONA_FLL_MIN_FVCO * fll->vco_mult; + if (Fout * ARIZONA_FLL_MAX_OUTDIV < Fvco_min) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + + return 0; +} + +static int arizona_find_fratio(unsigned int Fref, int *fratio) +{ + int i; + + /* Find an appropriate FLL_FRATIO */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + if (fratio) + *fratio = fll_fratios[i].fratio; + return fll_fratios[i].ratio; + } + } + + return -EINVAL; +} + +static int arizona_calc_fratio(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int target, + unsigned int Fref, bool sync) +{ + int init_ratio, ratio; + int refdiv, div; - /* Fref must be <=13.5MHz */ + /* Fref must be <=13.5MHz, find initial refdiv */ div = 1; cfg->refdiv = 0; - while ((Fref / div) > 13500000) { + while (Fref > ARIZONA_FLL_MAX_FREF) { div *= 2; + Fref /= 2; cfg->refdiv++; - if (div > 8) { - arizona_fll_err(fll, - "Can't scale %dMHz in to <=13.5MHz\n", - Fref); + if (div > ARIZONA_FLL_MAX_REFDIV) return -EINVAL; + } + + /* Find an appropriate FLL_FRATIO */ + init_ratio = arizona_find_fratio(Fref, &cfg->fratio); + if (init_ratio < 0) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return init_ratio; + } + + switch (fll->arizona->type) { + case WM5110: + if (fll->arizona->rev < 3 || sync) + return init_ratio; + break; + default: + return init_ratio; + } + + cfg->fratio = init_ratio - 1; + + /* Adjust FRATIO/refdiv to avoid integer mode if possible */ + refdiv = cfg->refdiv; + + while (div <= ARIZONA_FLL_MAX_REFDIV) { + for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + ratio++) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + return ratio; + } } + + for (ratio = init_ratio - 1; ratio >= 0; ratio--) { + if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < + Fref) + break; + + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + return ratio; + } + } + + div *= 2; + Fref /= 2; + refdiv++; + init_ratio = arizona_find_fratio(Fref, NULL); } - /* Apply the division for our remaining calculations */ - Fref /= div; + arizona_fll_warn(fll, "Falling back to integer mode operation\n"); + return cfg->fratio + 1; +} + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, bool sync) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout); /* Fvco should be over the targt; don't check the upper bound */ - div = 1; - while (Fout * div < 90000000 * fll->vco_mult) { + div = ARIZONA_FLL_MIN_OUTDIV; + while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; - if (div > 7) { - arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", - Fout); + if (div > ARIZONA_FLL_MAX_OUTDIV) return -EINVAL; - } } - target = Fout * div / fll->vco_mult; + target = fll->fout * div / fll->vco_mult; cfg->outdiv = div; arizona_fll_dbg(fll, "Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ - for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { - if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { - cfg->fratio = fll_fratios[i].fratio; - ratio = fll_fratios[i].ratio; - break; - } - } - if (i == ARRAY_SIZE(fll_fratios)) { - arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", - Fref); - return -EINVAL; - } + /* Find an appropriate FLL_FRATIO and refdiv */ + ratio = arizona_calc_fratio(fll, cfg, target, Fref, sync); + if (ratio < 0) + return ratio; - for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { - if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { - cfg->gain = fll_gains[i].gain; - break; - } - } - if (i == ARRAY_SIZE(fll_gains)) { - arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", - Fref); - return -EINVAL; - } + /* Apply the division for our remaining calculations */ + Fref = Fref / (1 << cfg->refdiv); cfg->n = target / (ratio * Fref); @@ -1469,6 +1559,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->lambda >>= 1; } + for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { + if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { + cfg->gain = fll_gains[i].gain; + break; + } + } + if (i == ARRAY_SIZE(fll_gains)) { + arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", + Fref); + return -EINVAL; + } + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", @@ -1496,14 +1598,18 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); - if (sync) - regmap_update_bits_async(arizona->regmap, base + 0x7, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); - else - regmap_update_bits_async(arizona->regmap, base + 0x9, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + if (sync) { + regmap_update_bits(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + } else { + regmap_update_bits(arizona->regmap, base + 0x5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + } regmap_update_bits_async(arizona->regmap, base + 2, ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, @@ -1526,13 +1632,12 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } -static void arizona_enable_fll(struct arizona_fll *fll, - struct arizona_fll_cfg *ref, - struct arizona_fll_cfg *sync) +static void arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; int ret; bool use_sync = false; + struct arizona_fll_cfg cfg; /* * If we have both REFCLK and SYNCCLK then enable both, @@ -1540,23 +1645,21 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - regmap_update_bits_async(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + arizona_calc_fll(fll, &cfg, fll->ref_freq, false); - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_apply_fll(arizona, fll->base + 0x10, sync, + arizona_calc_fll(fll, &cfg, fll->sync_freq, true); + + arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - regmap_update_bits_async(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + arizona_calc_fll(fll, &cfg, fll->sync_freq, false); - arizona_apply_fll(arizona, fll->base, sync, + arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); regmap_update_bits_async(arizona->regmap, fll->base + 0x11, @@ -1618,32 +1721,22 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll_refclk(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona_fll_cfg ref, sync; int ret; if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout) { - if (Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; - } - - if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, - fll->fout); - if (ret != 0) - return ret; - } + if (fll->fout && Fref > 0) { + ret = arizona_validate_fll(fll, Fref, fll->fout); + if (ret != 0) + return ret; } fll->ref_src = source; fll->ref_freq = Fref; if (fll->fout && Fref > 0) { - arizona_enable_fll(fll, &ref, &sync); + arizona_enable_fll(fll); } return 0; @@ -1653,7 +1746,6 @@ EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona_fll_cfg ref, sync; int ret; if (fll->sync_src == source && @@ -1662,13 +1754,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source, if (Fout) { if (fll->ref_src >= 0) { - ret = arizona_calc_fll(fll, &ref, fll->ref_freq, - Fout); + ret = arizona_validate_fll(fll, fll->ref_freq, Fout); if (ret != 0) return ret; } - ret = arizona_calc_fll(fll, &sync, Fref, Fout); + ret = arizona_validate_fll(fll, Fref, Fout); if (ret != 0) return ret; } @@ -1678,7 +1769,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->fout = Fout; if (Fout) { - arizona_enable_fll(fll, &ref, &sync); + arizona_enable_fll(fll); } else { arizona_disable_fll(fll); } diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 43737a27d79c..1e25c7af853b 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -138,9 +138,8 @@ static int cq93vc_probe(struct snd_soc_codec *codec) struct davinci_vc *davinci_vc = codec->dev->platform_data; davinci_vc->cq93vc.codec = codec; - codec->control_data = davinci_vc->regmap; - snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, davinci_vc->regmap); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 83c835d9fd88..3920e6264948 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -506,15 +506,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will - * then do the I2C transactions itself. - */ - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - return ret; - } - /* Disable auto-mute. This feature appears to be buggy. In some * situations, auto-mute will not deactivate when it should, so we want * this feature disabled by default. An application (e.g. alsactl) can diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd93dc74..aef4965750c7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -159,7 +159,6 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) } struct cs4271_private { - /* SND_SOC_I2C or SND_SOC_SPI */ unsigned int mclk; bool master; bool deemph; @@ -540,14 +539,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; - int gpio_nreset = -EINVAL; bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { - gpio_nreset = of_get_named_gpio(codec->dev->of_node, - "reset-gpio", 0); - if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; @@ -559,27 +554,19 @@ static int cs4271_probe(struct snd_soc_codec *codec) #endif if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_nreset >= 0) - if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) - gpio_nreset = -EINVAL; - if (gpio_nreset >= 0) { + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ - gpio_direction_output(gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); udelay(1); - gpio_set_value(gpio_nreset, 1); + gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ udelay(1); } - cs4271->gpio_nreset = gpio_nreset; - ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); @@ -625,6 +612,36 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; +static int cs4271_common_probe(struct device *dev, + struct cs4271_private **c) +{ + struct cs4271_platform_data *cs4271plat = dev->platform_data; + struct cs4271_private *cs4271; + + cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); + if (!cs4271) + return -ENOMEM; + + if (of_match_device(cs4271_dt_ids, dev)) + cs4271->gpio_nreset = + of_get_named_gpio(dev->of_node, "reset-gpio", 0); + + if (cs4271plat) + cs4271->gpio_nreset = cs4271plat->gpio_nreset; + + if (gpio_is_valid(cs4271->gpio_nreset)) { + int ret; + + ret = devm_gpio_request(dev, cs4271->gpio_nreset, + "CS4271 Reset"); + if (ret < 0) + return ret; + } + + *c = cs4271; + return 0; +} + #if defined(CONFIG_SPI_MASTER) static const struct regmap_config cs4271_spi_regmap = { @@ -644,10 +661,11 @@ static const struct regmap_config cs4271_spi_regmap = { static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; + int ret; - cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); - if (!cs4271) - return -ENOMEM; + ret = cs4271_common_probe(&spi->dev, &cs4271); + if (ret < 0) + return ret; spi_set_drvdata(spi, cs4271); cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); @@ -698,10 +716,11 @@ static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct cs4271_private *cs4271; + int ret; - cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); - if (!cs4271) - return -ENOMEM; + ret = cs4271_common_probe(&client->dev, &cs4271); + if (ret < 0) + return ret; i2c_set_clientdata(client, cs4271); cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..6c0da2baa154 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -30,6 +30,7 @@ #include <sound/pcm_params.h> #include <sound/pcm.h> #include <linux/i2c.h> +#include <linux/regmap.h> #include "cs42l51.h" @@ -40,7 +41,6 @@ enum master_slave_mode { }; struct cs42l51_private { - enum snd_soc_control_type control_type; unsigned int mclk; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; @@ -52,24 +52,6 @@ struct cs42l51_private { SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -static int cs42l51_fill_cache(struct snd_soc_codec *codec) -{ - u8 *cache = codec->reg_cache + 1; - struct i2c_client *i2c_client = to_i2c_client(codec->dev); - s32 length; - - length = i2c_smbus_read_i2c_block_data(i2c_client, - CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache); - if (length != CS42L51_NUMREGS) { - dev_err(&i2c_client->dev, - "I2C read failure, addr=0x%x (ret=%d vs %d)\n", - i2c_client->addr, length, CS42L51_NUMREGS); - return -EIO; - } - - return 0; -} - static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -124,9 +106,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); -/* This is a lie. after -102 db, it stays at -102 */ -/* maybe a range would be better */ -static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); static const char *chan_mix[] = { @@ -135,13 +116,12 @@ static const char *chan_mix[] = { "R L", }; -static const struct soc_enum cs42l51_chan_mix = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix); +static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, chan_mix); static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", @@ -149,7 +129,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), @@ -192,22 +172,22 @@ static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, static const char *cs42l51_dac_names[] = {"Direct PCM", "DSP PCM", "ADC"}; -static const struct soc_enum cs42l51_dac_mux_enum = - SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names); +static SOC_ENUM_SINGLE_DECL(cs42l51_dac_mux_enum, + CS42L51_DAC_CTL, 6, cs42l51_dac_names); static const struct snd_kcontrol_new cs42l51_dac_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum); static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left", "MIC Left", "MIC+preamp Left"}; -static const struct soc_enum cs42l51_adcl_mux_enum = - SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names); +static SOC_ENUM_SINGLE_DECL(cs42l51_adcl_mux_enum, + CS42L51_ADC_INPUT, 4, cs42l51_adcl_names); static const struct snd_kcontrol_new cs42l51_adcl_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum); static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right", "MIC Right", "MIC+preamp Right"}; -static const struct soc_enum cs42l51_adcr_mux_enum = - SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names); +static SOC_ENUM_SINGLE_DECL(cs42l51_adcr_mux_enum, + CS42L51_ADC_INPUT, 6, cs42l51_adcr_names); static const struct snd_kcontrol_new cs42l51_adcr_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum); @@ -505,21 +485,8 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { - struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; - ret = cs42l51_fill_cache(codec); - if (ret < 0) { - dev_err(codec->dev, "failed to fill register cache\n"); - return ret; - } - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l51->control_type); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* * DAC configuration * - Use signal processor @@ -538,8 +505,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_probe, - .reg_cache_size = CS42L51_NUMREGS + 1, - .reg_word_size = sizeof(u8), .controls = cs42l51_snd_controls, .num_controls = ARRAY_SIZE(cs42l51_snd_controls), @@ -549,38 +514,53 @@ static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; +static const struct regmap_config cs42l51_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L51_CHARGE_FREQ, + .cache_type = REGCACHE_RBTREE, +}; + static int cs42l51_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l51_private *cs42l51; + struct regmap *regmap; + unsigned int val; int ret; + regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c_client->dev, "Failed to create regmap: %d\n", + ret); + return ret; + } + /* Verify that we have a CS42L51 */ - ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID); + ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { dev_err(&i2c_client->dev, "failed to read I2C\n"); goto error; } - if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && - (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { - dev_err(&i2c_client->dev, "Invalid chip id\n"); + if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && + (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { + dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val); ret = -ENODEV; goto error; } dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", - ret & 7); + val & 7); cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); - if (!cs42l51) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l51) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs42l51); - cs42l51->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); @@ -600,10 +580,17 @@ static const struct i2c_device_id cs42l51_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_id); +static const struct of_device_id cs42l51_of_match[] = { + { .compatible = "cirrus,cs42l51", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l51_of_match); + static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51-codec", .owner = THIS_MODULE, + .of_match_table = cs42l51_of_match, }, .id_table = cs42l51_id, .probe = cs42l51_i2c_probe, diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..f0ca6bee6771 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -210,13 +210,11 @@ static const char * const cs42l52_adca_text[] = { static const char * const cs42l52_adcb_text[] = { "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; -static const struct soc_enum adca_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, - ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); +static SOC_ENUM_SINGLE_DECL(adca_enum, + CS42L52_ADC_PGA_A, 5, cs42l52_adca_text); -static const struct soc_enum adcb_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, - ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); +static SOC_ENUM_SINGLE_DECL(adcb_enum, + CS42L52_ADC_PGA_B, 5, cs42l52_adcb_text); static const struct snd_kcontrol_new adca_mux = SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); @@ -229,26 +227,22 @@ static const char * const mic_bias_level_text[] = { "0.8 +VA", "0.83 +VA", "0.91 +VA" }; -static const struct soc_enum mic_bias_level_enum = - SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, - ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); +static SOC_ENUM_SINGLE_DECL(mic_bias_level_enum, + CS42L52_IFACE_CTL2, 0, mic_bias_level_text); static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" }; -static const struct soc_enum mica_enum = - SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, - ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); +static SOC_ENUM_SINGLE_DECL(mica_enum, + CS42L52_MICA_CTL, 5, cs42l52_mic_text); -static const struct soc_enum micb_enum = - SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, - ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); +static SOC_ENUM_SINGLE_DECL(micb_enum, + CS42L52_MICB_CTL, 5, cs42l52_mic_text); static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; -static const struct soc_enum digital_output_mux_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, - ARRAY_SIZE(digital_output_mux_text), - digital_output_mux_text); +static SOC_ENUM_SINGLE_DECL(digital_output_mux_enum, + CS42L52_ADC_MISC_CTL, 6, + digital_output_mux_text); static const struct snd_kcontrol_new digital_output_mux = SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); @@ -258,18 +252,18 @@ static const char * const hp_gain_num_text[] = { "0.7099", "0.8399", "1.000", "1.1430" }; -static const struct soc_enum hp_gain_enum = - SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5, - ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); +static SOC_ENUM_SINGLE_DECL(hp_gain_enum, + CS42L52_PB_CTL1, 5, + hp_gain_num_text); static const char * const beep_pitch_text[] = { "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" }; -static const struct soc_enum beep_pitch_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, - ARRAY_SIZE(beep_pitch_text), beep_pitch_text); +static SOC_ENUM_SINGLE_DECL(beep_pitch_enum, + CS42L52_BEEP_FREQ, 4, + beep_pitch_text); static const char * const beep_ontime_text[] = { "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", @@ -277,66 +271,66 @@ static const char * const beep_ontime_text[] = { "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" }; -static const struct soc_enum beep_ontime_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, - ARRAY_SIZE(beep_ontime_text), beep_ontime_text); +static SOC_ENUM_SINGLE_DECL(beep_ontime_enum, + CS42L52_BEEP_FREQ, 0, + beep_ontime_text); static const char * const beep_offtime_text[] = { "1.23 s", "2.58 s", "3.90 s", "5.20 s", "6.60 s", "8.05 s", "9.35 s", "10.80 s" }; -static const struct soc_enum beep_offtime_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, - ARRAY_SIZE(beep_offtime_text), beep_offtime_text); +static SOC_ENUM_SINGLE_DECL(beep_offtime_enum, + CS42L52_BEEP_VOL, 5, + beep_offtime_text); static const char * const beep_config_text[] = { "Off", "Single", "Multiple", "Continuous" }; -static const struct soc_enum beep_config_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, - ARRAY_SIZE(beep_config_text), beep_config_text); +static SOC_ENUM_SINGLE_DECL(beep_config_enum, + CS42L52_BEEP_TONE_CTL, 6, + beep_config_text); static const char * const beep_bass_text[] = { "50 Hz", "100 Hz", "200 Hz", "250 Hz" }; -static const struct soc_enum beep_bass_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, - ARRAY_SIZE(beep_bass_text), beep_bass_text); +static SOC_ENUM_SINGLE_DECL(beep_bass_enum, + CS42L52_BEEP_TONE_CTL, 1, + beep_bass_text); static const char * const beep_treble_text[] = { "5 kHz", "7 kHz", "10 kHz", " 15 kHz" }; -static const struct soc_enum beep_treble_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, - ARRAY_SIZE(beep_treble_text), beep_treble_text); +static SOC_ENUM_SINGLE_DECL(beep_treble_enum, + CS42L52_BEEP_TONE_CTL, 3, + beep_treble_text); static const char * const ng_threshold_text[] = { "-34dB", "-37dB", "-40dB", "-43dB", "-46dB", "-52dB", "-58dB", "-64dB" }; -static const struct soc_enum ng_threshold_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, - ARRAY_SIZE(ng_threshold_text), ng_threshold_text); +static SOC_ENUM_SINGLE_DECL(ng_threshold_enum, + CS42L52_NOISE_GATE_CTL, 2, + ng_threshold_text); static const char * const cs42l52_ng_delay_text[] = { "50ms", "100ms", "150ms", "200ms"}; -static const struct soc_enum ng_delay_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, - ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); +static SOC_ENUM_SINGLE_DECL(ng_delay_enum, + CS42L52_NOISE_GATE_CTL, 0, + cs42l52_ng_delay_text); static const char * const cs42l52_ng_type_text[] = { "Apply Specific", "Apply All" }; -static const struct soc_enum ng_type_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, - ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); +static SOC_ENUM_SINGLE_DECL(ng_type_enum, + CS42L52_NOISE_GATE_CTL, 6, + cs42l52_ng_type_text); static const char * const left_swap_text[] = { "Left", "LR 2", "Right"}; @@ -347,7 +341,7 @@ static const char * const right_swap_text[] = { static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -356,7 +350,7 @@ static const struct snd_kcontrol_new adca_mixer = SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -365,7 +359,7 @@ static const struct snd_kcontrol_new pcma_mixer = SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -374,7 +368,7 @@ static const struct snd_kcontrol_new adcb_mixer = SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -1115,14 +1109,7 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) static int cs42l52_probe(struct snd_soc_codec *codec) { struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); - int ret; - codec->control_data = cs42l52->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } regcache_cache_only(cs42l52->regmap, true); cs42l52_add_mic_controls(codec); @@ -1134,7 +1121,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; - return ret; + return 0; } static int cs42l52_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..0ee60a19a263 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -278,13 +278,13 @@ static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; -static const struct soc_enum pgaa_enum = - SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, - ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); +static SOC_ENUM_SINGLE_DECL(pgaa_enum, + CS42L73_ADCIPC, 3, + cs42l73_pgaa_text); -static const struct soc_enum pgab_enum = - SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, - ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); +static SOC_ENUM_SINGLE_DECL(pgab_enum, + CS42L73_ADCIPC, 7, + cs42l73_pgab_text); static const struct snd_kcontrol_new pgaa_mux = SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); @@ -309,9 +309,9 @@ static const struct snd_kcontrol_new input_right_mixer[] = { static const char * const cs42l73_ng_delay_text[] = { "50ms", "100ms", "150ms", "200ms" }; -static const struct soc_enum ng_delay_enum = - SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, - ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); +static SOC_ENUM_SINGLE_DECL(ng_delay_enum, + CS42L73_NGCAB, 0, + cs42l73_ng_delay_text); static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -319,7 +319,7 @@ static const char * const cs42l73_mono_mix_texts[] = { static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; static const struct soc_enum spk_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -337,7 +337,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer = SOC_DAPM_ENUM("Route", spk_xsp_enum); static const struct soc_enum esl_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -346,7 +346,7 @@ static const struct snd_kcontrol_new esl_asp_mixer = SOC_DAPM_ENUM("Route", esl_asp_enum); static const struct soc_enum esl_xsp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -357,19 +357,19 @@ static const struct snd_kcontrol_new esl_xsp_mixer = static const char * const cs42l73_ip_swap_text[] = { "Stereo", "Mono A", "Mono B", "Swap A-B"}; -static const struct soc_enum ip_swap_enum = - SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, - ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); +static SOC_ENUM_SINGLE_DECL(ip_swap_enum, + CS42L73_MIOPC, 6, + cs42l73_ip_swap_text); static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; -static const struct soc_enum vsp_output_mux_enum = - SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, - ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); +static SOC_ENUM_SINGLE_DECL(vsp_output_mux_enum, + CS42L73_MIXERCTL, 5, + cs42l73_spo_mixer_text); -static const struct soc_enum xsp_output_mux_enum = - SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, - ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); +static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum, + CS42L73_MIXERCTL, 4, + cs42l73_spo_mixer_text); static const struct snd_kcontrol_new vsp_output_mux = SOC_DAPM_ENUM("Route", vsp_output_mux_enum); @@ -1108,7 +1108,7 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -static u32 cs42l73_asrc_rates[] = { +static const unsigned int cs42l73_asrc_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; @@ -1241,7 +1241,7 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) 0x7F, tristate << 7); } -static struct snd_pcm_hw_constraint_list constraints_12_24 = { +static const struct snd_pcm_hw_constraint_list constraints_12_24 = { .count = ARRAY_SIZE(cs42l73_asrc_rates), .list = cs42l73_asrc_rates, }; @@ -1255,9 +1255,6 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, return 0; } -/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ -#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) - #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1278,14 +1275,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "XSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "XSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1298,14 +1295,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "ASP Playback", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "ASP Capture", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1318,14 +1315,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "VSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "VSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1348,17 +1345,8 @@ static int cs42l73_resume(struct snd_soc_codec *codec) static int cs42l73_probe(struct snd_soc_codec *codec) { - int ret; struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - codec->control_data = cs42l73->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Set Charge Pump Frequency */ @@ -1371,7 +1359,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; - return ret; + return 0; } static int cs42l73_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c new file mode 100644 index 000000000000..657dce27eade --- /dev/null +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -0,0 +1,64 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen@freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <sound/soc.h> + +#include "cs42xx8.h" + +static int cs42xx8_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42xx8_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42xx8_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42xx8_i2c_id[] = { + {"cs42448", (kernel_ulong_t)&cs42448_data}, + {"cs42888", (kernel_ulong_t)&cs42888_data}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); + +static struct i2c_driver cs42xx8_i2c_driver = { + .driver = { + .name = "cs42xx8", + .owner = THIS_MODULE, + .pm = &cs42xx8_pm, + }, + .probe = cs42xx8_i2c_probe, + .remove = cs42xx8_i2c_remove, + .id_table = cs42xx8_i2c_id, +}; + +module_i2c_driver(cs42xx8_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c new file mode 100644 index 000000000000..082299a4e2fa --- /dev/null +++ b/sound/soc/codecs/cs42xx8.c @@ -0,0 +1,602 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen@freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/of_device.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs42xx8.h" + +#define CS42XX8_NUM_SUPPLIES 4 +static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42xx8_priv { + struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES]; + const struct cs42xx8_driver_data *drvdata; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single); +static const struct soc_enum adc3_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc); + +static const struct snd_kcontrol_new cs42xx8_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1, + CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3, + CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5, + CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7, + CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1, + CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3, + CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5, + CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0), + SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum), +}; + +static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1), + + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), +}; + +static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { + /* Capture */ + { "ADC3", NULL, "AIN3L" }, + { "ADC3", NULL, "AIN3R" }, + { "ADC3", NULL, "PWR" }, +}; + +struct cs42xx8_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42xx8_ratios cs42xx8_ratios[] = { + { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, + { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, + { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, + { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, + { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, + { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, + { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, + { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + cs42xx8->sysclk = freq; + + return 0; +} + +static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF, + CS42XX8_INTF_DAC_DIF_MASK | + CS42XX8_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42xx8->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42xx8->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42xx8_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42xx8->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { + if (cs42xx8_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42xx8_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mclk; + + fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, + CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42xx8_dai_ops = { + .set_fmt = cs42xx8_set_dai_fmt, + .set_sysclk = cs42xx8_set_dai_sysclk, + .hw_params = cs42xx8_hw_params, + .digital_mute = cs42xx8_digital_mute, +}; + +static struct snd_soc_dai_driver cs42xx8_dai = { + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .ops = &cs42xx8_dai_ops, +}; + +static const struct reg_default cs42xx8_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x15, 0x00 }, /* Volume Control AIN5 */ + { 0x16, 0x00 }, /* Volume Control AIN6 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_STATUS: + return true; + default: + return false; + } +} + +static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_CHIPID: + case CS42XX8_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42xx8_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42XX8_LASTREG, + .reg_defaults = cs42xx8_reg, + .num_reg_defaults = ARRAY_SIZE(cs42xx8_reg), + .volatile_reg = cs42xx8_volatile_register, + .writeable_reg = cs42xx8_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42xx8_regmap_config); + +static int cs42xx8_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + switch (cs42xx8->drvdata->num_adcs) { + case 3: + snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls, + ARRAY_SIZE(cs42xx8_adc3_snd_controls)); + snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets, + ARRAY_SIZE(cs42xx8_adc3_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes, + ARRAY_SIZE(cs42xx8_adc3_dapm_routes)); + break; + default: + break; + } + + /* Mute all DAC channels */ + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42xx8_driver = { + .probe = cs42xx8_codec_probe, + .idle_bias_off = true, + + .controls = cs42xx8_snd_controls, + .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), + .dapm_widgets = cs42xx8_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), + .dapm_routes = cs42xx8_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), +}; + +const struct cs42xx8_driver_data cs42448_data = { + .name = "cs42448", + .num_adcs = 3, +}; +EXPORT_SYMBOL_GPL(cs42448_data); + +const struct cs42xx8_driver_data cs42888_data = { + .name = "cs42888", + .num_adcs = 2, +}; +EXPORT_SYMBOL_GPL(cs42888_data); + +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); +EXPORT_SYMBOL_GPL(cs42xx8_of_match); + +int cs42xx8_probe(struct device *dev, struct regmap *regmap) +{ + const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev); + struct cs42xx8_priv *cs42xx8; + int ret, val, i; + + cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL); + if (cs42xx8 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42xx8); + + if (of_id) + cs42xx8->drvdata = of_id->data; + + if (!cs42xx8->drvdata) { + dev_err(dev, "failed to find driver data\n"); + return -EINVAL; + } + + cs42xx8->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42xx8->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42xx8->clk)); + return -EINVAL; + } + + cs42xx8->sysclk = clk_get_rate(cs42xx8->clk); + + for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++) + cs42xx8->supplies[i].supply = cs42xx8_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42xx8->regmap = regmap; + if (IS_ERR(cs42xx8->regmap)) { + ret = PTR_ERR(cs42xx8->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42xx8->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42XX8_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42XX8_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42xx8->regmap, false); + + cs42xx8_dai.name = cs42xx8->drvdata->name; + + /* Each adc supports stereo input */ + cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2; + + ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42xx8->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42xx8_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42xx8_runtime_resume(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42xx8->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42xx8->regmap, false); + + ret = regcache_sync(cs42xx8->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); +err_clk: + clk_disable_unprepare(cs42xx8->clk); + + return ret; +} + +static int cs42xx8_runtime_suspend(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + + regcache_cache_only(cs42xx8->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + clk_disable_unprepare(cs42xx8->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42xx8_pm = { + SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42xx8_pm); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h new file mode 100644 index 000000000000..da0b94aee419 --- /dev/null +++ b/sound/soc/codecs/cs42xx8.h @@ -0,0 +1,238 @@ +/* + * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen <Guangyu.Chen@freescale.com> + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42XX8_H +#define _CS42XX8_H + +struct cs42xx8_driver_data { + char name[32]; + int num_adcs; +}; + +extern const struct dev_pm_ops cs42xx8_pm; +extern const struct cs42xx8_driver_data cs42448_data; +extern const struct cs42xx8_driver_data cs42888_data; +extern const struct regmap_config cs42xx8_regmap_config; +int cs42xx8_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42XX8_CHIPID 0x01 /* Chip ID */ +#define CS42XX8_PWRCTL 0x02 /* Power Control */ +#define CS42XX8_FUNCMOD 0x03 /* Functional Mode */ +#define CS42XX8_INTF 0x04 /* Interface Formats */ +#define CS42XX8_ADCCTL 0x05 /* ADC Control */ +#define CS42XX8_TXCTL 0x06 /* Transition Control */ +#define CS42XX8_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42XX8_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42XX8_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42XX8_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42XX8_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42XX8_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42XX8_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42XX8_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42XX8_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42XX8_DACINV 0x10 /* DAC Channel Invert */ +#define CS42XX8_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42XX8_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42XX8_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42XX8_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42XX8_VOLAIN5 0x15 /* Volume Control AIN5 */ +#define CS42XX8_VOLAIN6 0x16 /* Volume Control AIN6 */ +#define CS42XX8_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42XX8_STATUSCTL 0x18 /* Status Control */ +#define CS42XX8_STATUS 0x19 /* Status */ +#define CS42XX8_STATUSM 0x1A /* Status Mask */ +#define CS42XX8_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42XX8_FIRSTREG CS42XX8_CHIPID +#define CS42XX8_LASTREG CS42XX8_MUTEC +#define CS42XX8_NUMREGS (CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1) +#define CS42XX8_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42XX8_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42XX8_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT 7 +#define CS42XX8_PWRCTL_PDN_ADC3_MASK (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC3 (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42XX8_PWRCTL_PDN_ADC2_MASK (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2 (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42XX8_PWRCTL_PDN_ADC1_MASK (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1 (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42XX8_PWRCTL_PDN_DAC4_MASK (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4 (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42XX8_PWRCTL_PDN_DAC3_MASK (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3 (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42XX8_PWRCTL_PDN_DAC2_MASK (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2 (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42XX8_PWRCTL_PDN_DAC1_MASK (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1 (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_SHIFT 0 +#define CS42XX8_PWRCTL_PDN_MASK (1 << CS42XX8_PWRCTL_PDN_SHIFT) +#define CS42XX8_PWRCTL_PDN (1 << CS42XX8_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42XX8_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42XX8_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_DAC_FM_MASK (((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_DAC_FM(v) ((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42XX8_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_ADC_FM_MASK (((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM(v) ((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK) +#define CS42XX8_FUNCMOD_xC_FM(x, v) ((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v)) +#define CS42XX8_FUNCMOD_MFREQ_SHIFT 1 +#define CS42XX8_FUNCMOD_MFREQ_WIDTH 3 +#define CS42XX8_FUNCMOD_MFREQ_MASK (((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT) +#define CS42XX8_FUNCMOD_MFREQ_256(s) ((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_384(s) ((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_512(s) ((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_768(s) ((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_1024(s) ((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42XX8_FM_SINGLE 0 +#define CS42XX8_FM_DOUBLE 1 +#define CS42XX8_FM_QUAD 2 +#define CS42XX8_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42XX8_INTF_FREEZE_SHIFT 7 +#define CS42XX8_INTF_FREEZE_MASK (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_FREEZE (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_AUX_DIF_SHIFT 6 +#define CS42XX8_INTF_AUX_DIF_MASK (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_AUX_DIF (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_SHIFT 3 +#define CS42XX8_INTF_DAC_DIF_WIDTH 3 +#define CS42XX8_INTF_DAC_DIF_MASK (((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_LEFTJ (0 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_I2S (1 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_SHIFT 0 +#define CS42XX8_INTF_ADC_DIF_WIDTH 3 +#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_LEFTJ (0 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_I2S (1 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42XX8_ADCCTL_DAC_DEM_MASK (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT 2 +#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT 1 +#define CS42XX8_ADCCTL_AIN5_MUX_MASK (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT 0 +#define CS42XX8_ADCCTL_AIN6_MUX_MASK (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42XX8_TXCTL_DAC_SNGVOL_MASK (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SNGVOL (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SHIFT 5 +#define CS42XX8_TXCTL_DAC_SZC_WIDTH 2 +#define CS42XX8_TXCTL_DAC_SZC_MASK (((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_IC (0 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_ZC (1 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SR (2 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SRZC (3 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_AMUTE_SHIFT 4 +#define CS42XX8_TXCTL_AMUTE_MASK (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_AMUTE (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42XX8_TXCTL_ADC_SNGVOL_MASK (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SHIFT 0 +#define CS42XX8_TXCTL_ADC_SZC_MASK (((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_IC (0 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_ZC (1 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SR (2 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SRZC (3 << CS42XX8_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42XX8_DACMUTE_AOUT(n) (0x1 << n) +#define CS42XX8_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42XX8_STATUSCTL_INI_SHIFT 2 +#define CS42XX8_STATUSCTL_INI_WIDTH 2 +#define CS42XX8_STATUSCTL_INI_MASK (((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW (1 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN (2 << CS42XX8_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42XX8_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42XX8_MUTEC_MCPOLARITY_MASK (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42XX8_H */ diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e62e294a8033..137e8ebc092c 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -307,29 +307,29 @@ static const char * const da7210_hpf_cutoff_txt[] = { "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" }; -static const struct soc_enum da7210_dac_hpf_cutoff = - SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_dac_hpf_cutoff, + DA7210_DAC_HPF, 0, da7210_hpf_cutoff_txt); -static const struct soc_enum da7210_adc_hpf_cutoff = - SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_adc_hpf_cutoff, + DA7210_ADC_HPF, 0, da7210_hpf_cutoff_txt); /* ADC and DAC voice (8kHz) high pass cutoff value */ static const char * const da7210_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da7210_dac_vf_cutoff = - SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_dac_vf_cutoff, + DA7210_DAC_HPF, 4, da7210_vf_cutoff_txt); -static const struct soc_enum da7210_adc_vf_cutoff = - SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_adc_vf_cutoff, + DA7210_ADC_HPF, 4, da7210_vf_cutoff_txt); static const char *da7210_hp_mode_txt[] = { "Class H", "Class G" }; -static const struct soc_enum da7210_hp_mode_sel = - SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +static SOC_ENUM_SINGLE_DECL(da7210_hp_mode_sel, + DA7210_HP_CFG, 0, da7210_hp_mode_txt); /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, @@ -1071,17 +1071,9 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); - int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ da7210->master = 0; /* This will be set from set_fmt() */ diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 0c77e7ad7423..738fa18a50d2 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -63,30 +63,30 @@ static const char * const da7213_voice_hpf_corner_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da7213_dac_voice_hpf_corner = - SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, - DA7213_VOICE_HPF_CORNER_MAX, - da7213_voice_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_voice_hpf_corner, + DA7213_DAC_FILTERS1, + DA7213_VOICE_HPF_CORNER_SHIFT, + da7213_voice_hpf_corner_txt); -static const struct soc_enum da7213_adc_voice_hpf_corner = - SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, - DA7213_VOICE_HPF_CORNER_MAX, - da7213_voice_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_adc_voice_hpf_corner, + DA7213_ADC_FILTERS1, + DA7213_VOICE_HPF_CORNER_SHIFT, + da7213_voice_hpf_corner_txt); /* ADC and DAC high pass filter cutoff value */ static const char * const da7213_audio_hpf_corner_txt[] = { "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" }; -static const struct soc_enum da7213_dac_audio_hpf_corner = - SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, - DA7213_AUDIO_HPF_CORNER_MAX, - da7213_audio_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_audio_hpf_corner, + DA7213_DAC_FILTERS1 + , DA7213_AUDIO_HPF_CORNER_SHIFT, + da7213_audio_hpf_corner_txt); -static const struct soc_enum da7213_adc_audio_hpf_corner = - SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, - DA7213_AUDIO_HPF_CORNER_MAX, - da7213_audio_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_adc_audio_hpf_corner, + DA7213_ADC_FILTERS1, + DA7213_AUDIO_HPF_CORNER_SHIFT, + da7213_audio_hpf_corner_txt); /* Gain ramping rate value */ static const char * const da7213_gain_ramp_rate_txt[] = { @@ -94,52 +94,50 @@ static const char * const da7213_gain_ramp_rate_txt[] = { "nominal rate / 32" }; -static const struct soc_enum da7213_gain_ramp_rate = - SOC_ENUM_SINGLE(DA7213_GAIN_RAMP_CTRL, DA7213_GAIN_RAMP_RATE_SHIFT, - DA7213_GAIN_RAMP_RATE_MAX, da7213_gain_ramp_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_gain_ramp_rate, + DA7213_GAIN_RAMP_CTRL, + DA7213_GAIN_RAMP_RATE_SHIFT, + da7213_gain_ramp_rate_txt); /* DAC noise gate setup time value */ static const char * const da7213_dac_ng_setup_time_txt[] = { "256 samples", "512 samples", "1024 samples", "2048 samples" }; -static const struct soc_enum da7213_dac_ng_setup_time = - SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, - DA7213_DAC_NG_SETUP_TIME_SHIFT, - DA7213_DAC_NG_SETUP_TIME_MAX, - da7213_dac_ng_setup_time_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_setup_time, + DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_SETUP_TIME_SHIFT, + da7213_dac_ng_setup_time_txt); /* DAC noise gate rampup rate value */ static const char * const da7213_dac_ng_rampup_txt[] = { "0.02 ms/dB", "0.16 ms/dB" }; -static const struct soc_enum da7213_dac_ng_rampup_rate = - SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, - DA7213_DAC_NG_RAMPUP_RATE_SHIFT, - DA7213_DAC_NG_RAMP_RATE_MAX, - da7213_dac_ng_rampup_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampup_rate, + DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPUP_RATE_SHIFT, + da7213_dac_ng_rampup_txt); /* DAC noise gate rampdown rate value */ static const char * const da7213_dac_ng_rampdown_txt[] = { "0.64 ms/dB", "20.48 ms/dB" }; -static const struct soc_enum da7213_dac_ng_rampdown_rate = - SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, - DA7213_DAC_NG_RAMPDN_RATE_SHIFT, - DA7213_DAC_NG_RAMP_RATE_MAX, - da7213_dac_ng_rampdown_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampdown_rate, + DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPDN_RATE_SHIFT, + da7213_dac_ng_rampdown_txt); /* DAC soft mute rate value */ static const char * const da7213_dac_soft_mute_rate_txt[] = { "1", "2", "4", "8", "16", "32", "64" }; -static const struct soc_enum da7213_dac_soft_mute_rate = - SOC_ENUM_SINGLE(DA7213_DAC_FILTERS5, DA7213_DAC_SOFTMUTE_RATE_SHIFT, - DA7213_DAC_SOFTMUTE_RATE_MAX, - da7213_dac_soft_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_soft_mute_rate, + DA7213_DAC_FILTERS5, + DA7213_DAC_SOFTMUTE_RATE_SHIFT, + da7213_dac_soft_mute_rate_txt); /* ALC Attack Rate select */ static const char * const da7213_alc_attack_rate_txt[] = { @@ -147,9 +145,10 @@ static const char * const da7213_alc_attack_rate_txt[] = { "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da7213_alc_attack_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_ATTACK_SHIFT, - DA7213_ALC_ATTACK_MAX, da7213_alc_attack_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_attack_rate, + DA7213_ALC_CTRL2, + DA7213_ALC_ATTACK_SHIFT, + da7213_alc_attack_rate_txt); /* ALC Release Rate select */ static const char * const da7213_alc_release_rate_txt[] = { @@ -157,9 +156,10 @@ static const char * const da7213_alc_release_rate_txt[] = { "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da7213_alc_release_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_RELEASE_SHIFT, - DA7213_ALC_RELEASE_MAX, da7213_alc_release_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_release_rate, + DA7213_ALC_CTRL2, + DA7213_ALC_RELEASE_SHIFT, + da7213_alc_release_rate_txt); /* ALC Hold Time select */ static const char * const da7213_alc_hold_time_txt[] = { @@ -168,22 +168,25 @@ static const char * const da7213_alc_hold_time_txt[] = { "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" }; -static const struct soc_enum da7213_alc_hold_time = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_HOLD_SHIFT, - DA7213_ALC_HOLD_MAX, da7213_alc_hold_time_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_hold_time, + DA7213_ALC_CTRL3, + DA7213_ALC_HOLD_SHIFT, + da7213_alc_hold_time_txt); /* ALC Input Signal Tracking rate select */ static const char * const da7213_alc_integ_rate_txt[] = { "1/4", "1/16", "1/256", "1/65536" }; -static const struct soc_enum da7213_alc_integ_attack_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_ATTACK_SHIFT, - DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_attack_rate, + DA7213_ALC_CTRL3, + DA7213_ALC_INTEG_ATTACK_SHIFT, + da7213_alc_integ_rate_txt); -static const struct soc_enum da7213_alc_integ_release_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_RELEASE_SHIFT, - DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_release_rate, + DA7213_ALC_CTRL3, + DA7213_ALC_INTEG_RELEASE_SHIFT, + da7213_alc_integ_rate_txt); /* @@ -584,15 +587,17 @@ static const char * const da7213_mic_amp_in_sel_txt[] = { "Differential", "MIC_P", "MIC_N" }; -static const struct soc_enum da7213_mic_1_amp_in_sel = - SOC_ENUM_SINGLE(DA7213_MIC_1_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, - DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static SOC_ENUM_SINGLE_DECL(da7213_mic_1_amp_in_sel, + DA7213_MIC_1_CTRL, + DA7213_MIC_AMP_IN_SEL_SHIFT, + da7213_mic_amp_in_sel_txt); static const struct snd_kcontrol_new da7213_mic_1_amp_in_sel_mux = SOC_DAPM_ENUM("Mic 1 Amp Source MUX", da7213_mic_1_amp_in_sel); -static const struct soc_enum da7213_mic_2_amp_in_sel = - SOC_ENUM_SINGLE(DA7213_MIC_2_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, - DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static SOC_ENUM_SINGLE_DECL(da7213_mic_2_amp_in_sel, + DA7213_MIC_2_CTRL, + DA7213_MIC_AMP_IN_SEL_SHIFT, + da7213_mic_amp_in_sel_txt); static const struct snd_kcontrol_new da7213_mic_2_amp_in_sel_mux = SOC_DAPM_ENUM("Mic 2 Amp Source MUX", da7213_mic_2_amp_in_sel); @@ -601,15 +606,17 @@ static const char * const da7213_dai_src_txt[] = { "ADC Left", "ADC Right", "DAI Input Left", "DAI Input Right" }; -static const struct soc_enum da7213_dai_l_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_L_SRC_SHIFT, - DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dai_l_src, + DA7213_DIG_ROUTING_DAI, + DA7213_DAI_L_SRC_SHIFT, + da7213_dai_src_txt); static const struct snd_kcontrol_new da7213_dai_l_src_mux = SOC_DAPM_ENUM("DAI Left Source MUX", da7213_dai_l_src); -static const struct soc_enum da7213_dai_r_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_R_SRC_SHIFT, - DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dai_r_src, + DA7213_DIG_ROUTING_DAI, + DA7213_DAI_R_SRC_SHIFT, + da7213_dai_src_txt); static const struct snd_kcontrol_new da7213_dai_r_src_mux = SOC_DAPM_ENUM("DAI Right Source MUX", da7213_dai_r_src); @@ -619,15 +626,17 @@ static const char * const da7213_dac_src_txt[] = { "DAI Input Right" }; -static const struct soc_enum da7213_dac_l_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_L_SRC_SHIFT, - DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_l_src, + DA7213_DIG_ROUTING_DAC, + DA7213_DAC_L_SRC_SHIFT, + da7213_dac_src_txt); static const struct snd_kcontrol_new da7213_dac_l_src_mux = SOC_DAPM_ENUM("DAC Left Source MUX", da7213_dac_l_src); -static const struct soc_enum da7213_dac_r_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_R_SRC_SHIFT, - DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_r_src, + DA7213_DIG_ROUTING_DAC, + DA7213_DAC_R_SRC_SHIFT, + da7213_dac_src_txt); static const struct snd_kcontrol_new da7213_dac_r_src_mux = SOC_DAPM_ENUM("DAC Right Source MUX", da7213_dac_r_src); @@ -1384,17 +1393,9 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, static int da7213_probe(struct snd_soc_codec *codec) { - int ret; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); struct da7213_platform_data *pdata = da7213->pdata; - codec->control_data = da7213->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Default to using ALC auto offset calibration mode. */ snd_soc_update_bits(codec, DA7213_ALC_CTRL1, DA7213_ALC_CALIB_MODE_MAN, 0); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b6569910..7d168ec71cd7 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -269,81 +269,65 @@ static const char *da732x_hpf_voice[] = { "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da732x_dac1_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac1_hpf_mode_enum, + DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_dac2_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac2_hpf_mode_enum, + DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_dac3_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac3_hpf_mode_enum, + DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_adc1_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc1_hpf_mode_enum, + DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_adc2_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc2_hpf_mode_enum, + DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_dac1_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac1_hp_filter_enum, + DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_dac2_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac2_hp_filter_enum, + DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_dac3_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac3_hp_filter_enum, + DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_adc1_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc1_hp_filter_enum, + DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_adc2_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc2_hp_filter_enum, + DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_dac1_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac1_voice_filter_enum, + DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); -static const struct soc_enum da732x_dac2_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac2_voice_filter_enum, + DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); -static const struct soc_enum da732x_dac3_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac3_voice_filter_enum, + DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); -static const struct soc_enum da732x_adc1_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; - -static const struct soc_enum da732x_adc2_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc1_voice_filter_enum, + DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); +static SOC_ENUM_SINGLE_DECL(da732x_adc2_voice_filter_enum, + DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); static int da732x_hpf_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -714,65 +698,65 @@ static const char *enable_text[] = { }; /* ADC1LMUX */ -static const struct soc_enum adc1l_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, - DA732X_ADCL_MUX_MAX, adcl_text); +static SOC_ENUM_SINGLE_DECL(adc1l_enum, + DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, + adcl_text); static const struct snd_kcontrol_new adc1l_mux = SOC_DAPM_ENUM("ADC Route", adc1l_enum); /* ADC1RMUX */ -static const struct soc_enum adc1r_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, - DA732X_ADCR_MUX_MAX, adcr_text); +static SOC_ENUM_SINGLE_DECL(adc1r_enum, + DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, + adcr_text); static const struct snd_kcontrol_new adc1r_mux = SOC_DAPM_ENUM("ADC Route", adc1r_enum); /* ADC2LMUX */ -static const struct soc_enum adc2l_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, - DA732X_ADCL_MUX_MAX, adcl_text); +static SOC_ENUM_SINGLE_DECL(adc2l_enum, + DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, + adcl_text); static const struct snd_kcontrol_new adc2l_mux = SOC_DAPM_ENUM("ADC Route", adc2l_enum); /* ADC2RMUX */ -static const struct soc_enum adc2r_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, - DA732X_ADCR_MUX_MAX, adcr_text); +static SOC_ENUM_SINGLE_DECL(adc2r_enum, + DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, + adcr_text); static const struct snd_kcontrol_new adc2r_mux = SOC_DAPM_ENUM("ADC Route", adc2r_enum); -static const struct soc_enum da732x_hp_left_output = - SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_hp_left_output, + DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new hpl_mux = SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output); -static const struct soc_enum da732x_hp_right_output = - SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_hp_right_output, + DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new hpr_mux = SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output); -static const struct soc_enum da732x_speaker_output = - SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_speaker_output, + DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new spk_mux = SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output); -static const struct soc_enum da732x_lout4_output = - SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_lout4_output, + DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new lout4_mux = SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output); -static const struct soc_enum da732x_lout2_output = - SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_lout2_output, + DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new lout2_mux = SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output); @@ -1268,11 +1252,23 @@ static struct snd_soc_dai_driver da732x_dai[] = { }, }; +static bool da732x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case DA732X_REG_HPL_DAC_OFF_CNTL: + case DA732X_REG_HPR_DAC_OFF_CNTL: + return true; + default: + return false; + } +} + static const struct regmap_config da732x_regmap = { .reg_bits = 8, .val_bits = 8, .max_register = DA732X_MAX_REG, + .volatile_reg = da732x_volatile, .reg_defaults = da732x_reg_cache, .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), .cache_type = REGCACHE_RBTREE, @@ -1301,9 +1297,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check DAC offset sign */ - sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + sign[DA732X_HPL_DAC] = (snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); - sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + sign[DA732X_HPR_DAC] = (snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); /* Binary search DAC offset values (both channels at once) */ @@ -1320,10 +1316,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) offset[DA732X_HPL_DAC] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) offset[DA732X_HPR_DAC] &= ~step; @@ -1364,9 +1360,9 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check output offset sign */ - sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + sign[DA732X_HPL_AMP] = snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO; - sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + sign[DA732X_HPR_AMP] = snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO; snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | @@ -1387,10 +1383,10 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL) & + if ((snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) offset[DA732X_HPL_AMP] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR) & + if ((snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) offset[DA732X_HPR_AMP] &= ~step; @@ -1487,8 +1483,8 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, da732x_hp_dc_offset_cancellation(codec); - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(da732x->regmap, false); + regcache_sync(da732x->regmap); } else { snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_BOOST_MASK, @@ -1499,7 +1495,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(da732x->regmap, true); da732x_set_charge_pump(codec, DA732X_DISABLE_CP); snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, DA732X_BIAS_DIS); @@ -1516,23 +1512,14 @@ static int da732x_probe(struct snd_soc_codec *codec) { struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; da732x->codec = codec; dapm->idle_bias_off = false; - codec->control_data = da732x->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to register codec.\n"); - goto err; - } - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -err: - return ret; + + return 0; } static int da732x_remove(struct snd_soc_codec *codec) @@ -1554,7 +1541,6 @@ static struct snd_soc_codec_driver soc_codec_dev_da732x = { .dapm_routes = da732x_dapm_routes, .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), .set_pll = da732x_set_dai_pll, - .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), }; static int da732x_i2c_probe(struct i2c_client *i2c, diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index c8ce5475de22..1dceafeec415 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -113,9 +113,6 @@ #define DA732X_EQ_OVERALL_VOL_DB_MIN -1800 #define DA732X_EQ_OVERALL_VOL_DB_INC 600 -#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \ - {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext} - enum da732x_sysctl { DA732X_SR_8KHZ = 0x1, DA732X_SR_11_025KHZ = 0x2, diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..4ff06b50fbba 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -18,6 +18,8 @@ #include <linux/regmap.h> #include <linux/slab.h> #include <linux/module.h> +#include <linux/of.h> +#include <linux/of_device.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -321,22 +323,22 @@ static const char * const da9055_hpf_cutoff_txt[] = { "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" }; -static const struct soc_enum da9055_dac_hpf_cutoff = - SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_hpf_cutoff, + DA9055_DAC_FILTERS1, 4, da9055_hpf_cutoff_txt); -static const struct soc_enum da9055_adc_hpf_cutoff = - SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_adc_hpf_cutoff, + DA9055_ADC_FILTERS1, 4, da9055_hpf_cutoff_txt); /* ADC and DAC voice mode (8kHz) high pass cutoff value */ static const char * const da9055_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da9055_dac_vf_cutoff = - SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_vf_cutoff, + DA9055_DAC_FILTERS1, 0, da9055_vf_cutoff_txt); -static const struct soc_enum da9055_adc_vf_cutoff = - SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_adc_vf_cutoff, + DA9055_ADC_FILTERS1, 0, da9055_vf_cutoff_txt); /* Gain ramping rate value */ static const char * const da9055_gain_ramping_txt[] = { @@ -344,44 +346,44 @@ static const char * const da9055_gain_ramping_txt[] = { "nominal rate / 8" }; -static const struct soc_enum da9055_gain_ramping_rate = - SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); +static SOC_ENUM_SINGLE_DECL(da9055_gain_ramping_rate, + DA9055_GAIN_RAMP_CTRL, 0, da9055_gain_ramping_txt); /* DAC noise gate setup time value */ static const char * const da9055_dac_ng_setup_time_txt[] = { "256 samples", "512 samples", "1024 samples", "2048 samples" }; -static const struct soc_enum da9055_dac_ng_setup_time = - SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, - da9055_dac_ng_setup_time_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_setup_time, + DA9055_DAC_NG_SETUP_TIME, 0, + da9055_dac_ng_setup_time_txt); /* DAC noise gate rampup rate value */ static const char * const da9055_dac_ng_rampup_txt[] = { "0.02 ms/dB", "0.16 ms/dB" }; -static const struct soc_enum da9055_dac_ng_rampup_rate = - SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, - da9055_dac_ng_rampup_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampup_rate, + DA9055_DAC_NG_SETUP_TIME, 2, + da9055_dac_ng_rampup_txt); /* DAC noise gate rampdown rate value */ static const char * const da9055_dac_ng_rampdown_txt[] = { "0.64 ms/dB", "20.48 ms/dB" }; -static const struct soc_enum da9055_dac_ng_rampdown_rate = - SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, - da9055_dac_ng_rampdown_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampdown_rate, + DA9055_DAC_NG_SETUP_TIME, 3, + da9055_dac_ng_rampdown_txt); /* DAC soft mute rate value */ static const char * const da9055_dac_soft_mute_rate_txt[] = { "1", "2", "4", "8", "16", "32", "64" }; -static const struct soc_enum da9055_dac_soft_mute_rate = - SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, - da9055_dac_soft_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_soft_mute_rate, + DA9055_DAC_FILTERS5, 4, + da9055_dac_soft_mute_rate_txt); /* DAC routing select */ static const char * const da9055_dac_src_txt[] = { @@ -389,40 +391,40 @@ static const char * const da9055_dac_src_txt[] = { "AIF input right" }; -static const struct soc_enum da9055_dac_l_src = - SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_l_src, + DA9055_DIG_ROUTING_DAC, 0, da9055_dac_src_txt); -static const struct soc_enum da9055_dac_r_src = - SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_r_src, + DA9055_DIG_ROUTING_DAC, 4, da9055_dac_src_txt); /* MIC PGA Left source select */ static const char * const da9055_mic_l_src_txt[] = { "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" }; -static const struct soc_enum da9055_mic_l_src = - SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_mic_l_src, + DA9055_MIXIN_L_SELECT, 4, da9055_mic_l_src_txt); /* MIC PGA Right source select */ static const char * const da9055_mic_r_src_txt[] = { "MIC2_R_L", "MIC2_R", "MIC2_L" }; -static const struct soc_enum da9055_mic_r_src = - SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_mic_r_src, + DA9055_MIXIN_R_SELECT, 4, da9055_mic_r_src_txt); /* ALC Input Signal Tracking rate select */ static const char * const da9055_signal_tracking_rate_txt[] = { "1/4", "1/16", "1/256", "1/65536" }; -static const struct soc_enum da9055_integ_attack_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, - da9055_signal_tracking_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_integ_attack_rate, + DA9055_ALC_CTRL3, 4, + da9055_signal_tracking_rate_txt); -static const struct soc_enum da9055_integ_release_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, - da9055_signal_tracking_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_integ_release_rate, + DA9055_ALC_CTRL3, 6, + da9055_signal_tracking_rate_txt); /* ALC Attack Rate select */ static const char * const da9055_attack_rate_txt[] = { @@ -430,8 +432,8 @@ static const char * const da9055_attack_rate_txt[] = { "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da9055_attack_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_attack_rate, + DA9055_ALC_CTRL2, 0, da9055_attack_rate_txt); /* ALC Release Rate select */ static const char * const da9055_release_rate_txt[] = { @@ -439,8 +441,8 @@ static const char * const da9055_release_rate_txt[] = { "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da9055_release_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_release_rate, + DA9055_ALC_CTRL2, 4, da9055_release_rate_txt); /* ALC Hold Time select */ static const char * const da9055_hold_time_txt[] = { @@ -449,8 +451,8 @@ static const char * const da9055_hold_time_txt[] = { "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" }; -static const struct soc_enum da9055_hold_time = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); +static SOC_ENUM_SINGLE_DECL(da9055_hold_time, + DA9055_ALC_CTRL3, 0, da9055_hold_time_txt); static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) { @@ -1381,16 +1383,8 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec, static int da9055_probe(struct snd_soc_codec *codec) { - int ret; struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); - codec->control_data = da9055->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Enable all Gain Ramps */ snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); @@ -1523,17 +1517,30 @@ static int da9055_remove(struct i2c_client *client) return 0; } +/* + * DO NOT change the device Ids. The naming is intentionally specific as both + * the CODEC and PMIC parts of this chip are instantiated separately as I2C + * devices (both have configurable I2C addresses, and are to all intents and + * purposes separate). As a result there are specific DA9055 Ids for CODEC + * and PMIC, which must be different to operate together. + */ static const struct i2c_device_id da9055_i2c_id[] = { - { "da9055", 0 }, + { "da9055-codec", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); +static const struct of_device_id da9055_of_match[] = { + { .compatible = "dlg,da9055-codec", }, + { } +}; + /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { - .name = "da9055", + .name = "da9055-codec", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(da9055_of_match), }, .probe = da9055_i2c_probe, .remove = da9055_remove, diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..3a89ce66d51d 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -140,13 +140,17 @@ static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; static const struct soc_enum isabelle_rx1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, + ARRAY_SIZE(isabelle_rx1_texts), isabelle_rx1_texts), }; static const struct soc_enum isabelle_rx2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), - SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, + ARRAY_SIZE(isabelle_rx2_texts), isabelle_rx2_texts), }; /* Headset DAC playback switches */ @@ -161,13 +165,17 @@ static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; static const struct soc_enum isabelle_atx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_atx_texts), isabelle_atx_texts), }; static const struct soc_enum isabelle_vtx_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), - SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, + ARRAY_SIZE(isabelle_vtx_texts), isabelle_vtx_texts), }; static const struct snd_kcontrol_new atx_mux_controls = @@ -183,17 +191,13 @@ static const char *isabelle_amic1_texts[] = { /* Left analog microphone selection */ static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; -static const struct soc_enum isabelle_amic1_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, - ARRAY_SIZE(isabelle_amic1_texts), - isabelle_amic1_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic1_enum, + ISABELLE_AMIC_CFG_REG, 5, + isabelle_amic1_texts); -static const struct soc_enum isabelle_amic2_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, - ARRAY_SIZE(isabelle_amic2_texts), - isabelle_amic2_texts), -}; +static SOC_ENUM_SINGLE_DECL(isabelle_amic2_enum, + ISABELLE_AMIC_CFG_REG, 4, + isabelle_amic2_texts); static const struct snd_kcontrol_new amic1_control = SOC_DAPM_ENUM("Route", isabelle_amic1_enum); @@ -206,16 +210,20 @@ static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; static const struct soc_enum isabelle_st_audio_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), - SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_audio_texts), isabelle_st_audio_texts), }; static const struct soc_enum isabelle_st_voice_enum[] = { - SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), - SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, + ARRAY_SIZE(isabelle_st_voice_texts), isabelle_st_voice_texts), }; @@ -910,8 +918,7 @@ static int isabelle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 aif = 0; unsigned int fs_val = 0; @@ -1082,23 +1089,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = { }, }; -static int isabelle_probe(struct snd_soc_codec *codec) -{ - int ret = 0; - - codec->control_data = dev_get_regmap(codec->dev, NULL); - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_isabelle = { - .probe = isabelle_probe, .set_bias_level = isabelle_set_bias_level, .controls = isabelle_snd_controls, .num_controls = ARRAY_SIZE(isabelle_snd_controls), diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 0e5743ea79df..4f048db9f55f 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -101,8 +101,7 @@ static const char *lm4857_mode[] = { "Headphone", }; -static const struct soc_enum lm4857_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode); +static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode); static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN"), diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e19490cfb3a8..275b3f72f3f4 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -195,33 +195,31 @@ struct lm49453_priv { static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; -static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, - lm49453_mic2mode_text); +static SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; -static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, - LM49453_P0_DIGITAL_MIC1_CONFIG_REG, - 7, lm49453_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, 7, + lm49453_dmic_cfg_text); -static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, - LM49453_P0_DIGITAL_MIC2_CONFIG_REG, - 7, lm49453_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, 7, + lm49453_dmic_cfg_text); /* MUX Controls */ static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; -static const struct soc_enum lm49453_adcl_enum = - SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, - ARRAY_SIZE(lm49453_adcl_mux_text), - lm49453_adcl_mux_text); +static SOC_ENUM_SINGLE_DECL(lm49453_adcl_enum, + LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + lm49453_adcl_mux_text); -static const struct soc_enum lm49453_adcr_enum = - SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, - ARRAY_SIZE(lm49453_adcr_mux_text), - lm49453_adcr_mux_text); +static SOC_ENUM_SINGLE_DECL(lm49453_adcr_enum, + LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + lm49453_adcr_mux_text); static const struct snd_kcontrol_new lm49453_adcl_mux_control = SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); @@ -1409,22 +1407,6 @@ static int lm49453_resume(struct snd_soc_codec *codec) return 0; } -static int lm49453_probe(struct snd_soc_codec *codec) -{ - struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); - int ret = 0; - - codec->control_data = lm49453->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1433,7 +1415,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { - .probe = lm49453_probe, .remove = lm49453_remove, .suspend = lm49453_suspend, .resume = lm49453_resume, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 31f91560e9f6..ec481fc428c7 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -135,11 +135,6 @@ static int max9768_probe(struct snd_soc_codec *codec) struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = max9768->regmap; - ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP); - if (ret) - return ret; - if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) { ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM); if (ret) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ee660e2d3df3..ef7cf89f5623 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -597,28 +597,27 @@ static const unsigned int max98088_exmode_values[] = { 0x00, 0x43, 0x10, 0x20, 0x30, 0x40, 0x11, 0x22, 0x32 }; -static const struct soc_enum max98088_exmode_enum = - SOC_VALUE_ENUM_SINGLE(M98088_REG_41_SPKDHP, 0, 127, - ARRAY_SIZE(max98088_exmode_texts), - max98088_exmode_texts, - max98088_exmode_values); +static SOC_VALUE_ENUM_SINGLE_DECL(max98088_exmode_enum, + M98088_REG_41_SPKDHP, 0, 127, + max98088_exmode_texts, + max98088_exmode_values); static const char *max98088_ex_thresh[] = { /* volts PP */ "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; -static const struct soc_enum max98088_ex_thresh_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_42_SPKDHP_THRESH, 0, 8, - max98088_ex_thresh), -}; +static SOC_ENUM_SINGLE_DECL(max98088_ex_thresh_enum, + M98088_REG_42_SPKDHP_THRESH, 0, + max98088_ex_thresh); static const char *max98088_fltr_mode[] = {"Voice", "Music" }; -static const struct soc_enum max98088_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 7, 2, max98088_fltr_mode), -}; +static SOC_ENUM_SINGLE_DECL(max98088_filter_mode_enum, + M98088_REG_18_DAI1_FILTERS, 7, + max98088_fltr_mode); static const char *max98088_extmic_text[] = { "None", "MIC1", "MIC2" }; -static const struct soc_enum max98088_extmic_enum = - SOC_ENUM_SINGLE(M98088_REG_48_CFG_MIC, 0, 3, max98088_extmic_text); +static SOC_ENUM_SINGLE_DECL(max98088_extmic_enum, + M98088_REG_48_CFG_MIC, 0, + max98088_extmic_text); static const struct snd_kcontrol_new max98088_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98088_extmic_enum); @@ -626,12 +625,12 @@ static const struct snd_kcontrol_new max98088_extmic_mux = static const char *max98088_dai1_fltr[] = { "Off", "fc=258/fs=16k", "fc=500/fs=16k", "fc=258/fs=8k", "fc=500/fs=8k", "fc=200"}; -static const struct soc_enum max98088_dai1_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 0, 6, max98088_dai1_fltr), -}; -static const struct soc_enum max98088_dai1_adc_filter_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 4, 6, max98088_dai1_fltr), -}; +static SOC_ENUM_SINGLE_DECL(max98088_dai1_dac_filter_enum, + M98088_REG_18_DAI1_FILTERS, 0, + max98088_dai1_fltr); +static SOC_ENUM_SINGLE_DECL(max98088_dai1_adc_filter_enum, + M98088_REG_18_DAI1_FILTERS, 4, + max98088_dai1_fltr); static int max98088_mic1pre_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1849,7 +1848,7 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec) /* Now point the soc_enum to .texts array items */ max98088->eq_enum.texts = max98088->eq_texts; - max98088->eq_enum.max = max98088->eq_textcnt; + max98088->eq_enum.items = max98088->eq_textcnt; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) @@ -1915,12 +1914,6 @@ static int max98088_probe(struct snd_soc_codec *codec) regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* initialize private data */ max98088->sysclk = (unsigned)-1; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d16b41..98c6e104357c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -336,6 +336,7 @@ static bool max98090_readable_register(struct device *dev, unsigned int reg) case M98090_REG_RECORD_TDM_SLOT: case M98090_REG_SAMPLE_RATE: case M98090_REG_DMIC34_BIQUAD_BASE ... M98090_REG_DMIC34_BIQUAD_BASE + 0x0E: + case M98090_REG_REVISION_ID: return true; default: return false; @@ -512,65 +513,75 @@ static const char *max98090_perf_pwr_text[] = static const char *max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; -static const struct soc_enum max98090_vcmbandgap_enum = - SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, - ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); +static SOC_ENUM_SINGLE_DECL(max98090_vcmbandgap_enum, + M98090_REG_BIAS_CONTROL, + M98090_VCM_MODE_SHIFT, + max98090_pwr_perf_text); static const char *max98090_osr128_text[] = { "64*fs", "128*fs" }; -static const struct soc_enum max98090_osr128_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, - ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); +static SOC_ENUM_SINGLE_DECL(max98090_osr128_enum, + M98090_REG_ADC_CONTROL, + M98090_OSR128_SHIFT, + max98090_osr128_text); static const char *max98090_mode_text[] = { "Voice", "Music" }; -static const struct soc_enum max98090_mode_enum = - SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT, - ARRAY_SIZE(max98090_mode_text), max98090_mode_text); +static SOC_ENUM_SINGLE_DECL(max98090_mode_enum, + M98090_REG_FILTER_CONFIG, + M98090_MODE_SHIFT, + max98090_mode_text); -static const struct soc_enum max98090_filter_dmic34mode_enum = - SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, - M98090_FLT_DMIC34MODE_SHIFT, - ARRAY_SIZE(max98090_mode_text), max98090_mode_text); +static SOC_ENUM_SINGLE_DECL(max98090_filter_dmic34mode_enum, + M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34MODE_SHIFT, + max98090_mode_text); static const char *max98090_drcatk_text[] = { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; -static const struct soc_enum max98090_drcatk_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, - ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); +static SOC_ENUM_SINGLE_DECL(max98090_drcatk_enum, + M98090_REG_DRC_TIMING, + M98090_DRCATK_SHIFT, + max98090_drcatk_text); static const char *max98090_drcrls_text[] = { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; -static const struct soc_enum max98090_drcrls_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, - ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); +static SOC_ENUM_SINGLE_DECL(max98090_drcrls_enum, + M98090_REG_DRC_TIMING, + M98090_DRCRLS_SHIFT, + max98090_drcrls_text); static const char *max98090_alccmp_text[] = { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; -static const struct soc_enum max98090_alccmp_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, - ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); +static SOC_ENUM_SINGLE_DECL(max98090_alccmp_enum, + M98090_REG_DRC_COMPRESSOR, + M98090_DRCCMP_SHIFT, + max98090_alccmp_text); static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; -static const struct soc_enum max98090_drcexp_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, - ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text); +static SOC_ENUM_SINGLE_DECL(max98090_drcexp_enum, + M98090_REG_DRC_EXPANDER, + M98090_DRCEXP_SHIFT, + max98090_drcexp_text); -static const struct soc_enum max98090_dac_perfmode_enum = - SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT, - ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); +static SOC_ENUM_SINGLE_DECL(max98090_dac_perfmode_enum, + M98090_REG_DAC_CONTROL, + M98090_PERFMODE_SHIFT, + max98090_perf_pwr_text); -static const struct soc_enum max98090_dachp_enum = - SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT, - ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); +static SOC_ENUM_SINGLE_DECL(max98090_dachp_enum, + M98090_REG_DAC_CONTROL, + M98090_DACHP_SHIFT, + max98090_pwr_perf_text); -static const struct soc_enum max98090_adchp_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT, - ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); +static SOC_ENUM_SINGLE_DECL(max98090_adchp_enum, + M98090_REG_ADC_CONTROL, + M98090_ADCHP_SHIFT, + max98090_pwr_perf_text); static const struct snd_kcontrol_new max98090_snd_controls[] = { SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), @@ -841,39 +852,42 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, static const char *mic1_mux_text[] = { "IN12", "IN56" }; -static const struct soc_enum mic1_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT, - ARRAY_SIZE(mic1_mux_text), mic1_mux_text); +static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, + M98090_REG_INPUT_MODE, + M98090_EXTMIC1_SHIFT, + mic1_mux_text); static const struct snd_kcontrol_new max98090_mic1_mux = SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); static const char *mic2_mux_text[] = { "IN34", "IN56" }; -static const struct soc_enum mic2_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT, - ARRAY_SIZE(mic2_mux_text), mic2_mux_text); +static SOC_ENUM_SINGLE_DECL(mic2_mux_enum, + M98090_REG_INPUT_MODE, + M98090_EXTMIC2_SHIFT, + mic2_mux_text); static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); static const char *dmic_mux_text[] = { "ADC", "DMIC" }; -static const struct soc_enum dmic_mux_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_mux_text), dmic_mux_text); +static SOC_ENUM_SINGLE_VIRT_DECL(dmic_mux_enum, dmic_mux_text); static const struct snd_kcontrol_new max98090_dmic_mux = SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum); static const char *max98090_micpre_text[] = { "Off", "On" }; -static const struct soc_enum max98090_pa1en_enum = - SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, - ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); +static SOC_ENUM_SINGLE_DECL(max98090_pa1en_enum, + M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PA1EN_SHIFT, + max98090_micpre_text); -static const struct soc_enum max98090_pa2en_enum = - SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, - ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); +static SOC_ENUM_SINGLE_DECL(max98090_pa2en_enum, + M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PA2EN_SHIFT, + max98090_micpre_text); /* LINEA mixer switch */ static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { @@ -937,13 +951,15 @@ static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; -static const struct soc_enum ltenl_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, - ARRAY_SIZE(lten_mux_text), lten_mux_text); +static SOC_ENUM_SINGLE_DECL(ltenl_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LTEN_SHIFT, + lten_mux_text); -static const struct soc_enum ltenr_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, - ARRAY_SIZE(lten_mux_text), lten_mux_text); +static SOC_ENUM_SINGLE_DECL(ltenr_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LTEN_SHIFT, + lten_mux_text); static const struct snd_kcontrol_new max98090_ltenl_mux = SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); @@ -953,13 +969,15 @@ static const struct snd_kcontrol_new max98090_ltenr_mux = static const char *lben_mux_text[] = { "Normal", "Loopback" }; -static const struct soc_enum lbenl_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, - ARRAY_SIZE(lben_mux_text), lben_mux_text); +static SOC_ENUM_SINGLE_DECL(lbenl_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LBEN_SHIFT, + lben_mux_text); -static const struct soc_enum lbenr_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, - ARRAY_SIZE(lben_mux_text), lben_mux_text); +static SOC_ENUM_SINGLE_DECL(lbenr_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LBEN_SHIFT, + lben_mux_text); static const struct snd_kcontrol_new max98090_lbenl_mux = SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); @@ -971,13 +989,15 @@ static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" }; -static const struct soc_enum stenl_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT, - ARRAY_SIZE(stenl_mux_text), stenl_mux_text); +static SOC_ENUM_SINGLE_DECL(stenl_mux_enum, + M98090_REG_ADC_SIDETONE, + M98090_DSTSL_SHIFT, + stenl_mux_text); -static const struct soc_enum stenr_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT, - ARRAY_SIZE(stenr_mux_text), stenr_mux_text); +static SOC_ENUM_SINGLE_DECL(stenr_mux_enum, + M98090_REG_ADC_SIDETONE, + M98090_DSTSR_SHIFT, + stenr_mux_text); static const struct snd_kcontrol_new max98090_stenl_mux = SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum); @@ -1085,9 +1105,10 @@ static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; -static const struct soc_enum linmod_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT, - ARRAY_SIZE(linmod_mux_text), linmod_mux_text); +static SOC_ENUM_SINGLE_DECL(linmod_mux_enum, + M98090_REG_LOUTR_MIXER, + M98090_LINMOD_SHIFT, + linmod_mux_text); static const struct snd_kcontrol_new max98090_linmod_mux = SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); @@ -1097,16 +1118,18 @@ static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; /* * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable */ -static const struct soc_enum mixhplsel_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT, - ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); +static SOC_ENUM_SINGLE_DECL(mixhplsel_mux_enum, + M98090_REG_HP_CONTROL, + M98090_MIXHPLSEL_SHIFT, + mixhpsel_mux_text); static const struct snd_kcontrol_new max98090_mixhplsel_mux = SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); -static const struct soc_enum mixhprsel_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, - ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); +static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum, + M98090_REG_HP_CONTROL, + M98090_MIXHPRSEL_SHIFT, + mixhpsel_mux_text); static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); @@ -1769,16 +1792,6 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { - ret = regcache_sync(max98090->regmap); - - if (ret != 0) { - dev_err(codec->dev, - "Failed to sync cache: %d\n", ret); - return ret; - } - } - if (max98090->jack_state == M98090_JACK_STATE_HEADSET) { /* * Set to normal bias level. @@ -1792,6 +1805,16 @@ static int max98090_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regcache_sync(max98090->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: /* Set internal pull-up to lowest power mode */ snd_soc_update_bits(codec, M98090_REG_JACK_DETECT, @@ -2195,14 +2218,6 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->codec = codec; - codec->control_data = max98090->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Reset the codec, the DSP core, and disable all interrupts */ max98090_reset(max98090); @@ -2328,7 +2343,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->devtype = id->driver_data; i2c_set_clientdata(i2c, max98090); - max98090->control_data = i2c; max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f249053..1a4e2334a7b2 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1523,7 +1523,6 @@ struct max98090_priv { struct regmap *regmap; struct snd_soc_codec *codec; enum max98090_type devtype; - void *control_data; struct max98090_pdata *pdata; unsigned int sysclk; unsigned int bclk; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 3ba1170ebb53..03f0536e6f61 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -560,25 +560,27 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, } static const char * const max98095_fltr_mode[] = { "Voice", "Music" }; -static const struct soc_enum max98095_dai1_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode), -}; -static const struct soc_enum max98095_dai2_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode), -}; +static SOC_ENUM_SINGLE_DECL(max98095_dai1_filter_mode_enum, + M98095_02E_DAI1_FILTERS, 7, + max98095_fltr_mode); +static SOC_ENUM_SINGLE_DECL(max98095_dai2_filter_mode_enum, + M98095_038_DAI2_FILTERS, 7, + max98095_fltr_mode); static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" }; -static const struct soc_enum max98095_extmic_enum = - SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text); +static SOC_ENUM_SINGLE_DECL(max98095_extmic_enum, + M98095_087_CFG_MIC, 0, + max98095_extmic_text); static const struct snd_kcontrol_new max98095_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum); static const char * const max98095_linein_text[] = { "INA", "INB" }; -static const struct soc_enum max98095_linein_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text); +static SOC_ENUM_SINGLE_DECL(max98095_linein_enum, + M98095_086_CFG_LINE, 6, + max98095_linein_text); static const struct snd_kcontrol_new max98095_linein_mux = SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum); @@ -586,24 +588,26 @@ static const struct snd_kcontrol_new max98095_linein_mux = static const char * const max98095_line_mode_text[] = { "Stereo", "Differential"}; -static const struct soc_enum max98095_linein_mode_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text); +static SOC_ENUM_SINGLE_DECL(max98095_linein_mode_enum, + M98095_086_CFG_LINE, 7, + max98095_line_mode_text); -static const struct soc_enum max98095_lineout_mode_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text); +static SOC_ENUM_SINGLE_DECL(max98095_lineout_mode_enum, + M98095_086_CFG_LINE, 4, + max98095_line_mode_text); static const char * const max98095_dai_fltr[] = { "Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k", "Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"}; -static const struct soc_enum max98095_dai1_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 0, 6, max98095_dai_fltr), -}; -static const struct soc_enum max98095_dai2_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 0, 6, max98095_dai_fltr), -}; -static const struct soc_enum max98095_dai3_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_042_DAI3_FILTERS, 0, 6, max98095_dai_fltr), -}; +static SOC_ENUM_SINGLE_DECL(max98095_dai1_dac_filter_enum, + M98095_02E_DAI1_FILTERS, 0, + max98095_dai_fltr); +static SOC_ENUM_SINGLE_DECL(max98095_dai2_dac_filter_enum, + M98095_038_DAI2_FILTERS, 0, + max98095_dai_fltr); +static SOC_ENUM_SINGLE_DECL(max98095_dai3_dac_filter_enum, + M98095_042_DAI3_FILTERS, 0, + max98095_dai_fltr); static int max98095_mic1pre_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1861,7 +1865,7 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) /* Now point the soc_enum to .texts array items */ max98095->eq_enum.texts = max98095->eq_texts; - max98095->eq_enum.max = max98095->eq_textcnt; + max98095->eq_enum.items = max98095->eq_textcnt; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) @@ -2016,7 +2020,7 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) /* Now point the soc_enum to .texts array items */ max98095->bq_enum.texts = max98095->bq_texts; - max98095->bq_enum.max = max98095->bq_textcnt; + max98095->bq_enum.items = max98095->bq_textcnt; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) @@ -2234,12 +2238,6 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 82757ebf0301..4fdf5aaa236f 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -312,14 +312,6 @@ static int max9850_resume(struct snd_soc_codec *codec) static int max9850_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* enable zero-detect */ snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); /* enable slew-rate control */ diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bbd42cb..2c59b1fb69dc 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -106,8 +106,7 @@ static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); int i; @@ -126,8 +125,7 @@ static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); unsigned int val; @@ -408,8 +406,7 @@ static const char * const adcl_enum_text[] = { "MC1L", "RXINL", }; -static const struct soc_enum adcl_enum = - SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adcl_enum, adcl_enum_text); static const struct snd_kcontrol_new left_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcl_enum); @@ -418,8 +415,7 @@ static const char * const adcr_enum_text[] = { "MC1R", "MC2", "RXINR", "TXIN", }; -static const struct soc_enum adcr_enum = - SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adcr_enum, adcr_enum_text); static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); @@ -430,8 +426,8 @@ static const struct snd_kcontrol_new samp_ctl = static const char * const speaker_amp_source_text[] = { "CODEC", "Right" }; -static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, - speaker_amp_source_text); +static SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, + speaker_amp_source_text); static const struct snd_kcontrol_new speaker_amp_source_mux = SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); @@ -439,8 +435,8 @@ static const char * const headset_amp_source_text[] = { "CODEC", "Mixer" }; -static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, - headset_amp_source_text); +static SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, + headset_amp_source_text); static const struct snd_kcontrol_new headset_amp_source_mux = SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); @@ -580,9 +576,9 @@ static struct snd_soc_dapm_route mc13783_routes[] = { static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", "Mono", "Mono Mix"}; -static const struct soc_enum mc13783_enum_3d_mixer = - SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), - mc13783_3d_mixer); +static SOC_ENUM_SINGLE_DECL(mc13783_enum_3d_mixer, + MC13783_AUDIO_RX1, 16, + mc13783_3d_mixer); static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), @@ -614,8 +610,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, + dev_get_regmap(codec->dev->parent, NULL)); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 185fa3bc3052..e661e8420e3d 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -73,11 +73,11 @@ static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); static const char * const ml26124_companding[] = {"16bit PCM", "u-law", "A-law"}; -static const struct soc_enum ml26124_adc_companding_enum - = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static SOC_ENUM_SINGLE_DECL(ml26124_adc_companding_enum, + ML26124_SAI_TRANS_CTL, 6, ml26124_companding); -static const struct soc_enum ml26124_dac_companding_enum - = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); +static SOC_ENUM_SINGLE_DECL(ml26124_dac_companding_enum, + ML26124_SAI_RCV_CTL, 6, ml26124_companding); static const struct snd_kcontrol_new ml26124_snd_controls[] = { SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, @@ -136,8 +136,8 @@ static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", "Digital MIC in", "Analog MIC Differential in"}; -static const struct soc_enum ml26124_insel_enum = - SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select); +static SOC_ENUM_SINGLE_DECL(ml26124_insel_enum, + ML26124_MIC_IF_CTL, 0, ml26124_input_select); static const struct snd_kcontrol_new ml26124_input_mux_controls = SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); @@ -586,16 +586,6 @@ static int ml26124_resume(struct snd_soc_codec *codec) static int ml26124_probe(struct snd_soc_codec *codec) { - int ret; - struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); - codec->control_data = priv->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Software Reset */ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 73f9c3630e2c..e427544183d7 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -172,16 +172,21 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); int val = 0, ret; - int pcm_format = params_format(params); priv->rate = params_rate(params); switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) - val = 0x00; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x03; + switch (params_width(params)) { + case 24: + val = 0; + break; + case 16: + val = 3; + break; + default: + return -EINVAL; + } break; case SND_SOC_DAIFMT_I2S: val = 0x04; diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 7146653a8e16..3a80ba4452df 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -107,24 +107,35 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); int val = 0, ret; - int pcm_format = params_format(params); priv->rate = params_rate(params); switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || - pcm_format == SNDRV_PCM_FORMAT_S32_LE) - val = 0x02; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x00; + switch (params_width(params)) { + case 24: + case 32: + val = 2; + break; + case 16: + val = 0; + break; + default: + return -EINVAL; + } break; case SND_SOC_DAIFMT_I2S: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || - pcm_format == SNDRV_PCM_FORMAT_S32_LE) - val = 0x05; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x04; + switch (params_width(params)) { + case 24: + case 32: + val = 5; + break; + case 16: + val = 4; + break; + default: + return -EINVAL; + } break; default: dev_err(codec->dev, "Invalid DAI format\n"); diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c new file mode 100644 index 000000000000..4d62230bd378 --- /dev/null +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -0,0 +1,71 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie@linaro.org> + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/i2c.h> + +#include "pcm512x.h" + +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_i2c_driver(pcm512x_i2c_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - I2C"); +MODULE_AUTHOR("Mark Brown <broonie@linaro.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c new file mode 100644 index 000000000000..f297058c0038 --- /dev/null +++ b/sound/soc/codecs/pcm512x-spi.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie@linaro.org> + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/spi/spi.h> + +#include "pcm512x.h" + +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_spi_driver(pcm512x_spi_driver); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c new file mode 100644 index 000000000000..4b4c0c7bb918 --- /dev/null +++ b/sound/soc/codecs/pcm512x.c @@ -0,0 +1,589 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie@linaro.org> + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "pcm512x.h" + +#define PCM512x_NUM_SUPPLIES 3 +static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "CPVDD", +}; + +struct pcm512x_priv { + struct regmap *regmap; + struct clk *sclk; + struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES]; + struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES]; +}; + +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define PCM512x_REGULATOR_EVENT(n) \ +static int pcm512x_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \ + supply_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + regcache_mark_dirty(pcm512x->regmap); \ + regcache_cache_only(pcm512x->regmap, true); \ + } \ + return 0; \ +} + +PCM512x_REGULATOR_EVENT(0) +PCM512x_REGULATOR_EVENT(1) +PCM512x_REGULATOR_EVENT(2) + +static const struct reg_default pcm512x_reg_defaults[] = { + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_OUTPUT_AMPLITUDE, 0x00 }, + { PCM512x_ANALOG_GAIN_CTRL, 0x00 }, + { PCM512x_UNDERVOLTAGE_PROT, 0x00 }, + { PCM512x_ANALOG_MUTE_CTRL, 0x00 }, + { PCM512x_ANALOG_GAIN_BOOST, 0x00 }, + { PCM512x_VCOM_CTRL_1, 0x00 }, + { PCM512x_VCOM_CTRL_2, 0x01 }, +}; + +static bool pcm512x_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_RESET: + case PCM512x_POWER: + case PCM512x_MUTE: + case PCM512x_PLL_EN: + case PCM512x_SPI_MISO_FUNCTION: + case PCM512x_DSP: + case PCM512x_GPIO_EN: + case PCM512x_BCLK_LRCLK_CFG: + case PCM512x_DSP_GPIO_INPUT: + case PCM512x_MASTER_MODE: + case PCM512x_PLL_REF: + case PCM512x_PLL_COEFF_0: + case PCM512x_PLL_COEFF_1: + case PCM512x_PLL_COEFF_2: + case PCM512x_PLL_COEFF_3: + case PCM512x_PLL_COEFF_4: + case PCM512x_DSP_CLKDIV: + case PCM512x_DAC_CLKDIV: + case PCM512x_NCP_CLKDIV: + case PCM512x_OSR_CLKDIV: + case PCM512x_MASTER_CLKDIV_1: + case PCM512x_MASTER_CLKDIV_2: + case PCM512x_FS_SPEED_MODE: + case PCM512x_IDAC_1: + case PCM512x_IDAC_2: + case PCM512x_ERROR_DETECT: + case PCM512x_I2S_1: + case PCM512x_I2S_2: + case PCM512x_DAC_ROUTING: + case PCM512x_DSP_PROGRAM: + case PCM512x_CLKDET: + case PCM512x_AUTO_MUTE: + case PCM512x_DIGITAL_VOLUME_1: + case PCM512x_DIGITAL_VOLUME_2: + case PCM512x_DIGITAL_VOLUME_3: + case PCM512x_DIGITAL_MUTE_1: + case PCM512x_DIGITAL_MUTE_2: + case PCM512x_DIGITAL_MUTE_3: + case PCM512x_GPIO_OUTPUT_1: + case PCM512x_GPIO_OUTPUT_2: + case PCM512x_GPIO_OUTPUT_3: + case PCM512x_GPIO_OUTPUT_4: + case PCM512x_GPIO_OUTPUT_5: + case PCM512x_GPIO_OUTPUT_6: + case PCM512x_GPIO_CONTROL_1: + case PCM512x_GPIO_CONTROL_2: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_OUTPUT_AMPLITUDE: + case PCM512x_ANALOG_GAIN_CTRL: + case PCM512x_UNDERVOLTAGE_PROT: + case PCM512x_ANALOG_MUTE_CTRL: + case PCM512x_ANALOG_GAIN_BOOST: + case PCM512x_VCOM_CTRL_1: + case PCM512x_VCOM_CTRL_2: + case PCM512x_CRAM_CTRL: + return true; + default: + /* There are 256 raw register addresses */ + return reg < 0xff; + } +} + +static bool pcm512x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_PLL_EN: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_CRAM_CTRL: + return true; + default: + /* There are 256 raw register addresses */ + return reg < 0xff; + } +} + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); +static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); + +static const char * const pcm512x_dsp_program_texts[] = { + "FIR interpolation with de-emphasis", + "Low latency IIR with de-emphasis", + "Fixed process flow", + "High attenuation with de-emphasis", + "Ringing-less low latency FIR", +}; + +static const unsigned int pcm512x_dsp_program_values[] = { + 1, + 2, + 3, + 5, + 7, +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); + +static const char * const pcm512x_clk_missing_text[] = { + "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" +}; + +static const struct soc_enum pcm512x_clk_missing = + SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text); + +static const char * const pcm512x_autom_text[] = { + "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" +}; + +static const struct soc_enum pcm512x_autom_l = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 8, + pcm512x_autom_text); + +static const struct soc_enum pcm512x_autom_r = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8, + pcm512x_autom_text); + +static const char * const pcm512x_ramp_rate_text[] = { + "1 sample/update", "2 samples/update", "4 samples/update", + "Immediate" +}; + +static const struct soc_enum pcm512x_vndf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vnuf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vedf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const char * const pcm512x_ramp_step_text[] = { + "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" +}; + +static const struct soc_enum pcm512x_vnds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_vnus = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_veds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct snd_kcontrol_new pcm512x_controls[] = { +SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, + PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, + PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), +SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, + PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), +SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, + PCM512x_RQMR_SHIFT, 1, 1), + +SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), +SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program), + +SOC_ENUM("Clock Missing Period", pcm512x_clk_missing), +SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l), +SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r), +SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3, + PCM512x_ACTL_SHIFT, 1, 0), +SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT, + PCM512x_AMLR_SHIFT, 1, 0), + +SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf), +SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds), +SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), +SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), +SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), +SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), +}; + +static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { + { "DACL", NULL, "Playback" }, + { "DACR", NULL, "Playback" }, + + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +static int pcm512x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to remove standby: %d\n", + ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(codec->dev, "Failed to request standby: %d\n", + ret); + return ret; + } + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_dai_driver pcm512x_dai = { + .name = "pcm512x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver pcm512x_codec_driver = { + .set_bias_level = pcm512x_set_bias_level, + .idle_bias_off = true, + + .controls = pcm512x_controls, + .num_controls = ARRAY_SIZE(pcm512x_controls), + .dapm_widgets = pcm512x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets), + .dapm_routes = pcm512x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), +}; + +static const struct regmap_range_cfg pcm512x_range = { + .name = "Pages", .range_min = PCM512x_VIRT_BASE, + .range_max = PCM512x_MAX_REGISTER, + .selector_reg = PCM512x_PAGE, + .selector_mask = 0xff, + .window_start = 0, .window_len = 0x100, +}; + +const struct regmap_config pcm512x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = pcm512x_readable, + .volatile_reg = pcm512x_volatile, + + .ranges = &pcm512x_range, + .num_ranges = 1, + + .max_register = PCM512x_MAX_REGISTER, + .reg_defaults = pcm512x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(pcm512x_regmap); + +int pcm512x_probe(struct device *dev, struct regmap *regmap) +{ + struct pcm512x_priv *pcm512x; + int i, ret; + + pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL); + if (!pcm512x) + return -ENOMEM; + + dev_set_drvdata(dev, pcm512x); + pcm512x->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) + pcm512x->supplies[i].supply = pcm512x_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0; + pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1; + pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) { + ret = regulator_register_notifier(pcm512x->supplies[i].consumer, + &pcm512x->supply_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the device, verifying I/O in the process for I2C */ + ret = regmap_write(regmap, PCM512x_RESET, + PCM512x_RSTM | PCM512x_RSTR); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + ret = regmap_write(regmap, PCM512x_RESET, 0); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + pcm512x->sclk = devm_clk_get(dev, NULL); + if (IS_ERR(pcm512x->sclk)) { + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + dev_info(dev, "No SCLK, using BCLK: %ld\n", + PTR_ERR(pcm512x->sclk)); + + /* Disable reporting of missing SCLK as an error */ + regmap_update_bits(regmap, PCM512x_ERROR_DETECT, + PCM512x_IDCH, PCM512x_IDCH); + + /* Switch PLL input to BCLK */ + regmap_update_bits(regmap, PCM512x_PLL_REF, + PCM512x_SREF, PCM512x_SREF); + } else { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + /* Default to standby mode */ + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(dev, "Failed to request standby: %d\n", + ret); + goto err_clk; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &pcm512x_codec_driver, + &pcm512x_dai, 1); + if (ret != 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_clk: + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); +err: + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + return ret; +} +EXPORT_SYMBOL_GPL(pcm512x_probe); + +void pcm512x_remove(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); +} +EXPORT_SYMBOL_GPL(pcm512x_remove); + +static int pcm512x_suspend(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, PCM512x_RQPD); + if (ret != 0) { + dev_err(dev, "Failed to request power down: %d\n", ret); + return ret; + } + + ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to disable supplies: %d\n", ret); + return ret; + } + + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + + return 0; +} + +static int pcm512x_resume(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(pcm512x->sclk)) { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(pcm512x->regmap, false); + ret = regcache_sync(pcm512x->regmap); + if (ret != 0) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, 0); + if (ret != 0) { + dev_err(dev, "Failed to remove power down: %d\n", ret); + return ret; + } + + return 0; +} + +const struct dev_pm_ops pcm512x_pm_ops = { + SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) +}; +EXPORT_SYMBOL_GPL(pcm512x_pm_ops); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver"); +MODULE_AUTHOR("Mark Brown <broonie@linaro.org>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h new file mode 100644 index 000000000000..6ee76aaca09a --- /dev/null +++ b/sound/soc/codecs/pcm512x.h @@ -0,0 +1,171 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown <broonie@linaro.org> + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef _SND_SOC_PCM512X +#define _SND_SOC_PCM512X + +#include <linux/pm.h> +#include <linux/regmap.h> + +#define PCM512x_VIRT_BASE 0x100 +#define PCM512x_PAGE_LEN 0x100 +#define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) + +#define PCM512x_PAGE 0 + +#define PCM512x_RESET (PCM512x_PAGE_BASE(0) + 1) +#define PCM512x_POWER (PCM512x_PAGE_BASE(0) + 2) +#define PCM512x_MUTE (PCM512x_PAGE_BASE(0) + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_BASE(0) + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_BASE(0) + 6) +#define PCM512x_DSP (PCM512x_PAGE_BASE(0) + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_BASE(0) + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_BASE(0) + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_BASE(0) + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_BASE(0) + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_BASE(0) + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_BASE(0) + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_BASE(0) + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_BASE(0) + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_BASE(0) + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_BASE(0) + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_BASE(0) + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_BASE(0) + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_BASE(0) + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_BASE(0) + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_BASE(0) + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_BASE(0) + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_BASE(0) + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_BASE(0) + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_BASE(0) + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_BASE(0) + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_BASE(0) + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_BASE(0) + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_BASE(0) + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_BASE(0) + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_BASE(0) + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_BASE(0) + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_BASE(0) + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_BASE(0) + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_BASE(0) + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_BASE(0) + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_BASE(0) + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_BASE(0) + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_BASE(0) + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_BASE(0) + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_BASE(0) + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_BASE(0) + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108) +#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120) + +#define PCM512x_OUTPUT_AMPLITUDE (PCM512x_PAGE_BASE(1) + 1) +#define PCM512x_ANALOG_GAIN_CTRL (PCM512x_PAGE_BASE(1) + 2) +#define PCM512x_UNDERVOLTAGE_PROT (PCM512x_PAGE_BASE(1) + 5) +#define PCM512x_ANALOG_MUTE_CTRL (PCM512x_PAGE_BASE(1) + 6) +#define PCM512x_ANALOG_GAIN_BOOST (PCM512x_PAGE_BASE(1) + 7) +#define PCM512x_VCOM_CTRL_1 (PCM512x_PAGE_BASE(1) + 8) +#define PCM512x_VCOM_CTRL_2 (PCM512x_PAGE_BASE(1) + 9) + +#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1) + +/* Page 0, Register 1 - reset */ +#define PCM512x_RSTR (1 << 0) +#define PCM512x_RSTM (1 << 4) + +/* Page 0, Register 2 - power */ +#define PCM512x_RQPD (1 << 0) +#define PCM512x_RQPD_SHIFT 0 +#define PCM512x_RQST (1 << 4) +#define PCM512x_RQST_SHIFT 4 + +/* Page 0, Register 3 - mute */ +#define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML_SHIFT 4 + +/* Page 0, Register 4 - PLL */ +#define PCM512x_PLCE (1 << 0) +#define PCM512x_RLCE_SHIFT 0 +#define PCM512x_PLCK (1 << 4) +#define PCM512x_PLCK_SHIFT 4 + +/* Page 0, Register 7 - DSP */ +#define PCM512x_SDSL (1 << 0) +#define PCM512x_SDSL_SHIFT 0 +#define PCM512x_DEMP (1 << 4) +#define PCM512x_DEMP_SHIFT 4 + +/* Page 0, Register 13 - PLL reference */ +#define PCM512x_SREF (1 << 4) + +/* Page 0, Register 37 - Error detection */ +#define PCM512x_IPLK (1 << 0) +#define PCM512x_DCAS (1 << 1) +#define PCM512x_IDCM (1 << 2) +#define PCM512x_IDCH (1 << 3) +#define PCM512x_IDSK (1 << 4) +#define PCM512x_IDBK (1 << 5) +#define PCM512x_IDFS (1 << 6) + +/* Page 0, Register 42 - DAC routing */ +#define PCM512x_AUPR_SHIFT 0 +#define PCM512x_AUPL_SHIFT 4 + +/* Page 0, Register 59 - auto mute */ +#define PCM512x_ATMR_SHIFT 0 +#define PCM512x_ATML_SHIFT 4 + +/* Page 0, Register 63 - ramp rates */ +#define PCM512x_VNDF_SHIFT 6 +#define PCM512x_VNDS_SHIFT 4 +#define PCM512x_VNUF_SHIFT 2 +#define PCM512x_VNUS_SHIFT 0 + +/* Page 0, Register 64 - emergency ramp rates */ +#define PCM512x_VEDF_SHIFT 6 +#define PCM512x_VEDS_SHIFT 4 + +/* Page 0, Register 65 - Digital mute enables */ +#define PCM512x_ACTL_SHIFT 2 +#define PCM512x_AMLE_SHIFT 1 +#define PCM512x_AMLR_SHIFT 0 + +/* Page 1, Register 2 - analog volume control */ +#define PCM512x_RAGN_SHIFT 0 +#define PCM512x_LAGN_SHIFT 4 + +/* Page 1, Register 7 - analog boost control */ +#define PCM512x_AGBR_SHIFT 0 +#define PCM512x_AGBL_SHIFT 4 + +extern const struct dev_pm_ops pcm512x_pm_ops; +extern const struct regmap_config pcm512x_regmap; + +int pcm512x_probe(struct device *dev, struct regmap *regmap); +void pcm512x_remove(struct device *dev); + +#endif diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 912c9cbc2724..d4c229f0233f 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -210,26 +210,22 @@ static int rt5631_dmic_put(struct snd_kcontrol *kcontrol, static const char *rt5631_input_mode[] = { "Single ended", "Differential"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1, - RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode); +static SOC_ENUM_SINGLE_DECL(rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode); -static const SOC_ENUM_SINGLE_DECL( - rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1, - RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode); +static SOC_ENUM_SINGLE_DECL(rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode); /* MONO Input Type */ -static const SOC_ENUM_SINGLE_DECL( - rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL, - RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode); +static SOC_ENUM_SINGLE_DECL(rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL, + RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode); /* SPK Ratio Gain Control */ static const char *rt5631_spk_ratio[] = {"1.00x", "1.09x", "1.27x", "1.44x", "1.56x", "1.68x", "1.99x", "2.34x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, - RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio); +static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, + RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio); static const struct snd_kcontrol_new rt5631_snd_controls[] = { /* MIC */ @@ -759,9 +755,8 @@ static const struct snd_kcontrol_new rt5631_monomix_mixer_controls[] = { /* Left SPK Volume Input */ static const char *rt5631_spkvoll_sel[] = {"Vmid", "SPKMIXL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL, - RT5631_L_EN_SHIFT, rt5631_spkvoll_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_spkvoll_sel); static const struct snd_kcontrol_new rt5631_spkvoll_mux_control = SOC_DAPM_ENUM("Left SPKVOL SRC", rt5631_spkvoll_enum); @@ -769,9 +764,8 @@ static const struct snd_kcontrol_new rt5631_spkvoll_mux_control = /* Left HP Volume Input */ static const char *rt5631_hpvoll_sel[] = {"Vmid", "OUTMIXL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpvoll_enum, RT5631_HP_OUT_VOL, - RT5631_L_EN_SHIFT, rt5631_hpvoll_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpvoll_enum, RT5631_HP_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_hpvoll_sel); static const struct snd_kcontrol_new rt5631_hpvoll_mux_control = SOC_DAPM_ENUM("Left HPVOL SRC", rt5631_hpvoll_enum); @@ -779,9 +773,8 @@ static const struct snd_kcontrol_new rt5631_hpvoll_mux_control = /* Left Out Volume Input */ static const char *rt5631_outvoll_sel[] = {"Vmid", "OUTMIXL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL, - RT5631_L_EN_SHIFT, rt5631_outvoll_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_L_EN_SHIFT, rt5631_outvoll_sel); static const struct snd_kcontrol_new rt5631_outvoll_mux_control = SOC_DAPM_ENUM("Left OUTVOL SRC", rt5631_outvoll_enum); @@ -789,9 +782,8 @@ static const struct snd_kcontrol_new rt5631_outvoll_mux_control = /* Right Out Volume Input */ static const char *rt5631_outvolr_sel[] = {"Vmid", "OUTMIXR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL, - RT5631_R_EN_SHIFT, rt5631_outvolr_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_R_EN_SHIFT, rt5631_outvolr_sel); static const struct snd_kcontrol_new rt5631_outvolr_mux_control = SOC_DAPM_ENUM("Right OUTVOL SRC", rt5631_outvolr_enum); @@ -799,9 +791,8 @@ static const struct snd_kcontrol_new rt5631_outvolr_mux_control = /* Right HP Volume Input */ static const char *rt5631_hpvolr_sel[] = {"Vmid", "OUTMIXR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpvolr_enum, RT5631_HP_OUT_VOL, - RT5631_R_EN_SHIFT, rt5631_hpvolr_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpvolr_enum, RT5631_HP_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_hpvolr_sel); static const struct snd_kcontrol_new rt5631_hpvolr_mux_control = SOC_DAPM_ENUM("Right HPVOL SRC", rt5631_hpvolr_enum); @@ -809,9 +800,8 @@ static const struct snd_kcontrol_new rt5631_hpvolr_mux_control = /* Right SPK Volume Input */ static const char *rt5631_spkvolr_sel[] = {"Vmid", "SPKMIXR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL, - RT5631_R_EN_SHIFT, rt5631_spkvolr_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_spkvolr_sel); static const struct snd_kcontrol_new rt5631_spkvolr_mux_control = SOC_DAPM_ENUM("Right SPKVOL SRC", rt5631_spkvolr_enum); @@ -820,9 +810,8 @@ static const struct snd_kcontrol_new rt5631_spkvolr_mux_control = static const char *rt5631_spol_src_sel[] = { "SPOLMIX", "MONOIN_RX", "VDAC", "DACL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel); static const struct snd_kcontrol_new rt5631_spol_mux_control = SOC_DAPM_ENUM("SPOL SRC", rt5631_spol_src_enum); @@ -831,9 +820,8 @@ static const struct snd_kcontrol_new rt5631_spol_mux_control = static const char *rt5631_spor_src_sel[] = { "SPORMIX", "MONOIN_RX", "VDAC", "DACR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel); static const struct snd_kcontrol_new rt5631_spor_mux_control = SOC_DAPM_ENUM("SPOR SRC", rt5631_spor_src_enum); @@ -841,9 +829,8 @@ static const struct snd_kcontrol_new rt5631_spor_mux_control = /* MONO Input */ static const char *rt5631_mono_src_sel[] = {"MONOMIX", "MONOIN_RX", "VDAC"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel); static const struct snd_kcontrol_new rt5631_mono_mux_control = SOC_DAPM_ENUM("MONO SRC", rt5631_mono_src_enum); @@ -851,9 +838,8 @@ static const struct snd_kcontrol_new rt5631_mono_mux_control = /* Left HPO Input */ static const char *rt5631_hpl_src_sel[] = {"Left HPVOL", "Left DAC"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel); static const struct snd_kcontrol_new rt5631_hpl_mux_control = SOC_DAPM_ENUM("HPL SRC", rt5631_hpl_src_enum); @@ -861,9 +847,8 @@ static const struct snd_kcontrol_new rt5631_hpl_mux_control = /* Right HPO Input */ static const char *rt5631_hpr_src_sel[] = {"Right HPVOL", "Right DAC"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel); static const struct snd_kcontrol_new rt5631_hpr_mux_control = SOC_DAPM_ENUM("HPR SRC", rt5631_hpr_src_enum); @@ -1585,15 +1570,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) { struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); unsigned int val; - int ret; - - codec->control_data = rt5631->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3); if (val & 0x0002) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..0061ae6b6716 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -361,25 +361,24 @@ static unsigned int bst_tlv[] = { static const char * const rt5640_data_select[] = { "Normal", "left copy to right", "right copy to left", "Swap"}; -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); /* Class D speaker gain ratio */ static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, - RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, + RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ @@ -753,9 +752,8 @@ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, - RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5640_stereo_adc1_enum); @@ -764,9 +762,8 @@ static const char * const rt5640_stereo_adc2_src[] = { "DMIC1", "DMIC2", "DIG MIX" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, - RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5640_stereo_adc2_enum); @@ -776,9 +773,8 @@ static const char * const rt5640_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADCL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = SOC_DAPM_ENUM("Mono ADC1 left source", rt5640_mono_adc_l1_enum); @@ -787,9 +783,8 @@ static const char * const rt5640_mono_adc_l2_src[] = { "DMIC L1", "DMIC L2", "Mono DAC MIXL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = SOC_DAPM_ENUM("Mono ADC2 left source", rt5640_mono_adc_l2_enum); @@ -798,9 +793,8 @@ static const char * const rt5640_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADCR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = SOC_DAPM_ENUM("Mono ADC1 right source", rt5640_mono_adc_r1_enum); @@ -809,9 +803,8 @@ static const char * const rt5640_mono_adc_r2_src[] = { "DMIC R1", "DMIC R2", "Mono DAC MIXR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = SOC_DAPM_ENUM("Mono ADC2 right source", rt5640_mono_adc_r2_enum); @@ -826,9 +819,9 @@ static int rt5640_dac_l2_values[] = { 3, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( - rt5640_dac_l2_enum, RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT, - 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_l2_enum, + RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT, + 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values); static const struct snd_kcontrol_new rt5640_dac_l2_mux = SOC_DAPM_VALUE_ENUM("DAC2 left channel source", rt5640_dac_l2_enum); @@ -841,9 +834,9 @@ static int rt5640_dac_r2_values[] = { 0, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( - rt5640_dac_r2_enum, RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT, - 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_r2_enum, + RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT, + 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values); static const struct snd_kcontrol_new rt5640_dac_r2_mux = SOC_DAPM_ENUM("DAC2 right channel source", rt5640_dac_r2_enum); @@ -860,9 +853,10 @@ static int rt5640_dai_iis_map_values[] = { 7, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( - rt5640_dai_iis_map_enum, RT5640_I2S1_SDP, RT5640_I2S_IF_SFT, - 0x7, rt5640_dai_iis_map, rt5640_dai_iis_map_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dai_iis_map_enum, + RT5640_I2S1_SDP, RT5640_I2S_IF_SFT, + 0x7, rt5640_dai_iis_map, + rt5640_dai_iis_map_values); static const struct snd_kcontrol_new rt5640_dai_mux = SOC_DAPM_VALUE_ENUM("DAI select", rt5640_dai_iis_map_enum); @@ -872,9 +866,8 @@ static const char * const rt5640_sdi_sel[] = { "IF1", "IF2" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_sdi_sel_enum, RT5640_I2S2_SDP, - RT5640_I2S2_SDI_SFT, rt5640_sdi_sel); +static SOC_ENUM_SINGLE_DECL(rt5640_sdi_sel_enum, RT5640_I2S2_SDP, + RT5640_I2S2_SDI_SFT, rt5640_sdi_sel); static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); @@ -1601,8 +1594,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; @@ -1943,16 +1935,8 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - int ret; rt5640->codec = codec; - codec->control_data = rt5640->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } codec->dapm.idle_bias_off = 1; rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); @@ -2093,6 +2077,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, + { "10EC5640", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0fcbe90f3ef2..d3ed1be5a186 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -187,8 +187,9 @@ static const char *adc_mux_text[] = { "MIC_IN", "LINE_IN" }; -static const struct soc_enum adc_enum = -SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); +static SOC_ENUM_SINGLE_DECL(adc_enum, + SGTL5000_CHIP_ANA_CTRL, 2, + adc_mux_text); static const struct snd_kcontrol_new adc_mux = SOC_DAPM_ENUM("Capture Mux", adc_enum); @@ -198,8 +199,9 @@ static const char *dac_mux_text[] = { "DAC", "LINE_IN" }; -static const struct soc_enum dac_enum = -SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); +static SOC_ENUM_SINGLE_DECL(dac_enum, + SGTL5000_CHIP_ANA_CTRL, 6, + dac_mux_text); static const struct snd_kcontrol_new dac_mux = SOC_DAPM_ENUM("Headphone Mux", dac_enum); @@ -1350,14 +1352,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) int ret; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - /* setup i2c data ops */ - codec->control_data = sgtl5000->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = sgtl5000_enable_regulators(codec); if (ret) return ret; diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 52e7cb08434b..244c097cd905 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -21,6 +21,7 @@ #include <linux/slab.h> #include <sound/pcm.h> #include <sound/pcm_params.h> +#include <linux/regmap.h> #include <sound/soc.h> #include <sound/initval.h> @@ -209,8 +210,9 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return 0; + struct regmap *regmap = dev_get_regmap(codec->dev->parent, NULL); + + return snd_soc_codec_set_cache_io(codec, regmap); } static struct snd_soc_dai_ops si476x_dai_ops = { diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c new file mode 100644 index 000000000000..58e7c1f23771 --- /dev/null +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -0,0 +1,524 @@ +/* + * SiRF audio codec driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/io.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "sirf-audio-codec.h" + +struct sirf_audio_codec { + struct clk *clk; + struct regmap *regmap; + u32 reg_ctrl0, reg_ctrl1; +}; + +static const char * const input_mode_mux[] = {"Single-ended", + "Differential"}; + +static const struct soc_enum input_mode_mux_enum = + SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux); + +static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control = + SOC_DAPM_ENUM("Route", input_mode_mux_enum); + +static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0); +static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0); +static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6, + 0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0), + 0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0), +); + +static struct snd_kcontrol_new volume_controls_atlas6[] = { + SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10, + 0x3F, 0, capture_vol_tlv_atlas6), +}; + +static struct snd_kcontrol_new volume_controls_prima2[] = { + SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10, + 0x1F, 0, capture_vol_tlv_prima2), +}; + +static struct snd_kcontrol_new left_input_path_controls[] = { + SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0), +}; + +static struct snd_kcontrol_new right_input_path_controls[] = { + SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0), +}; + +static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0); + +static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0); + +static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0); + +static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0); + +/* After enable adc, Delay 200ms to avoid pop noise */ +static int adc_enable_delay_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(200); + break; + default: + break; + } + + return 0; +} + +static void enable_and_reset_codec(struct regmap *regmap, + u32 codec_enable_bits, u32 codec_reset_bits) +{ + regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, + codec_enable_bits | codec_reset_bits, + codec_enable_bits | ~codec_reset_bits); + msleep(20); + regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, + codec_reset_bits, codec_reset_bits); +} + +static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ +#define ATLAS6_CODEC_ENABLE_BITS (1 << 29) +#define ATLAS6_CODEC_RESET_BITS (1 << 28) + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + enable_and_reset_codec(sirf_audio_codec->regmap, + ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, + ~ATLAS6_CODEC_ENABLE_BITS); + break; + default: + break; + } + + return 0; +} + +static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ +#define PRIMA2_CODEC_ENABLE_BITS (1 << 27) +#define PRIMA2_CODEC_RESET_BITS (1 << 26) + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + enable_and_reset_codec(sirf_audio_codec->regmap, + PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, + ~PRIMA2_CODEC_ENABLE_BITS); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1, + 25, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1, + 26, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1, + 27, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1, + 23, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1, + 24, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1, + 25, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget = + SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0, + atlas6_codec_enable_and_reset_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD); + +static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget = + SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0, + prima2_codec_enable_and_reset_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD); + +static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0), + SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0), + SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0), + + SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &left_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &right_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + + SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0, + adc_enable_delay_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0, + adc_enable_delay_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0, + &left_input_path_controls[0], + ARRAY_SIZE(left_input_path_controls)), + SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0, + &right_input_path_controls[0], + ARRAY_SIZE(right_input_path_controls)), + + SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0, + &sirf_audio_codec_input_mode_control), + SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0), + SND_SOC_DAPM_INPUT("MICIN1"), + SND_SOC_DAPM_INPUT("MICIN2"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + + SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0, + 30, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route sirf_audio_codec_map[] = { + {"SPKOUT", NULL, "Speaker Driver"}, + {"Speaker Driver", NULL, "Speaker amp driver"}, + {"Speaker amp driver", NULL, "Left dac to speaker lineout"}, + {"Speaker amp driver", NULL, "Right dac to speaker lineout"}, + {"Left dac to speaker lineout", "Switch", "DAC left"}, + {"Right dac to speaker lineout", "Switch", "DAC right"}, + {"HPOUTL", NULL, "HP Left Driver"}, + {"HPOUTR", NULL, "HP Right Driver"}, + {"HP Left Driver", NULL, "HP amp left driver"}, + {"HP Right Driver", NULL, "HP amp right driver"}, + {"HP amp left driver", NULL, "Right dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"HP amp left driver", NULL, "Left dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"Right dac to hp left amp", "Switch", "DAC left"}, + {"Right dac to hp right amp", "Switch", "DAC right"}, + {"Left dac to hp left amp", "Switch", "DAC left"}, + {"Left dac to hp right amp", "Switch", "DAC right"}, + {"DAC left", NULL, "codecclk"}, + {"DAC right", NULL, "codecclk"}, + {"DAC left", NULL, "Playback"}, + {"DAC right", NULL, "Playback"}, + {"DAC left", NULL, "HSL Phase Opposite"}, + {"DAC right", NULL, "HSL Phase Opposite"}, + + {"Capture", NULL, "ADC left"}, + {"Capture", NULL, "ADC right"}, + {"ADC left", NULL, "codecclk"}, + {"ADC right", NULL, "codecclk"}, + {"ADC left", NULL, "Left PGA mixer"}, + {"ADC right", NULL, "Right PGA mixer"}, + {"Left PGA mixer", "Line Left Switch", "LINEIN2"}, + {"Right PGA mixer", "Line Right Switch", "LINEIN1"}, + {"Left PGA mixer", "Mic Left Switch", "MICIN2"}, + {"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"}, + {"Mic input mode mux", "Single-ended", "MICIN1"}, + {"Mic input mode mux", "Differential", "MICIN1"}, +}; + +static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, + int cmd, + struct snd_soc_dai *dai) +{ + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + u32 val = 0; + + /* + * This is a workaround, When stop playback, + * need disable HP amp, avoid the current noise. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (playback) + val = IC_HSLEN | IC_HSREN; + break; + default: + return -EINVAL; + } + + if (playback) + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, val); + return 0; +} + +struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { + .trigger = sirf_audio_codec_trigger, +}; + +struct snd_soc_dai_driver sirf_audio_codec_dai = { + .name = "sirf-audio-codec", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &sirf_audio_codec_dai_ops, +}; + +static int sirf_audio_codec_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + + pm_runtime_enable(codec->dev); + + if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { + snd_soc_dapm_new_controls(dapm, + prima2_output_driver_dapm_widgets, + ARRAY_SIZE(prima2_output_driver_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, + &prima2_codec_clock_dapm_widget, 1); + return snd_soc_add_codec_controls(codec, + volume_controls_prima2, + ARRAY_SIZE(volume_controls_prima2)); + } + if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) { + snd_soc_dapm_new_controls(dapm, + atlas6_output_driver_dapm_widgets, + ARRAY_SIZE(atlas6_output_driver_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, + &atlas6_codec_clock_dapm_widget, 1); + return snd_soc_add_codec_controls(codec, + volume_controls_atlas6, + ARRAY_SIZE(volume_controls_atlas6)); + } + + return -EINVAL; +} + +static int sirf_audio_codec_remove(struct snd_soc_codec *codec) +{ + pm_runtime_disable(codec->dev); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { + .probe = sirf_audio_codec_probe, + .remove = sirf_audio_codec_remove, + .dapm_widgets = sirf_audio_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), + .dapm_routes = sirf_audio_codec_map, + .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + .idle_bias_off = true, +}; + +static const struct of_device_id sirf_audio_codec_of_match[] = { + { .compatible = "sirf,prima2-audio-codec" }, + { .compatible = "sirf,atlas6-audio-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match); + +static const struct regmap_config sirf_audio_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AUDIO_IC_CODEC_CTRL3, + .cache_type = REGCACHE_NONE, +}; + +static int sirf_audio_codec_driver_probe(struct platform_device *pdev) +{ + int ret; + struct sirf_audio_codec *sirf_audio_codec; + void __iomem *base; + struct resource *mem_res; + const struct of_device_id *match; + + match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); + + sirf_audio_codec = devm_kzalloc(&pdev->dev, + sizeof(struct sirf_audio_codec), GFP_KERNEL); + if (!sirf_audio_codec) + return -ENOMEM; + + platform_set_drvdata(pdev, sirf_audio_codec); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (base == NULL) + return -ENOMEM; + + sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sirf_audio_codec_regmap_config); + if (IS_ERR(sirf_audio_codec->regmap)) + return PTR_ERR(sirf_audio_codec->regmap); + + sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(sirf_audio_codec->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(sirf_audio_codec->clk); + } + + ret = clk_prepare_enable(sirf_audio_codec->clk); + if (ret) { + dev_err(&pdev->dev, "Enable clock failed.\n"); + return ret; + } + + ret = snd_soc_register_codec(&(pdev->dev), + &soc_codec_device_sirf_audio_codec, + &sirf_audio_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio Codec dai failed.\n"); + goto err_clk_put; + } + + /* + * Always open charge pump, if not, when the charge pump closed the + * adc will not stable + */ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + IC_CPFREQ, IC_CPFREQ); + + if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec")) + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN); + return 0; + +err_clk_put: + clk_disable_unprepare(sirf_audio_codec->clk); + return ret; +} + +static int sirf_audio_codec_driver_remove(struct platform_device *pdev) +{ + struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev); + + clk_disable_unprepare(sirf_audio_codec->clk); + snd_soc_unregister_codec(&(pdev->dev)); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int sirf_audio_codec_suspend(struct device *dev) +{ + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev); + + regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + &sirf_audio_codec->reg_ctrl0); + regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1, + &sirf_audio_codec->reg_ctrl1); + clk_disable_unprepare(sirf_audio_codec->clk); + + return 0; +} + +static int sirf_audio_codec_resume(struct device *dev) +{ + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(sirf_audio_codec->clk); + if (ret) + return ret; + + regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + sirf_audio_codec->reg_ctrl0); + regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1, + sirf_audio_codec->reg_ctrl1); + + return 0; +} +#endif + +static const struct dev_pm_ops sirf_audio_codec_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume) +}; + +static struct platform_driver sirf_audio_codec_driver = { + .driver = { + .name = "sirf-audio-codec", + .owner = THIS_MODULE, + .of_match_table = sirf_audio_codec_of_match, + .pm = &sirf_audio_codec_pm_ops, + }, + .probe = sirf_audio_codec_driver_probe, + .remove = sirf_audio_codec_driver_remove, +}; + +module_platform_driver(sirf_audio_codec_driver); + +MODULE_DESCRIPTION("SiRF audio codec driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h new file mode 100644 index 000000000000..d4c187b8e54a --- /dev/null +++ b/sound/soc/codecs/sirf-audio-codec.h @@ -0,0 +1,75 @@ +/* + * SiRF inner codec controllers define + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_AUDIO_CODEC_H +#define _SIRF_AUDIO_CODEC_H + + +#define AUDIO_IC_CODEC_PWR (0x00E0) +#define AUDIO_IC_CODEC_CTRL0 (0x00E4) +#define AUDIO_IC_CODEC_CTRL1 (0x00E8) +#define AUDIO_IC_CODEC_CTRL2 (0x00EC) +#define AUDIO_IC_CODEC_CTRL3 (0x00F0) + +#define MICBIASEN (1 << 3) + +#define IC_RDACEN (1 << 0) +#define IC_LDACEN (1 << 1) +#define IC_HSREN (1 << 2) +#define IC_HSLEN (1 << 3) +#define IC_SPEN (1 << 4) +#define IC_CPEN (1 << 5) + +#define IC_HPRSELR (1 << 6) +#define IC_HPLSELR (1 << 7) +#define IC_HPRSELL (1 << 8) +#define IC_HPLSELL (1 << 9) +#define IC_SPSELR (1 << 10) +#define IC_SPSELL (1 << 11) + +#define IC_MONOR (1 << 12) +#define IC_MONOL (1 << 13) + +#define IC_RXOSRSEL (1 << 28) +#define IC_CPFREQ (1 << 29) +#define IC_HSINVEN (1 << 30) + +#define IC_MICINREN (1 << 0) +#define IC_MICINLEN (1 << 1) +#define IC_MICIN1SEL (1 << 2) +#define IC_MICIN2SEL (1 << 3) +#define IC_MICDIFSEL (1 << 4) +#define IC_LINEIN1SEL (1 << 5) +#define IC_LINEIN2SEL (1 << 6) +#define IC_RADCEN (1 << 7) +#define IC_LADCEN (1 << 8) +#define IC_ALM (1 << 9) + +#define IC_DIGMICEN (1 << 22) +#define IC_DIGMICFREQ (1 << 23) +#define IC_ADC14B_12 (1 << 24) +#define IC_FIRDAC_HSL_EN (1 << 25) +#define IC_FIRDAC_HSR_EN (1 << 26) +#define IC_FIRDAC_LOUT_EN (1 << 27) +#define IC_POR (1 << 28) +#define IC_CODEC_CLK_EN (1 << 29) +#define IC_HP_3DB_BOOST (1 << 30) + +#define IC_ADC_LEFT_GAIN_SHIFT 16 +#define IC_ADC_RIGHT_GAIN_SHIFT 10 +#define IC_ADC_GAIN_MASK 0x3F +#define IC_MIC_MAX_GAIN 0x39 + +#define IC_RXPGAR_MASK 0x3F +#define IC_RXPGAR_SHIFT 14 +#define IC_RXPGAL_MASK 0x3F +#define IC_RXPGAL_SHIFT 21 +#define IC_RXPGAR 0x7B +#define IC_RXPGAL 0x7B + +#endif /*__SIRF_AUDIO_CODEC_H*/ diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 13045f2af4d3..42dff26b3a2a 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -312,14 +312,14 @@ static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, /* mux controls */ static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; -static const struct soc_enum sn95031_micl_enum = - SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_micl_enum, + SN95031_ADCCONFIG, 1, sn95031_mic_texts); static const struct snd_kcontrol_new sn95031_micl_mux_control = SOC_DAPM_ENUM("Route", sn95031_micl_enum); -static const struct soc_enum sn95031_micr_enum = - SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_micr_enum, + SN95031_ADCCONFIG, 3, sn95031_mic_texts); static const struct snd_kcontrol_new sn95031_micr_mux_control = SOC_DAPM_ENUM("Route", sn95031_micr_enum); @@ -328,26 +328,26 @@ static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5", "DMIC6", "ADC Left", "ADC Right" }; -static const struct soc_enum sn95031_input1_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input1_enum, + SN95031_AUDIOMUX12, 0, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input1_mux_control = SOC_DAPM_ENUM("Route", sn95031_input1_enum); -static const struct soc_enum sn95031_input2_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input2_enum, + SN95031_AUDIOMUX12, 4, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input2_mux_control = SOC_DAPM_ENUM("Route", sn95031_input2_enum); -static const struct soc_enum sn95031_input3_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input3_enum, + SN95031_AUDIOMUX34, 0, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input3_mux_control = SOC_DAPM_ENUM("Route", sn95031_input3_enum); -static const struct soc_enum sn95031_input4_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input4_enum, + SN95031_AUDIOMUX34, 4, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input4_mux_control = SOC_DAPM_ENUM("Route", sn95031_input4_enum); @@ -359,19 +359,19 @@ static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; /* 0dB to 30dB in 10dB steps */ static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0); -static const struct soc_enum sn95031_micmode1_enum = - SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); -static const struct soc_enum sn95031_micmode2_enum = - SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); +static SOC_ENUM_SINGLE_DECL(sn95031_micmode1_enum, + SN95031_MICAMP1, 1, sn95031_micmode_text); +static SOC_ENUM_SINGLE_DECL(sn95031_micmode2_enum, + SN95031_MICAMP2, 1, sn95031_micmode_text); static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; -static const struct soc_enum sn95031_dmic12_cfg_enum = - SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); -static const struct soc_enum sn95031_dmic34_cfg_enum = - SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); -static const struct soc_enum sn95031_dmic56_cfg_enum = - SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(sn95031_dmic12_cfg_enum, + SN95031_DMICMUX, 0, sn95031_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(sn95031_dmic34_cfg_enum, + SN95031_DMICMUX, 1, sn95031_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(sn95031_dmic56_cfg_enum, + SN95031_DMICMUX, 2, sn95031_dmic_cfg_text); static const struct snd_kcontrol_new sn95031_snd_controls[] = { SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), @@ -825,8 +825,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index cc8debce752f..56adb3e2def9 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -169,19 +169,19 @@ static const char * const ssm2518_drc_hold_time_text[] = { "682.24 ms", "1364 ms", }; -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); static const struct snd_kcontrol_new ssm2518_snd_controls[] = { @@ -648,16 +648,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { static int ssm2518_probe(struct snd_soc_codec *codec) { - struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ssm2518->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c new file mode 100644 index 000000000000..abd63d537173 --- /dev/null +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -0,0 +1,57 @@ +/* + * SSM2602/SSM2603/SSM2604 I2C audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "ssm2602.h" + +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return ssm2602_probe(&client->dev, id->driver_data, + devm_regmap_init_i2c(client, &ssm2602_regmap_config)); +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", SSM2602 }, + { "ssm2603", SSM2602 }, + { "ssm2604", SSM2604 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); + +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; +module_i2c_driver(ssm2602_i2c_driver); + +MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 I2C driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c new file mode 100644 index 000000000000..2bf55e24a7bb --- /dev/null +++ b/sound/soc/codecs/ssm2602-spi.c @@ -0,0 +1,41 @@ +/* + * SSM2602 SPI audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/spi/spi.h> +#include <linux/regmap.h> + +#include <sound/soc.h> + +#include "ssm2602.h" + +static int ssm2602_spi_probe(struct spi_device *spi) +{ + return ssm2602_probe(&spi->dev, SSM2602, + devm_regmap_init_spi(spi, &ssm2602_regmap_config)); +} + +static int ssm2602_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver ssm2602_spi_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_spi_probe, + .remove = ssm2602_spi_remove, +}; +module_spi_driver(ssm2602_spi_driver); + +MODULE_DESCRIPTION("ASoC SSM2602 SPI driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index af76bbd1b24f..97b0454eb346 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -27,32 +27,20 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> -#include <linux/i2c.h> -#include <linux/spi/spi.h> #include <linux/regmap.h> #include <linux/slab.h> -#include <sound/core.h> + #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> -#include <sound/initval.h> #include <sound/tlv.h> #include "ssm2602.h" -enum ssm2602_type { - SSM2602, - SSM2604, -}; - /* codec private data */ struct ssm2602_priv { unsigned int sysclk; - struct snd_pcm_hw_constraint_list *sysclk_constraints; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; struct regmap *regmap; @@ -75,15 +63,16 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { - "Line", "Mic", "None", "None", "None", - "None", "None", "None", + "Line", "Mic", }; static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; static const struct soc_enum ssm2602_enum[] = { - SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select), - SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), + SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select), + ssm2602_input_select), + SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph), + ssm2602_deemph), }; static const unsigned int ssm260x_outmix_tlv[] = { @@ -197,7 +186,7 @@ static const unsigned int ssm2602_rates_12288000[] = { 8000, 16000, 32000, 48000, 96000, }; -static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { +static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { .list = ssm2602_rates_12288000, .count = ARRAY_SIZE(ssm2602_rates_12288000), }; @@ -206,7 +195,7 @@ static const unsigned int ssm2602_rates_11289600[] = { 8000, 44100, 88200, }; -static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { +static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { .list = ssm2602_rates_11289600, .count = ARRAY_SIZE(ssm2602_rates_11289600), }; @@ -529,7 +518,7 @@ static int ssm2602_resume(struct snd_soc_codec *codec) return 0; } -static int ssm2602_probe(struct snd_soc_codec *codec) +static int ssm2602_codec_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; @@ -554,7 +543,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec) ARRAY_SIZE(ssm2602_routes)); } -static int ssm2604_probe(struct snd_soc_codec *codec) +static int ssm2604_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; @@ -568,18 +557,11 @@ static int ssm2604_probe(struct snd_soc_codec *codec) ARRAY_SIZE(ssm2604_routes)); } -static int ssm260x_probe(struct snd_soc_codec *codec) +static int ssm260x_codec_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ssm2602->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); @@ -597,10 +579,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec) switch (ssm2602->type) { case SSM2602: - ret = ssm2602_probe(codec); + ret = ssm2602_codec_probe(codec); break; case SSM2604: - ret = ssm2604_probe(codec); + ret = ssm2604_codec_probe(codec); break; } @@ -620,7 +602,7 @@ static int ssm2602_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { - .probe = ssm260x_probe, + .probe = ssm260x_codec_probe, .remove = ssm2602_remove, .suspend = ssm2602_suspend, .resume = ssm2602_resume, @@ -639,7 +621,7 @@ static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) return reg == SSM2602_RESET; } -static const struct regmap_config ssm2602_regmap_config = { +const struct regmap_config ssm2602_regmap_config = { .val_bits = 9, .reg_bits = 7, @@ -650,134 +632,28 @@ static const struct regmap_config ssm2602_regmap_config = { .reg_defaults_raw = ssm2602_reg, .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), }; +EXPORT_SYMBOL_GPL(ssm2602_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int ssm2602_spi_probe(struct spi_device *spi) +int ssm2602_probe(struct device *dev, enum ssm2602_type type, + struct regmap *regmap) { struct ssm2602_priv *ssm2602; - int ret; - - ssm2602 = devm_kzalloc(&spi->dev, sizeof(struct ssm2602_priv), - GFP_KERNEL); - if (ssm2602 == NULL) - return -ENOMEM; - - spi_set_drvdata(spi, ssm2602); - ssm2602->type = SSM2602; - ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config); - if (IS_ERR(ssm2602->regmap)) - return PTR_ERR(ssm2602->regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - return ret; -} - -static int ssm2602_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver ssm2602_spi_driver = { - .driver = { - .name = "ssm2602", - .owner = THIS_MODULE, - }, - .probe = ssm2602_spi_probe, - .remove = ssm2602_spi_remove, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -/* - * ssm2602 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ -static int ssm2602_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct ssm2602_priv *ssm2602; - int ret; - - ssm2602 = devm_kzalloc(&i2c->dev, sizeof(struct ssm2602_priv), - GFP_KERNEL); + ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL); if (ssm2602 == NULL) return -ENOMEM; - i2c_set_clientdata(i2c, ssm2602); - ssm2602->type = id->driver_data; - - ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config); - if (IS_ERR(ssm2602->regmap)) - return PTR_ERR(ssm2602->regmap); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - return ret; -} - -static int ssm2602_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id ssm2602_i2c_id[] = { - { "ssm2602", SSM2602 }, - { "ssm2603", SSM2602 }, - { "ssm2604", SSM2604 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); - -/* corgi i2c codec control layer */ -static struct i2c_driver ssm2602_i2c_driver = { - .driver = { - .name = "ssm2602", - .owner = THIS_MODULE, - }, - .probe = ssm2602_i2c_probe, - .remove = ssm2602_i2c_remove, - .id_table = ssm2602_i2c_id, -}; -#endif - - -static int __init ssm2602_modinit(void) -{ - int ret = 0; - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&ssm2602_spi_driver); - if (ret) - return ret; -#endif - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ssm2602_i2c_driver); - if (ret) - return ret; -#endif - - return ret; -} -module_init(ssm2602_modinit); - -static void __exit ssm2602_exit(void) -{ -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&ssm2602_spi_driver); -#endif + dev_set_drvdata(dev, ssm2602); + ssm2602->type = SSM2602; + ssm2602->regmap = regmap; -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ssm2602_i2c_driver); -#endif + return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, + &ssm2602_dai, 1); } -module_exit(ssm2602_exit); +EXPORT_SYMBOL_GPL(ssm2602_probe); MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver"); MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h index fbd07d7b73ca..747538847689 100644 --- a/sound/soc/codecs/ssm2602.h +++ b/sound/soc/codecs/ssm2602.h @@ -28,6 +28,20 @@ #ifndef _SSM2602_H #define _SSM2602_H +#include <linux/regmap.h> + +struct device; + +enum ssm2602_type { + SSM2602, + SSM2604, +}; + +extern const struct regmap_config ssm2602_regmap_config; + +int ssm2602_probe(struct device *dev, enum ssm2602_type type, + struct regmap *regmap); + /* SSM2602 Codec Register definitions */ #define SSM2602_LINVOL 0x00 diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 06edb396e733..12577749b17b 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -187,42 +187,42 @@ static const unsigned int sta32x_limiter_drc_release_tlv[] = { 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), }; -static const struct soc_enum sta32x_drc_ac_enum = - SOC_ENUM_SINGLE(STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, - 2, sta32x_drc_ac); -static const struct soc_enum sta32x_auto_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, - 3, sta32x_auto_eq_mode); -static const struct soc_enum sta32x_auto_gc_enum = - SOC_ENUM_SINGLE(STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, - 4, sta32x_auto_gc_mode); -static const struct soc_enum sta32x_auto_xo_enum = - SOC_ENUM_SINGLE(STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, - 16, sta32x_auto_xo_mode); -static const struct soc_enum sta32x_preset_eq_enum = - SOC_ENUM_SINGLE(STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, - 32, sta32x_preset_eq_mode); -static const struct soc_enum sta32x_limiter_ch1_enum = - SOC_ENUM_SINGLE(STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch2_enum = - SOC_ENUM_SINGLE(STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter_ch3_enum = - SOC_ENUM_SINGLE(STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, - 3, sta32x_limiter_select); -static const struct soc_enum sta32x_limiter1_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter2_attack_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxA_SHIFT, - 16, sta32x_limiter_attack_rate); -static const struct soc_enum sta32x_limiter1_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L1AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); -static const struct soc_enum sta32x_limiter2_release_rate_enum = - SOC_ENUM_SINGLE(STA32X_L2AR, STA32X_LxR_SHIFT, - 16, sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_drc_ac_enum, + STA32X_CONFD, STA32X_CONFD_DRC_SHIFT, + sta32x_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_eq_enum, + STA32X_AUTO1, STA32X_AUTO1_AMEQ_SHIFT, + sta32x_auto_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_gc_enum, + STA32X_AUTO1, STA32X_AUTO1_AMGC_SHIFT, + sta32x_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_auto_xo_enum, + STA32X_AUTO2, STA32X_AUTO2_XO_SHIFT, + sta32x_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_preset_eq_enum, + STA32X_AUTO3, STA32X_AUTO3_PEQ_SHIFT, + sta32x_preset_eq_mode); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch1_enum, + STA32X_C1CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch2_enum, + STA32X_C2CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter_ch3_enum, + STA32X_C3CFG, STA32X_CxCFG_LS_SHIFT, + sta32x_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_attack_rate_enum, + STA32X_L1AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_attack_rate_enum, + STA32X_L2AR, STA32X_LxA_SHIFT, + sta32x_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter1_release_rate_enum, + STA32X_L1AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta32x_limiter2_release_rate_enum, + STA32X_L2AR, STA32X_LxR_SHIFT, + sta32x_limiter_release_rate); /* byte array controls for setting biquad, mixer, scaling coefficients; * for biquads all five coefficients need to be set in one go, @@ -331,7 +331,7 @@ static int sta32x_sync_coef_shadow(struct snd_soc_codec *codec) static int sta32x_cache_sync(struct snd_soc_codec *codec) { - struct sta32x_priv *sta32x = codec->control_data; + struct sta32x_priv *sta32x = snd_soc_codec_get_drvdata(codec); unsigned int mute; int rc; @@ -434,7 +434,7 @@ SOC_SINGLE_TLV("Treble Tone Control", STA32X_TONE, STA32X_TONE_TTC_SHIFT, 15, 0, SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta32x_limiter1_attack_rate_enum), SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta32x_limiter2_attack_rate_enum), SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), -SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta32x_limiter2_release_rate_enum), /* depending on mode, the attack/release thresholds have * two different enum definitions; provide both @@ -872,16 +872,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will - * then do the I2C transactions itself. - */ - codec->control_data = sta32x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - goto err; - } - /* Chip documentation explicitly requires that the reset values * of reserved register bits are left untouched. * Write the register default value to cache for reserved registers, @@ -946,10 +936,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; - -err: - regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - return ret; } static int sta32x_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 40c07be9b581..a40c4b0196a3 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -141,7 +141,7 @@ static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary", static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0); static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0); -static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); +static SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); static const struct snd_kcontrol_new sta529_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0, @@ -193,8 +193,7 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; @@ -322,16 +321,6 @@ static struct snd_soc_dai_driver sta529_dai = { static int sta529_probe(struct snd_soc_codec *codec) { - struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = sta529->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index a5455c1aea42..53b810d23fea 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -62,25 +62,25 @@ static const char *stac9766_boost1[] = {"0dB", "10dB"}; static const char *stac9766_boost2[] = {"0dB", "20dB"}; static const char *stac9766_stereo_mic[] = {"Off", "On"}; -static const struct soc_enum stac9766_record_enum = - SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); -static const struct soc_enum stac9766_mono_enum = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); -static const struct soc_enum stac9766_mic_enum = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); -static const struct soc_enum stac9766_SPDIF_enum = - SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); -static const struct soc_enum stac9766_popbypass_enum = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); -static const struct soc_enum stac9766_record_all_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, - stac9766_record_all_mux); -static const struct soc_enum stac9766_boost1_enum = - SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ -static const struct soc_enum stac9766_boost2_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ -static const struct soc_enum stac9766_stereo_mic_enum = - SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); +static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum, + AC97_REC_SEL, 8, 0, stac9766_record_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum, + AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum, + AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum, + AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum, + AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum, + AC97_STAC_ANALOG_SPECIAL, 12, + stac9766_record_all_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum, + AC97_MIC, 6, stac9766_boost1); /* 0/10dB */ +static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum, + AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */ +static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum, + AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic); static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c new file mode 100644 index 000000000000..20fc46092c2c --- /dev/null +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -0,0 +1,59 @@ +/* + * ALSA SoC TLV320AIC23 codec driver I2C interface + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> + +#include "tlv320aic23.h" + +static int tlv320aic23_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct regmap *regmap; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); + return tlv320aic23_probe(&i2c->dev, regmap); +} + +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23-codec", + }, + .probe = tlv320aic23_i2c_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +module_i2c_driver(tlv320aic23_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver I2C"); +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c new file mode 100644 index 000000000000..3b387e41d75d --- /dev/null +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -0,0 +1,56 @@ +/* + * ALSA SoC TLV320AIC23 codec driver SPI interface + * + * Author: Arun KS, <arunks@mistralsolutions.com> + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <sound/soc.h> + +#include "tlv320aic23.h" + +static int aic23_spi_probe(struct spi_device *spi) +{ + int ret; + struct regmap *regmap; + + dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); + + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap); + return tlv320aic23_probe(&spi->dev, regmap); +} + +static int aic23_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver aic23_spi = { + .driver = { + .name = "tlv320aic23", + .owner = THIS_MODULE, + }, + .probe = aic23_spi_probe, + .remove = aic23_spi_remove, +}; + +module_spi_driver(aic23_spi); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver SPI"); +MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc56f51..20864ee8793b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -23,7 +23,6 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> @@ -51,7 +50,7 @@ static const struct reg_default tlv320aic23_reg[] = { { 9, 0x0000 }, }; -static const struct regmap_config tlv320aic23_regmap = { +const struct regmap_config tlv320aic23_regmap = { .reg_bits = 7, .val_bits = 9, @@ -60,20 +59,21 @@ static const struct regmap_config tlv320aic23_regmap = { .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL(tlv320aic23_regmap); static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const struct soc_enum rec_src_enum = - SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static SOC_ENUM_SINGLE_DECL(rec_src_enum, + TLV320AIC23_ANLG, 2, rec_src_text); static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = SOC_DAPM_ENUM("Input Select", rec_src_enum); -static const struct soc_enum tlv320aic23_rec_src = - SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); -static const struct soc_enum tlv320aic23_deemph = - SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); +static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src, + TLV320AIC23_ANLG, 2, rec_src_text); +static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph, + TLV320AIC23_DIGT, 1, deemph_text); static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); @@ -400,7 +400,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ - if (!codec->active) { + if (!snd_soc_codec_is_active(codec)) { udelay(50); snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } @@ -557,16 +557,8 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) return 0; } -static int tlv320aic23_probe(struct snd_soc_codec *codec) +static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); @@ -604,7 +596,7 @@ static int tlv320aic23_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { - .probe = tlv320aic23_probe, + .probe = tlv320aic23_codec_probe, .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, @@ -617,56 +609,25 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -/* - * If the i2c layer weren't so broken, we could pass this kind of data - * around - */ -static int tlv320aic23_codec_probe(struct i2c_client *i2c, - const struct i2c_device_id *i2c_id) +int tlv320aic23_probe(struct device *dev, struct regmap *regmap) { struct aic23 *aic23; - int ret; - if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - aic23 = devm_kzalloc(&i2c->dev, sizeof(struct aic23), GFP_KERNEL); + aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL); if (aic23 == NULL) return -ENOMEM; - aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); - if (IS_ERR(aic23->regmap)) - return PTR_ERR(aic23->regmap); + aic23->regmap = regmap; - i2c_set_clientdata(i2c, aic23); + dev_set_drvdata(dev, aic23); - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); - return ret; + return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, + &tlv320aic23_dai, 1); } -static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; -} - -static const struct i2c_device_id tlv320aic23_id[] = { - {"tlv320aic23", 0}, - {} -}; - -MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); - -static struct i2c_driver tlv320aic23_i2c_driver = { - .driver = { - .name = "tlv320aic23-codec", - }, - .probe = tlv320aic23_codec_probe, - .remove = __exit_p(tlv320aic23_i2c_remove), - .id_table = tlv320aic23_id, -}; - -module_i2c_driver(tlv320aic23_i2c_driver); +EXPORT_SYMBOL(tlv320aic23_probe); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h index e804120bd3da..3a7235a04a89 100644 --- a/sound/soc/codecs/tlv320aic23.h +++ b/sound/soc/codecs/tlv320aic23.h @@ -12,6 +12,12 @@ #ifndef _TLV320AIC23_H #define _TLV320AIC23_H +struct device; +struct regmap_config; + +extern const struct regmap_config tlv320aic23_regmap; +int tlv320aic23_probe(struct device *dev, struct regmap *regmap); + /* Codec TLV320AIC23 */ #define TLV320AIC23_LINVOL 0x00 #define TLV320AIC23_RINVOL 0x01 diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 94a658fa6d97..43069de3d56a 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -238,8 +238,9 @@ static struct snd_soc_dai_driver aic26_dai = { * ALSA controls */ static const char *aic26_capture_src_text[] = {"Mic", "Aux"}; -static const struct soc_enum aic26_capture_src_enum = - SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12, 2, aic26_capture_src_text); +static SOC_ENUM_SINGLE_DECL(aic26_capture_src_enum, + AIC26_REG_AUDIO_CTRL1, 12, + aic26_capture_src_text); static const struct snd_kcontrol_new aic26_snd_controls[] = { /* Output */ @@ -295,8 +296,6 @@ static int aic26_probe(struct snd_soc_codec *codec) struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, reg; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - aic26->codec = codec; /* Reset the codec to power on defaults */ diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 000000000000..fa158cfe9b32 --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1280 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha <jsarha@ti.com> + * + * Based on ground work by: Ajit Kulkarni <x0175765@ti.com> + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated + * high performance codec which provides a stereo DAC, a mono ADC, + * and mono/stereo Class-D speaker driver. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <dt-bindings/sound/tlv320aic31xx-micbias.h> + +#include "tlv320aic31xx.h" + +static const struct reg_default aic31xx_reg_defaults[] = { + { AIC31XX_CLKMUX, 0x00 }, + { AIC31XX_PLLPR, 0x11 }, + { AIC31XX_PLLJ, 0x04 }, + { AIC31XX_PLLDMSB, 0x00 }, + { AIC31XX_PLLDLSB, 0x00 }, + { AIC31XX_NDAC, 0x01 }, + { AIC31XX_MDAC, 0x01 }, + { AIC31XX_DOSRMSB, 0x00 }, + { AIC31XX_DOSRLSB, 0x80 }, + { AIC31XX_NADC, 0x01 }, + { AIC31XX_MADC, 0x01 }, + { AIC31XX_AOSR, 0x80 }, + { AIC31XX_IFACE1, 0x00 }, + { AIC31XX_DATA_OFFSET, 0x00 }, + { AIC31XX_IFACE2, 0x00 }, + { AIC31XX_BCLKN, 0x01 }, + { AIC31XX_DACSETUP, 0x14 }, + { AIC31XX_DACMUTE, 0x0c }, + { AIC31XX_LDACVOL, 0x00 }, + { AIC31XX_RDACVOL, 0x00 }, + { AIC31XX_ADCSETUP, 0x00 }, + { AIC31XX_ADCFGA, 0x80 }, + { AIC31XX_ADCVOL, 0x00 }, + { AIC31XX_HPDRIVER, 0x04 }, + { AIC31XX_SPKAMP, 0x06 }, + { AIC31XX_DACMIXERROUTE, 0x00 }, + { AIC31XX_LANALOGHPL, 0x7f }, + { AIC31XX_RANALOGHPR, 0x7f }, + { AIC31XX_LANALOGSPL, 0x7f }, + { AIC31XX_RANALOGSPR, 0x7f }, + { AIC31XX_HPLGAIN, 0x02 }, + { AIC31XX_HPRGAIN, 0x02 }, + { AIC31XX_SPLGAIN, 0x00 }, + { AIC31XX_SPRGAIN, 0x00 }, + { AIC31XX_MICBIAS, 0x00 }, + { AIC31XX_MICPGA, 0x80 }, + { AIC31XX_MICPGAPI, 0x00 }, + { AIC31XX_MICPGAMI, 0x00 }, +}; + +static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return false; + } + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = AIC31XX_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct regmap_config aic31xx_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = aic31xx_writeable, + .volatile_reg = aic31xx_volatile, + .reg_defaults = aic31xx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = aic31xx_ranges, + .num_ranges = ARRAY_SIZE(aic31xx_ranges), + .max_register = 12 * 128, +}; + +#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = { + "HPVDD", + "SPRVDD", + "SPLVDD", + "AVDD", + "IOVDD", + "DVDD", +}; + +struct aic31xx_disable_nb { + struct notifier_block nb; + struct aic31xx_priv *aic31xx; +}; + +struct aic31xx_priv { + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct aic31xx_pdata pdata; + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + unsigned int sysclk; + int rate_div_line; +}; + +struct aic31xx_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; +}; + +/* ADC dividers can be disabled by cofiguring them to 0 */ +static const struct aic31xx_rate_divs aic31xx_divs[] = { + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* 8k rate */ + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + /* 11.025k rate */ + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + /* 16k rate */ + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + /* 22.05k rate */ + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + /* 32k rate */ + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + /* 44.1k rate */ + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + /* 48k rate */ + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + /* 88.2k rate */ + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + /* 96k rate */ + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + /* 176.4k rate */ + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + /* 192k rate */ + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, +}; + +static const char * const ldac_in_text[] = { + "Off", "Left Data", "Right Data", "Mono" +}; + +static const char * const rdac_in_text[] = { + "Off", "Right Data", "Left Data", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text); + +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text); + +static const char * const mic_select_text[] = { + "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" +}; + +static const +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); + +static const +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); + +/* + * controls to be exported to the user space + */ +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, + adc_fgain_tlv), + + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1), + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, + 0, -24, 40, 6, 0, adc_cgain_tlv), + + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, + 119, 0, mic_pga_tlv), + + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic311x_snd_controls[] = { + SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new ldac_in_control = + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum); + +static const struct snd_kcontrol_new rdac_in_control = + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); + +static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) +{ + unsigned int bits; + int counter = count; + int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { + usleep_range(sleep, sleep * 2); + ret = regmap_read(aic31xx->regmap, reg, &bits); + counter--; + } + if ((bits & mask) != wbits) { + dev_err(aic31xx->dev, + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n", + __func__, reg, bits, wbits, ret, mask, + (count - counter) * sleep); + ret = -1; + } + return ret; +} + +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) + +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + unsigned int reg = AIC31XX_DACFLAG1; + unsigned int mask; + + switch (WIDGET_BIT(w->reg, w->shift)) { + case WIDGET_BIT(AIC31XX_DACSETUP, 7): + mask = AIC31XX_LDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_DACSETUP, 6): + mask = AIC31XX_RDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 7): + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 6): + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + break; + default: + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + case SND_SOC_DAPM_POST_PMD: + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); + default: + dev_dbg(w->codec->dev, + "Unhandled dapm widget event %d from %s\n", + event, w->name); + } + return 0; +} + +static const struct snd_kcontrol_new left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + +static const struct snd_kcontrol_new p_term_mic1lp = + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1rp = + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum); + +static const struct snd_kcontrol_new m_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum); + +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0); + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, + aic31xx->pdata.micbias_vg << + AIC31XX_MICBIAS_SHIFT); + dev_dbg(codec->dev, "%s: turned on\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback", + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback", + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + left_output_switches, + ARRAY_SIZE(left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + right_output_switches, + ARRAY_SIZE(right_output_switches)), + + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpl_switch), + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpr_switch), + + /* Output drivers */ + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lp), + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1rp), + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lm), + + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, + &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, + 7, 1, NULL, 0), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spr_switch), + SND_SOC_DAPM_OUTPUT("SPL"), + SND_SOC_DAPM_OUTPUT("SPR"), +}; + +/* AIC3100 and AIC3120 have only mono class-D amplifier */ +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_OUTPUT("SPK"), +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { + /* DAC Input Routing */ + {"DAC Left Input", "Left Data", "DAC IN"}, + {"DAC Left Input", "Right Data", "DAC IN"}, + {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Left Data", "DAC IN"}, + {"DAC Right Input", "Right Data", "DAC IN"}, + {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left", NULL, "DAC Left Input"}, + {"DAC Right", NULL, "DAC Right Input"}, + + /* Mic input */ + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"}, + + {"ADC", NULL, "MIC_GAIN_CTL"}, + + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From MIC1LP", "MIC1LP"}, + {"Output Left", "From MIC1RP", "MIC1RP"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From MIC1RP", "MIC1RP"}, + + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +aic311x_audio_map[] = { + /* SP L path */ + {"Speaker Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "Speaker Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"Speaker Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "Speaker Right"}, + {"SPR", NULL, "SPR ClassD"}, +}; + +static const struct snd_soc_dapm_route +aic310x_audio_map[] = { + /* SP L path */ + {"Speaker", "Switch", "Output Left"}, + {"SPK ClassD", NULL, "Speaker"}, + {"SPK", NULL, "SPK ClassD"}, +}; + +static int aic31xx_add_controls(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +static int aic31xx_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, aic311x_dapm_widgets, + ARRAY_SIZE(aic311x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return ret; + } + + return 0; +} + +static int aic31xx_setup_pll(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_n = 0; + int i; + + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); + + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { + if (aic31xx_divs[i].rate == params_rate(params) && + aic31xx_divs[i].mclk == aic31xx->sysclk) + break; + } + + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + __func__, params_rate(params)); + return -EINVAL; + } + + /* PLL configuration */ + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + (aic31xx_divs[i].p_val << 4) | 0x01); + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + + snd_soc_write(codec, AIC31XX_PLLDMSB, + aic31xx_divs[i].pll_d >> 8); + snd_soc_write(codec, AIC31XX_PLLDLSB, + aic31xx_divs[i].pll_d & 0xff); + + /* DAC dividers configuration */ + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].ndac); + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].mdac); + + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + + /* ADC dividers configuration. Write reset value 1 if not used. */ + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); + + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + + /* Bit clock divider configuration. */ + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) + / snd_soc_params_to_frame_size(params); + if (bclk_n == 0) { + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", + __func__); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_BCLKN, + AIC31XX_PLL_MASK, bclk_n); + + aic31xx->rate_div_line = i; + + dev_dbg(codec->dev, + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, + aic31xx_divs[i].madc, bclk_n); + + return 0; +} + +static int aic31xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", + __func__, params_format(params), params_width(params), + params_rate(params)); + + switch (params_width(params)) { + case 16: + break; + case 20: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 24: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 32: + data = (AIC31XX_WORD_LEN_32BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + default: + dev_err(codec->dev, "%s: Unsupported format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATALEN_MASK, + data); + + return aic31xx_setup_pll(codec, params); +} + +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, + AIC31XX_DACMUTE_MASK); + } else { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, 0x0); + } + + return 0; +} + +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + u8 dsp_a_val = 0; + + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + dsp_a_val = 0x1; + case SND_SOC_DAIFMT_DSP_B: + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface_reg3 |= AIC31XX_BCLKINV_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + iface_reg1 |= (AIC31XX_DSP_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATATYPE_MASK | + AIC31XX_IFACE1_MASTER_MASK, + iface_reg1); + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + AIC31XX_DATA_OFFSET_MASK, + dsp_a_val); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BCLKINV_MASK, + iface_reg3); + + return 0; +} + +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + __func__, clk_id, freq, dir); + + for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; + } + } + + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + clk_id << AIC31XX_PLL_CLKIN_SHIFT); + + aic31xx->sysclk = freq; + return 0; +} + +static int aic31xx_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic31xx_disable_nb *disable_nb = + container_of(nb, struct aic31xx_disable_nb, nb); + struct aic31xx_priv *aic31xx = disable_nb->aic31xx; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and as at least one of the + * supplies was disabled. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + regcache_mark_dirty(aic31xx->regmap); + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + } + + return 0; +} + +static void aic31xx_clk_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + aic31xx_divs[aic31xx->rate_div_line].rate); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + mdelay(10); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].nadc) + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].madc) + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); +} + +static void aic31xx_clk_off(struct snd_soc_codec *codec) +{ + u8 mask = AIC31XX_PM_MASK; + u8 off = 0; + + dev_dbg(codec->dev, "codec clock -> off\n"); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); +} + +static int aic31xx_power_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + udelay(100); + } + regcache_cache_only(aic31xx->regmap, false); + ret = regcache_sync(aic31xx->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + regcache_cache_only(aic31xx->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + return ret; + } + return 0; +} + +static int aic31xx_power_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + regcache_cache_only(aic31xx->regmap, true); + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + + return ret; +} + +static int aic31xx_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + codec->dapm.bias_level, level); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(codec); + break; + case SND_SOC_BIAS_STANDBY: + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + aic31xx_power_on(codec); + break; + case SND_SOC_BIAS_PREPARE: + aic31xx_clk_off(codec); + break; + default: + BUG(); + } + break; + case SND_SOC_BIAS_OFF: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(codec); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int aic31xx_suspend(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic31xx_resume(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic31xx_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(aic31xx->dev, "## %s\n", __func__); + + aic31xx = snd_soc_codec_get_drvdata(codec); + + aic31xx->codec = codec; + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { + aic31xx->disable_nb[i].nb.notifier_call = + aic31xx_regulator_event; + aic31xx->disable_nb[i].aic31xx = aic31xx; + ret = regulator_register_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + return ret; + } + } + + regcache_cache_only(aic31xx->regmap, true); + regcache_mark_dirty(aic31xx->regmap); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = aic31xx_add_widgets(codec); + + return ret; +} + +static int aic31xx_codec_remove(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + /* power down chip */ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + regulator_unregister_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { + .probe = aic31xx_codec_probe, + .remove = aic31xx_codec_remove, + .suspend = aic31xx_suspend, + .resume = aic31xx_resume, + .set_bias_level = aic31xx_set_bias_level, + .controls = aic31xx_snd_controls, + .num_controls = ARRAY_SIZE(aic31xx_snd_controls), + .dapm_widgets = aic31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), + .dapm_routes = aic31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), +}; + +static struct snd_soc_dai_ops aic31xx_dai_ops = { + .hw_params = aic31xx_hw_params, + .set_sysclk = aic31xx_set_dai_sysclk, + .set_fmt = aic31xx_set_dai_fmt, + .digital_mute = aic31xx_dac_mute, +}; + +static struct snd_soc_dai_driver aic31xx_dai_driver[] = { + { + .name = "tlv320aic31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic31xx_of_match[] = { + { .compatible = "ti,tlv320aic310x" }, + { .compatible = "ti,tlv320aic311x" }, + { .compatible = "ti,tlv320aic3100" }, + { .compatible = "ti,tlv320aic3110" }, + { .compatible = "ti,tlv320aic3120" }, + { .compatible = "ti,tlv320aic3111" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match); + +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ + struct device_node *np = aic31xx->dev->of_node; + unsigned int value = MICBIAS_2_0V; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case MICBIAS_2_0V: + case MICBIAS_2_5V: + case MICBIAS_AVDDV: + aic31xx->pdata.micbias_vg = value; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret > 0) + aic31xx->pdata.gpio_reset = ret; +} +#else /* CONFIG_OF */ +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ +} +#endif /* CONFIG_OF */ + +static void aic31xx_device_init(struct aic31xx_priv *aic31xx) +{ + int ret, i; + + dev_set_drvdata(aic31xx->dev, aic31xx); + + if (dev_get_platdata(aic31xx->dev)) + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev), + sizeof(aic31xx->pdata)); + else if (aic31xx->dev->of_node) + aic31xx_pdata_from_of(aic31xx); + + if (aic31xx->pdata.gpio_reset) { + ret = devm_gpio_request_one(aic31xx->dev, + aic31xx->pdata.gpio_reset, + GPIOF_OUT_INIT_HIGH, + "aic31xx-reset-pin"); + if (ret < 0) { + dev_err(aic31xx->dev, "not able to acquire gpio\n"); + return; + } + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + aic31xx->supplies[i].supply = aic31xx_supply_names[i]; + + ret = devm_regulator_bulk_get(aic31xx->dev, + ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret != 0) + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + +} + +static int aic31xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic31xx_priv *aic31xx; + int ret; + const struct regmap_config *regmap_config; + + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__, + id->name, (int) id->driver_data); + + regmap_config = &aic31xx_i2c_regmap; + + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL); + if (aic31xx == NULL) + return -ENOMEM; + + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); + if (IS_ERR(aic31xx->regmap)) { + ret = PTR_ERR(aic31xx->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + aic31xx->dev = &i2c->dev; + + aic31xx->pdata.codec_type = id->driver_data; + + aic31xx_device_init(aic31xx); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id aic31xx_i2c_id[] = { + { "tlv320aic310x", AIC3100 }, + { "tlv320aic311x", AIC3110 }, + { "tlv320aic3100", AIC3100 }, + { "tlv320aic3110", AIC3110 }, + { "tlv320aic3120", AIC3120 }, + { "tlv320aic3111", AIC3111 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); + +static struct i2c_driver aic31xx_i2c_driver = { + .driver = { + .name = "tlv320aic31xx-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + }, + .probe = aic31xx_i2c_probe, + .remove = aic31xx_i2c_remove, + .id_table = aic31xx_i2c_id, +}; + +module_i2c_driver(aic31xx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver"); +MODULE_AUTHOR("Jyri Sarha"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h new file mode 100644 index 000000000000..52ed57c69dfa --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,258 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ +#ifndef _TLV320AIC31XX_H +#define _TLV320AIC31XX_H + +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 + +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + + +#define AIC31XX_STEREO_CLASS_D_BIT 0x1 +#define AIC31XX_MINIDSP_BIT 0x2 + +enum aic31xx_type { + AIC3100 = 0, + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, + AIC3120 = AIC31XX_MINIDSP_BIT, + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + int micbias_vg; +}; + +/* Page Control Register */ +#define AIC31XX_PAGECTL 0x00 + +/* Page 0 Registers */ +/* Software reset register */ +#define AIC31XX_RESET 0x01 +/* OT FLAG register */ +#define AIC31XX_OT_FLAG 0x03 +/* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_CLKMUX 0x04 +/* PLL P and R-VAL register */ +#define AIC31XX_PLLPR 0x05 +/* PLL J-VAL register */ +#define AIC31XX_PLLJ 0x06 +/* PLL D-VAL MSB register */ +#define AIC31XX_PLLDMSB 0x07 +/* PLL D-VAL LSB register */ +#define AIC31XX_PLLDLSB 0x08 +/* DAC NDAC_VAL register*/ +#define AIC31XX_NDAC 0x0B +/* DAC MDAC_VAL register */ +#define AIC31XX_MDAC 0x0C +/* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRMSB 0x0D +/* DAC OSR setting register 2, LSB value */ +#define AIC31XX_DOSRLSB 0x0E +#define AIC31XX_MINI_DSP_INPOL 0x10 +/* Clock setting register 8, PLL */ +#define AIC31XX_NADC 0x12 +/* Clock setting register 9, PLL */ +#define AIC31XX_MADC 0x13 +/* ADC Oversampling (AOSR) Register */ +#define AIC31XX_AOSR 0x14 +/* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMUX 0x19 +/* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_CLKOUTMVAL 0x1A +/* Audio Interface Setting Register 1 */ +#define AIC31XX_IFACE1 0x1B +/* Audio Data Slot Offset Programming */ +#define AIC31XX_DATA_OFFSET 0x1C +/* Audio Interface Setting Register 2 */ +#define AIC31XX_IFACE2 0x1D +/* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_BCLKN 0x1E +/* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC1 0x1F +/* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC2 0x20 +/* Audio Interface Setting Register 5 */ +#define AIC31XX_IFACESEC3 0x21 +/* I2C Bus Condition */ +#define AIC31XX_I2C 0x22 +/* ADC FLAG */ +#define AIC31XX_ADCFLAG 0x24 +/* DAC Flag Registers */ +#define AIC31XX_DACFLAG1 0x25 +#define AIC31XX_DACFLAG2 0x26 +/* Sticky Interrupt flag (overflow) */ +#define AIC31XX_OFFLAG 0x27 +/* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRDACFLAG 0x2C +/* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRADCFLAG 0x2D +/* DAC Interrupt flags 2 */ +#define AIC31XX_INTRDACFLAG2 0x2E +/* ADC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 0x2F +/* INT1 interrupt control */ +#define AIC31XX_INT1CTRL 0x30 +/* INT2 interrupt control */ +#define AIC31XX_INT2CTRL 0x31 +/* GPIO1 control */ +#define AIC31XX_GPIO1 0x33 + +#define AIC31XX_DACPRB 0x3C +/* ADC Instruction Set Register */ +#define AIC31XX_ADCPRB 0x3D +/* DAC channel setup register */ +#define AIC31XX_DACSETUP 0x3F +/* DAC Mute and volume control register */ +#define AIC31XX_DACMUTE 0x40 +/* Left DAC channel digital volume control */ +#define AIC31XX_LDACVOL 0x41 +/* Right DAC channel digital volume control */ +#define AIC31XX_RDACVOL 0x42 +/* Headset detection */ +#define AIC31XX_HSDETECT 0x43 +/* ADC Digital Mic */ +#define AIC31XX_ADCSETUP 0x51 +/* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCFGA 0x52 +/* ADC Digital Volume Control Coarse Adjust */ +#define AIC31XX_ADCVOL 0x53 + + +/* Page 1 Registers */ +/* Headphone drivers */ +#define AIC31XX_HPDRIVER 0x9F +/* Class-D Speakear Amplifier */ +#define AIC31XX_SPKAMP 0xA0 +/* HP Output Drivers POP Removal Settings */ +#define AIC31XX_HPPOP 0xA1 +/* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_SPPGARAMP 0xA2 +/* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_DACMIXERROUTE 0xA3 +/* Left Analog Vol to HPL */ +#define AIC31XX_LANALOGHPL 0xA4 +/* Right Analog Vol to HPR */ +#define AIC31XX_RANALOGHPR 0xA5 +/* Left Analog Vol to SPL */ +#define AIC31XX_LANALOGSPL 0xA6 +/* Right Analog Vol to SPR */ +#define AIC31XX_RANALOGSPR 0xA7 +/* HPL Driver */ +#define AIC31XX_HPLGAIN 0xA8 +/* HPR Driver */ +#define AIC31XX_HPRGAIN 0xA9 +/* SPL Driver */ +#define AIC31XX_SPLGAIN 0xAA +/* SPR Driver */ +#define AIC31XX_SPRGAIN 0xAB +/* HP Driver Control */ +#define AIC31XX_HPCONTROL 0xAC +/* MIC Bias Control */ +#define AIC31XX_MICBIAS 0xAE +/* MIC PGA*/ +#define AIC31XX_MICPGA 0xAF +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAPI 0xB0 +/* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGAMI 0xB1 +/* Input CM Settings */ +#define AIC31XX_MICPGACM 0xB2 + +/* Bits, masks and shifts */ + +/* AIC31XX_CLKMUX */ +#define AIC31XX_PLL_CLKIN_MASK 0x0c +#define AIC31XX_PLL_CLKIN_SHIFT 2 +#define AIC31XX_PLL_CLKIN_MCLK 0 +#define AIC31XX_CODEC_CLKIN_MASK 0x03 +#define AIC31XX_CODEC_CLKIN_SHIFT 0 +#define AIC31XX_CODEC_CLKIN_PLL 3 +#define AIC31XX_CODEC_CLKIN_BCLK 1 + +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, + AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK 0x7f +#define AIC31XX_PM_MASK 0x80 + +/* AIC31XX_IFACE1 */ +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_DATALEN_MASK 0x30 +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6) +#define AIC31XX_I2S_MODE 0x00 +#define AIC31XX_DSP_MODE 0x01 +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 +#define AIC31XX_IFACE1_MASTER_MASK 0x0C +#define AIC31XX_BCLK_MASTER 0x08 +#define AIC31XX_WCLK_MASTER 0x04 + +/* AIC31XX_DATA_OFFSET */ +#define AIC31XX_DATA_OFFSET_MASK 0xFF + +/* AIC31XX_IFACE2 */ +#define AIC31XX_BCLKINV_MASK 0x08 +#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_DAC2BCLK 0x00 +#define AIC31XX_DACMOD2BCLK 0x01 +#define AIC31XX_ADC2BCLK 0x02 +#define AIC31XX_ADCMOD2BCLK 0x03 + +/* AIC31XX_ADCFLAG */ +#define AIC31XX_ADCPWRSTATUS_MASK 0x40 + +/* AIC31XX_DACFLAG1 */ +#define AIC31XX_LDACPWRSTATUS_MASK 0x80 +#define AIC31XX_RDACPWRSTATUS_MASK 0x08 +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 + +/* AIC31XX_INTRDACFLAG */ +#define AIC31XX_HPSCDETECT_MASK 0x80 +#define AIC31XX_BUTTONPRESS_MASK 0x20 +#define AIC31XX_HSPLUG_MASK 0x10 +#define AIC31XX_LDRCTHRES_MASK 0x08 +#define AIC31XX_RDRCTHRES_MASK 0x04 +#define AIC31XX_DACSINT_MASK 0x02 +#define AIC31XX_DACAINT_MASK 0x01 + +/* AIC31XX_INT1CTRL */ +#define AIC31XX_HSPLUGDET_MASK 0x80 +#define AIC31XX_BUTTONPRESSDET_MASK 0x40 +#define AIC31XX_DRCTHRES_MASK 0x20 +#define AIC31XX_AGCNOISE_MASK 0x10 +#define AIC31XX_OC_MASK 0x08 +#define AIC31XX_ENGINE_MASK 0x04 + +/* AIC31XX_DACSETUP */ +#define AIC31XX_SOFTSTEP_MASK 0x03 + +/* AIC31XX_DACMUTE */ +#define AIC31XX_DACMUTE_MASK 0x0C + +/* AIC31XX_MICBIAS */ +#define AIC31XX_MICBIAS_MASK 0x03 +#define AIC31XX_MICBIAS_SHIFT 0 + +#endif /* _TLV320AIC31XX_H */ diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 688151ba309a..1d9b117345a3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -29,9 +29,12 @@ #include <linux/delay.h> #include <linux/pm.h> #include <linux/gpio.h> +#include <linux/of_gpio.h> #include <linux/i2c.h> #include <linux/cdev.h> #include <linux/slab.h> +#include <linux/clk.h> +#include <linux/regulator/consumer.h> #include <sound/tlv320aic32x4.h> #include <sound/core.h> @@ -66,20 +69,32 @@ struct aic32x4_priv { u32 micpga_routing; bool swapdacs; int rstn_gpio; + struct clk *mclk; + + struct regulator *supply_ldo; + struct regulator *supply_iov; + struct regulator *supply_dv; + struct regulator *supply_av; }; -/* 0dB min, 1dB steps */ -static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0); /* 0dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0); +/* -63.5dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0); +/* -6dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); +/* -12dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); static const struct snd_kcontrol_new aic32x4_snd_controls[] = { - SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL, - AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5), - SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, - AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1), - SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, - AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_DOUBLE_R_S_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, -0x6, 0x1d, 5, 0, + tlv_driver_gain), + SOC_DOUBLE_R_S_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, -0x6, 0x1d, 5, 0, + tlv_driver_gain), SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, AIC32X4_HPRGAIN, 6, 0x01, 1), SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, @@ -90,8 +105,8 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0), SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0), - SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL, - AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5), + SOC_DOUBLE_R_S_TLV("ADC Level Volume", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, -0x18, 0x28, 6, 0, tlv_adc_vol), SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL, AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5), @@ -480,8 +495,18 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: + /* Switch on master clock */ + ret = clk_prepare_enable(aic32x4->mclk); + if (ret) { + dev_err(codec->dev, "Failed to enable master clock\n"); + return ret; + } + /* Switch on PLL */ snd_soc_update_bits(codec, AIC32X4_PLLPR, AIC32X4_PLLEN, AIC32X4_PLLEN); @@ -509,29 +534,32 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* Switch off PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); + /* Switch off BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); - /* Switch off NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); + /* Switch off MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); + + /* Switch off NADC Divider */ + snd_soc_update_bits(codec, AIC32X4_NADC, + AIC32X4_NADCEN, 0); /* Switch off MDAC Divider */ snd_soc_update_bits(codec, AIC32X4_MDAC, AIC32X4_MDACEN, 0); - /* Switch off NADC Divider */ - snd_soc_update_bits(codec, AIC32X4_NADC, - AIC32X4_NADCEN, 0); + /* Switch off NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); - /* Switch off MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, 0); + /* Switch off PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); - /* Switch off BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); + /* Switch off master clock */ + clk_disable_unprepare(aic32x4->mclk); break; case SND_SOC_BIAS_OFF: break; @@ -586,9 +614,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (aic32x4->rstn_gpio >= 0) { + if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -663,11 +689,122 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; +static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, + struct device_node *np) +{ + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0); + + return 0; +} + +static void aic32x4_disable_regulators(struct aic32x4_priv *aic32x4) +{ + regulator_disable(aic32x4->supply_iov); + + if (!IS_ERR(aic32x4->supply_ldo)) + regulator_disable(aic32x4->supply_ldo); + + if (!IS_ERR(aic32x4->supply_dv)) + regulator_disable(aic32x4->supply_dv); + + if (!IS_ERR(aic32x4->supply_av)) + regulator_disable(aic32x4->supply_av); +} + +static int aic32x4_setup_regulators(struct device *dev, + struct aic32x4_priv *aic32x4) +{ + int ret = 0; + + aic32x4->supply_ldo = devm_regulator_get_optional(dev, "ldoin"); + aic32x4->supply_iov = devm_regulator_get(dev, "iov"); + aic32x4->supply_dv = devm_regulator_get_optional(dev, "dv"); + aic32x4->supply_av = devm_regulator_get_optional(dev, "av"); + + /* Check if the regulator requirements are fulfilled */ + + if (IS_ERR(aic32x4->supply_iov)) { + dev_err(dev, "Missing supply 'iov'\n"); + return PTR_ERR(aic32x4->supply_iov); + } + + if (IS_ERR(aic32x4->supply_ldo)) { + if (PTR_ERR(aic32x4->supply_ldo) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + if (IS_ERR(aic32x4->supply_dv)) { + dev_err(dev, "Missing supply 'dv' or 'ldoin'\n"); + return PTR_ERR(aic32x4->supply_dv); + } + if (IS_ERR(aic32x4->supply_av)) { + dev_err(dev, "Missing supply 'av' or 'ldoin'\n"); + return PTR_ERR(aic32x4->supply_av); + } + } else { + if (IS_ERR(aic32x4->supply_dv) && + PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (IS_ERR(aic32x4->supply_av) && + PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + ret = regulator_enable(aic32x4->supply_iov); + if (ret) { + dev_err(dev, "Failed to enable regulator iov\n"); + return ret; + } + + if (!IS_ERR(aic32x4->supply_ldo)) { + ret = regulator_enable(aic32x4->supply_ldo); + if (ret) { + dev_err(dev, "Failed to enable regulator ldo\n"); + goto error_ldo; + } + } + + if (!IS_ERR(aic32x4->supply_dv)) { + ret = regulator_enable(aic32x4->supply_dv); + if (ret) { + dev_err(dev, "Failed to enable regulator dv\n"); + goto error_dv; + } + } + + if (!IS_ERR(aic32x4->supply_av)) { + ret = regulator_enable(aic32x4->supply_av); + if (ret) { + dev_err(dev, "Failed to enable regulator av\n"); + goto error_av; + } + } + + if (!IS_ERR(aic32x4->supply_ldo) && IS_ERR(aic32x4->supply_av)) + aic32x4->power_cfg |= AIC32X4_PWR_AIC32X4_LDO_ENABLE; + + return 0; + +error_av: + if (!IS_ERR(aic32x4->supply_dv)) + regulator_disable(aic32x4->supply_dv); + +error_dv: + if (!IS_ERR(aic32x4->supply_ldo)) + regulator_disable(aic32x4->supply_ldo); + +error_ldo: + regulator_disable(aic32x4->supply_iov); + return ret; +} + static int aic32x4_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct aic32x4_pdata *pdata = i2c->dev.platform_data; struct aic32x4_priv *aic32x4; + struct device_node *np = i2c->dev.of_node; int ret; aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv), @@ -686,6 +823,12 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->swapdacs = pdata->swapdacs; aic32x4->micpga_routing = pdata->micpga_routing; aic32x4->rstn_gpio = pdata->rstn_gpio; + } else if (np) { + ret = aic32x4_parse_dt(aic32x4, np); + if (ret) { + dev_err(&i2c->dev, "Failed to parse DT node\n"); + return ret; + } } else { aic32x4->power_cfg = 0; aic32x4->swapdacs = false; @@ -693,20 +836,44 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - if (aic32x4->rstn_gpio >= 0) { + aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(aic32x4->mclk)) { + dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + return PTR_ERR(aic32x4->mclk); + } + + if (gpio_is_valid(aic32x4->rstn_gpio)) { ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) return ret; } + ret = aic32x4_setup_regulators(&i2c->dev, aic32x4); + if (ret) { + dev_err(&i2c->dev, "Failed to setup regulators\n"); + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); - return ret; + if (ret) { + dev_err(&i2c->dev, "Failed to register codec\n"); + aic32x4_disable_regulators(aic32x4); + return ret; + } + + i2c_set_clientdata(i2c, aic32x4); + + return 0; } static int aic32x4_i2c_remove(struct i2c_client *client) { + struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client); + + aic32x4_disable_regulators(aic32x4); + snd_soc_unregister_codec(&client->dev); return 0; } @@ -717,10 +884,17 @@ static const struct i2c_device_id aic32x4_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4", .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, }, .probe = aic32x4_i2c_probe, .remove = aic32x4_i2c_remove, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb4b386..b1835103e9b4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1344,12 +1344,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f358393d6d6..6bfc8a17331b 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -122,7 +122,6 @@ struct tlv320dac33_priv { unsigned int uthr; enum dac33_state state; - enum snd_soc_control_type control_type; void *control_data; }; @@ -461,7 +460,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, if (dac33->fifo_mode == ucontrol->value.integer.value[0]) return 0; /* Do not allow changes while stream is running*/ - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || @@ -478,9 +477,7 @@ static const char *dac33_fifo_mode_texts[] = { "Bypass", "Mode 1", "Mode 7" }; -static const struct soc_enum dac33_fifo_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts), - dac33_fifo_mode_texts); +static SOC_ENUM_SINGLE_EXT_DECL(dac33_fifo_mode_enum, dac33_fifo_mode_texts); /* L/R Line Output Gain */ static const char *lr_lineout_gain_texts[] = { @@ -488,15 +485,13 @@ static const char *lr_lineout_gain_texts[] = { "Line 0dB DAC 12dB", "Line 6dB DAC 18dB", }; -static const struct soc_enum l_lineout_gain_enum = - SOC_ENUM_SINGLE(DAC33_LDAC_PWR_CTRL, 0, - ARRAY_SIZE(lr_lineout_gain_texts), - lr_lineout_gain_texts); +static SOC_ENUM_SINGLE_DECL(l_lineout_gain_enum, + DAC33_LDAC_PWR_CTRL, 0, + lr_lineout_gain_texts); -static const struct soc_enum r_lineout_gain_enum = - SOC_ENUM_SINGLE(DAC33_RDAC_PWR_CTRL, 0, - ARRAY_SIZE(lr_lineout_gain_texts), - lr_lineout_gain_texts); +static SOC_ENUM_SINGLE_DECL(r_lineout_gain_enum, + DAC33_RDAC_PWR_CTRL, 0, + lr_lineout_gain_texts); /* * DACL/R digital volume control: @@ -534,18 +529,16 @@ static const struct snd_kcontrol_new dac33_dapm_abypassr_control = /* LOP L/R invert selection */ static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"}; -static const struct soc_enum dac33_left_lom_enum = - SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3, - ARRAY_SIZE(dac33_lr_lom_texts), - dac33_lr_lom_texts); +static SOC_ENUM_SINGLE_DECL(dac33_left_lom_enum, + DAC33_OUT_AMP_CTRL, 3, + dac33_lr_lom_texts); static const struct snd_kcontrol_new dac33_dapm_left_lom_control = SOC_DAPM_ENUM("Route", dac33_left_lom_enum); -static const struct soc_enum dac33_right_lom_enum = - SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2, - ARRAY_SIZE(dac33_lr_lom_texts), - dac33_lr_lom_texts); +static SOC_ENUM_SINGLE_DECL(dac33_right_lom_enum, + DAC33_OUT_AMP_CTRL, 2, + dac33_lr_lom_texts); static const struct snd_kcontrol_new dac33_dapm_right_lom_control = SOC_DAPM_ENUM("Route", dac33_right_lom_enum); diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 00665ada23e2..975e0f760ac1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -415,10 +415,9 @@ static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { static const char *twl4030_handsfreel_texts[] = {"Voice", "AudioL1", "AudioL2", "AudioR2"}; -static const struct soc_enum twl4030_handsfreel_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, - ARRAY_SIZE(twl4030_handsfreel_texts), - twl4030_handsfreel_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_handsfreel_enum, + TWL4030_REG_HFL_CTL, 0, + twl4030_handsfreel_texts); static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control = SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); @@ -431,10 +430,9 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = static const char *twl4030_handsfreer_texts[] = {"Voice", "AudioR1", "AudioR2", "AudioL2"}; -static const struct soc_enum twl4030_handsfreer_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, - ARRAY_SIZE(twl4030_handsfreer_texts), - twl4030_handsfreer_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_handsfreer_enum, + TWL4030_REG_HFR_CTL, 0, + twl4030_handsfreer_texts); static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); @@ -448,10 +446,9 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = static const char *twl4030_vibra_texts[] = {"AudioL1", "AudioR1", "AudioL2", "AudioR2"}; -static const struct soc_enum twl4030_vibra_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2, - ARRAY_SIZE(twl4030_vibra_texts), - twl4030_vibra_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibra_enum, + TWL4030_REG_VIBRA_CTL, 2, + twl4030_vibra_texts); static const struct snd_kcontrol_new twl4030_dapm_vibra_control = SOC_DAPM_ENUM("Route", twl4030_vibra_enum); @@ -460,10 +457,9 @@ SOC_DAPM_ENUM("Route", twl4030_vibra_enum); static const char *twl4030_vibrapath_texts[] = {"Local vibrator", "Audio"}; -static const struct soc_enum twl4030_vibrapath_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4, - ARRAY_SIZE(twl4030_vibrapath_texts), - twl4030_vibrapath_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibrapath_enum, + TWL4030_REG_VIBRA_CTL, 4, + twl4030_vibrapath_texts); static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control = SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); @@ -490,10 +486,9 @@ static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { static const char *twl4030_micpathtx1_texts[] = {"Analog", "Digimic0"}; -static const struct soc_enum twl4030_micpathtx1_enum = - SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0, - ARRAY_SIZE(twl4030_micpathtx1_texts), - twl4030_micpathtx1_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx1_enum, + TWL4030_REG_ADCMICSEL, 0, + twl4030_micpathtx1_texts); static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control = SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum); @@ -502,10 +497,9 @@ SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum); static const char *twl4030_micpathtx2_texts[] = {"Analog", "Digimic1"}; -static const struct soc_enum twl4030_micpathtx2_enum = - SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2, - ARRAY_SIZE(twl4030_micpathtx2_texts), - twl4030_micpathtx2_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx2_enum, + TWL4030_REG_ADCMICSEL, 2, + twl4030_micpathtx2_texts); static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control = SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum); @@ -955,19 +949,15 @@ static const char *twl4030_op_modes_texts[] = { "Option 2 (voice/audio)", "Option 1 (audio)" }; -static const struct soc_enum twl4030_op_modes_enum = - SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, - ARRAY_SIZE(twl4030_op_modes_texts), - twl4030_op_modes_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_op_modes_enum, + TWL4030_REG_CODEC_MODE, 0, + twl4030_op_modes_texts); static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; if (twl4030->configured) { dev_err(codec->dev, @@ -975,19 +965,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - if (ucontrol->value.enumerated.item[0] > e->max - 1) - return -EINVAL; - - val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) - return -EINVAL; - val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= e->mask << e->shift_r; - } - - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_put_enum_double(kcontrol, ucontrol); } /* @@ -1044,10 +1022,9 @@ static const char *twl4030_avadc_clk_priority_texts[] = { "Voice high priority", "HiFi high priority" }; -static const struct soc_enum twl4030_avadc_clk_priority_enum = - SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, - ARRAY_SIZE(twl4030_avadc_clk_priority_texts), - twl4030_avadc_clk_priority_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_avadc_clk_priority_enum, + TWL4030_REG_AVADC_CTL, 2, + twl4030_avadc_clk_priority_texts); static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", @@ -1055,40 +1032,36 @@ static const char *twl4030_rampdelay_texts[] = { "3495/2581/1748 ms" }; -static const struct soc_enum twl4030_rampdelay_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2, - ARRAY_SIZE(twl4030_rampdelay_texts), - twl4030_rampdelay_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_rampdelay_enum, + TWL4030_REG_HS_POPN_SET, 2, + twl4030_rampdelay_texts); /* Vibra H-bridge direction mode */ static const char *twl4030_vibradirmode_texts[] = { "Vibra H-bridge direction", "Audio data MSB", }; -static const struct soc_enum twl4030_vibradirmode_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5, - ARRAY_SIZE(twl4030_vibradirmode_texts), - twl4030_vibradirmode_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibradirmode_enum, + TWL4030_REG_VIBRA_CTL, 5, + twl4030_vibradirmode_texts); /* Vibra H-bridge direction */ static const char *twl4030_vibradir_texts[] = { "Positive polarity", "Negative polarity", }; -static const struct soc_enum twl4030_vibradir_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1, - ARRAY_SIZE(twl4030_vibradir_texts), - twl4030_vibradir_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibradir_enum, + TWL4030_REG_VIBRA_CTL, 1, + twl4030_vibradir_texts); /* Digimic Left and right swapping */ static const char *twl4030_digimicswap_texts[] = { "Not swapped", "Swapped", }; -static const struct soc_enum twl4030_digimicswap_enum = - SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0, - ARRAY_SIZE(twl4030_digimicswap_texts), - twl4030_digimicswap_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_digimicswap_enum, + TWL4030_REG_MISC_SET_1, 0, + twl4030_digimicswap_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Codec operation mode control */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0afe8bef6765..bd3a20647fdf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -81,7 +81,7 @@ struct twl6040_data { }; /* set of rates for each pll: low-power and high-performance */ -static unsigned int lp_rates[] = { +static const unsigned int lp_rates[] = { 8000, 11250, 16000, @@ -93,7 +93,7 @@ static unsigned int lp_rates[] = { 96000, }; -static unsigned int hp_rates[] = { +static const unsigned int hp_rates[] = { 8000, 16000, 32000, @@ -101,7 +101,7 @@ static unsigned int hp_rates[] = { 96000, }; -static struct snd_pcm_hw_constraint_list sysclk_constraints[] = { +static const struct snd_pcm_hw_constraint_list sysclk_constraints[] = { { .count = ARRAY_SIZE(lp_rates), .list = lp_rates, }, { .count = ARRAY_SIZE(hp_rates), .list = hp_rates, }, }; @@ -392,8 +392,10 @@ static const char *twl6040_amicr_texts[] = {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; static const struct soc_enum twl6040_enum[] = { - SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts), - SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, + ARRAY_SIZE(twl6040_amicl_texts), twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, + ARRAY_SIZE(twl6040_amicr_texts), twl6040_amicr_texts), }; static const char *twl6040_hs_texts[] = { @@ -476,9 +478,8 @@ static const char *twl6040_power_mode_texts[] = { "Low-Power", "High-Performance", }; -static const struct soc_enum twl6040_power_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl6040_power_mode_texts), - twl6040_power_mode_texts); +static SOC_ENUM_SINGLE_EXT_DECL(twl6040_power_mode_enum, + twl6040_power_mode_texts); static int twl6040_headset_power_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c94d4c1e3dac..edf27acc1d77 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -203,8 +203,7 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); u8 hw_params; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 726df6d43c2b..e62e70781ec2 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -108,7 +108,7 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, /* the interpolator & decimator regs must only be written when the * codec DAI is active. */ - if (!codec->active && (reg >= UDA1380_MVOL)) + if (!snd_soc_codec_is_active(codec) && (reg >= UDA1380_MVOL)) return 0; pr_debug("uda1380: hw write %x val %x\n", reg, value); if (codec->hw_write(codec->control_data, data, 3) == 3) { @@ -237,25 +237,27 @@ static const char *uda1380_os_setting[] = { }; static const struct soc_enum uda1380_deemp_enum[] = { - SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), - SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, ARRAY_SIZE(uda1380_deemp), + uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, ARRAY_SIZE(uda1380_deemp), + uda1380_deemp), }; -static const struct soc_enum uda1380_input_sel_enum = - SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ -static const struct soc_enum uda1380_output_sel_enum = - SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ -static const struct soc_enum uda1380_spf_enum = - SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ -static const struct soc_enum uda1380_capture_sel_enum = - SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ -static const struct soc_enum uda1380_sel_ns_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ -static const struct soc_enum uda1380_mix_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ -static const struct soc_enum uda1380_sdet_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ -static const struct soc_enum uda1380_os_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ +static SOC_ENUM_SINGLE_DECL(uda1380_input_sel_enum, + UDA1380_ADC, 2, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ +static SOC_ENUM_SINGLE_DECL(uda1380_output_sel_enum, + UDA1380_PM, 7, uda1380_output_sel); /* R02_EN_AVC */ +static SOC_ENUM_SINGLE_DECL(uda1380_spf_enum, + UDA1380_MODE, 14, uda1380_spf_mode); /* M */ +static SOC_ENUM_SINGLE_DECL(uda1380_capture_sel_enum, + UDA1380_IFACE, 6, uda1380_capture_sel); /* SEL_SOURCE */ +static SOC_ENUM_SINGLE_DECL(uda1380_sel_ns_enum, + UDA1380_MIXER, 14, uda1380_sel_ns); /* SEL_NS */ +static SOC_ENUM_SINGLE_DECL(uda1380_mix_enum, + UDA1380_MIXER, 12, uda1380_mix_control); /* MIX, MIX_POS */ +static SOC_ENUM_SINGLE_DECL(uda1380_sdet_enum, + UDA1380_MIXER, 4, uda1380_sdet_setting); /* SD_VALUE */ +static SOC_ENUM_SINGLE_DECL(uda1380_os_enum, + UDA1380_MIXER, 0, uda1380_os_setting); /* OS */ /* * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) @@ -564,8 +566,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index b7ab2ef567c8..6be5f80b65f1 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -197,7 +197,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, return 0; /* Do not allow changes while stream is running */ - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || @@ -209,8 +209,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, return 1; } -static const struct soc_enum wl1273_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route); +static SOC_ENUM_SINGLE_EXT_DECL(wl1273_enum, wl1273_audio_route); static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -247,9 +246,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, static const char * const wl1273_audio_strings[] = { "Digital", "Analog" }; -static const struct soc_enum wl1273_audio_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), - wl1273_audio_strings); +static SOC_ENUM_SINGLE_EXT_DECL(wl1273_audio_enum, wl1273_audio_strings); static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8ae50274ea8f..83a2c872925c 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -786,8 +786,6 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); - /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 57ba315d0c84..2e721e06671b 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1113,11 +1113,10 @@ static const char *wm2200_rxanc_input_sel_texts[] = { "None", "IN1", "IN2", "IN3", }; -static const struct soc_enum wm2200_rxanc_input_sel = - SOC_ENUM_SINGLE(WM2200_RXANC_SRC, - WM2200_IN_RXANC_SEL_SHIFT, - ARRAY_SIZE(wm2200_rxanc_input_sel_texts), - wm2200_rxanc_input_sel_texts); +static SOC_ENUM_SINGLE_DECL(wm2200_rxanc_input_sel, + WM2200_RXANC_SRC, + WM2200_IN_RXANC_SEL_SHIFT, + wm2200_rxanc_input_sel_texts); static const struct snd_kcontrol_new wm2200_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, @@ -1288,11 +1287,10 @@ static const char *wm2200_aec_loopback_texts[] = { "OUT1L", "OUT1R", "OUT2L", "OUT2R", }; -static const struct soc_enum wm2200_aec_loopback = - SOC_ENUM_SINGLE(WM2200_DAC_AEC_CONTROL_1, - WM2200_AEC_LOOPBACK_SRC_SHIFT, - ARRAY_SIZE(wm2200_aec_loopback_texts), - wm2200_aec_loopback_texts); +static SOC_ENUM_SINGLE_DECL(wm2200_aec_loopback, + WM2200_DAC_AEC_CONTROL_1, + WM2200_AEC_LOOPBACK_SRC_SHIFT, + wm2200_aec_loopback_texts); static const struct snd_kcontrol_new wm2200_aec_loopback_mux = SOC_DAPM_ENUM("AEC Loopback", wm2200_aec_loopback); @@ -1556,15 +1554,8 @@ static int wm2200_probe(struct snd_soc_codec *codec) int ret; wm2200->codec = codec; - codec->control_data = wm2200->regmap; codec->dapm.bias_level = SND_SOC_BIAS_OFF; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4e3e31aaf509..eca983fad891 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -506,21 +506,21 @@ static const char *wm5100_lhpf_mode_text[] = { "Low-pass", "High-pass" }; -static const struct soc_enum wm5100_lhpf1_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf1_mode, + WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, + wm5100_lhpf_mode_text); -static const struct soc_enum wm5100_lhpf2_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf2_mode, + WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, + wm5100_lhpf_mode_text); -static const struct soc_enum wm5100_lhpf3_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf3_mode, + WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, + wm5100_lhpf_mode_text); -static const struct soc_enum wm5100_lhpf4_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf4_mode, + WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, + wm5100_lhpf_mode_text); static const struct snd_kcontrol_new wm5100_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM5100_IN1L_CONTROL, @@ -2100,6 +2100,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) { struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; if (jack) { wm5100->jack = jack; @@ -2117,9 +2118,14 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) WM5100_ACCDET_RATE_MASK); /* We need the charge pump to power MICBIAS */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "CP2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK"); + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); /* We start off just enabling microphone detection - even a * plain headphone will trigger detection. @@ -2337,13 +2343,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) int ret, i; wm5100->codec = codec; - codec->control_data = wm5100->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ce9c8e14d4bd..dcf1d12cfef8 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -582,7 +582,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; @@ -622,13 +622,16 @@ static const unsigned int wm5102_osr_val[] = { static const struct soc_enum wm5102_hpout_osr[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT1_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT2_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT3_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), }; @@ -685,15 +688,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -705,6 +701,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -716,6 +714,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -727,6 +727,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, @@ -1758,9 +1760,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2c3c962d9a85..df5a38dd8328 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -136,7 +136,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; @@ -247,15 +247,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -267,6 +260,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -278,6 +273,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -289,6 +286,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, @@ -1588,10 +1587,9 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; priv->core.arizona->dapm = &codec->dapm; - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index a183dcf3d5c1..757256bf7672 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1505,9 +1505,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - codec->control_data = wm8350->regmap; - - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, wm8350->regmap); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 48dc7d2fee36..146564feaea0 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -117,19 +117,23 @@ static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8400_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8400_left_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_left_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACL_SHIFT, + wm8400_digital_sidetone); -static const struct soc_enum wm8400_right_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE, - WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8400_right_digital_sidetone_enum, + WM8400_DIGITAL_SIDE_TONE, + WM8400_ADC_TO_DACR_SHIFT, + wm8400_digital_sidetone); static const char *wm8400_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8400_right_adcmode_enum = -SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8400_right_adcmode_enum, + WM8400_ADC_CTRL, + WM8400_ADC_HPF_CUT_SHIFT, + wm8400_adcmode); static const struct snd_kcontrol_new wm8400_snd_controls[] = { /* INMIXL */ @@ -422,9 +426,10 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT, static const char *wm8400_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8400_ainlmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT, - ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainlmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINLMODE_SHIFT, + wm8400_ainlmux); static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); @@ -435,9 +440,10 @@ SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum); static const char *wm8400_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8400_ainrmux_enum = -SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT, - ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8400_ainrmux_enum, + WM8400_INPUT_MIXER1, + WM8400_AINRMODE_SHIFT, + wm8400_ainrmux); static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum); @@ -1310,10 +1316,9 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, priv); priv->wm8400 = wm8400; - codec->control_data = wm8400->regmap; priv->codec = codec; - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, wm8400->regmap); ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 7df7d4572755..1c1e328feeb8 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -589,20 +589,12 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); - return ret; - } - wm8510_reset(codec); /* power on device */ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 74d106dc7667..601ee8178af1 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -75,8 +75,8 @@ static const char *wm8523_zd_count_text[] = { "2048", }; -static const struct soc_enum wm8523_zc_count = - SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); +static SOC_ENUM_SINGLE_DECL(wm8523_zc_count, WM8523_ZERO_DETECT, 0, + wm8523_zd_count_text); static const struct snd_kcontrol_new wm8523_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, @@ -392,18 +392,11 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - int ret; wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8523_DAC_GAINR, WM8523_DACR_VU, WM8523_DACR_VU); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989acbbe5..af7ed8b5d4e1 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -504,8 +504,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); u16 paifa = 0; u16 paifb = 0; @@ -869,12 +868,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); if (ret != 0) { diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948c513c..b0fbcb377baf 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -201,7 +201,7 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; /* deactivate */ - if (!codec->active) { + if (!snd_soc_codec_is_active(codec)) { udelay(50); snd_soc_write(codec, WM8711_ACTIVE, 0x0); } @@ -367,12 +367,6 @@ static int wm8711_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8711_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index cd89033e84c0..bac7fc28fe71 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -228,19 +228,10 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", - ret); - return ret; - } - /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } static int wm8728_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 029720366ff8..d74f43975b90 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -83,8 +83,8 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg) static const char *wm8731_input_select[] = {"Line In", "Mic"}; -static const struct soc_enum wm8731_insel_enum = - SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select); +static SOC_ENUM_SINGLE_DECL(wm8731_insel_enum, + WM8731_APANA, 2, wm8731_input_select); static int wm8731_deemph[] = { 0, 32000, 44100, 48000 }; @@ -583,13 +583,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; - codec->control_data = wm8731->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 2f167a8ca01b..b27f26cdc049 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -99,29 +99,29 @@ static const char *micbias_enum_text[] = { "100%", }; -static const struct soc_enum micbias_enum = - SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 0, 4, micbias_enum_text); +static SOC_ENUM_SINGLE_DECL(micbias_enum, + WM8737_MIC_PREAMP_CONTROL, 0, micbias_enum_text); static const char *low_cutoff_text[] = { "Low", "High" }; -static const struct soc_enum low_3d = - SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 6, 2, low_cutoff_text); +static SOC_ENUM_SINGLE_DECL(low_3d, + WM8737_3D_ENHANCE, 6, low_cutoff_text); static const char *high_cutoff_text[] = { "High", "Low" }; -static const struct soc_enum high_3d = - SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 5, 2, high_cutoff_text); +static SOC_ENUM_SINGLE_DECL(high_3d, + WM8737_3D_ENHANCE, 5, high_cutoff_text); static const char *alc_fn_text[] = { "Disabled", "Right", "Left", "Stereo" }; -static const struct soc_enum alc_fn = - SOC_ENUM_SINGLE(WM8737_ALC1, 7, 4, alc_fn_text); +static SOC_ENUM_SINGLE_DECL(alc_fn, + WM8737_ALC1, 7, alc_fn_text); static const char *alc_hold_text[] = { "0", "2.67ms", "5.33ms", "10.66ms", "21.32ms", "42.64ms", "85.28ms", @@ -129,24 +129,24 @@ static const char *alc_hold_text[] = { "10.916s", "21.832s", "43.691s" }; -static const struct soc_enum alc_hold = - SOC_ENUM_SINGLE(WM8737_ALC2, 0, 16, alc_hold_text); +static SOC_ENUM_SINGLE_DECL(alc_hold, + WM8737_ALC2, 0, alc_hold_text); static const char *alc_atk_text[] = { "8.4ms", "16.8ms", "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s", "4.3s", "8.6s" }; -static const struct soc_enum alc_atk = - SOC_ENUM_SINGLE(WM8737_ALC3, 0, 11, alc_atk_text); +static SOC_ENUM_SINGLE_DECL(alc_atk, + WM8737_ALC3, 0, alc_atk_text); static const char *alc_dcy_text[] = { "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s", "4.3s", "8.6s", "17.2s", "34.41s" }; -static const struct soc_enum alc_dcy = - SOC_ENUM_SINGLE(WM8737_ALC3, 4, 11, alc_dcy_text); +static SOC_ENUM_SINGLE_DECL(alc_dcy, + WM8737_ALC3, 4, alc_dcy_text); static const struct snd_kcontrol_new wm8737_snd_controls[] = { SOC_DOUBLE_R_TLV("Mic Boost Volume", WM8737_AUDIO_PATH_L, WM8737_AUDIO_PATH_R, @@ -191,8 +191,8 @@ static const char *linsel_text[] = { "LINPUT1", "LINPUT2", "LINPUT3", "LINPUT1 DC", }; -static const struct soc_enum linsel_enum = - SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_L, 7, 4, linsel_text); +static SOC_ENUM_SINGLE_DECL(linsel_enum, + WM8737_AUDIO_PATH_L, 7, linsel_text); static const struct snd_kcontrol_new linsel_mux = SOC_DAPM_ENUM("LINSEL", linsel_enum); @@ -202,8 +202,8 @@ static const char *rinsel_text[] = { "RINPUT1", "RINPUT2", "RINPUT3", "RINPUT1 DC", }; -static const struct soc_enum rinsel_enum = - SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_R, 7, 4, rinsel_text); +static SOC_ENUM_SINGLE_DECL(rinsel_enum, + WM8737_AUDIO_PATH_R, 7, rinsel_text); static const struct snd_kcontrol_new rinsel_mux = SOC_DAPM_ENUM("RINSEL", rinsel_enum); @@ -212,15 +212,15 @@ static const char *bypass_text[] = { "Direct", "Preamp" }; -static const struct soc_enum lbypass_enum = - SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 2, 2, bypass_text); +static SOC_ENUM_SINGLE_DECL(lbypass_enum, + WM8737_MIC_PREAMP_CONTROL, 2, bypass_text); static const struct snd_kcontrol_new lbypass_mux = SOC_DAPM_ENUM("Left Bypass", lbypass_enum); -static const struct soc_enum rbypass_enum = - SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 3, 2, bypass_text); +static SOC_ENUM_SINGLE_DECL(rbypass_enum, + WM8737_MIC_PREAMP_CONTROL, 3, bypass_text); static const struct snd_kcontrol_new rbypass_mux = SOC_DAPM_ENUM("Left Bypass", rbypass_enum); @@ -570,12 +570,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { @@ -644,7 +638,7 @@ static const struct regmap_config wm8737_regmap = { .volatile_reg = wm8737_volatile, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -758,7 +752,7 @@ static struct spi_driver wm8737_spi_driver = { static int __init wm8737_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8737_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8737 I2C driver: %d\n", @@ -781,7 +775,7 @@ static void __exit wm8737_exit(void) #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8737_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8737_i2c_driver); #endif } diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 2895c8d3b5e4..b33542a04607 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -44,7 +44,7 @@ struct wm8741_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; - struct snd_pcm_hw_constraint_list *sysclk_constraints; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; }; static const struct reg_default wm8741_reg_defaults[] = { @@ -122,74 +122,74 @@ static struct { { 6, 768 }, }; -static unsigned int rates_11289[] = { +static const unsigned int rates_11289[] = { 44100, 88235, }; -static struct snd_pcm_hw_constraint_list constraints_11289 = { +static const struct snd_pcm_hw_constraint_list constraints_11289 = { .count = ARRAY_SIZE(rates_11289), .list = rates_11289, }; -static unsigned int rates_12288[] = { +static const unsigned int rates_12288[] = { 32000, 48000, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_12288 = { +static const struct snd_pcm_hw_constraint_list constraints_12288 = { .count = ARRAY_SIZE(rates_12288), .list = rates_12288, }; -static unsigned int rates_16384[] = { +static const unsigned int rates_16384[] = { 32000, }; -static struct snd_pcm_hw_constraint_list constraints_16384 = { +static const struct snd_pcm_hw_constraint_list constraints_16384 = { .count = ARRAY_SIZE(rates_16384), .list = rates_16384, }; -static unsigned int rates_16934[] = { +static const unsigned int rates_16934[] = { 44100, 88235, }; -static struct snd_pcm_hw_constraint_list constraints_16934 = { +static const struct snd_pcm_hw_constraint_list constraints_16934 = { .count = ARRAY_SIZE(rates_16934), .list = rates_16934, }; -static unsigned int rates_18432[] = { +static const unsigned int rates_18432[] = { 48000, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_18432 = { +static const struct snd_pcm_hw_constraint_list constraints_18432 = { .count = ARRAY_SIZE(rates_18432), .list = rates_18432, }; -static unsigned int rates_22579[] = { +static const unsigned int rates_22579[] = { 44100, 88235, 1764000 }; -static struct snd_pcm_hw_constraint_list constraints_22579 = { +static const struct snd_pcm_hw_constraint_list constraints_22579 = { .count = ARRAY_SIZE(rates_22579), .list = rates_22579, }; -static unsigned int rates_24576[] = { +static const unsigned int rates_24576[] = { 32000, 48000, 96000, 192000 }; -static struct snd_pcm_hw_constraint_list constraints_24576 = { +static const struct snd_pcm_hw_constraint_list constraints_24576 = { .count = ARRAY_SIZE(rates_24576), .list = rates_24576, }; -static unsigned int rates_36864[] = { +static const unsigned int rates_36864[] = { 48000, 96000, 19200 }; -static struct snd_pcm_hw_constraint_list constraints_36864 = { +static const struct snd_pcm_hw_constraint_list constraints_36864 = { .count = ARRAY_SIZE(rates_36864), .list = rates_36864, }; @@ -429,12 +429,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_get; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_enable; - } - ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 78616a638a55..33990b63d214 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -702,12 +702,6 @@ static int wm8750_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8750_reset(codec); if (ret < 0) { printk(KERN_ERR "wm8750: failed to reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da93a268..cbb8d55052a4 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -251,7 +251,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, if (wm8753->dai_func == ucontrol->value.integer.value[0]) return 0; - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EBUSY; ioctl = snd_soc_read(codec, WM8753_IOCTL); @@ -1314,7 +1314,7 @@ static int wm8753_mute(struct snd_soc_dai *dai, int mute) /* the digital mute covers the HiFi and Voice DAC's on the WM8753. * make sure we check if they are not both active when we mute */ if (mute && wm8753->dai_func == 1) { - if (!codec->active) + if (!snd_soc_codec_is_active(codec)) snd_soc_write(codec, WM8753_DAC, mute_reg | 0x8); } else { if (mute) @@ -1440,7 +1440,6 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - codec->cache_sync = 1; return 0; } @@ -1471,13 +1470,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); - codec->control_data = wm8753->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8753_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 89a18d82f303..c61aeb38efb8 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -196,8 +196,8 @@ static const char *ain_text[] = { "AIN5", "AIN6", "AIN7", "AIN8" }; -static const struct soc_enum ain_enum = - SOC_ENUM_DOUBLE(WM8770_ADCMUX, 0, 4, 8, ain_text); +static SOC_ENUM_DOUBLE_DECL(ain_enum, + WM8770_ADCMUX, 0, 4, ain_text); static const struct snd_kcontrol_new ain_mux = SOC_DAPM_ENUM("Capture Mux", ain_enum); @@ -580,12 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies), wm8770->supplies); if (ret) { diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ef8246725232..70952ceb278b 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -430,12 +430,6 @@ static int wm8776_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8776_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206a6807..ee76f0fb4299 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -92,7 +92,7 @@ WM8804_REGULATOR_EVENT(0) WM8804_REGULATOR_EVENT(1) static const char *txsrc_text[] = { "S/PDIF RX", "AIF" }; -static const SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text); +static SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text); static const struct snd_kcontrol_new wm8804_snd_controls[] = { SOC_ENUM_EXT("Input Source", txsrc, txsrc_get, txsrc_put), @@ -546,14 +546,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); - codec->control_data = wm8804->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e98bc7038a08..d09fdce57f5a 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -304,53 +304,53 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1); static const char *mic_bias_level_txt[] = { "0.9*AVDD", "0.65*AVDD" }; -static const struct soc_enum mic_bias_level = -SOC_ENUM_SINGLE(WM8900_REG_INCTL, 8, 2, mic_bias_level_txt); +static SOC_ENUM_SINGLE_DECL(mic_bias_level, + WM8900_REG_INCTL, 8, mic_bias_level_txt); static const char *dac_mute_rate_txt[] = { "Fast", "Slow" }; -static const struct soc_enum dac_mute_rate = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 7, 2, dac_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(dac_mute_rate, + WM8900_REG_DACCTRL, 7, dac_mute_rate_txt); static const char *dac_deemphasis_txt[] = { "Disabled", "32kHz", "44.1kHz", "48kHz" }; -static const struct soc_enum dac_deemphasis = -SOC_ENUM_SINGLE(WM8900_REG_DACCTRL, 4, 4, dac_deemphasis_txt); +static SOC_ENUM_SINGLE_DECL(dac_deemphasis, + WM8900_REG_DACCTRL, 4, dac_deemphasis_txt); static const char *adc_hpf_cut_txt[] = { "Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3" }; -static const struct soc_enum adc_hpf_cut = -SOC_ENUM_SINGLE(WM8900_REG_ADCCTRL, 5, 4, adc_hpf_cut_txt); +static SOC_ENUM_SINGLE_DECL(adc_hpf_cut, + WM8900_REG_ADCCTRL, 5, adc_hpf_cut_txt); static const char *lr_txt[] = { "Left", "Right" }; -static const struct soc_enum aifl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifl_src, + WM8900_REG_AUDIO1, 15, lr_txt); -static const struct soc_enum aifr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO1, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(aifr_src, + WM8900_REG_AUDIO1, 14, lr_txt); -static const struct soc_enum dacl_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 15, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacl_src, + WM8900_REG_AUDIO2, 15, lr_txt); -static const struct soc_enum dacr_src = -SOC_ENUM_SINGLE(WM8900_REG_AUDIO2, 14, 2, lr_txt); +static SOC_ENUM_SINGLE_DECL(dacr_src, + WM8900_REG_AUDIO2, 14, lr_txt); static const char *sidetone_txt[] = { "Disabled", "Left ADC", "Right ADC" }; -static const struct soc_enum dacl_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 2, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone, + WM8900_REG_SIDETONE, 2, sidetone_txt); -static const struct soc_enum dacr_sidetone = -SOC_ENUM_SINGLE(WM8900_REG_SIDETONE, 0, 3, sidetone_txt); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone, + WM8900_REG_SIDETONE, 0, sidetone_txt); static const struct snd_kcontrol_new wm8900_snd_controls[] = { SOC_ENUM("Mic Bias Level", mic_bias_level), @@ -496,8 +496,8 @@ SOC_DAPM_SINGLE("RINPUT3 Switch", WM8900_REG_INCTL, 0, 1, 0), static const char *wm8900_lp_mux[] = { "Disabled", "Enabled" }; -static const struct soc_enum wm8900_lineout2_lp_mux = -SOC_ENUM_SINGLE(WM8900_REG_LOUTMIXCTL1, 1, 2, wm8900_lp_mux); +static SOC_ENUM_SINGLE_DECL(wm8900_lineout2_lp_mux, + WM8900_REG_LOUTMIXCTL1, 1, wm8900_lp_mux); static const struct snd_kcontrol_new wm8900_lineout2_lp = SOC_DAPM_ENUM("Route", wm8900_lineout2_lp_mux); @@ -1178,13 +1178,7 @@ static int wm8900_resume(struct snd_soc_codec *codec) static int wm8900_probe(struct snd_soc_codec *codec) { - int ret = 0, reg; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } + int reg; reg = snd_soc_read(codec, WM8900_REG_ID); if (reg != 0x8900) { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index eebcb1da3b7b..b0084a127d18 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -489,28 +489,28 @@ static const char *hpf_mode_text[] = { "Hi-fi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum hpf_mode = - SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(hpf_mode, + WM8903_ADC_DIGITAL_0, 5, hpf_mode_text); static const char *osr_text[] = { "Low power", "High performance" }; -static const struct soc_enum adc_osr = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(adc_osr, + WM8903_ANALOGUE_ADC_0, 0, osr_text); -static const struct soc_enum dac_osr = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(dac_osr, + WM8903_DAC_DIGITAL_1, 0, osr_text); static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; -static const struct soc_enum drc_slope_r0 = - SOC_ENUM_SINGLE(WM8903_DRC_2, 3, 6, drc_slope_text); +static SOC_ENUM_SINGLE_DECL(drc_slope_r0, + WM8903_DRC_2, 3, drc_slope_text); -static const struct soc_enum drc_slope_r1 = - SOC_ENUM_SINGLE(WM8903_DRC_2, 0, 6, drc_slope_text); +static SOC_ENUM_SINGLE_DECL(drc_slope_r1, + WM8903_DRC_2, 0, drc_slope_text); static const char *drc_attack_text[] = { "instantaneous", @@ -518,125 +518,125 @@ static const char *drc_attack_text[] = { "46.4ms", "92.8ms", "185.6ms" }; -static const struct soc_enum drc_attack = - SOC_ENUM_SINGLE(WM8903_DRC_1, 12, 11, drc_attack_text); +static SOC_ENUM_SINGLE_DECL(drc_attack, + WM8903_DRC_1, 12, drc_attack_text); static const char *drc_decay_text[] = { "186ms", "372ms", "743ms", "1.49s", "2.97s", "5.94s", "11.89s", "23.87s", "47.56s" }; -static const struct soc_enum drc_decay = - SOC_ENUM_SINGLE(WM8903_DRC_1, 8, 9, drc_decay_text); +static SOC_ENUM_SINGLE_DECL(drc_decay, + WM8903_DRC_1, 8, drc_decay_text); static const char *drc_ff_delay_text[] = { "5 samples", "9 samples" }; -static const struct soc_enum drc_ff_delay = - SOC_ENUM_SINGLE(WM8903_DRC_0, 5, 2, drc_ff_delay_text); +static SOC_ENUM_SINGLE_DECL(drc_ff_delay, + WM8903_DRC_0, 5, drc_ff_delay_text); static const char *drc_qr_decay_text[] = { "0.725ms", "1.45ms", "5.8ms" }; -static const struct soc_enum drc_qr_decay = - SOC_ENUM_SINGLE(WM8903_DRC_1, 4, 3, drc_qr_decay_text); +static SOC_ENUM_SINGLE_DECL(drc_qr_decay, + WM8903_DRC_1, 4, drc_qr_decay_text); static const char *drc_smoothing_text[] = { "Low", "Medium", "High" }; -static const struct soc_enum drc_smoothing = - SOC_ENUM_SINGLE(WM8903_DRC_0, 11, 3, drc_smoothing_text); +static SOC_ENUM_SINGLE_DECL(drc_smoothing, + WM8903_DRC_0, 11, drc_smoothing_text); static const char *soft_mute_text[] = { "Fast (fs/2)", "Slow (fs/32)" }; -static const struct soc_enum soft_mute = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 10, 2, soft_mute_text); +static SOC_ENUM_SINGLE_DECL(soft_mute, + WM8903_DAC_DIGITAL_1, 10, soft_mute_text); static const char *mute_mode_text[] = { "Hard", "Soft" }; -static const struct soc_enum mute_mode = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text); +static SOC_ENUM_SINGLE_DECL(mute_mode, + WM8903_DAC_DIGITAL_1, 9, mute_mode_text); static const char *companding_text[] = { "ulaw", "alaw" }; -static const struct soc_enum dac_companding = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 0, 2, companding_text); +static SOC_ENUM_SINGLE_DECL(dac_companding, + WM8903_AUDIO_INTERFACE_0, 0, companding_text); -static const struct soc_enum adc_companding = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 2, 2, companding_text); +static SOC_ENUM_SINGLE_DECL(adc_companding, + WM8903_AUDIO_INTERFACE_0, 2, companding_text); static const char *input_mode_text[] = { "Single-Ended", "Differential Line", "Differential Mic" }; -static const struct soc_enum linput_mode_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(linput_mode_enum, + WM8903_ANALOGUE_LEFT_INPUT_1, 0, input_mode_text); -static const struct soc_enum rinput_mode_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(rinput_mode_enum, + WM8903_ANALOGUE_RIGHT_INPUT_1, 0, input_mode_text); static const char *linput_mux_text[] = { "IN1L", "IN2L", "IN3L" }; -static const struct soc_enum linput_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 2, 3, linput_mux_text); +static SOC_ENUM_SINGLE_DECL(linput_enum, + WM8903_ANALOGUE_LEFT_INPUT_1, 2, linput_mux_text); -static const struct soc_enum linput_inv_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 4, 3, linput_mux_text); +static SOC_ENUM_SINGLE_DECL(linput_inv_enum, + WM8903_ANALOGUE_LEFT_INPUT_1, 4, linput_mux_text); static const char *rinput_mux_text[] = { "IN1R", "IN2R", "IN3R" }; -static const struct soc_enum rinput_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 2, 3, rinput_mux_text); +static SOC_ENUM_SINGLE_DECL(rinput_enum, + WM8903_ANALOGUE_RIGHT_INPUT_1, 2, rinput_mux_text); -static const struct soc_enum rinput_inv_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text); +static SOC_ENUM_SINGLE_DECL(rinput_inv_enum, + WM8903_ANALOGUE_RIGHT_INPUT_1, 4, rinput_mux_text); static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum lsidetone_enum = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(lsidetone_enum, + WM8903_DAC_DIGITAL_0, 2, sidetone_text); -static const struct soc_enum rsidetone_enum = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(rsidetone_enum, + WM8903_DAC_DIGITAL_0, 0, sidetone_text); static const char *adcinput_text[] = { "ADC", "DMIC" }; -static const struct soc_enum adcinput_enum = - SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); +static SOC_ENUM_SINGLE_DECL(adcinput_enum, + WM8903_CLOCK_RATE_TEST_4, 9, adcinput_text); static const char *aif_text[] = { "Left", "Right" }; -static const struct soc_enum lcapture_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(lcapture_enum, + WM8903_AUDIO_INTERFACE_0, 7, aif_text); -static const struct soc_enum rcapture_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(rcapture_enum, + WM8903_AUDIO_INTERFACE_0, 6, aif_text); -static const struct soc_enum lplay_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(lplay_enum, + WM8903_AUDIO_INTERFACE_0, 5, aif_text); -static const struct soc_enum rplay_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(rplay_enum, + WM8903_AUDIO_INTERFACE_0, 4, aif_text); static const struct snd_kcontrol_new wm8903_snd_controls[] = { @@ -1897,21 +1897,13 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903) static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int ret; wm8903->codec = codec; - codec->control_data = wm8903->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 53bbfac6a83a..49c35c36935e 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -552,18 +552,20 @@ static const char *input_mode_text[] = { "Single-Ended", "Differential Line", "Differential Mic" }; -static const struct soc_enum lin_mode = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(lin_mode, + WM8904_ANALOGUE_LEFT_INPUT_1, 0, + input_mode_text); -static const struct soc_enum rin_mode = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(rin_mode, + WM8904_ANALOGUE_RIGHT_INPUT_1, 0, + input_mode_text); static const char *hpf_mode_text[] = { "Hi-fi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum hpf_mode = - SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(hpf_mode, WM8904_ADC_DIGITAL_0, 5, + hpf_mode_text); static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -611,8 +613,7 @@ static const char *drc_path_text[] = { "ADC", "DAC" }; -static const struct soc_enum drc_path = - SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text); +static SOC_ENUM_SINGLE_DECL(drc_path, WM8904_DRC_0, 14, drc_path_text); static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = { SOC_SINGLE_TLV("Digital Playback Boost Volume", @@ -858,14 +859,14 @@ static const char *lin_text[] = { "IN1L", "IN2L", "IN3L" }; -static const struct soc_enum lin_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text); +static SOC_ENUM_SINGLE_DECL(lin_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 2, + lin_text); static const struct snd_kcontrol_new lin_mux = SOC_DAPM_ENUM("Left Capture Mux", lin_enum); -static const struct soc_enum lin_inv_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text); +static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, + lin_text); static const struct snd_kcontrol_new lin_inv_mux = SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); @@ -874,14 +875,14 @@ static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" }; -static const struct soc_enum rin_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text); +static SOC_ENUM_SINGLE_DECL(rin_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 2, + rin_text); static const struct snd_kcontrol_new rin_mux = SOC_DAPM_ENUM("Right Capture Mux", rin_enum); -static const struct soc_enum rin_inv_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text); +static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, + rin_text); static const struct snd_kcontrol_new rin_inv_mux = SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); @@ -890,26 +891,26 @@ static const char *aif_text[] = { "Left", "Right" }; -static const struct soc_enum aifoutl_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutl_enum, WM8904_AUDIO_INTERFACE_0, 7, + aif_text); static const struct snd_kcontrol_new aifoutl_mux = SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); -static const struct soc_enum aifoutr_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutr_enum, WM8904_AUDIO_INTERFACE_0, 6, + aif_text); static const struct snd_kcontrol_new aifoutr_mux = SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); -static const struct soc_enum aifinl_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinl_enum, WM8904_AUDIO_INTERFACE_0, 5, + aif_text); static const struct snd_kcontrol_new aifinl_mux = SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); -static const struct soc_enum aifinr_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinr_enum, WM8904_AUDIO_INTERFACE_0, 4, + aif_text); static const struct snd_kcontrol_new aifinr_mux = SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); @@ -991,26 +992,26 @@ static const char *out_mux_text[] = { "DAC", "Bypass" }; -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(hpl_enum, WM8904_ANALOGUE_OUT12_ZC, 3, + out_mux_text); static const struct snd_kcontrol_new hpl_mux = SOC_DAPM_ENUM("HPL Mux", hpl_enum); -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(hpr_enum, WM8904_ANALOGUE_OUT12_ZC, 2, + out_mux_text); static const struct snd_kcontrol_new hpr_mux = SOC_DAPM_ENUM("HPR Mux", hpr_enum); -static const struct soc_enum linel_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(linel_enum, WM8904_ANALOGUE_OUT12_ZC, 1, + out_mux_text); static const struct snd_kcontrol_new linel_mux = SOC_DAPM_ENUM("LINEL Mux", linel_enum); -static const struct soc_enum liner_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(liner_enum, WM8904_ANALOGUE_OUT12_ZC, 0, + out_mux_text); static const struct snd_kcontrol_new liner_mux = SOC_DAPM_ENUM("LINER Mux", liner_enum); @@ -1019,14 +1020,14 @@ static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum dacl_sidetone_enum = - SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone_enum, WM8904_DAC_DIGITAL_0, 2, + sidetone_text); static const struct snd_kcontrol_new dacl_sidetone_mux = SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum); -static const struct soc_enum dacr_sidetone_enum = - SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone_enum, WM8904_DAC_DIGITAL_0, 0, + sidetone_text); static const struct snd_kcontrol_new dacr_sidetone_mux = SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum); @@ -1981,7 +1982,7 @@ static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", wm8904->num_retune_mobile_texts); - wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts; + wm8904->retune_mobile_enum.items = wm8904->num_retune_mobile_texts; wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts; ret = snd_soc_add_codec_controls(codec, &control, 1); @@ -2022,7 +2023,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_drc_cfgs; i++) wm8904->drc_texts[i] = pdata->drc_cfgs[i].name; - wm8904->drc_enum.max = pdata->num_drc_cfgs; + wm8904->drc_enum.items = pdata->num_drc_cfgs; wm8904->drc_enum.texts = wm8904->drc_texts; ret = snd_soc_add_codec_controls(codec, &control, 1); @@ -2047,9 +2048,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = wm8904->regmap; switch (wm8904->devtype) { case WM8904: @@ -2063,12 +2061,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) return -EINVAL; } - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b404c26c1753..fc6eec9ad66b 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -154,22 +154,22 @@ static const struct reg_default wm8940_reg_defaults[] = { }; static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; -static const struct soc_enum wm8940_adc_companding_enum -= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding); -static const struct soc_enum wm8940_dac_companding_enum -= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding); +static SOC_ENUM_SINGLE_DECL(wm8940_adc_companding_enum, + WM8940_COMPANDINGCTL, 1, wm8940_companding); +static SOC_ENUM_SINGLE_DECL(wm8940_dac_companding_enum, + WM8940_COMPANDINGCTL, 3, wm8940_companding); static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"}; -static const struct soc_enum wm8940_alc_mode_enum -= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text); +static SOC_ENUM_SINGLE_DECL(wm8940_alc_mode_enum, + WM8940_ALC3, 8, wm8940_alc_mode_text); static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"}; -static const struct soc_enum wm8940_mic_bias_level_enum -= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text); +static SOC_ENUM_SINGLE_DECL(wm8940_mic_bias_level_enum, + WM8940_INPUTCTL, 8, wm8940_mic_bias_level_text); static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; -static const struct soc_enum wm8940_filter_mode_enum -= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text); +static SOC_ENUM_SINGLE_DECL(wm8940_filter_mode_enum, + WM8940_ADC, 7, wm8940_filter_mode_text); static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); @@ -712,12 +712,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8940_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 82c8ba975720..fecd4e4f4c57 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -416,22 +416,21 @@ static const char *bass_mode_text[] = { "Linear", "Adaptive", }; -static const struct soc_enum bass_mode = - SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 7, 2, bass_mode_text); +static SOC_ENUM_SINGLE_DECL(bass_mode, WM8955_BASS_CONTROL, 7, bass_mode_text); static const char *bass_cutoff_text[] = { "Low", "High" }; -static const struct soc_enum bass_cutoff = - SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 6, 2, bass_cutoff_text); +static SOC_ENUM_SINGLE_DECL(bass_cutoff, WM8955_BASS_CONTROL, 6, + bass_cutoff_text); static const char *treble_cutoff_text[] = { "High", "Low" }; -static const struct soc_enum treble_cutoff = - SOC_ENUM_SINGLE(WM8955_TREBLE_CONTROL, 6, 2, treble_cutoff_text); +static SOC_ENUM_SINGLE_DECL(treble_cutoff, WM8955_TREBLE_CONTROL, 2, + treble_cutoff_text); static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(atten_tlv, -600, 600, 0); @@ -896,14 +895,6 @@ static int wm8955_probe(struct snd_soc_codec *codec) struct wm8955_pdata *pdata = dev_get_platdata(codec->dev); int ret, i; - codec->control_data = wm8955->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++) wm8955->supplies[i].supply = wm8955_supply_names[i]; diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b7488f190d2b..7ac2e511403c 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -153,7 +153,7 @@ static int wm8958_dsp2_fw(struct snd_soc_codec *codec, const char *name, data32 &= 0xffffff; - wm8994_bulk_write(codec->control_data, + wm8994_bulk_write(wm8994->wm8994, data32 & 0xffffff, block_len / 2, (void *)(data + 8)); @@ -944,7 +944,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_mbc_cfgs; i++) wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name; - wm8994->mbc_enum.max = pdata->num_mbc_cfgs; + wm8994->mbc_enum.items = pdata->num_mbc_cfgs; wm8994->mbc_enum.texts = wm8994->mbc_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, @@ -973,7 +973,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_vss_cfgs; i++) wm8994->vss_texts[i] = pdata->vss_cfgs[i].name; - wm8994->vss_enum.max = pdata->num_vss_cfgs; + wm8994->vss_enum.items = pdata->num_vss_cfgs; wm8994->vss_enum.texts = wm8994->vss_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, @@ -1003,7 +1003,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_vss_hpf_cfgs; i++) wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name; - wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; + wm8994->vss_hpf_enum.items = pdata->num_vss_hpf_cfgs; wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, @@ -1034,7 +1034,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_enh_eq_cfgs; i++) wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name; - wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; + wm8994->enh_eq_enum.items = pdata->num_enh_eq_cfgs; wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f156010e52bc..d04e9cad445c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -976,12 +976,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8960_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 900328e28a15..9c88f04442b3 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -317,15 +317,15 @@ static const char *adc_hpf_text[] = { "Hi-fi", "Voice 1", "Voice 2", "Voice 3", }; -static const struct soc_enum adc_hpf = - SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(adc_hpf, + WM8961_ADC_DAC_CONTROL_2, 7, adc_hpf_text); static const char *dac_deemph_text[] = { "None", "32kHz", "44.1kHz", "48kHz", }; -static const struct soc_enum dac_deemph = - SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text); +static SOC_ENUM_SINGLE_DECL(dac_deemph, + WM8961_ADC_DAC_CONTROL_1, 1, dac_deemph_text); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0); @@ -385,11 +385,11 @@ static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum dacl_sidetone = - SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone, + WM8961_DSP_SIDETONE_0, 2, sidetone_text); -static const struct soc_enum dacr_sidetone = - SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone, + WM8961_DSP_SIDETONE_1, 2, sidetone_text); static const struct snd_kcontrol_new dacl_mux = SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone); @@ -836,15 +836,8 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Enable class W */ reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B); reg |= WM8961_CP_DYN_PWR_MASK; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b45b411..5522d2566c67 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1479,7 +1479,9 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) { - return regcache_sync_region(codec->control_data, + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + return regcache_sync_region(wm8962->regmap, WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER); } @@ -1658,16 +1660,16 @@ static const char *cap_hpf_mode_text[] = { "Hi-fi", "Application" }; -static const struct soc_enum cap_hpf_mode = - SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(cap_hpf_mode, + WM8962_ADC_DAC_CONTROL_2, 10, cap_hpf_mode_text); static const char *cap_lhpf_mode_text[] = { "LPF", "HPF" }; -static const struct soc_enum cap_lhpf_mode = - SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(cap_lhpf_mode, + WM8962_LHPF1, 1, cap_lhpf_mode_text); static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2014,40 +2016,40 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, static const char *st_text[] = { "None", "Left", "Right" }; -static const struct soc_enum str_enum = - SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); +static SOC_ENUM_SINGLE_DECL(str_enum, + WM8962_DAC_DSP_MIXING_1, 2, st_text); static const struct snd_kcontrol_new str_mux = SOC_DAPM_ENUM("Right Sidetone", str_enum); -static const struct soc_enum stl_enum = - SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_2, 2, 3, st_text); +static SOC_ENUM_SINGLE_DECL(stl_enum, + WM8962_DAC_DSP_MIXING_2, 2, st_text); static const struct snd_kcontrol_new stl_mux = SOC_DAPM_ENUM("Left Sidetone", stl_enum); static const char *outmux_text[] = { "DAC", "Mixer" }; -static const struct soc_enum spkoutr_enum = - SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_2, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(spkoutr_enum, + WM8962_SPEAKER_MIXER_2, 7, outmux_text); static const struct snd_kcontrol_new spkoutr_mux = SOC_DAPM_ENUM("SPKOUTR Mux", spkoutr_enum); -static const struct soc_enum spkoutl_enum = - SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_1, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(spkoutl_enum, + WM8962_SPEAKER_MIXER_1, 7, outmux_text); static const struct snd_kcontrol_new spkoutl_mux = SOC_DAPM_ENUM("SPKOUTL Mux", spkoutl_enum); -static const struct soc_enum hpoutr_enum = - SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_2, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(hpoutr_enum, + WM8962_HEADPHONE_MIXER_2, 7, outmux_text); static const struct snd_kcontrol_new hpoutr_mux = SOC_DAPM_ENUM("HPOUTR Mux", hpoutr_enum); -static const struct soc_enum hpoutl_enum = - SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_1, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(hpoutl_enum, + WM8962_HEADPHONE_MIXER_1, 7, outmux_text); static const struct snd_kcontrol_new hpoutl_mux = SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum); @@ -2884,9 +2886,13 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); - try_wait_for_completion(&wm8962->fll_lock); + reinit_completion(&wm8962->fll_lock); - pm_runtime_get_sync(codec->dev); + ret = pm_runtime_get_sync(codec->dev); + if (ret < 0) { + dev_err(codec->dev, "Failed to resume device: %d\n", ret); + return ret; + } snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | @@ -2894,8 +2900,6 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); - ret = 0; - /* This should be a massive overestimate but go even * higher if we'll error out */ @@ -2909,14 +2913,17 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (timeout == 0 && wm8962->irq) { dev_err(codec->dev, "FLL lock timed out"); - ret = -ETIMEDOUT; + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, + WM8962_FLL_ENA, 0); + pm_runtime_put(codec->dev); + return -ETIMEDOUT; } wm8962->fll_fref = Fref; wm8962->fll_fout = Fout; wm8962->fll_src = source; - return ret; + return 0; } static int wm8962_mute(struct snd_soc_dai *dai, int mute) @@ -3003,9 +3010,16 @@ static irqreturn_t wm8962_irq(int irq, void *data) unsigned int active; int reg, ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) { + dev_err(dev, "Failed to resume: %d\n", ret); + return IRQ_NONE; + } + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK, &mask); if (ret != 0) { + pm_runtime_put(dev); dev_err(dev, "Failed to read interrupt mask: %d\n", ret); return IRQ_NONE; @@ -3013,14 +3027,17 @@ static irqreturn_t wm8962_irq(int irq, void *data) ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active); if (ret != 0) { + pm_runtime_put(dev); dev_err(dev, "Failed to read interrupt: %d\n", ret); return IRQ_NONE; } active &= ~mask; - if (!active) + if (!active) { + pm_runtime_put(dev); return IRQ_NONE; + } /* Acknowledge the interrupts */ ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active); @@ -3070,6 +3087,8 @@ static irqreturn_t wm8962_irq(int irq, void *data) msecs_to_jiffies(250)); } + pm_runtime_put(dev); + return IRQ_HANDLED; } @@ -3089,6 +3108,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int irq_mask, enable; wm8962->jack = jack; @@ -3109,14 +3129,18 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) snd_soc_jack_report(wm8962->jack, 0, SND_JACK_MICROPHONE | SND_JACK_BTN_0); + snd_soc_dapm_mutex_lock(dapm); + if (jack) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "SYSCLK"); - snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "SYSCLK"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); } + snd_soc_dapm_mutex_unlock(dapm); + return 0; } EXPORT_SYMBOL_GPL(wm8962_mic_detect); @@ -3400,13 +3424,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) bool dmicclk, dmicdat; wm8962->codec = codec; - codec->control_data = wm8962->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0; wm8962->disable_nb[1].notifier_call = wm8962_regulator_event_1; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 67aba78a7ca5..09b7b4200221 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -648,12 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); - return ret; - } - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); wm8971_workq = create_workqueue("wm8971"); if (wm8971_workq == NULL) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 15f45c7bd833..0627c56fa44e 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -84,8 +84,8 @@ static const struct soc_enum wm8974_enum[] = { static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" }; -static const struct soc_enum wm8974_auxmode = - SOC_ENUM_SINGLE(WM8974_INPUT, 3, 2, wm8974_auxmode_text); +static SOC_ENUM_SINGLE_DECL(wm8974_auxmode, + WM8974_INPUT, 3, wm8974_auxmode_text); static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); @@ -593,12 +593,6 @@ static int wm8974_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8974_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index d8fc531c0e59..28ef46c91f62 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -117,21 +117,21 @@ static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; static const char *wm8978_alc3[] = {"ALC", "Limiter"}; static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; -static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, - wm8978_companding); -static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, - wm8978_companding); -static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); -static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); -static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); -static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); -static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); -static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); -static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); -static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); -static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); -static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); -static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); +static SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); @@ -975,19 +975,13 @@ static const int update_reg[] = { static int wm8978_probe(struct snd_soc_codec *codec) { struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); - int ret = 0, i; + int i; /* * Set default system clock to PLL, it is more precise, this is also the * default hardware setting */ wm8978->sysclk = WM8978_PLL; - codec->control_data = wm8978->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* * Set the update bit in all registers, that have one. This way all diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index aa41ba0dfff4..2b9bfa53efbf 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -205,49 +205,44 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" }; -static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7, - alc_sel_text); +static SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7, alc_sel_text); static const char *alc_mode_text[] = { "ALC", "Limiter" }; -static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8, - alc_mode_text); +static SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8, alc_mode_text); static const char *filter_mode_text[] = { "Audio", "Application" }; -static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7, - filter_mode_text); +static SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7, + filter_mode_text); static const char *eq_bw_text[] = { "Narrow", "Wide" }; static const char *eqmode_text[] = { "Capture", "Playback" }; -static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); +static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); static const char *eq1_cutoff_text[] = { "80Hz", "105Hz", "135Hz", "175Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5, - eq1_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5, + eq1_cutoff_text); static const char *eq2_cutoff_text[] = { "230Hz", "300Hz", "385Hz", "500Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5, - eq2_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5, eq2_cutoff_text); static const char *eq3_cutoff_text[] = { "650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5, - eq3_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5, eq3_cutoff_text); static const char *eq4_cutoff_text[] = { "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5, - eq4_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5, eq4_cutoff_text); static const char *eq5_cutoff_text[] = { "5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5, - eq5_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5, + eq5_cutoff_text); static const char *depth_3d_text[] = { "Off", @@ -267,8 +262,8 @@ static const char *depth_3d_text[] = { "93.3%", "100%" }; -static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0, - depth_3d_text); +static SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0, + depth_3d_text); static const struct snd_kcontrol_new wm8983_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8983_COMPANDING_CONTROL, @@ -1000,12 +995,6 @@ static int wm8983_probe(struct snd_soc_codec *codec) int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); @@ -1129,7 +1118,7 @@ static struct spi_driver wm8983_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8983_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1182,7 +1171,7 @@ static int __init wm8983_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8983_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8983 I2C driver: %d\n", @@ -1202,7 +1191,7 @@ module_init(wm8983_modinit); static void __exit wm8983_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8983_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 271b517911a4..5473dc969585 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -226,52 +226,48 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" }; -static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7, - alc_sel_text); +static SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7, alc_sel_text); static const char *alc_mode_text[] = { "ALC", "Limiter" }; -static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8, - alc_mode_text); +static SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8, alc_mode_text); static const char *filter_mode_text[] = { "Audio", "Application" }; -static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7, - filter_mode_text); +static SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7, + filter_mode_text); static const char *eq_bw_text[] = { "Narrow", "Wide" }; static const char *eqmode_text[] = { "Capture", "Playback" }; -static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); +static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); static const char *eq1_cutoff_text[] = { "80Hz", "105Hz", "135Hz", "175Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5, - eq1_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5, + eq1_cutoff_text); static const char *eq2_cutoff_text[] = { "230Hz", "300Hz", "385Hz", "500Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5, - eq2_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5, eq2_cutoff_text); static const char *eq3_cutoff_text[] = { "650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5, - eq3_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5, + eq3_cutoff_text); static const char *eq4_cutoff_text[] = { "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5, - eq4_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5, eq4_cutoff_text); static const char *eq5_cutoff_text[] = { "5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5, +static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5, eq5_cutoff_text); static const char *speaker_mode_text[] = { "Class A/B", "Class D" }; -static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text); +static SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text); static const char *depth_3d_text[] = { "Off", @@ -291,8 +287,7 @@ static const char *depth_3d_text[] = { "93.3%", "100%" }; -static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, - depth_3d_text); +static SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, depth_3d_text); static const struct snd_kcontrol_new wm8985_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8985_COMPANDING_CONTROL, @@ -1000,13 +995,6 @@ static int wm8985_probe(struct snd_soc_codec *codec) int ret; wm8985 = snd_soc_codec_get_drvdata(codec); - codec->control_data = wm8985->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } for (i = 0; i < ARRAY_SIZE(wm8985->supplies); i++) wm8985->supplies[i].supply = wm8985_supply_names[i]; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index a55e1c2c382e..3a1ae4f5164d 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -116,7 +116,7 @@ static bool wm8988_writeable(struct device *dev, unsigned int reg) struct wm8988_priv { struct regmap *regmap; unsigned int sysclk; - struct snd_pcm_hw_constraint_list *sysclk_constraints; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; }; #define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) @@ -126,46 +126,46 @@ struct wm8988_priv { */ static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"}; -static const struct soc_enum bass_boost = - SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt); +static SOC_ENUM_SINGLE_DECL(bass_boost, + WM8988_BASS, 7, bass_boost_txt); static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; -static const struct soc_enum bass_filter = - SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt); +static SOC_ENUM_SINGLE_DECL(bass_filter, + WM8988_BASS, 6, bass_filter_txt); static const char *treble_txt[] = {"8kHz", "4kHz"}; -static const struct soc_enum treble = - SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt); +static SOC_ENUM_SINGLE_DECL(treble, + WM8988_TREBLE, 6, treble_txt); static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"}; -static const struct soc_enum stereo_3d_lc = - SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt); +static SOC_ENUM_SINGLE_DECL(stereo_3d_lc, + WM8988_3D, 5, stereo_3d_lc_txt); static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"}; -static const struct soc_enum stereo_3d_uc = - SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt); +static SOC_ENUM_SINGLE_DECL(stereo_3d_uc, + WM8988_3D, 6, stereo_3d_uc_txt); static const char *stereo_3d_func_txt[] = {"Capture", "Playback"}; -static const struct soc_enum stereo_3d_func = - SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt); +static SOC_ENUM_SINGLE_DECL(stereo_3d_func, + WM8988_3D, 7, stereo_3d_func_txt); static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"}; -static const struct soc_enum alc_func = - SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt); +static SOC_ENUM_SINGLE_DECL(alc_func, + WM8988_ALC1, 7, alc_func_txt); static const char *ng_type_txt[] = {"Constant PGA Gain", "Mute ADC Output"}; -static const struct soc_enum ng_type = - SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt); +static SOC_ENUM_SINGLE_DECL(ng_type, + WM8988_NGATE, 1, ng_type_txt); static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const struct soc_enum deemph = - SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt); +static SOC_ENUM_SINGLE_DECL(deemph, + WM8988_ADCDAC, 1, deemph_txt); static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert", "L + R Invert"}; -static const struct soc_enum adcpol = - SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt); +static SOC_ENUM_SINGLE_DECL(adcpol, + WM8988_ADCDAC, 5, adcpol_txt); static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); @@ -317,16 +317,16 @@ static const struct snd_kcontrol_new wm8988_right_pga_controls = /* Differential Mux */ static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"}; -static const struct soc_enum diffmux = - SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel); +static SOC_ENUM_SINGLE_DECL(diffmux, + WM8988_ADCIN, 8, wm8988_diff_sel); static const struct snd_kcontrol_new wm8988_diffmux_controls = SOC_DAPM_ENUM("Route", diffmux); /* Mono ADC Mux */ static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)", "Mono (Right)", "Digital Mono"}; -static const struct soc_enum monomux = - SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux); +static SOC_ENUM_SINGLE_DECL(monomux, + WM8988_ADCIN, 6, wm8988_mono_mux); static const struct snd_kcontrol_new wm8988_monomux_controls = SOC_DAPM_ENUM("Route", monomux); @@ -521,30 +521,30 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ -static unsigned int rates_12288[] = { +static const unsigned int rates_12288[] = { 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_12288 = { +static const struct snd_pcm_hw_constraint_list constraints_12288 = { .count = ARRAY_SIZE(rates_12288), .list = rates_12288, }; -static unsigned int rates_112896[] = { +static const unsigned int rates_112896[] = { 8000, 11025, 22050, 44100, }; -static struct snd_pcm_hw_constraint_list constraints_112896 = { +static const struct snd_pcm_hw_constraint_list constraints_112896 = { .count = ARRAY_SIZE(rates_112896), .list = rates_112896, }; -static unsigned int rates_12[] = { +static const unsigned int rates_12[] = { 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, 48000, 88235, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_12 = { +static const struct snd_pcm_hw_constraint_list constraints_12 = { .count = ARRAY_SIZE(rates_12), .list = rates_12, }; @@ -810,16 +810,8 @@ static int wm8988_resume(struct snd_soc_codec *codec) static int wm8988_probe(struct snd_soc_codec *codec) { - struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); int ret = 0; - codec->control_data = wm8988->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8988_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 0ccd4d8d043b..c413c1991453 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -157,26 +157,23 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8990_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8990_left_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, - WM8990_ADC_TO_DACL_SHIFT, - WM8990_ADC_TO_DACL_MASK, - wm8990_digital_sidetone); - -static const struct soc_enum wm8990_right_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, - WM8990_ADC_TO_DACR_SHIFT, - WM8990_ADC_TO_DACR_MASK, - wm8990_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8990_left_digital_sidetone_enum, + WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACL_SHIFT, + wm8990_digital_sidetone); + +static SOC_ENUM_SINGLE_DECL(wm8990_right_digital_sidetone_enum, + WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACR_SHIFT, + wm8990_digital_sidetone); static const char *wm8990_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8990_right_adcmode_enum = -SOC_ENUM_SINGLE(WM8990_ADC_CTRL, - WM8990_ADC_HPF_CUT_SHIFT, - WM8990_ADC_HPF_CUT_MASK, - wm8990_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8990_right_adcmode_enum, + WM8990_ADC_CTRL, + WM8990_ADC_HPF_CUT_SHIFT, + wm8990_adcmode); static const struct snd_kcontrol_new wm8990_snd_controls[] = { /* INMIXL */ @@ -475,9 +472,9 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, static const char *wm8990_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8990_ainlmux_enum = -SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, - ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8990_ainlmux_enum, + WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, + wm8990_ainlmux); static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); @@ -488,9 +485,9 @@ SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); static const char *wm8990_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8990_ainrmux_enum = -SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, - ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8990_ainrmux_enum, + WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, + wm8990_ainrmux); static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum); @@ -1292,14 +1289,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) */ static int wm8990_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); - return ret; - } - wm8990_reset(codec); /* charge output caps */ diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index dba0306c42a5..844cc4a60d66 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -171,26 +171,23 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8991_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8991_left_digital_sidetone_enum = - SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, - WM8991_ADC_TO_DACL_SHIFT, - WM8991_ADC_TO_DACL_MASK, - wm8991_digital_sidetone); - -static const struct soc_enum wm8991_right_digital_sidetone_enum = - SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, - WM8991_ADC_TO_DACR_SHIFT, - WM8991_ADC_TO_DACR_MASK, - wm8991_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8991_left_digital_sidetone_enum, + WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACL_SHIFT, + wm8991_digital_sidetone); + +static SOC_ENUM_SINGLE_DECL(wm8991_right_digital_sidetone_enum, + WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACR_SHIFT, + wm8991_digital_sidetone); static const char *wm8991_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8991_right_adcmode_enum = - SOC_ENUM_SINGLE(WM8991_ADC_CTRL, - WM8991_ADC_HPF_CUT_SHIFT, - WM8991_ADC_HPF_CUT_MASK, - wm8991_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8991_right_adcmode_enum, + WM8991_ADC_CTRL, + WM8991_ADC_HPF_CUT_SHIFT, + wm8991_adcmode); static const struct snd_kcontrol_new wm8991_snd_controls[] = { /* INMIXL */ @@ -486,9 +483,9 @@ static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = { static const char *wm8991_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8991_ainlmux_enum = - SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, - ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8991_ainlmux_enum, + WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, + wm8991_ainlmux); static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum); @@ -499,9 +496,9 @@ static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = static const char *wm8991_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8991_ainrmux_enum = - SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, - ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8991_ainrmux_enum, + WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, + wm8991_ainrmux); static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum); @@ -1251,17 +1248,6 @@ static int wm8991_remove(struct snd_soc_codec *codec) static int wm8991_probe(struct snd_soc_codec *codec) { - struct wm8991_priv *wm8991; - int ret; - - wm8991 = snd_soc_codec_get_drvdata(codec); - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a0f3ef..f825dc04ebe1 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -646,8 +646,8 @@ static const char *dac_deemph_text[] = { "48kHz", }; -static const struct soc_enum dac_deemph = - SOC_ENUM_SINGLE(WM8993_DAC_CTRL, 4, 4, dac_deemph_text); +static SOC_ENUM_SINGLE_DECL(dac_deemph, + WM8993_DAC_CTRL, 4, dac_deemph_text); static const char *adc_hpf_text[] = { "Hi-Fi", @@ -656,16 +656,16 @@ static const char *adc_hpf_text[] = { "Voice 3", }; -static const struct soc_enum adc_hpf = - SOC_ENUM_SINGLE(WM8993_ADC_CTRL, 5, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(adc_hpf, + WM8993_ADC_CTRL, 5, adc_hpf_text); static const char *drc_path_text[] = { "ADC", "DAC" }; -static const struct soc_enum drc_path = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 14, 2, drc_path_text); +static SOC_ENUM_SINGLE_DECL(drc_path, + WM8993_DRC_CONTROL_1, 14, drc_path_text); static const char *drc_r0_text[] = { "1", @@ -676,8 +676,8 @@ static const char *drc_r0_text[] = { "0", }; -static const struct soc_enum drc_r0 = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 8, 6, drc_r0_text); +static SOC_ENUM_SINGLE_DECL(drc_r0, + WM8993_DRC_CONTROL_3, 8, drc_r0_text); static const char *drc_r1_text[] = { "1", @@ -687,8 +687,8 @@ static const char *drc_r1_text[] = { "0", }; -static const struct soc_enum drc_r1 = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_4, 13, 5, drc_r1_text); +static SOC_ENUM_SINGLE_DECL(drc_r1, + WM8993_DRC_CONTROL_4, 13, drc_r1_text); static const char *drc_attack_text[] = { "Reserved", @@ -705,8 +705,8 @@ static const char *drc_attack_text[] = { "185.6ms", }; -static const struct soc_enum drc_attack = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 12, 12, drc_attack_text); +static SOC_ENUM_SINGLE_DECL(drc_attack, + WM8993_DRC_CONTROL_2, 12, drc_attack_text); static const char *drc_decay_text[] = { "186ms", @@ -720,16 +720,16 @@ static const char *drc_decay_text[] = { "47.56ms", }; -static const struct soc_enum drc_decay = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 8, 9, drc_decay_text); +static SOC_ENUM_SINGLE_DECL(drc_decay, + WM8993_DRC_CONTROL_2, 8, drc_decay_text); static const char *drc_ff_text[] = { "5 samples", "9 samples", }; -static const struct soc_enum drc_ff = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 7, 2, drc_ff_text); +static SOC_ENUM_SINGLE_DECL(drc_ff, + WM8993_DRC_CONTROL_3, 7, drc_ff_text); static const char *drc_qr_rate_text[] = { "0.725ms", @@ -737,8 +737,8 @@ static const char *drc_qr_rate_text[] = { "5.8ms", }; -static const struct soc_enum drc_qr_rate = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 0, 3, drc_qr_rate_text); +static SOC_ENUM_SINGLE_DECL(drc_qr_rate, + WM8993_DRC_CONTROL_3, 0, drc_qr_rate_text); static const char *drc_smooth_text[] = { "Low", @@ -746,8 +746,8 @@ static const char *drc_smooth_text[] = { "High", }; -static const struct soc_enum drc_smooth = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 4, 3, drc_smooth_text); +static SOC_ENUM_SINGLE_DECL(drc_smooth, + WM8993_DRC_CONTROL_1, 4, drc_smooth_text); static const struct snd_kcontrol_new wm8993_snd_controls[] = { SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE, @@ -841,26 +841,26 @@ static const char *aif_text[] = { "Left", "Right" }; -static const struct soc_enum aifoutl_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutl_enum, + WM8993_AUDIO_INTERFACE_1, 15, aif_text); static const struct snd_kcontrol_new aifoutl_mux = SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); -static const struct soc_enum aifoutr_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutr_enum, + WM8993_AUDIO_INTERFACE_1, 14, aif_text); static const struct snd_kcontrol_new aifoutr_mux = SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); -static const struct soc_enum aifinl_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinl_enum, + WM8993_AUDIO_INTERFACE_2, 15, aif_text); static const struct snd_kcontrol_new aifinl_mux = SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); -static const struct soc_enum aifinr_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinr_enum, + WM8993_AUDIO_INTERFACE_2, 14, aif_text); static const struct snd_kcontrol_new aifinr_mux = SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); @@ -869,14 +869,14 @@ static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum sidetonel_enum = - SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetonel_enum, + WM8993_DIGITAL_SIDE_TONE, 2, sidetone_text); static const struct snd_kcontrol_new sidetonel_mux = SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum); -static const struct soc_enum sidetoner_enum = - SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetoner_enum, + WM8993_DIGITAL_SIDE_TONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetoner_mux = SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum); @@ -1493,13 +1493,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.dcs_codes_r = -2; wm8993->hubs_data.series_startup = 1; - codec->control_data = wm8993->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Latch volume update bits and default ZC on */ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, WM8993_DAC_VU, WM8993_DAC_VU); @@ -1559,10 +1552,7 @@ static int wm8993_probe(struct snd_soc_codec *codec) static int wm8993_remove(struct snd_soc_codec *codec) { - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); - regulator_bulk_free(ARRAY_SIZE(wm8993->supplies), wm8993->supplies); return 0; } diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cbc4603..6303537f54c6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -265,21 +265,21 @@ static const char *sidetone_hpf_text[] = { "2.7kHz", "1.35kHz", "675Hz", "370Hz", "180Hz", "90Hz", "45Hz" }; -static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 7, 7, sidetone_hpf_text); +static SOC_ENUM_SINGLE_DECL(sidetone_hpf, + WM8994_SIDETONE, 7, sidetone_hpf_text); static const char *adc_hpf_text[] = { "HiFi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum aif1adc1_hpf = - SOC_ENUM_SINGLE(WM8994_AIF1_ADC1_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif1adc1_hpf, + WM8994_AIF1_ADC1_FILTERS, 13, adc_hpf_text); -static const struct soc_enum aif1adc2_hpf = - SOC_ENUM_SINGLE(WM8994_AIF1_ADC2_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif1adc2_hpf, + WM8994_AIF1_ADC2_FILTERS, 13, adc_hpf_text); -static const struct soc_enum aif2adc_hpf = - SOC_ENUM_SINGLE(WM8994_AIF2_ADC_FILTERS, 13, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(aif2adc_hpf, + WM8994_AIF2_ADC_FILTERS, 13, adc_hpf_text); static const DECLARE_TLV_DB_SCALE(aif_tlv, 0, 600, 0); static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); @@ -501,39 +501,39 @@ static const char *aif_chan_src_text[] = { "Left", "Right" }; -static const struct soc_enum aif1adcl_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1adcl_src, + WM8994_AIF1_CONTROL_1, 15, aif_chan_src_text); -static const struct soc_enum aif1adcr_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_1, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1adcr_src, + WM8994_AIF1_CONTROL_1, 14, aif_chan_src_text); -static const struct soc_enum aif2adcl_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2adcl_src, + WM8994_AIF2_CONTROL_1, 15, aif_chan_src_text); -static const struct soc_enum aif2adcr_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_1, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2adcr_src, + WM8994_AIF2_CONTROL_1, 14, aif_chan_src_text); -static const struct soc_enum aif1dacl_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1dacl_src, + WM8994_AIF1_CONTROL_2, 15, aif_chan_src_text); -static const struct soc_enum aif1dacr_src = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif1dacr_src, + WM8994_AIF1_CONTROL_2, 14, aif_chan_src_text); -static const struct soc_enum aif2dacl_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 15, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacl_src, + WM8994_AIF2_CONTROL_2, 15, aif_chan_src_text); -static const struct soc_enum aif2dacr_src = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, 14, 2, aif_chan_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacr_src, + WM8994_AIF2_CONTROL_2, 14, aif_chan_src_text); static const char *osr_text[] = { "Low Power", "High Performance", }; -static const struct soc_enum dac_osr = - SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(dac_osr, + WM8994_OVERSAMPLING, 0, osr_text); -static const struct soc_enum adc_osr = - SOC_ENUM_SINGLE(WM8994_OVERSAMPLING, 1, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(adc_osr, + WM8994_OVERSAMPLING, 1, osr_text); static const struct snd_kcontrol_new wm8994_snd_controls[] = { SOC_DOUBLE_R_TLV("AIF1ADC1 Volume", WM8994_AIF1_ADC1_LEFT_VOLUME, @@ -690,17 +690,20 @@ static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; -static const struct soc_enum wm8958_aif1dac1_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF1_DAC1_NOISE_GATE, - WM8958_AIF1DAC1_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac1_ng_hold, + WM8958_AIF1_DAC1_NOISE_GATE, + WM8958_AIF1DAC1_NG_THR_SHIFT, + wm8958_ng_text); -static const struct soc_enum wm8958_aif1dac2_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF1_DAC2_NOISE_GATE, - WM8958_AIF1DAC2_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif1dac2_ng_hold, + WM8958_AIF1_DAC2_NOISE_GATE, + WM8958_AIF1DAC2_NG_THR_SHIFT, + wm8958_ng_text); -static const struct soc_enum wm8958_aif2dac_ng_hold = - SOC_ENUM_SINGLE(WM8958_AIF2_DAC_NOISE_GATE, - WM8958_AIF2DAC_NG_THR_SHIFT, 4, wm8958_ng_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif2dac_ng_hold, + WM8958_AIF2_DAC_NOISE_GATE, + WM8958_AIF2DAC_NG_THR_SHIFT, + wm8958_ng_text); static const struct snd_kcontrol_new wm8958_snd_controls[] = { SOC_SINGLE_TLV("AIF3 Boost Volume", WM8958_AIF3_CONTROL_2, 10, 3, 0, aif_tlv), @@ -1341,8 +1344,7 @@ static const char *adc_mux_text[] = { "DMIC", }; -static const struct soc_enum adc_enum = - SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); @@ -1478,14 +1480,14 @@ static const char *sidetone_text[] = { "ADC/DMIC1", "DMIC2", }; -static const struct soc_enum sidetone1_enum = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone1_enum, + WM8994_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetone1_mux = SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); -static const struct soc_enum sidetone2_enum = - SOC_ENUM_SINGLE(WM8994_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone2_enum, + WM8994_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new sidetone2_mux = SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); @@ -1498,22 +1500,24 @@ static const char *loopback_text[] = { "None", "ADCDAT", }; -static const struct soc_enum aif1_loopback_enum = - SOC_ENUM_SINGLE(WM8994_AIF1_CONTROL_2, WM8994_AIF1_LOOPBACK_SHIFT, 2, - loopback_text); +static SOC_ENUM_SINGLE_DECL(aif1_loopback_enum, + WM8994_AIF1_CONTROL_2, + WM8994_AIF1_LOOPBACK_SHIFT, + loopback_text); static const struct snd_kcontrol_new aif1_loopback = SOC_DAPM_ENUM("AIF1 Loopback", aif1_loopback_enum); -static const struct soc_enum aif2_loopback_enum = - SOC_ENUM_SINGLE(WM8994_AIF2_CONTROL_2, WM8994_AIF2_LOOPBACK_SHIFT, 2, - loopback_text); +static SOC_ENUM_SINGLE_DECL(aif2_loopback_enum, + WM8994_AIF2_CONTROL_2, + WM8994_AIF2_LOOPBACK_SHIFT, + loopback_text); static const struct snd_kcontrol_new aif2_loopback = SOC_DAPM_ENUM("AIF2 Loopback", aif2_loopback_enum); -static const struct soc_enum aif1dac_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 0, 2, aif1dac_text); +static SOC_ENUM_SINGLE_DECL(aif1dac_enum, + WM8994_POWER_MANAGEMENT_6, 0, aif1dac_text); static const struct snd_kcontrol_new aif1dac_mux = SOC_DAPM_ENUM("AIF1DAC Mux", aif1dac_enum); @@ -1522,8 +1526,8 @@ static const char *aif2dac_text[] = { "AIF2DACDAT", "AIF3DACDAT", }; -static const struct soc_enum aif2dac_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 1, 2, aif2dac_text); +static SOC_ENUM_SINGLE_DECL(aif2dac_enum, + WM8994_POWER_MANAGEMENT_6, 1, aif2dac_text); static const struct snd_kcontrol_new aif2dac_mux = SOC_DAPM_ENUM("AIF2DAC Mux", aif2dac_enum); @@ -1532,8 +1536,8 @@ static const char *aif2adc_text[] = { "AIF2ADCDAT", "AIF3DACDAT", }; -static const struct soc_enum aif2adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 2, 2, aif2adc_text); +static SOC_ENUM_SINGLE_DECL(aif2adc_enum, + WM8994_POWER_MANAGEMENT_6, 2, aif2adc_text); static const struct snd_kcontrol_new aif2adc_mux = SOC_DAPM_ENUM("AIF2ADC Mux", aif2adc_enum); @@ -1542,14 +1546,14 @@ static const char *aif3adc_text[] = { "AIF1ADCDAT", "AIF2ADCDAT", "AIF2DACDAT", "Mono PCM", }; -static const struct soc_enum wm8994_aif3adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 3, aif3adc_text); +static SOC_ENUM_SINGLE_DECL(wm8994_aif3adc_enum, + WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text); static const struct snd_kcontrol_new wm8994_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8994_aif3adc_enum); -static const struct soc_enum wm8958_aif3adc_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 3, 4, aif3adc_text); +static SOC_ENUM_SINGLE_DECL(wm8958_aif3adc_enum, + WM8994_POWER_MANAGEMENT_6, 3, aif3adc_text); static const struct snd_kcontrol_new wm8958_aif3adc_mux = SOC_DAPM_ENUM("AIF3ADC Mux", wm8958_aif3adc_enum); @@ -1558,8 +1562,8 @@ static const char *mono_pcm_out_text[] = { "None", "AIF2ADCL", "AIF2ADCR", }; -static const struct soc_enum mono_pcm_out_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 9, 3, mono_pcm_out_text); +static SOC_ENUM_SINGLE_DECL(mono_pcm_out_enum, + WM8994_POWER_MANAGEMENT_6, 9, mono_pcm_out_text); static const struct snd_kcontrol_new mono_pcm_out_mux = SOC_DAPM_ENUM("Mono PCM Out Mux", mono_pcm_out_enum); @@ -1569,14 +1573,14 @@ static const char *aif2dac_src_text[] = { }; /* Note that these two control shouldn't be simultaneously switched to AIF3 */ -static const struct soc_enum aif2dacl_src_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 7, 2, aif2dac_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacl_src_enum, + WM8994_POWER_MANAGEMENT_6, 7, aif2dac_src_text); static const struct snd_kcontrol_new aif2dacl_src_mux = SOC_DAPM_ENUM("AIF2DACL Mux", aif2dacl_src_enum); -static const struct soc_enum aif2dacr_src_enum = - SOC_ENUM_SINGLE(WM8994_POWER_MANAGEMENT_6, 8, 2, aif2dac_src_text); +static SOC_ENUM_SINGLE_DECL(aif2dacr_src_enum, + WM8994_POWER_MANAGEMENT_6, 8, aif2dac_src_text); static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); @@ -2549,43 +2553,52 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; switch (mode) { case WM8994_VMID_NORMAL: + snd_soc_dapm_mutex_lock(dapm); + if (wm8994->hubs.lineout1_se) { - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT1N Driver"); - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT1P Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT1N Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT1P Driver"); } if (wm8994->hubs.lineout2_se) { - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT2N Driver"); - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT2P Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT2N Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT2P Driver"); } /* Do the sync with the old mode to allow it to clean up */ - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync_unlocked(dapm); wm8994->vmid_mode = mode; + + snd_soc_dapm_mutex_unlock(dapm); break; case WM8994_VMID_FORCE: + snd_soc_dapm_mutex_lock(dapm); + if (wm8994->hubs.lineout1_se) { - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT1N Driver"); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT1P Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT1N Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT1P Driver"); } if (wm8994->hubs.lineout2_se) { - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT2N Driver"); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT2P Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT2N Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT2P Driver"); } wm8994->vmid_mode = mode; - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); break; default: @@ -3237,7 +3250,7 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", wm8994->num_retune_mobile_texts); - wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; + wm8994->retune_mobile_enum.items = wm8994->num_retune_mobile_texts; wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, @@ -3293,7 +3306,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) for (i = 0; i < pdata->num_drc_cfgs; i++) wm8994->drc_texts[i] = pdata->drc_cfgs[i].name; - wm8994->drc_enum.max = pdata->num_drc_cfgs; + wm8994->drc_enum.items = pdata->num_drc_cfgs; wm8994->drc_enum.texts = wm8994->drc_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, @@ -3985,9 +3998,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) int ret, i; wm8994->hubs.codec = codec; - codec->control_data = control->regmap; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, control->regmap); mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caff1783..d3152cf5bd56 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -423,24 +423,24 @@ static const char *in1l_text[] = { "Differential", "Single-ended IN1LN", "Single-ended IN1LP" }; -static const SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL, - 2, in1l_text); +static SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL, + 2, in1l_text); static const char *in1r_text[] = { "Differential", "Single-ended IN1RN", "Single-ended IN1RP" }; -static const SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL, - 0, in1r_text); +static SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL, + 0, in1r_text); static const char *dmic_src_text[] = { "DMICDAT1", "DMICDAT2", "DMICDAT3" }; -static const SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5, - 8, dmic_src_text); -static const SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5, - 6, dmic_src_text); +static SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5, + 8, dmic_src_text); +static SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5, + 6, dmic_src_text); static const struct snd_kcontrol_new wm8995_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC1 Volume", WM8995_DAC1_LEFT_VOLUME, @@ -561,10 +561,8 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec; - struct wm8995_priv *wm8995; codec = w->codec; - wm8995 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -783,14 +781,12 @@ static const char *sidetone_text[] = { "ADC/DMIC1", "DMIC2", }; -static const struct soc_enum sidetone1_enum = - SOC_ENUM_SINGLE(WM8995_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone1_enum, WM8995_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetone1_mux = SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); -static const struct soc_enum sidetone2_enum = - SOC_ENUM_SINGLE(WM8995_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone2_enum, WM8995_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new sidetone2_mux = SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); @@ -886,8 +882,7 @@ static const char *adc_mux_text[] = { "DMIC", }; -static const struct soc_enum adc_enum = - SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); @@ -899,14 +894,14 @@ static const char *spk_src_text[] = { "DAC1L", "DAC1R", "DAC2L", "DAC2R" }; -static const SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1, - 0, spk_src_text); -static const SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1, - 0, spk_src_text); -static const SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2, - 0, spk_src_text); -static const SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2, - 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1, + 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1, + 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2, + 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2, + 0, spk_src_text); static const struct snd_kcontrol_new spk1l_mux = SOC_DAPM_ENUM("SPK1L SRC", spk1l_src_enum); @@ -2047,13 +2042,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; - codec->control_data = wm8995->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) wm8995->supplies[i].supply = wm8995_supply_names[i]; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1a7655b0aa22..c6cbb3b8ace9 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -311,28 +311,28 @@ static const char *sidetone_hpf_text[] = { "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" }; -static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); +static SOC_ENUM_SINGLE_DECL(sidetone_hpf, + WM8996_SIDETONE, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" }; -static const struct soc_enum dsp1tx_hpf_mode = - SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 3, 3, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_mode, + WM8996_DSP1_TX_FILTERS, 3, hpf_mode_text); -static const struct soc_enum dsp2tx_hpf_mode = - SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 3, 3, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_mode, + WM8996_DSP2_TX_FILTERS, 3, hpf_mode_text); static const char *hpf_cutoff_text[] = { "50Hz", "75Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum dsp1tx_hpf_cutoff = - SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 0, 7, hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_cutoff, + WM8996_DSP1_TX_FILTERS, 0, hpf_cutoff_text); -static const struct soc_enum dsp2tx_hpf_cutoff = - SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 0, 7, hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_cutoff, + WM8996_DSP2_TX_FILTERS, 0, hpf_cutoff_text); static void wm8996_set_retune_mobile(struct snd_soc_codec *codec, int block) { @@ -780,14 +780,14 @@ static const char *sidetone_text[] = { "IN1", "IN2", }; -static const struct soc_enum left_sidetone_enum = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(left_sidetone_enum, + WM8996_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new left_sidetone = SOC_DAPM_ENUM("Left Sidetone", left_sidetone_enum); -static const struct soc_enum right_sidetone_enum = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(right_sidetone_enum, + WM8996_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new right_sidetone = SOC_DAPM_ENUM("Right Sidetone", right_sidetone_enum); @@ -796,14 +796,14 @@ static const char *spk_text[] = { "DAC1L", "DAC1R", "DAC2L", "DAC2R" }; -static const struct soc_enum spkl_enum = - SOC_ENUM_SINGLE(WM8996_LEFT_PDM_SPEAKER, 0, 4, spk_text); +static SOC_ENUM_SINGLE_DECL(spkl_enum, + WM8996_LEFT_PDM_SPEAKER, 0, spk_text); static const struct snd_kcontrol_new spkl_mux = SOC_DAPM_ENUM("SPKL", spkl_enum); -static const struct soc_enum spkr_enum = - SOC_ENUM_SINGLE(WM8996_RIGHT_PDM_SPEAKER, 0, 4, spk_text); +static SOC_ENUM_SINGLE_DECL(spkr_enum, + WM8996_RIGHT_PDM_SPEAKER, 0, spk_text); static const struct snd_kcontrol_new spkr_mux = SOC_DAPM_ENUM("SPKR", spkr_enum); @@ -812,8 +812,8 @@ static const char *dsp1rx_text[] = { "AIF1", "AIF2" }; -static const struct soc_enum dsp1rx_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 0, 2, dsp1rx_text); +static SOC_ENUM_SINGLE_DECL(dsp1rx_enum, + WM8996_POWER_MANAGEMENT_8, 0, dsp1rx_text); static const struct snd_kcontrol_new dsp1rx = SOC_DAPM_ENUM("DSP1RX", dsp1rx_enum); @@ -822,8 +822,8 @@ static const char *dsp2rx_text[] = { "AIF2", "AIF1" }; -static const struct soc_enum dsp2rx_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 4, 2, dsp2rx_text); +static SOC_ENUM_SINGLE_DECL(dsp2rx_enum, + WM8996_POWER_MANAGEMENT_8, 4, dsp2rx_text); static const struct snd_kcontrol_new dsp2rx = SOC_DAPM_ENUM("DSP2RX", dsp2rx_enum); @@ -832,8 +832,8 @@ static const char *aif2tx_text[] = { "DSP2", "DSP1", "AIF1" }; -static const struct soc_enum aif2tx_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 6, 3, aif2tx_text); +static SOC_ENUM_SINGLE_DECL(aif2tx_enum, + WM8996_POWER_MANAGEMENT_8, 6, aif2tx_text); static const struct snd_kcontrol_new aif2tx = SOC_DAPM_ENUM("AIF2TX", aif2tx_enum); @@ -842,14 +842,14 @@ static const char *inmux_text[] = { "ADC", "DMIC1", "DMIC2" }; -static const struct soc_enum in1_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 0, 3, inmux_text); +static SOC_ENUM_SINGLE_DECL(in1_enum, + WM8996_POWER_MANAGEMENT_7, 0, inmux_text); static const struct snd_kcontrol_new in1_mux = SOC_DAPM_ENUM("IN1 Mux", in1_enum); -static const struct soc_enum in2_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 4, 3, inmux_text); +static SOC_ENUM_SINGLE_DECL(in2_enum, + WM8996_POWER_MANAGEMENT_7, 4, inmux_text); static const struct snd_kcontrol_new in2_mux = SOC_DAPM_ENUM("IN2 Mux", in2_enum); @@ -1608,8 +1608,8 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, msleep(5); } - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(wm8996->regmap, false); + regcache_sync(wm8996->regmap); } /* Bypass the MICBIASes for lowest power */ @@ -1620,10 +1620,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); @@ -2251,6 +2251,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8996_polarity_fn polarity_cb) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; wm8996->jack = jack; wm8996->detecting = true; @@ -2267,8 +2268,12 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, WM8996_MICB2_DISCH, 0); /* LDO2 powers the microphones, SYSCLK clocks detection */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK"); + + snd_soc_dapm_mutex_unlock(dapm); /* We start off just enabling microphone detection - even a * plain headphone will trigger detection. @@ -2595,7 +2600,7 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", wm8996->num_retune_mobile_texts); - wm8996->retune_mobile_enum.max = wm8996->num_retune_mobile_texts; + wm8996->retune_mobile_enum.items = wm8996->num_retune_mobile_texts; wm8996->retune_mobile_enum.texts = wm8996->retune_mobile_texts; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); @@ -2628,14 +2633,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->dcs_done); init_completion(&wm8996->fll_lock); - codec->control_data = wm8996->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err; - } - if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else @@ -2674,13 +2671,11 @@ static int wm8996_probe(struct snd_soc_codec *codec) } else { dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; } } return 0; - -err: - return ret; } static int wm8996_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 555115ee2159..004186b6bd48 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -86,7 +86,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; @@ -123,10 +123,12 @@ static const unsigned int wm8997_osr_val[] = { static const struct soc_enum wm8997_hpout_osr[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT1_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm8997_osr_text), wm8997_osr_text, wm8997_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT3_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm8997_osr_text), wm8997_osr_text, wm8997_osr_val), }; @@ -170,15 +172,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -190,6 +185,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -201,6 +198,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -212,6 +211,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, @@ -1052,9 +1053,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 0982c1d38ec4..d18eff31fbbc 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -268,8 +268,7 @@ static const char *drc_high_text[] = { "0", }; -static const struct soc_enum drc_high = - SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text); +static SOC_ENUM_SINGLE_DECL(drc_high, WM9081_DRC_3, 3, drc_high_text); static const char *drc_low_text[] = { "1", @@ -279,8 +278,7 @@ static const char *drc_low_text[] = { "0", }; -static const struct soc_enum drc_low = - SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text); +static SOC_ENUM_SINGLE_DECL(drc_low, WM9081_DRC_3, 0, drc_low_text); static const char *drc_atk_text[] = { "181us", @@ -297,8 +295,7 @@ static const char *drc_atk_text[] = { "185.6ms", }; -static const struct soc_enum drc_atk = - SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text); +static SOC_ENUM_SINGLE_DECL(drc_atk, WM9081_DRC_2, 12, drc_atk_text); static const char *drc_dcy_text[] = { "186ms", @@ -312,8 +309,7 @@ static const char *drc_dcy_text[] = { "47.56s", }; -static const struct soc_enum drc_dcy = - SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text); +static SOC_ENUM_SINGLE_DECL(drc_dcy, WM9081_DRC_2, 8, drc_dcy_text); static const char *drc_qr_dcy_text[] = { "0.725ms", @@ -321,8 +317,7 @@ static const char *drc_qr_dcy_text[] = { "5.8ms", }; -static const struct soc_enum drc_qr_dcy = - SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text); +static SOC_ENUM_SINGLE_DECL(drc_qr_dcy, WM9081_DRC_2, 4, drc_qr_dcy_text); static const char *dac_deemph_text[] = { "None", @@ -331,16 +326,16 @@ static const char *dac_deemph_text[] = { "48kHz", }; -static const struct soc_enum dac_deemph = - SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text); +static SOC_ENUM_SINGLE_DECL(dac_deemph, WM9081_DAC_DIGITAL_2, 1, + dac_deemph_text); static const char *speaker_mode_text[] = { "Class D", "Class AB", }; -static const struct soc_enum speaker_mode = - SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text); +static SOC_ENUM_SINGLE_DECL(speaker_mode, WM9081_ANALOGUE_SPEAKER_2, 6, + speaker_mode_text); static int speaker_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1265,15 +1260,6 @@ static struct snd_soc_dai_driver wm9081_dai = { static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = wm9081->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* Enable zero cross by default */ snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, @@ -1288,7 +1274,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm9081_eq_controls)); } - return ret; + return 0; } static int wm9081_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a07fe1618eec..87934171f063 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -522,16 +522,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { - struct wm9090_priv *wm9090 = dev_get_drvdata(codec->dev); - int ret; - - codec->control_data = wm9090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Configure some defaults; they will be written out when we * bring the bias up. */ diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 70ce6793c5bd..c0b7f45dfa37 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -67,12 +67,12 @@ static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC", "Line", "Stereo Mix", "Mono Mix", "Phone"}; -static const struct soc_enum wm9705_enum_mic = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic); -static const struct soc_enum wm9705_enum_rec_l = - SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel); -static const struct soc_enum wm9705_enum_rec_r = - SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel); +static SOC_ENUM_SINGLE_DECL(wm9705_enum_mic, + AC97_GENERAL_PURPOSE, 8, wm9705_mic); +static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_l, + AC97_REC_SEL, 8, wm9705_rec_sel); +static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_r, + AC97_REC_SEL, 0, wm9705_rec_sel); /* Headphone Mixer */ static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = { diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 444626fcab40..bb5f7b4e3ebb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,24 +684,38 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = wm_adsp_buf_alloc(region->data, - le32_to_cpu(region->len), - &buf_list); - if (!buf) { - adsp_err(dsp, "Out of memory\n"); - ret = -ENOMEM; - goto out_fw; - } + size_t to_write = PAGE_SIZE; + size_t remain = le32_to_cpu(region->len); + const u8 *data = region->data; + + while (remain > 0) { + if (remain < PAGE_SIZE) + to_write = remain; + + buf = wm_adsp_buf_alloc(data, + to_write, + &buf_list); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + ret = -ENOMEM; + goto out_fw; + } - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(region->len)); - if (ret != 0) { - adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", - file, regions, - le32_to_cpu(region->len), offset, - region_name, ret); - goto out_fw; + ret = regmap_raw_write_async(regmap, reg, + buf->buf, + to_write); + if (ret != 0) { + adsp_err(dsp, + "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", + file, regions, + to_write, offset, + region_name, ret); + goto out_fw; + } + + data += to_write; + reg += to_write / 2; + remain -= to_write; } } @@ -1679,6 +1693,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_dbg(dsp, "Shutdown complete\n"); break; default: diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index b371066dd5bc..b6209662ab13 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -50,16 +50,16 @@ static const char *speaker_ref_text[] = { "VMID", }; -static const struct soc_enum speaker_ref = - SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text); +static SOC_ENUM_SINGLE_DECL(speaker_ref, + WM8993_SPEAKER_MIXER, 8, speaker_ref_text); static const char *speaker_mode_text[] = { "Class D", "Class AB", }; -static const struct soc_enum speaker_mode = - SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text); +static SOC_ENUM_SINGLE_DECL(speaker_mode, + WM8993_SPKMIXR_ATTENUATION, 8, speaker_mode_text); static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { @@ -735,15 +735,15 @@ static const char *hp_mux_text[] = { "DAC", }; -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); +static SOC_ENUM_SINGLE_DECL(hpl_enum, + WM8993_OUTPUT_MIXER1, 8, hp_mux_text); const struct snd_kcontrol_new wm_hubs_hpl_mux = WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum); EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux); -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); +static SOC_ENUM_SINGLE_DECL(hpr_enum, + WM8993_OUTPUT_MIXER2, 8, hp_mux_text); const struct snd_kcontrol_new wm_hubs_hpr_mux = WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum); diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 70ff3772079f..cab98a580053 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -17,6 +17,7 @@ #include <linux/platform_data/edma.h> #include <linux/i2c.h> #include <linux/of_platform.h> +#include <linux/clk.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -30,9 +31,34 @@ #include "davinci-i2s.h" struct snd_soc_card_drvdata_davinci { + struct clk *mclk; unsigned sysclk; }; +static int evm_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *soc_card = rtd->codec->card; + struct snd_soc_card_drvdata_davinci *drvdata = + snd_soc_card_get_drvdata(soc_card); + + if (drvdata->mclk) + return clk_prepare_enable(drvdata->mclk); + + return 0; +} + +static void evm_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *soc_card = rtd->codec->card; + struct snd_soc_card_drvdata_davinci *drvdata = + snd_soc_card_get_drvdata(soc_card); + + if (drvdata->mclk) + clk_disable_unprepare(drvdata->mclk); +} + static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -59,6 +85,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, } static struct snd_soc_ops evm_ops = { + .startup = evm_startup, + .shutdown = evm_shutdown, .hw_params = evm_hw_params, }; @@ -95,35 +123,29 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Logic for a aic3x as connected on a davinci-evm */ static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_card *card = rtd->card; struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; struct device_node *np = codec->card->dev->of_node; int ret; /* Add davinci-evm specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, + snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets, ARRAY_SIZE(aic3x_dapm_widgets)); if (np) { - ret = snd_soc_of_parse_audio_routing(codec->card, - "ti,audio-routing"); + ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing"); if (ret) return ret; } else { /* Set up davinci-evm specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_add_routes(&card->dapm, audio_map, + ARRAY_SIZE(audio_map)); } /* not connected */ - snd_soc_dapm_disable_pin(dapm, "MONO_LOUT"); - snd_soc_dapm_disable_pin(dapm, "HPLCOM"); - snd_soc_dapm_disable_pin(dapm, "HPRCOM"); - - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Out"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line In"); + snd_soc_dapm_nc_pin(&codec->dapm, "MONO_LOUT"); + snd_soc_dapm_nc_pin(&codec->dapm, "HPLCOM"); + snd_soc_dapm_nc_pin(&codec->dapm, "HPRCOM"); return 0; } @@ -348,6 +370,7 @@ static int davinci_evm_probe(struct platform_device *pdev) of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev); struct snd_soc_dai_link *dai = (struct snd_soc_dai_link *) match->data; struct snd_soc_card_drvdata_davinci *drvdata = NULL; + struct clk *mclk; int ret = 0; evm_soc_card.dai_link = dai; @@ -367,13 +390,38 @@ static int davinci_evm_probe(struct platform_device *pdev) if (ret) return ret; + mclk = devm_clk_get(&pdev->dev, "mclk"); + if (PTR_ERR(mclk) == -EPROBE_DEFER) { + return -EPROBE_DEFER; + } else if (IS_ERR(mclk)) { + dev_dbg(&pdev->dev, "mclk not found.\n"); + mclk = NULL; + } + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); if (!drvdata) return -ENOMEM; + drvdata->mclk = mclk; + ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk); - if (ret < 0) - return -EINVAL; + + if (ret < 0) { + if (!drvdata->mclk) { + dev_err(&pdev->dev, + "No clock or clock rate defined.\n"); + return -EINVAL; + } + drvdata->sysclk = clk_get_rate(drvdata->mclk); + } else if (drvdata->mclk) { + unsigned int requestd_rate = drvdata->sysclk; + clk_set_rate(drvdata->mclk, drvdata->sysclk); + drvdata->sysclk = clk_get_rate(drvdata->mclk); + if (drvdata->sysclk != requestd_rate) + dev_warn(&pdev->dev, + "Could not get requested rate %u using %u.\n", + requestd_rate, drvdata->sysclk); + } snd_soc_card_set_drvdata(&evm_soc_card, drvdata); ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card); @@ -399,6 +447,7 @@ static struct platform_driver davinci_evm_driver = { .driver = { .name = "davinci_evm", .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, .of_match_table = of_match_ptr(davinci_evm_dt_ids), }, }; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index b7858bfa0295..a01ae97c90aa 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -37,6 +37,16 @@ #include "davinci-pcm.h" #include "davinci-mcasp.h" +struct davinci_mcasp_context { + u32 txfmtctl; + u32 rxfmtctl; + u32 txfmt; + u32 rxfmt; + u32 aclkxctl; + u32 aclkrctl; + u32 pdir; +}; + struct davinci_mcasp { struct davinci_pcm_dma_params dma_params[2]; struct snd_dmaengine_dai_dma_data dma_data[2]; @@ -53,6 +63,9 @@ struct davinci_mcasp { u16 bclk_lrclk_ratio; int streams; + int sysclk_freq; + bool bclk_master; + /* McASP FIFO related */ u8 txnumevt; u8 rxnumevt; @@ -60,15 +73,7 @@ struct davinci_mcasp { bool dat_port; #ifdef CONFIG_PM_SLEEP - struct { - u32 txfmtctl; - u32 rxfmtctl; - u32 txfmt; - u32 rxfmt; - u32 aclkxctl; - u32 aclkrctl; - u32 pdir; - } context; + struct davinci_mcasp_context context; #endif }; @@ -263,7 +268,9 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + pm_runtime_get_sync(mcasp->dev); switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: case SND_SOC_DAIFMT_AC97: @@ -292,6 +299,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 1; break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ @@ -303,6 +311,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | ACLKR); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AFSX | AFSR); + mcasp->bclk_master = 0; break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ @@ -314,10 +323,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + mcasp->bclk_master = 0; break; default: - return -EINVAL; + ret = -EINVAL; + goto out; } switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -354,10 +365,12 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, break; default: - return -EINVAL; + ret = -EINVAL; + break; } - - return 0; +out: + pm_runtime_put_sync(mcasp->dev); + return ret; } static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) @@ -405,6 +418,8 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } + mcasp->sysclk_freq = freq; + return 0; } @@ -448,7 +463,7 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, return 0; } -static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, +static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int channels) { int i; @@ -524,12 +539,18 @@ static int davinci_hw_common_param(struct davinci_mcasp *mcasp, int stream, return 0; } -static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) +static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) { int i, active_slots; u32 mask = 0; u32 busel = 0; + if ((mcasp->tdm_slots < 2) || (mcasp->tdm_slots > 32)) { + dev_err(mcasp->dev, "tdm slot %d not supported\n", + mcasp->tdm_slots); + return -EINVAL; + } + active_slots = (mcasp->tdm_slots > 31) ? 32 : mcasp->tdm_slots; for (i = 0; i < active_slots; i++) mask |= (1 << i); @@ -539,35 +560,21 @@ static void davinci_hw_param(struct davinci_mcasp *mcasp, int stream) if (!mcasp->dat_port) busel = TXSEL; - if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, - FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); - else - printk(KERN_ERR "playback tdm slot %d not supported\n", - mcasp->tdm_slots); - } else { - /* bit stream is MSB first with no delay */ - /* DSP_B mode */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); - - if ((mcasp->tdm_slots >= 2) && (mcasp->tdm_slots <= 32)) - mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, - FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); - else - printk(KERN_ERR "capture tdm slot %d not supported\n", - mcasp->tdm_slots); - } + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, + FSXMOD(mcasp->tdm_slots), FSXMOD(0x1FF)); + + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask); + mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, + FSRMOD(mcasp->tdm_slots), FSRMOD(0x1FF)); + + return 0; } /* S/PDIF */ -static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) { /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ @@ -589,6 +596,8 @@ static void davinci_hw_dit_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + + return 0; } static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, @@ -604,24 +613,31 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, u8 fifo_level; u8 slots = mcasp->tdm_slots; u8 active_serializers; - int channels; - struct snd_interval *pcm_channels = hw_param_interval(params, - SNDRV_PCM_HW_PARAM_CHANNELS); - channels = pcm_channels->min; + int channels = params_channels(params); + int ret; - active_serializers = (channels + slots - 1) / slots; + /* If mcasp is BCLK master we need to set BCLK divider */ + if (mcasp->bclk_master) { + unsigned int bclk_freq = snd_soc_params_to_bclk(params); + if (mcasp->sysclk_freq % bclk_freq != 0) { + dev_err(mcasp->dev, "Can't produce requred BCLK\n"); + return -EINVAL; + } + davinci_mcasp_set_clkdiv( + cpu_dai, 1, mcasp->sysclk_freq / bclk_freq); + } - if (davinci_hw_common_param(mcasp, substream->stream, channels) == -EINVAL) - return -EINVAL; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - fifo_level = mcasp->txnumevt * active_serializers; - else - fifo_level = mcasp->rxnumevt * active_serializers; + ret = mcasp_common_hw_param(mcasp, substream->stream, channels); + if (ret) + return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - davinci_hw_dit_param(mcasp); + ret = mcasp_dit_hw_param(mcasp); else - davinci_hw_param(mcasp, substream->stream); + ret = mcasp_i2s_hw_param(mcasp, substream->stream); + + if (ret) + return ret; switch (params_format(params)) { case SNDRV_PCM_FORMAT_U8: @@ -655,6 +671,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + /* Calculate FIFO level */ + active_serializers = (channels + slots - 1) / slots; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + fifo_level = mcasp->txnumevt * active_serializers; + else + fifo_level = mcasp->rxnumevt * active_serializers; + if (mcasp->version == MCASP_VERSION_2 && !fifo_level) dma_params->acnt = 4; else @@ -678,19 +701,9 @@ static int davinci_mcasp_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - ret = pm_runtime_get_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_get_sync() failed\n"); davinci_mcasp_start(mcasp, substream->stream); break; - case SNDRV_PCM_TRIGGER_SUSPEND: - davinci_mcasp_stop(mcasp, substream->stream); - ret = pm_runtime_put_sync(mcasp->dev); - if (IS_ERR_VALUE(ret)) - dev_err(mcasp->dev, "pm_runtime_put_sync() failed\n"); - break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: davinci_mcasp_stop(mcasp, substream->stream); @@ -726,6 +739,43 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_sysclk = davinci_mcasp_set_sysclk, }; +#ifdef CONFIG_PM_SLEEP +static int davinci_mcasp_suspend(struct snd_soc_dai *dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp_context *context = &mcasp->context; + + context->txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); + context->rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); + context->txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); + context->rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); + context->aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); + context->aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); + context->pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); + + return 0; +} + +static int davinci_mcasp_resume(struct snd_soc_dai *dai) +{ + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + struct davinci_mcasp_context *context = &mcasp->context; + + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, context->txfmtctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, context->rxfmtctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, context->txfmt); + mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, context->rxfmt); + mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, context->aclkxctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, context->aclkrctl); + mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, context->pdir); + + return 0; +} +#else +#define davinci_mcasp_suspend NULL +#define davinci_mcasp_resume NULL +#endif + #define DAVINCI_MCASP_RATES SNDRV_PCM_RATE_8000_192000 #define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \ @@ -742,6 +792,8 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", + .suspend = davinci_mcasp_suspend, + .resume = davinci_mcasp_resume, .playback = { .channels_min = 2, .channels_max = 32 * 16, @@ -775,28 +827,28 @@ static const struct snd_soc_component_driver davinci_mcasp_component = { }; /* Some HW specific values and defaults. The rest is filled in from DT. */ -static struct snd_platform_data dm646x_mcasp_pdata = { +static struct davinci_mcasp_pdata dm646x_mcasp_pdata = { .tx_dma_offset = 0x400, .rx_dma_offset = 0x400, .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_1, }; -static struct snd_platform_data da830_mcasp_pdata = { +static struct davinci_mcasp_pdata da830_mcasp_pdata = { .tx_dma_offset = 0x2000, .rx_dma_offset = 0x2000, .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_2, }; -static struct snd_platform_data am33xx_mcasp_pdata = { +static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { .tx_dma_offset = 0, .rx_dma_offset = 0, .asp_chan_q = EVENTQ_0, .version = MCASP_VERSION_3, }; -static struct snd_platform_data dra7_mcasp_pdata = { +static struct davinci_mcasp_pdata dra7_mcasp_pdata = { .tx_dma_offset = 0x200, .rx_dma_offset = 0x284, .asp_chan_q = EVENTQ_0, @@ -864,11 +916,11 @@ err1: return ret; } -static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( +static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct snd_platform_data *pdata = NULL; + struct davinci_mcasp_pdata *pdata = NULL; const struct of_device_id *match = of_match_device(mcasp_dt_ids, &pdev->dev); struct of_phandle_args dma_spec; @@ -881,7 +933,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( pdata = pdev->dev.platform_data; return pdata; } else if (match) { - pdata = (struct snd_platform_data *) match->data; + pdata = (struct davinci_mcasp_pdata*) match->data; } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; @@ -973,9 +1025,10 @@ nodata: static int davinci_mcasp_probe(struct platform_device *pdev) { - struct davinci_pcm_dma_params *dma_data; + struct davinci_pcm_dma_params *dma_params; + struct snd_dmaengine_dai_dma_data *dma_data; struct resource *mem, *ioarea, *res, *dat; - struct snd_platform_data *pdata; + struct davinci_mcasp_pdata *pdata; struct davinci_mcasp *mcasp; int ret; @@ -1042,41 +1095,49 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) mcasp->dat_port = true; - dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; - dma_data->asp_chan_q = pdata->asp_chan_q; - dma_data->ram_chan_q = pdata->ram_chan_q; - dma_data->sram_pool = pdata->sram_pool; - dma_data->sram_size = pdata->sram_size_playback; + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_PLAYBACK]; + dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_playback; if (dat) - dma_data->dma_addr = dat->start; + dma_params->dma_addr = dat->start; else - dma_data->dma_addr = mem->start + pdata->tx_dma_offset; + dma_params->dma_addr = mem->start + pdata->tx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr = dma_data->dma_addr; + dma_data->addr = dma_params->dma_addr; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (res) - dma_data->channel = res->start; + dma_params->channel = res->start; else - dma_data->channel = pdata->tx_dma_channel; + dma_params->channel = pdata->tx_dma_channel; - dma_data = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; - dma_data->asp_chan_q = pdata->asp_chan_q; - dma_data->ram_chan_q = pdata->ram_chan_q; - dma_data->sram_pool = pdata->sram_pool; - dma_data->sram_size = pdata->sram_size_capture; + /* dmaengine filter data for DT and non-DT boot */ + if (pdev->dev.of_node) + dma_data->filter_data = "tx"; + else + dma_data->filter_data = &dma_params->channel; + + dma_params = &mcasp->dma_params[SNDRV_PCM_STREAM_CAPTURE]; + dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + dma_params->asp_chan_q = pdata->asp_chan_q; + dma_params->ram_chan_q = pdata->ram_chan_q; + dma_params->sram_pool = pdata->sram_pool; + dma_params->sram_size = pdata->sram_size_capture; if (dat) - dma_data->dma_addr = dat->start; + dma_params->dma_addr = dat->start; else - dma_data->dma_addr = mem->start + pdata->rx_dma_offset; + dma_params->dma_addr = mem->start + pdata->rx_dma_offset; /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr = dma_data->dma_addr; + dma_data->addr = dma_params->dma_addr; if (mcasp->version < MCASP_VERSION_3) { mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE; - /* dma_data->dma_addr is pointing to the data port address */ + /* dma_params->dma_addr is pointing to the data port address */ mcasp->dat_port = true; } else { mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; @@ -1084,13 +1145,15 @@ static int davinci_mcasp_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_DMA, 1); if (res) - dma_data->channel = res->start; + dma_params->channel = res->start; else - dma_data->channel = pdata->rx_dma_channel; + dma_params->channel = pdata->rx_dma_channel; - /* Unconditional dmaengine stuff */ - mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data = "tx"; - mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data = "rx"; + /* dmaengine filter data for DT and non-DT boot */ + if (pdev->dev.of_node) + dma_data->filter_data = "rx"; + else + dma_data->filter_data = &dma_params->channel; dev_set_drvdata(&pdev->dev, mcasp); @@ -1134,49 +1197,12 @@ static int davinci_mcasp_remove(struct platform_device *pdev) return 0; } -#ifdef CONFIG_PM_SLEEP -static int davinci_mcasp_suspend(struct device *dev) -{ - struct davinci_mcasp *mcasp = dev_get_drvdata(dev); - - mcasp->context.txfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG); - mcasp->context.rxfmtctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG); - mcasp->context.txfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXFMT_REG); - mcasp->context.rxfmt = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMT_REG); - mcasp->context.aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG); - mcasp->context.aclkrctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG); - mcasp->context.pdir = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG); - - return 0; -} - -static int davinci_mcasp_resume(struct device *dev) -{ - struct davinci_mcasp *mcasp = dev_get_drvdata(dev); - - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, mcasp->context.txfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG, mcasp->context.rxfmtctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMT_REG, mcasp->context.txfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_RXFMT_REG, mcasp->context.rxfmt); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, mcasp->context.aclkxctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, mcasp->context.aclkrctl); - mcasp_set_reg(mcasp, DAVINCI_MCASP_PDIR_REG, mcasp->context.pdir); - - return 0; -} -#endif - -SIMPLE_DEV_PM_OPS(davinci_mcasp_pm_ops, - davinci_mcasp_suspend, - davinci_mcasp_resume); - static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, - .pm = &davinci_mcasp_pm_ops, .of_match_table = mcasp_dt_ids, }, }; diff --git a/sound/soc/davinci/edma-pcm.c b/sound/soc/davinci/edma-pcm.c new file mode 100644 index 000000000000..d38afb1c61ae --- /dev/null +++ b/sound/soc/davinci/edma-pcm.c @@ -0,0 +1,57 @@ +/* + * edma-pcm.c - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> + * + * Based on: sound/soc/tegra/tegra_pcm.c + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> +#include <linux/edma.h> + +static const struct snd_pcm_hardware edma_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_INTERLEAVED, + .buffer_bytes_max = 128 * 1024, + .period_bytes_min = 32, + .period_bytes_max = 64 * 1024, + .periods_min = 2, + .periods_max = 19, /* Limit by edma dmaengine driver */ +}; + +static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = { + .pcm_hardware = &edma_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = edma_filter_fn, + .prealloc_buffer_size = 128 * 1024, +}; + +int edma_pcm_platform_register(struct device *dev) +{ + return devm_snd_dmaengine_pcm_register(dev, &edma_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_COMPAT); +} +EXPORT_SYMBOL_GPL(edma_pcm_platform_register); + +MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>"); +MODULE_DESCRIPTION("eDMA PCM ASoC platform driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/davinci/edma-pcm.h b/sound/soc/davinci/edma-pcm.h new file mode 100644 index 000000000000..894c378c0f74 --- /dev/null +++ b/sound/soc/davinci/edma-pcm.h @@ -0,0 +1,25 @@ +/* + * edma-pcm.h - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Peter Ujfalusi <peter.ujfalusi@ti.com> + * + * Based on: sound/soc/tegra/tegra_pcm.h + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef __EDMA_PCM_H__ +#define __EDMA_PCM_H__ + +int edma_pcm_platform_register(struct device *dev); + +#endif /* __EDMA_PCM_H__ */ diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 07f8f141727d..597962ec28fa 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,5 +1,6 @@ config SND_SOC_FSL_SAI tristate + select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SOC_FSL_SSI @@ -7,9 +8,11 @@ config SND_SOC_FSL_SSI config SND_SOC_FSL_SPDIF tristate + select REGMAP_MMIO config SND_SOC_FSL_ESAI tristate + select REGMAP_MMIO config SND_SOC_FSL_UTILS tristate @@ -168,12 +171,14 @@ config SND_SOC_EUKREA_TLV320 depends on MACH_EUKREA_MBIMX27_BASEBOARD \ || MACH_EUKREA_MBIMXSD25_BASEBOARD \ || MACH_EUKREA_MBIMXSD35_BASEBOARD \ - || MACH_EUKREA_MBIMXSD51_BASEBOARD + || MACH_EUKREA_MBIMXSD51_BASEBOARD \ + || (OF && ARM) depends on I2C - select SND_SOC_TLV320AIC23 - select SND_SOC_IMX_PCM_FIQ + select SND_SOC_TLV320AIC23_I2C select SND_SOC_IMX_AUDMUX select SND_SOC_IMX_SSI + select SND_SOC_FSL_SSI + select SND_SOC_IMX_PCM_DMA help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface @@ -204,7 +209,6 @@ config SND_SOC_IMX_SPDIF tristate "SoC Audio support for i.MX boards with S/PDIF" select SND_SOC_IMX_PCM_DMA select SND_SOC_FSL_SPDIF - select REGMAP_MMIO help SoC Audio support for i.MX boards with S/PDIF Say Y if you want to add support for SoC audio on an i.MX board with diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 5983740be123..eb093d5b85c4 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -15,8 +15,11 @@ * */ +#include <linux/errno.h> #include <linux/module.h> #include <linux/moduleparam.h> +#include <linux/of.h> +#include <linux/of_platform.h> #include <linux/device.h> #include <linux/i2c.h> #include <sound/core.h> @@ -26,6 +29,7 @@ #include "../codecs/tlv320aic23.h" #include "imx-ssi.h" +#include "fsl_ssi.h" #include "imx-audmux.h" #define CODEC_CLOCK 12000000 @@ -41,7 +45,8 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); - if (ret) { + /* fsl_ssi lacks the set_fmt ops. */ + if (ret && ret != -ENOTSUPP) { dev_err(cpu_dai->dev, "Failed to set the cpu dai format.\n"); return ret; @@ -63,11 +68,13 @@ static int eukrea_tlv320_hw_params(struct snd_pcm_substream *substream, "Failed to set the codec sysclk.\n"); return ret; } + snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0); ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, SND_SOC_CLOCK_IN); - if (ret) { + /* fsl_ssi lacks the set_sysclk ops */ + if (ret && ret != -EINVAL) { dev_err(cpu_dai->dev, "Can't set the IMX_SSP_SYS_CLK CPU system clock.\n"); return ret; @@ -84,14 +91,10 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "imx-ssi.0", - .codec_name = "tlv320aic23-codec.0-001a", - .cpu_dai_name = "imx-ssi.0", .ops = &eukrea_tlv320_snd_ops, }; static struct snd_soc_card eukrea_tlv320 = { - .name = "cpuimx-audio", .owner = THIS_MODULE, .dai_link = &eukrea_tlv320_dai, .num_links = 1, @@ -101,8 +104,65 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) { int ret; int int_port = 0, ext_port; + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; - if (machine_is_eukrea_cpuimx27()) { + eukrea_tlv320.dev = &pdev->dev; + if (np) { + ret = snd_soc_of_parse_card_name(&eukrea_tlv320, + "eukrea,model"); + if (ret) { + dev_err(&pdev->dev, + "eukrea,model node missing or invalid.\n"); + goto err; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, + "ssi-controller", 0); + if (!ssi_np) { + dev_err(&pdev->dev, + "ssi-controller missing or invalid.\n"); + ret = -ENODEV; + goto err; + } + + codec_np = of_parse_phandle(ssi_np, "codec-handle", 0); + if (codec_np) + eukrea_tlv320_dai.codec_of_node = codec_np; + else + dev_err(&pdev->dev, "codec-handle node missing or invalid.\n"); + + ret = of_property_read_u32(np, "fsl,mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-int-port node missing or invalid.\n"); + return ret; + } + ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, + "fsl,mux-ext-port node missing or invalid.\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + + eukrea_tlv320_dai.cpu_of_node = ssi_np; + eukrea_tlv320_dai.platform_of_node = ssi_np; + } else { + eukrea_tlv320_dai.cpu_dai_name = "imx-ssi.0"; + eukrea_tlv320_dai.platform_name = "imx-ssi.0"; + eukrea_tlv320_dai.codec_name = "tlv320aic23-codec.0-001a"; + eukrea_tlv320.name = "cpuimx-audio"; + } + + if (machine_is_eukrea_cpuimx27() || + of_find_compatible_node(NULL, NULL, "fsl,imx21-audmux")) { imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFSDIR | @@ -119,8 +179,12 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) ); } else if (machine_is_eukrea_cpuimx25sd() || machine_is_eukrea_cpuimx35sd() || - machine_is_eukrea_cpuimx51sd()) { - ext_port = machine_is_eukrea_cpuimx25sd() ? 4 : 3; + machine_is_eukrea_cpuimx51sd() || + of_find_compatible_node(NULL, NULL, "fsl,imx31-audmux")) { + if (!np) + ext_port = machine_is_eukrea_cpuimx25sd() ? + 4 : 3; + imx_audmux_v2_configure_port(int_port, IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | @@ -134,14 +198,27 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) ); } else { - /* return happy. We might run on a totally different machine */ - return 0; + if (np) { + /* The eukrea,asoc-tlv320 driver was explicitely + * requested (through the device tree). + */ + dev_err(&pdev->dev, + "Missing or invalid audmux DT node.\n"); + return -ENODEV; + } else { + /* Return happy. + * We might run on a totally different machine. + */ + return 0; + } } - eukrea_tlv320.dev = &pdev->dev; ret = snd_soc_register_card(&eukrea_tlv320); +err: if (ret) dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (np) + of_node_put(ssi_np); return ret; } @@ -153,10 +230,17 @@ static int eukrea_tlv320_remove(struct platform_device *pdev) return 0; } +static const struct of_device_id imx_tlv320_dt_ids[] = { + { .compatible = "eukrea,asoc-tlv320"}, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_tlv320_dt_ids); + static struct platform_driver eukrea_tlv320_driver = { .driver = { .name = "eukrea_tlv320", .owner = THIS_MODULE, + .of_match_table = imx_tlv320_dt_ids, }, .probe = eukrea_tlv320_probe, .remove = eukrea_tlv320_remove, diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0c72ed261e7..0ba37005ab04 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -326,7 +326,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(tx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); @@ -334,7 +334,7 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMA_xS_MASK, ESAI_xSMB_xS(rx_mask)); + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); esai_priv->slot_width = slot_width; @@ -431,17 +431,26 @@ static int fsl_esai_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) static int fsl_esai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { + int ret; struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); /* * Some platforms might use the same bit to gate all three or two of * clocks, so keep all clocks open/close at the same time for safety */ - clk_prepare_enable(esai_priv->coreclk); - if (!IS_ERR(esai_priv->extalclk)) - clk_prepare_enable(esai_priv->extalclk); - if (!IS_ERR(esai_priv->fsysclk)) - clk_prepare_enable(esai_priv->fsysclk); + ret = clk_prepare_enable(esai_priv->coreclk); + if (ret) + return ret; + if (!IS_ERR(esai_priv->extalclk)) { + ret = clk_prepare_enable(esai_priv->extalclk); + if (ret) + goto err_extalck; + } + if (!IS_ERR(esai_priv->fsysclk)) { + ret = clk_prepare_enable(esai_priv->fsysclk); + if (ret) + goto err_fsysclk; + } if (!dai->active) { /* Reset Port C */ @@ -463,6 +472,14 @@ static int fsl_esai_startup(struct snd_pcm_substream *substream, } return 0; + +err_fsysclk: + if (!IS_ERR(esai_priv->extalclk)) + clk_disable_unprepare(esai_priv->extalclk); +err_extalck: + clk_disable_unprepare(esai_priv->coreclk); + + return ret; } static int fsl_esai_hw_params(struct snd_pcm_substream *substream, @@ -661,7 +678,7 @@ static bool fsl_esai_writeable_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config fsl_esai_regmap_config = { +static struct regmap_config fsl_esai_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -687,6 +704,9 @@ static int fsl_esai_probe(struct platform_device *pdev) esai_priv->pdev = pdev; strcpy(esai_priv->name, np->name); + if (of_property_read_bool(np, "big-endian")) + fsl_esai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_esai.h b/sound/soc/fsl/fsl_esai.h index 9c9f957fcae1..75e14033e8d8 100644 --- a/sound/soc/fsl/fsl_esai.h +++ b/sound/soc/fsl/fsl_esai.h @@ -322,7 +322,7 @@ #define ESAI_xSMB_xS_SHIFT 0 #define ESAI_xSMB_xS_WIDTH 16 #define ESAI_xSMB_xS_MASK (((1 << ESAI_xSMB_xS_WIDTH) - 1) << ESAI_xSMB_xS_SHIFT) -#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMA_xS_MASK) +#define ESAI_xSMB_xS(v) (((v) >> ESAI_xSMA_xS_WIDTH) & ESAI_xSMB_xS_MASK) /* Port C Direction Register -- REG_ESAI_PRRC 0xF8 */ #define ESAI_PRRC_PDC_SHIFT 0 diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index cdd3fa830704..c4a423111673 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -15,6 +15,7 @@ #include <linux/dmaengine.h> #include <linux/module.h> #include <linux/of_address.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/dmaengine_pcm.h> @@ -22,34 +23,6 @@ #include "fsl_sai.h" -static inline u32 sai_readl(struct fsl_sai *sai, - const void __iomem *addr) -{ - u32 val; - - val = __raw_readl(addr); - - if (likely(sai->big_endian_regs)) - val = be32_to_cpu(val); - else - val = le32_to_cpu(val); - rmb(); - - return val; -} - -static inline void sai_writel(struct fsl_sai *sai, - u32 val, void __iomem *addr) -{ - wmb(); - if (likely(sai->big_endian_regs)) - val = cpu_to_be32(val); - else - val = cpu_to_le32(val); - - __raw_writel(val, addr); -} - static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { @@ -61,7 +34,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, else reg_cr2 = FSL_SAI_RCR2; - val_cr2 = sai_readl(sai, sai->base + reg_cr2); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; switch (clk_id) { @@ -81,7 +55,7 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); + regmap_write(sai->regmap, reg_cr2, val_cr2); return 0; } @@ -89,32 +63,22 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, static int fsl_sai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; if (dir == SND_SOC_CLOCK_IN) return 0; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx sysclk: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_sysclk_tr(cpu_dai, clk_id, freq, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx sysclk: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -133,43 +97,84 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, reg_cr4 = FSL_SAI_RCR4; } - val_cr2 = sai_readl(sai, sai->base + reg_cr2); - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr2, &val_cr2); + regmap_read(sai->regmap, reg_cr4, &val_cr4); if (sai->big_endian_data) val_cr4 &= ~FSL_SAI_CR4_MF; else val_cr4 |= FSL_SAI_CR4_MF; + /* DAI mode */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: + /* + * Frame low, 1clk before data, one word length for frame sync, + * frame sync starts one serial clock cycle earlier, + * that is, together with the last bit of the previous + * data word. + */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* + * Frame high, one word length for frame sync, + * frame sync asserts with the first bit of the frame. + */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); + break; + case SND_SOC_DAIFMT_DSP_A: + /* + * Frame high, 1clk before data, one bit for frame sync, + * frame sync starts one serial clock cycle earlier, + * that is, together with the last bit of the previous + * data word. + */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; + sai->is_dsp_mode = true; + break; + case SND_SOC_DAIFMT_DSP_B: + /* + * Frame high, one bit for frame sync, + * frame sync asserts with the first bit of the frame. + */ + val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); + sai->is_dsp_mode = true; break; + case SND_SOC_DAIFMT_RIGHT_J: + /* To be done */ default: return -EINVAL; } + /* DAI clock inversion */ switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_IB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert both clocks */ + val_cr2 ^= FSL_SAI_CR2_BCP; + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_IB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 &= ~FSL_SAI_CR2_BCP; + /* Invert bit clock */ + val_cr2 ^= FSL_SAI_CR2_BCP; break; case SND_SOC_DAIFMT_NB_IF: - val_cr4 |= FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Invert frame clock */ + val_cr4 ^= FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_NB_NF: - val_cr4 &= ~FSL_SAI_CR4_FSP; - val_cr2 |= FSL_SAI_CR2_BCP; + /* Nothing to do for both normal cases */ break; default: return -EINVAL; } + /* DAI clock master masks */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; @@ -179,39 +184,37 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; + case SND_SOC_DAIFMT_CBS_CFM: + val_cr2 |= FSL_SAI_CR2_BCD_MSTR; + val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; + val_cr4 |= FSL_SAI_CR4_FSD_MSTR; + break; default: return -EINVAL; } - sai_writel(sai, val_cr2, sai->base + reg_cr2); - sai_writel(sai, val_cr4, sai->base + reg_cr4); + regmap_write(sai->regmap, reg_cr2, val_cr2); + regmap_write(sai->regmap, reg_cr4, val_cr4); return 0; } static int fsl_sai_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { - struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); int ret; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_TRANSMITTER); if (ret) { dev_err(cpu_dai->dev, "Cannot set tx format: %d\n", ret); - goto err_clk; + return ret; } ret = fsl_sai_set_dai_fmt_tr(cpu_dai, fmt, FSL_FMT_RECEIVER); - if (ret) { + if (ret) dev_err(cpu_dai->dev, "Cannot set rx format: %d\n", ret); - goto err_clk; - } - -err_clk: - clk_disable_unprepare(sai->clk); return ret; } @@ -235,16 +238,19 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, reg_mr = FSL_SAI_RMR; } - val_cr4 = sai_readl(sai, sai->base + reg_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr4); + regmap_read(sai->regmap, reg_cr4, &val_cr5); + val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; - val_cr5 = sai_readl(sai, sai->base + reg_cr5); val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; - val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + if (!sai->is_dsp_mode) + val_cr4 |= FSL_SAI_CR4_SYWD(word_width); + val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); @@ -257,9 +263,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(channels); val_mr = ~0UL - ((1 << channels) - 1); - sai_writel(sai, val_cr4, sai->base + reg_cr4); - sai_writel(sai, val_cr5, sai->base + reg_cr5); - sai_writel(sai, val_mr, sai->base + reg_mr); + regmap_write(sai->regmap, reg_cr4, val_cr4); + regmap_write(sai->regmap, reg_cr5, val_cr5); + regmap_write(sai->regmap, reg_mr, val_mr); return 0; } @@ -268,44 +274,42 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 tcsr, rcsr, val_cr2, val_cr3, reg_cr3; - - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_TCR2); - val_cr2 &= ~FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_TCR2); + u32 tcsr, rcsr; - val_cr2 = sai_readl(sai, sai->base + FSL_SAI_RCR2); - val_cr2 |= FSL_SAI_CR2_SYNC; - sai_writel(sai, val_cr2, sai->base + FSL_SAI_RCR2); + /* + * The transmitter bit clock and frame sync are to be + * used by both the transmitter and receiver. + */ + regmap_update_bits(sai->regmap, FSL_SAI_TCR2, FSL_SAI_CR2_SYNC, + ~FSL_SAI_CR2_SYNC); + regmap_update_bits(sai->regmap, FSL_SAI_RCR2, FSL_SAI_CR2_SYNC, + FSL_SAI_CR2_SYNC); - tcsr = sai_readl(sai, sai->base + FSL_SAI_TCSR); - rcsr = sai_readl(sai, sai->base + FSL_SAI_RCSR); + regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); + regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { tcsr |= FSL_SAI_CSR_FRDE; rcsr &= ~FSL_SAI_CSR_FRDE; - reg_cr3 = FSL_SAI_TCR3; } else { rcsr |= FSL_SAI_CSR_FRDE; tcsr &= ~FSL_SAI_CSR_FRDE; - reg_cr3 = FSL_SAI_RCR3; } - val_cr3 = sai_readl(sai, sai->base + reg_cr3); - + /* + * It is recommended that the transmitter is the last enabled + * and the first disabled. + */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: tcsr |= FSL_SAI_CSR_TERE; rcsr |= FSL_SAI_CSR_TERE; - val_cr3 |= FSL_SAI_CR3_TRCE; - sai_writel(sai, val_cr3, sai->base + reg_cr3); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); break; - case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -314,11 +318,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, rcsr &= ~FSL_SAI_CSR_TERE; } - val_cr3 &= ~FSL_SAI_CR3_TRCE; - - sai_writel(sai, tcsr, sai->base + FSL_SAI_TCSR); - sai_writel(sai, rcsr, sai->base + FSL_SAI_RCSR); - sai_writel(sai, val_cr3, sai->base + reg_cr3); + regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -331,16 +332,32 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = FSL_SAI_TCR3; + else + reg = FSL_SAI_RCR3; + + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + FSL_SAI_CR3_TRCE); - return clk_prepare_enable(sai->clk); + return 0; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + u32 reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + reg = FSL_SAI_TCR3; + else + reg = FSL_SAI_RCR3; - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + ~FSL_SAI_CR3_TRCE); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -355,18 +372,13 @@ static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - int ret; - ret = clk_prepare_enable(sai->clk); - if (ret) - return ret; - - sai_writel(sai, 0x0, sai->base + FSL_SAI_RCSR); - sai_writel(sai, 0x0, sai->base + FSL_SAI_TCSR); - sai_writel(sai, FSL_SAI_MAXBURST_TX * 2, sai->base + FSL_SAI_TCR1); - sai_writel(sai, FSL_SAI_MAXBURST_RX - 1, sai->base + FSL_SAI_RCR1); - - clk_disable_unprepare(sai->clk); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_TX * 2); + regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, + FSL_SAI_MAXBURST_RX - 1); snd_soc_dai_init_dma_data(cpu_dai, &sai->dma_params_tx, &sai->dma_params_rx); @@ -397,26 +409,109 @@ static const struct snd_soc_component_driver fsl_component = { .name = "fsl-sai", }; +static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TFR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RDR: + case FSL_SAI_RFR: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TFR: + case FSL_SAI_RFR: + case FSL_SAI_TDR: + case FSL_SAI_RDR: + return true; + default: + return false; + } + +} + +static bool fsl_sai_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_TCR1: + case FSL_SAI_TCR2: + case FSL_SAI_TCR3: + case FSL_SAI_TCR4: + case FSL_SAI_TCR5: + case FSL_SAI_TDR: + case FSL_SAI_TMR: + case FSL_SAI_RCSR: + case FSL_SAI_RCR1: + case FSL_SAI_RCR2: + case FSL_SAI_RCR3: + case FSL_SAI_RCR4: + case FSL_SAI_RCR5: + case FSL_SAI_RMR: + return true; + default: + return false; + } +} + +static struct regmap_config fsl_sai_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + + .max_register = FSL_SAI_RMR, + .readable_reg = fsl_sai_readable_reg, + .volatile_reg = fsl_sai_volatile_reg, + .writeable_reg = fsl_sai_writeable_reg, +}; + static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct fsl_sai *sai; struct resource *res; + void __iomem *base; int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) return -ENOMEM; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); + if (sai->big_endian_regs) + fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + + sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - sai->base = devm_ioremap_resource(&pdev->dev, res); - if (IS_ERR(sai->base)) - return PTR_ERR(sai->base); - - sai->clk = devm_clk_get(&pdev->dev, "sai"); - if (IS_ERR(sai->clk)) { - dev_err(&pdev->dev, "Cannot get SAI's clock\n"); - return PTR_ERR(sai->clk); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return PTR_ERR(base); + + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); + if (IS_ERR(sai->regmap)) { + dev_err(&pdev->dev, "regmap init failed\n"); + return PTR_ERR(sai->regmap); } sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; @@ -424,9 +519,6 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; sai->dma_params_tx.maxburst = FSL_SAI_MAXBURST_TX; - sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); - platform_set_drvdata(pdev, sai); ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 41bb62e69361..e432260be598 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -15,31 +15,36 @@ SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) +/* SAI Register Map Register */ +#define FSL_SAI_TCSR 0x00 /* SAI Transmit Control */ +#define FSL_SAI_TCR1 0x04 /* SAI Transmit Configuration 1 */ +#define FSL_SAI_TCR2 0x08 /* SAI Transmit Configuration 2 */ +#define FSL_SAI_TCR3 0x0c /* SAI Transmit Configuration 3 */ +#define FSL_SAI_TCR4 0x10 /* SAI Transmit Configuration 4 */ +#define FSL_SAI_TCR5 0x14 /* SAI Transmit Configuration 5 */ +#define FSL_SAI_TDR 0x20 /* SAI Transmit Data */ +#define FSL_SAI_TFR 0x40 /* SAI Transmit FIFO */ +#define FSL_SAI_TMR 0x60 /* SAI Transmit Mask */ +#define FSL_SAI_RCSR 0x80 /* SAI Receive Control */ +#define FSL_SAI_RCR1 0x84 /* SAI Receive Configuration 1 */ +#define FSL_SAI_RCR2 0x88 /* SAI Receive Configuration 2 */ +#define FSL_SAI_RCR3 0x8c /* SAI Receive Configuration 3 */ +#define FSL_SAI_RCR4 0x90 /* SAI Receive Configuration 4 */ +#define FSL_SAI_RCR5 0x94 /* SAI Receive Configuration 5 */ +#define FSL_SAI_RDR 0xa0 /* SAI Receive Data */ +#define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ +#define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ + /* SAI Transmit/Recieve Control Register */ -#define FSL_SAI_TCSR 0x00 -#define FSL_SAI_RCSR 0x80 #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRIE BIT(8) #define FSL_SAI_CSR_FRDE BIT(0) -/* SAI Transmit Data/FIFO/MASK Register */ -#define FSL_SAI_TDR 0x20 -#define FSL_SAI_TFR 0x40 -#define FSL_SAI_TMR 0x60 - -/* SAI Recieve Data/FIFO/MASK Register */ -#define FSL_SAI_RDR 0xa0 -#define FSL_SAI_RFR 0xc0 -#define FSL_SAI_RMR 0xe0 - /* SAI Transmit and Recieve Configuration 1 Register */ -#define FSL_SAI_TCR1 0x04 -#define FSL_SAI_RCR1 0x84 +#define FSL_SAI_CR1_RFW_MASK 0x1f /* SAI Transmit and Recieve Configuration 2 Register */ -#define FSL_SAI_TCR2 0x08 -#define FSL_SAI_RCR2 0x88 #define FSL_SAI_CR2_SYNC BIT(30) #define FSL_SAI_CR2_MSEL_MASK (0xff << 26) #define FSL_SAI_CR2_MSEL_BUS 0 @@ -50,15 +55,11 @@ #define FSL_SAI_CR2_BCD_MSTR BIT(24) /* SAI Transmit and Recieve Configuration 3 Register */ -#define FSL_SAI_TCR3 0x0c -#define FSL_SAI_RCR3 0x8c #define FSL_SAI_CR3_TRCE BIT(16) #define FSL_SAI_CR3_WDFL(x) (x) #define FSL_SAI_CR3_WDFL_MASK 0x1f /* SAI Transmit and Recieve Configuration 4 Register */ -#define FSL_SAI_TCR4 0x10 -#define FSL_SAI_RCR4 0x90 #define FSL_SAI_CR4_FRSZ(x) (((x) - 1) << 16) #define FSL_SAI_CR4_FRSZ_MASK (0x1f << 16) #define FSL_SAI_CR4_SYWD(x) (((x) - 1) << 8) @@ -69,8 +70,6 @@ #define FSL_SAI_CR4_FSD_MSTR BIT(0) /* SAI Transmit and Recieve Configuration 5 Register */ -#define FSL_SAI_TCR5 0x14 -#define FSL_SAI_RCR5 0x94 #define FSL_SAI_CR5_WNW(x) (((x) - 1) << 24) #define FSL_SAI_CR5_WNW_MASK (0x1f << 24) #define FSL_SAI_CR5_W0W(x) (((x) - 1) << 16) @@ -100,12 +99,11 @@ #define FSL_SAI_MAXBURST_RX 6 struct fsl_sai { - struct clk *clk; - - void __iomem *base; + struct regmap *regmap; bool big_endian_regs; bool big_endian_data; + bool is_dsp_mode; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 4d075f1abe78..6452ca83d889 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -911,8 +911,8 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) { struct fsl_spdif_priv *spdif_private = snd_soc_dai_get_drvdata(dai); - dai->playback_dma_data = &spdif_private->dma_params_tx; - dai->capture_dma_data = &spdif_private->dma_params_rx; + snd_soc_dai_init_dma_data(dai, &spdif_private->dma_params_tx, + &spdif_private->dma_params_rx); snd_soc_add_dai_controls(dai, fsl_spdif_ctrls, ARRAY_SIZE(fsl_spdif_ctrls)); @@ -985,7 +985,7 @@ static bool fsl_spdif_writeable_reg(struct device *dev, unsigned int reg) } } -static const struct regmap_config fsl_spdif_regmap_config = { +static struct regmap_config fsl_spdif_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, @@ -1105,6 +1105,9 @@ static int fsl_spdif_probe(struct platform_device *pdev) memcpy(&spdif_priv->cpu_dai_drv, &fsl_spdif_dai, sizeof(fsl_spdif_dai)); spdif_priv->cpu_dai_drv.name = spdif_priv->name; + if (of_property_read_bool(np, "big-endian")) + fsl_spdif_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index 79cee782dbbf..a2fd7321b5a9 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -160,7 +160,6 @@ static struct platform_driver imx_mc13783_audio_driver = { .driver = { .name = "imx_mc13783", .owner = THIS_MODULE, - .pm = &snd_soc_pm_ops, }, .probe = imx_mc13783_probe, .remove = imx_mc13783_remove diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 6553202dd48c..7abf6a079574 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -270,18 +270,17 @@ static int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) - goto out; + return ret; } if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = imx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) - goto out; + return ret; } -out: - return ret; + return 0; } static int ssi_irq = 0; diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index f2beae78969f..1cb22dd034eb 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -33,8 +33,7 @@ struct imx_sgtl5000_data { static int imx_sgtl5000_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct imx_sgtl5000_data *data = container_of(rtd->card, - struct imx_sgtl5000_data, card); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(rtd->card); struct device *dev = rtd->card->dev; int ret; @@ -159,13 +158,15 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) data->card.dapm_widgets = imx_sgtl5000_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_sgtl5000_dapm_widgets); + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -184,7 +185,8 @@ fail: static int imx_sgtl5000_remove(struct platform_device *pdev) { - struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_sgtl5000_data *data = snd_soc_card_get_drvdata(card); clk_put(data->codec_clk); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 3fd76bc391de..3a3d17ce6ba4 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -71,7 +71,7 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; unsigned int pll_out; int ret; @@ -137,7 +137,7 @@ static int imx_wm8962_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; struct imx_priv *priv = &card_priv; - struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); struct device *dev = &priv->pdev->dev; int ret; @@ -264,13 +264,15 @@ static int imx_wm8962_probe(struct platform_device *pdev) data->card.late_probe = imx_wm8962_late_probe; data->card.set_bias_level = imx_wm8962_set_bias_level; + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); goto clk_fail; } - platform_set_drvdata(pdev, data); of_node_put(ssi_np); of_node_put(codec_np); @@ -289,7 +291,8 @@ fail: static int imx_wm8962_remove(struct platform_device *pdev) { - struct imx_wm8962_data *data = platform_get_drvdata(pdev); + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct imx_wm8962_data *data = snd_soc_card_get_drvdata(card); if (!IS_ERR(data->codec_clk)) clk_disable_unprepare(data->codec_clk); diff --git a/sound/soc/fsl/wm1133-ev1.c b/sound/soc/fsl/wm1133-ev1.c index fce63252bdbb..804749a6c61e 100644 --- a/sound/soc/fsl/wm1133-ev1.c +++ b/sound/soc/fsl/wm1133-ev1.c @@ -214,12 +214,6 @@ static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets, - ARRAY_SIZE(wm1133_ev1_widgets)); - - snd_soc_dapm_add_routes(dapm, wm1133_ev1_map, - ARRAY_SIZE(wm1133_ev1_map)); - /* Headphone jack detection */ snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack); snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), @@ -257,6 +251,11 @@ static struct snd_soc_card wm1133_ev1 = { .owner = THIS_MODULE, .dai_link = &wm1133_ev1_dai, .num_links = 1, + + .dapm_widgets = wm1133_ev1_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets), + .dapm_routes = wm1133_ev1_map, + .num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map), }; static struct platform_device *wm1133_ev1_snd_device; diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 2a1b1b5b5221..2ee8ed56bcf1 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -9,48 +9,72 @@ * published by the Free Software Foundation. */ #include <linux/clk.h> +#include <linux/device.h> #include <linux/module.h> #include <linux/of.h> #include <linux/platform_device.h> #include <linux/string.h> #include <sound/simple_card.h> +#include <sound/soc-dai.h> +#include <sound/soc.h> + +struct simple_card_data { + struct snd_soc_card snd_card; + struct asoc_simple_dai cpu_dai; + struct asoc_simple_dai codec_dai; + struct snd_soc_dai_link snd_link; +}; static int __asoc_simple_card_dai_init(struct snd_soc_dai *dai, - struct asoc_simple_dai *set, - unsigned int daifmt) + struct asoc_simple_dai *set) { - int ret = 0; + int ret; - daifmt |= set->fmt; + if (set->fmt) { + ret = snd_soc_dai_set_fmt(dai, set->fmt); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_fmt error\n"); + goto err; + } + } - if (daifmt) - ret = snd_soc_dai_set_fmt(dai, daifmt); + if (set->sysclk) { + ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_sysclk error\n"); + goto err; + } + } - if (ret == -ENOTSUPP) { - dev_dbg(dai->dev, "ASoC: set_fmt is not supported\n"); - ret = 0; + if (set->slots) { + ret = snd_soc_dai_set_tdm_slot(dai, 0, 0, + set->slots, + set->slot_width); + if (ret && ret != -ENOTSUPP) { + dev_err(dai->dev, "simple-card: set_tdm_slot error\n"); + goto err; + } } - if (!ret && set->sysclk) - ret = snd_soc_dai_set_sysclk(dai, 0, set->sysclk, 0); + ret = 0; +err: return ret; } static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) { - struct asoc_simple_card_info *info = + struct simple_card_data *priv = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_dai *codec = rtd->codec_dai; struct snd_soc_dai *cpu = rtd->cpu_dai; - unsigned int daifmt = info->daifmt; int ret; - ret = __asoc_simple_card_dai_init(codec, &info->codec_dai, daifmt); + ret = __asoc_simple_card_dai_init(codec, &priv->codec_dai); if (ret < 0) return ret; - ret = __asoc_simple_card_dai_init(cpu, &info->cpu_dai, daifmt); + ret = __asoc_simple_card_dai_init(cpu, &priv->cpu_dai); if (ret < 0) return ret; @@ -59,9 +83,12 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) static int asoc_simple_card_sub_parse_of(struct device_node *np, + unsigned int daifmt, struct asoc_simple_dai *dai, - struct device_node **node) + const struct device_node **p_node, + const char **name) { + struct device_node *node; struct clk *clk; int ret; @@ -69,14 +96,20 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * get node via "sound-dai = <&phandle port>" * it will be used as xxx_of_node on soc_bind_dai_link() */ - *node = of_parse_phandle(np, "sound-dai", 0); - if (!*node) + node = of_parse_phandle(np, "sound-dai", 0); + if (!node) return -ENODEV; + *p_node = node; /* get dai->name */ - ret = snd_soc_of_get_dai_name(np, &dai->name); + ret = snd_soc_of_get_dai_name(np, name); if (ret < 0) - goto parse_error; + return ret; + + /* parse TDM slot */ + ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); + if (ret) + return ret; /* * bitclock-inversion, frame-inversion @@ -84,6 +117,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, * and specific "format" if it has */ dai->fmt = snd_soc_of_parse_daifmt(np, NULL); + dai->fmt |= daifmt; /* * dai->sysclk come from @@ -95,7 +129,7 @@ asoc_simple_card_sub_parse_of(struct device_node *np, clk = of_clk_get(np, 0); if (IS_ERR(clk)) { ret = PTR_ERR(clk); - goto parse_error; + return ret; } dai->sysclk = clk_get_rate(clk); @@ -104,37 +138,44 @@ asoc_simple_card_sub_parse_of(struct device_node *np, "system-clock-frequency", &dai->sysclk); } else { - clk = of_clk_get(*node, 0); + clk = of_clk_get(node, 0); if (!IS_ERR(clk)) dai->sysclk = clk_get_rate(clk); } - ret = 0; - -parse_error: - of_node_put(*node); - - return ret; + return 0; } static int asoc_simple_card_parse_of(struct device_node *node, - struct asoc_simple_card_info *info, - struct device *dev, - struct device_node **of_cpu, - struct device_node **of_codec, - struct device_node **of_platform) + struct simple_card_data *priv, + struct device *dev) { + struct snd_soc_dai_link *dai_link = priv->snd_card.dai_link; + struct asoc_simple_dai *codec_dai = &priv->codec_dai; + struct asoc_simple_dai *cpu_dai = &priv->cpu_dai; struct device_node *np; char *name; + unsigned int daifmt; int ret; + /* parsing the card name from DT */ + snd_soc_of_parse_card_name(&priv->snd_card, "simple-audio-card,name"); + /* get CPU/CODEC common format via simple-audio-card,format */ - info->daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & + daifmt = snd_soc_of_parse_daifmt(node, "simple-audio-card,") & (SND_SOC_DAIFMT_FORMAT_MASK | SND_SOC_DAIFMT_INV_MASK); + /* off-codec widgets */ + if (of_property_read_bool(node, "simple-audio-card,widgets")) { + ret = snd_soc_of_parse_audio_simple_widgets(&priv->snd_card, + "simple-audio-card,widgets"); + if (ret) + return ret; + } + /* DAPM routes */ if (of_property_read_bool(node, "simple-audio-card,routing")) { - ret = snd_soc_of_parse_audio_routing(&info->snd_card, + ret = snd_soc_of_parse_audio_routing(&priv->snd_card, "simple-audio-card,routing"); if (ret) return ret; @@ -143,125 +184,173 @@ static int asoc_simple_card_parse_of(struct device_node *node, /* CPU sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,cpu"); - if (np) - ret = asoc_simple_card_sub_parse_of(np, - &info->cpu_dai, - of_cpu); + if (np) { + ret = asoc_simple_card_sub_parse_of(np, daifmt, + cpu_dai, + &dai_link->cpu_of_node, + &dai_link->cpu_dai_name); + of_node_put(np); + } if (ret < 0) return ret; /* CODEC sub-node */ ret = -EINVAL; np = of_get_child_by_name(node, "simple-audio-card,codec"); - if (np) - ret = asoc_simple_card_sub_parse_of(np, - &info->codec_dai, - of_codec); + if (np) { + ret = asoc_simple_card_sub_parse_of(np, daifmt, + codec_dai, + &dai_link->codec_of_node, + &dai_link->codec_dai_name); + of_node_put(np); + } if (ret < 0) return ret; - if (!info->cpu_dai.name || !info->codec_dai.name) + /* + * overwrite cpu_dai->fmt as its DAIFMT_MASTER bit is based on CODEC + * while the other bits should be identical unless buggy SW/HW design. + */ + cpu_dai->fmt = codec_dai->fmt; + + if (!dai_link->cpu_dai_name || !dai_link->codec_dai_name) return -EINVAL; /* card name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(info->cpu_dai.name) + - strlen(info->codec_dai.name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + 2, GFP_KERNEL); - sprintf(name, "%s-%s", info->cpu_dai.name, info->codec_dai.name); - info->name = info->card = name; + sprintf(name, "%s-%s", dai_link->cpu_dai_name, + dai_link->codec_dai_name); + if (!priv->snd_card.name) + priv->snd_card.name = name; + dai_link->name = dai_link->stream_name = name; /* simple-card assumes platform == cpu */ - *of_platform = *of_cpu; + dai_link->platform_of_node = dai_link->cpu_of_node; - dev_dbg(dev, "card-name : %s\n", info->card); - dev_dbg(dev, "platform : %04x\n", info->daifmt); + dev_dbg(dev, "card-name : %s\n", name); + dev_dbg(dev, "platform : %04x\n", daifmt); dev_dbg(dev, "cpu : %s / %04x / %d\n", - info->cpu_dai.name, - info->cpu_dai.fmt, - info->cpu_dai.sysclk); + dai_link->cpu_dai_name, + cpu_dai->fmt, + cpu_dai->sysclk); dev_dbg(dev, "codec : %s / %04x / %d\n", - info->codec_dai.name, - info->codec_dai.fmt, - info->codec_dai.sysclk); + dai_link->codec_dai_name, + codec_dai->fmt, + codec_dai->sysclk); + + /* + * soc_bind_dai_link() will check cpu name + * after of_node matching if dai_link has cpu_dai_name. + * but, it will never match if name was created by fmt_single_name() + * remove cpu_dai_name to escape name matching. + * see + * fmt_single_name() + * fmt_multiple_name() + */ + dai_link->cpu_dai_name = NULL; return 0; } +/* update the reference count of the devices nodes at end of probe */ +static int asoc_simple_card_unref(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct snd_soc_dai_link *dai_link; + struct device_node *np; + int num_links; + + for (num_links = 0, dai_link = card->dai_link; + num_links < card->num_links; + num_links++, dai_link++) { + np = (struct device_node *) dai_link->cpu_of_node; + if (np) + of_node_put(np); + np = (struct device_node *) dai_link->codec_of_node; + if (np) + of_node_put(np); + } + return 0; +} + static int asoc_simple_card_probe(struct platform_device *pdev) { - struct asoc_simple_card_info *cinfo; + struct simple_card_data *priv; + struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; - struct device_node *of_cpu, *of_codec, *of_platform; struct device *dev = &pdev->dev; int ret; - cinfo = NULL; - of_cpu = NULL; - of_codec = NULL; - of_platform = NULL; - - cinfo = devm_kzalloc(dev, sizeof(*cinfo), GFP_KERNEL); - if (!cinfo) + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) return -ENOMEM; + /* + * init snd_soc_card + */ + priv->snd_card.owner = THIS_MODULE; + priv->snd_card.dev = dev; + dai_link = &priv->snd_link; + priv->snd_card.dai_link = dai_link; + priv->snd_card.num_links = 1; + if (np && of_device_is_available(np)) { - cinfo->snd_card.dev = dev; - ret = asoc_simple_card_parse_of(np, cinfo, dev, - &of_cpu, - &of_codec, - &of_platform); + ret = asoc_simple_card_parse_of(np, priv, dev); if (ret < 0) { if (ret != -EPROBE_DEFER) dev_err(dev, "parse error %d\n", ret); - return ret; + goto err; } } else { - if (!dev->platform_data) { + struct asoc_simple_card_info *cinfo; + + cinfo = dev->platform_data; + if (!cinfo) { dev_err(dev, "no info for asoc-simple-card\n"); return -EINVAL; } - memcpy(cinfo, dev->platform_data, sizeof(*cinfo)); - cinfo->snd_card.dev = dev; - } + if (!cinfo->name || + !cinfo->codec_dai.name || + !cinfo->codec || + !cinfo->platform || + !cinfo->cpu_dai.name) { + dev_err(dev, "insufficient asoc_simple_card_info settings\n"); + return -EINVAL; + } - if (!cinfo->name || - !cinfo->card || - !cinfo->codec_dai.name || - !(cinfo->codec || of_codec) || - !(cinfo->platform || of_platform) || - !(cinfo->cpu_dai.name || of_cpu)) { - dev_err(dev, "insufficient asoc_simple_card_info settings\n"); - return -EINVAL; + priv->snd_card.name = (cinfo->card) ? cinfo->card : cinfo->name; + dai_link->name = cinfo->name; + dai_link->stream_name = cinfo->name; + dai_link->platform_name = cinfo->platform; + dai_link->codec_name = cinfo->codec; + dai_link->cpu_dai_name = cinfo->cpu_dai.name; + dai_link->codec_dai_name = cinfo->codec_dai.name; + memcpy(&priv->cpu_dai, &cinfo->cpu_dai, + sizeof(priv->cpu_dai)); + memcpy(&priv->codec_dai, &cinfo->codec_dai, + sizeof(priv->codec_dai)); + + priv->cpu_dai.fmt |= cinfo->daifmt; + priv->codec_dai.fmt |= cinfo->daifmt; } /* * init snd_soc_dai_link */ - cinfo->snd_link.name = cinfo->name; - cinfo->snd_link.stream_name = cinfo->name; - cinfo->snd_link.cpu_dai_name = cinfo->cpu_dai.name; - cinfo->snd_link.platform_name = cinfo->platform; - cinfo->snd_link.codec_name = cinfo->codec; - cinfo->snd_link.codec_dai_name = cinfo->codec_dai.name; - cinfo->snd_link.cpu_of_node = of_cpu; - cinfo->snd_link.codec_of_node = of_codec; - cinfo->snd_link.platform_of_node = of_platform; - cinfo->snd_link.init = asoc_simple_card_dai_init; + dai_link->init = asoc_simple_card_dai_init; - /* - * init snd_soc_card - */ - cinfo->snd_card.name = cinfo->card; - cinfo->snd_card.owner = THIS_MODULE; - cinfo->snd_card.dai_link = &cinfo->snd_link; - cinfo->snd_card.num_links = 1; + snd_soc_card_set_drvdata(&priv->snd_card, priv); - snd_soc_card_set_drvdata(&cinfo->snd_card, cinfo); + ret = devm_snd_soc_register_card(&pdev->dev, &priv->snd_card); - return devm_snd_soc_register_card(&pdev->dev, &cinfo->snd_card); +err: + asoc_simple_card_unref(pdev); + return ret; } static const struct of_device_id asoc_simple_of_match[] = { diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 61c10bf503d2..3c81b3891209 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -2,12 +2,50 @@ config SND_MFLD_MACHINE tristate "SOC Machine Audio driver for Intel Medfield MID platform" depends on INTEL_SCU_IPC select SND_SOC_SN95031 - select SND_SST_PLATFORM + select SND_SST_MFLD_PLATFORM help This adds support for ASoC machine driver for Intel(R) MID Medfield platform used as alsa device in audio substem in Intel(R) MID devices Say Y if you have such a device If unsure select "N". -config SND_SST_PLATFORM +config SND_SST_MFLD_PLATFORM tristate + +config SND_SOC_INTEL_SST + tristate "ASoC support for Intel(R) Smart Sound Technology" + select SND_SOC_INTEL_SST_ACPI if ACPI + depends on (X86 || COMPILE_TEST) + help + This adds support for Intel(R) Smart Sound Technology (SST). + Say Y if you have such a device + If unsure select "N". + +config SND_SOC_INTEL_SST_ACPI + tristate + +config SND_SOC_INTEL_HASWELL + tristate + +config SND_SOC_INTEL_BAYTRAIL + tristate + +config SND_SOC_INTEL_HASWELL_MACH + tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C + select SND_SOC_INTEL_HASWELL + select SND_SOC_RT5640 + help + This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell + Ultrabook platforms. + Say Y if you have such a device + If unsure select "N". + +config SND_SOC_INTEL_BYT_RT5640_MACH + tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && I2C + select SND_SOC_INTEL_BAYTRAIL + select SND_SOC_RT5640 + help + This adds audio driver for Intel Baytrail platform based boards + with the RT5640 audio codec. diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 639883339465..edeb79ae3dff 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -1,5 +1,28 @@ -snd-soc-sst-platform-objs := sst_platform.o +# Core support +snd-soc-sst-dsp-objs := sst-dsp.o sst-firmware.o +snd-soc-sst-acpi-objs := sst-acpi.o + +snd-soc-sst-mfld-platform-objs := sst-mfld-platform.o snd-soc-mfld-machine-objs := mfld_machine.o -obj-$(CONFIG_SND_SST_PLATFORM) += snd-soc-sst-platform.o +obj-$(CONFIG_SND_SST_MFLD_PLATFORM) += snd-soc-sst-mfld-platform.o obj-$(CONFIG_SND_MFLD_MACHINE) += snd-soc-mfld-machine.o + +obj-$(CONFIG_SND_SOC_INTEL_SST) += snd-soc-sst-dsp.o +obj-$(CONFIG_SND_SOC_INTEL_SST_ACPI) += snd-soc-sst-acpi.o + +# Platform Support +snd-soc-sst-haswell-pcm-objs := \ + sst-haswell-ipc.o sst-haswell-pcm.o sst-haswell-dsp.o +snd-soc-sst-baytrail-pcm-objs := \ + sst-baytrail-ipc.o sst-baytrail-pcm.o sst-baytrail-dsp.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL) += snd-soc-sst-haswell-pcm.o +obj-$(CONFIG_SND_SOC_INTEL_BAYTRAIL) += snd-soc-sst-baytrail-pcm.o + +# Machine support +snd-soc-sst-haswell-objs := haswell.o +snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o + +obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o +obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c new file mode 100644 index 000000000000..eff97c8e5218 --- /dev/null +++ b/sound/soc/intel/byt-rt5640.c @@ -0,0 +1,187 @@ +/* + * Intel Baytrail SST RT5640 machine driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../codecs/rt5640.h" + +#include "sst-dsp.h" + +static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Internal Mic", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { + {"IN2P", NULL, "Headset Mic"}, + {"IN2N", NULL, "Headset Mic"}, + {"DMIC1", NULL, "Internal Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Speaker", NULL, "SPOLP"}, + {"Speaker", NULL, "SPOLN"}, + {"Speaker", NULL, "SPORP"}, + {"Speaker", NULL, "SPORN"}, +}; + +static const struct snd_kcontrol_new byt_rt5640_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int byt_rt5640_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_PLL1, + params_rate(params) * 256, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set codec clock %d\n", ret); + return ret; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5640_PLL1_S_BCLK1, + params_rate(params) * 64, + params_rate(params) * 256); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set codec pll: %d\n", ret); + return ret; + } + return 0; +} + +static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_codec *codec = runtime->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = runtime->card; + + card->dapm.idle_bias_off = true; + + ret = snd_soc_add_card_controls(card, byt_rt5640_controls, + ARRAY_SIZE(byt_rt5640_controls)); + if (ret) { + dev_err(card->dev, "unable to add card controls\n"); + return ret; + } + + snd_soc_dapm_ignore_suspend(dapm, "HPOL"); + snd_soc_dapm_ignore_suspend(dapm, "HPOR"); + + snd_soc_dapm_ignore_suspend(dapm, "SPOLP"); + snd_soc_dapm_ignore_suspend(dapm, "SPOLN"); + snd_soc_dapm_ignore_suspend(dapm, "SPORP"); + snd_soc_dapm_ignore_suspend(dapm, "SPORN"); + + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphone"); + snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin(dapm, "Internal Mic"); + + snd_soc_dapm_sync(dapm); + return ret; +} + +static struct snd_soc_ops byt_rt5640_ops = { + .hw_params = byt_rt5640_hw_params, +}; + +static struct snd_soc_dai_link byt_rt5640_dais[] = { + { + .name = "Baytrail Audio", + .stream_name = "Audio", + .cpu_dai_name = "Front-cpu-dai", + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .platform_name = "baytrail-pcm-audio", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = byt_rt5640_init, + .ignore_suspend = 1, + .ops = &byt_rt5640_ops, + }, + { + .name = "Baytrail Voice", + .stream_name = "Voice", + .cpu_dai_name = "Mic1-cpu-dai", + .codec_dai_name = "rt5640-aif1", + .codec_name = "i2c-10EC5640:00", + .platform_name = "baytrail-pcm-audio", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .init = NULL, + .ignore_suspend = 1, + .ops = &byt_rt5640_ops, + }, +}; + +static struct snd_soc_card byt_rt5640_card = { + .name = "byt-rt5640", + .dai_link = byt_rt5640_dais, + .num_links = ARRAY_SIZE(byt_rt5640_dais), + .dapm_widgets = byt_rt5640_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_rt5640_widgets), + .dapm_routes = byt_rt5640_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_rt5640_audio_map), +}; + +static int byt_rt5640_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &byt_rt5640_card; + struct device *dev = &pdev->dev; + + card->dev = &pdev->dev; + dev_set_drvdata(dev, card); + return snd_soc_register_card(card); +} + +static int byt_rt5640_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver byt_rt5640_audio = { + .probe = byt_rt5640_probe, + .remove = byt_rt5640_remove, + .driver = { + .name = "byt-rt5640", + .owner = THIS_MODULE, + }, +}; +module_platform_driver(byt_rt5640_audio) + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail Machine driver"); +MODULE_AUTHOR("Omair Md Abdullah, Jarkko Nikula"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:byt-rt5640"); diff --git a/sound/soc/intel/haswell.c b/sound/soc/intel/haswell.c new file mode 100644 index 000000000000..54345a2a7386 --- /dev/null +++ b/sound/soc/intel/haswell.c @@ -0,0 +1,235 @@ +/* + * Intel Haswell Lynxpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#include "sst-dsp.h" +#include "sst-haswell-ipc.h" + +#include "../codecs/rt5640.h" + +/* Haswell ULT platforms have a Headphone and Mic jack */ +static const struct snd_soc_dapm_widget haswell_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route haswell_rt5640_map[] = { + + {"Headphones", NULL, "HPOR"}, + {"Headphones", NULL, "HPOL"}, + {"IN2P", NULL, "Mic"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static int haswell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP0 to 16 bit */ + snd_mask_set(¶ms->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S16_LE); + return 0; +} + +static int haswell_rt5640_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5640_SCLK_S_MCLK, 12288000, + SND_SOC_CLOCK_IN); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + /* set correct codec filter for DAI format and clock config */ + snd_soc_update_bits(rtd->codec, 0x83, 0xffff, 0x8000); + + return ret; +} + +static struct snd_soc_ops haswell_rt5640_ops = { + .hw_params = haswell_rt5640_hw_params, +}; + +static int haswell_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sst_pdata *pdata = dev_get_platdata(rtd->platform->dev); + struct sst_hsw *haswell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings */ + ret = sst_hsw_device_set_config(haswell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_CLOCK_MASTER, 9); + if (ret < 0) { + dev_err(rtd->dev, "failed to set device config\n"); + return ret; + } + + /* always connected */ + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Mic"); + + return 0; +} + +static struct snd_soc_dai_link haswell_rt5640_dais[] = { + /* Front End DAI links */ + { + .name = "System", + .stream_name = "System Playback", + .cpu_dai_name = "System Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = haswell_rtd_init, + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload0", + .stream_name = "Offload0 Playback", + .cpu_dai_name = "Offload0 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Offload1", + .stream_name = "Offload1 Playback", + .cpu_dai_name = "Offload1 Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + { + .name = "Loopback", + .stream_name = "Loopback", + .cpu_dai_name = "Loopback Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 0, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + { + .name = "Capture", + .stream_name = "Capture", + .cpu_dai_name = "Capture Pin", + .platform_name = "haswell-pcm-audio", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = {SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .be_id = 0, + .cpu_dai_name = "snd-soc-dummy-dai", + .platform_name = "snd-soc-dummy", + .no_pcm = 1, + .codec_name = "i2c-INT33CA:00", + .codec_dai_name = "rt5640-aif1", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = haswell_ssp0_fixup, + .ops = &haswell_rt5640_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, +}; + +/* audio machine driver for Haswell Lynxpoint DSP + RT5640 */ +static struct snd_soc_card haswell_rt5640 = { + .name = "haswell-rt5640", + .owner = THIS_MODULE, + .dai_link = haswell_rt5640_dais, + .num_links = ARRAY_SIZE(haswell_rt5640_dais), + .dapm_widgets = haswell_widgets, + .num_dapm_widgets = ARRAY_SIZE(haswell_widgets), + .dapm_routes = haswell_rt5640_map, + .num_dapm_routes = ARRAY_SIZE(haswell_rt5640_map), + .fully_routed = true, +}; + +static int haswell_audio_probe(struct platform_device *pdev) +{ + haswell_rt5640.dev = &pdev->dev; + + return snd_soc_register_card(&haswell_rt5640); +} + +static int haswell_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&haswell_rt5640); + return 0; +} + +static struct platform_driver haswell_audio = { + .probe = haswell_audio_probe, + .remove = haswell_audio_remove, + .driver = { + .name = "haswell-audio", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(haswell_audio) + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Intel SST Audio for Haswell Lynxpoint"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:haswell-audio"); diff --git a/sound/soc/intel/mfld_machine.c b/sound/soc/intel/mfld_machine.c index d3d4c32434f7..031d78783fc8 100644 --- a/sound/soc/intel/mfld_machine.c +++ b/sound/soc/intel/mfld_machine.c @@ -53,6 +53,7 @@ enum soc_mic_bias_zones { static unsigned int hs_switch; static unsigned int lo_dac; +static struct snd_soc_codec *mfld_codec; struct mfld_mc_private { void __iomem *int_base; @@ -100,40 +101,47 @@ static int headset_get_switch(struct snd_kcontrol *kcontrol, static int headset_set_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &card->dapm; if (ucontrol->value.integer.value[0] == hs_switch) return 0; + snd_soc_dapm_mutex_lock(dapm); + if (ucontrol->value.integer.value[0]) { pr_debug("hs_set HS path\n"); - snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); } else { pr_debug("hs_set EP path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); } - snd_soc_dapm_sync(&codec->dapm); + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + hs_switch = ucontrol->value.integer.value[0]; return 0; } -static void lo_enable_out_pins(struct snd_soc_codec *codec) +static void lo_enable_out_pins(struct snd_soc_dapm_context *dapm) { - snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "IHFOUTR"); - snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTL"); - snd_soc_dapm_enable_pin(&codec->dapm, "LINEOUTR"); - snd_soc_dapm_enable_pin(&codec->dapm, "VIB1OUT"); - snd_soc_dapm_enable_pin(&codec->dapm, "VIB2OUT"); + snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTL"); + snd_soc_dapm_enable_pin_unlocked(dapm, "IHFOUTR"); + snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTL"); + snd_soc_dapm_enable_pin_unlocked(dapm, "LINEOUTR"); + snd_soc_dapm_enable_pin_unlocked(dapm, "VIB1OUT"); + snd_soc_dapm_enable_pin_unlocked(dapm, "VIB2OUT"); if (hs_switch) { - snd_soc_dapm_enable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_enable_pin(&codec->dapm, "EPOUT"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_enable_pin_unlocked(dapm, "EPOUT"); } } @@ -147,45 +155,53 @@ static int lo_get_switch(struct snd_kcontrol *kcontrol, static int lo_set_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = &card->dapm; if (ucontrol->value.integer.value[0] == lo_dac) return 0; + snd_soc_dapm_mutex_lock(dapm); + /* we dont want to work with last state of lineout so just enable all * pins and then disable pins not required */ - lo_enable_out_pins(codec); + lo_enable_out_pins(dapm); + switch (ucontrol->value.integer.value[0]) { case 0: pr_debug("set vibra path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "VIB1OUT"); - snd_soc_dapm_disable_pin(&codec->dapm, "VIB2OUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0); + snd_soc_dapm_disable_pin_unlocked(dapm, "VIB1OUT"); + snd_soc_dapm_disable_pin_unlocked(dapm, "VIB2OUT"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0); break; case 1: pr_debug("set hs path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "Headphones"); - snd_soc_dapm_disable_pin(&codec->dapm, "EPOUT"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x22); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphones"); + snd_soc_dapm_disable_pin_unlocked(dapm, "EPOUT"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x22); break; case 2: pr_debug("set spkr path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "IHFOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x44); + snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTL"); + snd_soc_dapm_disable_pin_unlocked(dapm, "IHFOUTR"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x44); break; case 3: pr_debug("set null path\n"); - snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTL"); - snd_soc_dapm_disable_pin(&codec->dapm, "LINEOUTR"); - snd_soc_update_bits(codec, SN95031_LOCTL, 0x66, 0x66); + snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTL"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LINEOUTR"); + snd_soc_update_bits(mfld_codec, SN95031_LOCTL, 0x66, 0x66); break; } - snd_soc_dapm_sync(&codec->dapm); + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); + lo_dac = ucontrol->value.integer.value[0]; return 0; } @@ -221,26 +237,11 @@ static void mfld_jack_check(unsigned int intr_status) static int mfld_init(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_codec *codec = runtime->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &runtime->card->dapm; int ret_val; - /* Add jack sense widgets */ - snd_soc_dapm_new_controls(dapm, mfld_widgets, ARRAY_SIZE(mfld_widgets)); - - /* Set up the map */ - snd_soc_dapm_add_routes(dapm, mfld_map, ARRAY_SIZE(mfld_map)); + mfld_codec = runtime->codec; - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Headphones"); - snd_soc_dapm_enable_pin(dapm, "Mic"); - - ret_val = snd_soc_add_codec_controls(codec, mfld_snd_controls, - ARRAY_SIZE(mfld_snd_controls)); - if (ret_val) { - pr_err("soc_add_controls failed %d", ret_val); - return ret_val; - } /* default is earpiece pin, userspace sets it explcitly */ snd_soc_dapm_disable_pin(dapm, "Headphones"); /* default is lineout NC, userspace sets it explcitly */ @@ -253,7 +254,7 @@ static int mfld_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_disable_pin(dapm, "LINEINR"); /* Headset and button jack detection */ - ret_val = snd_soc_jack_new(codec, "Intel(R) MID Audio Jack", + ret_val = snd_soc_jack_new(mfld_codec, "Intel(R) MID Audio Jack", SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1, &mfld_jack); if (ret_val) { @@ -335,6 +336,13 @@ static struct snd_soc_card snd_soc_card_mfld = { .owner = THIS_MODULE, .dai_link = mfld_msic_dailink, .num_links = ARRAY_SIZE(mfld_msic_dailink), + + .controls = mfld_snd_controls, + .num_controls = ARRAY_SIZE(mfld_snd_controls), + .dapm_widgets = mfld_widgets, + .num_dapm_widgets = ARRAY_SIZE(mfld_widgets), + .dapm_routes = mfld_map, + .num_dapm_routes = ARRAY_SIZE(mfld_map), }; static irqreturn_t snd_mfld_jack_intr_handler(int irq, void *dev) diff --git a/sound/soc/intel/sst-acpi.c b/sound/soc/intel/sst-acpi.c new file mode 100644 index 000000000000..5d06eecb6198 --- /dev/null +++ b/sound/soc/intel/sst-acpi.c @@ -0,0 +1,284 @@ +/* + * Intel SST loader on ACPI systems + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/firmware.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include "sst-dsp.h" + +#define SST_LPT_DSP_DMA_ADDR_OFFSET 0x0F0000 +#define SST_WPT_DSP_DMA_ADDR_OFFSET 0x0FE000 +#define SST_LPT_DSP_DMA_SIZE (1024 - 1) + +/* Descriptor for SST ASoC machine driver */ +struct sst_acpi_mach { + /* ACPI ID for the matching machine driver. Audio codec for instance */ + const u8 id[ACPI_ID_LEN]; + /* machine driver name */ + const char *drv_name; + /* firmware file name */ + const char *fw_filename; +}; + +/* Descriptor for setting up SST platform data */ +struct sst_acpi_desc { + const char *drv_name; + struct sst_acpi_mach *machines; + /* Platform resource indexes. Must set to -1 if not used */ + int resindex_lpe_base; + int resindex_pcicfg_base; + int resindex_fw_base; + int irqindex_host_ipc; + int resindex_dma_base; + /* Unique number identifying the SST core on platform */ + int sst_id; + /* DMA only valid when resindex_dma_base != -1*/ + int dma_engine; + int dma_size; +}; + +struct sst_acpi_priv { + struct platform_device *pdev_mach; + struct platform_device *pdev_pcm; + struct sst_pdata sst_pdata; + struct sst_acpi_desc *desc; + struct sst_acpi_mach *mach; +}; + +static void sst_acpi_fw_cb(const struct firmware *fw, void *context) +{ + struct platform_device *pdev = context; + struct device *dev = &pdev->dev; + struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; + struct sst_acpi_desc *desc = sst_acpi->desc; + struct sst_acpi_mach *mach = sst_acpi->mach; + + sst_pdata->fw = fw; + if (!fw) { + dev_err(dev, "Cannot load firmware %s\n", mach->fw_filename); + return; + } + + /* register PCM and DAI driver */ + sst_acpi->pdev_pcm = + platform_device_register_data(dev, desc->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_pcm)) { + dev_err(dev, "Cannot register device %s. Error %d\n", + desc->drv_name, (int)PTR_ERR(sst_acpi->pdev_pcm)); + } + + return; +} + +static acpi_status sst_acpi_mach_match(acpi_handle handle, u32 level, + void *context, void **ret) +{ + *(bool *)context = true; + return AE_OK; +} + +static struct sst_acpi_mach *sst_acpi_find_machine( + struct sst_acpi_mach *machines) +{ + struct sst_acpi_mach *mach; + bool found = false; + + for (mach = machines; mach->id[0]; mach++) + if (ACPI_SUCCESS(acpi_get_devices(mach->id, + sst_acpi_mach_match, + &found, NULL)) && found) + return mach; + + return NULL; +} + +static int sst_acpi_probe(struct platform_device *pdev) +{ + const struct acpi_device_id *id; + struct device *dev = &pdev->dev; + struct sst_acpi_priv *sst_acpi; + struct sst_pdata *sst_pdata; + struct sst_acpi_mach *mach; + struct sst_acpi_desc *desc; + struct resource *mmio; + int ret = 0; + + sst_acpi = devm_kzalloc(dev, sizeof(*sst_acpi), GFP_KERNEL); + if (sst_acpi == NULL) + return -ENOMEM; + + id = acpi_match_device(dev->driver->acpi_match_table, dev); + if (!id) + return -ENODEV; + + desc = (struct sst_acpi_desc *)id->driver_data; + mach = sst_acpi_find_machine(desc->machines); + if (mach == NULL) { + dev_err(dev, "No matching ASoC machine driver found\n"); + return -ENODEV; + } + + sst_pdata = &sst_acpi->sst_pdata; + sst_pdata->id = desc->sst_id; + sst_acpi->desc = desc; + sst_acpi->mach = mach; + + if (desc->resindex_dma_base >= 0) { + sst_pdata->dma_engine = desc->dma_engine; + sst_pdata->dma_base = desc->resindex_dma_base; + sst_pdata->dma_size = desc->dma_size; + } + + if (desc->irqindex_host_ipc >= 0) + sst_pdata->irq = platform_get_irq(pdev, desc->irqindex_host_ipc); + + if (desc->resindex_lpe_base >= 0) { + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_lpe_base); + if (mmio) { + sst_pdata->lpe_base = mmio->start; + sst_pdata->lpe_size = resource_size(mmio); + } + } + + if (desc->resindex_pcicfg_base >= 0) { + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_pcicfg_base); + if (mmio) { + sst_pdata->pcicfg_base = mmio->start; + sst_pdata->pcicfg_size = resource_size(mmio); + } + } + + if (desc->resindex_fw_base >= 0) { + mmio = platform_get_resource(pdev, IORESOURCE_MEM, + desc->resindex_fw_base); + if (mmio) { + sst_pdata->fw_base = mmio->start; + sst_pdata->fw_size = resource_size(mmio); + } + } + + platform_set_drvdata(pdev, sst_acpi); + + /* register machine driver */ + sst_acpi->pdev_mach = + platform_device_register_data(dev, mach->drv_name, -1, + sst_pdata, sizeof(*sst_pdata)); + if (IS_ERR(sst_acpi->pdev_mach)) + return PTR_ERR(sst_acpi->pdev_mach); + + /* continue SST probing after firmware is loaded */ + ret = request_firmware_nowait(THIS_MODULE, true, mach->fw_filename, + dev, GFP_KERNEL, pdev, sst_acpi_fw_cb); + if (ret) + platform_device_unregister(sst_acpi->pdev_mach); + + return ret; +} + +static int sst_acpi_remove(struct platform_device *pdev) +{ + struct sst_acpi_priv *sst_acpi = platform_get_drvdata(pdev); + struct sst_pdata *sst_pdata = &sst_acpi->sst_pdata; + + platform_device_unregister(sst_acpi->pdev_mach); + if (!IS_ERR_OR_NULL(sst_acpi->pdev_pcm)) + platform_device_unregister(sst_acpi->pdev_pcm); + release_firmware(sst_pdata->fw); + + return 0; +} + +static struct sst_acpi_mach haswell_machines[] = { + { "INT33CA", "haswell-audio", "intel/IntcSST1.bin" }, + {} +}; + +static struct sst_acpi_desc sst_acpi_haswell_desc = { + .drv_name = "haswell-pcm-audio", + .machines = haswell_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_fw_base = -1, + .irqindex_host_ipc = 0, + .sst_id = SST_DEV_ID_LYNX_POINT, + .dma_engine = SST_DMA_TYPE_DW, + .resindex_dma_base = SST_LPT_DSP_DMA_ADDR_OFFSET, + .dma_size = SST_LPT_DSP_DMA_SIZE, +}; + +static struct sst_acpi_mach broadwell_machines[] = { + { "INT343A", "broadwell-audio", "intel/IntcSST2.bin" }, + {} +}; + +static struct sst_acpi_desc sst_acpi_broadwell_desc = { + .drv_name = "haswell-pcm-audio", + .machines = broadwell_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_fw_base = -1, + .irqindex_host_ipc = 0, + .sst_id = SST_DEV_ID_WILDCAT_POINT, + .dma_engine = SST_DMA_TYPE_DW, + .resindex_dma_base = SST_WPT_DSP_DMA_ADDR_OFFSET, + .dma_size = SST_LPT_DSP_DMA_SIZE, +}; + +static struct sst_acpi_mach baytrail_machines[] = { + { "10EC5640", "byt-rt5640", "intel/fw_sst_0f28.bin-i2s_master" }, + {} +}; + +static struct sst_acpi_desc sst_acpi_baytrail_desc = { + .drv_name = "baytrail-pcm-audio", + .machines = baytrail_machines, + .resindex_lpe_base = 0, + .resindex_pcicfg_base = 1, + .resindex_fw_base = 2, + .irqindex_host_ipc = 5, + .sst_id = SST_DEV_ID_BYT, + .resindex_dma_base = -1, +}; + +static struct acpi_device_id sst_acpi_match[] = { + { "INT33C8", (unsigned long)&sst_acpi_haswell_desc }, + { "INT3438", (unsigned long)&sst_acpi_broadwell_desc }, + { "80860F28", (unsigned long)&sst_acpi_baytrail_desc }, + { } +}; +MODULE_DEVICE_TABLE(acpi, sst_acpi_match); + +static struct platform_driver sst_acpi_driver = { + .probe = sst_acpi_probe, + .remove = sst_acpi_remove, + .driver = { + .name = "sst-acpi", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(sst_acpi_match), + }, +}; +module_platform_driver(sst_acpi_driver); + +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@linux.intel.com>"); +MODULE_DESCRIPTION("Intel SST loader on ACPI systems"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/intel/sst-baytrail-dsp.c b/sound/soc/intel/sst-baytrail-dsp.c new file mode 100644 index 000000000000..a50bf7fc0e3a --- /dev/null +++ b/sound/soc/intel/sst-baytrail-dsp.c @@ -0,0 +1,372 @@ +/* + * Intel Baytrail SST DSP driver + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include <linux/delay.h> +#include <linux/fs.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-baytrail-ipc.h" + +#define SST_BYT_FW_SIGNATURE_SIZE 4 +#define SST_BYT_FW_SIGN "$SST" + +#define SST_BYT_IRAM_OFFSET 0xC0000 +#define SST_BYT_DRAM_OFFSET 0x100000 +#define SST_BYT_SHIM_OFFSET 0x140000 + +enum sst_ram_type { + SST_BYT_IRAM = 1, + SST_BYT_DRAM = 2, + SST_BYT_CACHE = 3, +}; + +struct dma_block_info { + enum sst_ram_type type; /* IRAM/DRAM */ + u32 size; /* Bytes */ + u32 ram_offset; /* Offset in I/DRAM */ + u32 rsvd; /* Reserved field */ +}; + +struct fw_header { + unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE]; + u32 file_size; /* size of fw minus this header */ + u32 modules; /* # of modules */ + u32 file_format; /* version of header format */ + u32 reserved[4]; +}; + +struct sst_byt_fw_module_header { + unsigned char signature[SST_BYT_FW_SIGNATURE_SIZE]; + u32 mod_size; /* size of module */ + u32 blocks; /* # of blocks */ + u32 type; /* codec type, pp lib */ + u32 entry_point; +}; + +static int sst_byt_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, + struct sst_byt_fw_module_header *module) +{ + struct dma_block_info *block; + struct sst_module *mod; + struct sst_module_data block_data; + struct sst_module_template template; + int count; + + memset(&template, 0, sizeof(template)); + template.id = module->type; + template.entry = module->entry_point; + template.p.type = SST_MEM_DRAM; + template.p.data_type = SST_DATA_P; + template.s.type = SST_MEM_DRAM; + template.s.data_type = SST_DATA_S; + + mod = sst_module_new(fw, &template, NULL); + if (mod == NULL) + return -ENOMEM; + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + + if (block->size <= 0) { + dev_err(dsp->dev, "block %d size invalid\n", count); + return -EINVAL; + } + + switch (block->type) { + case SST_BYT_IRAM: + block_data.offset = block->ram_offset + + dsp->addr.iram_offset; + block_data.type = SST_MEM_IRAM; + break; + case SST_BYT_DRAM: + block_data.offset = block->ram_offset + + dsp->addr.dram_offset; + block_data.type = SST_MEM_DRAM; + break; + case SST_BYT_CACHE: + block_data.offset = block->ram_offset + + (dsp->addr.fw_ext - dsp->addr.lpe); + block_data.type = SST_MEM_CACHE; + break; + default: + dev_err(dsp->dev, "wrong ram type 0x%x in block0x%x\n", + block->type, count); + return -EINVAL; + } + + block_data.size = block->size; + block_data.data_type = SST_DATA_M; + block_data.data = (void *)block + sizeof(*block); + + sst_module_insert_fixed_block(mod, &block_data); + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +static int sst_byt_parse_fw_image(struct sst_fw *sst_fw) +{ + struct fw_header *header; + struct sst_byt_fw_module_header *module; + struct sst_dsp *dsp = sst_fw->dsp; + int ret, count; + + /* Read the header information from the data pointer */ + header = (struct fw_header *)sst_fw->dma_buf; + + /* verify FW */ + if ((strncmp(header->signature, SST_BYT_FW_SIGN, 4) != 0) || + (sst_fw->size != header->file_size + sizeof(*header))) { + /* Invalid FW signature */ + dev_err(dsp->dev, "Invalid FW sign/filesize mismatch\n"); + return -EINVAL; + } + + dev_dbg(dsp->dev, + "header sign=%4s size=0x%x modules=0x%x fmt=0x%x size=%zu\n", + header->signature, header->file_size, header->modules, + header->file_format, sizeof(*header)); + + module = (void *)sst_fw->dma_buf + sizeof(*header); + for (count = 0; count < header->modules; count++) { + /* module */ + ret = sst_byt_parse_module(dsp, sst_fw, module); + if (ret < 0) { + dev_err(dsp->dev, "invalid module %d\n", count); + return ret; + } + module = (void *)module + sizeof(*module) + module->mod_size; + } + + return 0; +} + +static void sst_byt_dump_shim(struct sst_dsp *sst) +{ + int i; + u64 reg; + + for (i = 0; i <= 0xF0; i += 8) { + reg = sst_dsp_shim_read64_unlocked(sst, i); + if (reg) + dev_dbg(sst->dev, "shim 0x%2.2x value 0x%16.16llx\n", + i, reg); + } + + for (i = 0x00; i <= 0xff; i += 4) { + reg = readl(sst->addr.pci_cfg + i); + if (reg) + dev_dbg(sst->dev, "pci 0x%2.2x value 0x%8.8x\n", + i, (u32)reg); + } +} + +static irqreturn_t sst_byt_irq(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + u64 isrx; + irqreturn_t ret = IRQ_NONE; + + spin_lock(&sst->spinlock); + + isrx = sst_dsp_shim_read64_unlocked(sst, SST_ISRX); + if (isrx & SST_ISRX_DONE) { + /* ADSP has processed the message request from IA */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCX, + SST_BYT_IPCX_DONE, 0); + ret = IRQ_WAKE_THREAD; + } + if (isrx & SST_BYT_ISRX_REQUEST) { + /* mask message request from ADSP and do processing later */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX, + SST_BYT_IMRX_REQUEST, + SST_BYT_IMRX_REQUEST); + ret = IRQ_WAKE_THREAD; + } + + spin_unlock(&sst->spinlock); + + return ret; +} + +static void sst_byt_boot(struct sst_dsp *sst) +{ + int tries = 10; + + /* release stall and wait to unstall */ + sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_STALL, 0x0); + while (tries--) { + if (!(sst_dsp_shim_read64(sst, SST_CSR) & + SST_BYT_CSR_PWAITMODE)) + break; + msleep(100); + } + if (tries < 0) { + dev_err(sst->dev, "unable to start DSP\n"); + sst_byt_dump_shim(sst); + } +} + +static void sst_byt_reset(struct sst_dsp *sst) +{ + /* put DSP into reset, set reset vector and stall */ + sst_dsp_shim_update_bits64(sst, SST_CSR, + SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL, + SST_BYT_CSR_RST | SST_BYT_CSR_VECTOR_SEL | SST_BYT_CSR_STALL); + + udelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits64(sst, SST_CSR, SST_BYT_CSR_RST, 0); +} + +struct sst_adsp_memregion { + u32 start; + u32 end; + int blocks; + enum sst_mem_type type; +}; + +/* BYT test stuff */ +static const struct sst_adsp_memregion byt_region[] = { + {0xC0000, 0x100000, 8, SST_MEM_IRAM}, /* I-SRAM - 8 * 32kB */ + {0x100000, 0x140000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ +}; + +static int sst_byt_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + sst->addr.lpe_base = pdata->lpe_base; + sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); + if (!sst->addr.lpe) + return -ENODEV; + + /* ADSP PCI MMIO config space */ + sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size); + if (!sst->addr.pci_cfg) { + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Extended FW allocation */ + sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size); + if (!sst->addr.fw_ext) { + iounmap(sst->addr.pci_cfg); + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Shim */ + sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset; + + sst_dsp_mailbox_init(sst, SST_BYT_MAILBOX_OFFSET + 0x204, + SST_BYT_IPC_MAX_PAYLOAD_SIZE, + SST_BYT_MAILBOX_OFFSET, + SST_BYT_IPC_MAX_PAYLOAD_SIZE); + + sst->irq = pdata->irq; + + return 0; +} + +static int sst_byt_init(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + const struct sst_adsp_memregion *region; + struct device *dev; + int ret = -ENODEV, i, j, region_count; + u32 offset, size; + + dev = sst->dev; + + switch (sst->id) { + case SST_DEV_ID_BYT: + region = byt_region; + region_count = ARRAY_SIZE(byt_region); + sst->addr.iram_offset = SST_BYT_IRAM_OFFSET; + sst->addr.dram_offset = SST_BYT_DRAM_OFFSET; + sst->addr.shim_offset = SST_BYT_SHIM_OFFSET; + break; + default: + dev_err(dev, "failed to get mem resources\n"); + return ret; + } + + ret = sst_byt_resource_map(sst, pdata); + if (ret < 0) { + dev_err(dev, "failed to map resources\n"); + return ret; + } + + /* + * save the physical address of extended firmware block in the first + * 4 bytes of the mailbox + */ + memcpy_toio(sst->addr.lpe + SST_BYT_MAILBOX_OFFSET, + &pdata->fw_base, sizeof(u32)); + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + /* enable Interrupt from both sides */ + sst_dsp_shim_update_bits64(sst, SST_IMRX, 0x3, 0x0); + sst_dsp_shim_update_bits64(sst, SST_IMRD, 0x3, 0x0); + + /* register DSP memory blocks - ideally we should get this from ACPI */ + for (i = 0; i < region_count; i++) { + offset = region[i].start; + size = (region[i].end - region[i].start) / region[i].blocks; + + /* register individual memory blocks */ + for (j = 0; j < region[i].blocks; j++) { + sst_mem_block_register(sst, offset, size, + region[i].type, NULL, j, sst); + offset += size; + } + } + + return 0; +} + +static void sst_byt_free(struct sst_dsp *sst) +{ + sst_mem_block_unregister_all(sst); + iounmap(sst->addr.lpe); + iounmap(sst->addr.pci_cfg); + iounmap(sst->addr.fw_ext); +} + +struct sst_ops sst_byt_ops = { + .reset = sst_byt_reset, + .boot = sst_byt_boot, + .write = sst_shim32_write, + .read = sst_shim32_read, + .write64 = sst_shim32_write64, + .read64 = sst_shim32_read64, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .irq_handler = sst_byt_irq, + .init = sst_byt_init, + .free = sst_byt_free, + .parse_fw = sst_byt_parse_fw_image, +}; diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c new file mode 100644 index 000000000000..d0eaeee21be4 --- /dev/null +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -0,0 +1,867 @@ +/* + * Intel Baytrail SST IPC Support + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/list.h> +#include <linux/device.h> +#include <linux/wait.h> +#include <linux/spinlock.h> +#include <linux/workqueue.h> +#include <linux/export.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/list.h> +#include <linux/platform_device.h> +#include <linux/kthread.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <asm/div64.h> + +#include "sst-baytrail-ipc.h" +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +/* IPC message timeout */ +#define IPC_TIMEOUT_MSECS 300 +#define IPC_BOOT_MSECS 200 + +#define IPC_EMPTY_LIST_SIZE 8 + +/* IPC header bits */ +#define IPC_HEADER_MSG_ID_MASK 0xff +#define IPC_HEADER_MSG_ID(x) ((x) & IPC_HEADER_MSG_ID_MASK) +#define IPC_HEADER_STR_ID_SHIFT 8 +#define IPC_HEADER_STR_ID_MASK 0x1f +#define IPC_HEADER_STR_ID(x) (((x) & 0x1f) << IPC_HEADER_STR_ID_SHIFT) +#define IPC_HEADER_LARGE_SHIFT 13 +#define IPC_HEADER_LARGE(x) (((x) & 0x1) << IPC_HEADER_LARGE_SHIFT) +#define IPC_HEADER_DATA_SHIFT 16 +#define IPC_HEADER_DATA_MASK 0x3fff +#define IPC_HEADER_DATA(x) (((x) & 0x3fff) << IPC_HEADER_DATA_SHIFT) + +/* mask for differentiating between notification and reply message */ +#define IPC_NOTIFICATION (0x1 << 7) + +/* I2L Stream config/control msgs */ +#define IPC_IA_ALLOC_STREAM 0x20 +#define IPC_IA_FREE_STREAM 0x21 +#define IPC_IA_PAUSE_STREAM 0x24 +#define IPC_IA_RESUME_STREAM 0x25 +#define IPC_IA_DROP_STREAM 0x26 +#define IPC_IA_START_STREAM 0x30 + +/* notification messages */ +#define IPC_IA_FW_INIT_CMPLT 0x81 +#define IPC_SST_PERIOD_ELAPSED 0x97 + +/* IPC messages between host and ADSP */ +struct sst_byt_address_info { + u32 addr; + u32 size; +} __packed; + +struct sst_byt_str_type { + u8 codec_type; + u8 str_type; + u8 operation; + u8 protected_str; + u8 time_slots; + u8 reserved; + u16 result; +} __packed; + +struct sst_byt_pcm_params { + u8 num_chan; + u8 pcm_wd_sz; + u8 use_offload_path; + u8 reserved; + u32 sfreq; + u8 channel_map[8]; +} __packed; + +struct sst_byt_frames_info { + u16 num_entries; + u16 rsrvd; + u32 frag_size; + struct sst_byt_address_info ring_buf_info[8]; +} __packed; + +struct sst_byt_alloc_params { + struct sst_byt_str_type str_type; + struct sst_byt_pcm_params pcm_params; + struct sst_byt_frames_info frame_info; +} __packed; + +struct sst_byt_alloc_response { + struct sst_byt_str_type str_type; + u8 reserved[88]; +} __packed; + +struct sst_byt_start_stream_params { + u32 byte_offset; +} __packed; + +struct sst_byt_tstamp { + u64 ring_buffer_counter; + u64 hardware_counter; + u64 frames_decoded; + u64 bytes_decoded; + u64 bytes_copied; + u32 sampling_frequency; + u32 channel_peak[8]; +} __packed; + +/* driver internal IPC message structure */ +struct ipc_message { + struct list_head list; + u64 header; + + /* direction wrt host CPU */ + char tx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE]; + size_t tx_size; + char rx_data[SST_BYT_IPC_MAX_PAYLOAD_SIZE]; + size_t rx_size; + + wait_queue_head_t waitq; + bool complete; + bool wait; + int errno; +}; + +struct sst_byt_stream; +struct sst_byt; + +/* stream infomation */ +struct sst_byt_stream { + struct list_head node; + + /* configuration */ + struct sst_byt_alloc_params request; + struct sst_byt_alloc_response reply; + + /* runtime info */ + struct sst_byt *byt; + int str_id; + bool commited; + bool running; + + /* driver callback */ + u32 (*notify_position)(struct sst_byt_stream *stream, void *data); + void *pdata; +}; + +/* SST Baytrail IPC data */ +struct sst_byt { + struct device *dev; + struct sst_dsp *dsp; + + /* stream */ + struct list_head stream_list; + + /* boot */ + wait_queue_head_t boot_wait; + bool boot_complete; + + /* IPC messaging */ + struct list_head tx_list; + struct list_head rx_list; + struct list_head empty_list; + wait_queue_head_t wait_txq; + struct task_struct *tx_thread; + struct kthread_worker kworker; + struct kthread_work kwork; + struct ipc_message *msg; +}; + +static inline u64 sst_byt_header(int msg_id, int data, bool large, int str_id) +{ + u64 header; + + header = IPC_HEADER_MSG_ID(msg_id) | + IPC_HEADER_STR_ID(str_id) | + IPC_HEADER_LARGE(large) | + IPC_HEADER_DATA(data) | + SST_BYT_IPCX_BUSY; + + return header; +} + +static inline u16 sst_byt_header_msg_id(u64 header) +{ + return header & IPC_HEADER_MSG_ID_MASK; +} + +static inline u8 sst_byt_header_str_id(u64 header) +{ + return (header >> IPC_HEADER_STR_ID_SHIFT) & IPC_HEADER_STR_ID_MASK; +} + +static inline u16 sst_byt_header_data(u64 header) +{ + return (header >> IPC_HEADER_DATA_SHIFT) & IPC_HEADER_DATA_MASK; +} + +static struct sst_byt_stream *sst_byt_get_stream(struct sst_byt *byt, + int stream_id) +{ + struct sst_byt_stream *stream; + + list_for_each_entry(stream, &byt->stream_list, node) { + if (stream->str_id == stream_id) + return stream; + } + + return NULL; +} + +static void sst_byt_ipc_shim_dbg(struct sst_byt *byt, const char *text) +{ + struct sst_dsp *sst = byt->dsp; + u64 isr, ipcd, imrx, ipcx; + + ipcx = sst_dsp_shim_read64_unlocked(sst, SST_IPCX); + isr = sst_dsp_shim_read64_unlocked(sst, SST_ISRX); + ipcd = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + imrx = sst_dsp_shim_read64_unlocked(sst, SST_IMRX); + + dev_err(byt->dev, + "ipc: --%s-- ipcx 0x%llx isr 0x%llx ipcd 0x%llx imrx 0x%llx\n", + text, ipcx, isr, ipcd, imrx); +} + +/* locks held by caller */ +static struct ipc_message *sst_byt_msg_get_empty(struct sst_byt *byt) +{ + struct ipc_message *msg = NULL; + + if (!list_empty(&byt->empty_list)) { + msg = list_first_entry(&byt->empty_list, + struct ipc_message, list); + list_del(&msg->list); + } + + return msg; +} + +static void sst_byt_ipc_tx_msgs(struct kthread_work *work) +{ + struct sst_byt *byt = + container_of(work, struct sst_byt, kwork); + struct ipc_message *msg; + u64 ipcx; + unsigned long flags; + + spin_lock_irqsave(&byt->dsp->spinlock, flags); + if (list_empty(&byt->tx_list)) { + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return; + } + + /* if the DSP is busy we will TX messages after IRQ */ + ipcx = sst_dsp_shim_read64_unlocked(byt->dsp, SST_IPCX); + if (ipcx & SST_BYT_IPCX_BUSY) { + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return; + } + + msg = list_first_entry(&byt->tx_list, struct ipc_message, list); + + list_move(&msg->list, &byt->rx_list); + + /* send the message */ + if (msg->header & IPC_HEADER_LARGE(true)) + sst_dsp_outbox_write(byt->dsp, msg->tx_data, msg->tx_size); + sst_dsp_shim_write64_unlocked(byt->dsp, SST_IPCX, msg->header); + + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); +} + +static inline void sst_byt_tx_msg_reply_complete(struct sst_byt *byt, + struct ipc_message *msg) +{ + msg->complete = true; + + if (!msg->wait) + list_add_tail(&msg->list, &byt->empty_list); + else + wake_up(&msg->waitq); +} + +static int sst_byt_tx_wait_done(struct sst_byt *byt, struct ipc_message *msg, + void *rx_data) +{ + unsigned long flags; + int ret; + + /* wait for DSP completion */ + ret = wait_event_timeout(msg->waitq, msg->complete, + msecs_to_jiffies(IPC_TIMEOUT_MSECS)); + + spin_lock_irqsave(&byt->dsp->spinlock, flags); + if (ret == 0) { + list_del(&msg->list); + sst_byt_ipc_shim_dbg(byt, "message timeout"); + + ret = -ETIMEDOUT; + } else { + + /* copy the data returned from DSP */ + if (msg->rx_size) + memcpy(rx_data, msg->rx_data, msg->rx_size); + ret = msg->errno; + } + + list_add_tail(&msg->list, &byt->empty_list); + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return ret; +} + +static int sst_byt_ipc_tx_message(struct sst_byt *byt, u64 header, + void *tx_data, size_t tx_bytes, + void *rx_data, size_t rx_bytes, int wait) +{ + unsigned long flags; + struct ipc_message *msg; + + spin_lock_irqsave(&byt->dsp->spinlock, flags); + + msg = sst_byt_msg_get_empty(byt); + if (msg == NULL) { + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + return -EBUSY; + } + + msg->header = header; + msg->tx_size = tx_bytes; + msg->rx_size = rx_bytes; + msg->wait = wait; + msg->errno = 0; + msg->complete = false; + + if (tx_bytes) { + /* msg content = lower 32-bit of the header + data */ + *(u32 *)msg->tx_data = (u32)(header & (u32)-1); + memcpy(msg->tx_data + sizeof(u32), tx_data, tx_bytes); + msg->tx_size += sizeof(u32); + } + + list_add_tail(&msg->list, &byt->tx_list); + spin_unlock_irqrestore(&byt->dsp->spinlock, flags); + + queue_kthread_work(&byt->kworker, &byt->kwork); + + if (wait) + return sst_byt_tx_wait_done(byt, msg, rx_data); + else + return 0; +} + +static inline int sst_byt_ipc_tx_msg_wait(struct sst_byt *byt, u64 header, + void *tx_data, size_t tx_bytes, + void *rx_data, size_t rx_bytes) +{ + return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes, + rx_data, rx_bytes, 1); +} + +static inline int sst_byt_ipc_tx_msg_nowait(struct sst_byt *byt, u64 header, + void *tx_data, size_t tx_bytes) +{ + return sst_byt_ipc_tx_message(byt, header, tx_data, tx_bytes, + NULL, 0, 0); +} + +static struct ipc_message *sst_byt_reply_find_msg(struct sst_byt *byt, + u64 header) +{ + struct ipc_message *msg = NULL, *_msg; + u64 mask; + + /* match reply to message sent based on msg and stream IDs */ + mask = IPC_HEADER_MSG_ID_MASK | + IPC_HEADER_STR_ID_MASK << IPC_HEADER_STR_ID_SHIFT; + header &= mask; + + if (list_empty(&byt->rx_list)) { + dev_err(byt->dev, + "ipc: rx list is empty but received 0x%llx\n", header); + goto out; + } + + list_for_each_entry(_msg, &byt->rx_list, list) { + if ((_msg->header & mask) == header) { + msg = _msg; + break; + } + } + +out: + return msg; +} + +static void sst_byt_stream_update(struct sst_byt *byt, struct ipc_message *msg) +{ + struct sst_byt_stream *stream; + u64 header = msg->header; + u8 stream_id = sst_byt_header_str_id(header); + u8 stream_msg = sst_byt_header_msg_id(header); + + stream = sst_byt_get_stream(byt, stream_id); + if (stream == NULL) + return; + + switch (stream_msg) { + case IPC_IA_DROP_STREAM: + case IPC_IA_PAUSE_STREAM: + case IPC_IA_FREE_STREAM: + stream->running = false; + break; + case IPC_IA_START_STREAM: + case IPC_IA_RESUME_STREAM: + stream->running = true; + break; + } +} + +static int sst_byt_process_reply(struct sst_byt *byt, u64 header) +{ + struct ipc_message *msg; + + msg = sst_byt_reply_find_msg(byt, header); + if (msg == NULL) + return 1; + + if (header & IPC_HEADER_LARGE(true)) { + msg->rx_size = sst_byt_header_data(header); + sst_dsp_inbox_read(byt->dsp, msg->rx_data, msg->rx_size); + } + + /* update any stream states */ + sst_byt_stream_update(byt, msg); + + list_del(&msg->list); + /* wake up */ + sst_byt_tx_msg_reply_complete(byt, msg); + + return 1; +} + +static void sst_byt_fw_ready(struct sst_byt *byt, u64 header) +{ + dev_dbg(byt->dev, "ipc: DSP is ready 0x%llX\n", header); + + byt->boot_complete = true; + wake_up(&byt->boot_wait); +} + +static int sst_byt_process_notification(struct sst_byt *byt, + unsigned long *flags) +{ + struct sst_dsp *sst = byt->dsp; + struct sst_byt_stream *stream; + u64 header; + u8 msg_id, stream_id; + int handled = 1; + + header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + msg_id = sst_byt_header_msg_id(header); + + switch (msg_id) { + case IPC_SST_PERIOD_ELAPSED: + stream_id = sst_byt_header_str_id(header); + stream = sst_byt_get_stream(byt, stream_id); + if (stream && stream->running && stream->notify_position) { + spin_unlock_irqrestore(&sst->spinlock, *flags); + stream->notify_position(stream, stream->pdata); + spin_lock_irqsave(&sst->spinlock, *flags); + } + break; + case IPC_IA_FW_INIT_CMPLT: + sst_byt_fw_ready(byt, header); + break; + } + + return handled; +} + +static irqreturn_t sst_byt_irq_thread(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + struct sst_byt *byt = sst_dsp_get_thread_context(sst); + u64 header; + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + + header = sst_dsp_shim_read64_unlocked(sst, SST_IPCD); + if (header & SST_BYT_IPCD_BUSY) { + if (header & IPC_NOTIFICATION) { + /* message from ADSP */ + sst_byt_process_notification(byt, &flags); + } else { + /* reply from ADSP */ + sst_byt_process_reply(byt, header); + } + /* + * clear IPCD BUSY bit and set DONE bit. Tell DSP we have + * processed the message and can accept new. Clear data part + * of the header + */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IPCD, + SST_BYT_IPCD_DONE | SST_BYT_IPCD_BUSY | + IPC_HEADER_DATA(IPC_HEADER_DATA_MASK), + SST_BYT_IPCD_DONE); + /* unmask message request interrupts */ + sst_dsp_shim_update_bits64_unlocked(sst, SST_IMRX, + SST_BYT_IMRX_REQUEST, 0); + } + + spin_unlock_irqrestore(&sst->spinlock, flags); + + /* continue to send any remaining messages... */ + queue_kthread_work(&byt->kworker, &byt->kwork); + + return IRQ_HANDLED; +} + +/* stream API */ +struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id, + u32 (*notify_position)(struct sst_byt_stream *stream, void *data), + void *data) +{ + struct sst_byt_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (stream == NULL) + return NULL; + + list_add(&stream->node, &byt->stream_list); + stream->notify_position = notify_position; + stream->pdata = data; + stream->byt = byt; + stream->str_id = id; + + return stream; +} + +int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream, + int bits) +{ + stream->request.pcm_params.pcm_wd_sz = bits; + return 0; +} + +int sst_byt_stream_set_channels(struct sst_byt *byt, + struct sst_byt_stream *stream, u8 channels) +{ + stream->request.pcm_params.num_chan = channels; + return 0; +} + +int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream, + unsigned int rate) +{ + stream->request.pcm_params.sfreq = rate; + return 0; +} + +/* stream sonfiguration */ +int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream, + int codec_type, int stream_type, int operation) +{ + stream->request.str_type.codec_type = codec_type; + stream->request.str_type.str_type = stream_type; + stream->request.str_type.operation = operation; + stream->request.str_type.time_slots = 0xc; + + return 0; +} + +int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream, + uint32_t buffer_addr, uint32_t buffer_size) +{ + stream->request.frame_info.num_entries = 1; + stream->request.frame_info.ring_buf_info[0].addr = buffer_addr; + stream->request.frame_info.ring_buf_info[0].size = buffer_size; + /* calculate bytes per 4 ms fragment */ + stream->request.frame_info.frag_size = + stream->request.pcm_params.sfreq * + stream->request.pcm_params.num_chan * + stream->request.pcm_params.pcm_wd_sz / 8 * + 4 / 1000; + return 0; +} + +int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + struct sst_byt_alloc_params *str_req = &stream->request; + struct sst_byt_alloc_response *reply = &stream->reply; + u64 header; + int ret; + + header = sst_byt_header(IPC_IA_ALLOC_STREAM, + sizeof(*str_req) + sizeof(u32), + true, stream->str_id); + ret = sst_byt_ipc_tx_msg_wait(byt, header, str_req, sizeof(*str_req), + reply, sizeof(*reply)); + if (ret < 0) { + dev_err(byt->dev, "ipc: error stream commit failed\n"); + return ret; + } + + stream->commited = true; + + return 0; +} + +int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + u64 header; + int ret = 0; + + if (!stream->commited) + goto out; + + header = sst_byt_header(IPC_IA_FREE_STREAM, 0, false, stream->str_id); + ret = sst_byt_ipc_tx_msg_wait(byt, header, NULL, 0, NULL, 0); + if (ret < 0) { + dev_err(byt->dev, "ipc: free stream %d failed\n", + stream->str_id); + return -EAGAIN; + } + + stream->commited = false; +out: + list_del(&stream->node); + kfree(stream); + + return ret; +} + +static int sst_byt_stream_operations(struct sst_byt *byt, int type, + int stream_id, int wait) +{ + struct sst_byt_start_stream_params start_stream; + u64 header; + void *tx_msg = NULL; + size_t size = 0; + + if (type != IPC_IA_START_STREAM) { + header = sst_byt_header(type, 0, false, stream_id); + } else { + start_stream.byte_offset = 0; + header = sst_byt_header(IPC_IA_START_STREAM, + sizeof(start_stream) + sizeof(u32), + true, stream_id); + tx_msg = &start_stream; + size = sizeof(start_stream); + } + + if (wait) + return sst_byt_ipc_tx_msg_wait(byt, header, + tx_msg, size, NULL, 0); + else + return sst_byt_ipc_tx_msg_nowait(byt, header, tx_msg, size); +} + +/* stream ALSA trigger operations */ +int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + ret = sst_byt_stream_operations(byt, IPC_IA_START_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to start stream %d\n", + stream->str_id); + + return ret; +} + +int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + /* don't stop streams that are not commited */ + if (!stream->commited) + return 0; + + ret = sst_byt_stream_operations(byt, IPC_IA_DROP_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to stop stream %d\n", + stream->str_id); + return ret; +} + +int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + ret = sst_byt_stream_operations(byt, IPC_IA_PAUSE_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to pause stream %d\n", + stream->str_id); + + return ret; +} + +int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream) +{ + int ret; + + ret = sst_byt_stream_operations(byt, IPC_IA_RESUME_STREAM, + stream->str_id, 0); + if (ret < 0) + dev_err(byt->dev, "ipc: error failed to resume stream %d\n", + stream->str_id); + + return ret; +} + +int sst_byt_get_dsp_position(struct sst_byt *byt, + struct sst_byt_stream *stream, int buffer_size) +{ + struct sst_dsp *sst = byt->dsp; + struct sst_byt_tstamp fw_tstamp; + u8 str_id = stream->str_id; + u32 tstamp_offset; + + tstamp_offset = SST_BYT_TIMESTAMP_OFFSET + str_id * sizeof(fw_tstamp); + memcpy_fromio(&fw_tstamp, + sst->addr.lpe + tstamp_offset, sizeof(fw_tstamp)); + + return do_div(fw_tstamp.ring_buffer_counter, buffer_size); +} + +static int msg_empty_list_init(struct sst_byt *byt) +{ + struct ipc_message *msg; + int i; + + byt->msg = kzalloc(sizeof(*msg) * IPC_EMPTY_LIST_SIZE, GFP_KERNEL); + if (byt->msg == NULL) + return -ENOMEM; + + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + init_waitqueue_head(&byt->msg[i].waitq); + list_add(&byt->msg[i].list, &byt->empty_list); + } + + return 0; +} + +struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt) +{ + return byt->dsp; +} + +static struct sst_dsp_device byt_dev = { + .thread = sst_byt_irq_thread, + .ops = &sst_byt_ops, +}; + +int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_byt *byt; + struct sst_fw *byt_sst_fw; + int err; + + dev_dbg(dev, "initialising Byt DSP IPC\n"); + + byt = devm_kzalloc(dev, sizeof(*byt), GFP_KERNEL); + if (byt == NULL) + return -ENOMEM; + + byt->dev = dev; + INIT_LIST_HEAD(&byt->stream_list); + INIT_LIST_HEAD(&byt->tx_list); + INIT_LIST_HEAD(&byt->rx_list); + INIT_LIST_HEAD(&byt->empty_list); + init_waitqueue_head(&byt->boot_wait); + init_waitqueue_head(&byt->wait_txq); + + err = msg_empty_list_init(byt); + if (err < 0) + return -ENOMEM; + + /* start the IPC message thread */ + init_kthread_worker(&byt->kworker); + byt->tx_thread = kthread_run(kthread_worker_fn, + &byt->kworker, + dev_name(byt->dev)); + if (IS_ERR(byt->tx_thread)) { + err = PTR_ERR(byt->tx_thread); + dev_err(byt->dev, "error failed to create message TX task\n"); + goto err_free_msg; + } + init_kthread_work(&byt->kwork, sst_byt_ipc_tx_msgs); + + byt_dev.thread_context = byt; + + /* init SST shim */ + byt->dsp = sst_dsp_new(dev, &byt_dev, pdata); + if (byt->dsp == NULL) { + err = -ENODEV; + goto err_free_msg; + } + + /* keep the DSP in reset state for base FW loading */ + sst_dsp_reset(byt->dsp); + + byt_sst_fw = sst_fw_new(byt->dsp, pdata->fw, byt); + if (byt_sst_fw == NULL) { + err = -ENODEV; + dev_err(dev, "error: failed to load firmware\n"); + goto fw_err; + } + + /* wait for DSP boot completion */ + sst_dsp_boot(byt->dsp); + err = wait_event_timeout(byt->boot_wait, byt->boot_complete, + msecs_to_jiffies(IPC_BOOT_MSECS)); + if (err == 0) { + err = -EIO; + dev_err(byt->dev, "ipc: error DSP boot timeout\n"); + goto boot_err; + } + + pdata->dsp = byt; + + return 0; + +boot_err: + sst_dsp_reset(byt->dsp); + sst_fw_free(byt_sst_fw); +fw_err: + sst_dsp_free(byt->dsp); +err_free_msg: + kfree(byt->msg); + + return err; +} +EXPORT_SYMBOL_GPL(sst_byt_dsp_init); + +void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_byt *byt = pdata->dsp; + + sst_dsp_reset(byt->dsp); + sst_fw_free_all(byt->dsp); + sst_dsp_free(byt->dsp); + kfree(byt->msg); +} +EXPORT_SYMBOL_GPL(sst_byt_dsp_free); diff --git a/sound/soc/intel/sst-baytrail-ipc.h b/sound/soc/intel/sst-baytrail-ipc.h new file mode 100644 index 000000000000..f172b6440fa9 --- /dev/null +++ b/sound/soc/intel/sst-baytrail-ipc.h @@ -0,0 +1,69 @@ +/* + * Intel Baytrail SST IPC Support + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#ifndef __SST_BYT_IPC_H +#define __SST_BYT_IPC_H + +#include <linux/types.h> + +struct sst_byt; +struct sst_byt_stream; +struct sst_pdata; +extern struct sst_ops sst_byt_ops; + + +#define SST_BYT_MAILBOX_OFFSET 0x144000 +#define SST_BYT_TIMESTAMP_OFFSET (SST_BYT_MAILBOX_OFFSET + 0x800) + +/** + * Upfront defined maximum message size that is + * expected by the in/out communication pipes in FW. + */ +#define SST_BYT_IPC_MAX_PAYLOAD_SIZE 200 + +/* stream API */ +struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id, + uint32_t (*get_write_position)(struct sst_byt_stream *stream, + void *data), + void *data); + +/* stream configuration */ +int sst_byt_stream_set_bits(struct sst_byt *byt, struct sst_byt_stream *stream, + int bits); +int sst_byt_stream_set_channels(struct sst_byt *byt, + struct sst_byt_stream *stream, u8 channels); +int sst_byt_stream_set_rate(struct sst_byt *byt, struct sst_byt_stream *stream, + unsigned int rate); +int sst_byt_stream_type(struct sst_byt *byt, struct sst_byt_stream *stream, + int codec_type, int stream_type, int operation); +int sst_byt_stream_buffer(struct sst_byt *byt, struct sst_byt_stream *stream, + uint32_t buffer_addr, uint32_t buffer_size); +int sst_byt_stream_commit(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream); + +/* stream ALSA trigger operations */ +int sst_byt_stream_start(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_stop(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_pause(struct sst_byt *byt, struct sst_byt_stream *stream); +int sst_byt_stream_resume(struct sst_byt *byt, struct sst_byt_stream *stream); + +int sst_byt_get_dsp_position(struct sst_byt *byt, + struct sst_byt_stream *stream, int buffer_size); + +/* init */ +int sst_byt_dsp_init(struct device *dev, struct sst_pdata *pdata); +void sst_byt_dsp_free(struct device *dev, struct sst_pdata *pdata); +struct sst_dsp *sst_byt_get_dsp(struct sst_byt *byt); + +#endif diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c new file mode 100644 index 000000000000..6d101f3813b4 --- /dev/null +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -0,0 +1,422 @@ +/* + * Intel Baytrail SST PCM Support + * Copyright (c) 2014, Intel Corporation. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + */ + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "sst-baytrail-ipc.h" +#include "sst-dsp-priv.h" +#include "sst-dsp.h" + +#define BYT_PCM_COUNT 2 + +static const struct snd_pcm_hardware sst_byt_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FORMAT_S24_LE, + .period_bytes_min = 384, + .period_bytes_max = 48000, + .periods_min = 2, + .periods_max = 250, + .buffer_bytes_max = 96000, +}; + +/* private data for each PCM DSP stream */ +struct sst_byt_pcm_data { + struct sst_byt_stream *stream; + struct snd_pcm_substream *substream; + struct mutex mutex; +}; + +/* private data for the driver */ +struct sst_byt_priv_data { + /* runtime DSP */ + struct sst_byt *byt; + + /* DAI data */ + struct sst_byt_pcm_data pcm[BYT_PCM_COUNT]; +}; + +/* this may get called several times by oss emulation */ +static int sst_byt_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + u32 rate, bits; + u8 channels; + int ret, playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + + dev_dbg(rtd->dev, "PCM: hw_params, pcm_data %p\n", pcm_data); + + ret = sst_byt_stream_type(byt, pcm_data->stream, + 1, 1, !playback); + if (ret < 0) { + dev_err(rtd->dev, "failed to set stream format %d\n", ret); + return ret; + } + + rate = params_rate(params); + ret = sst_byt_stream_set_rate(byt, pcm_data->stream, rate); + if (ret < 0) { + dev_err(rtd->dev, "could not set rate %d\n", rate); + return ret; + } + + bits = snd_pcm_format_width(params_format(params)); + ret = sst_byt_stream_set_bits(byt, pcm_data->stream, bits); + if (ret < 0) { + dev_err(rtd->dev, "could not set formats %d\n", + params_rate(params)); + return ret; + } + + channels = (u8)(params_channels(params) & 0xF); + ret = sst_byt_stream_set_channels(byt, pcm_data->stream, channels); + if (ret < 0) { + dev_err(rtd->dev, "could not set channels %d\n", + params_rate(params)); + return ret; + } + + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + + ret = sst_byt_stream_buffer(byt, pcm_data->stream, + substream->dma_buffer.addr, + params_buffer_bytes(params)); + if (ret < 0) { + dev_err(rtd->dev, "PCM: failed to set DMA buffer %d\n", ret); + return ret; + } + + ret = sst_byt_stream_commit(byt, pcm_data->stream); + if (ret < 0) { + dev_err(rtd->dev, "PCM: failed stream commit %d\n", ret); + return ret; + } + + return 0; +} + +static int sst_byt_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->dev, "PCM: hw_free\n"); + snd_pcm_lib_free_pages(substream); + + return 0; +} + +static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + + dev_dbg(rtd->dev, "PCM: trigger %d\n", cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + sst_byt_stream_start(byt, pcm_data->stream); + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_byt_stream_resume(byt, pcm_data->stream); + break; + case SNDRV_PCM_TRIGGER_STOP: + sst_byt_stream_stop(byt, pcm_data->stream); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_byt_stream_pause(byt, pcm_data->stream); + break; + default: + break; + } + + return 0; +} + +static u32 byt_notify_pointer(struct sst_byt_stream *stream, void *data) +{ + struct sst_byt_pcm_data *pcm_data = data; + struct snd_pcm_substream *substream = pcm_data->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u32 pos; + + pos = frames_to_bytes(runtime, + (runtime->control->appl_ptr % + runtime->buffer_size)); + + dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + + snd_pcm_period_elapsed(substream); + return pos; +} + +static snd_pcm_uframes_t sst_byt_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + snd_pcm_uframes_t offset; + int pos; + + pos = sst_byt_get_dsp_position(byt, pcm_data->stream, + snd_pcm_lib_buffer_bytes(substream)); + offset = bytes_to_frames(runtime, pos); + + dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n", + frames_to_bytes(runtime, (u32)offset)); + return offset; +} + +static int sst_byt_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + + dev_dbg(rtd->dev, "PCM: open\n"); + + pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + mutex_lock(&pcm_data->mutex); + + snd_soc_pcm_set_drvdata(rtd, pcm_data); + pcm_data->substream = substream; + + snd_soc_set_runtime_hwparams(substream, &sst_byt_pcm_hardware); + + pcm_data->stream = sst_byt_stream_new(byt, rtd->cpu_dai->id + 1, + byt_notify_pointer, pcm_data); + if (pcm_data->stream == NULL) { + dev_err(rtd->dev, "failed to create stream\n"); + mutex_unlock(&pcm_data->mutex); + return -EINVAL; + } + + mutex_unlock(&pcm_data->mutex); + return 0; +} + +static int sst_byt_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sst_byt_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct sst_byt_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_byt *byt = pdata->byt; + int ret; + + dev_dbg(rtd->dev, "PCM: close\n"); + + mutex_lock(&pcm_data->mutex); + ret = sst_byt_stream_free(byt, pcm_data->stream); + if (ret < 0) { + dev_dbg(rtd->dev, "Free stream fail\n"); + goto out; + } + pcm_data->stream = NULL; + +out: + mutex_unlock(&pcm_data->mutex); + return ret; +} + +static int sst_byt_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->dev, "PCM: mmap\n"); + return snd_pcm_lib_default_mmap(substream, vma); +} + +static struct snd_pcm_ops sst_byt_pcm_ops = { + .open = sst_byt_pcm_open, + .close = sst_byt_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sst_byt_pcm_hw_params, + .hw_free = sst_byt_pcm_hw_free, + .trigger = sst_byt_pcm_trigger, + .pointer = sst_byt_pcm_pointer, + .mmap = sst_byt_pcm_mmap, +}; + +static void sst_byt_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int sst_byt_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + size_t size; + int ret = 0; + + ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + size = sst_byt_pcm_hardware.buffer_bytes_max; + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->dev, + size, size); + if (ret) { + dev_err(rtd->dev, "dma buffer allocation failed %d\n", + ret); + return ret; + } + } + + return ret; +} + +static struct snd_soc_dai_driver byt_dais[] = { + { + .name = "Front-cpu-dai", + .playback = { + .stream_name = "System Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S16_LE, + }, + }, + { + .name = "Mic1-cpu-dai", + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, +}; + +static int sst_byt_pcm_probe(struct snd_soc_platform *platform) +{ + struct sst_pdata *plat_data = dev_get_platdata(platform->dev); + struct sst_byt_priv_data *priv_data; + int i; + + if (!plat_data) + return -ENODEV; + + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), + GFP_KERNEL); + priv_data->byt = plat_data->dsp; + snd_soc_platform_set_drvdata(platform, priv_data); + + for (i = 0; i < ARRAY_SIZE(byt_dais); i++) + mutex_init(&priv_data->pcm[i].mutex); + + return 0; +} + +static int sst_byt_pcm_remove(struct snd_soc_platform *platform) +{ + return 0; +} + +static struct snd_soc_platform_driver byt_soc_platform = { + .probe = sst_byt_pcm_probe, + .remove = sst_byt_pcm_remove, + .ops = &sst_byt_pcm_ops, + .pcm_new = sst_byt_pcm_new, + .pcm_free = sst_byt_pcm_free, +}; + +static const struct snd_soc_component_driver byt_dai_component = { + .name = "byt-dai", +}; + +static int sst_byt_pcm_dev_probe(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + int ret; + + ret = sst_byt_dsp_init(&pdev->dev, sst_pdata); + if (ret < 0) + return -ENODEV; + + ret = snd_soc_register_platform(&pdev->dev, &byt_soc_platform); + if (ret < 0) + goto err_plat; + + ret = snd_soc_register_component(&pdev->dev, &byt_dai_component, + byt_dais, ARRAY_SIZE(byt_dais)); + if (ret < 0) + goto err_comp; + + return 0; + +err_comp: + snd_soc_unregister_platform(&pdev->dev); +err_plat: + sst_byt_dsp_free(&pdev->dev, sst_pdata); + return ret; +} + +static int sst_byt_pcm_dev_remove(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + sst_byt_dsp_free(&pdev->dev, sst_pdata); + + return 0; +} + +static struct platform_driver sst_byt_pcm_driver = { + .driver = { + .name = "baytrail-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = sst_byt_pcm_dev_probe, + .remove = sst_byt_pcm_dev_remove, +}; +module_platform_driver(sst_byt_pcm_driver); + +MODULE_AUTHOR("Jarkko Nikula"); +MODULE_DESCRIPTION("Baytrail PCM"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:baytrail-pcm-audio"); diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h new file mode 100644 index 000000000000..fe8e81aad646 --- /dev/null +++ b/sound/soc/intel/sst-dsp-priv.h @@ -0,0 +1,309 @@ +/* + * Intel Smart Sound Technology + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_SOC_SST_DSP_PRIV_H +#define __SOUND_SOC_SST_DSP_PRIV_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/firmware.h> + +struct sst_mem_block; +struct sst_module; +struct sst_fw; + +/* + * DSP Operations exported by platform Audio DSP driver. + */ +struct sst_ops { + /* DSP core boot / reset */ + void (*boot)(struct sst_dsp *); + void (*reset)(struct sst_dsp *); + + /* Shim IO */ + void (*write)(void __iomem *addr, u32 offset, u32 value); + u32 (*read)(void __iomem *addr, u32 offset); + void (*write64)(void __iomem *addr, u32 offset, u64 value); + u64 (*read64)(void __iomem *addr, u32 offset); + + /* DSP I/DRAM IO */ + void (*ram_read)(struct sst_dsp *sst, void *dest, void __iomem *src, + size_t bytes); + void (*ram_write)(struct sst_dsp *sst, void __iomem *dest, void *src, + size_t bytes); + + void (*dump)(struct sst_dsp *); + + /* IRQ handlers */ + irqreturn_t (*irq_handler)(int irq, void *context); + + /* SST init and free */ + int (*init)(struct sst_dsp *sst, struct sst_pdata *pdata); + void (*free)(struct sst_dsp *sst); + + /* FW module parser/loader */ + int (*parse_fw)(struct sst_fw *sst_fw); +}; + +/* + * Audio DSP memory offsets and addresses. + */ +struct sst_addr { + u32 lpe_base; + u32 shim_offset; + u32 iram_offset; + u32 dram_offset; + void __iomem *lpe; + void __iomem *shim; + void __iomem *pci_cfg; + void __iomem *fw_ext; +}; + +/* + * Audio DSP Mailbox configuration. + */ +struct sst_mailbox { + void __iomem *in_base; + void __iomem *out_base; + size_t in_size; + size_t out_size; +}; + +/* + * Audio DSP Firmware data types. + */ +enum sst_data_type { + SST_DATA_M = 0, /* module block data */ + SST_DATA_P = 1, /* peristant data (text, data) */ + SST_DATA_S = 2, /* scratch data (usually buffers) */ +}; + +/* + * Audio DSP memory block types. + */ +enum sst_mem_type { + SST_MEM_IRAM = 0, + SST_MEM_DRAM = 1, + SST_MEM_ANY = 2, + SST_MEM_CACHE= 3, +}; + +/* + * Audio DSP Generic Firmware File. + * + * SST Firmware files can consist of 1..N modules. This generic structure is + * used to manage each firmware file and it's modules regardless of SST firmware + * type. A SST driver may load multiple FW files. + */ +struct sst_fw { + struct sst_dsp *dsp; + + /* base addresses of FW file data */ + dma_addr_t dmable_fw_paddr; /* physical address of fw data */ + void *dma_buf; /* virtual address of fw data */ + u32 size; /* size of fw data */ + + /* lists */ + struct list_head list; /* DSP list of FW */ + struct list_head module_list; /* FW list of modules */ + + void *private; /* core doesn't touch this */ +}; + +/* + * Audio DSP Generic Module data. + * + * This is used to dsecribe any sections of persistent (text and data) and + * scratch (buffers) of module data in ADSP memory space. + */ +struct sst_module_data { + + enum sst_mem_type type; /* destination memory type */ + enum sst_data_type data_type; /* type of module data */ + + u32 size; /* size in bytes */ + u32 offset; /* offset in FW file */ + u32 data_offset; /* offset in ADSP memory space */ + void *data; /* module data */ +}; + +/* + * Audio DSP Generic Module Template. + * + * Used to define and register a new FW module. This data is extracted from + * FW module header information. + */ +struct sst_module_template { + u32 id; + u32 entry; /* entry point */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ +}; + +/* + * Audio DSP Generic Module. + * + * Each Firmware file can consist of 1..N modules. A module can span multiple + * ADSP memory blocks. The simplest FW will be a file with 1 module. + */ +struct sst_module { + struct sst_dsp *dsp; + struct sst_fw *sst_fw; /* parent FW we belong too */ + + /* module configuration */ + u32 id; + u32 entry; /* module entry point */ + u32 offset; /* module offset in firmware file */ + u32 size; /* module size */ + struct sst_module_data s; /* scratch data */ + struct sst_module_data p; /* peristant data */ + + /* runtime */ + u32 usage_count; /* can be unloaded if count == 0 */ + void *private; /* core doesn't touch this */ + + /* lists */ + struct list_head block_list; /* Module list of blocks in use */ + struct list_head list; /* DSP list of modules */ + struct list_head list_fw; /* FW list of modules */ +}; + +/* + * SST Memory Block operations. + */ +struct sst_block_ops { + int (*enable)(struct sst_mem_block *block); + int (*disable)(struct sst_mem_block *block); +}; + +/* + * SST Generic Memory Block. + * + * SST ADP memory has multiple IRAM and DRAM blocks. Some ADSP blocks can be + * power gated. + */ +struct sst_mem_block { + struct sst_dsp *dsp; + struct sst_module *module; /* module that uses this block */ + + /* block config */ + u32 offset; /* offset from base */ + u32 size; /* block size */ + u32 index; /* block index 0..N */ + enum sst_mem_type type; /* block memory type IRAM/DRAM */ + struct sst_block_ops *ops; /* block operations, if any */ + + /* block status */ + enum sst_data_type data_type; /* data type held in this block */ + u32 bytes_used; /* bytes in use by modules */ + void *private; /* generic core does not touch this */ + int users; /* number of modules using this block */ + + /* block lists */ + struct list_head module_list; /* Module list of blocks */ + struct list_head list; /* Map list of free/used blocks */ +}; + +/* + * Generic SST Shim Interface. + */ +struct sst_dsp { + + /* runtime */ + struct sst_dsp_device *sst_dev; + spinlock_t spinlock; /* IPC locking */ + struct mutex mutex; /* DSP FW lock */ + struct device *dev; + void *thread_context; + int irq; + u32 id; + + /* list of free and used ADSP memory blocks */ + struct list_head used_block_list; + struct list_head free_block_list; + + /* operations */ + struct sst_ops *ops; + + /* debug FS */ + struct dentry *debugfs_root; + + /* base addresses */ + struct sst_addr addr; + + /* mailbox */ + struct sst_mailbox mailbox; + + /* SST FW files loaded and their modules */ + struct list_head module_list; + struct list_head fw_list; + + /* platform data */ + struct sst_pdata *pdata; + + /* DMA FW loading */ + struct sst_dma *dma; + bool fw_use_dma; +}; + +/* Size optimised DRAM/IRAM memcpy */ +static inline void sst_dsp_write(struct sst_dsp *sst, void *src, + u32 dest_offset, size_t bytes) +{ + sst->ops->ram_write(sst, sst->addr.lpe + dest_offset, src, bytes); +} + +static inline void sst_dsp_read(struct sst_dsp *sst, void *dest, + u32 src_offset, size_t bytes) +{ + sst->ops->ram_read(sst, dest, sst->addr.lpe + src_offset, bytes); +} + +static inline void *sst_dsp_get_thread_context(struct sst_dsp *sst) +{ + return sst->thread_context; +} + +/* Create/Free FW files - can contain multiple modules */ +struct sst_fw *sst_fw_new(struct sst_dsp *dsp, + const struct firmware *fw, void *private); +void sst_fw_free(struct sst_fw *sst_fw); +void sst_fw_free_all(struct sst_dsp *dsp); + +/* Create/Free firmware modules */ +struct sst_module *sst_module_new(struct sst_fw *sst_fw, + struct sst_module_template *template, void *private); +void sst_module_free(struct sst_module *sst_module); +int sst_module_insert(struct sst_module *sst_module); +int sst_module_remove(struct sst_module *sst_module); +int sst_module_insert_fixed_block(struct sst_module *module, + struct sst_module_data *data); +struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id); + +/* allocate/free pesistent/scratch memory regions managed by drv */ +struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp); +void sst_mem_block_free_scratch(struct sst_dsp *dsp, + struct sst_module *scratch); +int sst_block_module_remove(struct sst_module *module); + +/* Register the DSPs memory blocks - would be nice to read from ACPI */ +struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, + u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, + void *private); +void sst_mem_block_unregister_all(struct sst_dsp *dsp); + +#endif diff --git a/sound/soc/intel/sst-dsp.c b/sound/soc/intel/sst-dsp.c new file mode 100644 index 000000000000..0c129fd85ecf --- /dev/null +++ b/sound/soc/intel/sst-dsp.c @@ -0,0 +1,385 @@ +/* + * Intel Smart Sound Technology (SST) DSP Core Driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/slab.h> +#include <linux/export.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +#define CREATE_TRACE_POINTS +#include <trace/events/intel-sst.h> + +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value) +{ + writel(value, addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_write); + +u32 sst_shim32_read(void __iomem *addr, u32 offset) +{ + return readl(addr + offset); +} +EXPORT_SYMBOL_GPL(sst_shim32_read); + +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value) +{ + memcpy_toio(addr + offset, &value, sizeof(value)); +} +EXPORT_SYMBOL_GPL(sst_shim32_write64); + +u64 sst_shim32_read64(void __iomem *addr, u32 offset) +{ + u64 val; + + memcpy_fromio(&val, addr + offset, sizeof(val)); + return val; +} +EXPORT_SYMBOL_GPL(sst_shim32_read64); + +static inline void _sst_memcpy_toio_32(volatile u32 __iomem *dest, + u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + writel(src[i], dest + i); +} + +static inline void _sst_memcpy_fromio_32(u32 *dest, + const volatile __iomem u32 *src, size_t bytes) +{ + int i, words = bytes >> 2; + + for (i = 0; i < words; i++) + dest[i] = readl(src + i); +} + +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes) +{ + _sst_memcpy_toio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_toio_32); + +void sst_memcpy_fromio_32(struct sst_dsp *sst, void *dest, + void __iomem *src, size_t bytes) +{ + _sst_memcpy_fromio_32(dest, src, bytes); +} +EXPORT_SYMBOL_GPL(sst_memcpy_fromio_32); + +/* Public API */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write); + +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u32 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read); + +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value) +{ + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + sst->ops->write64(sst->addr.shim, offset, value); + spin_unlock_irqrestore(&sst->spinlock, flags); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write64); + +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset) +{ + unsigned long flags; + u64 val; + + spin_lock_irqsave(&sst->spinlock, flags); + val = sst->ops->read64(sst->addr.shim, offset); + spin_unlock_irqrestore(&sst->spinlock, flags); + + return val; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read64); + +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value) +{ + sst->ops->write(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write_unlocked); + +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read(sst->addr.shim, offset); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read_unlocked); + +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value) +{ + sst->ops->write64(sst->addr.shim, offset, value); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_write64_unlocked); + +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset) +{ + return sst->ops->read64(sst->addr.shim, offset); +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_read64_unlocked); + +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + bool change; + unsigned int old, new; + u32 ret; + + ret = sst_dsp_shim_read_unlocked(sst, offset); + + old = ret; + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits_unlocked); + +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + bool change; + u64 old, new; + + old = sst_dsp_shim_read64_unlocked(sst, offset); + + new = (old & (~mask)) | (value & mask); + + change = (old != new); + if (change) + sst_dsp_shim_write64_unlocked(sst, offset, new); + + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64_unlocked); + +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sst->spinlock, flags); + change = sst_dsp_shim_update_bits_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits); + +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value) +{ + unsigned long flags; + bool change; + + spin_lock_irqsave(&sst->spinlock, flags); + change = sst_dsp_shim_update_bits64_unlocked(sst, offset, mask, value); + spin_unlock_irqrestore(&sst->spinlock, flags); + return change; +} +EXPORT_SYMBOL_GPL(sst_dsp_shim_update_bits64); + +void sst_dsp_dump(struct sst_dsp *sst) +{ + sst->ops->dump(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_dump); + +void sst_dsp_reset(struct sst_dsp *sst) +{ + sst->ops->reset(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_reset); + +int sst_dsp_boot(struct sst_dsp *sst) +{ + sst->ops->boot(sst); + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_boot); + +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg) +{ + sst_dsp_shim_write_unlocked(dsp, SST_IPCX, msg | SST_IPCX_BUSY); + trace_sst_ipc_msg_tx(msg); +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_tx); + +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp) +{ + u32 msg; + + msg = sst_dsp_shim_read_unlocked(dsp, SST_IPCX); + trace_sst_ipc_msg_rx(msg); + + return msg; +} +EXPORT_SYMBOL_GPL(sst_dsp_ipc_msg_rx); + +int sst_dsp_mailbox_init(struct sst_dsp *sst, u32 inbox_offset, size_t inbox_size, + u32 outbox_offset, size_t outbox_size) +{ + sst->mailbox.in_base = sst->addr.lpe + inbox_offset; + sst->mailbox.out_base = sst->addr.lpe + outbox_offset; + sst->mailbox.in_size = inbox_size; + sst->mailbox.out_size = outbox_size; + return 0; +} +EXPORT_SYMBOL_GPL(sst_dsp_mailbox_init); + +void sst_dsp_outbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_outbox_write(bytes); + + memcpy_toio(sst->mailbox.out_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_wdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_outbox_write); + +void sst_dsp_outbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_outbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.out_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_outbox_rdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_outbox_read); + +void sst_dsp_inbox_write(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_inbox_write(bytes); + + memcpy_toio(sst->mailbox.in_base, message, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_wdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_inbox_write); + +void sst_dsp_inbox_read(struct sst_dsp *sst, void *message, size_t bytes) +{ + u32 i; + + trace_sst_ipc_inbox_read(bytes); + + memcpy_fromio(message, sst->mailbox.in_base, bytes); + + for (i = 0; i < bytes; i += 4) + trace_sst_ipc_inbox_rdata(i, *(u32 *)(message + i)); +} +EXPORT_SYMBOL_GPL(sst_dsp_inbox_read); + +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata) +{ + struct sst_dsp *sst; + int err; + + dev_dbg(dev, "initialising audio DSP id 0x%x\n", pdata->id); + + sst = devm_kzalloc(dev, sizeof(*sst), GFP_KERNEL); + if (sst == NULL) + return NULL; + + spin_lock_init(&sst->spinlock); + mutex_init(&sst->mutex); + sst->dev = dev; + sst->thread_context = sst_dev->thread_context; + sst->sst_dev = sst_dev; + sst->id = pdata->id; + sst->irq = pdata->irq; + sst->ops = sst_dev->ops; + sst->pdata = pdata; + INIT_LIST_HEAD(&sst->used_block_list); + INIT_LIST_HEAD(&sst->free_block_list); + INIT_LIST_HEAD(&sst->module_list); + INIT_LIST_HEAD(&sst->fw_list); + + /* Initialise SST Audio DSP */ + if (sst->ops->init) { + err = sst->ops->init(sst, pdata); + if (err < 0) + return NULL; + } + + /* Register the ISR */ + err = request_threaded_irq(sst->irq, sst->ops->irq_handler, + sst_dev->thread, IRQF_SHARED, "AudioDSP", sst); + if (err) + goto irq_err; + + return sst; + +irq_err: + if (sst->ops->free) + sst->ops->free(sst); + + return NULL; +} +EXPORT_SYMBOL_GPL(sst_dsp_new); + +void sst_dsp_free(struct sst_dsp *sst) +{ + free_irq(sst->irq, sst); + if (sst->ops->free) + sst->ops->free(sst); +} +EXPORT_SYMBOL_GPL(sst_dsp_free); + +/* Module information */ +MODULE_AUTHOR("Liam Girdwood"); +MODULE_DESCRIPTION("Intel SST Core"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/intel/sst-dsp.h b/sound/soc/intel/sst-dsp.h new file mode 100644 index 000000000000..74052b59485c --- /dev/null +++ b/sound/soc/intel/sst-dsp.h @@ -0,0 +1,233 @@ +/* + * Intel Smart Sound Technology (SST) Core + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SOUND_SOC_SST_DSP_H +#define __SOUND_SOC_SST_DSP_H + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> + +/* SST Device IDs */ +#define SST_DEV_ID_LYNX_POINT 0x33C8 +#define SST_DEV_ID_WILDCAT_POINT 0x3438 +#define SST_DEV_ID_BYT 0x0F28 + +/* Supported SST DMA Devices */ +#define SST_DMA_TYPE_DW 1 +#define SST_DMA_TYPE_MID 2 + +/* SST Shim register map + * The register naming can differ between products. Some products also + * contain extra functionality. + */ +#define SST_CSR 0x00 +#define SST_PISR 0x08 +#define SST_PIMR 0x10 +#define SST_ISRX 0x18 +#define SST_ISRD 0x20 +#define SST_IMRX 0x28 +#define SST_IMRD 0x30 +#define SST_IPCX 0x38 /* IPC IA -> SST */ +#define SST_IPCD 0x40 /* IPC SST -> IA */ +#define SST_ISRSC 0x48 +#define SST_ISRLPESC 0x50 +#define SST_IMRSC 0x58 +#define SST_IMRLPESC 0x60 +#define SST_IPCSC 0x68 +#define SST_IPCLPESC 0x70 +#define SST_CLKCTL 0x78 +#define SST_CSR2 0x80 +#define SST_LTRC 0xE0 +#define SST_HDMC 0xE8 +#define SST_DBGO 0xF0 + +#define SST_SHIM_SIZE 0x100 +#define SST_PWMCTRL 0x1000 + +/* SST Shim Register bits + * The register bit naming can differ between products. Some products also + * contain extra functionality. + */ + +/* CSR / CS */ +#define SST_CSR_RST (0x1 << 1) +#define SST_CSR_SBCS0 (0x1 << 2) +#define SST_CSR_SBCS1 (0x1 << 3) +#define SST_CSR_DCS(x) (x << 4) +#define SST_CSR_DCS_MASK (0x7 << 4) +#define SST_CSR_STALL (0x1 << 10) +#define SST_CSR_S0IOCS (0x1 << 21) +#define SST_CSR_S1IOCS (0x1 << 23) +#define SST_CSR_LPCS (0x1 << 31) +#define SST_BYT_CSR_RST (0x1 << 0) +#define SST_BYT_CSR_VECTOR_SEL (0x1 << 1) +#define SST_BYT_CSR_STALL (0x1 << 2) +#define SST_BYT_CSR_PWAITMODE (0x1 << 3) + +/* ISRX / ISC */ +#define SST_ISRX_BUSY (0x1 << 1) +#define SST_ISRX_DONE (0x1 << 0) +#define SST_BYT_ISRX_REQUEST (0x1 << 1) + +/* ISRD / ISD */ +#define SST_ISRD_BUSY (0x1 << 1) +#define SST_ISRD_DONE (0x1 << 0) + +/* IMRX / IMC */ +#define SST_IMRX_BUSY (0x1 << 1) +#define SST_IMRX_DONE (0x1 << 0) +#define SST_BYT_IMRX_REQUEST (0x1 << 1) + +/* IPCX / IPCC */ +#define SST_IPCX_DONE (0x1 << 30) +#define SST_IPCX_BUSY (0x1 << 31) +#define SST_BYT_IPCX_DONE ((u64)0x1 << 62) +#define SST_BYT_IPCX_BUSY ((u64)0x1 << 63) + +/* IPCD */ +#define SST_IPCD_DONE (0x1 << 30) +#define SST_IPCD_BUSY (0x1 << 31) +#define SST_BYT_IPCD_DONE ((u64)0x1 << 62) +#define SST_BYT_IPCD_BUSY ((u64)0x1 << 63) + +/* CLKCTL */ +#define SST_CLKCTL_SMOS(x) (x << 24) +#define SST_CLKCTL_MASK (3 << 24) +#define SST_CLKCTL_DCPLCG (1 << 18) +#define SST_CLKCTL_SCOE1 (1 << 17) +#define SST_CLKCTL_SCOE0 (1 << 16) + +/* CSR2 / CS2 */ +#define SST_CSR2_SDFD_SSP0 (1 << 1) +#define SST_CSR2_SDFD_SSP1 (1 << 2) + +/* LTRC */ +#define SST_LTRC_VAL(x) (x << 0) + +/* HDMC */ +#define SST_HDMC_HDDA0(x) (x << 0) +#define SST_HDMC_HDDA1(x) (x << 7) + + +/* SST Vendor Defined Registers and bits */ +#define SST_VDRTCTL0 0xa0 +#define SST_VDRTCTL1 0xa4 +#define SST_VDRTCTL2 0xa8 +#define SST_VDRTCTL3 0xaC + +/* VDRTCTL0 */ +#define SST_VDRTCL0_DSRAMPGE_SHIFT 16 +#define SST_VDRTCL0_DSRAMPGE_MASK (0xffff << SST_VDRTCL0_DSRAMPGE_SHIFT) +#define SST_VDRTCL0_ISRAMPGE_SHIFT 6 +#define SST_VDRTCL0_ISRAMPGE_MASK (0x3ff << SST_VDRTCL0_ISRAMPGE_SHIFT) + +struct sst_dsp; + +/* + * SST Device. + * + * This structure is populated by the SST core driver. + */ +struct sst_dsp_device { + /* Mandatory fields */ + struct sst_ops *ops; + irqreturn_t (*thread)(int irq, void *context); + void *thread_context; +}; + +/* + * SST Platform Data. + */ +struct sst_pdata { + /* ACPI data */ + u32 lpe_base; + u32 lpe_size; + u32 pcicfg_base; + u32 pcicfg_size; + u32 fw_base; + u32 fw_size; + int irq; + + /* Firmware */ + const struct firmware *fw; + + /* DMA */ + u32 dma_base; + u32 dma_size; + int dma_engine; + + /* DSP */ + u32 id; + void *dsp; +}; + +/* Initialization */ +struct sst_dsp *sst_dsp_new(struct device *dev, + struct sst_dsp_device *sst_dev, struct sst_pdata *pdata); +void sst_dsp_free(struct sst_dsp *sst); + +/* SHIM Read / Write */ +void sst_dsp_shim_write(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* SHIM Read / Write Unlocked for callers already holding sst lock */ +void sst_dsp_shim_write_unlocked(struct sst_dsp *sst, u32 offset, u32 value); +u32 sst_dsp_shim_read_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits_unlocked(struct sst_dsp *sst, u32 offset, + u32 mask, u32 value); +void sst_dsp_shim_write64_unlocked(struct sst_dsp *sst, u32 offset, u64 value); +u64 sst_dsp_shim_read64_unlocked(struct sst_dsp *sst, u32 offset); +int sst_dsp_shim_update_bits64_unlocked(struct sst_dsp *sst, u32 offset, + u64 mask, u64 value); + +/* Internal generic low-level SST IO functions - can be overidden */ +void sst_shim32_write(void __iomem *addr, u32 offset, u32 value); +u32 sst_shim32_read(void __iomem *addr, u32 offset); +void sst_shim32_write64(void __iomem *addr, u32 offset, u64 value); +u64 sst_shim32_read64(void __iomem *addr, u32 offset); +void sst_memcpy_toio_32(struct sst_dsp *sst, + void __iomem *dest, void *src, size_t bytes); +void sst_memcpy_fromio_32(struct sst_dsp *sst, + void *dest, void __iomem *src, size_t bytes); + +/* DSP reset & boot */ +void sst_dsp_reset(struct sst_dsp *sst); +int sst_dsp_boot(struct sst_dsp *sst); + +/* Msg IO */ +void sst_dsp_ipc_msg_tx(struct sst_dsp *dsp, u32 msg); +u32 sst_dsp_ipc_msg_rx(struct sst_dsp *dsp); + +/* Mailbox management */ +int sst_dsp_mailbox_init(struct sst_dsp *dsp, u32 inbox_offset, + size_t inbox_size, u32 outbox_offset, size_t outbox_size); +void sst_dsp_inbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_inbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_write(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_outbox_read(struct sst_dsp *dsp, void *message, size_t bytes); +void sst_dsp_mailbox_dump(struct sst_dsp *dsp, size_t bytes); + +/* Debug */ +void sst_dsp_dump(struct sst_dsp *sst); + +#endif diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c new file mode 100644 index 000000000000..f7687107cf7f --- /dev/null +++ b/sound/soc/intel/sst-firmware.c @@ -0,0 +1,587 @@ +/* + * Intel SST Firmware Loader + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/kernel.h> +#include <linux/slab.h> +#include <linux/sched.h> +#include <linux/firmware.h> +#include <linux/export.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/dmaengine.h> +#include <linux/pci.h> + +#include <asm/page.h> +#include <asm/pgtable.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +static void sst_memcpy32(volatile void __iomem *dest, void *src, u32 bytes) +{ + u32 i; + + /* copy one 32 bit word at a time as 64 bit access is not supported */ + for (i = 0; i < bytes; i += 4) + memcpy_toio(dest + i, src + i, 4); +} + +/* create new generic firmware object */ +struct sst_fw *sst_fw_new(struct sst_dsp *dsp, + const struct firmware *fw, void *private) +{ + struct sst_fw *sst_fw; + int err; + + if (!dsp->ops->parse_fw) + return NULL; + + sst_fw = kzalloc(sizeof(*sst_fw), GFP_KERNEL); + if (sst_fw == NULL) + return NULL; + + sst_fw->dsp = dsp; + sst_fw->private = private; + sst_fw->size = fw->size; + + err = dma_coerce_mask_and_coherent(dsp->dev, DMA_BIT_MASK(32)); + if (err < 0) { + kfree(sst_fw); + return NULL; + } + + /* allocate DMA buffer to store FW data */ + sst_fw->dma_buf = dma_alloc_coherent(dsp->dev, sst_fw->size, + &sst_fw->dmable_fw_paddr, GFP_DMA | GFP_KERNEL); + if (!sst_fw->dma_buf) { + dev_err(dsp->dev, "error: DMA alloc failed\n"); + kfree(sst_fw); + return NULL; + } + + /* copy FW data to DMA-able memory */ + memcpy((void *)sst_fw->dma_buf, (void *)fw->data, fw->size); + + /* call core specific FW paser to load FW data into DSP */ + err = dsp->ops->parse_fw(sst_fw); + if (err < 0) { + dev_err(dsp->dev, "error: parse fw failed %d\n", err); + goto parse_err; + } + + mutex_lock(&dsp->mutex); + list_add(&sst_fw->list, &dsp->fw_list); + mutex_unlock(&dsp->mutex); + + return sst_fw; + +parse_err: + dma_free_coherent(dsp->dev, sst_fw->size, + sst_fw->dma_buf, + sst_fw->dmable_fw_paddr); + kfree(sst_fw); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_fw_new); + +/* free single firmware object */ +void sst_fw_free(struct sst_fw *sst_fw) +{ + struct sst_dsp *dsp = sst_fw->dsp; + + mutex_lock(&dsp->mutex); + list_del(&sst_fw->list); + mutex_unlock(&dsp->mutex); + + dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf, + sst_fw->dmable_fw_paddr); + kfree(sst_fw); +} +EXPORT_SYMBOL_GPL(sst_fw_free); + +/* free all firmware objects */ +void sst_fw_free_all(struct sst_dsp *dsp) +{ + struct sst_fw *sst_fw, *t; + + mutex_lock(&dsp->mutex); + list_for_each_entry_safe(sst_fw, t, &dsp->fw_list, list) { + + list_del(&sst_fw->list); + dma_free_coherent(dsp->dev, sst_fw->size, sst_fw->dma_buf, + sst_fw->dmable_fw_paddr); + kfree(sst_fw); + } + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_fw_free_all); + +/* create a new SST generic module from FW template */ +struct sst_module *sst_module_new(struct sst_fw *sst_fw, + struct sst_module_template *template, void *private) +{ + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_module *sst_module; + + sst_module = kzalloc(sizeof(*sst_module), GFP_KERNEL); + if (sst_module == NULL) + return NULL; + + sst_module->id = template->id; + sst_module->dsp = dsp; + sst_module->sst_fw = sst_fw; + + memcpy(&sst_module->s, &template->s, sizeof(struct sst_module_data)); + memcpy(&sst_module->p, &template->p, sizeof(struct sst_module_data)); + + INIT_LIST_HEAD(&sst_module->block_list); + + mutex_lock(&dsp->mutex); + list_add(&sst_module->list, &dsp->module_list); + mutex_unlock(&dsp->mutex); + + return sst_module; +} +EXPORT_SYMBOL_GPL(sst_module_new); + +/* free firmware module and remove from available list */ +void sst_module_free(struct sst_module *sst_module) +{ + struct sst_dsp *dsp = sst_module->dsp; + + mutex_lock(&dsp->mutex); + list_del(&sst_module->list); + mutex_unlock(&dsp->mutex); + + kfree(sst_module); +} +EXPORT_SYMBOL_GPL(sst_module_free); + +static struct sst_mem_block *find_block(struct sst_dsp *dsp, int type, + u32 offset) +{ + struct sst_mem_block *block; + + list_for_each_entry(block, &dsp->free_block_list, list) { + if (block->type == type && block->offset == offset) + return block; + } + + return NULL; +} + +static int block_alloc_contiguous(struct sst_module *module, + struct sst_module_data *data, u32 offset, int size) +{ + struct list_head tmp = LIST_HEAD_INIT(tmp); + struct sst_dsp *dsp = module->dsp; + struct sst_mem_block *block; + + while (size > 0) { + block = find_block(dsp, data->type, offset); + if (!block) { + list_splice(&tmp, &dsp->free_block_list); + return -ENOMEM; + } + + list_move_tail(&block->list, &tmp); + offset += block->size; + size -= block->size; + } + + list_splice(&tmp, &dsp->used_block_list); + return 0; +} + +/* allocate free DSP blocks for module data - callers hold locks */ +static int block_alloc(struct sst_module *module, + struct sst_module_data *data) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_mem_block *block, *tmp; + int ret = 0; + + if (data->size == 0) + return 0; + + /* find first free whole blocks that can hold module */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + + /* ignore blocks with wrong type */ + if (block->type != data->type) + continue; + + if (data->size > block->size) + continue; + + data->offset = block->offset; + block->data_type = data->data_type; + block->bytes_used = data->size % block->size; + list_add(&block->module_list, &module->block_list); + list_move(&block->list, &dsp->used_block_list); + dev_dbg(dsp->dev, " *module %d added block %d:%d\n", + module->id, block->type, block->index); + return 0; + } + + /* then find free multiple blocks that can hold module */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + + /* ignore blocks with wrong type */ + if (block->type != data->type) + continue; + + /* do we span > 1 blocks */ + if (data->size > block->size) { + ret = block_alloc_contiguous(module, data, + block->offset + block->size, + data->size - block->size); + if (ret == 0) + return ret; + } + } + + /* not enough free block space */ + return -ENOMEM; +} + +/* remove module from memory - callers hold locks */ +static void block_module_remove(struct sst_module *module) +{ + struct sst_mem_block *block, *tmp; + struct sst_dsp *dsp = module->dsp; + int err; + + /* disable each block */ + list_for_each_entry(block, &module->block_list, module_list) { + + if (block->ops && block->ops->disable) { + err = block->ops->disable(block); + if (err < 0) + dev_err(dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + } + } + + /* mark each block as free */ + list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + list_del(&block->module_list); + list_move(&block->list, &dsp->free_block_list); + } +} + +/* prepare the memory block to receive data from host - callers hold locks */ +static int block_module_prepare(struct sst_module *module) +{ + struct sst_mem_block *block; + int ret = 0; + + /* enable each block so that's it'e ready for module P/S data */ + list_for_each_entry(block, &module->block_list, module_list) { + + if (block->ops && block->ops->enable) { + ret = block->ops->enable(block); + if (ret < 0) { + dev_err(module->dsp->dev, + "error: cant disable block %d:%d\n", + block->type, block->index); + goto err; + } + } + } + return ret; + +err: + list_for_each_entry(block, &module->block_list, module_list) { + if (block->ops && block->ops->disable) + block->ops->disable(block); + } + return ret; +} + +/* allocate memory blocks for static module addresses - callers hold locks */ +static int block_alloc_fixed(struct sst_module *module, + struct sst_module_data *data) +{ + struct sst_dsp *dsp = module->dsp; + struct sst_mem_block *block, *tmp; + u32 end = data->offset + data->size, block_end; + int err; + + /* only IRAM/DRAM blocks are managed */ + if (data->type != SST_MEM_IRAM && data->type != SST_MEM_DRAM) + return 0; + + /* are blocks already attached to this module */ + list_for_each_entry_safe(block, tmp, &module->block_list, module_list) { + + /* force compacting mem blocks of the same data_type */ + if (block->data_type != data->data_type) + continue; + + block_end = block->offset + block->size; + + /* find block that holds section */ + if (data->offset >= block->offset && end < block_end) + return 0; + + /* does block span more than 1 section */ + if (data->offset >= block->offset && data->offset < block_end) { + + err = block_alloc_contiguous(module, data, + block->offset + block->size, + data->size - block->size + data->offset - block->offset); + if (err < 0) + return -ENOMEM; + + /* module already owns blocks */ + return 0; + } + } + + /* find first free blocks that can hold section in free list */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + block_end = block->offset + block->size; + + /* find block that holds section */ + if (data->offset >= block->offset && end < block_end) { + + /* add block */ + block->data_type = data->data_type; + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, &module->block_list); + return 0; + } + + /* does block span more than 1 section */ + if (data->offset >= block->offset && data->offset < block_end) { + + err = block_alloc_contiguous(module, data, + block->offset + block->size, + data->size - block->size); + if (err < 0) + return -ENOMEM; + + /* add block */ + block->data_type = data->data_type; + list_move(&block->list, &dsp->used_block_list); + list_add(&block->module_list, &module->block_list); + return 0; + } + + } + + return -ENOMEM; +} + +/* Load fixed module data into DSP memory blocks */ +int sst_module_insert_fixed_block(struct sst_module *module, + struct sst_module_data *data) +{ + struct sst_dsp *dsp = module->dsp; + int ret; + + mutex_lock(&dsp->mutex); + + /* alloc blocks that includes this section */ + ret = block_alloc_fixed(module, data); + if (ret < 0) { + dev_err(dsp->dev, + "error: no free blocks for section at offset 0x%x size 0x%x\n", + data->offset, data->size); + mutex_unlock(&dsp->mutex); + return -ENOMEM; + } + + /* prepare DSP blocks for module copy */ + ret = block_module_prepare(module); + if (ret < 0) { + dev_err(dsp->dev, "error: fw module prepare failed\n"); + goto err; + } + + /* copy partial module data to blocks */ + sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size); + + mutex_unlock(&dsp->mutex); + return ret; + +err: + block_module_remove(module); + mutex_unlock(&dsp->mutex); + return ret; +} +EXPORT_SYMBOL_GPL(sst_module_insert_fixed_block); + +/* Unload entire module from DSP memory */ +int sst_block_module_remove(struct sst_module *module) +{ + struct sst_dsp *dsp = module->dsp; + + mutex_lock(&dsp->mutex); + block_module_remove(module); + mutex_unlock(&dsp->mutex); + return 0; +} +EXPORT_SYMBOL_GPL(sst_block_module_remove); + +/* register a DSP memory block for use with FW based modules */ +struct sst_mem_block *sst_mem_block_register(struct sst_dsp *dsp, u32 offset, + u32 size, enum sst_mem_type type, struct sst_block_ops *ops, u32 index, + void *private) +{ + struct sst_mem_block *block; + + block = kzalloc(sizeof(*block), GFP_KERNEL); + if (block == NULL) + return NULL; + + block->offset = offset; + block->size = size; + block->index = index; + block->type = type; + block->dsp = dsp; + block->private = private; + block->ops = ops; + + mutex_lock(&dsp->mutex); + list_add(&block->list, &dsp->free_block_list); + mutex_unlock(&dsp->mutex); + + return block; +} +EXPORT_SYMBOL_GPL(sst_mem_block_register); + +/* unregister all DSP memory blocks */ +void sst_mem_block_unregister_all(struct sst_dsp *dsp) +{ + struct sst_mem_block *block, *tmp; + + mutex_lock(&dsp->mutex); + + /* unregister used blocks */ + list_for_each_entry_safe(block, tmp, &dsp->used_block_list, list) { + list_del(&block->list); + kfree(block); + } + + /* unregister free blocks */ + list_for_each_entry_safe(block, tmp, &dsp->free_block_list, list) { + list_del(&block->list); + kfree(block); + } + + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_mem_block_unregister_all); + +/* allocate scratch buffer blocks */ +struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp) +{ + struct sst_module *sst_module, *scratch; + struct sst_mem_block *block, *tmp; + u32 block_size; + int ret = 0; + + scratch = kzalloc(sizeof(struct sst_module), GFP_KERNEL); + if (scratch == NULL) + return NULL; + + mutex_lock(&dsp->mutex); + + /* calculate required scratch size */ + list_for_each_entry(sst_module, &dsp->module_list, list) { + if (scratch->s.size > sst_module->s.size) + scratch->s.size = scratch->s.size; + else + scratch->s.size = sst_module->s.size; + } + + dev_dbg(dsp->dev, "scratch buffer required is %d bytes\n", + scratch->s.size); + + /* init scratch module */ + scratch->dsp = dsp; + scratch->s.type = SST_MEM_DRAM; + scratch->s.data_type = SST_DATA_S; + INIT_LIST_HEAD(&scratch->block_list); + + /* check free blocks before looking at used blocks for space */ + if (!list_empty(&dsp->free_block_list)) + block = list_first_entry(&dsp->free_block_list, + struct sst_mem_block, list); + else + block = list_first_entry(&dsp->used_block_list, + struct sst_mem_block, list); + block_size = block->size; + + /* allocate blocks for module scratch buffers */ + dev_dbg(dsp->dev, "allocating scratch blocks\n"); + ret = block_alloc(scratch, &scratch->s); + if (ret < 0) { + dev_err(dsp->dev, "error: can't alloc scratch blocks\n"); + goto err; + } + + /* assign the same offset of scratch to each module */ + list_for_each_entry(sst_module, &dsp->module_list, list) + sst_module->s.offset = scratch->s.offset; + + mutex_unlock(&dsp->mutex); + return scratch; + +err: + list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list) + list_del(&block->module_list); + mutex_unlock(&dsp->mutex); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_mem_block_alloc_scratch); + +/* free all scratch blocks */ +void sst_mem_block_free_scratch(struct sst_dsp *dsp, + struct sst_module *scratch) +{ + struct sst_mem_block *block, *tmp; + + mutex_lock(&dsp->mutex); + + list_for_each_entry_safe(block, tmp, &scratch->block_list, module_list) + list_del(&block->module_list); + + mutex_unlock(&dsp->mutex); +} +EXPORT_SYMBOL_GPL(sst_mem_block_free_scratch); + +/* get a module from it's unique ID */ +struct sst_module *sst_module_get_from_id(struct sst_dsp *dsp, u32 id) +{ + struct sst_module *module; + + mutex_lock(&dsp->mutex); + + list_for_each_entry(module, &dsp->module_list, list) { + if (module->id == id) { + mutex_unlock(&dsp->mutex); + return module; + } + } + + mutex_unlock(&dsp->mutex); + return NULL; +} +EXPORT_SYMBOL_GPL(sst_module_get_from_id); diff --git a/sound/soc/intel/sst-haswell-dsp.c b/sound/soc/intel/sst-haswell-dsp.c new file mode 100644 index 000000000000..f5ebf36af889 --- /dev/null +++ b/sound/soc/intel/sst-haswell-dsp.c @@ -0,0 +1,517 @@ +/* + * Intel Haswell SST DSP driver + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/delay.h> +#include <linux/fs.h> +#include <linux/slab.h> +#include <linux/device.h> +#include <linux/sched.h> +#include <linux/export.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/platform_device.h> +#include <linux/pci.h> +#include <linux/firmware.h> +#include <linux/pm_runtime.h> + +#include <linux/acpi.h> +#include <acpi/acpi_bus.h> + +#include "sst-dsp.h" +#include "sst-dsp-priv.h" +#include "sst-haswell-ipc.h" + +#include <trace/events/hswadsp.h> + +#define SST_HSW_FW_SIGNATURE_SIZE 4 +#define SST_HSW_FW_SIGN "$SST" +#define SST_HSW_FW_LIB_SIGN "$LIB" + +#define SST_WPT_SHIM_OFFSET 0xFB000 +#define SST_LP_SHIM_OFFSET 0xE7000 +#define SST_WPT_IRAM_OFFSET 0xA0000 +#define SST_LP_IRAM_OFFSET 0x80000 + +#define SST_SHIM_PM_REG 0x84 + +#define SST_HSW_IRAM 1 +#define SST_HSW_DRAM 2 +#define SST_HSW_REGS 3 + +struct dma_block_info { + __le32 type; /* IRAM/DRAM */ + __le32 size; /* Bytes */ + __le32 ram_offset; /* Offset in I/DRAM */ + __le32 rsvd; /* Reserved field */ +} __attribute__((packed)); + +struct fw_module_info { + __le32 persistent_size; + __le32 scratch_size; +} __attribute__((packed)); + +struct fw_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* FW signature */ + __le32 file_size; /* size of fw minus this header */ + __le32 modules; /* # of modules */ + __le32 file_format; /* version of header format */ + __le32 reserved[4]; +} __attribute__((packed)); + +struct fw_module_header { + unsigned char signature[SST_HSW_FW_SIGNATURE_SIZE]; /* module signature */ + __le32 mod_size; /* size of module */ + __le32 blocks; /* # of blocks */ + __le16 padding; + __le16 type; /* codec type, pp lib */ + __le32 entry_point; + struct fw_module_info info; +} __attribute__((packed)); + +static void hsw_free(struct sst_dsp *sst); + +static int hsw_parse_module(struct sst_dsp *dsp, struct sst_fw *fw, + struct fw_module_header *module) +{ + struct dma_block_info *block; + struct sst_module *mod; + struct sst_module_data block_data; + struct sst_module_template template; + int count; + void __iomem *ram; + + /* TODO: allowed module types need to be configurable */ + if (module->type != SST_HSW_MODULE_BASE_FW + && module->type != SST_HSW_MODULE_PCM_SYSTEM + && module->type != SST_HSW_MODULE_PCM + && module->type != SST_HSW_MODULE_PCM_REFERENCE + && module->type != SST_HSW_MODULE_PCM_CAPTURE + && module->type != SST_HSW_MODULE_LPAL) + return 0; + + dev_dbg(dsp->dev, "new module sign 0x%s size 0x%x blocks 0x%x type 0x%x\n", + module->signature, module->mod_size, + module->blocks, module->type); + dev_dbg(dsp->dev, " entrypoint 0x%x\n", module->entry_point); + dev_dbg(dsp->dev, " persistent 0x%x scratch 0x%x\n", + module->info.persistent_size, module->info.scratch_size); + + memset(&template, 0, sizeof(template)); + template.id = module->type; + template.entry = module->entry_point; + template.p.size = module->info.persistent_size; + template.p.type = SST_MEM_DRAM; + template.p.data_type = SST_DATA_P; + template.s.size = module->info.scratch_size; + template.s.type = SST_MEM_DRAM; + template.s.data_type = SST_DATA_S; + + mod = sst_module_new(fw, &template, NULL); + if (mod == NULL) + return -ENOMEM; + + block = (void *)module + sizeof(*module); + + for (count = 0; count < module->blocks; count++) { + + if (block->size <= 0) { + dev_err(dsp->dev, + "error: block %d size invalid\n", count); + sst_module_free(mod); + return -EINVAL; + } + + switch (block->type) { + case SST_HSW_IRAM: + ram = dsp->addr.lpe; + block_data.offset = + block->ram_offset + dsp->addr.iram_offset; + block_data.type = SST_MEM_IRAM; + break; + case SST_HSW_DRAM: + ram = dsp->addr.lpe; + block_data.offset = block->ram_offset; + block_data.type = SST_MEM_DRAM; + break; + default: + dev_err(dsp->dev, "error: bad type 0x%x for block 0x%x\n", + block->type, count); + sst_module_free(mod); + return -EINVAL; + } + + block_data.size = block->size; + block_data.data_type = SST_DATA_M; + block_data.data = (void *)block + sizeof(*block); + block_data.data_offset = block_data.data - fw->dma_buf; + + dev_dbg(dsp->dev, "copy firmware block %d type 0x%x " + "size 0x%x ==> ram %p offset 0x%x\n", + count, block->type, block->size, ram, + block->ram_offset); + + sst_module_insert_fixed_block(mod, &block_data); + + block = (void *)block + sizeof(*block) + block->size; + } + return 0; +} + +static int hsw_parse_fw_image(struct sst_fw *sst_fw) +{ + struct fw_header *header; + struct sst_module *scratch; + struct fw_module_header *module; + struct sst_dsp *dsp = sst_fw->dsp; + struct sst_hsw *hsw = sst_fw->private; + int ret, count; + + /* Read the header information from the data pointer */ + header = (struct fw_header *)sst_fw->dma_buf; + + /* verify FW */ + if ((strncmp(header->signature, SST_HSW_FW_SIGN, 4) != 0) || + (sst_fw->size != header->file_size + sizeof(*header))) { + dev_err(dsp->dev, "error: invalid fw sign/filesize mismatch\n"); + return -EINVAL; + } + + dev_dbg(dsp->dev, "header size=0x%x modules=0x%x fmt=0x%x size=%zu\n", + header->file_size, header->modules, + header->file_format, sizeof(*header)); + + /* parse each module */ + module = (void *)sst_fw->dma_buf + sizeof(*header); + for (count = 0; count < header->modules; count++) { + + /* module */ + ret = hsw_parse_module(dsp, sst_fw, module); + if (ret < 0) { + dev_err(dsp->dev, "error: invalid module %d\n", count); + return ret; + } + module = (void *)module + sizeof(*module) + module->mod_size; + } + + /* allocate persistent/scratch mem regions */ + scratch = sst_mem_block_alloc_scratch(dsp); + if (scratch == NULL) + return -ENOMEM; + + sst_hsw_set_scratch_module(hsw, scratch); + + return 0; +} + +static irqreturn_t hsw_irq(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + u32 isr; + int ret = IRQ_NONE; + + spin_lock(&sst->spinlock); + + /* Interrupt arrived, check src */ + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + if (isr & SST_ISRX_DONE) { + trace_sst_irq_done(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Done interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, SST_IMRX_DONE); + ret = IRQ_WAKE_THREAD; + } + + if (isr & SST_ISRX_BUSY) { + trace_sst_irq_busy(isr, + sst_dsp_shim_read_unlocked(sst, SST_IMRX)); + + /* Mask Busy interrupt before return */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, SST_IMRX_BUSY); + ret = IRQ_WAKE_THREAD; + } + + spin_unlock(&sst->spinlock); + return ret; +} + +static void hsw_boot(struct sst_dsp *sst) +{ + /* select SSP1 19.2MHz base clock, SSP clock 0, turn off Low Power Clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_S1IOCS | SST_CSR_SBCS1 | SST_CSR_LPCS, 0x0); + + /* stall DSP core, set clk to 192/96Mhz */ + sst_dsp_shim_update_bits_unlocked(sst, + SST_CSR, SST_CSR_STALL | SST_CSR_DCS_MASK, + SST_CSR_STALL | SST_CSR_DCS(4)); + + /* Set 24MHz MCLK, prevent local clock gating, enable SSP0 clock */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CLKCTL, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0, + SST_CLKCTL_MASK | SST_CLKCTL_DCPLCG | SST_CLKCTL_SCOE0); + + /* disable DMA finish function for SSP0 & SSP1 */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR2, SST_CSR2_SDFD_SSP1, + SST_CSR2_SDFD_SSP1); + + /* enable DMA engine 0,1 all channels to access host memory */ + sst_dsp_shim_update_bits_unlocked(sst, SST_HDMC, + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff), + SST_HDMC_HDDA1(0xff) | SST_HDMC_HDDA0(0xff)); + + /* disable all clock gating */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL2); + + /* set DSP to RUN */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, SST_CSR_STALL, 0x0); +} + +static void hsw_reset(struct sst_dsp *sst) +{ + /* put DSP into reset and stall */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_RST | SST_CSR_STALL); + + /* keep in reset for 10ms */ + mdelay(10); + + /* take DSP out of reset and keep stalled for FW loading */ + sst_dsp_shim_update_bits_unlocked(sst, SST_CSR, + SST_CSR_RST | SST_CSR_STALL, SST_CSR_STALL); +} + +struct sst_adsp_memregion { + u32 start; + u32 end; + int blocks; + enum sst_mem_type type; +}; + +/* lynx point ADSP mem regions */ +static const struct sst_adsp_memregion lp_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xE0000, 12, SST_MEM_IRAM}, /* I-SRAM - 12 * 32kB */ +}; + +/* wild cat point ADSP mem regions */ +static const struct sst_adsp_memregion wpt_region[] = { + {0x00000, 0x40000, 8, SST_MEM_DRAM}, /* D-SRAM0 - 8 * 32kB */ + {0x40000, 0x80000, 8, SST_MEM_DRAM}, /* D-SRAM1 - 8 * 32kB */ + {0x80000, 0xA0000, 4, SST_MEM_DRAM}, /* D-SRAM2 - 4 * 32kB */ + {0xA0000, 0xF0000, 10, SST_MEM_IRAM}, /* I-SRAM - 10 * 32kB */ +}; + +static int hsw_acpi_resource_map(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + /* ADSP DRAM & IRAM */ + sst->addr.lpe_base = pdata->lpe_base; + sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); + if (!sst->addr.lpe) + return -ENODEV; + + /* ADSP PCI MMIO config space */ + sst->addr.pci_cfg = ioremap(pdata->pcicfg_base, pdata->pcicfg_size); + if (!sst->addr.pci_cfg) { + iounmap(sst->addr.lpe); + return -ENODEV; + } + + /* SST Shim */ + sst->addr.shim = sst->addr.lpe + sst->addr.shim_offset; + return 0; +} + +static u32 hsw_block_get_bit(struct sst_mem_block *block) +{ + u32 bit = 0, shift = 0; + + switch (block->type) { + case SST_MEM_DRAM: + shift = 16; + break; + case SST_MEM_IRAM: + shift = 6; + break; + default: + return 0; + } + + bit = 1 << (block->index + shift); + + return bit; +} + +/* enable 32kB memory block - locks held by caller */ +static int hsw_block_enable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (block->users++ > 0) + return 0; + + dev_dbg(block->dsp->dev, " enabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val & ~bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + /* wait 18 DSP clock ticks */ + udelay(10); + + return 0; +} + +/* disable 32kB memory block - locks held by caller */ +static int hsw_block_disable(struct sst_mem_block *block) +{ + struct sst_dsp *sst = block->dsp; + u32 bit, val; + + if (--block->users > 0) + return 0; + + dev_dbg(block->dsp->dev, " disabled block %d:%d at offset 0x%x\n", + block->type, block->index, block->offset); + + val = readl(sst->addr.pci_cfg + SST_VDRTCTL0); + bit = hsw_block_get_bit(block); + writel(val | bit, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static struct sst_block_ops sst_hsw_ops = { + .enable = hsw_block_enable, + .disable = hsw_block_disable, +}; + +static int hsw_enable_shim(struct sst_dsp *sst) +{ + int tries = 10; + u32 reg; + + /* enable shim */ + reg = readl(sst->addr.pci_cfg + SST_SHIM_PM_REG); + writel(reg & ~0x3, sst->addr.pci_cfg + SST_SHIM_PM_REG); + + /* check that ADSP shim is enabled */ + while (tries--) { + reg = sst_dsp_shim_read_unlocked(sst, SST_CSR); + if (reg != 0xffffffff) + return 0; + + msleep(1); + } + + return -ENODEV; +} + +static int hsw_init(struct sst_dsp *sst, struct sst_pdata *pdata) +{ + const struct sst_adsp_memregion *region; + struct device *dev; + int ret = -ENODEV, i, j, region_count; + u32 offset, size; + + dev = sst->dev; + + switch (sst->id) { + case SST_DEV_ID_LYNX_POINT: + region = lp_region; + region_count = ARRAY_SIZE(lp_region); + sst->addr.iram_offset = SST_LP_IRAM_OFFSET; + sst->addr.shim_offset = SST_LP_SHIM_OFFSET; + break; + case SST_DEV_ID_WILDCAT_POINT: + region = wpt_region; + region_count = ARRAY_SIZE(wpt_region); + sst->addr.iram_offset = SST_WPT_IRAM_OFFSET; + sst->addr.shim_offset = SST_WPT_SHIM_OFFSET; + break; + default: + dev_err(dev, "error: failed to get mem resources\n"); + return ret; + } + + ret = hsw_acpi_resource_map(sst, pdata); + if (ret < 0) { + dev_err(dev, "error: failed to map resources\n"); + return ret; + } + + /* enable the DSP SHIM */ + ret = hsw_enable_shim(sst); + if (ret < 0) { + dev_err(dev, "error: failed to set DSP D0 and reset SHIM\n"); + return ret; + } + + ret = dma_coerce_mask_and_coherent(dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + /* Enable Interrupt from both sides */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, 0x3, 0x0); + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRD, + (0x3 | 0x1 << 16 | 0x3 << 21), 0x0); + + /* register DSP memory blocks - ideally we should get this from ACPI */ + for (i = 0; i < region_count; i++) { + offset = region[i].start; + size = (region[i].end - region[i].start) / region[i].blocks; + + /* register individual memory blocks */ + for (j = 0; j < region[i].blocks; j++) { + sst_mem_block_register(sst, offset, size, + region[i].type, &sst_hsw_ops, j, sst); + offset += size; + } + } + + /* set default power gating mask */ + writel(0x0, sst->addr.pci_cfg + SST_VDRTCTL0); + + return 0; +} + +static void hsw_free(struct sst_dsp *sst) +{ + sst_mem_block_unregister_all(sst); + iounmap(sst->addr.lpe); + iounmap(sst->addr.pci_cfg); +} + +struct sst_ops haswell_ops = { + .reset = hsw_reset, + .boot = hsw_boot, + .write = sst_shim32_write, + .read = sst_shim32_read, + .write64 = sst_shim32_write64, + .read64 = sst_shim32_read64, + .ram_read = sst_memcpy_fromio_32, + .ram_write = sst_memcpy_toio_32, + .irq_handler = hsw_irq, + .init = hsw_init, + .free = hsw_free, + .parse_fw = hsw_parse_fw_image, +}; diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c new file mode 100644 index 000000000000..f46bb4ddde6f --- /dev/null +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -0,0 +1,1785 @@ +/* + * Intel SST Haswell/Broadwell IPC Support + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/list.h> +#include <linux/device.h> +#include <linux/wait.h> +#include <linux/spinlock.h> +#include <linux/workqueue.h> +#include <linux/export.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/sched.h> +#include <linux/list.h> +#include <linux/platform_device.h> +#include <linux/kthread.h> +#include <linux/firmware.h> +#include <linux/dma-mapping.h> +#include <linux/debugfs.h> + +#include "sst-haswell-ipc.h" +#include "sst-dsp.h" +#include "sst-dsp-priv.h" + +/* Global Message - Generic */ +#define IPC_GLB_TYPE_SHIFT 24 +#define IPC_GLB_TYPE_MASK (0x1f << IPC_GLB_TYPE_SHIFT) +#define IPC_GLB_TYPE(x) (x << IPC_GLB_TYPE_SHIFT) + +/* Global Message - Reply */ +#define IPC_GLB_REPLY_SHIFT 0 +#define IPC_GLB_REPLY_MASK (0x1f << IPC_GLB_REPLY_SHIFT) +#define IPC_GLB_REPLY_TYPE(x) (x << IPC_GLB_REPLY_TYPE_SHIFT) + +/* Stream Message - Generic */ +#define IPC_STR_TYPE_SHIFT 20 +#define IPC_STR_TYPE_MASK (0xf << IPC_STR_TYPE_SHIFT) +#define IPC_STR_TYPE(x) (x << IPC_STR_TYPE_SHIFT) +#define IPC_STR_ID_SHIFT 16 +#define IPC_STR_ID_MASK (0xf << IPC_STR_ID_SHIFT) +#define IPC_STR_ID(x) (x << IPC_STR_ID_SHIFT) + +/* Stream Message - Reply */ +#define IPC_STR_REPLY_SHIFT 0 +#define IPC_STR_REPLY_MASK (0x1f << IPC_STR_REPLY_SHIFT) + +/* Stream Stage Message - Generic */ +#define IPC_STG_TYPE_SHIFT 12 +#define IPC_STG_TYPE_MASK (0xf << IPC_STG_TYPE_SHIFT) +#define IPC_STG_TYPE(x) (x << IPC_STG_TYPE_SHIFT) +#define IPC_STG_ID_SHIFT 10 +#define IPC_STG_ID_MASK (0x3 << IPC_STG_ID_SHIFT) +#define IPC_STG_ID(x) (x << IPC_STG_ID_SHIFT) + +/* Stream Stage Message - Reply */ +#define IPC_STG_REPLY_SHIFT 0 +#define IPC_STG_REPLY_MASK (0x1f << IPC_STG_REPLY_SHIFT) + +/* Debug Log Message - Generic */ +#define IPC_LOG_OP_SHIFT 20 +#define IPC_LOG_OP_MASK (0xf << IPC_LOG_OP_SHIFT) +#define IPC_LOG_OP_TYPE(x) (x << IPC_LOG_OP_SHIFT) +#define IPC_LOG_ID_SHIFT 16 +#define IPC_LOG_ID_MASK (0xf << IPC_LOG_ID_SHIFT) +#define IPC_LOG_ID(x) (x << IPC_LOG_ID_SHIFT) + +/* IPC message timeout (msecs) */ +#define IPC_TIMEOUT_MSECS 300 +#define IPC_BOOT_MSECS 200 +#define IPC_MSG_WAIT 0 +#define IPC_MSG_NOWAIT 1 + +/* Firmware Ready Message */ +#define IPC_FW_READY (0x1 << 29) +#define IPC_STATUS_MASK (0x3 << 30) + +#define IPC_EMPTY_LIST_SIZE 8 +#define IPC_MAX_STREAMS 4 + +/* Mailbox */ +#define IPC_MAX_MAILBOX_BYTES 256 + +/* Global Message - Types and Replies */ +enum ipc_glb_type { + IPC_GLB_GET_FW_VERSION = 0, /* Retrieves firmware version */ + IPC_GLB_PERFORMANCE_MONITOR = 1, /* Performance monitoring actions */ + IPC_GLB_ALLOCATE_STREAM = 3, /* Request to allocate new stream */ + IPC_GLB_FREE_STREAM = 4, /* Request to free stream */ + IPC_GLB_GET_FW_CAPABILITIES = 5, /* Retrieves firmware capabilities */ + IPC_GLB_STREAM_MESSAGE = 6, /* Message directed to stream or its stages */ + /* Request to store firmware context during D0->D3 transition */ + IPC_GLB_REQUEST_DUMP = 7, + /* Request to restore firmware context during D3->D0 transition */ + IPC_GLB_RESTORE_CONTEXT = 8, + IPC_GLB_GET_DEVICE_FORMATS = 9, /* Set device format */ + IPC_GLB_SET_DEVICE_FORMATS = 10, /* Get device format */ + IPC_GLB_SHORT_REPLY = 11, + IPC_GLB_ENTER_DX_STATE = 12, + IPC_GLB_GET_MIXER_STREAM_INFO = 13, /* Request mixer stream params */ + IPC_GLB_DEBUG_LOG_MESSAGE = 14, /* Message to or from the debug logger. */ + IPC_GLB_REQUEST_TRANSFER = 16, /* < Request Transfer for host */ + IPC_GLB_MAX_IPC_MESSAGE_TYPE = 17, /* Maximum message number */ +}; + +enum ipc_glb_reply { + IPC_GLB_REPLY_SUCCESS = 0, /* The operation was successful. */ + IPC_GLB_REPLY_ERROR_INVALID_PARAM = 1, /* Invalid parameter was passed. */ + IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE = 2, /* Uknown message type was resceived. */ + IPC_GLB_REPLY_OUT_OF_RESOURCES = 3, /* No resources to satisfy the request. */ + IPC_GLB_REPLY_BUSY = 4, /* The system or resource is busy. */ + IPC_GLB_REPLY_PENDING = 5, /* The action was scheduled for processing. */ + IPC_GLB_REPLY_FAILURE = 6, /* Critical error happened. */ + IPC_GLB_REPLY_INVALID_REQUEST = 7, /* Request can not be completed. */ + IPC_GLB_REPLY_STAGE_UNINITIALIZED = 8, /* Processing stage was uninitialized. */ + IPC_GLB_REPLY_NOT_FOUND = 9, /* Required resource can not be found. */ + IPC_GLB_REPLY_SOURCE_NOT_STARTED = 10, /* Source was not started. */ +}; + +/* Stream Message - Types */ +enum ipc_str_operation { + IPC_STR_RESET = 0, + IPC_STR_PAUSE = 1, + IPC_STR_RESUME = 2, + IPC_STR_STAGE_MESSAGE = 3, + IPC_STR_NOTIFICATION = 4, + IPC_STR_MAX_MESSAGE +}; + +/* Stream Stage Message Types */ +enum ipc_stg_operation { + IPC_STG_GET_VOLUME = 0, + IPC_STG_SET_VOLUME, + IPC_STG_SET_WRITE_POSITION, + IPC_STG_SET_FX_ENABLE, + IPC_STG_SET_FX_DISABLE, + IPC_STG_SET_FX_GET_PARAM, + IPC_STG_SET_FX_SET_PARAM, + IPC_STG_SET_FX_GET_INFO, + IPC_STG_MUTE_LOOPBACK, + IPC_STG_MAX_MESSAGE +}; + +/* Stream Stage Message Types For Notification*/ +enum ipc_stg_operation_notify { + IPC_POSITION_CHANGED = 0, + IPC_STG_GLITCH, + IPC_STG_MAX_NOTIFY +}; + +enum ipc_glitch_type { + IPC_GLITCH_UNDERRUN = 1, + IPC_GLITCH_DECODER_ERROR, + IPC_GLITCH_DOUBLED_WRITE_POS, + IPC_GLITCH_MAX +}; + +/* Debug Control */ +enum ipc_debug_operation { + IPC_DEBUG_ENABLE_LOG = 0, + IPC_DEBUG_DISABLE_LOG = 1, + IPC_DEBUG_REQUEST_LOG_DUMP = 2, + IPC_DEBUG_NOTIFY_LOG_DUMP = 3, + IPC_DEBUG_MAX_DEBUG_LOG +}; + +/* Firmware Ready */ +struct sst_hsw_ipc_fw_ready { + u32 inbox_offset; + u32 outbox_offset; + u32 inbox_size; + u32 outbox_size; + u32 fw_info_size; + u8 fw_info[1]; +} __attribute__((packed)); + +struct ipc_message { + struct list_head list; + u32 header; + + /* direction wrt host CPU */ + char tx_data[IPC_MAX_MAILBOX_BYTES]; + size_t tx_size; + char rx_data[IPC_MAX_MAILBOX_BYTES]; + size_t rx_size; + + wait_queue_head_t waitq; + bool pending; + bool complete; + bool wait; + int errno; +}; + +struct sst_hsw_stream; +struct sst_hsw; + +/* Stream infomation */ +struct sst_hsw_stream { + /* configuration */ + struct sst_hsw_ipc_stream_alloc_req request; + struct sst_hsw_ipc_stream_alloc_reply reply; + struct sst_hsw_ipc_stream_free_req free_req; + + /* Mixer info */ + u32 mute_volume[SST_HSW_NO_CHANNELS]; + u32 mute[SST_HSW_NO_CHANNELS]; + + /* runtime info */ + struct sst_hsw *hsw; + int host_id; + bool commited; + bool running; + + /* Notification work */ + struct work_struct notify_work; + u32 header; + + /* Position info from DSP */ + struct sst_hsw_ipc_stream_set_position wpos; + struct sst_hsw_ipc_stream_get_position rpos; + struct sst_hsw_ipc_stream_glitch_position glitch; + + /* Volume info */ + struct sst_hsw_ipc_volume_req vol_req; + + /* driver callback */ + u32 (*notify_position)(struct sst_hsw_stream *stream, void *data); + void *pdata; + + struct list_head node; +}; + +/* FW log ring information */ +struct sst_hsw_log_stream { + dma_addr_t dma_addr; + unsigned char *dma_area; + unsigned char *ring_descr; + int pages; + int size; + + /* Notification work */ + struct work_struct notify_work; + wait_queue_head_t readers_wait_q; + struct mutex rw_mutex; + + u32 last_pos; + u32 curr_pos; + u32 reader_pos; + + /* fw log config */ + u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS]; + + struct sst_hsw *hsw; +}; + +/* SST Haswell IPC data */ +struct sst_hsw { + struct device *dev; + struct sst_dsp *dsp; + struct platform_device *pdev_pcm; + + /* FW config */ + struct sst_hsw_ipc_fw_ready fw_ready; + struct sst_hsw_ipc_fw_version version; + struct sst_module *scratch; + bool fw_done; + + /* stream */ + struct list_head stream_list; + + /* global mixer */ + struct sst_hsw_ipc_stream_info_reply mixer_info; + enum sst_hsw_volume_curve curve_type; + u32 curve_duration; + u32 mute[SST_HSW_NO_CHANNELS]; + u32 mute_volume[SST_HSW_NO_CHANNELS]; + + /* DX */ + struct sst_hsw_ipc_dx_reply dx; + + /* boot */ + wait_queue_head_t boot_wait; + bool boot_complete; + bool shutdown; + + /* IPC messaging */ + struct list_head tx_list; + struct list_head rx_list; + struct list_head empty_list; + wait_queue_head_t wait_txq; + struct task_struct *tx_thread; + struct kthread_worker kworker; + struct kthread_work kwork; + bool pending; + struct ipc_message *msg; + + /* FW log stream */ + struct sst_hsw_log_stream log_stream; +}; + +#define CREATE_TRACE_POINTS +#include <trace/events/hswadsp.h> + +static inline u32 msg_get_global_type(u32 msg) +{ + return (msg & IPC_GLB_TYPE_MASK) >> IPC_GLB_TYPE_SHIFT; +} + +static inline u32 msg_get_global_reply(u32 msg) +{ + return (msg & IPC_GLB_REPLY_MASK) >> IPC_GLB_REPLY_SHIFT; +} + +static inline u32 msg_get_stream_type(u32 msg) +{ + return (msg & IPC_STR_TYPE_MASK) >> IPC_STR_TYPE_SHIFT; +} + +static inline u32 msg_get_stage_type(u32 msg) +{ + return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; +} + +static inline u32 msg_set_stage_type(u32 msg, u32 type) +{ + return (msg & ~IPC_STG_TYPE_MASK) + + (type << IPC_STG_TYPE_SHIFT); +} + +static inline u32 msg_get_stream_id(u32 msg) +{ + return (msg & IPC_STR_ID_MASK) >> IPC_STR_ID_SHIFT; +} + +static inline u32 msg_get_notify_reason(u32 msg) +{ + return (msg & IPC_STG_TYPE_MASK) >> IPC_STG_TYPE_SHIFT; +} + +u32 create_channel_map(enum sst_hsw_channel_config config) +{ + switch (config) { + case SST_HSW_CHANNEL_CONFIG_MONO: + return (0xFFFFFFF0 | SST_HSW_CHANNEL_CENTER); + case SST_HSW_CHANNEL_CONFIG_STEREO: + return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_RIGHT << 4)); + case SST_HSW_CHANNEL_CONFIG_2_POINT_1: + return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_RIGHT << 4) + | (SST_HSW_CHANNEL_LFE << 8 )); + case SST_HSW_CHANNEL_CONFIG_3_POINT_0: + return (0xFFFFF000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8)); + case SST_HSW_CHANNEL_CONFIG_3_POINT_1: + return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_LFE << 12)); + case SST_HSW_CHANNEL_CONFIG_QUATRO: + return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_RIGHT << 4) + | (SST_HSW_CHANNEL_LEFT_SURROUND << 8) + | (SST_HSW_CHANNEL_RIGHT_SURROUND << 12)); + case SST_HSW_CHANNEL_CONFIG_4_POINT_0: + return (0xFFFF0000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_CENTER_SURROUND << 12)); + case SST_HSW_CHANNEL_CONFIG_5_POINT_0: + return (0xFFF00000 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_CENTER << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_LEFT_SURROUND << 12) + | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16)); + case SST_HSW_CHANNEL_CONFIG_5_POINT_1: + return (0xFF000000 | SST_HSW_CHANNEL_CENTER + | (SST_HSW_CHANNEL_LEFT << 4) + | (SST_HSW_CHANNEL_RIGHT << 8) + | (SST_HSW_CHANNEL_LEFT_SURROUND << 12) + | (SST_HSW_CHANNEL_RIGHT_SURROUND << 16) + | (SST_HSW_CHANNEL_LFE << 20)); + case SST_HSW_CHANNEL_CONFIG_DUAL_MONO: + return (0xFFFFFF00 | SST_HSW_CHANNEL_LEFT + | (SST_HSW_CHANNEL_LEFT << 4)); + default: + return 0xFFFFFFFF; + } +} + +static struct sst_hsw_stream *get_stream_by_id(struct sst_hsw *hsw, + int stream_id) +{ + struct sst_hsw_stream *stream; + + list_for_each_entry(stream, &hsw->stream_list, node) { + if (stream->reply.stream_hw_id == stream_id) + return stream; + } + + return NULL; +} + +static void ipc_shim_dbg(struct sst_hsw *hsw, const char *text) +{ + struct sst_dsp *sst = hsw->dsp; + u32 isr, ipcd, imrx, ipcx; + + ipcx = sst_dsp_shim_read_unlocked(sst, SST_IPCX); + isr = sst_dsp_shim_read_unlocked(sst, SST_ISRX); + ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + imrx = sst_dsp_shim_read_unlocked(sst, SST_IMRX); + + dev_err(hsw->dev, "ipc: --%s-- ipcx 0x%8.8x isr 0x%8.8x ipcd 0x%8.8x imrx 0x%8.8x\n", + text, ipcx, isr, ipcd, imrx); +} + +/* locks held by caller */ +static struct ipc_message *msg_get_empty(struct sst_hsw *hsw) +{ + struct ipc_message *msg = NULL; + + if (!list_empty(&hsw->empty_list)) { + msg = list_first_entry(&hsw->empty_list, struct ipc_message, + list); + list_del(&msg->list); + } + + return msg; +} + +static void ipc_tx_msgs(struct kthread_work *work) +{ + struct sst_hsw *hsw = + container_of(work, struct sst_hsw, kwork); + struct ipc_message *msg; + unsigned long flags; + u32 ipcx; + + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + + if (list_empty(&hsw->tx_list) || hsw->pending) { + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return; + } + + /* if the DSP is busy we will TX messages after IRQ */ + ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); + if (ipcx & SST_IPCX_BUSY) { + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return; + } + + msg = list_first_entry(&hsw->tx_list, struct ipc_message, list); + + list_move(&msg->list, &hsw->rx_list); + + /* send the message */ + sst_dsp_outbox_write(hsw->dsp, msg->tx_data, msg->tx_size); + sst_dsp_ipc_msg_tx(hsw->dsp, msg->header | SST_IPCX_BUSY); + + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); +} + +/* locks held by caller */ +static void tx_msg_reply_complete(struct sst_hsw *hsw, struct ipc_message *msg) +{ + msg->complete = true; + trace_ipc_reply("completed", msg->header); + + if (!msg->wait) + list_add_tail(&msg->list, &hsw->empty_list); + else + wake_up(&msg->waitq); +} + +static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, + void *rx_data) +{ + unsigned long flags; + int ret; + + /* wait for DSP completion (in all cases atm inc pending) */ + ret = wait_event_timeout(msg->waitq, msg->complete, + msecs_to_jiffies(IPC_TIMEOUT_MSECS)); + + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + if (ret == 0) { + ipc_shim_dbg(hsw, "message timeout"); + + trace_ipc_error("error message timeout for", msg->header); + ret = -ETIMEDOUT; + } else { + + /* copy the data returned from DSP */ + if (msg->rx_size) + memcpy(rx_data, msg->rx_data, msg->rx_size); + ret = msg->errno; + } + + list_add_tail(&msg->list, &hsw->empty_list); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return ret; +} + +static int ipc_tx_message(struct sst_hsw *hsw, u32 header, void *tx_data, + size_t tx_bytes, void *rx_data, size_t rx_bytes, int wait) +{ + struct ipc_message *msg; + unsigned long flags; + + spin_lock_irqsave(&hsw->dsp->spinlock, flags); + + msg = msg_get_empty(hsw); + if (msg == NULL) { + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + return -EBUSY; + } + + if (tx_bytes) + memcpy(msg->tx_data, tx_data, tx_bytes); + + msg->header = header; + msg->tx_size = tx_bytes; + msg->rx_size = rx_bytes; + msg->wait = wait; + msg->errno = 0; + msg->pending = false; + msg->complete = false; + + list_add_tail(&msg->list, &hsw->tx_list); + spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); + + queue_kthread_work(&hsw->kworker, &hsw->kwork); + + if (wait) + return tx_wait_done(hsw, msg, rx_data); + else + return 0; +} + +static inline int ipc_tx_message_wait(struct sst_hsw *hsw, u32 header, + void *tx_data, size_t tx_bytes, void *rx_data, size_t rx_bytes) +{ + return ipc_tx_message(hsw, header, tx_data, tx_bytes, rx_data, + rx_bytes, 1); +} + +static inline int ipc_tx_message_nowait(struct sst_hsw *hsw, u32 header, + void *tx_data, size_t tx_bytes) +{ + return ipc_tx_message(hsw, header, tx_data, tx_bytes, NULL, 0, 0); +} + +static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) +{ + struct sst_hsw_ipc_fw_ready fw_ready; + u32 offset; + + offset = (header & 0x1FFFFFFF) << 3; + + dev_dbg(hsw->dev, "ipc: DSP is ready 0x%8.8x offset %d\n", + header, offset); + + /* copy data from the DSP FW ready offset */ + sst_dsp_read(hsw->dsp, &fw_ready, offset, sizeof(fw_ready)); + + sst_dsp_mailbox_init(hsw->dsp, fw_ready.inbox_offset, + fw_ready.inbox_size, fw_ready.outbox_offset, + fw_ready.outbox_size); + + hsw->boot_complete = true; + wake_up(&hsw->boot_wait); + + dev_dbg(hsw->dev, " mailbox upstream 0x%x - size 0x%x\n", + fw_ready.inbox_offset, fw_ready.inbox_size); + dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", + fw_ready.outbox_offset, fw_ready.outbox_size); +} + +static void hsw_notification_work(struct work_struct *work) +{ + struct sst_hsw_stream *stream = container_of(work, + struct sst_hsw_stream, notify_work); + struct sst_hsw_ipc_stream_glitch_position *glitch = &stream->glitch; + struct sst_hsw_ipc_stream_get_position *pos = &stream->rpos; + struct sst_hsw *hsw = stream->hsw; + u32 reason; + + reason = msg_get_notify_reason(stream->header); + + switch (reason) { + case IPC_STG_GLITCH: + trace_ipc_notification("DSP stream under/overrun", + stream->reply.stream_hw_id); + sst_dsp_inbox_read(hsw->dsp, glitch, sizeof(*glitch)); + + dev_err(hsw->dev, "glitch %d pos 0x%x write pos 0x%x\n", + glitch->glitch_type, glitch->present_pos, + glitch->write_pos); + break; + + case IPC_POSITION_CHANGED: + trace_ipc_notification("DSP stream position changed for", + stream->reply.stream_hw_id); + sst_dsp_inbox_read(hsw->dsp, pos, sizeof(pos)); + + if (stream->notify_position) + stream->notify_position(stream, stream->pdata); + + break; + default: + dev_err(hsw->dev, "error: unknown notification 0x%x\n", + stream->header); + break; + } + + /* tell DSP that notification has been handled */ + sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IPCD, + SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); + + /* unmask busy interrupt */ + sst_dsp_shim_update_bits_unlocked(hsw->dsp, SST_IMRX, SST_IMRX_BUSY, 0); +} + +static struct ipc_message *reply_find_msg(struct sst_hsw *hsw, u32 header) +{ + struct ipc_message *msg; + + /* clear reply bits & status bits */ + header &= ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK); + + if (list_empty(&hsw->rx_list)) { + dev_err(hsw->dev, "error: rx list empty but received 0x%x\n", + header); + return NULL; + } + + list_for_each_entry(msg, &hsw->rx_list, list) { + if (msg->header == header) + return msg; + } + + return NULL; +} + +static void hsw_stream_update(struct sst_hsw *hsw, struct ipc_message *msg) +{ + struct sst_hsw_stream *stream; + u32 header = msg->header & ~(IPC_STATUS_MASK | IPC_GLB_REPLY_MASK); + u32 stream_id = msg_get_stream_id(header); + u32 stream_msg = msg_get_stream_type(header); + + stream = get_stream_by_id(hsw, stream_id); + if (stream == NULL) + return; + + switch (stream_msg) { + case IPC_STR_STAGE_MESSAGE: + case IPC_STR_NOTIFICATION: + case IPC_STR_RESET: + break; + case IPC_STR_PAUSE: + stream->running = false; + trace_ipc_notification("stream paused", + stream->reply.stream_hw_id); + break; + case IPC_STR_RESUME: + stream->running = true; + trace_ipc_notification("stream running", + stream->reply.stream_hw_id); + break; + } +} + +static int hsw_process_reply(struct sst_hsw *hsw, u32 header) +{ + struct ipc_message *msg; + u32 reply = msg_get_global_reply(header); + + trace_ipc_reply("processing -->", header); + + msg = reply_find_msg(hsw, header); + if (msg == NULL) { + trace_ipc_error("error: can't find message header", header); + return -EIO; + } + + /* first process the header */ + switch (reply) { + case IPC_GLB_REPLY_PENDING: + trace_ipc_pending_reply("received", header); + msg->pending = true; + hsw->pending = true; + return 1; + case IPC_GLB_REPLY_SUCCESS: + if (msg->pending) { + trace_ipc_pending_reply("completed", header); + sst_dsp_inbox_read(hsw->dsp, msg->rx_data, + msg->rx_size); + hsw->pending = false; + } else { + /* copy data from the DSP */ + sst_dsp_outbox_read(hsw->dsp, msg->rx_data, + msg->rx_size); + } + break; + /* these will be rare - but useful for debug */ + case IPC_GLB_REPLY_UNKNOWN_MESSAGE_TYPE: + trace_ipc_error("error: unknown message type", header); + msg->errno = -EBADMSG; + break; + case IPC_GLB_REPLY_OUT_OF_RESOURCES: + trace_ipc_error("error: out of resources", header); + msg->errno = -ENOMEM; + break; + case IPC_GLB_REPLY_BUSY: + trace_ipc_error("error: reply busy", header); + msg->errno = -EBUSY; + break; + case IPC_GLB_REPLY_FAILURE: + trace_ipc_error("error: reply failure", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_STAGE_UNINITIALIZED: + trace_ipc_error("error: stage uninitialized", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_NOT_FOUND: + trace_ipc_error("error: reply not found", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_SOURCE_NOT_STARTED: + trace_ipc_error("error: source not started", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_INVALID_REQUEST: + trace_ipc_error("error: invalid request", header); + msg->errno = -EINVAL; + break; + case IPC_GLB_REPLY_ERROR_INVALID_PARAM: + trace_ipc_error("error: invalid parameter", header); + msg->errno = -EINVAL; + break; + default: + trace_ipc_error("error: unknown reply", header); + msg->errno = -EINVAL; + break; + } + + /* update any stream states */ + hsw_stream_update(hsw, msg); + + /* wake up and return the error if we have waiters on this message ? */ + list_del(&msg->list); + tx_msg_reply_complete(hsw, msg); + + return 1; +} + +static int hsw_stream_message(struct sst_hsw *hsw, u32 header) +{ + u32 stream_msg, stream_id, stage_type; + struct sst_hsw_stream *stream; + int handled = 0; + + stream_msg = msg_get_stream_type(header); + stream_id = msg_get_stream_id(header); + stage_type = msg_get_stage_type(header); + + stream = get_stream_by_id(hsw, stream_id); + if (stream == NULL) + return handled; + + stream->header = header; + + switch (stream_msg) { + case IPC_STR_STAGE_MESSAGE: + dev_err(hsw->dev, "error: stage msg not implemented 0x%8.8x\n", + header); + break; + case IPC_STR_NOTIFICATION: + schedule_work(&stream->notify_work); + break; + default: + /* handle pending message complete request */ + handled = hsw_process_reply(hsw, header); + break; + } + + return handled; +} + +static int hsw_log_message(struct sst_hsw *hsw, u32 header) +{ + u32 operation = (header & IPC_LOG_OP_MASK) >> IPC_LOG_OP_SHIFT; + struct sst_hsw_log_stream *stream = &hsw->log_stream; + int ret = 1; + + if (operation != IPC_DEBUG_REQUEST_LOG_DUMP) { + dev_err(hsw->dev, + "error: log msg not implemented 0x%8.8x\n", header); + return 0; + } + + mutex_lock(&stream->rw_mutex); + stream->last_pos = stream->curr_pos; + sst_dsp_inbox_read( + hsw->dsp, &stream->curr_pos, sizeof(stream->curr_pos)); + mutex_unlock(&stream->rw_mutex); + + schedule_work(&stream->notify_work); + + return ret; +} + +static int hsw_process_notification(struct sst_hsw *hsw) +{ + struct sst_dsp *sst = hsw->dsp; + u32 type, header; + int handled = 1; + + header = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + type = msg_get_global_type(header); + + trace_ipc_request("processing -->", header); + + /* FW Ready is a special case */ + if (!hsw->boot_complete && header & IPC_FW_READY) { + hsw_fw_ready(hsw, header); + return handled; + } + + switch (type) { + case IPC_GLB_GET_FW_VERSION: + case IPC_GLB_ALLOCATE_STREAM: + case IPC_GLB_FREE_STREAM: + case IPC_GLB_GET_FW_CAPABILITIES: + case IPC_GLB_REQUEST_DUMP: + case IPC_GLB_GET_DEVICE_FORMATS: + case IPC_GLB_SET_DEVICE_FORMATS: + case IPC_GLB_ENTER_DX_STATE: + case IPC_GLB_GET_MIXER_STREAM_INFO: + case IPC_GLB_MAX_IPC_MESSAGE_TYPE: + case IPC_GLB_RESTORE_CONTEXT: + case IPC_GLB_SHORT_REPLY: + dev_err(hsw->dev, "error: message type %d header 0x%x\n", + type, header); + break; + case IPC_GLB_STREAM_MESSAGE: + handled = hsw_stream_message(hsw, header); + break; + case IPC_GLB_DEBUG_LOG_MESSAGE: + handled = hsw_log_message(hsw, header); + break; + default: + dev_err(hsw->dev, "error: unexpected type %d hdr 0x%8.8x\n", + type, header); + break; + } + + return handled; +} + +static irqreturn_t hsw_irq_thread(int irq, void *context) +{ + struct sst_dsp *sst = (struct sst_dsp *) context; + struct sst_hsw *hsw = sst_dsp_get_thread_context(sst); + u32 ipcx, ipcd; + int handled; + unsigned long flags; + + spin_lock_irqsave(&sst->spinlock, flags); + + ipcx = sst_dsp_ipc_msg_rx(hsw->dsp); + ipcd = sst_dsp_shim_read_unlocked(sst, SST_IPCD); + + /* reply message from DSP */ + if (ipcx & SST_IPCX_DONE) { + + /* Handle Immediate reply from DSP Core */ + handled = hsw_process_reply(hsw, ipcx); + + if (handled > 0) { + /* clear DONE bit - tell DSP we have completed */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCX, + SST_IPCX_DONE, 0); + + /* unmask Done interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_DONE, 0); + } + } + + /* new message from DSP */ + if (ipcd & SST_IPCD_BUSY) { + + /* Handle Notification and Delayed reply from DSP Core */ + handled = hsw_process_notification(hsw); + + /* clear BUSY bit and set DONE bit - accept new messages */ + if (handled > 0) { + sst_dsp_shim_update_bits_unlocked(sst, SST_IPCD, + SST_IPCD_BUSY | SST_IPCD_DONE, SST_IPCD_DONE); + + /* unmask busy interrupt */ + sst_dsp_shim_update_bits_unlocked(sst, SST_IMRX, + SST_IMRX_BUSY, 0); + } + } + + spin_unlock_irqrestore(&sst->spinlock, flags); + + /* continue to send any remaining messages... */ + queue_kthread_work(&hsw->kworker, &hsw->kwork); + + return IRQ_HANDLED; +} + +int sst_hsw_fw_get_version(struct sst_hsw *hsw, + struct sst_hsw_ipc_fw_version *version) +{ + int ret; + + ret = ipc_tx_message_wait(hsw, IPC_GLB_TYPE(IPC_GLB_GET_FW_VERSION), + NULL, 0, version, sizeof(*version)); + if (ret < 0) + dev_err(hsw->dev, "error: get version failed\n"); + + return ret; +} + +/* Mixer Controls */ +int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel) +{ + int ret; + + ret = sst_hsw_stream_get_volume(hsw, stream, stage_id, channel, + &stream->mute_volume[channel]); + if (ret < 0) + return ret; + + ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + stream->reply.stream_hw_id, channel); + return ret; + } + + stream->mute[channel] = 1; + return 0; +} + +int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel) + +{ + int ret; + + stream->mute[channel] = 0; + ret = sst_hsw_stream_set_volume(hsw, stream, stage_id, channel, + stream->mute_volume[channel]); + if (ret < 0) { + dev_err(hsw->dev, "error: can't unmute stream %d channel %d\n", + stream->reply.stream_hw_id, channel); + return ret; + } + + return 0; +} + +int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel, u32 *volume) +{ + if (channel > 1) + return -EINVAL; + + sst_dsp_read(hsw->dsp, volume, + stream->reply.volume_register_address[channel], sizeof(volume)); + + return 0; +} + +int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u64 curve_duration, + enum sst_hsw_volume_curve curve) +{ + /* curve duration in steps of 100ns */ + stream->vol_req.curve_duration = curve_duration; + stream->vol_req.curve_type = curve; + + return 0; +} + +/* stream volume */ +int sst_hsw_stream_set_volume(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume) +{ + struct sst_hsw_ipc_volume_req *req; + u32 header; + int ret; + + trace_ipc_request("set stream volume", stream->reply.stream_hw_id); + + if (channel > 1) + return -EINVAL; + + if (stream->mute[channel]) { + stream->mute_volume[channel] = volume; + return 0; + } + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | + IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); + header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); + header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT); + header |= (stage_id << IPC_STG_ID_SHIFT); + + req = &stream->vol_req; + req->channel = channel; + req->target_volume = volume; + + ret = ipc_tx_message_wait(hsw, header, req, sizeof(*req), NULL, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: set stream volume failed\n"); + return ret; + } + + return 0; +} + +int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel) +{ + int ret; + + ret = sst_hsw_mixer_get_volume(hsw, stage_id, channel, + &hsw->mute_volume[channel]); + if (ret < 0) + return ret; + + ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + channel); + return ret; + } + + hsw->mute[channel] = 1; + return 0; +} + +int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel) +{ + int ret; + + ret = sst_hsw_mixer_set_volume(hsw, stage_id, channel, + hsw->mixer_info.volume_register_address[channel]); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to unmute mixer channel %d\n", + channel); + return ret; + } + + hsw->mute[channel] = 0; + return 0; +} + +int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 *volume) +{ + if (channel > 1) + return -EINVAL; + + sst_dsp_read(hsw->dsp, volume, + hsw->mixer_info.volume_register_address[channel], + sizeof(*volume)); + + return 0; +} + +int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, + u64 curve_duration, enum sst_hsw_volume_curve curve) +{ + /* curve duration in steps of 100ns */ + hsw->curve_duration = curve_duration; + hsw->curve_type = curve; + + return 0; +} + +/* global mixer volume */ +int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 volume) +{ + struct sst_hsw_ipc_volume_req req; + u32 header; + int ret; + + trace_ipc_request("set mixer volume", volume); + + /* set both at same time ? */ + if (channel == 2) { + if (hsw->mute[0] && hsw->mute[1]) { + hsw->mute_volume[0] = hsw->mute_volume[1] = volume; + return 0; + } else if (hsw->mute[0]) + req.channel = 1; + else if (hsw->mute[1]) + req.channel = 0; + else + req.channel = 0xffffffff; + } else { + /* set only 1 channel */ + if (hsw->mute[channel]) { + hsw->mute_volume[channel] = volume; + return 0; + } + req.channel = channel; + } + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | + IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); + header |= (hsw->mixer_info.mixer_hw_id << IPC_STR_ID_SHIFT); + header |= (IPC_STG_SET_VOLUME << IPC_STG_TYPE_SHIFT); + header |= (stage_id << IPC_STG_ID_SHIFT); + + req.curve_duration = hsw->curve_duration; + req.curve_type = hsw->curve_type; + req.target_volume = volume; + + ret = ipc_tx_message_wait(hsw, header, &req, sizeof(req), NULL, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: set mixer volume failed\n"); + return ret; + } + + return 0; +} + +/* Stream API */ +struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, + u32 (*notify_position)(struct sst_hsw_stream *stream, void *data), + void *data) +{ + struct sst_hsw_stream *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (stream == NULL) + return NULL; + + list_add(&stream->node, &hsw->stream_list); + stream->notify_position = notify_position; + stream->pdata = data; + stream->hsw = hsw; + stream->host_id = id; + + /* work to process notification messages */ + INIT_WORK(&stream->notify_work, hsw_notification_work); + + return stream; +} + +int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) +{ + u32 header; + int ret = 0; + + /* dont free DSP streams that are not commited */ + if (!stream->commited) + goto out; + + trace_ipc_request("stream free", stream->host_id); + + stream->free_req.stream_id = stream->reply.stream_hw_id; + header = IPC_GLB_TYPE(IPC_GLB_FREE_STREAM); + + ret = ipc_tx_message_wait(hsw, header, &stream->free_req, + sizeof(stream->free_req), NULL, 0); + if (ret < 0) { + dev_err(hsw->dev, "error: free stream %d failed\n", + stream->free_req.stream_id); + return -EAGAIN; + } + + trace_hsw_stream_free_req(stream, &stream->free_req); + +out: + list_del(&stream->node); + kfree(stream); + + return ret; +} + +int sst_hsw_stream_set_bits(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_bitdepth bits) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set bits\n"); + return -EINVAL; + } + + stream->request.format.bitdepth = bits; + return 0; +} + +int sst_hsw_stream_set_channels(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, int channels) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set channels\n"); + return -EINVAL; + } + + /* stereo is only supported atm */ + if (channels != 2) + return -EINVAL; + + stream->request.format.ch_num = channels; + return 0; +} + +int sst_hsw_stream_set_rate(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, int rate) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set rate\n"); + return -EINVAL; + } + + stream->request.format.frequency = rate; + return 0; +} + +int sst_hsw_stream_set_map_config(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 map, + enum sst_hsw_channel_config config) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set map\n"); + return -EINVAL; + } + + stream->request.format.map = map; + stream->request.format.config = config; + return 0; +} + +int sst_hsw_stream_set_style(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_interleaving style) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set style\n"); + return -EINVAL; + } + + stream->request.format.style = style; + return 0; +} + +int sst_hsw_stream_set_valid(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 bits) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set valid bits\n"); + return -EINVAL; + } + + stream->request.format.valid_bit = bits; + return 0; +} + +/* Stream Configuration */ +int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_stream_path_id path_id, + enum sst_hsw_stream_type stream_type, + enum sst_hsw_stream_format format_id) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set format\n"); + return -EINVAL; + } + + stream->request.path_id = path_id; + stream->request.stream_type = stream_type; + stream->request.format_id = format_id; + + trace_hsw_stream_alloc_request(stream, &stream->request); + + return 0; +} + +int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 ring_pt_address, u32 num_pages, + u32 ring_size, u32 ring_offset, u32 ring_first_pfn) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for buffer\n"); + return -EINVAL; + } + + stream->request.ringinfo.ring_pt_address = ring_pt_address; + stream->request.ringinfo.num_pages = num_pages; + stream->request.ringinfo.ring_size = ring_size; + stream->request.ringinfo.ring_offset = ring_offset; + stream->request.ringinfo.ring_first_pfn = ring_first_pfn; + + trace_hsw_stream_buffer(stream); + + return 0; +} + +int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id, + u32 entry_point) +{ + struct sst_hsw_module_map *map = &stream->request.map; + + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set module\n"); + return -EINVAL; + } + + /* only support initial module atm */ + map->module_entries_count = 1; + map->module_entries[0].module_id = module_id; + map->module_entries[0].entry_point = entry_point; + + return 0; +} + +int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set pmem\n"); + return -EINVAL; + } + + stream->request.persistent_mem.offset = offset; + stream->request.persistent_mem.size = size; + + return 0; +} + +int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size) +{ + if (stream->commited) { + dev_err(hsw->dev, "error: stream committed for set smem\n"); + return -EINVAL; + } + + stream->request.scratch_mem.offset = offset; + stream->request.scratch_mem.size = size; + + return 0; +} + +int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream) +{ + struct sst_hsw_ipc_stream_alloc_req *str_req = &stream->request; + struct sst_hsw_ipc_stream_alloc_reply *reply = &stream->reply; + u32 header; + int ret; + + trace_ipc_request("stream alloc", stream->host_id); + + header = IPC_GLB_TYPE(IPC_GLB_ALLOCATE_STREAM); + + ret = ipc_tx_message_wait(hsw, header, str_req, sizeof(*str_req), + reply, sizeof(*reply)); + if (ret < 0) { + dev_err(hsw->dev, "error: stream commit failed\n"); + return ret; + } + + stream->commited = 1; + trace_hsw_stream_alloc_reply(stream); + + return 0; +} + +/* Stream Information - these calls could be inline but we want the IPC + ABI to be opaque to client PCM drivers to cope with any future ABI changes */ +int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.stream_hw_id; +} + +int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.mixer_hw_id; +} + +u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.read_position_register_address; +} + +u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->reply.presentation_position_register_address; +} + +u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel) +{ + if (channel >= 2) + return 0; + + return stream->reply.peak_meter_register_address[channel]; +} + +u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel) +{ + if (channel >= 2) + return 0; + + return stream->reply.volume_register_address[channel]; +} + +int sst_hsw_mixer_get_info(struct sst_hsw *hsw) +{ + struct sst_hsw_ipc_stream_info_reply *reply; + u32 header; + int ret; + + reply = &hsw->mixer_info; + header = IPC_GLB_TYPE(IPC_GLB_GET_MIXER_STREAM_INFO); + + trace_ipc_request("get global mixer info", 0); + + ret = ipc_tx_message_wait(hsw, header, NULL, 0, reply, sizeof(*reply)); + if (ret < 0) { + dev_err(hsw->dev, "error: get stream info failed\n"); + return ret; + } + + trace_hsw_mixer_info_reply(reply); + + return 0; +} + +/* Send stream command */ +static int sst_hsw_stream_operations(struct sst_hsw *hsw, int type, + int stream_id, int wait) +{ + u32 header; + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | IPC_STR_TYPE(type); + header |= (stream_id << IPC_STR_ID_SHIFT); + + if (wait) + return ipc_tx_message_wait(hsw, header, NULL, 0, NULL, 0); + else + return ipc_tx_message_nowait(hsw, header, NULL, 0); +} + +/* Stream ALSA trigger operations */ +int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait) +{ + int ret; + + trace_ipc_request("stream pause", stream->reply.stream_hw_id); + + ret = sst_hsw_stream_operations(hsw, IPC_STR_PAUSE, + stream->reply.stream_hw_id, wait); + if (ret < 0) + dev_err(hsw->dev, "error: failed to pause stream %d\n", + stream->reply.stream_hw_id); + + return ret; +} + +int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait) +{ + int ret; + + trace_ipc_request("stream resume", stream->reply.stream_hw_id); + + ret = sst_hsw_stream_operations(hsw, IPC_STR_RESUME, + stream->reply.stream_hw_id, wait); + if (ret < 0) + dev_err(hsw->dev, "error: failed to resume stream %d\n", + stream->reply.stream_hw_id); + + return ret; +} + +int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream) +{ + int ret, tries = 10; + + /* dont reset streams that are not commited */ + if (!stream->commited) + return 0; + + /* wait for pause to complete before we reset the stream */ + while (stream->running && tries--) + msleep(1); + if (!tries) { + dev_err(hsw->dev, "error: reset stream %d still running\n", + stream->reply.stream_hw_id); + return -EINVAL; + } + + trace_ipc_request("stream reset", stream->reply.stream_hw_id); + + ret = sst_hsw_stream_operations(hsw, IPC_STR_RESET, + stream->reply.stream_hw_id, 1); + if (ret < 0) + dev_err(hsw->dev, "error: failed to reset stream %d\n", + stream->reply.stream_hw_id); + return ret; +} + +/* Stream pointer positions */ +int sst_hsw_get_dsp_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream) +{ + return stream->rpos.position; +} + +int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 position) +{ + u32 header; + int ret; + + trace_stream_write_position(stream->reply.stream_hw_id, position); + + header = IPC_GLB_TYPE(IPC_GLB_STREAM_MESSAGE) | + IPC_STR_TYPE(IPC_STR_STAGE_MESSAGE); + header |= (stream->reply.stream_hw_id << IPC_STR_ID_SHIFT); + header |= (IPC_STG_SET_WRITE_POSITION << IPC_STG_TYPE_SHIFT); + header |= (stage_id << IPC_STG_ID_SHIFT); + stream->wpos.position = position; + + ret = ipc_tx_message_nowait(hsw, header, &stream->wpos, + sizeof(stream->wpos)); + if (ret < 0) + dev_err(hsw->dev, "error: stream %d set position %d failed\n", + stream->reply.stream_hw_id, position); + + return ret; +} + +/* physical BE config */ +int sst_hsw_device_set_config(struct sst_hsw *hsw, + enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, + enum sst_hsw_device_mode mode, u32 clock_divider) +{ + struct sst_hsw_ipc_device_config_req config; + u32 header; + int ret; + + trace_ipc_request("set device config", dev); + + config.ssp_interface = dev; + config.clock_frequency = mclk; + config.mode = mode; + config.clock_divider = clock_divider; + + trace_hsw_device_config_req(&config); + + header = IPC_GLB_TYPE(IPC_GLB_SET_DEVICE_FORMATS); + + ret = ipc_tx_message_wait(hsw, header, &config, sizeof(config), + NULL, 0); + if (ret < 0) + dev_err(hsw->dev, "error: set device formats failed\n"); + + return ret; +} +EXPORT_SYMBOL_GPL(sst_hsw_device_set_config); + +/* DX Config */ +int sst_hsw_dx_set_state(struct sst_hsw *hsw, + enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx) +{ + u32 header, state_; + int ret; + + header = IPC_GLB_TYPE(IPC_GLB_ENTER_DX_STATE); + state_ = state; + + trace_ipc_request("PM enter Dx state", state); + + ret = ipc_tx_message_wait(hsw, header, &state_, sizeof(state_), + dx, sizeof(dx)); + if (ret < 0) { + dev_err(hsw->dev, "ipc: error set dx state %d failed\n", state); + return ret; + } + + dev_dbg(hsw->dev, "ipc: got %d entry numbers for state %d\n", + dx->entries_no, state); + + memcpy(&hsw->dx, dx, sizeof(*dx)); + return 0; +} + +/* Used to save state into hsw->dx_reply */ +int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, + u32 *offset, u32 *size, u32 *source) +{ + struct sst_hsw_ipc_dx_memory_item *dx_mem; + struct sst_hsw_ipc_dx_reply *dx_reply; + int entry_no; + + dx_reply = &hsw->dx; + entry_no = dx_reply->entries_no; + + trace_ipc_request("PM get Dx state", entry_no); + + if (item >= entry_no) + return -EINVAL; + + dx_mem = &dx_reply->mem_info[item]; + *offset = dx_mem->offset; + *size = dx_mem->size; + *source = dx_mem->source; + + return 0; +} + +static int msg_empty_list_init(struct sst_hsw *hsw) +{ + int i; + + hsw->msg = kzalloc(sizeof(struct ipc_message) * + IPC_EMPTY_LIST_SIZE, GFP_KERNEL); + if (hsw->msg == NULL) + return -ENOMEM; + + for (i = 0; i < IPC_EMPTY_LIST_SIZE; i++) { + init_waitqueue_head(&hsw->msg[i].waitq); + list_add(&hsw->msg[i].list, &hsw->empty_list); + } + + return 0; +} + +void sst_hsw_set_scratch_module(struct sst_hsw *hsw, + struct sst_module *scratch) +{ + hsw->scratch = scratch; +} + +struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw) +{ + return hsw->dsp; +} + +static struct sst_dsp_device hsw_dev = { + .thread = hsw_irq_thread, + .ops = &haswell_ops, +}; + +int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_hsw_ipc_fw_version version; + struct sst_hsw *hsw; + struct sst_fw *hsw_sst_fw; + int ret; + + dev_dbg(dev, "initialising Audio DSP IPC\n"); + + hsw = devm_kzalloc(dev, sizeof(*hsw), GFP_KERNEL); + if (hsw == NULL) + return -ENOMEM; + + hsw->dev = dev; + INIT_LIST_HEAD(&hsw->stream_list); + INIT_LIST_HEAD(&hsw->tx_list); + INIT_LIST_HEAD(&hsw->rx_list); + INIT_LIST_HEAD(&hsw->empty_list); + init_waitqueue_head(&hsw->boot_wait); + init_waitqueue_head(&hsw->wait_txq); + + ret = msg_empty_list_init(hsw); + if (ret < 0) + goto list_err; + + /* start the IPC message thread */ + init_kthread_worker(&hsw->kworker); + hsw->tx_thread = kthread_run(kthread_worker_fn, + &hsw->kworker, + dev_name(hsw->dev)); + if (IS_ERR(hsw->tx_thread)) { + ret = PTR_ERR(hsw->tx_thread); + dev_err(hsw->dev, "error: failed to create message TX task\n"); + goto list_err; + } + init_kthread_work(&hsw->kwork, ipc_tx_msgs); + + hsw_dev.thread_context = hsw; + + /* init SST shim */ + hsw->dsp = sst_dsp_new(dev, &hsw_dev, pdata); + if (hsw->dsp == NULL) { + ret = -ENODEV; + goto list_err; + } + + /* keep the DSP in reset state for base FW loading */ + sst_dsp_reset(hsw->dsp); + + hsw_sst_fw = sst_fw_new(hsw->dsp, pdata->fw, hsw); + + if (hsw_sst_fw == NULL) { + ret = -ENODEV; + dev_err(dev, "error: failed to load firmware\n"); + goto fw_err; + } + + /* wait for DSP boot completion */ + sst_dsp_boot(hsw->dsp); + ret = wait_event_timeout(hsw->boot_wait, hsw->boot_complete, + msecs_to_jiffies(IPC_BOOT_MSECS)); + if (ret == 0) { + ret = -EIO; + dev_err(hsw->dev, "error: ADSP boot timeout\n"); + goto boot_err; + } + + /* get the FW version */ + sst_hsw_fw_get_version(hsw, &version); + dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", + version.type, version.major, version.minor, version.build); + + /* get the globalmixer */ + ret = sst_hsw_mixer_get_info(hsw); + if (ret < 0) { + dev_err(hsw->dev, "error: failed to get stream info\n"); + goto boot_err; + } + + pdata->dsp = hsw; + return 0; + +boot_err: + sst_dsp_reset(hsw->dsp); + sst_fw_free(hsw_sst_fw); +fw_err: + sst_dsp_free(hsw->dsp); + kfree(hsw->msg); +list_err: + return ret; +} +EXPORT_SYMBOL_GPL(sst_hsw_dsp_init); + +void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata) +{ + struct sst_hsw *hsw = pdata->dsp; + + sst_dsp_reset(hsw->dsp); + sst_fw_free_all(hsw->dsp); + sst_dsp_free(hsw->dsp); + kfree(hsw->scratch); + kfree(hsw->msg); +} +EXPORT_SYMBOL_GPL(sst_hsw_dsp_free); diff --git a/sound/soc/intel/sst-haswell-ipc.h b/sound/soc/intel/sst-haswell-ipc.h new file mode 100644 index 000000000000..d517929ccc38 --- /dev/null +++ b/sound/soc/intel/sst-haswell-ipc.h @@ -0,0 +1,488 @@ +/* + * Intel SST Haswell/Broadwell IPC Support + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#ifndef __SST_HASWELL_IPC_H +#define __SST_HASWELL_IPC_H + +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/platform_device.h> + +#define SST_HSW_NO_CHANNELS 2 +#define SST_HSW_MAX_DX_REGIONS 14 + +#define SST_HSW_FW_LOG_CONFIG_DWORDS 12 +#define SST_HSW_GLOBAL_LOG 15 + +/** + * Upfront defined maximum message size that is + * expected by the in/out communication pipes in FW. + */ +#define SST_HSW_IPC_MAX_PAYLOAD_SIZE 400 +#define SST_HSW_MAX_INFO_SIZE 64 +#define SST_HSW_BUILD_HASH_LENGTH 40 + +struct sst_hsw; +struct sst_hsw_stream; +struct sst_hsw_log_stream; +struct sst_pdata; +struct sst_module; +extern struct sst_ops haswell_ops; + +/* Stream Allocate Path ID */ +enum sst_hsw_stream_path_id { + SST_HSW_STREAM_PATH_SSP0_OUT = 0, + SST_HSW_STREAM_PATH_SSP0_IN = 1, + SST_HSW_STREAM_PATH_MAX_PATH_ID = 2, +}; + +/* Stream Allocate Stream Type */ +enum sst_hsw_stream_type { + SST_HSW_STREAM_TYPE_RENDER = 0, + SST_HSW_STREAM_TYPE_SYSTEM = 1, + SST_HSW_STREAM_TYPE_CAPTURE = 2, + SST_HSW_STREAM_TYPE_LOOPBACK = 3, + SST_HSW_STREAM_TYPE_MAX_STREAM_TYPE = 4, +}; + +/* Stream Allocate Stream Format */ +enum sst_hsw_stream_format { + SST_HSW_STREAM_FORMAT_PCM_FORMAT = 0, + SST_HSW_STREAM_FORMAT_MP3_FORMAT = 1, + SST_HSW_STREAM_FORMAT_AAC_FORMAT = 2, + SST_HSW_STREAM_FORMAT_MAX_FORMAT_ID = 3, +}; + +/* Device ID */ +enum sst_hsw_device_id { + SST_HSW_DEVICE_SSP_0 = 0, + SST_HSW_DEVICE_SSP_1 = 1, +}; + +/* Device Master Clock Frequency */ +enum sst_hsw_device_mclk { + SST_HSW_DEVICE_MCLK_OFF = 0, + SST_HSW_DEVICE_MCLK_FREQ_6_MHZ = 1, + SST_HSW_DEVICE_MCLK_FREQ_12_MHZ = 2, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ = 3, +}; + +/* Device Clock Master */ +enum sst_hsw_device_mode { + SST_HSW_DEVICE_CLOCK_SLAVE = 0, + SST_HSW_DEVICE_CLOCK_MASTER = 1, +}; + +/* DX Power State */ +enum sst_hsw_dx_state { + SST_HSW_DX_STATE_D0 = 0, + SST_HSW_DX_STATE_D1 = 1, + SST_HSW_DX_STATE_D3 = 3, + SST_HSW_DX_STATE_MAX = 3, +}; + +/* Audio stream stage IDs */ +enum sst_hsw_fx_stage_id { + SST_HSW_STAGE_ID_WAVES = 0, + SST_HSW_STAGE_ID_DTS = 1, + SST_HSW_STAGE_ID_DOLBY = 2, + SST_HSW_STAGE_ID_BOOST = 3, + SST_HSW_STAGE_ID_MAX_FX_ID +}; + +/* DX State Type */ +enum sst_hsw_dx_type { + SST_HSW_DX_TYPE_FW_IMAGE = 0, + SST_HSW_DX_TYPE_MEMORY_DUMP = 1 +}; + +/* Volume Curve Type*/ +enum sst_hsw_volume_curve { + SST_HSW_VOLUME_CURVE_NONE = 0, + SST_HSW_VOLUME_CURVE_FADE = 1 +}; + +/* Sample ordering */ +enum sst_hsw_interleaving { + SST_HSW_INTERLEAVING_PER_CHANNEL = 0, + SST_HSW_INTERLEAVING_PER_SAMPLE = 1, +}; + +/* Channel indices */ +enum sst_hsw_channel_index { + SST_HSW_CHANNEL_LEFT = 0, + SST_HSW_CHANNEL_CENTER = 1, + SST_HSW_CHANNEL_RIGHT = 2, + SST_HSW_CHANNEL_LEFT_SURROUND = 3, + SST_HSW_CHANNEL_CENTER_SURROUND = 3, + SST_HSW_CHANNEL_RIGHT_SURROUND = 4, + SST_HSW_CHANNEL_LFE = 7, + SST_HSW_CHANNEL_INVALID = 0xF, +}; + +/* List of supported channel maps. */ +enum sst_hsw_channel_config { + SST_HSW_CHANNEL_CONFIG_MONO = 0, /* mono only. */ + SST_HSW_CHANNEL_CONFIG_STEREO = 1, /* L & R. */ + SST_HSW_CHANNEL_CONFIG_2_POINT_1 = 2, /* L, R & LFE; PCM only. */ + SST_HSW_CHANNEL_CONFIG_3_POINT_0 = 3, /* L, C & R; MP3 & AAC only. */ + SST_HSW_CHANNEL_CONFIG_3_POINT_1 = 4, /* L, C, R & LFE; PCM only. */ + SST_HSW_CHANNEL_CONFIG_QUATRO = 5, /* L, R, Ls & Rs; PCM only. */ + SST_HSW_CHANNEL_CONFIG_4_POINT_0 = 6, /* L, C, R & Cs; MP3 & AAC only. */ + SST_HSW_CHANNEL_CONFIG_5_POINT_0 = 7, /* L, C, R, Ls & Rs. */ + SST_HSW_CHANNEL_CONFIG_5_POINT_1 = 8, /* L, C, R, Ls, Rs & LFE. */ + SST_HSW_CHANNEL_CONFIG_DUAL_MONO = 9, /* One channel replicated in two. */ + SST_HSW_CHANNEL_CONFIG_INVALID, +}; + +/* List of supported bit depths. */ +enum sst_hsw_bitdepth { + SST_HSW_DEPTH_8BIT = 8, + SST_HSW_DEPTH_16BIT = 16, + SST_HSW_DEPTH_24BIT = 24, /* Default. */ + SST_HSW_DEPTH_32BIT = 32, + SST_HSW_DEPTH_INVALID = 33, +}; + +enum sst_hsw_module_id { + SST_HSW_MODULE_BASE_FW = 0x0, + SST_HSW_MODULE_MP3 = 0x1, + SST_HSW_MODULE_AAC_5_1 = 0x2, + SST_HSW_MODULE_AAC_2_0 = 0x3, + SST_HSW_MODULE_SRC = 0x4, + SST_HSW_MODULE_WAVES = 0x5, + SST_HSW_MODULE_DOLBY = 0x6, + SST_HSW_MODULE_BOOST = 0x7, + SST_HSW_MODULE_LPAL = 0x8, + SST_HSW_MODULE_DTS = 0x9, + SST_HSW_MODULE_PCM_CAPTURE = 0xA, + SST_HSW_MODULE_PCM_SYSTEM = 0xB, + SST_HSW_MODULE_PCM_REFERENCE = 0xC, + SST_HSW_MODULE_PCM = 0xD, + SST_HSW_MODULE_BLUETOOTH_RENDER_MODULE = 0xE, + SST_HSW_MODULE_BLUETOOTH_CAPTURE_MODULE = 0xF, + SST_HSW_MAX_MODULE_ID, +}; + +enum sst_hsw_performance_action { + SST_HSW_PERF_START = 0, + SST_HSW_PERF_STOP = 1, +}; + +/* SST firmware module info */ +struct sst_hsw_module_info { + u8 name[SST_HSW_MAX_INFO_SIZE]; + u8 version[SST_HSW_MAX_INFO_SIZE]; +} __attribute__((packed)); + +/* Module entry point */ +struct sst_hsw_module_entry { + enum sst_hsw_module_id module_id; + u32 entry_point; +} __attribute__((packed)); + +/* Module map - alignement matches DSP */ +struct sst_hsw_module_map { + u8 module_entries_count; + struct sst_hsw_module_entry module_entries[1]; +} __attribute__((packed)); + +struct sst_hsw_memory_info { + u32 offset; + u32 size; +} __attribute__((packed)); + +struct sst_hsw_fx_enable { + struct sst_hsw_module_map module_map; + struct sst_hsw_memory_info persistent_mem; +} __attribute__((packed)); + +struct sst_hsw_get_fx_param { + u32 parameter_id; + u32 param_size; +} __attribute__((packed)); + +struct sst_hsw_perf_action { + u32 action; +} __attribute__((packed)); + +struct sst_hsw_perf_data { + u64 timestamp; + u64 cycles; + u64 datatime; +} __attribute__((packed)); + +/* FW version */ +struct sst_hsw_ipc_fw_version { + u8 build; + u8 minor; + u8 major; + u8 type; + u8 fw_build_hash[SST_HSW_BUILD_HASH_LENGTH]; + u32 fw_log_providers_hash; +} __attribute__((packed)); + +/* Stream ring info */ +struct sst_hsw_ipc_stream_ring { + u32 ring_pt_address; + u32 num_pages; + u32 ring_size; + u32 ring_offset; + u32 ring_first_pfn; +} __attribute__((packed)); + +/* Debug Dump Log Enable Request */ +struct sst_hsw_ipc_debug_log_enable_req { + struct sst_hsw_ipc_stream_ring ringinfo; + u32 config[SST_HSW_FW_LOG_CONFIG_DWORDS]; +} __attribute__((packed)); + +/* Debug Dump Log Reply */ +struct sst_hsw_ipc_debug_log_reply { + u32 log_buffer_begining; + u32 log_buffer_size; +} __attribute__((packed)); + +/* Stream glitch position */ +struct sst_hsw_ipc_stream_glitch_position { + u32 glitch_type; + u32 present_pos; + u32 write_pos; +} __attribute__((packed)); + +/* Stream get position */ +struct sst_hsw_ipc_stream_get_position { + u32 position; + u32 fw_cycle_count; +} __attribute__((packed)); + +/* Stream set position */ +struct sst_hsw_ipc_stream_set_position { + u32 position; + u32 end_of_buffer; +} __attribute__((packed)); + +/* Stream Free Request */ +struct sst_hsw_ipc_stream_free_req { + u8 stream_id; + u8 reserved[3]; +} __attribute__((packed)); + +/* Set Volume Request */ +struct sst_hsw_ipc_volume_req { + u32 channel; + u32 target_volume; + u64 curve_duration; + u32 curve_type; +} __attribute__((packed)); + +/* Device Configuration Request */ +struct sst_hsw_ipc_device_config_req { + u32 ssp_interface; + u32 clock_frequency; + u32 mode; + u16 clock_divider; + u16 reserved; +} __attribute__((packed)); + +/* Audio Data formats */ +struct sst_hsw_audio_data_format_ipc { + u32 frequency; + u32 bitdepth; + u32 map; + u32 config; + u32 style; + u8 ch_num; + u8 valid_bit; + u8 reserved[2]; +} __attribute__((packed)); + +/* Stream Allocate Request */ +struct sst_hsw_ipc_stream_alloc_req { + u8 path_id; + u8 stream_type; + u8 format_id; + u8 reserved; + struct sst_hsw_audio_data_format_ipc format; + struct sst_hsw_ipc_stream_ring ringinfo; + struct sst_hsw_module_map map; + struct sst_hsw_memory_info persistent_mem; + struct sst_hsw_memory_info scratch_mem; + u32 number_of_notifications; +} __attribute__((packed)); + +/* Stream Allocate Reply */ +struct sst_hsw_ipc_stream_alloc_reply { + u32 stream_hw_id; + u32 mixer_hw_id; // returns rate ???? + u32 read_position_register_address; + u32 presentation_position_register_address; + u32 peak_meter_register_address[SST_HSW_NO_CHANNELS]; + u32 volume_register_address[SST_HSW_NO_CHANNELS]; +} __attribute__((packed)); + +/* Get Mixer Stream Info */ +struct sst_hsw_ipc_stream_info_reply { + u32 mixer_hw_id; + u32 peak_meter_register_address[SST_HSW_NO_CHANNELS]; + u32 volume_register_address[SST_HSW_NO_CHANNELS]; +} __attribute__((packed)); + +/* DX State Request */ +struct sst_hsw_ipc_dx_req { + u8 state; + u8 reserved[3]; +} __attribute__((packed)); + +/* DX State Reply Memory Info Item */ +struct sst_hsw_ipc_dx_memory_item { + u32 offset; + u32 size; + u32 source; +} __attribute__((packed)); + +/* DX State Reply */ +struct sst_hsw_ipc_dx_reply { + u32 entries_no; + struct sst_hsw_ipc_dx_memory_item mem_info[SST_HSW_MAX_DX_REGIONS]; +} __attribute__((packed)); + +struct sst_hsw_ipc_fw_version; + +/* SST Init & Free */ +struct sst_hsw *sst_hsw_new(struct device *dev, const u8 *fw, size_t fw_length, + u32 fw_offset); +void sst_hsw_free(struct sst_hsw *hsw); +int sst_hsw_fw_get_version(struct sst_hsw *hsw, + struct sst_hsw_ipc_fw_version *version); +u32 create_channel_map(enum sst_hsw_channel_config config); + +/* Stream Mixer Controls - */ +int sst_hsw_stream_mute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel); +int sst_hsw_stream_unmute(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 stage_id, u32 channel); + +int sst_hsw_stream_set_volume(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 volume); +int sst_hsw_stream_get_volume(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 channel, u32 *volume); + +int sst_hsw_stream_set_volume_curve(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u64 curve_duration, + enum sst_hsw_volume_curve curve); + +/* Global Mixer Controls - */ +int sst_hsw_mixer_mute(struct sst_hsw *hsw, u32 stage_id, u32 channel); +int sst_hsw_mixer_unmute(struct sst_hsw *hsw, u32 stage_id, u32 channel); + +int sst_hsw_mixer_set_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 volume); +int sst_hsw_mixer_get_volume(struct sst_hsw *hsw, u32 stage_id, u32 channel, + u32 *volume); + +int sst_hsw_mixer_set_volume_curve(struct sst_hsw *hsw, + u64 curve_duration, enum sst_hsw_volume_curve curve); + +/* Stream API */ +struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, + u32 (*get_write_position)(struct sst_hsw_stream *stream, void *data), + void *data); + +int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream); + +/* Stream Configuration */ +int sst_hsw_stream_format(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_stream_path_id path_id, + enum sst_hsw_stream_type stream_type, + enum sst_hsw_stream_format format_id); + +int sst_hsw_stream_buffer(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 ring_pt_address, u32 num_pages, + u32 ring_size, u32 ring_offset, u32 ring_first_pfn); + +int sst_hsw_stream_commit(struct sst_hsw *hsw, struct sst_hsw_stream *stream); + +int sst_hsw_stream_set_valid(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + u32 bits); +int sst_hsw_stream_set_rate(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int rate); +int sst_hsw_stream_set_bits(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_bitdepth bits); +int sst_hsw_stream_set_channels(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, int channels); +int sst_hsw_stream_set_map_config(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 map, + enum sst_hsw_channel_config config); +int sst_hsw_stream_set_style(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + enum sst_hsw_interleaving style); +int sst_hsw_stream_set_module_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, enum sst_hsw_module_id module_id, + u32 entry_point); +int sst_hsw_stream_set_pmemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size); +int sst_hsw_stream_set_smemory_info(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 offset, u32 size); +int sst_hsw_stream_get_hw_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +int sst_hsw_stream_get_mixer_id(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +u32 sst_hsw_stream_get_read_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +u32 sst_hsw_stream_get_pointer_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); +u32 sst_hsw_stream_get_peak_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel); +u32 sst_hsw_stream_get_vol_reg(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 channel); +int sst_hsw_mixer_get_info(struct sst_hsw *hsw); + +/* Stream ALSA trigger operations */ +int sst_hsw_stream_pause(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait); +int sst_hsw_stream_resume(struct sst_hsw *hsw, struct sst_hsw_stream *stream, + int wait); +int sst_hsw_stream_reset(struct sst_hsw *hsw, struct sst_hsw_stream *stream); + +/* Stream pointer positions */ +int sst_hsw_stream_get_read_pos(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 *position); +int sst_hsw_stream_get_write_pos(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 *position); +int sst_hsw_stream_set_write_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream, u32 stage_id, u32 position); +int sst_hsw_get_dsp_position(struct sst_hsw *hsw, + struct sst_hsw_stream *stream); + +/* HW port config */ +int sst_hsw_device_set_config(struct sst_hsw *hsw, + enum sst_hsw_device_id dev, enum sst_hsw_device_mclk mclk, + enum sst_hsw_device_mode mode, u32 clock_divider); + +/* DX Config */ +int sst_hsw_dx_set_state(struct sst_hsw *hsw, + enum sst_hsw_dx_state state, struct sst_hsw_ipc_dx_reply *dx); +int sst_hsw_dx_get_state(struct sst_hsw *hsw, u32 item, + u32 *offset, u32 *size, u32 *source); + +/* init */ +int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata); +void sst_hsw_dsp_free(struct device *dev, struct sst_pdata *pdata); +struct sst_dsp *sst_hsw_get_dsp(struct sst_hsw *hsw); +void sst_hsw_set_scratch_module(struct sst_hsw *hsw, + struct sst_module *scratch); + +#endif diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c new file mode 100644 index 000000000000..0a32dd13a23d --- /dev/null +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -0,0 +1,872 @@ +/* + * Intel SST Haswell/Broadwell PCM Support + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/delay.h> +#include <asm/page.h> +#include <asm/pgtable.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/compress_driver.h> + +#include "sst-haswell-ipc.h" +#include "sst-dsp-priv.h" +#include "sst-dsp.h" + +#define HSW_PCM_COUNT 6 +#define HSW_VOLUME_MAX 0x7FFFFFFF /* 0dB */ + +/* simple volume table */ +static const u32 volume_map[] = { + HSW_VOLUME_MAX >> 30, + HSW_VOLUME_MAX >> 29, + HSW_VOLUME_MAX >> 28, + HSW_VOLUME_MAX >> 27, + HSW_VOLUME_MAX >> 26, + HSW_VOLUME_MAX >> 25, + HSW_VOLUME_MAX >> 24, + HSW_VOLUME_MAX >> 23, + HSW_VOLUME_MAX >> 22, + HSW_VOLUME_MAX >> 21, + HSW_VOLUME_MAX >> 20, + HSW_VOLUME_MAX >> 19, + HSW_VOLUME_MAX >> 18, + HSW_VOLUME_MAX >> 17, + HSW_VOLUME_MAX >> 16, + HSW_VOLUME_MAX >> 15, + HSW_VOLUME_MAX >> 14, + HSW_VOLUME_MAX >> 13, + HSW_VOLUME_MAX >> 12, + HSW_VOLUME_MAX >> 11, + HSW_VOLUME_MAX >> 10, + HSW_VOLUME_MAX >> 9, + HSW_VOLUME_MAX >> 8, + HSW_VOLUME_MAX >> 7, + HSW_VOLUME_MAX >> 6, + HSW_VOLUME_MAX >> 5, + HSW_VOLUME_MAX >> 4, + HSW_VOLUME_MAX >> 3, + HSW_VOLUME_MAX >> 2, + HSW_VOLUME_MAX >> 1, + HSW_VOLUME_MAX >> 0, +}; + +#define HSW_PCM_PERIODS_MAX 64 +#define HSW_PCM_PERIODS_MIN 2 + +static const struct snd_pcm_hardware hsw_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE, + .periods_min = HSW_PCM_PERIODS_MIN, + .periods_max = HSW_PCM_PERIODS_MAX, + .buffer_bytes_max = HSW_PCM_PERIODS_MAX * PAGE_SIZE, +}; + +/* private data for each PCM DSP stream */ +struct hsw_pcm_data { + int dai_id; + struct sst_hsw_stream *stream; + u32 volume[2]; + struct snd_pcm_substream *substream; + struct snd_compr_stream *cstream; + unsigned int wpos; + struct mutex mutex; +}; + +/* private data for the driver */ +struct hsw_priv_data { + /* runtime DSP */ + struct sst_hsw *hsw; + + /* page tables */ + unsigned char *pcm_pg[HSW_PCM_COUNT][2]; + + /* DAI data */ + struct hsw_pcm_data pcm[HSW_PCM_COUNT]; +}; + +static inline u32 hsw_mixer_to_ipc(unsigned int value) +{ + if (value >= ARRAY_SIZE(volume_map)) + return volume_map[0]; + else + return volume_map[value]; +} + +static inline unsigned int hsw_ipc_to_mixer(u32 value) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(volume_map); i++) { + if (volume_map[i] >= value) + return i; + } + + return i - 1; +} + +static int hsw_stream_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(platform); + struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct sst_hsw *hsw = pdata->hsw; + u32 volume; + + mutex_lock(&pcm_data->mutex); + + if (!pcm_data->stream) { + pcm_data->volume[0] = + hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + pcm_data->volume[1] = + hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + mutex_unlock(&pcm_data->mutex); + return 0; + } + + if (ucontrol->value.integer.value[0] == + ucontrol->value.integer.value[1]) { + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 2, volume); + } else { + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 0, volume); + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, 1, volume); + } + + mutex_unlock(&pcm_data->mutex); + return 0; +} + +static int hsw_stream_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(platform); + struct hsw_pcm_data *pcm_data = &pdata->pcm[mc->reg]; + struct sst_hsw *hsw = pdata->hsw; + u32 volume; + + mutex_lock(&pcm_data->mutex); + + if (!pcm_data->stream) { + ucontrol->value.integer.value[0] = + hsw_ipc_to_mixer(pcm_data->volume[0]); + ucontrol->value.integer.value[1] = + hsw_ipc_to_mixer(pcm_data->volume[1]); + mutex_unlock(&pcm_data->mutex); + return 0; + } + + sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 0, &volume); + ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume); + sst_hsw_stream_get_volume(hsw, pcm_data->stream, 0, 1, &volume); + ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume); + mutex_unlock(&pcm_data->mutex); + + return 0; +} + +static int hsw_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + u32 volume; + + if (ucontrol->value.integer.value[0] == + ucontrol->value.integer.value[1]) { + + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_mixer_set_volume(hsw, 0, 2, volume); + + } else { + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[0]); + sst_hsw_mixer_set_volume(hsw, 0, 0, volume); + + volume = hsw_mixer_to_ipc(ucontrol->value.integer.value[1]); + sst_hsw_mixer_set_volume(hsw, 0, 1, volume); + } + + return 0; +} + +static int hsw_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_platform *platform = snd_kcontrol_chip(kcontrol); + struct hsw_priv_data *pdata = snd_soc_platform_get_drvdata(platform); + struct sst_hsw *hsw = pdata->hsw; + unsigned int volume = 0; + + sst_hsw_mixer_get_volume(hsw, 0, 0, &volume); + ucontrol->value.integer.value[0] = hsw_ipc_to_mixer(volume); + + sst_hsw_mixer_get_volume(hsw, 0, 1, &volume); + ucontrol->value.integer.value[1] = hsw_ipc_to_mixer(volume); + + return 0; +} + +/* TLV used by both global and stream volumes */ +static const DECLARE_TLV_DB_SCALE(hsw_vol_tlv, -9000, 300, 1); + +/* System Pin has no volume control */ +static const struct snd_kcontrol_new hsw_volume_controls[] = { + /* Global DSP volume */ + SOC_DOUBLE_EXT_TLV("Master Playback Volume", 0, 0, 8, + ARRAY_SIZE(volume_map) -1, 0, + hsw_volume_get, hsw_volume_put, hsw_vol_tlv), + /* Offload 0 volume */ + SOC_DOUBLE_EXT_TLV("Media0 Playback Volume", 1, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* Offload 1 volume */ + SOC_DOUBLE_EXT_TLV("Media1 Playback Volume", 2, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* Loopback volume */ + SOC_DOUBLE_EXT_TLV("Loopback Capture Volume", 3, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), + /* Mic Capture volume */ + SOC_DOUBLE_EXT_TLV("Mic Capture Volume", 4, 0, 8, + ARRAY_SIZE(volume_map), 0, + hsw_stream_volume_get, hsw_stream_volume_put, hsw_vol_tlv), +}; + +/* Create DMA buffer page table for DSP */ +static int create_adsp_page_table(struct hsw_priv_data *pdata, + struct snd_soc_pcm_runtime *rtd, + unsigned char *dma_area, size_t size, int pcm, int stream) +{ + int i, pages; + + if (size % PAGE_SIZE) + pages = (size / PAGE_SIZE) + 1; + else + pages = size / PAGE_SIZE; + + dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dma_area, size, pages); + + for (i = 0; i < pages; i++) { + u32 idx = (((i << 2) + i)) >> 1; + u32 pfn = (virt_to_phys(dma_area + i * PAGE_SIZE)) >> PAGE_SHIFT; + u32 *pg_table; + + dev_dbg(rtd->dev, "pfn i %i idx %d pfn %x\n", i, idx, pfn); + + pg_table = (u32*)(pdata->pcm_pg[pcm][stream] + idx); + + if (i & 1) + *pg_table |= (pfn << 4); + else + *pg_table |= pfn; + } + + return 0; +} + +/* this may get called several times by oss emulation */ +static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + struct sst_module *module_data; + struct sst_dsp *dsp; + enum sst_hsw_stream_type stream_type; + enum sst_hsw_stream_path_id path_id; + u32 rate, bits, map, pages, module_id; + u8 channels; + int ret; + + /* stream direction */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + path_id = SST_HSW_STREAM_PATH_SSP0_OUT; + else + path_id = SST_HSW_STREAM_PATH_SSP0_IN; + + /* DSP stream type depends on DAI ID */ + switch (rtd->cpu_dai->id) { + case 0: + stream_type = SST_HSW_STREAM_TYPE_SYSTEM; + module_id = SST_HSW_MODULE_PCM_SYSTEM; + break; + case 1: + case 2: + stream_type = SST_HSW_STREAM_TYPE_RENDER; + module_id = SST_HSW_MODULE_PCM; + break; + case 3: + /* path ID needs to be OUT for loopback */ + stream_type = SST_HSW_STREAM_TYPE_LOOPBACK; + path_id = SST_HSW_STREAM_PATH_SSP0_OUT; + module_id = SST_HSW_MODULE_PCM_REFERENCE; + break; + case 4: + stream_type = SST_HSW_STREAM_TYPE_CAPTURE; + module_id = SST_HSW_MODULE_PCM_CAPTURE; + break; + default: + dev_err(rtd->dev, "error: invalid DAI ID %d\n", + rtd->cpu_dai->id); + return -EINVAL; + }; + + ret = sst_hsw_stream_format(hsw, pcm_data->stream, + path_id, stream_type, SST_HSW_STREAM_FORMAT_PCM_FORMAT); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set format %d\n", ret); + return ret; + } + + rate = params_rate(params); + ret = sst_hsw_stream_set_rate(hsw, pcm_data->stream, rate); + if (ret < 0) { + dev_err(rtd->dev, "error: could not set rate %d\n", rate); + return ret; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + bits = SST_HSW_DEPTH_16BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + bits = SST_HSW_DEPTH_24BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32); + break; + default: + dev_err(rtd->dev, "error: invalid format %d\n", + params_format(params)); + return -EINVAL; + } + + ret = sst_hsw_stream_set_bits(hsw, pcm_data->stream, bits); + if (ret < 0) { + dev_err(rtd->dev, "error: could not set bits %d\n", bits); + return ret; + } + + /* we only support stereo atm */ + channels = params_channels(params); + if (channels != 2) { + dev_err(rtd->dev, "error: invalid channels %d\n", channels); + return -EINVAL; + } + + map = create_channel_map(SST_HSW_CHANNEL_CONFIG_STEREO); + sst_hsw_stream_set_map_config(hsw, pcm_data->stream, + map, SST_HSW_CHANNEL_CONFIG_STEREO); + + ret = sst_hsw_stream_set_channels(hsw, pcm_data->stream, channels); + if (ret < 0) { + dev_err(rtd->dev, "error: could not set channels %d\n", + channels); + return ret; + } + + ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); + if (ret < 0) { + dev_err(rtd->dev, "error: could not allocate %d bytes for PCM %d\n", + params_buffer_bytes(params), ret); + return ret; + } + + ret = create_adsp_page_table(pdata, rtd, runtime->dma_area, + runtime->dma_bytes, rtd->cpu_dai->id, substream->stream); + if (ret < 0) + return ret; + + sst_hsw_stream_set_style(hsw, pcm_data->stream, + SST_HSW_INTERLEAVING_PER_CHANNEL); + + if (runtime->dma_bytes % PAGE_SIZE) + pages = (runtime->dma_bytes / PAGE_SIZE) + 1; + else + pages = runtime->dma_bytes / PAGE_SIZE; + + ret = sst_hsw_stream_buffer(hsw, pcm_data->stream, + virt_to_phys(pdata->pcm_pg[rtd->cpu_dai->id][substream->stream]), + pages, runtime->dma_bytes, 0, + (u32)(virt_to_phys(runtime->dma_area) >> PAGE_SHIFT)); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set DMA buffer %d\n", ret); + return ret; + } + + dsp = sst_hsw_get_dsp(hsw); + + module_data = sst_module_get_from_id(dsp, module_id); + if (module_data == NULL) { + dev_err(rtd->dev, "error: failed to get module config\n"); + return -EINVAL; + } + + /* we use hardcoded memory offsets atm, will be updated for new FW */ + if (stream_type == SST_HSW_STREAM_TYPE_CAPTURE) { + sst_hsw_stream_set_module_info(hsw, pcm_data->stream, + SST_HSW_MODULE_PCM_CAPTURE, module_data->entry); + sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, + 0x449400, 0x4000); + sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, + 0x400000, 0); + } else { /* stream_type == SST_HSW_STREAM_TYPE_SYSTEM */ + sst_hsw_stream_set_module_info(hsw, pcm_data->stream, + SST_HSW_MODULE_PCM_SYSTEM, module_data->entry); + + sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, + module_data->offset, module_data->size); + sst_hsw_stream_set_pmemory_info(hsw, pcm_data->stream, + 0x44d400, 0x3800); + + sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, + module_data->offset, module_data->size); + sst_hsw_stream_set_smemory_info(hsw, pcm_data->stream, + 0x400000, 0); + } + + ret = sst_hsw_stream_commit(hsw, pcm_data->stream); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to commit stream %d\n", ret); + return ret; + } + + ret = sst_hsw_stream_pause(hsw, pcm_data->stream, 1); + if (ret < 0) + dev_err(rtd->dev, "error: failed to pause %d\n", ret); + + return 0; +} + +static int hsw_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int hsw_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_hsw_stream_resume(hsw, pcm_data->stream, 0); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_hsw_stream_pause(hsw, pcm_data->stream, 0); + break; + default: + break; + } + + return 0; +} + +static u32 hsw_notify_pointer(struct sst_hsw_stream *stream, void *data) +{ + struct hsw_pcm_data *pcm_data = data; + struct snd_pcm_substream *substream = pcm_data->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u32 pos; + + pos = frames_to_bytes(runtime, + (runtime->control->appl_ptr % runtime->buffer_size)); + + dev_dbg(rtd->dev, "PCM: App pointer %d bytes\n", pos); + + /* let alsa know we have play a period */ + snd_pcm_period_elapsed(substream); + return pos; +} + +static snd_pcm_uframes_t hsw_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + snd_pcm_uframes_t offset; + + offset = bytes_to_frames(runtime, + sst_hsw_get_dsp_position(hsw, pcm_data->stream)); + + dev_dbg(rtd->dev, "PCM: DMA pointer %zu bytes\n", + frames_to_bytes(runtime, (u32)offset)); + return offset; +} + +static int hsw_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data; + struct sst_hsw *hsw = pdata->hsw; + + pcm_data = &pdata->pcm[rtd->cpu_dai->id]; + + mutex_lock(&pcm_data->mutex); + + snd_soc_pcm_set_drvdata(rtd, pcm_data); + pcm_data->substream = substream; + + snd_soc_set_runtime_hwparams(substream, &hsw_pcm_hardware); + + pcm_data->stream = sst_hsw_stream_new(hsw, rtd->cpu_dai->id, + hsw_notify_pointer, pcm_data); + if (pcm_data->stream == NULL) { + dev_err(rtd->dev, "error: failed to create stream\n"); + mutex_unlock(&pcm_data->mutex); + return -EINVAL; + } + + /* Set previous saved volume */ + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 0, pcm_data->volume[0]); + sst_hsw_stream_set_volume(hsw, pcm_data->stream, 0, + 1, pcm_data->volume[1]); + + mutex_unlock(&pcm_data->mutex); + return 0; +} + +static int hsw_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hsw_priv_data *pdata = + snd_soc_platform_get_drvdata(rtd->platform); + struct hsw_pcm_data *pcm_data = snd_soc_pcm_get_drvdata(rtd); + struct sst_hsw *hsw = pdata->hsw; + int ret; + + mutex_lock(&pcm_data->mutex); + ret = sst_hsw_stream_reset(hsw, pcm_data->stream); + if (ret < 0) { + dev_dbg(rtd->dev, "error: reset stream failed %d\n", ret); + goto out; + } + + ret = sst_hsw_stream_free(hsw, pcm_data->stream); + if (ret < 0) { + dev_dbg(rtd->dev, "error: free stream failed %d\n", ret); + goto out; + } + pcm_data->stream = NULL; + +out: + mutex_unlock(&pcm_data->mutex); + return ret; +} + +static struct snd_pcm_ops hsw_pcm_ops = { + .open = hsw_pcm_open, + .close = hsw_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hsw_pcm_hw_params, + .hw_free = hsw_pcm_hw_free, + .trigger = hsw_pcm_trigger, + .pointer = hsw_pcm_pointer, + .mmap = snd_pcm_lib_default_mmap, +}; + +static void hsw_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + ret = dma_coerce_mask_and_coherent(rtd->card->dev, DMA_BIT_MASK(32)); + if (ret) + return ret; + + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream || + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { + ret = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + rtd->card->dev, + hsw_pcm_hardware.buffer_bytes_max, + hsw_pcm_hardware.buffer_bytes_max); + if (ret) { + dev_err(rtd->dev, "dma buffer allocation failed %d\n", + ret); + return ret; + } + } + + return ret; +} + +#define HSW_FORMATS \ + (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver hsw_dais[] = { + { + .name = "System Pin", + .playback = { + .stream_name = "System Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + }, + { + /* PCM */ + .name = "Offload0 Pin", + .playback = { + .stream_name = "Offload0 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, + { + /* PCM */ + .name = "Offload1 Pin", + .playback = { + .stream_name = "Offload1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, + { + .name = "Loopback Pin", + .capture = { + .stream_name = "Loopback Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, + { + .name = "Capture Pin", + .capture = { + .stream_name = "Analog Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = HSW_FORMATS, + }, + }, +}; + +static const struct snd_soc_dapm_widget widgets[] = { + + /* Backend DAIs */ + SND_SOC_DAPM_AIF_IN("SSP0 CODEC IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SSP0 CODEC OUT", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SSP1 BT IN", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SSP1 BT OUT", NULL, 0, SND_SOC_NOPM, 0, 0), + + /* Global Playback Mixer */ + SND_SOC_DAPM_MIXER("Playback VMixer", SND_SOC_NOPM, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route graph[] = { + + /* Playback Mixer */ + {"Playback VMixer", NULL, "System Playback"}, + {"Playback VMixer", NULL, "Offload0 Playback"}, + {"Playback VMixer", NULL, "Offload1 Playback"}, + + {"SSP0 CODEC OUT", NULL, "Playback VMixer"}, + + {"Analog Capture", NULL, "SSP0 CODEC IN"}, +}; + +static int hsw_pcm_probe(struct snd_soc_platform *platform) +{ + struct sst_pdata *pdata = dev_get_platdata(platform->dev); + struct hsw_priv_data *priv_data; + int i; + + if (!pdata) + return -ENODEV; + + priv_data = devm_kzalloc(platform->dev, sizeof(*priv_data), GFP_KERNEL); + priv_data->hsw = pdata->dsp; + snd_soc_platform_set_drvdata(platform, priv_data); + + /* allocate DSP buffer page tables */ + for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { + + mutex_init(&priv_data->pcm[i].mutex); + + /* playback */ + if (hsw_dais[i].playback.channels_min) { + priv_data->pcm_pg[i][0] = kzalloc(PAGE_SIZE, GFP_DMA); + if (priv_data->pcm_pg[i][0] == NULL) + goto err; + } + + /* capture */ + if (hsw_dais[i].capture.channels_min) { + priv_data->pcm_pg[i][1] = kzalloc(PAGE_SIZE, GFP_DMA); + if (priv_data->pcm_pg[i][1] == NULL) + goto err; + } + } + + return 0; + +err: + for (;i >= 0; i--) { + if (hsw_dais[i].playback.channels_min) + kfree(priv_data->pcm_pg[i][0]); + if (hsw_dais[i].capture.channels_min) + kfree(priv_data->pcm_pg[i][1]); + } + return -ENOMEM; +} + +static int hsw_pcm_remove(struct snd_soc_platform *platform) +{ + struct hsw_priv_data *priv_data = + snd_soc_platform_get_drvdata(platform); + int i; + + for (i = 0; i < ARRAY_SIZE(hsw_dais); i++) { + if (hsw_dais[i].playback.channels_min) + kfree(priv_data->pcm_pg[i][0]); + if (hsw_dais[i].capture.channels_min) + kfree(priv_data->pcm_pg[i][1]); + } + + return 0; +} + +static struct snd_soc_platform_driver hsw_soc_platform = { + .probe = hsw_pcm_probe, + .remove = hsw_pcm_remove, + .ops = &hsw_pcm_ops, + .pcm_new = hsw_pcm_new, + .pcm_free = hsw_pcm_free, + .controls = hsw_volume_controls, + .num_controls = ARRAY_SIZE(hsw_volume_controls), + .dapm_widgets = widgets, + .num_dapm_widgets = ARRAY_SIZE(widgets), + .dapm_routes = graph, + .num_dapm_routes = ARRAY_SIZE(graph), +}; + +static const struct snd_soc_component_driver hsw_dai_component = { + .name = "haswell-dai", +}; + +static int hsw_pcm_dev_probe(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + int ret; + + ret = sst_hsw_dsp_init(&pdev->dev, sst_pdata); + if (ret < 0) + return -ENODEV; + + ret = snd_soc_register_platform(&pdev->dev, &hsw_soc_platform); + if (ret < 0) + goto err_plat; + + ret = snd_soc_register_component(&pdev->dev, &hsw_dai_component, + hsw_dais, ARRAY_SIZE(hsw_dais)); + if (ret < 0) + goto err_comp; + + return 0; + +err_comp: + snd_soc_unregister_platform(&pdev->dev); +err_plat: + sst_hsw_dsp_free(&pdev->dev, sst_pdata); + return 0; +} + +static int hsw_pcm_dev_remove(struct platform_device *pdev) +{ + struct sst_pdata *sst_pdata = dev_get_platdata(&pdev->dev); + + snd_soc_unregister_platform(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); + sst_hsw_dsp_free(&pdev->dev, sst_pdata); + + return 0; +} + +static struct platform_driver hsw_pcm_driver = { + .driver = { + .name = "haswell-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = hsw_pcm_dev_probe, + .remove = hsw_pcm_dev_remove, +}; +module_platform_driver(hsw_pcm_driver); + +MODULE_AUTHOR("Liam Girdwood, Xingchao Wang"); +MODULE_DESCRIPTION("Haswell/Lynxpoint + Broadwell/Wildcatpoint PCM"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:haswell-pcm-audio"); diff --git a/sound/soc/intel/sst_dsp.h b/sound/soc/intel/sst-mfld-dsp.h index 0fce1de284ff..3b63edc04b7f 100644 --- a/sound/soc/intel/sst_dsp.h +++ b/sound/soc/intel/sst-mfld-dsp.h @@ -1,7 +1,7 @@ -#ifndef __SST_DSP_H__ -#define __SST_DSP_H__ +#ifndef __SST_MFLD_DSP_H__ +#define __SST_MFLD_DSP_H__ /* - * sst_dsp.h - Intel SST Driver for audio engine + * sst_mfld_dsp.h - Intel SST Driver for audio engine * * Copyright (C) 2008-12 Intel Corporation * Authors: Vinod Koul <vinod.koul@linux.intel.com> @@ -131,4 +131,4 @@ struct snd_sst_params { struct snd_sst_alloc_params_ext aparams; }; -#endif /* __SST_DSP_H__ */ +#endif /* __SST_MFLD_DSP_H__ */ diff --git a/sound/soc/intel/sst_platform.c b/sound/soc/intel/sst-mfld-platform.c index f465a8180863..840306c2ef14 100644 --- a/sound/soc/intel/sst_platform.c +++ b/sound/soc/intel/sst-mfld-platform.c @@ -1,5 +1,5 @@ /* - * sst_platform.c - Intel MID Platform driver + * sst_mfld_platform.c - Intel MID Platform driver * * Copyright (C) 2010-2013 Intel Corp * Author: Vinod Koul <vinod.koul@intel.com> @@ -33,7 +33,7 @@ #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/compress_driver.h> -#include "sst_platform.h" +#include "sst-mfld-platform.h" static struct sst_device *sst; static DEFINE_MUTEX(sst_lock); @@ -709,7 +709,7 @@ static int sst_platform_remove(struct platform_device *pdev) static struct platform_driver sst_platform_driver = { .driver = { - .name = "sst-platform", + .name = "sst-mfld-platform", .owner = THIS_MODULE, }, .probe = sst_platform_probe, @@ -722,4 +722,4 @@ MODULE_DESCRIPTION("ASoC Intel(R) MID Platform driver"); MODULE_AUTHOR("Vinod Koul <vinod.koul@intel.com>"); MODULE_AUTHOR("Harsha Priya <priya.harsha@intel.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:sst-platform"); +MODULE_ALIAS("platform:sst-mfld-platform"); diff --git a/sound/soc/intel/sst_platform.h b/sound/soc/intel/sst-mfld-platform.h index bee64fb7d2ef..0c4e2ddcecb1 100644 --- a/sound/soc/intel/sst_platform.h +++ b/sound/soc/intel/sst-mfld-platform.h @@ -1,5 +1,5 @@ /* - * sst_platform.h - Intel MID Platform driver header file + * sst_mfld_platform.h - Intel MID Platform driver header file * * Copyright (C) 2010 Intel Corp * Author: Vinod Koul <vinod.koul@intel.com> @@ -27,7 +27,7 @@ #ifndef __SST_PLATFORMDRV_H__ #define __SST_PLATFORMDRV_H__ -#include "sst_dsp.h" +#include "sst-mfld-dsp.h" #define SST_MONO 1 #define SST_STEREO 2 diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 78ed4a42ad21..49f8437665de 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,11 +1,20 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" - depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST + depends on ARCH_KIRKWOOD || ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the audio interfaces to support below. +config SND_KIRKWOOD_SOC_ARMADA370_DB + tristate "SoC Audio support for Armada 370 DB" + depends on SND_KIRKWOOD_SOC && (ARCH_MVEBU || COMPILE_TEST) && I2C + select SND_SOC_CS42L51 + select SND_SOC_SPDIF + help + Say Y if you want to add support for SoC audio on + the Armada 370 Development Board. + config SND_KIRKWOOD_SOC_OPENRD tristate "SoC Audio support for Kirkwood Openrd Client" depends on SND_KIRKWOOD_SOC && (MACH_OPENRD_CLIENT || MACH_OPENRD_ULTIMATE || COMPILE_TEST) diff --git a/sound/soc/kirkwood/Makefile b/sound/soc/kirkwood/Makefile index 9e781385cb88..7c1d8fe09e6b 100644 --- a/sound/soc/kirkwood/Makefile +++ b/sound/soc/kirkwood/Makefile @@ -4,6 +4,8 @@ obj-$(CONFIG_SND_KIRKWOOD_SOC) += snd-soc-kirkwood.o snd-soc-openrd-objs := kirkwood-openrd.o snd-soc-t5325-objs := kirkwood-t5325.o +snd-soc-armada-370-db-objs := armada-370-db.o +obj-$(CONFIG_SND_KIRKWOOD_SOC_ARMADA370_DB) += snd-soc-armada-370-db.o obj-$(CONFIG_SND_KIRKWOOD_SOC_OPENRD) += snd-soc-openrd.o obj-$(CONFIG_SND_KIRKWOOD_SOC_T5325) += snd-soc-t5325.o diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c new file mode 100644 index 000000000000..c44333849259 --- /dev/null +++ b/sound/soc/kirkwood/armada-370-db.c @@ -0,0 +1,148 @@ +/* + * Copyright (C) 2014 Marvell + * + * Thomas Petazzoni <thomas.petazzoni@free-electrons.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License as + * published by the Free Software Foundation; either version 2 of the + * License, or (at your option) any later version. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <sound/soc.h> +#include <linux/of.h> +#include <linux/platform_data/asoc-kirkwood.h> +#include "../codecs/cs42l51.h" + +static int a370db_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int freq; + + switch (params_rate(params)) { + default: + case 44100: + freq = 11289600; + break; + case 48000: + freq = 12288000; + break; + case 96000: + freq = 24576000; + break; + } + + return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); +} + +static struct snd_soc_ops a370db_ops = { + .hw_params = a370db_hw_params, +}; + +static const struct snd_soc_dapm_widget a370db_dapm_widgets[] = { + SND_SOC_DAPM_HP("Out Jack", NULL), + SND_SOC_DAPM_LINE("In Jack", NULL), +}; + +static const struct snd_soc_dapm_route a370db_route[] = { + { "Out Jack", NULL, "HPL" }, + { "Out Jack", NULL, "HPR" }, + { "AIN1L", NULL, "In Jack" }, + { "AIN1L", NULL, "In Jack" }, +}; + +static struct snd_soc_dai_link a370db_dai[] = { +{ + .name = "CS42L51", + .stream_name = "analog", + .cpu_dai_name = "i2s", + .codec_dai_name = "cs42l51-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, + .ops = &a370db_ops, +}, +{ + .name = "S/PDIF out", + .stream_name = "spdif-out", + .cpu_dai_name = "spdif", + .codec_dai_name = "dit-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, +}, +{ + .name = "S/PDIF in", + .stream_name = "spdif-in", + .cpu_dai_name = "spdif", + .codec_dai_name = "dir-hifi", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBS_CFS, +}, +}; + +static struct snd_soc_card a370db = { + .name = "a370db", + .owner = THIS_MODULE, + .dai_link = a370db_dai, + .num_links = ARRAY_SIZE(a370db_dai), + .dapm_widgets = a370db_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(a370db_dapm_widgets), + .dapm_routes = a370db_route, + .num_dapm_routes = ARRAY_SIZE(a370db_route), +}; + +static int a370db_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &a370db; + + card->dev = &pdev->dev; + + a370db_dai[0].cpu_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-controller", 0); + a370db_dai[0].platform_of_node = a370db_dai[0].cpu_of_node; + + a370db_dai[0].codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-codec", 0); + + a370db_dai[1].cpu_of_node = a370db_dai[0].cpu_of_node; + a370db_dai[1].platform_of_node = a370db_dai[0].cpu_of_node; + + a370db_dai[1].codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-codec", 1); + + a370db_dai[2].cpu_of_node = a370db_dai[0].cpu_of_node; + a370db_dai[2].platform_of_node = a370db_dai[0].cpu_of_node; + + a370db_dai[2].codec_of_node = + of_parse_phandle(pdev->dev.of_node, + "marvell,audio-codec", 2); + + return devm_snd_soc_register_card(card->dev, card); +} + +static const struct of_device_id a370db_dt_ids[] = { + { .compatible = "marvell,a370db-audio" }, + { }, +}; + +static struct platform_driver a370db_driver = { + .driver = { + .name = "a370db-audio", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(a370db_dt_ids), + }, + .probe = a370db_probe, +}; + +module_platform_driver(a370db_driver); + +MODULE_AUTHOR("Thomas Petazzoni <thomas.petazzoni@free-electrons.com>"); +MODULE_DESCRIPTION("ALSA SoC a370db audio client"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:a370db-audio"); diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 3920a5e8125f..9f842222e798 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -633,6 +633,7 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev) static struct of_device_id mvebu_audio_of_match[] = { { .compatible = "marvell,kirkwood-audio" }, { .compatible = "marvell,dove-audio" }, + { .compatible = "marvell,armada370-audio" }, { } }; MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 22ad9c5654b5..e00659351a4e 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -58,7 +58,7 @@ config SND_OMAP_SOC_OSK5912 tristate "SoC Audio support for omap osk5912" depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23 + select SND_SOC_TLV320AIC23_I2C help Say Y if you want to add support for SoC audio on osk5912. @@ -66,7 +66,7 @@ config SND_OMAP_SOC_AM3517EVM tristate "SoC Audio support for OMAP3517 / AM3517 EVM" depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C select SND_OMAP_SOC_MCBSP - select SND_SOC_TLV320AIC23 + select SND_SOC_TLV320AIC23_I2C help Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517 EVM. diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 629446482a91..56a5219c0a00 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -103,60 +103,62 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, if (!codec->hw_write) return -EUNATCH; - if (ucontrol->value.enumerated.item[0] >= control->max) + if (ucontrol->value.enumerated.item[0] >= control->items) return -EINVAL; - mutex_lock(&codec->mutex); + snd_soc_dapm_mutex_lock(dapm); /* Translate selection to bitmap */ pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]]; /* Setup pins after corresponding bits if changed */ pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE)); + if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(dapm, "Mouthpiece"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece"); else - snd_soc_dapm_disable_pin(dapm, "Mouthpiece"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece"); } pin = !!(pins & (1 << AMS_DELTA_EARPIECE)); if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(dapm, "Earpiece"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece"); else - snd_soc_dapm_disable_pin(dapm, "Earpiece"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece"); } pin = !!(pins & (1 << AMS_DELTA_MICROPHONE)); if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(dapm, "Microphone"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone"); else - snd_soc_dapm_disable_pin(dapm, "Microphone"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone"); } pin = !!(pins & (1 << AMS_DELTA_SPEAKER)); if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) { changed = 1; if (pin) - snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker"); else - snd_soc_dapm_disable_pin(dapm, "Speaker"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker"); } pin = !!(pins & (1 << AMS_DELTA_AGC)); if (pin != ams_delta_audio_agc) { ams_delta_audio_agc = pin; changed = 1; if (pin) - snd_soc_dapm_enable_pin(dapm, "AGCIN"); + snd_soc_dapm_enable_pin_unlocked(dapm, "AGCIN"); else - snd_soc_dapm_disable_pin(dapm, "AGCIN"); + snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN"); } + if (changed) - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync_unlocked(dapm); - mutex_unlock(&codec->mutex); + snd_soc_dapm_mutex_unlock(dapm); return changed; } @@ -194,13 +196,11 @@ static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol, return 0; } -static const struct soc_enum ams_delta_audio_enum[] = { - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode), - ams_delta_audio_mode), -}; +static const SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum, + ams_delta_audio_mode); static const struct snd_kcontrol_new ams_delta_audio_controls[] = { - SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0], + SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum, ams_delta_get_audio_mode, ams_delta_set_audio_mode), }; @@ -315,12 +315,17 @@ static void cx81801_close(struct tty_struct *tty) v253_ops.close(tty); /* Revert back to default audio input/output constellation */ - snd_soc_dapm_disable_pin(dapm, "Mouthpiece"); - snd_soc_dapm_enable_pin(dapm, "Earpiece"); - snd_soc_dapm_enable_pin(dapm, "Microphone"); - snd_soc_dapm_disable_pin(dapm, "Speaker"); - snd_soc_dapm_disable_pin(dapm, "AGCIN"); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker"); + snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN"); + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } /* Line discipline .hangup() */ diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 3fde9e402710..fd4d9c809e50 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -68,26 +68,30 @@ static void n810_ext_control(struct snd_soc_dapm_context *dapm) break; } + snd_soc_dapm_mutex_lock(dapm); + if (n810_spk_func) - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(dapm, "Ext Spk"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk"); if (hp) - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); else - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); if (line1l) - snd_soc_dapm_enable_pin(dapm, "LINE1L"); + snd_soc_dapm_enable_pin_unlocked(dapm, "LINE1L"); else - snd_soc_dapm_disable_pin(dapm, "LINE1L"); + snd_soc_dapm_disable_pin_unlocked(dapm, "LINE1L"); if (n810_dmic_func) - snd_soc_dapm_enable_pin(dapm, "DMic"); + snd_soc_dapm_enable_pin_unlocked(dapm, "DMic"); else - snd_soc_dapm_disable_pin(dapm, "DMic"); + snd_soc_dapm_disable_pin_unlocked(dapm, "DMic"); + + snd_soc_dapm_sync_unlocked(dapm); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_mutex_unlock(dapm); } static int n810_startup(struct snd_pcm_substream *substream) @@ -305,7 +309,9 @@ static int __init n810_soc_init(void) int err; struct device *dev; - if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax())) + if (!of_have_populated_dt() || + (!of_machine_is_compatible("nokia,n810") && + !of_machine_is_compatible("nokia,n810-wimax"))) return -ENODEV; n810_snd_device = platform_device_alloc("soc-audio", -1); diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index ebb13906b3a0..024dafc3e298 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -203,8 +203,7 @@ static const struct snd_soc_dapm_route dmic_audio_map[] = { static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; return snd_soc_dapm_add_routes(dapm, dmic_audio_map, ARRAY_SIZE(dmic_audio_map)); diff --git a/sound/soc/omap/rx51.c b/sound/soc/omap/rx51.c index 611179c3bca4..7fb3d4b10370 100644 --- a/sound/soc/omap/rx51.c +++ b/sound/soc/omap/rx51.c @@ -74,26 +74,30 @@ static void rx51_ext_control(struct snd_soc_dapm_context *dapm) break; } + snd_soc_dapm_mutex_lock(dapm); + if (rx51_spk_func) - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(dapm, "Ext Spk"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk"); if (rx51_dmic_func) - snd_soc_dapm_enable_pin(dapm, "DMic"); + snd_soc_dapm_enable_pin_unlocked(dapm, "DMic"); else - snd_soc_dapm_disable_pin(dapm, "DMic"); + snd_soc_dapm_disable_pin_unlocked(dapm, "DMic"); if (hp) - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); else - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); if (hs) - snd_soc_dapm_enable_pin(dapm, "HS Mic"); + snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic"); else - snd_soc_dapm_disable_pin(dapm, "HS Mic"); + snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic"); gpio_set_value(RX51_TVOUT_SEL_GPIO, tvout); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } static int rx51_startup(struct snd_pcm_substream *substream) diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 1853d41034bf..5a88136aa800 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -47,64 +47,63 @@ static int corgi_spk_func; static void corgi_ext_control(struct snd_soc_dapm_context *dapm) { + snd_soc_dapm_mutex_lock(dapm); + /* set up jack connection */ switch (corgi_jack_func) { case CORGI_HP: /* set = unmute headphone */ gpio_set_value(CORGI_GPIO_MUTE_L, 1); gpio_set_value(CORGI_GPIO_MUTE_R, 1); - snd_soc_dapm_disable_pin(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); break; case CORGI_MIC: /* reset = mute headphone */ gpio_set_value(CORGI_GPIO_MUTE_L, 0); gpio_set_value(CORGI_GPIO_MUTE_R, 0); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); break; case CORGI_LINE: gpio_set_value(CORGI_GPIO_MUTE_L, 0); gpio_set_value(CORGI_GPIO_MUTE_R, 0); - snd_soc_dapm_disable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Jack"); - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); break; case CORGI_HEADSET: gpio_set_value(CORGI_GPIO_MUTE_L, 0); gpio_set_value(CORGI_GPIO_MUTE_R, 1); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Headset Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); break; } if (corgi_spk_func == CORGI_SPK_ON) - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(dapm, "Ext Spk"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk"); /* signal a DAPM event */ - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } static int corgi_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - corgi_ext_control(&codec->dapm); - - mutex_unlock(&codec->mutex); + corgi_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index 44b5c09d296b..c29fedab2f49 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -103,11 +103,6 @@ static int e740_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "PCBEEP"); snd_soc_dapm_nc_pin(dapm, "MIC2"); - snd_soc_dapm_new_controls(dapm, e740_dapm_widgets, - ARRAY_SIZE(e740_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -136,6 +131,11 @@ static struct snd_soc_card e740 = { .owner = THIS_MODULE, .dai_link = e740_dai, .num_links = ARRAY_SIZE(e740_dai), + + .dapm_widgets = e740_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(e740_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct gpio e740_audio_gpios[] = { diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index c34e447eb991..ee36aba88063 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -85,11 +85,6 @@ static int e750_ac97_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "PCBEEP"); snd_soc_dapm_nc_pin(dapm, "MIC2"); - snd_soc_dapm_new_controls(dapm, e750_dapm_widgets, - ARRAY_SIZE(e750_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -119,6 +114,11 @@ static struct snd_soc_card e750 = { .owner = THIS_MODULE, .dai_link = e750_dai, .num_links = ARRAY_SIZE(e750_dai), + + .dapm_widgets = e750_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(e750_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct gpio e750_audio_gpios[] = { diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 3137f800b43f..24c2078ce70b 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -71,19 +71,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"MIC2", NULL, "Mic (Internal2)"}, }; -static int e800_ac97_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, e800_dapm_widgets, - ARRAY_SIZE(e800_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - static struct snd_soc_dai_link e800_dai[] = { { .name = "AC97", @@ -92,7 +79,6 @@ static struct snd_soc_dai_link e800_dai[] = { .codec_dai_name = "wm9712-hifi", .platform_name = "pxa-pcm-audio", .codec_name = "wm9712-codec", - .init = e800_ac97_init, }, { .name = "AC97 Aux", @@ -109,6 +95,11 @@ static struct snd_soc_card e800 = { .owner = THIS_MODULE, .dai_link = e800_dai, .num_links = ARRAY_SIZE(e800_dai), + + .dapm_widgets = e800_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(e800_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct gpio e800_audio_gpios[] = { diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index aace19e0fe2c..259e048681c0 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -41,44 +41,42 @@ static int magician_hp_switch; static int magician_spk_switch = 1; static int magician_in_sel = MAGICIAN_MIC; -static void magician_ext_control(struct snd_soc_codec *codec) +static void magician_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_mutex_lock(dapm); if (magician_spk_switch) - snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker"); else - snd_soc_dapm_disable_pin(dapm, "Speaker"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker"); if (magician_hp_switch) - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); else - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); switch (magician_in_sel) { case MAGICIAN_MIC: - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_enable_pin(dapm, "Call Mic"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Call Mic"); break; case MAGICIAN_MIC_EXT: - snd_soc_dapm_disable_pin(dapm, "Call Mic"); - snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Call Mic"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Mic"); break; } - snd_soc_dapm_sync(dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } static int magician_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - magician_ext_control(codec); - - mutex_unlock(&codec->mutex); + magician_ext_control(&rtd->card->dapm); return 0; } @@ -277,13 +275,13 @@ static int magician_get_hp(struct snd_kcontrol *kcontrol, static int magician_set_hp(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (magician_hp_switch == ucontrol->value.integer.value[0]) return 0; magician_hp_switch = ucontrol->value.integer.value[0]; - magician_ext_control(codec); + magician_ext_control(&card->dapm); return 1; } @@ -297,13 +295,13 @@ static int magician_get_spk(struct snd_kcontrol *kcontrol, static int magician_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (magician_spk_switch == ucontrol->value.integer.value[0]) return 0; magician_spk_switch = ucontrol->value.integer.value[0]; - magician_ext_control(codec); + magician_ext_control(&card->dapm); return 1; } @@ -400,7 +398,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; /* NC codec pins */ snd_soc_dapm_nc_pin(dapm, "VOUTLHP"); @@ -410,19 +407,6 @@ static int magician_uda1380_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "VINL"); snd_soc_dapm_nc_pin(dapm, "VINR"); - /* Add magician specific controls */ - err = snd_soc_add_codec_controls(codec, uda1380_magician_controls, - ARRAY_SIZE(uda1380_magician_controls)); - if (err < 0) - return err; - - /* Add magician specific widgets */ - snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, - ARRAY_SIZE(uda1380_dapm_widgets)); - - /* Set up magician specific audio path interconnects */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -456,6 +440,12 @@ static struct snd_soc_card snd_soc_card_magician = { .dai_link = magician_dai, .num_links = ARRAY_SIZE(magician_dai), + .controls = uda1380_magician_controls, + .num_controls = ARRAY_SIZE(uda1380_magician_controls), + .dapm_widgets = uda1380_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *magician_snd_device; diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 160c5245448f..595eee341e90 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -127,16 +127,8 @@ static const struct snd_soc_dapm_route audio_map[] = { static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; unsigned short reg; - /* Add mioa701 specific widgets */ - snd_soc_dapm_new_controls(dapm, mioa701_dapm_widgets, - ARRAY_SIZE(mioa701_dapm_widgets)); - - /* Set up mioa701 specific audio path audio_mapnects */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - /* Prepare GPIO8 for rear speaker amplifier */ reg = codec->driver->read(codec, AC97_GPIO_CFG); codec->driver->write(codec, AC97_GPIO_CFG, reg | 0x0100); @@ -145,12 +137,6 @@ static int mioa701_wm9713_init(struct snd_soc_pcm_runtime *rtd) reg = codec->driver->read(codec, AC97_3D_CONTROL); codec->driver->write(codec, AC97_3D_CONTROL, reg | 0xc000); - snd_soc_dapm_enable_pin(dapm, "Front Speaker"); - snd_soc_dapm_enable_pin(dapm, "Rear Speaker"); - snd_soc_dapm_enable_pin(dapm, "Front Mic"); - snd_soc_dapm_enable_pin(dapm, "GSM Line In"); - snd_soc_dapm_enable_pin(dapm, "GSM Line Out"); - return 0; } @@ -183,6 +169,11 @@ static struct snd_soc_card mioa701 = { .owner = THIS_MODULE, .dai_link = mioa701_dai, .num_links = ARRAY_SIZE(mioa701_dai), + + .dapm_widgets = mioa701_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mioa701_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int mioa701_wm9713_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index c93e138d8dc3..c6bdc6c0eff6 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -74,14 +74,9 @@ static void poodle_ext_control(struct snd_soc_dapm_context *dapm) static int poodle_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - poodle_ext_control(&codec->dapm); - - mutex_unlock(&codec->mutex); + poodle_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index fc052d8247ff..1373b017a951 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -46,74 +46,74 @@ static int spitz_mic_gpio; static void spitz_ext_control(struct snd_soc_dapm_context *dapm) { + snd_soc_dapm_mutex_lock(dapm); + if (spitz_spk_func == SPITZ_SPK_ON) - snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk"); else - snd_soc_dapm_disable_pin(dapm, "Ext Spk"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk"); /* set up jack connection */ switch (spitz_jack_func) { case SPITZ_HP: /* enable and unmute hp jack, disable mic bias */ - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); - snd_soc_dapm_disable_pin(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 1); gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_MIC: /* enable mic jack and bias, mute hp */ - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_LINE: /* enable line jack, disable mic bias and mute hp */ - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); - snd_soc_dapm_disable_pin(dapm, "Mic Jack"); - snd_soc_dapm_enable_pin(dapm, "Line Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Line Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; case SPITZ_HEADSET: /* enable and unmute headset jack enable mic bias, mute L hp */ - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); - snd_soc_dapm_enable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 1); break; case SPITZ_HP_OFF: /* jack removed, everything off */ - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); - snd_soc_dapm_disable_pin(dapm, "Mic Jack"); - snd_soc_dapm_disable_pin(dapm, "Line Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Line Jack"); gpio_set_value(SPITZ_GPIO_MUTE_L, 0); gpio_set_value(SPITZ_GPIO_MUTE_R, 0); break; } - snd_soc_dapm_sync(dapm); + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } static int spitz_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - spitz_ext_control(&codec->dapm); - - mutex_unlock(&codec->mutex); + spitz_ext_control(&rtd->card->dapm); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 1d9c2ed223bc..4a956d1cb269 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -44,48 +44,46 @@ static int tosa_jack_func; static int tosa_spk_func; -static void tosa_ext_control(struct snd_soc_codec *codec) +static void tosa_ext_control(struct snd_soc_dapm_context *dapm) { - struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_mutex_lock(dapm); /* set up jack connection */ switch (tosa_jack_func) { case TOSA_HP: - snd_soc_dapm_disable_pin(dapm, "Mic (Internal)"); - snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); break; case TOSA_MIC_INT: - snd_soc_dapm_enable_pin(dapm, "Mic (Internal)"); - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_disable_pin(dapm, "Headset Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Mic (Internal)"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headset Jack"); break; case TOSA_HEADSET: - snd_soc_dapm_disable_pin(dapm, "Mic (Internal)"); - snd_soc_dapm_disable_pin(dapm, "Headphone Jack"); - snd_soc_dapm_enable_pin(dapm, "Headset Jack"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Mic (Internal)"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Headset Jack"); break; } if (tosa_spk_func == TOSA_SPK_ON) - snd_soc_dapm_enable_pin(dapm, "Speaker"); + snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker"); else - snd_soc_dapm_disable_pin(dapm, "Speaker"); + snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker"); + + snd_soc_dapm_sync_unlocked(dapm); - snd_soc_dapm_sync(dapm); + snd_soc_dapm_mutex_unlock(dapm); } static int tosa_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; - - mutex_lock(&codec->mutex); /* check the jack status at stream startup */ - tosa_ext_control(codec); - - mutex_unlock(&codec->mutex); + tosa_ext_control(&rtd->card->dapm); return 0; } @@ -104,13 +102,13 @@ static int tosa_get_jack(struct snd_kcontrol *kcontrol, static int tosa_set_jack(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (tosa_jack_func == ucontrol->value.integer.value[0]) return 0; tosa_jack_func = ucontrol->value.integer.value[0]; - tosa_ext_control(codec); + tosa_ext_control(&card->dapm); return 1; } @@ -124,13 +122,13 @@ static int tosa_get_spk(struct snd_kcontrol *kcontrol, static int tosa_set_spk(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); if (tosa_spk_func == ucontrol->value.integer.value[0]) return 0; tosa_spk_func = ucontrol->value.integer.value[0]; - tosa_ext_control(codec); + tosa_ext_control(&card->dapm); return 1; } @@ -191,24 +189,10 @@ static int tosa_ac97_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; snd_soc_dapm_nc_pin(dapm, "OUT3"); snd_soc_dapm_nc_pin(dapm, "MONOOUT"); - /* add tosa specific controls */ - err = snd_soc_add_codec_controls(codec, tosa_controls, - ARRAY_SIZE(tosa_controls)); - if (err < 0) - return err; - - /* add tosa specific widgets */ - snd_soc_dapm_new_controls(dapm, tosa_dapm_widgets, - ARRAY_SIZE(tosa_dapm_widgets)); - - /* set up tosa specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - return 0; } @@ -239,6 +223,13 @@ static struct snd_soc_card tosa = { .owner = THIS_MODULE, .dai_link = tosa_dai, .num_links = ARRAY_SIZE(tosa_dai), + + .controls = tosa_controls, + .num_controls = ARRAY_SIZE(tosa_controls), + .dapm_widgets = tosa_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tosa_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int tosa_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c index db8aadf8932d..23bf991e95d5 100644 --- a/sound/soc/pxa/zylonite.c +++ b/sound/soc/pxa/zylonite.c @@ -71,22 +71,10 @@ static const struct snd_soc_dapm_route audio_map[] = { static int zylonite_wm9713_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - if (clk_pout) snd_soc_dai_set_pll(rtd->codec_dai, 0, 0, clk_get_rate(pout), 0); - snd_soc_dapm_new_controls(dapm, zylonite_dapm_widgets, - ARRAY_SIZE(zylonite_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - /* Static setup for now */ - snd_soc_dapm_enable_pin(dapm, "Headphone"); - snd_soc_dapm_enable_pin(dapm, "Headset Earpiece"); - return 0; } @@ -256,6 +244,11 @@ static struct snd_soc_card zylonite = { .resume_pre = &zylonite_resume_pre, .dai_link = zylonite_dai, .num_links = ARRAY_SIZE(zylonite_dai), + + .dapm_widgets = zylonite_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(zylonite_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *zylonite_snd_ac97_device; diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c index 945e8abdc10f..0b21d1dc80c1 100644 --- a/sound/soc/s6000/s6105-ipcam.c +++ b/sound/soc/s6000/s6105-ipcam.c @@ -104,8 +104,8 @@ static int output_type_get(struct snd_kcontrol *kcontrol, static int output_type_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = kcontrol->private_data; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_card *card = kcontrol->private_data; + struct snd_soc_dapm_context *dapm = &card->dapm; unsigned int val = (ucontrol->value.enumerated.item[0] != 0); char *differential = "Audio Out Differential"; char *stereo = "Audio Out Stereo"; @@ -137,13 +137,7 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - - /* Add s6105 specific widgets */ - snd_soc_dapm_new_controls(dapm, aic3x_dapm_widgets, - ARRAY_SIZE(aic3x_dapm_widgets)); - - /* Set up s6105 specific audio path audio_map */ - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); + struct snd_soc_card *card = rtd->card; /* not present */ snd_soc_dapm_nc_pin(dapm, "MONO_LOUT"); @@ -157,17 +151,10 @@ static int s6105_aic3x_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "RLOUT"); snd_soc_dapm_nc_pin(dapm, "HPRCOM"); - /* always connected */ - snd_soc_dapm_enable_pin(dapm, "Audio In"); - /* must correspond to audio_out_mux.private_value initializer */ - snd_soc_dapm_disable_pin(dapm, "Audio Out Differential"); - snd_soc_dapm_sync(dapm); - snd_soc_dapm_enable_pin(dapm, "Audio Out Stereo"); - - snd_soc_dapm_sync(dapm); + snd_soc_dapm_disable_pin(&card->dapm, "Audio Out Differential"); - snd_ctl_add(codec->card->snd_card, snd_ctl_new1(&audio_out_mux, codec)); + snd_ctl_add(card->snd_card, snd_ctl_new1(&audio_out_mux, card)); return 0; } @@ -190,6 +177,11 @@ static struct snd_soc_card snd_soc_card_s6105 = { .owner = THIS_MODULE, .dai_link = &s6105_dai, .num_links = 1, + + .dapm_widgets = aic3x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic3x_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct s6000_snd_platform_data s6105_snd_data __initdata = { diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 454f41cfc828..f2e289180e46 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -59,7 +59,7 @@ config SND_SOC_SAMSUNG_JIVE_WM8750 select SND_SOC_WM8750 select SND_S3C2412_SOC_I2S help - Sat Y if you want to add support for SoC audio on the Jive. + Say Y if you want to add support for SoC audio on the Jive. config SND_SOC_SAMSUNG_SMDK_WM8580 tristate "SoC I2S Audio support for WM8580 on SMDK" @@ -117,7 +117,7 @@ config SND_SOC_SAMSUNG_SIMTEC_TLV320AIC23 tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" depends on SND_SOC_SAMSUNG && ARCH_S3C24XX select SND_S3C24XX_I2S - select SND_SOC_TLV320AIC23 + select SND_SOC_TLV320AIC23_I2C select SND_SOC_SAMSUNG_SIMTEC config SND_SOC_SAMSUNG_SIMTEC_HERMES @@ -145,11 +145,11 @@ config SND_SOC_SAMSUNG_RX1950_UDA1380 config SND_SOC_SAMSUNG_SMDK_WM9713 tristate "SoC AC97 Audio support for SMDK with WM9713" - depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110 || MACH_SMDKV310 || MACH_SMDKC210) + depends on SND_SOC_SAMSUNG && (MACH_SMDK6410 || MACH_SMDKC100 || MACH_SMDKV210 || MACH_SMDKC110) select SND_SOC_WM9713 select SND_SAMSUNG_AC97 help - Sat Y if you want to add support for SoC audio on the SMDK. + Say Y if you want to add support for SoC audio on the SMDK. config SND_SOC_SMARTQ tristate "SoC I2S Audio support for SmartQ board" diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index fbced589d077..88b09e022503 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -66,10 +66,6 @@ static int h1940_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.rate_min = hw_rates.list[0]; - runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); @@ -94,7 +90,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, div++; break; default: - dev_err(&rtd->dev, "%s: rate %d is not supported\n", + dev_err(rtd->dev, "%s: rate %d is not supported\n", __func__, rate); return -EINVAL; } @@ -181,7 +177,6 @@ static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 98a04c11202d..b0800337b79e 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -192,44 +192,6 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; -/* Shared routes and controls */ - -static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { - SND_SOC_DAPM_LINE("GSM Line Out", NULL), - SND_SOC_DAPM_LINE("GSM Line In", NULL), - SND_SOC_DAPM_MIC("Headset Mic", NULL), - SND_SOC_DAPM_MIC("Handset Mic", NULL), -}; - -static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { - /* Connections to the GSM Module */ - {"GSM Line Out", NULL, "MONO1"}, - {"GSM Line Out", NULL, "MONO2"}, - {"RXP", NULL, "GSM Line In"}, - {"RXN", NULL, "GSM Line In"}, - - /* Connections to Headset */ - {"MIC1", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Headset Mic"}, - - /* Call Mic */ - {"MIC2", NULL, "Mic Bias"}, - {"MIC2N", NULL, "Mic Bias"}, - {"Mic Bias", NULL, "Handset Mic"}, - - /* Connect the ALC pins */ - {"ACIN", NULL, "ACOP"}, -}; - -static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { - SOC_DAPM_PIN_SWITCH("GSM Line Out"), - SOC_DAPM_PIN_SWITCH("GSM Line In"), - SOC_DAPM_PIN_SWITCH("Headset Mic"), - SOC_DAPM_PIN_SWITCH("Handset Mic"), -}; - -/* GTA02 specific routes and controls */ - static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -257,7 +219,34 @@ static int lm4853_event(struct snd_soc_dapm_widget *w, return 0; } -static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { +static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Line Out", NULL), + SND_SOC_DAPM_LINE("GSM Line In", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Handset Mic", NULL), + SND_SOC_DAPM_SPK("Handset Spk", NULL), + SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), +}; + +static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = { + /* Connections to the GSM Module */ + {"GSM Line Out", NULL, "MONO1"}, + {"GSM Line Out", NULL, "MONO2"}, + {"RXP", NULL, "GSM Line In"}, + {"RXN", NULL, "GSM Line In"}, + + /* Connections to Headset */ + {"MIC1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Headset Mic"}, + + /* Call Mic */ + {"MIC2", NULL, "Mic Bias"}, + {"MIC2N", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Handset Mic"}, + + /* Connect the ALC pins */ + {"ACIN", NULL, "ACOP"}, + /* Connections to the amp */ {"Stereo Out", NULL, "LOUT1"}, {"Stereo Out", NULL, "ROUT1"}, @@ -267,7 +256,11 @@ static const struct snd_soc_dapm_route neo1973_gta02_routes[] = { {"Handset Spk", NULL, "ROUT2"}, }; -static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { +static const struct snd_kcontrol_new neo1973_wm8753_controls[] = { + SOC_DAPM_PIN_SWITCH("GSM Line Out"), + SOC_DAPM_PIN_SWITCH("GSM Line In"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Handset Mic"), SOC_DAPM_PIN_SWITCH("Handset Spk"), SOC_DAPM_PIN_SWITCH("Stereo Out"), @@ -276,86 +269,32 @@ static const struct snd_kcontrol_new neo1973_gta02_wm8753_controls[] = { lm4853_set_spk), }; -static const struct snd_soc_dapm_widget neo1973_gta02_wm8753_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Handset Spk", NULL), - SND_SOC_DAPM_SPK("Stereo Out", lm4853_event), -}; - -static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec) -{ - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; - - ret = snd_soc_dapm_new_controls(dapm, neo1973_gta02_wm8753_dapm_widgets, - ARRAY_SIZE(neo1973_gta02_wm8753_dapm_widgets)); - if (ret) - return ret; - - ret = snd_soc_dapm_add_routes(dapm, neo1973_gta02_routes, - ARRAY_SIZE(neo1973_gta02_routes)); - if (ret) - return ret; - - ret = snd_soc_add_card_controls(codec->card, neo1973_gta02_wm8753_controls, - ARRAY_SIZE(neo1973_gta02_wm8753_controls)); - if (ret) - return ret; - - snd_soc_dapm_disable_pin(dapm, "Stereo Out"); - snd_soc_dapm_disable_pin(dapm, "Handset Spk"); - snd_soc_dapm_ignore_suspend(dapm, "Stereo Out"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Spk"); - - return 0; -} - static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; + struct snd_soc_card *card = rtd->card; /* set up NC codec pins */ - snd_soc_dapm_nc_pin(dapm, "OUT3"); - snd_soc_dapm_nc_pin(dapm, "OUT4"); - snd_soc_dapm_nc_pin(dapm, "LINE1"); - snd_soc_dapm_nc_pin(dapm, "LINE2"); - - /* Add neo1973 specific widgets */ - ret = snd_soc_dapm_new_controls(dapm, neo1973_wm8753_dapm_widgets, - ARRAY_SIZE(neo1973_wm8753_dapm_widgets)); - if (ret) - return ret; - - /* add neo1973 specific controls */ - ret = snd_soc_add_card_controls(rtd->card, neo1973_wm8753_controls, - ARRAY_SIZE(neo1973_wm8753_controls)); - if (ret) - return ret; - - /* set up neo1973 specific audio routes */ - ret = snd_soc_dapm_add_routes(dapm, neo1973_wm8753_routes, - ARRAY_SIZE(neo1973_wm8753_routes)); - if (ret) - return ret; + snd_soc_dapm_nc_pin(&codec->dapm, "OUT3"); + snd_soc_dapm_nc_pin(&codec->dapm, "OUT4"); + snd_soc_dapm_nc_pin(&codec->dapm, "LINE1"); + snd_soc_dapm_nc_pin(&codec->dapm, "LINE2"); /* set endpoints to default off mode */ - snd_soc_dapm_disable_pin(dapm, "GSM Line Out"); - snd_soc_dapm_disable_pin(dapm, "GSM Line In"); - snd_soc_dapm_disable_pin(dapm, "Headset Mic"); - snd_soc_dapm_disable_pin(dapm, "Handset Mic"); + snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out"); + snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In"); + snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic"); + snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic"); + snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out"); + snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk"); /* allow audio paths from the GSM modem to run during suspend */ - snd_soc_dapm_ignore_suspend(dapm, "GSM Line Out"); - snd_soc_dapm_ignore_suspend(dapm, "GSM Line In"); - snd_soc_dapm_ignore_suspend(dapm, "Headset Mic"); - snd_soc_dapm_ignore_suspend(dapm, "Handset Mic"); - - if (machine_is_neo1973_gta02()) { - ret = neo1973_gta02_wm8753_init(codec); - if (ret) - return ret; - } + snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out"); + snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk"); return 0; } @@ -409,6 +348,13 @@ static struct snd_soc_card neo1973 = { .num_aux_devs = ARRAY_SIZE(neo1973_aux_devs), .codec_conf = neo1973_codec_conf, .num_configs = ARRAY_SIZE(neo1973_codec_conf), + + .controls = neo1973_wm8753_controls, + .num_controls = ARRAY_SIZE(neo1973_wm8753_controls), + .dapm_widgets = neo1973_wm8753_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets), + .dapm_routes = neo1973_wm8753_routes, + .num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes), }; static struct platform_device *neo1973_snd_device; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 06ebdc061770..2982d9e7f268 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -131,10 +131,6 @@ static int rx1950_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - runtime->hw.rate_min = hw_rates.list[0]; - runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; - runtime->hw.rates = SNDRV_PCM_RATE_KNOT; - return snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_rates); @@ -226,7 +222,6 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - int err; snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index d38ae98e2f32..682eb4f7ba0c 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -202,7 +202,7 @@ static int smdk_audio_probe(struct platform_device *pdev) static struct platform_driver smdk_audio_driver = { .driver = { - .name = "smdk-audio-wm8894", + .name = "smdk-audio-wm8994", .owner = THIS_MODULE, .of_match_table = of_match_ptr(samsung_wm8994_of_match), .pm = &snd_soc_pm_ops, diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index f21ff608a819..1807b75ccc12 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -44,6 +44,8 @@ static int tobermory_set_bias_level(struct snd_soc_card *card, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); + snd_soc_dai_set_pll(codec_dai, WM8962_FLL, + 0, 0, 0); return ret; } } diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 1967f44e7cd4..710a079a7377 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1711,9 +1711,9 @@ static int fsi_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - fsi->clk_master = 1; break; case SND_SOC_DAIFMT_CBS_CFS: + fsi->clk_master = 1; /* codec is slave, cpu is master */ break; default: return -EINVAL; diff --git a/sound/soc/sh/migor.c b/sound/soc/sh/migor.c index 5014a884afee..c58c2529f103 100644 --- a/sound/soc/sh/migor.c +++ b/sound/soc/sh/migor.c @@ -136,19 +136,6 @@ static const struct snd_soc_dapm_route audio_map[] = { { "Mic Bias", NULL, "External Microphone" }, }; -static int migor_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_dapm_context *dapm = &codec->dapm; - - snd_soc_dapm_new_controls(dapm, migor_dapm_widgets, - ARRAY_SIZE(migor_dapm_widgets)); - - snd_soc_dapm_add_routes(dapm, audio_map, ARRAY_SIZE(audio_map)); - - return 0; -} - /* migor digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link migor_dai = { .name = "wm8978", @@ -158,7 +145,6 @@ static struct snd_soc_dai_link migor_dai = { .platform_name = "siu-pcm-audio", .codec_name = "wm8978.0-001a", .ops = &migor_dai_ops, - .init = migor_dai_init, }; /* migor audio machine driver */ @@ -167,6 +153,11 @@ static struct snd_soc_card snd_soc_migor = { .owner = THIS_MODULE, .dai_link = &migor_dai, .num_links = 1, + + .dapm_widgets = migor_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(migor_dapm_widgets), + .dapm_routes = audio_map, + .num_dapm_routes = ARRAY_SIZE(audio_map), }; static struct platform_device *migor_snd_device; diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index e77f7716f1d7..215b668166be 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -525,10 +525,10 @@ static int rsnd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: - rdai->clk_master = 1; + rdai->clk_master = 0; break; case SND_SOC_DAIFMT_CBS_CFS: - rdai->clk_master = 0; + rdai->clk_master = 1; /* codec is slave, cpu is master */ break; default: return -EINVAL; diff --git a/sound/soc/sirf/Kconfig b/sound/soc/sirf/Kconfig new file mode 100644 index 000000000000..89e89429b04a --- /dev/null +++ b/sound/soc/sirf/Kconfig @@ -0,0 +1,14 @@ +config SND_SOC_SIRF + tristate "SoC Audio for the SiRF SoC chips" + depends on ARCH_SIRF || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_SIRF_AUDIO + tristate "SoC Audio support for SiRF internal audio codec" + depends on SND_SOC_SIRF + select SND_SOC_SIRF_AUDIO_CODEC + select SND_SOC_SIRF_AUDIO_PORT + +config SND_SOC_SIRF_AUDIO_PORT + select REGMAP_MMIO + tristate diff --git a/sound/soc/sirf/Makefile b/sound/soc/sirf/Makefile new file mode 100644 index 000000000000..913b93231d4e --- /dev/null +++ b/sound/soc/sirf/Makefile @@ -0,0 +1,5 @@ +snd-soc-sirf-audio-objs := sirf-audio.o +snd-soc-sirf-audio-port-objs := sirf-audio-port.o + +obj-$(CONFIG_SND_SOC_SIRF_AUDIO) += snd-soc-sirf-audio.o +obj-$(CONFIG_SND_SOC_SIRF_AUDIO_PORT) += snd-soc-sirf-audio-port.o diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c new file mode 100644 index 000000000000..b04a53f2b4f6 --- /dev/null +++ b/sound/soc/sirf/sirf-audio-port.c @@ -0,0 +1,194 @@ +/* + * SiRF Audio port driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ +#include <linux/module.h> +#include <linux/io.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/dmaengine_pcm.h> + +#include "sirf-audio-port.h" + +struct sirf_audio_port { + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data playback_dma_data; + struct snd_dmaengine_dai_dma_data capture_dma_data; +}; + +static void sirf_audio_port_tx_enable(struct sirf_audio_port *port) +{ + regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); + regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0); + regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); + regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_START, AUDIO_FIFO_START); + regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL, + IC_TX_ENABLE, IC_TX_ENABLE); +} + +static void sirf_audio_port_tx_disable(struct sirf_audio_port *port) +{ + regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); + regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL, + IC_TX_ENABLE, ~IC_TX_ENABLE); +} + +static void sirf_audio_port_rx_enable(struct sirf_audio_port *port, + int channels) +{ + regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); + regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_INT_MSK, 0); + regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0); + regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_START, AUDIO_FIFO_START); + if (channels == 1) + regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO); + else + regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO); +} + +static void sirf_audio_port_rx_disable(struct sirf_audio_port *port) +{ + regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO); +} + +static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai) +{ + struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai); + snd_soc_dai_init_dma_data(dai, &port->playback_dma_data, + &port->capture_dma_data); + return 0; +} + +static int sirf_audio_port_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai); + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (playback) + sirf_audio_port_tx_disable(port); + else + sirf_audio_port_rx_disable(port); + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (playback) + sirf_audio_port_tx_enable(port); + else + sirf_audio_port_rx_enable(port, + substream->runtime->channels); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dai_ops sirf_audio_port_dai_ops = { + .trigger = sirf_audio_port_trigger, +}; + +static struct snd_soc_dai_driver sirf_audio_port_dai = { + .probe = sirf_audio_port_dai_probe, + .name = "sirf-audio-port", + .id = 0, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &sirf_audio_port_dai_ops, +}; + +static const struct snd_soc_component_driver sirf_audio_port_component = { + .name = "sirf-audio-port", +}; + +static const struct regmap_config sirf_audio_port_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK, + .cache_type = REGCACHE_NONE, +}; + +static int sirf_audio_port_probe(struct platform_device *pdev) +{ + int ret; + struct sirf_audio_port *port; + void __iomem *base; + struct resource *mem_res; + + port = devm_kzalloc(&pdev->dev, + sizeof(struct sirf_audio_port), GFP_KERNEL); + if (!port) + return -ENOMEM; + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem_res) { + dev_err(&pdev->dev, "no mem resource?\n"); + return -ENODEV; + } + + base = devm_ioremap(&pdev->dev, mem_res->start, + resource_size(mem_res)); + if (base == NULL) + return -ENOMEM; + + port->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sirf_audio_port_regmap_config); + if (IS_ERR(port->regmap)) + return PTR_ERR(port->regmap); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sirf_audio_port_component, &sirf_audio_port_dai, 1); + if (ret) + return ret; + + platform_set_drvdata(pdev, port); + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); +} + +static const struct of_device_id sirf_audio_port_of_match[] = { + { .compatible = "sirf,audio-port", }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_audio_port_of_match); + +static struct platform_driver sirf_audio_port_driver = { + .driver = { + .name = "sirf-audio-port", + .owner = THIS_MODULE, + .of_match_table = sirf_audio_port_of_match, + }, + .probe = sirf_audio_port_probe, +}; + +module_platform_driver(sirf_audio_port_driver); + +MODULE_DESCRIPTION("SiRF Audio Port driver"); +MODULE_AUTHOR("RongJun Ying <Rongjun.Ying@csr.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/sirf/sirf-audio-port.h b/sound/soc/sirf/sirf-audio-port.h new file mode 100644 index 000000000000..f32dc54f4499 --- /dev/null +++ b/sound/soc/sirf/sirf-audio-port.h @@ -0,0 +1,62 @@ +/* + * SiRF Audio port controllers define + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_AUDIO_PORT_H +#define _SIRF_AUDIO_PORT_H + +#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F +#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 +#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 +#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 + +#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_SC_OFFSET) +#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_LC_OFFSET) +#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_HC_OFFSET) + +#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F +#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 +#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 +#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 + +#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_SC_OFFSET) +#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_LC_OFFSET) +#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_HC_OFFSET) +#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) +#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) + +#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) +#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) +#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) +#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) +#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) + +#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) +#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) +#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) +#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) +#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) + +#define AUDIO_FIFO_START (1 << 0) +#define AUDIO_FIFO_RESET (1 << 1) + +#define AUDIO_FIFO_FULL (1 << 0) +#define AUDIO_FIFO_EMPTY (1 << 1) +#define AUDIO_FIFO_OFLOW (1 << 2) +#define AUDIO_FIFO_UFLOW (1 << 3) + +#define IC_TX_ENABLE (0x03) +#define IC_RX_ENABLE_MONO (0x01) +#define IC_RX_ENABLE_STEREO (0x03) + +#endif /*__SIRF_AUDIO_PORT_H*/ diff --git a/sound/soc/sirf/sirf-audio.c b/sound/soc/sirf/sirf-audio.c new file mode 100644 index 000000000000..ecef51021653 --- /dev/null +++ b/sound/soc/sirf/sirf-audio.c @@ -0,0 +1,156 @@ +/* + * SiRF audio card driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include <linux/platform_device.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> + +struct sirf_audio_card { + unsigned int gpio_hp_pa; + unsigned int gpio_spk_pa; +}; + +static int sirf_audio_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *ctrl, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card); + int on = !SND_SOC_DAPM_EVENT_OFF(event); + if (gpio_is_valid(sirf_audio_card->gpio_hp_pa)) + gpio_set_value(sirf_audio_card->gpio_hp_pa, on); + return 0; +} + +static int sirf_audio_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *ctrl, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct sirf_audio_card *sirf_audio_card = snd_soc_card_get_drvdata(card); + int on = !SND_SOC_DAPM_EVENT_OFF(event); + + if (gpio_is_valid(sirf_audio_card->gpio_spk_pa)) + gpio_set_value(sirf_audio_card->gpio_spk_pa, on); + + return 0; +} +static const struct snd_soc_dapm_widget sirf_audio_dapm_widgets[] = { + SND_SOC_DAPM_HP("Hp", sirf_audio_hp_event), + SND_SOC_DAPM_SPK("Ext Spk", sirf_audio_spk_event), + SND_SOC_DAPM_MIC("Ext Mic", NULL), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"Hp", NULL, "HPOUTL"}, + {"Hp", NULL, "HPOUTR"}, + {"Ext Spk", NULL, "SPKOUT"}, + {"MICIN1", NULL, "Mic Bias"}, + {"Mic Bias", NULL, "Ext Mic"}, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sirf_audio_dai_link[] = { + { + .name = "SiRF audio card", + .stream_name = "SiRF audio HiFi", + .codec_dai_name = "sirf-audio-codec", + }, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sirf_audio_card = { + .name = "SiRF audio card", + .owner = THIS_MODULE, + .dai_link = sirf_audio_dai_link, + .num_links = ARRAY_SIZE(sirf_audio_dai_link), + .dapm_widgets = sirf_audio_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_audio_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), +}; + +static int sirf_audio_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snd_soc_sirf_audio_card; + struct sirf_audio_card *sirf_audio_card; + int ret; + + sirf_audio_card = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_card), + GFP_KERNEL); + if (sirf_audio_card == NULL) + return -ENOMEM; + + sirf_audio_dai_link[0].cpu_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0); + sirf_audio_dai_link[0].platform_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,audio-platform", 0); + sirf_audio_dai_link[0].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "sirf,audio-codec", 0); + sirf_audio_card->gpio_spk_pa = of_get_named_gpio(pdev->dev.of_node, + "spk-pa-gpios", 0); + sirf_audio_card->gpio_hp_pa = of_get_named_gpio(pdev->dev.of_node, + "hp-pa-gpios", 0); + if (gpio_is_valid(sirf_audio_card->gpio_spk_pa)) { + ret = devm_gpio_request_one(&pdev->dev, + sirf_audio_card->gpio_spk_pa, + GPIOF_OUT_INIT_LOW, "SPA_PA_SD"); + if (ret) { + dev_err(&pdev->dev, + "Failed to request GPIO_%d for reset: %d\n", + sirf_audio_card->gpio_spk_pa, ret); + return ret; + } + } + if (gpio_is_valid(sirf_audio_card->gpio_hp_pa)) { + ret = devm_gpio_request_one(&pdev->dev, + sirf_audio_card->gpio_hp_pa, + GPIOF_OUT_INIT_LOW, "HP_PA_SD"); + if (ret) { + dev_err(&pdev->dev, + "Failed to request GPIO_%d for reset: %d\n", + sirf_audio_card->gpio_hp_pa, ret); + return ret; + } + } + + card->dev = &pdev->dev; + snd_soc_card_set_drvdata(card, sirf_audio_card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); + + return ret; +} + +static const struct of_device_id sirf_audio_of_match[] = { + {.compatible = "sirf,sirf-audio-card", }, + { }, +}; +MODULE_DEVICE_TABLE(of, sirf_audio_of_match); + +static struct platform_driver sirf_audio_driver = { + .driver = { + .name = "sirf-audio-card", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = sirf_audio_of_match, + }, + .probe = sirf_audio_probe, +}; +module_platform_driver(sirf_audio_driver); + +MODULE_AUTHOR("RongJun Ying <RongJun.Ying@csr.com>"); +MODULE_DESCRIPTION("ALSA SoC SIRF audio card driver"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 375dc6dfba4e..bfed3e4c45ff 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -96,8 +96,7 @@ int snd_soc_cache_exit(struct snd_soc_codec *codec) { dev_dbg(codec->dev, "ASoC: Destroying cache for %s codec\n", codec->name); - if (!codec->reg_cache) - return 0; + kfree(codec->reg_cache); codec->reg_cache = NULL; return 0; @@ -117,8 +116,9 @@ int snd_soc_cache_read(struct snd_soc_codec *codec, return -EINVAL; mutex_lock(&codec->cache_rw_mutex); - *value = snd_soc_get_cache_val(codec->reg_cache, reg, - codec->driver->reg_word_size); + if (!ZERO_OR_NULL_PTR(codec->reg_cache)) + *value = snd_soc_get_cache_val(codec->reg_cache, reg, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); return 0; @@ -136,8 +136,9 @@ int snd_soc_cache_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { mutex_lock(&codec->cache_rw_mutex); - snd_soc_set_cache_val(codec->reg_cache, reg, value, - codec->driver->reg_word_size); + if (!ZERO_OR_NULL_PTR(codec->reg_cache)) + snd_soc_set_cache_val(codec->reg_cache, reg, value, + codec->driver->reg_word_size); mutex_unlock(&codec->cache_rw_mutex); return 0; diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 5e9690c85d8f..91083e6a6b38 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -30,8 +30,6 @@ static int soc_compr_open(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); @@ -52,17 +50,7 @@ static int soc_compr_open(struct snd_compr_stream *cstream) } } - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; + snd_soc_runtime_activate(rtd, cstream->direction); mutex_unlock(&rtd->pcm_mutex); @@ -81,8 +69,6 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) struct snd_soc_pcm_runtime *fe = cstream->private_data; struct snd_pcm_substream *fe_substream = fe->pcm->streams[0].substream; struct snd_soc_platform *platform = fe->platform; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; - struct snd_soc_dai *codec_dai = fe->codec_dai; struct snd_soc_dpcm *dpcm; struct snd_soc_dapm_widget_list *list; int stream; @@ -140,17 +126,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) fe->dpcm[stream].state = SND_SOC_DPCM_STATE_OPEN; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - - cpu_dai->active++; - codec_dai->active++; - fe->codec->active++; + snd_soc_runtime_activate(fe, stream); mutex_unlock(&fe->card->mutex); @@ -202,23 +178,18 @@ static int soc_compr_free(struct snd_compr_stream *cstream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + int stream; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (cstream->direction == SND_COMPRESS_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } + if (cstream->direction == SND_COMPRESS_PLAYBACK) + stream = SNDRV_PCM_STREAM_PLAYBACK; + else + stream = SNDRV_PCM_STREAM_CAPTURE; - snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); + snd_soc_runtime_deactivate(rtd, stream); - cpu_dai->active--; - codec_dai->active--; - codec->active--; + snd_soc_dai_digital_mute(codec_dai, 1, cstream->direction); if (!cpu_dai->active) cpu_dai->rate = 0; @@ -235,8 +206,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + if (snd_soc_runtime_ignore_pmdown_time(rtd)) { snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); @@ -261,26 +231,17 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) { struct snd_soc_pcm_runtime *fe = cstream->private_data; struct snd_soc_platform *platform = fe->platform; - struct snd_soc_dai *cpu_dai = fe->cpu_dai; - struct snd_soc_dai *codec_dai = fe->codec_dai; struct snd_soc_dpcm *dpcm; int stream, ret; mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); - if (cstream->direction == SND_COMPRESS_PLAYBACK) { + if (cstream->direction == SND_COMPRESS_PLAYBACK) stream = SNDRV_PCM_STREAM_PLAYBACK; - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { + else stream = SNDRV_PCM_STREAM_CAPTURE; - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - cpu_dai->active--; - codec_dai->active--; - fe->codec->active--; + snd_soc_runtime_deactivate(fe, stream); fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_FE; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50805a3..b322cf294d06 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -56,7 +56,6 @@ EXPORT_SYMBOL_GPL(snd_soc_debugfs_root); #endif static DEFINE_MUTEX(client_mutex); -static LIST_HEAD(dai_list); static LIST_HEAD(platform_list); static LIST_HEAD(codec_list); static LIST_HEAD(component_list); @@ -370,18 +369,22 @@ static ssize_t dai_list_read_file(struct file *file, char __user *user_buf, { char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL); ssize_t len, ret = 0; + struct snd_soc_component *component; struct snd_soc_dai *dai; if (!buf) return -ENOMEM; - list_for_each_entry(dai, &dai_list, list) { - len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", dai->name); - if (len >= 0) - ret += len; - if (ret > PAGE_SIZE) { - ret = PAGE_SIZE; - break; + list_for_each_entry(component, &component_list, list) { + list_for_each_entry(dai, &component->dai_list, list) { + len = snprintf(buf + ret, PAGE_SIZE - ret, "%s\n", + dai->name); + if (len >= 0) + ret += len; + if (ret > PAGE_SIZE) { + ret = PAGE_SIZE; + break; + } } } @@ -855,6 +858,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_component *component; struct snd_soc_codec *codec; struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai, *cpu_dai; @@ -863,18 +867,20 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) dev_dbg(card->dev, "ASoC: binding %s at idx %d\n", dai_link->name, num); /* Find CPU DAI from registered DAIs*/ - list_for_each_entry(cpu_dai, &dai_list, list) { + list_for_each_entry(component, &component_list, list) { if (dai_link->cpu_of_node && - (cpu_dai->dev->of_node != dai_link->cpu_of_node)) + component->dev->of_node != dai_link->cpu_of_node) continue; if (dai_link->cpu_name && - strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name)) - continue; - if (dai_link->cpu_dai_name && - strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + strcmp(dev_name(component->dev), dai_link->cpu_name)) continue; + list_for_each_entry(cpu_dai, &component->dai_list, list) { + if (dai_link->cpu_dai_name && + strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; - rtd->cpu_dai = cpu_dai; + rtd->cpu_dai = cpu_dai; + } } if (!rtd->cpu_dai) { @@ -899,12 +905,10 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) * CODEC found, so find CODEC DAI from registered DAIs from * this CODEC */ - list_for_each_entry(codec_dai, &dai_list, list) { - if (codec->dev == codec_dai->dev && - !strcmp(codec_dai->name, - dai_link->codec_dai_name)) { - + list_for_each_entry(codec_dai, &codec->component.dai_list, list) { + if (!strcmp(codec_dai->name, dai_link->codec_dai_name)) { rtd->codec_dai = codec_dai; + break; } } @@ -1128,15 +1132,21 @@ static int soc_probe_codec(struct snd_soc_card *card, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &dai_list, list) { - if (dai->dev != codec->dev) - continue; - + list_for_each_entry(dai, &codec->component.dai_list, list) snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); - } codec->dapm.idle_bias_off = driver->idle_bias_off; + if (!codec->write && dev_get_regmap(codec->dev, NULL)) { + /* Set the default I/O up try regmap */ + ret = snd_soc_codec_set_cache_io(codec, NULL); + if (ret < 0) { + dev_err(codec->dev, + "Failed to set cache I/O: %d\n", ret); + goto err_probe; + } + } + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { @@ -1150,10 +1160,6 @@ static int soc_probe_codec(struct snd_soc_card *card, codec->name); } - /* If the driver didn't set I/O up try regmap */ - if (!codec->write && dev_get_regmap(codec->dev, NULL)) - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); @@ -1180,6 +1186,7 @@ static int soc_probe_platform(struct snd_soc_card *card, { int ret = 0; const struct snd_soc_platform_driver *driver = platform->driver; + struct snd_soc_component *component; struct snd_soc_dai *dai; platform->card = card; @@ -1195,11 +1202,11 @@ static int soc_probe_platform(struct snd_soc_card *card, driver->dapm_widgets, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &dai_list, list) { - if (dai->dev != platform->dev) + list_for_each_entry(component, &component_list, list) { + if (component->dev != platform->dev) continue; - - snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); + list_for_each_entry(dai, &component->dai_list, list) + snd_soc_dapm_new_dai_widgets(&platform->dapm, dai); } platform->dapm.idle_bias_off = 1; @@ -2571,10 +2578,10 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->max; + uinfo->value.enumerated.items = e->items; - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; + if (uinfo->value.enumerated.item >= e->items) + uinfo->value.enumerated.item = e->items - 1; strlcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item], sizeof(uinfo->value.enumerated.name)); @@ -2596,14 +2603,18 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val; + unsigned int val, item; + unsigned int reg_val; - val = snd_soc_read(codec, e->reg); - ucontrol->value.enumerated.item[0] - = (val >> e->shift_l) & e->mask; - if (e->shift_l != e->shift_r) - ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & e->mask; + reg_val = snd_soc_read(codec, e->reg); + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[0] = item; + if (e->shift_l != e->shift_r) { + val = (reg_val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[1] = item; + } return 0; } @@ -2623,17 +2634,18 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; unsigned int val; unsigned int mask; - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (item[0] >= e->items) return -EINVAL; - val = ucontrol->value.enumerated.item[0] << e->shift_l; + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (item[1] >= e->items) return -EINVAL; - val |= ucontrol->value.enumerated.item[1] << e->shift_r; + val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; mask |= e->mask << e->shift_r; } @@ -2642,78 +2654,46 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); /** - * snd_soc_get_value_enum_double - semi enumerated double mixer get callback - * @kcontrol: mixer control - * @ucontrol: control element information + * snd_soc_read_signed - Read a codec register and interprete as signed value + * @codec: codec + * @reg: Register to read + * @mask: Mask to use after shifting the register value + * @shift: Right shift of register value + * @sign_bit: Bit that describes if a number is negative or not. * - * Callback to get the value of a double semi enumerated mixer. + * This functions reads a codec register. The register value is shifted right + * by 'shift' bits and masked with the given 'mask'. Afterwards it translates + * the given registervalue into a signed integer if sign_bit is non-zero. * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. - * - * Returns 0 for success. + * Returns the register value as signed int. */ -int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) +static int snd_soc_read_signed(struct snd_soc_codec *codec, unsigned int reg, + unsigned int mask, unsigned int shift, unsigned int sign_bit) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int reg_val, val, mux; + int ret; + unsigned int val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; - for (mux = 0; mux < e->max; mux++) { - if (val == e->values[mux]) - break; - } - ucontrol->value.enumerated.item[0] = mux; - if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; - for (mux = 0; mux < e->max; mux++) { - if (val == e->values[mux]) - break; - } - ucontrol->value.enumerated.item[1] = mux; - } + val = (snd_soc_read(codec, reg) >> shift) & mask; - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_get_value_enum_double); + if (!sign_bit) + return val; -/** - * snd_soc_put_value_enum_double - semi enumerated double mixer put callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a double semi enumerated mixer. - * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. - * - * Returns 0 for success. - */ -int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val; - unsigned int mask; + /* non-negative number */ + if (!(val & BIT(sign_bit))) + return val; - if (ucontrol->value.enumerated.item[0] > e->max - 1) - return -EINVAL; - val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) - return -EINVAL; - val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; - mask |= e->mask << e->shift_r; - } + ret = val; - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + /* + * The register most probably does not contain a full-sized int. + * Instead we have an arbitrary number of bits in a signed + * representation which has to be translated into a full-sized int. + * This is done by filling up all bits above the sign-bit. + */ + ret |= ~((int)(BIT(sign_bit) - 1)); + + return ret; } -EXPORT_SYMBOL_GPL(snd_soc_put_value_enum_double); /** * snd_soc_info_volsw - single mixer info callback @@ -2743,7 +2723,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = platform_max; + uinfo->value.integer.max = platform_max - mc->min; return 0; } EXPORT_SYMBOL_GPL(snd_soc_info_volsw); @@ -2769,11 +2749,16 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; int max = mc->max; + int min = mc->min; + int sign_bit = mc->sign_bit; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; - ucontrol->value.integer.value[0] = - (snd_soc_read(codec, reg) >> shift) & mask; + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + ucontrol->value.integer.value[0] = snd_soc_read_signed(codec, reg, mask, + shift, sign_bit) - min; if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; @@ -2781,10 +2766,12 @@ int snd_soc_get_volsw(struct snd_kcontrol *kcontrol, if (snd_soc_volsw_is_stereo(mc)) { if (reg == reg2) ucontrol->value.integer.value[1] = - (snd_soc_read(codec, reg) >> rshift) & mask; + snd_soc_read_signed(codec, reg, mask, rshift, + sign_bit) - min; else ucontrol->value.integer.value[1] = - (snd_soc_read(codec, reg2) >> shift) & mask; + snd_soc_read_signed(codec, reg2, mask, shift, + sign_bit) - min; if (invert) ucontrol->value.integer.value[1] = max - ucontrol->value.integer.value[1]; @@ -2815,20 +2802,25 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; int max = mc->max; + int min = mc->min; + unsigned int sign_bit = mc->sign_bit; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; int err; - bool type_2r = 0; + bool type_2r = false; unsigned int val2 = 0; unsigned int val, val_mask; - val = (ucontrol->value.integer.value[0] & mask); + if (sign_bit) + mask = BIT(sign_bit + 1) - 1; + + val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) val = max - val; val_mask = mask << shift; val = val << shift; if (snd_soc_volsw_is_stereo(mc)) { - val2 = (ucontrol->value.integer.value[1] & mask); + val2 = ((ucontrol->value.integer.value[1] + min) & mask); if (invert) val2 = max - val2; if (reg == reg2) { @@ -2836,7 +2828,7 @@ int snd_soc_put_volsw(struct snd_kcontrol *kcontrol, val |= val2 << rshift; } else { val2 = val2 << shift; - type_2r = 1; + type_2r = true; } } err = snd_soc_update_bits_locked(codec, reg, val_mask, val); @@ -3234,7 +3226,7 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct soc_bytes *params = (void *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); int ret, len; - unsigned int val; + unsigned int val, mask; void *data; if (!codec->using_regmap) @@ -3264,12 +3256,36 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, ((u8 *)data)[0] |= val; break; case 2: - ((u16 *)data)[0] &= cpu_to_be16(~params->mask); - ((u16 *)data)[0] |= cpu_to_be16(val); + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, + &mask, &mask); + if (ret != 0) + goto out; + + ((u16 *)data)[0] &= mask; + + ret = regmap_parse_val(codec->control_data, + &val, &val); + if (ret != 0) + goto out; + + ((u16 *)data)[0] |= val; break; case 4: - ((u32 *)data)[0] &= cpu_to_be32(~params->mask); - ((u32 *)data)[0] |= cpu_to_be32(val); + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, + &mask, &mask); + if (ret != 0) + goto out; + + ((u32 *)data)[0] &= mask; + + ret = regmap_parse_val(codec->control_data, + &val, &val); + if (ret != 0) + goto out; + + ((u32 *)data)[0] |= val; break; default: ret = -EINVAL; @@ -3609,6 +3625,30 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** + * snd_soc_of_xlate_tdm_slot - generate tx/rx slot mask. + * @slots: Number of slots in use. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. + * + * Generates the TDM tx and rx slot default masks for DAI. + */ +static int snd_soc_of_xlate_tdm_slot_mask(unsigned int slots, + unsigned int *tx_mask, + unsigned int *rx_mask) +{ + if (*tx_mask || *rx_mask) + return 0; + + if (!slots) + return -EINVAL; + + *tx_mask = (1 << slots) - 1; + *rx_mask = (1 << slots) - 1; + + return 0; +} + +/** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI * @tx_mask: bitmask representing active TX slots. @@ -3622,11 +3662,17 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { + if (dai->driver && dai->driver->ops->of_xlate_tdm_slot_mask) + dai->driver->ops->of_xlate_tdm_slot_mask(slots, + &tx_mask, &rx_mask); + else + snd_soc_of_xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); + if (dai->driver && dai->driver->ops->set_tdm_slot) return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); else - return -EINVAL; + return -ENOTSUPP; } EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); @@ -3882,95 +3928,42 @@ static inline char *fmt_multiple_name(struct device *dev, } /** - * snd_soc_register_dai - Register a DAI with the ASoC core + * snd_soc_unregister_dai - Unregister DAIs from the ASoC core * - * @dai: DAI to register + * @component: The component for which the DAIs should be unregistered */ -static int snd_soc_register_dai(struct device *dev, - struct snd_soc_dai_driver *dai_drv) +static void snd_soc_unregister_dais(struct snd_soc_component *component) { - struct snd_soc_codec *codec; - struct snd_soc_dai *dai; - - dev_dbg(dev, "ASoC: dai register %s\n", dev_name(dev)); + struct snd_soc_dai *dai, *_dai; - dai = kzalloc(sizeof(struct snd_soc_dai), GFP_KERNEL); - if (dai == NULL) - return -ENOMEM; - - /* create DAI component name */ - dai->name = fmt_single_name(dev, &dai->id); - if (dai->name == NULL) { + list_for_each_entry_safe(dai, _dai, &component->dai_list, list) { + dev_dbg(component->dev, "ASoC: Unregistered DAI '%s'\n", + dai->name); + list_del(&dai->list); + kfree(dai->name); kfree(dai); - return -ENOMEM; } - - dai->dev = dev; - dai->driver = dai_drv; - dai->dapm.dev = dev; - if (!dai->driver->ops) - dai->driver->ops = &null_dai_ops; - - mutex_lock(&client_mutex); - - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - - if (!dai->codec) - dai->dapm.idle_bias_off = 1; - - list_add(&dai->list, &dai_list); - - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); - - return 0; -} - -/** - * snd_soc_unregister_dai - Unregister a DAI from the ASoC core - * - * @dai: DAI to unregister - */ -static void snd_soc_unregister_dai(struct device *dev) -{ - struct snd_soc_dai *dai; - - list_for_each_entry(dai, &dai_list, list) { - if (dev == dai->dev) - goto found; - } - return; - -found: - mutex_lock(&client_mutex); - list_del(&dai->list); - mutex_unlock(&client_mutex); - - dev_dbg(dev, "ASoC: Unregistered DAI '%s'\n", dai->name); - kfree(dai->name); - kfree(dai); } /** - * snd_soc_register_dais - Register multiple DAIs with the ASoC core + * snd_soc_register_dais - Register a DAI with the ASoC core * - * @dai: Array of DAIs to register + * @component: The component the DAIs are registered for + * @codec: The CODEC that the DAIs are registered for, NULL if the component is + * not a CODEC. + * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs + * @legacy_dai_naming: Use the legacy naming scheme and let the DAI inherit the + * parent's name. */ -static int snd_soc_register_dais(struct device *dev, - struct snd_soc_dai_driver *dai_drv, size_t count) +static int snd_soc_register_dais(struct snd_soc_component *component, + struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv, + size_t count, bool legacy_dai_naming) { - struct snd_soc_codec *codec; + struct device *dev = component->dev; struct snd_soc_dai *dai; - int i, ret = 0; + unsigned int i; + int ret; dev_dbg(dev, "ASoC: dai register %s #%Zu\n", dev_name(dev), count); @@ -3982,70 +3975,54 @@ static int snd_soc_register_dais(struct device *dev, goto err; } - /* create DAI component name */ - dai->name = fmt_multiple_name(dev, &dai_drv[i]); + /* + * Back in the old days when we still had component-less DAIs, + * instead of having a static name, component-less DAIs would + * inherit the name of the parent device so it is possible to + * register multiple instances of the DAI. We still need to keep + * the same naming style even though those DAIs are not + * component-less anymore. + */ + if (count == 1 && legacy_dai_naming) { + dai->name = fmt_single_name(dev, &dai->id); + } else { + dai->name = fmt_multiple_name(dev, &dai_drv[i]); + if (dai_drv[i].id) + dai->id = dai_drv[i].id; + else + dai->id = i; + } if (dai->name == NULL) { kfree(dai); - ret = -EINVAL; + ret = -ENOMEM; goto err; } + dai->component = component; + dai->codec = codec; dai->dev = dev; dai->driver = &dai_drv[i]; - if (dai->driver->id) - dai->id = dai->driver->id; - else - dai->id = i; dai->dapm.dev = dev; if (!dai->driver->ops) dai->driver->ops = &null_dai_ops; - mutex_lock(&client_mutex); - - list_for_each_entry(codec, &codec_list, list) { - if (codec->dev == dev) { - dev_dbg(dev, - "ASoC: Mapped DAI %s to CODEC %s\n", - dai->name, codec->name); - dai->codec = codec; - break; - } - } - if (!dai->codec) dai->dapm.idle_bias_off = 1; - list_add(&dai->list, &dai_list); + list_add(&dai->list, &component->dai_list); - mutex_unlock(&client_mutex); - - dev_dbg(dai->dev, "ASoC: Registered DAI '%s'\n", dai->name); + dev_dbg(dev, "ASoC: Registered DAI '%s'\n", dai->name); } return 0; err: - for (i--; i >= 0; i--) - snd_soc_unregister_dai(dev); + snd_soc_unregister_dais(component); return ret; } /** - * snd_soc_unregister_dais - Unregister multiple DAIs from the ASoC core - * - * @dai: Array of DAIs to unregister - * @count: Number of DAIs - */ -static void snd_soc_unregister_dais(struct device *dev, size_t count) -{ - int i; - - for (i = 0; i < count; i++) - snd_soc_unregister_dai(dev); -} - -/** * snd_soc_register_component - Register a component with the ASoC core * */ @@ -4053,6 +4030,7 @@ static int __snd_soc_register_component(struct device *dev, struct snd_soc_component *cmpnt, const struct snd_soc_component_driver *cmpnt_drv, + struct snd_soc_codec *codec, struct snd_soc_dai_driver *dai_drv, int num_dai, bool allow_single_dai) { @@ -4075,20 +4053,10 @@ __snd_soc_register_component(struct device *dev, cmpnt->driver = cmpnt_drv; cmpnt->dai_drv = dai_drv; cmpnt->num_dai = num_dai; + INIT_LIST_HEAD(&cmpnt->dai_list); - /* - * snd_soc_register_dai() uses fmt_single_name(), and - * snd_soc_register_dais() uses fmt_multiple_name() - * for dai->name which is used for name based matching - * - * this function is used from cpu/codec. - * allow_single_dai flag can ignore "codec" driver reworking - * since it had been used snd_soc_register_dais(), - */ - if ((1 == num_dai) && allow_single_dai) - ret = snd_soc_register_dai(dev, dai_drv); - else - ret = snd_soc_register_dais(dev, dai_drv, num_dai); + ret = snd_soc_register_dais(cmpnt, codec, dai_drv, num_dai, + allow_single_dai); if (ret < 0) { dev_err(dev, "ASoC: Failed to regster DAIs: %d\n", ret); goto error_component_name; @@ -4121,7 +4089,9 @@ int snd_soc_register_component(struct device *dev, return -ENOMEM; } - return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, + cmpnt->ignore_pmdown_time = true; + + return __snd_soc_register_component(dev, cmpnt, cmpnt_drv, NULL, dai_drv, num_dai, true); } EXPORT_SYMBOL_GPL(snd_soc_register_component); @@ -4141,7 +4111,7 @@ void snd_soc_unregister_component(struct device *dev) return; found: - snd_soc_unregister_dais(dev, cmpnt->num_dai); + snd_soc_unregister_dais(cmpnt); mutex_lock(&client_mutex); list_del(&cmpnt->list); @@ -4319,7 +4289,7 @@ int snd_soc_register_codec(struct device *dev, codec->volatile_register = codec_drv->volatile_register; codec->readable_register = codec_drv->readable_register; codec->writable_register = codec_drv->writable_register; - codec->ignore_pmdown_time = codec_drv->ignore_pmdown_time; + codec->component.ignore_pmdown_time = codec_drv->ignore_pmdown_time; codec->dapm.bias_level = SND_SOC_BIAS_OFF; codec->dapm.dev = dev; codec->dapm.codec = codec; @@ -4342,7 +4312,7 @@ int snd_soc_register_codec(struct device *dev, /* register component */ ret = __snd_soc_register_component(dev, &codec->component, &codec_drv->component_driver, - dai_drv, num_dai, false); + codec, dai_drv, num_dai, false); if (ret < 0) { dev_err(codec->dev, "ASoC: Failed to regster component: %d\n", ret); goto fail_codec_name; @@ -4417,6 +4387,122 @@ int snd_soc_of_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(snd_soc_of_parse_card_name); +static const struct snd_soc_dapm_widget simple_widgets[] = { + SND_SOC_DAPM_MIC("Microphone", NULL), + SND_SOC_DAPM_LINE("Line", NULL), + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), +}; + +int snd_soc_of_parse_audio_simple_widgets(struct snd_soc_card *card, + const char *propname) +{ + struct device_node *np = card->dev->of_node; + struct snd_soc_dapm_widget *widgets; + const char *template, *wname; + int i, j, num_widgets, ret; + + num_widgets = of_property_count_strings(np, propname); + if (num_widgets < 0) { + dev_err(card->dev, + "ASoC: Property '%s' does not exist\n", propname); + return -EINVAL; + } + if (num_widgets & 1) { + dev_err(card->dev, + "ASoC: Property '%s' length is not even\n", propname); + return -EINVAL; + } + + num_widgets /= 2; + if (!num_widgets) { + dev_err(card->dev, "ASoC: Property '%s's length is zero\n", + propname); + return -EINVAL; + } + + widgets = devm_kcalloc(card->dev, num_widgets, sizeof(*widgets), + GFP_KERNEL); + if (!widgets) { + dev_err(card->dev, + "ASoC: Could not allocate memory for widgets\n"); + return -ENOMEM; + } + + for (i = 0; i < num_widgets; i++) { + ret = of_property_read_string_index(np, propname, + 2 * i, &template); + if (ret) { + dev_err(card->dev, + "ASoC: Property '%s' index %d read error:%d\n", + propname, 2 * i, ret); + return -EINVAL; + } + + for (j = 0; j < ARRAY_SIZE(simple_widgets); j++) { + if (!strncmp(template, simple_widgets[j].name, + strlen(simple_widgets[j].name))) { + widgets[i] = simple_widgets[j]; + break; + } + } + + if (j >= ARRAY_SIZE(simple_widgets)) { + dev_err(card->dev, + "ASoC: DAPM widget '%s' is not supported\n", + template); + return -EINVAL; + } + + ret = of_property_read_string_index(np, propname, + (2 * i) + 1, + &wname); + if (ret) { + dev_err(card->dev, + "ASoC: Property '%s' index %d read error:%d\n", + propname, (2 * i) + 1, ret); + return -EINVAL; + } + + widgets[i].name = wname; + } + + card->dapm_widgets = widgets; + card->num_dapm_widgets = num_widgets; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); + +int snd_soc_of_parse_tdm_slot(struct device_node *np, + unsigned int *slots, + unsigned int *slot_width) +{ + u32 val; + int ret; + + if (of_property_read_bool(np, "dai-tdm-slot-num")) { + ret = of_property_read_u32(np, "dai-tdm-slot-num", &val); + if (ret) + return ret; + + if (slots) + *slots = val; + } + + if (of_property_read_bool(np, "dai-tdm-slot-width")) { + ret = of_property_read_u32(np, "dai-tdm-slot-width", &val); + if (ret) + return ret; + + if (slot_width) + *slot_width = val; + } + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_of_parse_tdm_slot); + int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, const char *propname) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13187f7..c8a780d0d057 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -70,8 +70,6 @@ static int dapm_up_seq[] = { [snd_soc_dapm_aif_out] = 4, [snd_soc_dapm_mic] = 5, [snd_soc_dapm_mux] = 6, - [snd_soc_dapm_virt_mux] = 6, - [snd_soc_dapm_value_mux] = 6, [snd_soc_dapm_dac] = 7, [snd_soc_dapm_switch] = 8, [snd_soc_dapm_mixer] = 8, @@ -102,8 +100,6 @@ static int dapm_down_seq[] = { [snd_soc_dapm_mic] = 7, [snd_soc_dapm_micbias] = 8, [snd_soc_dapm_mux] = 9, - [snd_soc_dapm_virt_mux] = 9, - [snd_soc_dapm_value_mux] = 9, [snd_soc_dapm_aif_in] = 10, [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai_in] = 10, @@ -115,6 +111,12 @@ static int dapm_down_seq[] = { [snd_soc_dapm_post] = 14, }; +static void dapm_assert_locked(struct snd_soc_dapm_context *dapm) +{ + if (dapm->card && dapm->card->instantiated) + lockdep_assert_held(&dapm->card->dapm_mutex); +} + static void pop_wait(u32 pop_time) { if (pop_time) @@ -146,15 +148,16 @@ static bool dapm_dirty_widget(struct snd_soc_dapm_widget *w) return !list_empty(&w->dirty); } -void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) +static void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason) { + dapm_assert_locked(w->dapm); + if (!dapm_dirty_widget(w)) { dev_vdbg(w->dapm->dev, "Marking %s dirty due to %s\n", w->name, reason); list_add_tail(&w->dirty, &w->dapm->card->dapm_dirty); } } -EXPORT_SYMBOL_GPL(dapm_mark_dirty); void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm) { @@ -361,6 +364,8 @@ static void dapm_reset(struct snd_soc_card *card) { struct snd_soc_dapm_widget *w; + lockdep_assert_held(&card->dapm_mutex); + memset(&card->dapm_stats, 0, sizeof(card->dapm_stats)); list_for_each_entry(w, &card->widgets, list) { @@ -386,7 +391,8 @@ static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg, return -1; } -static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) +static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, + unsigned int val) { if (w->codec) return snd_soc_write(w->codec, reg, val); @@ -498,131 +504,40 @@ out: return ret; } -/* set up initial codec paths */ -static void dapm_set_path_status(struct snd_soc_dapm_widget *w, - struct snd_soc_dapm_path *p, int i) +/* connect mux widget to its interconnecting audio paths */ +static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, + struct snd_soc_dapm_path *path, const char *control_name, + const struct snd_kcontrol_new *kcontrol) { - switch (w->id) { - case snd_soc_dapm_switch: - case snd_soc_dapm_mixer: - case snd_soc_dapm_mixer_named_ctl: { - int val; - struct soc_mixer_control *mc = (struct soc_mixer_control *) - w->kcontrol_news[i].private_value; - int reg = mc->reg; - unsigned int shift = mc->shift; - int max = mc->max; - unsigned int mask = (1 << fls(max)) - 1; - unsigned int invert = mc->invert; - - if (reg != SND_SOC_NOPM) { - soc_widget_read(w, reg, &val); - val = (val >> shift) & mask; - if (invert) - val = max - val; - p->connect = !!val; - } else { - p->connect = 0; - } - - } - break; - case snd_soc_dapm_mux: { - struct soc_enum *e = (struct soc_enum *) - w->kcontrol_news[i].private_value; - int val, item; - - soc_widget_read(w, e->reg, &val); - item = (val >> e->shift_l) & e->mask; - - if (item < e->max && !strcmp(p->name, e->texts[item])) - p->connect = 1; - else - p->connect = 0; - } - break; - case snd_soc_dapm_virt_mux: { - struct soc_enum *e = (struct soc_enum *) - w->kcontrol_news[i].private_value; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int val, item; + int i; - p->connect = 0; + if (e->reg != SND_SOC_NOPM) { + soc_widget_read(dest, e->reg, &val); + val = (val >> e->shift_l) & e->mask; + item = snd_soc_enum_val_to_item(e, val); + } else { /* since a virtual mux has no backing registers to * decide which path to connect, it will try to match * with the first enumeration. This is to ensure * that the default mux choice (the first) will be * correctly powered up during initialization. */ - if (!strcmp(p->name, e->texts[0])) - p->connect = 1; - } - break; - case snd_soc_dapm_value_mux: { - struct soc_enum *e = (struct soc_enum *) - w->kcontrol_news[i].private_value; - int val, item; - - soc_widget_read(w, e->reg, &val); - val = (val >> e->shift_l) & e->mask; - for (item = 0; item < e->max; item++) { - if (val == e->values[item]) - break; - } - - if (item < e->max && !strcmp(p->name, e->texts[item])) - p->connect = 1; - else - p->connect = 0; - } - break; - /* does not affect routing - always connected */ - case snd_soc_dapm_pga: - case snd_soc_dapm_out_drv: - case snd_soc_dapm_output: - case snd_soc_dapm_adc: - case snd_soc_dapm_input: - case snd_soc_dapm_siggen: - case snd_soc_dapm_dac: - case snd_soc_dapm_micbias: - case snd_soc_dapm_vmid: - case snd_soc_dapm_supply: - case snd_soc_dapm_regulator_supply: - case snd_soc_dapm_clock_supply: - case snd_soc_dapm_aif_in: - case snd_soc_dapm_aif_out: - case snd_soc_dapm_dai_in: - case snd_soc_dapm_dai_out: - case snd_soc_dapm_hp: - case snd_soc_dapm_mic: - case snd_soc_dapm_spk: - case snd_soc_dapm_line: - case snd_soc_dapm_dai_link: - case snd_soc_dapm_kcontrol: - p->connect = 1; - break; - /* does affect routing - dynamically connected */ - case snd_soc_dapm_pre: - case snd_soc_dapm_post: - p->connect = 0; - break; + item = 0; } -} - -/* connect mux widget to its interconnecting audio paths */ -static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, - struct snd_soc_dapm_path *path, const char *control_name, - const struct snd_kcontrol_new *kcontrol) -{ - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - int i; - for (i = 0; i < e->max; i++) { + for (i = 0; i < e->items; i++) { if (!(strcmp(control_name, e->texts[i]))) { list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); path->name = (char*)e->texts[i]; - dapm_set_path_status(dest, path, 0); + if (i == item) + path->connect = 1; + else + path->connect = 0; return 0; } } @@ -630,6 +545,30 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, return -ENODEV; } +/* set up initial codec paths */ +static void dapm_set_mixer_path_status(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_path *p, int i) +{ + struct soc_mixer_control *mc = (struct soc_mixer_control *) + w->kcontrol_news[i].private_value; + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + unsigned int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val; + + if (reg != SND_SOC_NOPM) { + soc_widget_read(w, reg, &val); + val = (val >> shift) & mask; + if (invert) + val = max - val; + p->connect = !!val; + } else { + p->connect = 0; + } +} + /* connect mixer widget to its interconnecting audio paths */ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, struct snd_soc_dapm_widget *src, struct snd_soc_dapm_widget *dest, @@ -644,7 +583,7 @@ static int dapm_connect_mixer(struct snd_soc_dapm_context *dapm, list_add(&path->list_sink, &dest->sources); list_add(&path->list_source, &src->sinks); path->name = dest->kcontrol_news[i].name; - dapm_set_path_status(dest, path, i); + dapm_set_mixer_path_status(dest, path, i); return 0; } } @@ -723,8 +662,6 @@ static int dapm_create_or_share_mixmux_kcontrol(struct snd_soc_dapm_widget *w, kcname_in_long_name = true; break; case snd_soc_dapm_mux: - case snd_soc_dapm_virt_mux: - case snd_soc_dapm_value_mux: wname_in_long_name = true; kcname_in_long_name = false; break; @@ -1218,7 +1155,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, false); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to bypass %s: %d\n", + "ASoC: Failed to unbypass %s: %d\n", w->name, ret); } @@ -1228,7 +1165,7 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } @@ -1823,6 +1760,8 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) ASYNC_DOMAIN_EXCLUSIVE(async_domain); enum snd_soc_bias_level bias; + lockdep_assert_held(&card->dapm_mutex); + trace_snd_soc_dapm_start(card); list_for_each_entry(d, &card->dapm_list, list) { @@ -1897,10 +1836,14 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) trace_snd_soc_dapm_walk_done(card); - /* Run all the bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) - async_schedule_domain(dapm_pre_sequence_async, d, - &async_domain); + /* Run card bias changes at first */ + dapm_pre_sequence_async(&card->dapm, 0); + /* Run other bias changes in parallel */ + list_for_each_entry(d, &card->dapm_list, list) { + if (d != &card->dapm) + async_schedule_domain(dapm_pre_sequence_async, d, + &async_domain); + } async_synchronize_full_domain(&async_domain); list_for_each_entry(w, &down_list, power_list) { @@ -1920,10 +1863,14 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) dapm_seq_run(card, &up_list, event, true); /* Run all the bias changes in parallel */ - list_for_each_entry(d, &card->dapm_list, list) - async_schedule_domain(dapm_post_sequence_async, d, - &async_domain); + list_for_each_entry(d, &card->dapm_list, list) { + if (d != &card->dapm) + async_schedule_domain(dapm_post_sequence_async, d, + &async_domain); + } async_synchronize_full_domain(&async_domain); + /* Run card bias changes at last */ + dapm_post_sequence_async(&card->dapm, 0); /* do we need to notify any clients that DAPM event is complete */ list_for_each_entry(d, &card->dapm_list, list) { @@ -2110,6 +2057,8 @@ static int soc_dapm_mux_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { if (!path->name || !e->texts[mux]) @@ -2160,6 +2109,8 @@ static int soc_dapm_mixer_update_power(struct snd_soc_card *card, struct snd_soc_dapm_path *path; int found = 0; + lockdep_assert_held(&card->dapm_mutex); + /* find dapm widget path assoc with kcontrol */ dapm_kcontrol_for_each_path(path, kcontrol) { found = 1; @@ -2325,6 +2276,8 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + dapm_assert_locked(dapm); + if (!w) { dev_err(dapm->dev, "ASoC: DAPM unknown pin %s\n", pin); return -EINVAL; @@ -2341,18 +2294,18 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, } /** - * snd_soc_dapm_sync - scan and power dapm paths + * snd_soc_dapm_sync_unlocked - scan and power dapm paths * @dapm: DAPM context * * Walks all dapm audio paths and powers widgets according to their * stream or path usage. * + * Requires external locking. + * * Returns 0 for success. */ -int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) +int snd_soc_dapm_sync_unlocked(struct snd_soc_dapm_context *dapm) { - int ret; - /* * Suppress early reports (eg, jacks syncing their state) to avoid * silly DAPM runs during card startup. @@ -2360,8 +2313,25 @@ int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) if (!dapm->card || !dapm->card->instantiated) return 0; + return dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_sync_unlocked); + +/** + * snd_soc_dapm_sync - scan and power dapm paths + * @dapm: DAPM context + * + * Walks all dapm audio paths and powers widgets according to their + * stream or path usage. + * + * Returns 0 for success. + */ +int snd_soc_dapm_sync(struct snd_soc_dapm_context *dapm) +{ + int ret; + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - ret = dapm_power_widgets(dapm->card, SND_SOC_DAPM_STREAM_NOP); + ret = snd_soc_dapm_sync_unlocked(dapm); mutex_unlock(&dapm->card->dapm_mutex); return ret; } @@ -2444,8 +2414,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, path->connect = 1; return 0; case snd_soc_dapm_mux: - case snd_soc_dapm_virt_mux: - case snd_soc_dapm_value_mux: ret = dapm_connect_mux(dapm, wsource, wsink, path, control, &wsink->kcontrol_news[0]); if (ret != 0) @@ -2772,8 +2740,6 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) dapm_new_mixer(w); break; case snd_soc_dapm_mux: - case snd_soc_dapm_virt_mux: - case snd_soc_dapm_value_mux: dapm_new_mux(w); break; case snd_soc_dapm_pga: @@ -2935,213 +2901,75 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val; - - val = snd_soc_read(codec, e->reg); - ucontrol->value.enumerated.item[0] = (val >> e->shift_l) & e->mask; - if (e->shift_l != e->shift_r) - ucontrol->value.enumerated.item[1] = - (val >> e->shift_r) & e->mask; - - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); - -/** - * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to set the value of a dapm enumerated double mixer control. - * - * Returns 0 for success. - */ -int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux, change; - unsigned int mask; - struct snd_soc_dapm_update update; - int ret = 0; - - if (ucontrol->value.enumerated.item[0] > e->max - 1) - return -EINVAL; - mux = ucontrol->value.enumerated.item[0]; - val = mux << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) - return -EINVAL; - val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= e->mask << e->shift_r; - } - - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - - change = snd_soc_test_bits(codec, e->reg, mask, val); - if (change) { - update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; - card->update = &update; - - ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); - - card->update = NULL; - } - - mutex_unlock(&card->dapm_mutex); + unsigned int reg_val, val; - if (ret > 0) - soc_dpcm_runtime_update(card); - - return change; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); - -/** - * snd_soc_dapm_get_enum_virt - Get virtual DAPM mux - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Returns 0 for success. - */ -int snd_soc_dapm_get_enum_virt(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - ucontrol->value.enumerated.item[0] = dapm_kcontrol_get_value(kcontrol); - return 0; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_virt); - -/** - * snd_soc_dapm_put_enum_virt - Set virtual DAPM mux - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Returns 0 for success. - */ -int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct snd_soc_card *card = codec->card; - unsigned int value; - struct soc_enum *e = - (struct soc_enum *)kcontrol->private_value; - int change; - int ret = 0; - - if (ucontrol->value.enumerated.item[0] >= e->max) - return -EINVAL; - - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - - value = ucontrol->value.enumerated.item[0]; - change = dapm_kcontrol_set_value(kcontrol, value); - if (change) - ret = soc_dapm_mux_update_power(card, kcontrol, value, e); - - mutex_unlock(&card->dapm_mutex); - - if (ret > 0) - soc_dpcm_runtime_update(card); - - return change; -} -EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); - -/** - * snd_soc_dapm_get_value_enum_double - dapm semi enumerated double mixer get - * callback - * @kcontrol: mixer control - * @ucontrol: control element information - * - * Callback to get the value of a dapm semi enumerated double mixer control. - * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. - * - * Returns 0 for success. - */ -int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int reg_val, val, mux; + if (e->reg != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg); + else + reg_val = dapm_kcontrol_get_value(kcontrol); - reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; - for (mux = 0; mux < e->max; mux++) { - if (val == e->values[mux]) - break; - } - ucontrol->value.enumerated.item[0] = mux; + ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask; - for (mux = 0; mux < e->max; mux++) { - if (val == e->values[mux]) - break; - } - ucontrol->value.enumerated.item[1] = mux; + val = snd_soc_enum_val_to_item(e, val); + ucontrol->value.enumerated.item[1] = val; } return 0; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_get_value_enum_double); +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_double); /** - * snd_soc_dapm_put_value_enum_double - dapm semi enumerated double mixer set - * callback + * snd_soc_dapm_put_enum_double - dapm enumerated double mixer set callback * @kcontrol: mixer control * @ucontrol: control element information * - * Callback to set the value of a dapm semi enumerated double mixer control. - * - * Semi enumerated mixer: the enumerated items are referred as values. Can be - * used for handling bitfield coded enumeration for example. + * Callback to set the value of a dapm enumerated double mixer control. * * Returns 0 for success. */ -int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, +int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, mux, change; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, change; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0; - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (item[0] >= e->items) return -EINVAL; - mux = ucontrol->value.enumerated.item[0]; - val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; + + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (item[1] > e->items) return -EINVAL; - val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; + val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg, mask, val); + else + change = dapm_kcontrol_set_value(kcontrol, val); + if (change) { - update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; - card->update = &update; + if (e->reg != SND_SOC_NOPM) { + update.kcontrol = kcontrol; + update.reg = e->reg; + update.mask = mask; + update.val = val; + card->update = &update; + } - ret = soc_dapm_mux_update_power(card, kcontrol, mux, e); + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); card->update = NULL; } @@ -3153,7 +2981,7 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, return change; } -EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double); +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** * snd_soc_dapm_info_pin_switch - Info for a pin switch @@ -3210,15 +3038,11 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (ucontrol->value.integer.value[0]) snd_soc_dapm_enable_pin(&card->dapm, pin); else snd_soc_dapm_disable_pin(&card->dapm, pin); - mutex_unlock(&card->dapm_mutex); - snd_soc_dapm_sync(&card->dapm); return 0; } @@ -3248,7 +3072,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = regulator_allow_bypass(w->regulator, true); if (ret != 0) dev_warn(w->dapm->dev, - "ASoC: Failed to unbypass %s: %d\n", + "ASoC: Failed to bypass %s: %d\n", w->name, ret); } break; @@ -3287,8 +3111,6 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_mux: - case snd_soc_dapm_virt_mux: - case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; break; case snd_soc_dapm_dai_out: @@ -3767,23 +3589,52 @@ void snd_soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, } /** + * snd_soc_dapm_enable_pin_unlocked - enable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Enables input/output pin and its parents or children widgets iff there is + * a valid audio route and active audio stream. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 1); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin_unlocked); + +/** * snd_soc_dapm_enable_pin - enable pin. * @dapm: DAPM context * @pin: pin name * * Enables input/output pin and its parents or children widgets iff there is * a valid audio route and active audio stream. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ int snd_soc_dapm_enable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 1); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 1); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); /** - * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * snd_soc_dapm_force_enable_pin_unlocked - force a pin to be enabled * @dapm: DAPM context * @pin: pin name * @@ -3791,11 +3642,13 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin); * intended for use with microphone bias supplies used in microphone * jack detection. * + * Requires external locking. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ -int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, - const char *pin) +int snd_soc_dapm_force_enable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); @@ -3811,25 +3664,103 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, return 0; } +EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin_unlocked); + +/** + * snd_soc_dapm_force_enable_pin - force a pin to be enabled + * @dapm: DAPM context + * @pin: pin name + * + * Enables input/output pin regardless of any other state. This is + * intended for use with microphone bias supplies used in microphone + * jack detection. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_force_enable_pin_unlocked(dapm, pin); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; +} EXPORT_SYMBOL_GPL(snd_soc_dapm_force_enable_pin); /** + * snd_soc_dapm_disable_pin_unlocked - disable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Disables input/output pin and its parents or children widgets. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_disable_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin_unlocked); + +/** * snd_soc_dapm_disable_pin - disable pin. * @dapm: DAPM context * @pin: pin name * * Disables input/output pin and its parents or children widgets. + * * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to * do any widget power switching. */ int snd_soc_dapm_disable_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 0); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 0); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); /** + * snd_soc_dapm_nc_pin_unlocked - permanently disable pin. + * @dapm: DAPM context + * @pin: pin name + * + * Marks the specified pin as being not connected, disabling it along + * any parent or child widgets. At present this is identical to + * snd_soc_dapm_disable_pin() but in future it will be extended to do + * additional things such as disabling controls which only affect + * paths through the pin. + * + * Requires external locking. + * + * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to + * do any widget power switching. + */ +int snd_soc_dapm_nc_pin_unlocked(struct snd_soc_dapm_context *dapm, + const char *pin) +{ + return snd_soc_dapm_set_pin(dapm, pin, 0); +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin_unlocked); + +/** * snd_soc_dapm_nc_pin - permanently disable pin. * @dapm: DAPM context * @pin: pin name @@ -3845,7 +3776,15 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin); */ int snd_soc_dapm_nc_pin(struct snd_soc_dapm_context *dapm, const char *pin) { - return snd_soc_dapm_set_pin(dapm, pin, 0); + int ret; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_dapm_set_pin(dapm, pin, 0); + + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin); @@ -3985,7 +3924,7 @@ void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm) } EXPORT_SYMBOL_GPL(snd_soc_dapm_free); -static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) +static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) { struct snd_soc_card *card = dapm->card; struct snd_soc_dapm_widget *w; @@ -4025,14 +3964,21 @@ static void soc_dapm_shutdown_codec(struct snd_soc_dapm_context *dapm) */ void snd_soc_dapm_shutdown(struct snd_soc_card *card) { - struct snd_soc_codec *codec; + struct snd_soc_dapm_context *dapm; - list_for_each_entry(codec, &card->codec_dev_list, card_list) { - soc_dapm_shutdown_codec(&codec->dapm); - if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) - snd_soc_dapm_set_bias_level(&codec->dapm, - SND_SOC_BIAS_OFF); + list_for_each_entry(dapm, &card->dapm_list, list) { + if (dapm != &card->dapm) { + soc_dapm_shutdown_dapm(dapm); + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(dapm, + SND_SOC_BIAS_OFF); + } } + + soc_dapm_shutdown_dapm(&card->dapm); + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + snd_soc_dapm_set_bias_level(&card->dapm, + SND_SOC_BIAS_OFF); } /* Module information */ diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index aa886cca3ecf..260efc8466fc 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -23,21 +23,6 @@ static int hw_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { - int ret; - - if (!snd_soc_codec_volatile_register(codec, reg) && - reg < codec->driver->reg_cache_size && - !codec->cache_bypass) { - ret = snd_soc_cache_write(codec, reg, value); - if (ret < 0) - return -1; - } - - if (codec->cache_only) { - codec->cache_sync = 1; - return 0; - } - return regmap_write(codec->control_data, reg, value); } @@ -46,32 +31,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) int ret; unsigned int val; - if (reg >= codec->driver->reg_cache_size || - snd_soc_codec_volatile_register(codec, reg) || - codec->cache_bypass) { - if (codec->cache_only) - return -1; - - ret = regmap_read(codec->control_data, reg, &val); - if (ret == 0) - return val; - else - return -1; - } - - ret = snd_soc_cache_read(codec, reg, &val); - if (ret < 0) + ret = regmap_read(codec->control_data, reg, &val); + if (ret == 0) + return val; + else return -1; - return val; } /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * * @codec: CODEC to configure. - * @addr_bits: Number of bits of register address data. - * @data_bits: Number of bits of data per register. - * @control: Control bus used. + * @map: Register map to write to * * Register formats are frequently shared between many I2C and SPI * devices. In order to promote code reuse the ASoC core provides @@ -85,60 +56,36 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * volatile registers. */ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) + struct regmap *regmap) { - struct regmap_config config; int ret; - memset(&config, 0, sizeof(config)); - codec->write = hw_write; - codec->read = hw_read; - - config.reg_bits = addr_bits; - config.val_bits = data_bits; + /* Device has made its own regmap arrangements */ + if (!regmap) + codec->control_data = dev_get_regmap(codec->dev, NULL); + else + codec->control_data = regmap; - switch (control) { -#if IS_ENABLED(CONFIG_REGMAP_I2C) - case SND_SOC_I2C: - codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), - &config); - break; -#endif + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); -#if IS_ENABLED(CONFIG_REGMAP_SPI) - case SND_SOC_SPI: - codec->control_data = regmap_init_spi(to_spi_device(codec->dev), - &config); - break; -#endif - - case SND_SOC_REGMAP: - /* Device has made its own regmap arrangements */ - codec->using_regmap = true; - if (!codec->control_data) - codec->control_data = dev_get_regmap(codec->dev, NULL); + codec->write = hw_write; + codec->read = hw_read; - if (codec->control_data) { - ret = regmap_get_val_bytes(codec->control_data); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - codec->val_bytes = ret; - } - break; + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte + * multiples */ + if (ret > 0) + codec->val_bytes = ret; - default: - return -EINVAL; - } + codec->using_regmap = true; - return PTR_ERR_OR_ZERO(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) + struct regmap *regmap) { return -ENOTSUPP; } diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 23d43dac91da..b903f822d1b2 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -250,7 +250,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) report = 0; if (gpio->jack_status_check) - report = gpio->jack_status_check(); + report = gpio->jack_status_check(gpio->data); snd_soc_jack_report(jack, report, gpio->report); } @@ -342,7 +342,8 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, gpio_export(gpios[i].gpio, false); /* Update initial jack status */ - snd_soc_jack_gpio_detect(&gpios[i]); + schedule_delayed_work(&gpios[i].work, + msecs_to_jiffies(gpios[i].debounce_time)); } return 0; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 47e1ce771e65..2cedf09f6d96 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -35,6 +35,86 @@ #define DPCM_MAX_BE_USERS 8 /** + * snd_soc_runtime_activate() - Increment active count for PCM runtime components + * @rtd: ASoC PCM runtime that is activated + * @stream: Direction of the PCM stream + * + * Increments the active count for all the DAIs and components attached to a PCM + * runtime. Should typically be called when a stream is opened. + * + * Must be called with the rtd->pcm_mutex being held + */ +void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream) +{ + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + lockdep_assert_held(&rtd->pcm_mutex); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active++; + codec_dai->playback_active++; + } else { + cpu_dai->capture_active++; + codec_dai->capture_active++; + } + + cpu_dai->active++; + codec_dai->active++; + cpu_dai->component->active++; + codec_dai->component->active++; +} + +/** + * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components + * @rtd: ASoC PCM runtime that is deactivated + * @stream: Direction of the PCM stream + * + * Decrements the active count for all the DAIs and components attached to a PCM + * runtime. Should typically be called when a stream is closed. + * + * Must be called with the rtd->pcm_mutex being held + */ +void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream) +{ + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + lockdep_assert_held(&rtd->pcm_mutex); + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + cpu_dai->playback_active--; + codec_dai->playback_active--; + } else { + cpu_dai->capture_active--; + codec_dai->capture_active--; + } + + cpu_dai->active--; + codec_dai->active--; + cpu_dai->component->active--; + codec_dai->component->active--; +} + +/** + * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay + * @rtd: The ASoC PCM runtime that should be checked. + * + * This function checks whether the power down delay should be ignored for a + * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has + * been configured to ignore the delay, or if none of the components benefits + * from having the delay. + */ +bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd) +{ + if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time) + return true; + + return rtd->cpu_dai->component->ignore_pmdown_time && + rtd->codec_dai->component->ignore_pmdown_time; +} + +/** * snd_soc_set_runtime_hwparams - set the runtime hardware parameters * @substream: the pcm substream * @hw: the hardware parameters @@ -378,16 +458,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) runtime->hw.rate_max); dynamic: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active++; - codec_dai->playback_active++; - } else { - cpu_dai->capture_active++; - codec_dai->capture_active++; - } - cpu_dai->active++; - codec_dai->active++; - rtd->codec->active++; + + snd_soc_runtime_activate(rtd, substream->stream); + mutex_unlock(&rtd->pcm_mutex); return 0; @@ -459,21 +532,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - cpu_dai->playback_active--; - codec_dai->playback_active--; - } else { - cpu_dai->capture_active--; - codec_dai->capture_active--; - } - - cpu_dai->active--; - codec_dai->active--; - codec->active--; + snd_soc_runtime_deactivate(rtd, substream->stream); /* clear the corresponding DAIs rate when inactive */ if (!cpu_dai->active) @@ -496,8 +558,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) cpu_dai->runtime = NULL; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!rtd->pmdown_time || codec->ignore_pmdown_time || - rtd->dai_link->ignore_pmdown_time) { + if (snd_soc_runtime_ignore_pmdown_time(rtd)) { /* powered down playback stream now */ snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index fe99f461aff0..19cca043e6e4 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -213,10 +213,7 @@ static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) static int spdif_mute_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); ucontrol->value.integer.value[0] = host->saved_params.mute; @@ -226,10 +223,7 @@ static int spdif_mute_get(struct snd_kcontrol *kcontrol, static int spdif_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - struct snd_soc_card *card = codec->card; - struct snd_soc_pcm_runtime *rtd = card->rtd; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); if (host->saved_params.mute == ucontrol->value.integer.value[0]) diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 9f9c1856f822..31198cf7f88d 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -105,7 +105,7 @@ config SND_SOC_TEGRA_TRIMSLICE tristate "SoC Audio support for TrimSlice board" depends on SND_SOC_TEGRA && I2C select SND_SOC_TEGRA20_I2S if ARCH_TEGRA_2x_SOC - select SND_SOC_TLV320AIC23 + select SND_SOC_TLV320AIC23_I2C help Say Y or M here if you want to add support for SoC audio on the TrimSlice platform. diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index cf5e1cfe818d..0a59e2383ef3 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -306,7 +306,7 @@ static const struct regmap_config tegra20_ac97_regmap_config = { .readable_reg = tegra20_ac97_wr_rd_reg, .volatile_reg = tegra20_ac97_volatile_reg, .precious_reg = tegra20_ac97_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_ac97_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index e72392927bd2..a634f13b3ffc 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -128,7 +128,7 @@ static const struct regmap_config tegra20_das_regmap_config = { .max_register = LAST_REG(DAC_INPUT_DATA_CLK_SEL), .writeable_reg = tegra20_das_wr_rd_reg, .readable_reg = tegra20_das_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_das_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 42c1f6bfaf2e..79a9932ffe6e 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -333,7 +333,7 @@ static const struct regmap_config tegra20_i2s_regmap_config = { .readable_reg = tegra20_i2s_wr_rd_reg, .volatile_reg = tegra20_i2s_volatile_reg, .precious_reg = tegra20_i2s_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_i2s_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 8c7c1028e579..a0ce92400faf 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -259,7 +259,7 @@ static const struct regmap_config tegra20_spdif_regmap_config = { .readable_reg = tegra20_spdif_wr_rd_reg, .volatile_reg = tegra20_spdif_volatile_reg, .precious_reg = tegra20_spdif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static int tegra20_spdif_platform_probe(struct platform_device *pdev) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d6f4c9940e0c..0db68f49f4d9 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -471,7 +471,7 @@ static const struct regmap_config tegra30_ahub_apbif_regmap_config = { .readable_reg = tegra30_ahub_apbif_wr_rd_reg, .volatile_reg = tegra30_ahub_apbif_volatile_reg, .precious_reg = tegra30_ahub_apbif_precious_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static bool tegra30_ahub_ahub_wr_rd_reg(struct device *dev, unsigned int reg) @@ -490,7 +490,7 @@ static const struct regmap_config tegra30_ahub_ahub_regmap_config = { .max_register = LAST_REG(AUDIO_RX), .writeable_reg = tegra30_ahub_ahub_wr_rd_reg, .readable_reg = tegra30_ahub_ahub_wr_rd_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static struct tegra30_ahub_soc_data soc_data_tegra30 = { diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 49ad9366add8..f146c41dd3ec 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -357,7 +357,7 @@ static const struct regmap_config tegra30_i2s_regmap_config = { .writeable_reg = tegra30_i2s_wr_rd_reg, .readable_reg = tegra30_i2s_wr_rd_reg, .volatile_reg = tegra30_i2s_volatile_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_FLAT, }; static const struct tegra30_i2s_soc_data tegra30_i2s_config = { diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index e0305a148568..9edd68db9f48 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -183,14 +183,16 @@ static int txx9aclc_ac97_dev_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) return irq; + + drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); + if (!drvdata) + return -ENOMEM; + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); drvdata->base = devm_ioremap_resource(&pdev->dev, r); if (IS_ERR(drvdata->base)) return PTR_ERR(drvdata->base); - drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL); - if (!drvdata) - return -ENOMEM; platform_set_drvdata(pdev, drvdata); drvdata->physbase = r->start; if (sizeof(drvdata->physbase) > sizeof(r->start) && diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index de9408b83f75..e05a86b7c0da 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -14,6 +14,7 @@ config SND_USB_AUDIO select SND_HWDEP select SND_RAWMIDI select SND_PCM + select BITREVERSE help Say Y here to include support for USB audio and USB MIDI devices. diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 44b0ba4feab3..1bed780e21d9 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -883,6 +883,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, } break; + case USB_ID(0x046d, 0x0807): /* Logitech Webcam C500 */ case USB_ID(0x046d, 0x0808): case USB_ID(0x046d, 0x0809): case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 32af6b741ef5..d1d72ff50347 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -328,6 +328,11 @@ static struct usbmix_name_map gamecom780_map[] = { {} }; +static const struct usbmix_name_map kef_x300a_map[] = { + { 10, NULL }, /* firmware locks up (?) when we try to access this FU */ + { 0 } +}; + /* * Control map entries */ @@ -419,6 +424,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x200c, 0x1018), .map = ebox44_map, }, + { + .id = USB_ID(0x27ac, 0x1000), + .map = kef_x300a_map, + }, { 0 } /* terminator */ }; |