October 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Arecord issue
by kartik natarajan 11 Oct '12

11 Oct '12

Hi, I am working on a 2.6.37 kernel and on a DM365 davinci board with a tlv320aic3254 audio codec (i2c control lines, i2s data transfer) I wanted to get the audio up on my board so got the alsa-lib and alsa-utils 1.0.25 compiled them and got the aplay to work fine. When I got to working on the arecord things haven't worked fine. I have traced all the calls and have reached so far; codec gets set properly davinci_pcm_trigger executes with success snd_pcm_sw_params() executes successfully. aplay calls begin_wave() aplay calls pcm_read() that invokes snd_pcm_readi() Finally my app seems to not go beyond the snd_pcm_start() Any assistance shall be highly appreciated! Thanks in advance, N.Kartik

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[alsa-devel] [PATCH] ASoC: codecs: da9055: Minor improvement in ALC calibration process
by Ashish Chavan 11 Oct '12

11 Oct '12

This patch slightly improves ALC calibration process of da9055 codec driver by muting Mic PGAs during calibration. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 22 +++++++++++++++++++++- 1 files changed, 21 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 27c4454..abf3e43 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -179,6 +179,12 @@ #define DA9055_AIF_WORD_S24_LE (2 << 2) #define DA9055_AIF_WORD_S32_LE (3 << 2) +/* MIC_L_CTRL bit fields */ +#define DA9055_MIC_L_MUTE_EN (1 << 6) + +/* MIC_R_CTRL bit fields */ +#define DA9055_MIC_R_MUTE_EN (1 << 6) + /* MIXIN_L_CTRL bit fields */ #define DA9055_MIXIN_L_MIX_EN (1 << 3) @@ -477,7 +483,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u8 reg_val, adc_left, adc_right; + u8 reg_val, adc_left, adc_right, mic_left, mic_right; int avg_left_data, avg_right_data, offset_l, offset_r; if (ucontrol->value.integer.value[0]) { @@ -486,6 +492,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, * offsets must be done first */ + /* Save current values from Mic control registers */ + mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL); + mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL); + + /* Mute Mic PGA Left and Right */ + snd_soc_update_bits(codec, DA9055_MIC_L_CTRL, + DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_MIC_R_CTRL, + DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN); + /* Save current values from ADC control registers */ adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); @@ -521,6 +537,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, /* Restore original values of ADC control registers */ snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + + /* Restore original values of Mic control registers */ + snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left); + snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right); } return snd_soc_put_volsw(kcontrol, ucontrol); -- 1.7.1

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[alsa-devel] [PATCH] ASoC: codecs: da9055: Add support for DSP mode of AIF
by Ashish Chavan 11 Oct '12

11 Oct '12

This patch adds AIF DSP mode support for da9055 codec driver. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 4 ++++ 1 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 185d8dd..27c4454 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -173,6 +173,7 @@ #define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) #define DA9055_AIF_FORMAT_LEFT_J (1 << 0) #define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_FORMAT_DSP (3 << 0) #define DA9055_AIF_WORD_S16_LE (0 << 2) #define DA9055_AIF_WORD_S20_3LE (1 << 2) #define DA9055_AIF_WORD_S24_LE (2 << 2) @@ -1155,6 +1156,9 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_RIGHT_J: aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; break; + case SND_SOC_DAIFMT_DSP_A: + aif_ctrl = DA9055_AIF_FORMAT_DSP; + break; default: return -EINVAL; } -- 1.7.1

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[alsa-devel] [PATCH V7 12/15] ALSA: HDA: Make hda sound card usable for Loongson
by Huacai Chen 11 Oct '12

11 Oct '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Both A1004 and A1205 use the same pin configurations, but A1004 need to increase the default boost of internal mic. Signed-off-by: Jie Chen <chenj(a)lemote.com> Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- sound/pci/hda/patch_conexant.c | 44 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..3cc265e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4408,7 +4408,10 @@ static const struct hda_codec_ops cx_auto_patch_ops = { enum { CXT_PINCFG_LENOVO_X200, CXT_PINCFG_LENOVO_TP410, + CXT_PINCFG_LEMOTE_A1004, + CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, + CXT_FIXUP_INC_MIC_BOOST, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4418,6 +4421,19 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt5066_increase_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_amp_caps(codec, 0x17, HDA_OUTPUT, + (0x3 << AC_AMPCAP_OFFSET_SHIFT) | + (0x4 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -4435,6 +4451,18 @@ static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = { {} }; +/* Lemote A1004/A1205 with cxt5066 */ +static const struct hda_pintbl cxt_pincfg_lemote[] = { + { 0x1a, 0x90a10020 }, /* Internal mic */ + { 0x1b, 0x03a11020 }, /* External mic */ + { 0x1d, 0x400101f0 }, /* Not used */ + { 0x1e, 0x40a701f0 }, /* Not used */ + { 0x20, 0x404501f0 }, /* Not used */ + { 0x22, 0x404401f0 }, /* Not used */ + { 0x23, 0x40a701f0 }, /* Not used */ + {} +}; + static const struct hda_fixup cxt_fixups[] = { [CXT_PINCFG_LENOVO_X200] = { .type = HDA_FIXUP_PINS, @@ -4444,10 +4472,24 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lenovo_tp410, }, + [CXT_PINCFG_LEMOTE_A1004] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_INC_MIC_BOOST, + .v.pins = cxt_pincfg_lemote, + }, + [CXT_PINCFG_LEMOTE_A1205] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lemote, + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, }, + [CXT_FIXUP_INC_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt5066_increase_mic_boost, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4463,6 +4505,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), + SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} }; -- 1.7.7.3

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[alsa-devel] [PATCH] ASoC: codecs: da9055: Minor improvements
by Ashish Chavan 11 Oct '12

11 Oct '12

This patch includes two minor improvements for da9055 codec driver (1) Adding support for DSP mode of AIF (2) Muting Mic PGAs during ALC calibration process Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <david.chen(a)diasemi.com> --- sound/soc/codecs/da9055.c | 26 +++++++++++++++++++++++++- 1 files changed, 25 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 185d8dd..abf3e43 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -173,11 +173,18 @@ #define DA9055_AIF_FORMAT_I2S_MODE (0 << 0) #define DA9055_AIF_FORMAT_LEFT_J (1 << 0) #define DA9055_AIF_FORMAT_RIGHT_J (2 << 0) +#define DA9055_AIF_FORMAT_DSP (3 << 0) #define DA9055_AIF_WORD_S16_LE (0 << 2) #define DA9055_AIF_WORD_S20_3LE (1 << 2) #define DA9055_AIF_WORD_S24_LE (2 << 2) #define DA9055_AIF_WORD_S32_LE (3 << 2) +/* MIC_L_CTRL bit fields */ +#define DA9055_MIC_L_MUTE_EN (1 << 6) + +/* MIC_R_CTRL bit fields */ +#define DA9055_MIC_R_MUTE_EN (1 << 6) + /* MIXIN_L_CTRL bit fields */ #define DA9055_MIXIN_L_MIX_EN (1 << 3) @@ -476,7 +483,7 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); - u8 reg_val, adc_left, adc_right; + u8 reg_val, adc_left, adc_right, mic_left, mic_right; int avg_left_data, avg_right_data, offset_l, offset_r; if (ucontrol->value.integer.value[0]) { @@ -485,6 +492,16 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, * offsets must be done first */ + /* Save current values from Mic control registers */ + mic_left = snd_soc_read(codec, DA9055_MIC_L_CTRL); + mic_right = snd_soc_read(codec, DA9055_MIC_R_CTRL); + + /* Mute Mic PGA Left and Right */ + snd_soc_update_bits(codec, DA9055_MIC_L_CTRL, + DA9055_MIC_L_MUTE_EN, DA9055_MIC_L_MUTE_EN); + snd_soc_update_bits(codec, DA9055_MIC_R_CTRL, + DA9055_MIC_R_MUTE_EN, DA9055_MIC_R_MUTE_EN); + /* Save current values from ADC control registers */ adc_left = snd_soc_read(codec, DA9055_ADC_L_CTRL); adc_right = snd_soc_read(codec, DA9055_ADC_R_CTRL); @@ -520,6 +537,10 @@ static int da9055_put_alc_sw(struct snd_kcontrol *kcontrol, /* Restore original values of ADC control registers */ snd_soc_write(codec, DA9055_ADC_L_CTRL, adc_left); snd_soc_write(codec, DA9055_ADC_R_CTRL, adc_right); + + /* Restore original values of Mic control registers */ + snd_soc_write(codec, DA9055_MIC_L_CTRL, mic_left); + snd_soc_write(codec, DA9055_MIC_R_CTRL, mic_right); } return snd_soc_put_volsw(kcontrol, ucontrol); @@ -1155,6 +1176,9 @@ static int da9055_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) case SND_SOC_DAIFMT_RIGHT_J: aif_ctrl = DA9055_AIF_FORMAT_RIGHT_J; break; + case SND_SOC_DAIFMT_DSP_A: + aif_ctrl = DA9055_AIF_FORMAT_DSP; + break; default: return -EINVAL; } -- 1.7.1

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[alsa-devel] Bug 48381 - [Regression][Bisected]Sound gets permanently muted a few minutes after booting
by Da Fox 10 Oct '12

10 Oct '12

Hi all, Since the official bug-tracker at https://bugtrack.alsa-project.org/alsa-bug/ seems to be down (and moreover this appears to be a kernel bug), I have filed a bug at the kernel's bug-tracker last week ( https://bugzilla.kernel.org/show_bug.cgi?id=48381 ). However, it seems to have gone unnoticed thus-far. Quick summary of the issue: After a while of inactivity the sound-card seems to go into a sort of power-saving mode (you can hear a faint 'pop' from the speakers). Before the sound would automatically turn back on again as soon as a program attempted to play sound. However now it seems that after resuming from the power-saving mode the driver thinks that there is always a headphone connected, even when it is not. I think this is what happens because toggling the 'Auto-Mute Mode' (setting it to 'Disabled') in alsamixer allows the speakers to produce sound again. This is all on a laptop (please see the linked bug report for more detailed hardware description). Would a developer please take a look at it? I have bisected the issue down to a commit made by David Henningsson and/or Takashi Iwai: ---8<--------- 80c8bfbe76869bfd6bdf3d260d316e7a32f318c3 is the first bad commit commit 80c8bfbe76869bfd6bdf3d260d316e7a32f318c3 Author: David Henningsson <david.henningsson(a)canonical.com> Date: Mon Jun 4 09:33:51 2012 +0200 ALSA: HDA: Create phantom jacks for fixed inputs and outputs PulseAudio sometimes have difficulties knowing that there is a "Speaker" or "Internal Mic", if they have no individual volume controls or selectors. As a result, only e g "Headphone" might be created for a laptop, but no "Speaker". To help out, create phantom jacks (that are always present, at least for now) for "Speaker", "Internal Mic" etc, in case we detect them. The naming convention is e g "Speaker Phantom Jack". In order not to pollute the /dev/input namespace with even more devices, these are added to the kcontrols only, not the input devices. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> :040000 040000 7a954b731bef80cee731763521f64a424a7058a0 7d176b02ebb4553b0b0f97d200f7527cb69c122a M sound --->8--------- If any additional information is required please let me know. Best regards, Da Fox

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[alsa-devel] ASoC: special SPI format for that5173 driver in daisy chain mode
by Christian Schörkhuber 10 Oct '12

10 Oct '12

Hi all, I'm working on an ASoC driver for the THAT5173 microphone preamplifier which is controlled via SPI. When operating in daisy chain mode (4 amps) the desired format is a bit odd: I need to write four 16 bit spi commands (one gain setting for each amp) in a row without raising the chipselect line between commands. That is, I need to write one 64 bit SPI command that is composed of 4 different values (and 4 register addresses which are all 0). Usually I use regmap but I don't see how I can construct such a message with regmap. How can this be done (with or without regmap)? Thanks, Chris

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[alsa-devel] [PATCH] ASoC: fsl/pcm030 add unique dai_link names
by Eric Millbrandt 10 Oct '12

10 Oct '12

From: Eric Millbrandt <emillbrandt(a)dekaresearch.com Assign each dai_link a unique name to avoid this run-time error. [ 18.978043] pcm030-audio-fabric sound.2: wm9712-hifi <-> mpc5200-psc-ac97.0 mapping ok [ 19.003179] sysfs: cannot create duplicate filename '/devices/sound.2/AC97' Signed-off-by: Eric Millbrandt <emillbrandt(a)dekaresearch.com> --- sound/soc/fsl/pcm030-audio-fabric.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 4b63ec8..4d26192 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -29,14 +29,14 @@ struct pcm030_audio_data { static struct snd_soc_dai_link pcm030_fabric_dai[] = { { - .name = "AC97", + .name = "AC97.0", .stream_name = "AC97 Analog", .codec_dai_name = "wm9712-hifi", .cpu_dai_name = "mpc5200-psc-ac97.0", .codec_name = "wm9712-codec", }, { - .name = "AC97", + .name = "AC97.1", .stream_name = "AC97 IEC958", .codec_dai_name = "wm9712-aux", .cpu_dai_name = "mpc5200-psc-ac97.1", -- 1.7.8

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[alsa-devel] [PATCH] ALSA: ASoC: davinci-pcm: stub out gen_pool functions for !CONFIG_GENERIC_ALLOCATOR
by Daniel Mack 10 Oct '12

10 Oct '12

On platforms without CONFIG_GENERIC_ALLOCATOR, these functions can be no-ops. The code that uses the memory buffers does the right thing if they are NULL. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> Cc: Matt Porter <mporter(a)ti.com> Cc: Anil Kumar <anilkumar.v(a)ti.com> --- Alternatively, the gen_pool functions could stub out themselves for !CONFIG_GENERIC_ALLOCATOR. sound/soc/davinci/davinci-pcm.c | 29 ++++++++++++++++++++++++----- 1 file changed, 24 insertions(+), 5 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 01b3f17..afab81f 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -251,6 +251,7 @@ static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) } } +#ifdef CONFIG_GENERIC_ALLOCATOR static int allocate_sram(struct snd_pcm_substream *substream, struct gen_pool *sram_pool, unsigned size, struct snd_pcm_hardware *ppcm) @@ -284,6 +285,28 @@ exit1: return -ENOMEM; } +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ + struct davinci_runtime_data *prtd = substream->runtime->private_data; + struct gen_pool *sram_pool = prtd->params->sram_pool; + + gen_pool_free(sram_pool, (unsigned) iram_dma->area, iram_dma->bytes); +} +#else +static int allocate_sram(struct snd_pcm_substream *substream, + struct gen_pool *sram_pool, unsigned size, + struct snd_pcm_hardware *ppcm) +{ + return 0; +} + +static void davinci_free_sram(struct snd_pcm_substream *substream, + struct snd_dma_buffer *iram_dma) +{ +} +#endif + /* * Only used with ping/pong. * This is called after runtime->dma_addr, period_bytes and data_type are valid @@ -813,11 +836,7 @@ static void davinci_pcm_free(struct snd_pcm *pcm) buf->area = NULL; iram_dma = buf->private_data; if (iram_dma) { - struct davinci_runtime_data *prtd = - substream->runtime->private_data; - struct gen_pool *sram_pool = prtd->params->sram_pool; - gen_pool_free(sram_pool, (unsigned)iram_dma->area, - iram_dma->bytes); + davinci_free_sram(substream, iram_dma); kfree(iram_dma); } } -- 1.7.11.4

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[alsa-devel] [PATCH] ASoC: wm8958: Load firmwares simultaneously
by Mark Brown 10 Oct '12

10 Oct '12

As of commit 99c2aa (firmware loader: fix creation failure of fw loader device) we can have more than one firmware request outstanding at once so there is no need to daisychain our requests any more. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8958-dsp2.c | 31 +++++++++++-------------------- 1 file changed, 11 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index 2f191a0..b0710d8 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -883,13 +883,6 @@ static void wm8958_mbc_vss_loaded(const struct firmware *fw, void *context) wm8994->mbc_vss = fw; mutex_unlock(&codec->mutex); } - - /* We can't have more than one request outstanding at once so - * we daisy chain. - */ - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, - "wm8958_enh_eq.wfw", codec->dev, GFP_KERNEL, - codec, wm8958_enh_eq_loaded); } static void wm8958_mbc_loaded(const struct firmware *fw, void *context) @@ -897,19 +890,11 @@ static void wm8958_mbc_loaded(const struct firmware *fw, void *context) struct snd_soc_codec *codec = context; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - if (wm8958_dsp2_fw(codec, "MBC", fw, true) != 0) - return; - - mutex_lock(&codec->mutex); - wm8994->mbc = fw; - mutex_unlock(&codec->mutex); - - /* We can't have more than one request outstanding at once so - * we daisy chain. - */ - request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, - "wm8958_mbc_vss.wfw", codec->dev, GFP_KERNEL, - codec, wm8958_mbc_vss_loaded); + if (fw && (wm8958_dsp2_fw(codec, "MBC", fw, true) == 0)) { + mutex_lock(&codec->mutex); + wm8994->mbc = fw; + mutex_unlock(&codec->mutex); + } } void wm8958_dsp2_init(struct snd_soc_codec *codec) @@ -933,6 +918,12 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, "wm8958_mbc.wfw", codec->dev, GFP_KERNEL, codec, wm8958_mbc_loaded); + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_mbc_vss.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_mbc_vss_loaded); + request_firmware_nowait(THIS_MODULE, FW_ACTION_HOTPLUG, + "wm8958_enh_eq.wfw", codec->dev, GFP_KERNEL, + codec, wm8958_enh_eq_loaded); if (pdata->num_mbc_cfgs) { struct snd_kcontrol_new control[] = { -- 1.7.10.4

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