diff options
author | Takashi Iwai <tiwai@suse.de> | 2013-12-16 15:53:52 +0100 |
---|---|---|
committer | Takashi Iwai <tiwai@suse.de> | 2013-12-16 15:53:52 +0100 |
commit | d09476018bee39495d6ece7a2e069de29a9c0ed5 (patch) | |
tree | 866fff4323f94681e6b423f269f0549df0a34066 /sound | |
parent | ALSA: atmel_abdac: clk_round_rate() can return a zero upon error (diff) | |
parent | Merge tag 'asoc-v3.13-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/b... (diff) | |
download | linux-d09476018bee39495d6ece7a2e069de29a9c0ed5.tar.xz linux-d09476018bee39495d6ece7a2e069de29a9c0ed5.zip |
Merge branch 'for-linus' into for-next
Diffstat (limited to 'sound')
27 files changed, 288 insertions, 113 deletions
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index b78a3bc6f35e..3519518e25a0 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -357,7 +357,8 @@ static int set_sample_rates(struct atmel_abdac *dac) if (new_rate <= 0) break; /* make sure we are below the ABDAC clock */ - if (new_rate <= clk_get_rate(dac->pclk)) { + if (index < MAX_NUM_RATES && + new_rate <= clk_get_rate(dac->pclk)) { dac->rates[index] = new_rate / 256; index++; } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 44be167d1cad..8cebdcfdcfdc 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -475,6 +475,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx) memset(path, 0, sizeof(*path)); } +/* return a DAC if paired to the given pin by codec driver */ +static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + const hda_nid_t *list = spec->preferred_dacs; + + if (!list) + return 0; + for (; *list; list += 2) + if (*list == pin) + return list[1]; + return 0; +} + /* look for an empty DAC slot */ static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin, bool is_digital) @@ -1193,7 +1207,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, continue; } - dacs[i] = look_for_dac(codec, pin, false); + dacs[i] = get_preferred_dac(codec, pin); + if (dacs[i]) { + if (is_dac_already_used(codec, dacs[i])) + badness += bad->shared_primary; + } + + if (!dacs[i]) + dacs[i] = look_for_dac(codec, pin, false); if (!dacs[i] && !i) { /* try to steal the DAC of surrounds for the front */ for (j = 1; j < num_outs; j++) { @@ -4302,6 +4323,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return AC_PWRST_D3; } +/* mute all aamix inputs initially; parse up to the first leaves */ +static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) +{ + int i, nums; + const hda_nid_t *conn; + bool has_amp; + + nums = snd_hda_get_conn_list(codec, mix, &conn); + has_amp = nid_has_mute(codec, mix, HDA_INPUT); + for (i = 0; i < nums; i++) { + if (has_amp) + snd_hda_codec_amp_stereo(codec, mix, + HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); + else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) + snd_hda_codec_amp_stereo(codec, conn[i], + HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); + } +} /* * Parse the given BIOS configuration and set up the hda_gen_spec @@ -4454,6 +4495,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* mute all aamix input initially */ + if (spec->mixer_nid) + mute_all_mixer_nid(codec, spec->mixer_nid); + dig_only: parse_digital(codec); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 7e45cb44d151..0929a06df812 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -249,6 +249,9 @@ struct hda_gen_spec { const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; + /* preferred pin/DAC pairs; an array of paired NIDs */ + const hda_nid_t *preferred_dacs; + /* loopback mixing mode */ bool aamix_mode; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index af86c71f27bf..440c35546e57 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3446,6 +3446,10 @@ static void check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] = { + SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 05280033c302..b174eb1567b8 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) if (!spec->eapd_nid) return; + if (codec->inv_eapd) + enabled = !enabled; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); @@ -339,6 +341,14 @@ static int patch_ad1986a(struct hda_codec *codec) { int err; struct ad198x_spec *spec; + static hda_nid_t preferred_pairs[] = { + 0x1a, 0x03, + 0x1b, 0x03, + 0x1c, 0x04, + 0x1d, 0x05, + 0x1e, 0x03, + 0 + }; err = alloc_ad_spec(codec); if (err < 0) @@ -359,6 +369,11 @@ static int patch_ad1986a(struct hda_codec *codec) * So, let's disable the shared stream. */ spec->gen.multiout.no_share_stream = 1; + /* give fixed DAC/pin pairs */ + spec->gen.preferred_dacs = preferred_pairs; + + /* AD1986A can't manage the dynamic pin on/off smoothly */ + spec->gen.auto_mute_via_amp = 1; snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl, ad1986a_fixups); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1f2717f817a0..3fbf2883e06e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 1bd637b1dd0f..f5060fc7c303 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct hdmi_spec *spec = codec->spec; + int pin_idx = pin_nid_to_pin_index(spec, jack->nid); + if (pin_idx < 0) + return; + + if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) + snd_hda_jack_report_sync(codec); +} + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pin_nid; - int pin_idx; struct hda_jack_tbl *jack; int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) return; - pin_nid = jack->nid; jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), + codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); - pin_idx = pin_nid_to_pin_index(spec, pin_nid); - if (pin_idx < 0) - return; - - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + jack_callback(codec, jack); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -2080,7 +2082,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hda_nid_t pin_nid = per_pin->pin_nid; hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid, + codec->jackpoll_interval > 0 ? jack_callback : NULL); } return 0; } @@ -2323,8 +2326,9 @@ static int simple_playback_build_controls(struct hda_codec *codec) int err; per_cvt = get_cvt(spec, 0); - err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid, - per_cvt->cvt_nid); + err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid, + per_cvt->cvt_nid, + HDA_PCM_TYPE_HDMI); if (err < 0) return err; return simple_hdmi_build_jack(codec, 0); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 518e6ceeed20..3578f11c8e12 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1827,6 +1827,7 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC889_FIXUP_MBA21_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, ALC887_FIXUP_ASUS_BASS, @@ -1931,17 +1932,13 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, } } -/* Set VREF on speaker pins on imac91 */ -static void alc889_fixup_imac91_vref(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void alc889_fixup_mac_pins(struct hda_codec *codec, + const hda_nid_t *nids, int num_nids) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x18, 0x1a }; int i; - if (action != HDA_FIXUP_ACT_INIT) - return; - for (i = 0; i < ARRAY_SIZE(nids); i++) { + for (i = 0; i < num_nids; i++) { unsigned int val; val = snd_hda_codec_get_pin_target(codec, nids[i]); val |= AC_PINCTL_VREF_50; @@ -1950,6 +1947,26 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, spec->gen.keep_vref_in_automute = 1; } +/* Set VREF on speaker pins on imac91 */ +static void alc889_fixup_imac91_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[2] = { 0x18, 0x1a }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + +/* Set VREF on speaker pins on mba21 */ +static void alc889_fixup_mba21_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[2] = { 0x18, 0x19 }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -2149,6 +2166,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC889_FIXUP_MBA21_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba21_vref, + .chained = true, + .chain_id = ALC889_FIXUP_MBP_VREF, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2219,7 +2242,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), @@ -3340,6 +3363,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d60); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3368,6 +3392,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d50); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3653,11 +3678,6 @@ static void alc283_hp_automute_hook(struct hda_codec *codec, vref); } -static void alc283_chromebook_caps(struct hda_codec *codec) -{ - snd_hda_override_wcaps(codec, 0x03, 0); -} - static void alc283_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3666,9 +3686,26 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - alc283_chromebook_caps(codec); + snd_hda_override_wcaps(codec, 0x03, 0); /* Disable AA-loopback as it causes white noise */ spec->gen.mixer_nid = 0; + break; + case HDA_FIXUP_ACT_INIT: + /* Enable Line1 input control by verb */ + val = alc_read_coef_idx(codec, 0x1a); + alc_write_coef_idx(codec, 0x1a, val | (1 << 4)); + break; + } +} + +static void alc283_fixup_sense_combo_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + int val; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.hp_automute_hook = alc283_hp_automute_hook; break; case HDA_FIXUP_ACT_INIT: @@ -3676,9 +3713,6 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, /* Set to manual mode */ val = alc_read_coef_idx(codec, 0x06); alc_write_coef_idx(codec, 0x06, val & ~0x000c); - /* Enable Line1 input control by verb */ - val = alc_read_coef_idx(codec, 0x1a); - alc_write_coef_idx(codec, 0x1a, val | (1 << 4)); break; } } @@ -3868,12 +3902,14 @@ enum { ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, + ALC283_FIXUP_SENSE_COMBO_JACK, ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, ALC290_FIXUP_MONO_SPEAKERS, @@ -4129,6 +4165,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK, + }, [ALC269_FIXUP_ACER_AC700] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4173,6 +4215,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc283_fixup_chromebook, }, + [ALC283_FIXUP_SENSE_COMBO_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc283_fixup_sense_combo_jack, + .chained = true, + .chain_id = ALC283_FIXUP_CHROME_BOOK, + }, [ALC282_FIXUP_ASUS_TX300] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, @@ -4220,6 +4268,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4251,11 +4300,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "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), @@ -4264,7 +4316,6 @@ 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), - SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), 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), @@ -4372,6 +4423,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, + {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-chrome"}, + {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {} }; @@ -4612,6 +4665,7 @@ static const struct hda_fixup alc861_fixups[] = { static const struct snd_pci_quirk alc861_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), + SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), @@ -5044,8 +5098,11 @@ 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, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), 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_DELL_MIC_NO_PRESENCE), 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/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 8697cedccd21..1ead3c977a51 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; - ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", @@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, return 0; } +static int atmel_ssc_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + break; + default: + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + break; + } + + return 0; +} #ifdef CONFIG_PM static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) @@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, + .trigger = atmel_ssc_trigger, .hw_params = atmel_ssc_hw_params, .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv, diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38d5a15..7d6a9055874b 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) goto out; } + snd_soc_card_set_drvdata(card, priv); + card->dev = &pdev->dev; card->owner = THIS_MODULE; card->dai_link = dai; @@ -107,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) dai->stream_name = "WM8731 PCM"; dai->codec_dai_name = "wm8731-hifi"; dai->init = sam9x5_wm8731_init; - dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; ret = snd_soc_of_parse_card_name(card, "atmel,model"); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c3c7396a6181..99b359e19d35 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), -SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 1, 0), -SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, - ARIZONA_OUT4_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, - ARIZONA_OUT5_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, - ARIZONA_OUT6_OSR_SHIFT, 1, 0), - SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUTPUT_PATH_CONFIG_1R, - ARIZONA_OUT1L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUTPUT_PATH_CONFIG_2R, - ARIZONA_OUT2L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUTPUT_PATH_CONFIG_3R, - ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), - SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c6d759..bc7472c968e3 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 543c5c2631b6..0f17ed3e29f4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); + /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK. + * So we here provisionally enable it and then disable it afterward + * if current bias_level hasn't reached SND_SOC_BIAS_ON. + */ + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); + + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, 0); + if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); return; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 253c88bb7a4c..4f05fb88bddf 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); + + codec->cache_sync = 1; break; } diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 61e48852b9e8..3fd76bc391de 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, break; } - dapm->bias_level = level; - return 0; } diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index eb4373840bb6..3665f612819d 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op) return -ENOMEM; card->dev = &op->dev; - platform_set_drvdata(op, pdata); pdata->card = card; @@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + platform_set_drvdata(op, pdata); + return ret; } diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d34d91743e3f..0b18f654b413 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -33,6 +33,10 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define KIRKWOOD_SPDIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -244,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } - if (dai->id == 0) - ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ - else - ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_play; + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); @@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, @@ -493,7 +497,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, @@ -501,7 +505,7 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .rates = SNDRV_PCM_RATE_8000_192000 | SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6d216cb6c19b..3fde9e402710 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); n810_ext_control(&codec->dapm); - return clk_enable(sys_clkout2); + return clk_prepare_enable(sys_clkout2); } static void n810_shutdown(struct snd_pcm_substream *substream) { - clk_disable(sys_clkout2); + clk_disable_unprepare(sys_clkout2); } static int n810_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 14011d90d70a..ff60e11ecb56 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD + select REGMAP help This option enables R-Car SUR/SCU/SSIU/SSI sound support diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87e881d..a66783e13a9c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, break; case 2: ((u16 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be16(~params->mask); break; case 4: ((u32 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be32(~params->mask); break; default: return -EINVAL; diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d732255c02..3449c1e909ae 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res) */ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) { - struct device **ptr; + struct snd_soc_card **ptr; int ret; ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); @@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) ret = snd_soc_register_card(card); if (ret == 0) { - *ptr = dev; + *ptr = card; devres_add(dev, ptr); } else { devres_free(ptr); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96ce1f4..41949af3baae 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, } } +static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) +{ + unsigned int i; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; + i++) { + if (!pcm->chan[i]) + continue; + dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; + int ret; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) @@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev, dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) - return snd_soc_add_platform(dev, &pcm->platform, + ret = snd_soc_add_platform(dev, &pcm->platform, &dmaengine_no_residue_pcm_platform); else - return snd_soc_add_platform(dev, &pcm->platform, + ret = snd_soc_add_platform(dev, &pcm->platform, &dmaengine_pcm_platform); + if (ret) + goto err_free_dma; + + return 0; + +err_free_dma: + dmaengine_pcm_release_chan(pcm); + kfree(pcm); + return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) { struct snd_soc_platform *platform; struct dmaengine_pcm *pcm; - unsigned int i; platform = snd_soc_lookup_platform(dev); if (!platform) @@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) { - dma_release_channel(pcm->chan[i]); - if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) - break; - } - } - snd_soc_remove_platform(platform); + dmaengine_pcm_release_chan(pcm); kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c01e413..891b9a9bcbf8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } -static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, +static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *codec_stream, struct snd_soc_pcm_stream *cpu_stream) { - hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); - hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max); + struct snd_pcm_hardware *hw = &runtime->hw; + hw->channels_min = max(codec_stream->channels_min, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, @@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, if (cpu_stream->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + + snd_pcm_limit_hw_rates(runtime); + + hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); + hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); } /* @@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback, + soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, &cpu_dai_drv->playback); } else { - soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture, + soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, &cpu_dai_drv->capture); } ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", codec_dai->name, cpu_dai->name); @@ -594,12 +600,13 @@ static int soc_pcm_hw_free(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; + bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* apply codec digital mute */ - if (!codec->active) + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 364bf6a907e1..8c819f811470 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 08bc6931c7c7..8c7c1028e579 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 231a785b3921..02247fee1cf7 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 3454262358b3..f4b12c216f1c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1603,7 +1603,7 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return err; } - return err; + return 0; } int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) |