diff options
Diffstat (limited to 'sound')
34 files changed, 476 insertions, 209 deletions
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 14a286a7bf2b..857586135d18 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -419,6 +419,7 @@ EXPORT_SYMBOL(snd_ctl_make_virtual_master); * snd_ctl_add_vmaster_hook - Add a hook to a vmaster control * @kcontrol: vmaster kctl element * @hook: the hook function + * @private_data: the private_data pointer to be saved * * Adds the given hook to the vmaster control element so that it's called * at each time when the value is changed. diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index b4a6aa960f4b..8490f59709bb 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -1019,13 +1019,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) irq_cfg = get_irq_config(sscape->type, irq[dev]); if (irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } mpu_irq_cfg = get_irq_config(sscape->type, mpu_irq[dev]); if (mpu_irq_cfg == INVALID_IRQ) { snd_printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); - return -ENXIO; + err = -ENXIO; + goto _release_dma; } /* diff --git a/sound/last.c b/sound/last.c index bdd0857b8871..7ffc182e0844 100644 --- a/sound/last.c +++ b/sound/last.c @@ -38,4 +38,4 @@ static int __init alsa_sound_last_init(void) return 0; } -__initcall(alsa_sound_last_init); +late_initcall_sync(alsa_sound_last_init); diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index 2c79d60a725f..536c4c0514d3 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -1294,6 +1294,8 @@ static int __init calibrate_adc(WORD srate) static int upload_dsp_code(void) { + int ret = 0; + msnd_outb(HPBLKSEL_0, dev.io + HP_BLKS); #ifndef HAVE_DSPCODEH INITCODESIZE = mod_firmware_load(INITCODEFILE, &INITCODE); @@ -1312,7 +1314,8 @@ static int upload_dsp_code(void) memcpy_toio(dev.base, PERMCODE, PERMCODESIZE); if (msnd_upload_host(&dev, INITCODE, INITCODESIZE) < 0) { printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); - return -ENODEV; + ret = -ENODEV; + goto out; } #ifdef HAVE_DSPCODEH printk(KERN_INFO LOGNAME ": DSP firmware uploaded (resident)\n"); @@ -1320,12 +1323,13 @@ static int upload_dsp_code(void) printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); #endif +out: #ifndef HAVE_DSPCODEH vfree(INITCODE); vfree(PERMCODE); #endif - return 0; + return ret; } #ifdef MSND_CLASSIC diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 88168044375f..5ca0939e4223 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -2,8 +2,8 @@ config SND_TEA575X tristate - depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 - default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 + depends on SND_FM801_TEA575X_BOOL || SND_ES1968_RADIO || RADIO_SF16FMR2 || RADIO_MAXIRADIO + default SND_FM801 || SND_ES1968 || RADIO_SF16FMR2 || RADIO_MAXIRADIO menuconfig SND_PCI bool "PCI sound devices" diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 8c63200cf339..bc86cb726d79 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Copyright (C) 1997-2012 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -42,7 +42,7 @@ On error *pLockedMemHandle marked invalid, non-zero returned. If this function succeeds, then HpiOs_LockedMem_GetVirtAddr() and HpiOs_LockedMem_GetPyhsAddr() will always succed on the returned handle. */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_locked_mem_handle, /**< memory handle */ u32 size, /**< Size in bytes to allocate */ struct pci_dev *p_os_reference diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 87f4385fe8c7..5ef4fe964366 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Copyright (C) 1997-2012 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -39,11 +39,11 @@ void hpios_delay_micro_seconds(u32 num_micro_sec) } -/** Allocated an area of locked memory for bus master DMA operations. +/** Allocate an area of locked memory for bus master DMA operations. -On error, return -ENOMEM, and *pMemArea.size = 0 +If allocation fails, return 1, and *pMemArea.size = 0 */ -int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, +u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) { /*?? any benefit in using managed dmam_alloc_coherent? */ @@ -62,7 +62,7 @@ int hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, HPI_DEBUG_LOG(WARNING, "failed to allocate %d bytes locked memory\n", size); p_mem_area->size = 0; - return -ENOMEM; + return 1; } } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 9a9f372e1be4..56b4f74c0b13 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -851,6 +851,9 @@ struct hda_codec { unsigned int pin_amp_workaround:1; /* pin out-amp takes index * (e.g. Conexant codecs) */ + unsigned int single_adc_amp:1; /* adc in-amp takes no index + * (e.g. CX20549 codec) + */ unsigned int no_sticky_stream:1; /* no sticky-PCM stream assignment */ unsigned int pins_shutup:1; /* pins are shut up */ unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */ diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index b58b4b1687fa..4c054f4486b9 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -418,7 +418,7 @@ static void hdmi_show_short_audio_desc(struct cea_sad *a) else buf2[0] = '\0'; - printk(KERN_INFO "HDMI: supports coding type %s:" + _snd_printd(SND_PR_VERBOSE, "HDMI: supports coding type %s:" " channels = %d, rates =%s%s\n", cea_audio_coding_type_names[a->format], a->channels, @@ -442,14 +442,14 @@ void snd_hdmi_show_eld(struct hdmi_eld *e) { int i; - printk(KERN_INFO "HDMI: detected monitor %s at connection type %s\n", + _snd_printd(SND_PR_VERBOSE, "HDMI: detected monitor %s at connection type %s\n", e->monitor_name, eld_connection_type_names[e->conn_type]); if (e->spk_alloc) { char buf[SND_PRINT_CHANNEL_ALLOCATION_ADVISED_BUFSIZE]; snd_print_channel_allocation(e->spk_alloc, buf, sizeof(buf)); - printk(KERN_INFO "HDMI: available speakers:%s\n", buf); + _snd_printd(SND_PR_VERBOSE, "HDMI: available speakers:%s\n", buf); } for (i = 0; i < e->sad_count; i++) diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 254ab5204603..e59e2f059b6e 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -651,9 +651,16 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-In caps: "); print_amp_caps(buffer, codec, nid, HDA_INPUT); snd_iprintf(buffer, " Amp-In vals: "); - print_amp_vals(buffer, codec, nid, HDA_INPUT, - wid_caps & AC_WCAP_STEREO, - wid_type == AC_WID_PIN ? 1 : conn_len); + if (wid_type == AC_WID_PIN || + (codec->single_adc_amp && + wid_type == AC_WID_AUD_IN)) + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO, + 1); + else + print_amp_vals(buffer, codec, nid, HDA_INPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); } if (wid_caps & AC_WCAP_OUT_AMP) { snd_iprintf(buffer, " Amp-Out caps: "); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 8c6523bbc797..d906c5b74cf0 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -141,7 +141,6 @@ struct conexant_spec { unsigned int hp_laptop:1; unsigned int asus:1; unsigned int pin_eapd_ctrls:1; - unsigned int single_adc_amp:1; unsigned int adc_switching:1; @@ -687,27 +686,26 @@ static const struct hda_channel_mode cxt5045_modes[1] = { static const struct hda_input_mux cxt5045_capture_source = { .num_items = 2, .items = { - { "IntMic", 0x1 }, - { "ExtMic", 0x2 }, + { "Internal Mic", 0x1 }, + { "Mic", 0x2 }, } }; static const struct hda_input_mux cxt5045_capture_source_benq = { - .num_items = 5, + .num_items = 4, .items = { - { "IntMic", 0x1 }, - { "ExtMic", 0x2 }, - { "LineIn", 0x3 }, - { "CD", 0x4 }, - { "Mixer", 0x0 }, + { "Internal Mic", 0x1 }, + { "Mic", 0x2 }, + { "Line", 0x3 }, + { "Mixer", 0x0 }, } }; static const struct hda_input_mux cxt5045_capture_source_hp530 = { .num_items = 2, .items = { - { "ExtMic", 0x1 }, - { "IntMic", 0x2 }, + { "Mic", 0x1 }, + { "Internal Mic", 0x2 }, } }; @@ -798,10 +796,8 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static const struct snd_kcontrol_new cxt5045_mixers[] = { - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x1, HDA_INPUT), @@ -822,27 +818,15 @@ static const struct snd_kcontrol_new cxt5045_mixers[] = { }; static const struct snd_kcontrol_new cxt5045_benq_mixers[] = { - HDA_CODEC_VOLUME("CD Capture Volume", 0x1a, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Capture Switch", 0x1a, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x17, 0x4, HDA_INPUT), - - HDA_CODEC_VOLUME("Line In Capture Volume", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Line In Capture Switch", 0x1a, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Playback Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Line In Playback Switch", 0x17, 0x3, HDA_INPUT), - - HDA_CODEC_VOLUME("Mixer Capture Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mixer Capture Switch", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x3, HDA_INPUT), {} }; static const struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Internal Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0x2, HDA_INPUT), @@ -946,10 +930,10 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Output controls */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x10, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Node 11 Playback Volume", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Node 11 Playback Switch", 0x11, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Node 12 Playback Volume", 0x12, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Node 12 Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("HP-OUT Playback Volume", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("HP-OUT Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("LINE1 Playback Volume", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("LINE1 Playback Switch", 0x12, 0x0, HDA_OUTPUT), /* Modes for retasking pin widgets */ CXT_PIN_MODE("HP-OUT pin mode", 0x11, CXT_PIN_DIR_INOUT), @@ -960,16 +944,16 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { /* Loopback mixer controls */ - HDA_CODEC_VOLUME("Mixer-1 Volume", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-1 Switch", 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-2 Volume", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-2 Switch", 0x17, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-3 Volume", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-3 Switch", 0x17, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-4 Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-4 Switch", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Mixer-5 Volume", 0x17, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Mixer-5 Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("PCM Volume", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("PCM Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("MIC1 pin Volume", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("MIC1 pin Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("LINE1 pin Volume", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("LINE1 pin Switch", 0x17, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("HP-OUT pin Volume", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("HP-OUT pin Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("CD pin Volume", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD pin Switch", 0x17, 0x4, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -978,16 +962,8 @@ static const struct snd_kcontrol_new cxt5045_test_mixer[] = { .put = conexant_mux_enum_put, }, /* Audio input controls */ - HDA_CODEC_VOLUME("Input-1 Volume", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Input-1 Switch", 0x1a, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Input-2 Volume", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Input-2 Switch", 0x1a, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Input-3 Volume", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_MUTE("Input-3 Switch", 0x1a, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Input-4 Volume", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Input-4 Switch", 0x1a, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("Input-5 Volume", 0x1a, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Input-5 Switch", 0x1a, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x1a, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1a, 0x0, HDA_INPUT), { } /* end */ }; @@ -1009,10 +985,6 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x18, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Start with output sum widgets muted and their output gains at min */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - /* Unmute retasking pin widget output buffers since the default * state appears to be output. As the pin mode is changed by the * user the pin mode control will take care of enabling the pin's @@ -1027,11 +999,11 @@ static const struct hda_verb cxt5045_test_init_verbs[] = { /* Set ADC connection select to match default mixer setting (mic1 * pin) */ - {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x17, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x17, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Mute all inputs to mixer widget (even unconnected ones) */ - {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer pin */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* Mixer */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Mic1 pin */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* Line pin */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* HP pin */ @@ -1110,7 +1082,7 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - codec->pin_amp_workaround = 1; + codec->single_adc_amp = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -3999,9 +3971,14 @@ static void cx_auto_init_output(struct hda_codec *codec) int i; mute_outputs(codec, spec->multiout.num_dacs, spec->multiout.dac_nids); - for (i = 0; i < cfg->hp_outs; i++) + for (i = 0; i < cfg->hp_outs; i++) { + unsigned int val = PIN_OUT; + if (snd_hda_query_pin_caps(codec, cfg->hp_pins[i]) & + AC_PINCAP_HP_DRV) + val |= AC_PINCTL_HP_EN; snd_hda_codec_write(codec, cfg->hp_pins[i], 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + AC_VERB_SET_PIN_WIDGET_CONTROL, val); + } mute_outputs(codec, cfg->hp_outs, cfg->hp_pins); mute_outputs(codec, cfg->line_outs, cfg->line_out_pins); mute_outputs(codec, cfg->speaker_outs, cfg->speaker_pins); @@ -4220,7 +4197,7 @@ static int cx_auto_add_capture_volume(struct hda_codec *codec, hda_nid_t nid, int idx = get_input_connection(codec, adc_nid, nid); if (idx < 0) continue; - if (spec->single_adc_amp) + if (codec->single_adc_amp) idx = 0; return cx_auto_add_volume_idx(codec, label, pfx, cidx, adc_nid, HDA_INPUT, idx); @@ -4275,7 +4252,7 @@ static int cx_auto_build_input_controls(struct hda_codec *codec) if (cidx < 0) continue; input_conn[i] = spec->imux_info[i].adc; - if (!spec->single_adc_amp) + if (!codec->single_adc_amp) input_conn[i] |= cidx << 8; if (i > 0 && input_conn[i] != input_conn[0]) multi_connection = 1; @@ -4419,8 +4396,10 @@ static void apply_pin_fixup(struct hda_codec *codec, enum { CXT_PINCFG_LENOVO_X200, + CXT_PINCFG_LENOVO_TP410, }; +/* ThinkPad X200 & co with cxt5051 */ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ { 0x17, 0x21a11000 }, /* dock-mic */ @@ -4429,15 +4408,33 @@ static const struct cxt_pincfg cxt_pincfg_lenovo_x200[] = { {} }; +/* ThinkPad 410/420/510/520, X201 & co with cxt5066 */ +static const struct cxt_pincfg cxt_pincfg_lenovo_tp410[] = { + { 0x19, 0x042110ff }, /* HP (seq# overridden) */ + { 0x1a, 0x21a190f0 }, /* dock-mic */ + { 0x1c, 0x212140ff }, /* dock-HP */ + {} +}; + static const struct cxt_pincfg *cxt_pincfg_tbl[] = { [CXT_PINCFG_LENOVO_X200] = cxt_pincfg_lenovo_x200, + [CXT_PINCFG_LENOVO_TP410] = cxt_pincfg_lenovo_tp410, }; -static const struct snd_pci_quirk cxt_fixups[] = { +static const struct snd_pci_quirk cxt5051_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT_PINCFG_LENOVO_X200), {} }; +static const struct snd_pci_quirk cxt5066_fixups[] = { + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + {} +}; + /* add "fake" mute amp-caps to DACs on cx5051 so that mixer mute switches * can be created (bko#42825) */ @@ -4466,19 +4463,21 @@ static int patch_conexant_auto(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; - codec->pin_amp_workaround = 1; switch (codec->vendor_id) { case 0x14f15045: - spec->single_adc_amp = 1; + codec->single_adc_amp = 1; break; case 0x14f15051: add_cx5051_fake_mutes(codec); + codec->pin_amp_workaround = 1; + apply_pin_fixup(codec, cxt5051_fixups, cxt_pincfg_tbl); break; + default: + codec->pin_amp_workaround = 1; + apply_pin_fixup(codec, cxt5066_fixups, cxt_pincfg_tbl); } - apply_pin_fixup(codec, cxt_fixups, cxt_pincfg_tbl); - /* Show mute-led control only on HP laptops * This is a sort of white-list: on HP laptops, EAPD corresponds * only to the mute-LED without actualy amp function. Meanwhile, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 540cd13f7f15..83f345f3c961 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -757,8 +757,6 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) struct hdmi_spec *spec = codec->spec; int tag = res >> AC_UNSOL_RES_TAG_SHIFT; int pin_nid; - int pd = !!(res & AC_UNSOL_RES_PD); - int eldv = !!(res & AC_UNSOL_RES_ELDV); int pin_idx; struct hda_jack_tbl *jack; @@ -768,9 +766,10 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) pin_nid = jack->nid; jack->jack_dirty = 1; - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, pd, eldv); + codec->addr, pin_nid, + !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); pin_idx = pin_nid_to_pin_index(spec, pin_nid); if (pin_idx < 0) @@ -992,7 +991,7 @@ static void hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) if (eld->monitor_present) eld_valid = !!(present & AC_PINSENSE_ELDV); - printk(KERN_INFO + _snd_printd(SND_PR_VERBOSE, "HDMI status: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n", codec->addr, pin_nid, eld->monitor_present, eld_valid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9917e55d6f11..818f90bc7d57 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1445,6 +1445,13 @@ enum { ALC_FIXUP_ACT_BUILD, }; +static void alc_apply_pincfgs(struct hda_codec *codec, + const struct alc_pincfg *cfg) +{ + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); +} + static void alc_apply_fixup(struct hda_codec *codec, int action) { struct alc_spec *spec = codec->spec; @@ -1478,9 +1485,7 @@ static void alc_apply_fixup(struct hda_codec *codec, int action) snd_printdd(KERN_INFO "hda_codec: %s: " "Apply pincfg for %s\n", codec->chip_name, modelname); - for (; cfg->nid; cfg++) - snd_hda_codec_set_pincfg(codec, cfg->nid, - cfg->val); + alc_apply_pincfgs(codec, cfg); break; case ALC_FIXUP_VERBS: if (action != ALC_FIXUP_ACT_PROBE || !fix->v.verbs) @@ -3398,8 +3403,10 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) for (;;) { badness = fill_and_eval_dacs(codec, fill_hardwired, fill_mio_first); - if (badness < 0) + if (badness < 0) { + kfree(best_cfg); return badness; + } debug_badness("==> lo_type=%d, wired=%d, mio=%d, badness=0x%x\n", cfg->line_out_type, fill_hardwired, fill_mio_first, badness); @@ -3434,7 +3441,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_SPEAKER_OUT; fill_hardwired = true; continue; - } + } if (cfg->hp_outs > 0 && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) { cfg->speaker_outs = cfg->line_outs; @@ -3448,7 +3455,7 @@ static int alc_auto_fill_dac_nids(struct hda_codec *codec) cfg->line_out_type = AUTO_PIN_HP_OUT; fill_hardwired = true; continue; - } + } break; } @@ -4423,7 +4430,7 @@ static int alc_parse_auto_config(struct hda_codec *codec, static int alc880_parse_auto_config(struct hda_codec *codec) { static const hda_nid_t alc880_ignore[] = { 0x1d, 0 }; - static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; + static const hda_nid_t alc880_ssids[] = { 0x15, 0x1b, 0x14, 0 }; return alc_parse_auto_config(codec, alc880_ignore, alc880_ssids); } @@ -4859,6 +4866,7 @@ enum { ALC260_FIXUP_GPIO1_TOGGLE, ALC260_FIXUP_REPLACER, ALC260_FIXUP_HP_B1900, + ALC260_FIXUP_KN1, }; static void alc260_gpio1_automute(struct hda_codec *codec) @@ -4886,6 +4894,36 @@ static void alc260_fixup_gpio1_toggle(struct hda_codec *codec, } } +static void alc260_fixup_kn1(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct alc_pincfg pincfgs[] = { + { 0x0f, 0x02214000 }, /* HP/speaker */ + { 0x12, 0x90a60160 }, /* int mic */ + { 0x13, 0x02a19000 }, /* ext mic */ + { 0x18, 0x01446000 }, /* SPDIF out */ + /* disable bogus I/O pins */ + { 0x10, 0x411111f0 }, + { 0x11, 0x411111f0 }, + { 0x14, 0x411111f0 }, + { 0x15, 0x411111f0 }, + { 0x16, 0x411111f0 }, + { 0x17, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { } + }; + + switch (action) { + case ALC_FIXUP_ACT_PRE_PROBE: + alc_apply_pincfgs(codec, pincfgs); + break; + case ALC_FIXUP_ACT_PROBE: + spec->init_amp = ALC_INIT_NONE; + break; + } +} + static const struct alc_fixup alc260_fixups[] = { [ALC260_FIXUP_HP_DC5750] = { .type = ALC_FIXUP_PINS, @@ -4936,7 +4974,11 @@ static const struct alc_fixup alc260_fixups[] = { .v.func = alc260_fixup_gpio1_toggle, .chained = true, .chain_id = ALC260_FIXUP_COEF, - } + }, + [ALC260_FIXUP_KN1] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc260_fixup_kn1, + }, }; static const struct snd_pci_quirk alc260_fixup_tbl[] = { @@ -4946,6 +4988,7 @@ static const struct snd_pci_quirk alc260_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280a, "HP dc5750", ALC260_FIXUP_HP_DC5750), SND_PCI_QUIRK(0x103c, 0x30ba, "HP Presario B1900", ALC260_FIXUP_HP_B1900), SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FIXUP_GPIO1), + SND_PCI_QUIRK(0x152d, 0x0729, "Quanta KN1", ALC260_FIXUP_KN1), SND_PCI_QUIRK(0x161f, 0x2057, "Replacer 672V", ALC260_FIXUP_REPLACER), SND_PCI_QUIRK(0x1631, 0xc017, "PB V7900", ALC260_FIXUP_COEF), {} @@ -5269,7 +5312,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x16, 0x99130111 }, /* CLFE speaker */ { 0x17, 0x99130112 }, /* surround speaker */ { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC882_FIXUP_ACER_ASPIRE_8930G] = { .type = ALC_FIXUP_PINS, @@ -5312,7 +5357,9 @@ static const struct alc_fixup alc882_fixups[] = { { 0x20, AC_VERB_SET_COEF_INDEX, 0x07 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x3050 }, { } - } + }, + .chained = true, + .chain_id = ALC882_FIXUP_GPIO1, }, [ALC885_FIXUP_MACPRO_GPIO] = { .type = ALC_FIXUP_FUNC, @@ -5359,6 +5406,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { ALC882_FIXUP_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x0155, "Packard-Bell M5120", ALC882_FIXUP_PB_M5210), SND_PCI_QUIRK(0x1025, 0x0259, "Acer Aspire 5935", ALC889_FIXUP_DAC_ROUTE), + SND_PCI_QUIRK(0x1025, 0x026b, "Acer Aspire 8940G", ALC882_FIXUP_ACER_ASPIRE_8930G), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", ALC882_FIXUP_ACER_ASPIRE_7736), SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), @@ -5384,6 +5432,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4100, "Macmini 3,1", ALC889_FIXUP_IMAC91_VREF), + SND_PCI_QUIRK(0x106b, 0x4200, "Mac Pro 5,1", ALC885_FIXUP_MACPRO_GPIO), SND_PCI_QUIRK(0x106b, 0x4600, "MacbookPro 5,2", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC889_FIXUP_IMAC91_VREF), SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC889_FIXUP_IMAC91_VREF), @@ -5399,6 +5448,13 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { {} }; +static const struct alc_model_fixup alc882_fixup_models[] = { + {.id = ALC882_FIXUP_ACER_ASPIRE_4930G, .name = "acer-aspire-4930g"}, + {.id = ALC882_FIXUP_ACER_ASPIRE_8930G, .name = "acer-aspire-8930g"}, + {.id = ALC883_FIXUP_ACER_EAPD, .name = "acer-aspire"}, + {} +}; + /* * BIOS auto configuration */ @@ -5439,7 +5495,8 @@ static int patch_alc882(struct hda_codec *codec) if (err < 0) goto error; - alc_pick_fixup(codec, NULL, alc882_fixup_tbl, alc882_fixups); + alc_pick_fixup(codec, alc882_fixup_models, alc882_fixup_tbl, + alc882_fixups); alc_apply_fixup(codec, ALC_FIXUP_ACT_PRE_PROBE); alc_auto_parse_customize_define(codec); @@ -6052,6 +6109,7 @@ static const struct alc_fixup alc269_fixups[] = { static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS P901", ALC269_FIXUP_STEREO_DMIC), @@ -6079,7 +6137,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_FIXUP_AMIC), SND_PCI_QUIRK(0x1043, 0x1013, "ASUS N61Da", ALC269_FIXUP_AMIC), @@ -6296,7 +6354,7 @@ static void alc_fixup_no_jack_detect(struct hda_codec *codec, { if (action == ALC_FIXUP_ACT_PRE_PROBE) codec->no_jack_detect = 1; -} +} static const struct alc_fixup alc861_fixups[] = { [ALC861_FIXUP_FSC_AMILO_PI1505] = { @@ -6714,7 +6772,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { * Basically the device should work as is without the fixup table. * If BIOS doesn't give a proper info, enable the corresponding * fixup entry. - */ + */ SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC662_FIXUP_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC662_FIXUP_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1173, "ASUS K73Jn", ALC662_FIXUP_ASUS_MODE1), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 33a9946b492c..4742cac26aa9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5063,12 +5063,11 @@ static void stac92xx_update_led_status(struct hda_codec *codec, int enabled) if (spec->gpio_led_polarity) muted = !muted; - /*polarity defines *not* muted state level*/ if (!spec->vref_mute_led_nid) { if (muted) - spec->gpio_data &= ~spec->gpio_led; /* orange */ + spec->gpio_data |= spec->gpio_led; else - spec->gpio_data |= spec->gpio_led; /* white */ + spec->gpio_data &= ~spec->gpio_led; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); } else { diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c index df3ac73f8778..b39ad356b92b 100644 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ b/sound/soc/blackfin/bf5xx-ssm2602.c @@ -99,6 +99,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, { .name = "ssm2602", @@ -108,6 +109,7 @@ static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { .platform_name = "bfin-i2s-pcm-audio", .codec_name = "ssm2602.0-001b", .ops = &bf5xx_ssm2602_ops, + .dai_fmt = BF5XX_SSM2602_DAIFMT, }, }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6508e8b790bb..59d8efaa17e9 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,7 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE - select SND_SOC_TWL6040 if TWL4030_CORE + select SND_SOC_TWL6040 if TWL6040_CORE select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE @@ -276,7 +276,6 @@ config SND_SOC_TWL4030 tristate config SND_SOC_TWL6040 - select TWL6040_CORE tristate config SND_SOC_UDA134X diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index f8e10ced244a..b3e24f289421 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -140,7 +140,7 @@ * min : 0xFE : -115.0 dB * mute: 0xFF */ -static const DECLARE_TLV_DB_SCALE(out_tlv, -11500, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -11550, 50, 1); static const struct snd_kcontrol_new ak4642_snd_controls[] = { diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 78979b3e0e95..07c44b71f096 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -929,6 +929,8 @@ static int cs42l73_set_mclk(struct snd_soc_dai *dai, unsigned int freq) /* MCLKX -> MCLK */ mclkx_coeff = cs42l73_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; mclk = cs42l73_mclkx_coeffs[mclkx_coeff].mclkx / cs42l73_mclkx_coeffs[mclkx_coeff].ratio; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d1926266fe00..8e92fb88ed09 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -143,11 +143,11 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, } /* - * using codec assist to small pop, hp_powerup or lineout_powerup - * should stay setting until vag_powerup is fully ramped down, - * vag fully ramped down require 400ms. + * As manual described, ADC/DAC only works when VAG powerup, + * So enabled VAG before ADC/DAC up. + * In power down case, we need wait 400ms when vag fully ramped down. */ -static int small_pop_event(struct snd_soc_dapm_widget *w, +static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { @@ -156,7 +156,7 @@ static int small_pop_event(struct snd_soc_dapm_widget *w, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -201,12 +201,8 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { mic_bias_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), - SND_SOC_DAPM_PGA_E("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0, - small_pop_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA("HP", SGTL5000_CHIP_ANA_POWER, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), @@ -221,8 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), + SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, + power_vag_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), }; @@ -231,9 +230,11 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ + {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ + {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 16d55f91a653..df1e07ffac32 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -472,7 +472,7 @@ static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai, static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0xff7f; + u16 reg = snd_soc_read(codec, TLV320AIC23_PWR) & 0x17f; switch (level) { case SND_SOC_BIAS_ON: @@ -491,7 +491,7 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: /* everything off, dac mute, inactive */ snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); - snd_soc_write(codec, TLV320AIC23_PWR, 0xffff); + snd_soc_write(codec, TLV320AIC23_PWR, 0x1ff); break; } codec->dapm.bias_level = level; diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2d8c6b825e57..dc7509b9d53a 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -26,7 +26,6 @@ #include <linux/pm.h> #include <linux/platform_device.h> #include <linux/slab.h> -#include <linux/i2c/twl.h> #include <linux/mfd/twl6040.h> #include <sound/core.h> @@ -1528,7 +1527,7 @@ static int twl6040_resume(struct snd_soc_codec *codec) static int twl6040_probe(struct snd_soc_codec *codec) { struct twl6040_data *priv; - struct twl4030_codec_data *pdata = dev_get_platdata(codec->dev); + struct twl6040_codec_data *pdata = dev_get_platdata(codec->dev); struct platform_device *pdev = container_of(codec->dev, struct platform_device, dev); int ret = 0; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 8c4c9591ec05..aa12c6b6beeb 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -60,7 +60,7 @@ struct wm8350_jack_data { }; struct wm8350_data { - struct snd_soc_codec codec; + struct wm8350 *wm8350; struct wm8350_output out1; struct wm8350_output out2; struct wm8350_jack_data hpl; @@ -1309,7 +1309,7 @@ static void wm8350_hp_work(struct wm8350_data *priv, struct wm8350_jack_data *jack, u16 mask) { - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report; @@ -1342,7 +1342,7 @@ static void wm8350_hpr_work(struct work_struct *work) static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; struct wm8350_jack_data *jack = NULL; switch (irq - wm8350->irq_base) { @@ -1427,7 +1427,7 @@ EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect); static irqreturn_t wm8350_mic_handler(int irq, void *data) { struct wm8350_data *priv = data; - struct wm8350 *wm8350 = priv->codec.control_data; + struct wm8350 *wm8350 = priv->wm8350; u16 reg; int report = 0; @@ -1536,6 +1536,8 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) return -ENOMEM; snd_soc_codec_set_drvdata(codec, priv); + priv->wm8350 = wm8350; + for (i = 0; i < ARRAY_SIZE(supply_names); i++) priv->supplies[i].supply = supply_names[i]; @@ -1544,7 +1546,6 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - wm8350->codec.codec = codec; codec->control_data = wm8350; /* Put the codec into reset if it wasn't already */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7c49642af052..6c1fe3afd4b5 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1000,61 +1000,170 @@ static void wm8994_update_class_w(struct snd_soc_codec *codec) } } -static int late_enable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; + int mask = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC1R_ENA; + int dac; + int adc; + int val; + + switch (control->type) { + case WM8994: + case WM8958: + mask |= WM8994_AIF1DAC2L_ENA | WM8994_AIF1DAC2R_ENA; + break; + default: + break; + } switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (wm8994->aif1clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, - WM8994_AIF1CLK_ENA); - wm8994->aif1clk_enable = 0; - } - if (wm8994->aif2clk_enable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, - WM8994_AIF2CLK_ENA); - wm8994->aif2clk_enable = 0; - } + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_1); + if ((val & WM8994_AIF1ADCL_SRC) && + (val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA; + else if (!(val & WM8994_AIF1ADCL_SRC) && + !(val & WM8994_AIF1ADCR_SRC)) + adc = WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + else + adc = WM8994_AIF1ADC1R_ENA | WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC1L_ENA | WM8994_AIF1ADC2L_ENA; + + val = snd_soc_read(codec, WM8994_AIF1_CONTROL_2); + if ((val & WM8994_AIF1DACL_SRC) && + (val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA; + else if (!(val & WM8994_AIF1DACL_SRC) && + !(val & WM8994_AIF1DACR_SRC)) + dac = WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + else + dac = WM8994_AIF1DAC1R_ENA | WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC1L_ENA | WM8994_AIF1DAC2L_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF1DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, mask, + WM8994_AIF1ADC1R_ENA | + WM8994_AIF1ADC1L_ENA | + WM8994_AIF1ADC2R_ENA | + WM8994_AIF1ADC2L_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, mask, + WM8994_AIF1DAC1R_ENA | + WM8994_AIF1DAC1L_ENA | + WM8994_AIF1DAC2R_ENA | + WM8994_AIF1DAC2L_ENA); break; - } - /* We may also have postponed startup of DSP, handle that. */ - wm8958_aif_ev(w, kcontrol, event); + case SND_SOC_DAPM_PRE_PMD: + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + mask, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + mask, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF2DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF1DSPCLK_ENA, val); + break; + } return 0; } -static int late_disable_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; - struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + int dac; + int adc; + int val; switch (event) { + case SND_SOC_DAPM_PRE_PMU: + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_1); + if ((val & WM8994_AIF2ADCL_SRC) && + (val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCR_ENA; + else if (!(val & WM8994_AIF2ADCL_SRC) && + !(val & WM8994_AIF2ADCR_SRC)) + adc = WM8994_AIF2ADCL_ENA; + else + adc = WM8994_AIF2ADCL_ENA | WM8994_AIF2ADCR_ENA; + + + val = snd_soc_read(codec, WM8994_AIF2_CONTROL_2); + if ((val & WM8994_AIF2DACL_SRC) && + (val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACR_ENA; + else if (!(val & WM8994_AIF2DACL_SRC) && + !(val & WM8994_AIF2DACR_SRC)) + dac = WM8994_AIF2DACL_ENA; + else + dac = WM8994_AIF2DACL_ENA | WM8994_AIF2DACR_ENA; + + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, adc); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, dac); + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA, + WM8994_AIF2DSPCLK_ENA | + WM8994_SYSDSPCLK_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_4, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA); + break; + + case SND_SOC_DAPM_PRE_PMD: case SND_SOC_DAPM_POST_PMD: - if (wm8994->aif1clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, - WM8994_AIF1CLK_ENA_MASK, 0); - wm8994->aif1clk_disable = 0; - } - if (wm8994->aif2clk_disable) { - snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, - WM8994_AIF2CLK_ENA_MASK, 0); - wm8994->aif2clk_disable = 0; - } + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2DACL_ENA | + WM8994_AIF2DACR_ENA, 0); + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_5, + WM8994_AIF2ADCL_ENA | + WM8994_AIF2ADCR_ENA, 0); + + val = snd_soc_read(codec, WM8994_CLOCKING_1); + if (val & WM8994_AIF1DSPCLK_ENA) + val = WM8994_SYSDSPCLK_ENA; + else + val = 0; + snd_soc_update_bits(codec, WM8994_CLOCKING_1, + WM8994_SYSDSPCLK_ENA | + WM8994_AIF2DSPCLK_ENA, val); break; } return 0; } -static int aif1clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif1clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1071,8 +1180,8 @@ static int aif1clk_ev(struct snd_soc_dapm_widget *w, return 0; } -static int aif2clk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) +static int aif2clk_late_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec = w->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); @@ -1089,6 +1198,63 @@ static int aif2clk_ev(struct snd_soc_dapm_widget *w, return 0; } +static int late_enable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (wm8994->aif1clk_enable) { + aif1clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, + WM8994_AIF1CLK_ENA); + wm8994->aif1clk_enable = 0; + } + if (wm8994->aif2clk_enable) { + aif2clk_ev(w, kcontrol, event); + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, + WM8994_AIF2CLK_ENA); + wm8994->aif2clk_enable = 0; + } + break; + } + + /* We may also have postponed startup of DSP, handle that. */ + wm8958_aif_ev(w, kcontrol, event); + + return 0; +} + +static int late_disable_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMD: + if (wm8994->aif1clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1, + WM8994_AIF1CLK_ENA_MASK, 0); + aif1clk_ev(w, kcontrol, event); + wm8994->aif1clk_disable = 0; + } + if (wm8994->aif2clk_disable) { + snd_soc_update_bits(codec, WM8994_AIF2_CLOCKING_1, + WM8994_AIF2CLK_ENA_MASK, 0); + aif2clk_ev(w, kcontrol, event); + wm8994->aif2clk_disable = 0; + } + break; + } + + return 0; +} + static int adc_mux_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -1385,9 +1551,9 @@ static const struct snd_kcontrol_new aif2dacr_src_mux = SOC_DAPM_ENUM("AIF2DACR Mux", aif2dacr_src_enum); static const struct snd_soc_dapm_widget wm8994_lateclk_revd_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_ev, +SND_SOC_DAPM_SUPPLY("AIF1CLK", SND_SOC_NOPM, 0, 0, aif1clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), -SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_ev, +SND_SOC_DAPM_SUPPLY("AIF2CLK", SND_SOC_NOPM, 0, 0, aif2clk_late_ev, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_PGA_E("Late DAC1L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, @@ -1416,8 +1582,10 @@ SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { -SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, aif1clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, aif2clk_ev, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), @@ -1470,30 +1638,30 @@ SND_SOC_DAPM_SUPPLY("VMID", SND_SOC_NOPM, 0, 0, vmid_event, SND_SOC_DAPM_SUPPLY("CLK_SYS", SND_SOC_NOPM, 0, 0, clk_sys_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), -SND_SOC_DAPM_SUPPLY("DSP1CLK", WM8994_CLOCKING_1, 3, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSP2CLK", WM8994_CLOCKING_1, 2, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("DSPINTCLK", WM8994_CLOCKING_1, 1, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP1CLK", SND_SOC_NOPM, 3, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSP2CLK", SND_SOC_NOPM, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("DSPINTCLK", SND_SOC_NOPM, 1, 0, NULL, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 9, 0), + 0, SND_SOC_NOPM, 9, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC1R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 8, 0), + 0, SND_SOC_NOPM, 8, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 9, 0, wm8958_aif_ev, + SND_SOC_NOPM, 9, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC1R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 8, 0, wm8958_aif_ev, + SND_SOC_NOPM, 8, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_OUT("AIF1ADC2L", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 11, 0), + 0, SND_SOC_NOPM, 11, 0), SND_SOC_DAPM_AIF_OUT("AIF1ADC2R", NULL, - 0, WM8994_POWER_MANAGEMENT_4, 10, 0), + 0, SND_SOC_NOPM, 10, 0), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2L", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 11, 0, wm8958_aif_ev, + SND_SOC_NOPM, 11, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_AIF_IN_E("AIF1DAC2R", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 10, 0, wm8958_aif_ev, + SND_SOC_NOPM, 10, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_MIXER("AIF1ADC1L Mixer", SND_SOC_NOPM, 0, 0, @@ -1520,14 +1688,14 @@ SND_SOC_DAPM_MIXER("DAC1R Mixer", SND_SOC_NOPM, 0, 0, dac1r_mix, ARRAY_SIZE(dac1r_mix)), SND_SOC_DAPM_AIF_OUT("AIF2ADCL", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 13, 0), + SND_SOC_NOPM, 13, 0), SND_SOC_DAPM_AIF_OUT("AIF2ADCR", NULL, 0, - WM8994_POWER_MANAGEMENT_4, 12, 0), + SND_SOC_NOPM, 12, 0), SND_SOC_DAPM_AIF_IN_E("AIF2DACL", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 13, 0, wm8958_aif_ev, + SND_SOC_NOPM, 13, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN_E("AIF2DACR", NULL, 0, - WM8994_POWER_MANAGEMENT_5, 12, 0, wm8958_aif_ev, + SND_SOC_NOPM, 12, 0, wm8958_aif_ev, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_AIF_IN("AIF1DACDAT", NULL, 0, SND_SOC_NOPM, 0, 0), diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index f13f2886339c..6c028c470601 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1035,7 +1035,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); - int val; + int mask, val; switch (level) { case SND_SOC_BIAS_STANDBY: @@ -1047,6 +1047,13 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_ON: /* Turn off any unneded single ended outputs */ val = 0; + mask = 0; + + if (hubs->lineout1_se) + mask |= WM8993_LINEOUT1N_ENA | WM8993_LINEOUT1P_ENA; + + if (hubs->lineout2_se) + mask |= WM8993_LINEOUT2N_ENA | WM8993_LINEOUT2P_ENA; if (hubs->lineout1_se && hubs->lineout1n_ena) val |= WM8993_LINEOUT1N_ENA; @@ -1061,11 +1068,7 @@ void wm_hubs_set_bias_level(struct snd_soc_codec *codec, val |= WM8993_LINEOUT2P_ENA; snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_3, - WM8993_LINEOUT1N_ENA | - WM8993_LINEOUT1P_ENA | - WM8993_LINEOUT2N_ENA | - WM8993_LINEOUT2P_ENA, - val); + mask, val); /* Remove the input clamps */ snd_soc_update_bits(codec, WM8993_INPUTS_CLAMP_REG, diff --git a/sound/soc/imx/imx-audmux.c b/sound/soc/imx/imx-audmux.c index 1765a197acb0..f23700359c67 100644 --- a/sound/soc/imx/imx-audmux.c +++ b/sound/soc/imx/imx-audmux.c @@ -73,6 +73,9 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, if (!buf) return -ENOMEM; + if (!audmux_base) + return -ENOSYS; + if (audmux_clk) clk_prepare_enable(audmux_clk); @@ -152,7 +155,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 1; i < 8; i++) { + for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e00dd0b1139c..deafbfaacdbf 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -97,7 +97,7 @@ config SND_OMAP_SOC_SDP3430 config SND_OMAP_SOC_OMAP_ABE_TWL6040 tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec" - depends on TWL4030_CORE && SND_OMAP_SOC && ARCH_OMAP4 + depends on TWL6040_CORE && SND_OMAP_SOC && ARCH_OMAP4 select SND_OMAP_SOC_DMIC select SND_OMAP_SOC_MCPDM select SND_SOC_TWL6040 diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index a59bd352d342..5a649da9122a 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -401,6 +401,10 @@ static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) } out: + /* free preallocated buffers in case of error */ + if (ret) + omap_pcm_free_dma_buffers(pcm); + return ret; } diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 609abd51e55f..d08583790d23 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -17,6 +17,7 @@ #include <linux/delay.h> #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 72185078ddf8..79fbeea99d46 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -166,7 +166,7 @@ static struct snd_soc_dai_driver s3c2412_i2s_dai = { static __devinit int s3c2412_iis_dev_probe(struct platform_device *pdev) { - return snd_soc_register_dai(&pdev->dev, &s3c2412_i2s_dai); + return s3c_i2sv2_register_dai(&pdev->dev, -1, &s3c2412_i2s_dai); } static __devexit int s3c2412_iis_dev_remove(struct platform_device *pdev) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 378cc5b056d7..74ed2dffbffd 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1001,11 +1001,10 @@ static void fsi_dma_do_tasklet(unsigned long data) sg_dma_address(&sg) = buf; sg_dma_len(&sg) = len; - desc = chan->device->device_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | - DMA_CTRL_ACK); + desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { - dev_err(dai->dev, "device_prep_slave_sg() fail\n"); + dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e19c24ade414..c88d9741b9e7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1081,6 +1081,8 @@ static int soc_probe_platform(struct snd_soc_card *card, snd_soc_dapm_new_controls(&platform->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + platform->dapm.idle_bias_off = 1; + if (driver->probe) { ret = driver->probe(platform); if (ret < 0) { @@ -3111,6 +3113,7 @@ int snd_soc_register_card(struct snd_soc_card *card) GFP_KERNEL); if (card->rtd == NULL) return -ENOMEM; + card->num_rtd = 0; card->rtd_aux = &card->rtd[card->num_links]; for (i = 0; i < card->num_links; i++) @@ -3622,10 +3625,10 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, int i, ret; num_routes = of_property_count_strings(np, propname); - if (num_routes & 1) { + if (num_routes < 0 || num_routes & 1) { dev_err(card->dev, - "Property '%s's length is not even\n", - propname); + "Property '%s' does not exist or its length is not even\n", + propname); return -EINVAL; } num_routes /= 2; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 5cbd2d7623b8..1bb6d4a63cd8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -67,6 +67,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_out_drv] = 10, [snd_soc_dapm_hp] = 10, [snd_soc_dapm_spk] = 10, + [snd_soc_dapm_line] = 10, [snd_soc_dapm_post] = 11, }; @@ -75,6 +76,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_adc] = 1, [snd_soc_dapm_hp] = 2, [snd_soc_dapm_spk] = 2, + [snd_soc_dapm_line] = 2, [snd_soc_dapm_out_drv] = 2, [snd_soc_dapm_pga] = 4, [snd_soc_dapm_mixer_named_ctl] = 5, diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 33509de52540..e53349912b2e 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -79,11 +79,15 @@ static int tegra_i2s_show(struct seq_file *s, void *unused) struct tegra_i2s *i2s = s->private; int i; + clk_enable(i2s->clk_i2s); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_i2s_read(i2s, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(i2s->clk_i2s); + return 0; } @@ -112,7 +116,7 @@ static void tegra_i2s_debug_remove(struct tegra_i2s *i2s) debugfs_remove(i2s->debug); } #else -static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s, int id) +static inline void tegra_i2s_debug_add(struct tegra_i2s *i2s) { } diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c index 475428cf270e..9ff2c601445f 100644 --- a/sound/soc/tegra/tegra_spdif.c +++ b/sound/soc/tegra/tegra_spdif.c @@ -79,11 +79,15 @@ static int tegra_spdif_show(struct seq_file *s, void *unused) struct tegra_spdif *spdif = s->private; int i; + clk_enable(spdif->clk_spdif_out); + for (i = 0; i < ARRAY_SIZE(regs); i++) { u32 val = tegra_spdif_read(spdif, regs[i].offset); seq_printf(s, "%s = %08x\n", regs[i].name, val); } + clk_disable(spdif->clk_spdif_out); + return 0; } |