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udev attributes to detect compress-offload devices
by Carlos Rafael Giani 26 Jun '23

26 Jun '23

Are there attributes in udev that allow for unambiguously detect a compress-offload device? Currently, I have the problem that I can't autodetect compress-offload devices reliably, because I see nothing in udev that identifies a device as such. The next best thing thus far has to be to rely on the /dev/snd/comprC*D* name. But, these device nodes can be renamed, can't they?

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[PATCH 0/8] Few audio fixes on Tegra platforms
by Sameer Pujar 26 Jun '23

26 Jun '23

This series fixes some of the issues which were observed during an attempt to enhance automated test coverage on Jetson AGX Orin. Below is a short summary of the issues and fixes: * Sample rate coversion failures above 48kHz. * AMX and ADX test cases failures due to incorrect byte mask. * Atomic sleep in RT5640 codec which is present on Jetson AGX Orin. * AHUB clock fixes on Tegra234 and previous chips. * Minor cleanups in ASRC and AHUB driver. Sameer Pujar (4): ASoC: rt5640: Fix sleep in atomic context ASoC: tegra: Use normal system sleep for ASRC ASoC: tegra: Remove stale comments in AHUB arm64: tegra: Update AHUB clock parent and rate Sheetal (4): ASoC: tegra: Fix SFC conversion for few rates ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map arm64: tegra: Update AHUB clock parent and rate on Tegra234 arch/arm64/boot/dts/nvidia/tegra186.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra210.dtsi | 3 ++- arch/arm64/boot/dts/nvidia/tegra234.dtsi | 3 ++- sound/soc/codecs/rt5640.c | 20 ++++++++++++-------- sound/soc/tegra/tegra186_asrc.c | 4 ++-- sound/soc/tegra/tegra210_adx.c | 24 ++++++++++++------------ sound/soc/tegra/tegra210_ahub.c | 10 ---------- sound/soc/tegra/tegra210_amx.c | 30 ++++++++++++------------------ sound/soc/tegra/tegra210_sfc.c | 31 ++++++++++++++++++++++++++++++- sound/soc/tegra/tegra210_sfc.h | 4 ++-- 11 files changed, 78 insertions(+), 57 deletions(-) -- 2.7.4

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[PATCH] ASoC: loongson: fix address space confusion
by Arnd Bergmann 26 Jun '23

26 Jun '23

From: Arnd Bergmann <arnd(a)arndb.de> The i2s driver uses the mapped __iomem address of the FIFO as the DMA address for the device. This apparently works on loongarch because of the way it handles __iomem pointers as aliases of physical addresses, but this is not portable to other architectures and causes a compiler warning when dma addresses are not the same size as pointers: sound/soc/loongson/loongson_i2s_pci.c: In function 'loongson_i2s_pci_probe': sound/soc/loongson/loongson_i2s_pci.c:110:29: error: cast from pointer to integer of different size [-Werror=pointer-to-int-cast] 110 | tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; | ^ sound/soc/loongson/loongson_i2s_pci.c:113:29: error: cast from pointer to integer of different size [-Werror=pointer-to-int-cast] 113 | rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; | ^ Change the driver to instead use the physical address as stored in the PCI BAR resource directly. Since 'dev_addr' is a 32-bit address, I think this results in the same truncated address on loongarch but is otherwise closer to portable code and avoids the warning. Fixes: d84881e06836d ("ASoC: Add support for Loongson I2S controller") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- This is only build tested and I don't understand the loongarch specifics that well, please review or test for actual hardware requirements. --- sound/soc/loongson/loongson_i2s_pci.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/loongson/loongson_i2s_pci.c b/sound/soc/loongson/loongson_i2s_pci.c index 6dcfb17d3276d..fa90361865c6c 100644 --- a/sound/soc/loongson/loongson_i2s_pci.c +++ b/sound/soc/loongson/loongson_i2s_pci.c @@ -107,10 +107,10 @@ static int loongson_i2s_pci_probe(struct pci_dev *pdev, tx_data = &i2s->tx_dma_data; rx_data = &i2s->rx_dma_data; - tx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_TX_DATA; + tx_data->dev_addr = pci_resource_start(pdev, 0) + LS_I2S_TX_DATA; tx_data->order_addr = i2s->reg_base + LS_I2S_TX_ORDER; - rx_data->dev_addr = (dma_addr_t)i2s->reg_base + LS_I2S_RX_DATA; + rx_data->dev_addr = pci_resource_start(pdev, 0) + LS_I2S_RX_DATA; rx_data->order_addr = i2s->reg_base + LS_I2S_RX_ORDER; tx_data->irq = fwnode_irq_get_byname(fwnode, "tx"); -- 2.39.2

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[PATCH v2] ASoC: amd: update pm_runtime enable sequence
by Arun Gopal Kondaveeti 26 Jun '23

26 Jun '23

pm_runtime_allow() is not needed for ACP child platform devices. Replace pm_runtime_allow() with pm_runtime_mark_last_busy() & pm_runtime_set_active() in pm_runtime enable sequence for ACP child platform drivers. Signed-off-by: Arun Gopal <arungopal.kondaveeti(a)amd.com> --- Changes since v1: -- Updated commit message --- sound/soc/amd/ps/ps-pdm-dma.c | 3 ++- sound/soc/amd/raven/acp3x-pcm-dma.c | 3 ++- sound/soc/amd/renoir/acp3x-pdm-dma.c | 3 ++- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 4 ++-- sound/soc/amd/yc/acp6x-pdm-dma.c | 3 ++- 5 files changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index bdbbb797c74d..d48f7c5af289 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -391,8 +391,9 @@ static int acp63_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 7362dd15ad30..9538f3ffc5d9 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -416,8 +416,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 4e299f96521f..c3b47e9bd239 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -430,8 +430,9 @@ static int acp_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 29901ee4bfe3..587dec5bb33d 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -409,9 +409,9 @@ static int acp5x_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, 2000); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); - return 0; } diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index d818eba48546..72c4591e451b 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -383,8 +383,9 @@ static int acp6x_pdm_audio_probe(struct platform_device *pdev) } pm_runtime_set_autosuspend_delay(&pdev->dev, ACP_SUSPEND_DELAY_MS); pm_runtime_use_autosuspend(&pdev->dev); + pm_runtime_mark_last_busy(&pdev->dev); + pm_runtime_set_active(&pdev->dev); pm_runtime_enable(&pdev->dev); - pm_runtime_allow(&pdev->dev); return 0; } -- 2.25.1

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ALC287 problem
by ollierazdow＠protonmail.com 26 Jun '23

26 Jun '23

I have a problem with the ALC287 codec that might indicate a missing fix in the Kernel. There is digital noise in the output unless I run the command: sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0 I looked through the file /sound/pci/hda/patch_realtek.c and found ALC285_FIXUP_LENOVO_PC_BEEP_IN_NOISE but it does not affect my hardware. The latest Kernel I tried was 6.1.32. I'm not familiar enough with how the file is organized to write a patch myself, but If anyone has advice about how to address it I could try. I wanted to bring alert to this though because it seems like a missing fix. I also posted here with a bit more detail: https://bugzilla.kernel.org/show_bug.cgi?id=217580

1 0

USB-Audio: Add Steinberg UR44C
by GitHub pull_request - opened 25 Jun '23

25 Jun '23

alsa-project/alsa-ucm-conf pull request #330 was opened from cyberchrime: Adds support for the Steinberg UR44C. In fact, it is similar to the Steinberg UR44, but uses USB-C with USB 3.1 and has a different USB devnum (1730 instead of 1700). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/330 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/330.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MIC capture stops due to playback
by Arjav Parikh 25 Jun '23

25 Jun '23

Hi I am using alsa api's in my C Program to implement two-way talk i.e capturing data from mic and playing sound on speaker received from Cloud. I am using a Microsemi ZL38050 Audio DSP connected to an Ambarella CV25 SOC via I2S/SPI. Randomly I have observed that If I stop and resume receiving data from Cloud to play it on speaker using snd_pcm_writei() then my capture path stops. I do not hear any sound captured from MIC. Can someone help me to debug this issue because when this issue occurs there are no errors reported from Alsa nor do I see anything in dmesg as well. Thanks & Regards, Arjav Parikh

1 0

[PATCH] ALSA: aloop: Fix &cable->lock deadlock issues
by YE Chengfeng 25 Jun '23

25 Jun '23

The timer loopback_jiffies_timer_function is executed under bottom-half softirq context and require a spinlock, thus other process context code requiring the same lock (i.e., loopback_trigger, loopback_pointer) can deadlock with the timer if it is preempted while holding the lock. Deadlock scenario: loopback_trigger -> spin_lock(&cable->lock); <timer interrupt> -> loopback_jiffies_timer_function -> spin_lock_irqsave(&dpcm->cable->lock, flags); Fix the potential deadlock by using spin_lock_irqsave. Signed-off-by: Chengfeng Ye <cyeaa(a)connect.ust.hk> --- sound/drivers/aloop.c | 22 ++++++++++++---------- 1 file changed, 12 insertions(+), 10 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a38e602b4fc6..8ee93f8581b4 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -379,6 +379,7 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; struct loopback_cable *cable = dpcm->cable; + unsigned long flags; int err = 0, stream = 1 << substream->stream; switch (cmd) { @@ -389,39 +390,39 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) dpcm->last_jiffies = jiffies; dpcm->pcm_rate_shift = 0; dpcm->last_drift = 0; - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); cable->running |= stream; cable->pause &= ~stream; err = cable->ops->start(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_STOP: - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); cable->running &= ~stream; cable->pause &= ~stream; err = cable->ops->stop(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); cable->pause |= stream; err = cable->ops->stop(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: - spin_lock(&cable->lock); + spin_lock_irqsave(&cable->lock, flags); dpcm->last_jiffies = jiffies; cable->pause &= ~stream; err = cable->ops->start(dpcm); - spin_unlock(&cable->lock); + spin_unlock_irqrestore(&cable->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) loopback_active_notify(dpcm); break; @@ -865,12 +866,13 @@ static snd_pcm_uframes_t loopback_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct loopback_pcm *dpcm = runtime->private_data; snd_pcm_uframes_t pos; + unsigned long flags; - spin_lock(&dpcm->cable->lock); + spin_lock_irqsave(&dpcm->cable->lock, flags); if (dpcm->cable->ops->pos_update) dpcm->cable->ops->pos_update(dpcm->cable); pos = dpcm->buf_pos; - spin_unlock(&dpcm->cable->lock); + spin_unlock_irqrestore(&dpcm->cable->lock, flags); return bytes_to_frames(runtime, pos); } -- 2.17.1

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[PATCH] ALSA: pcm: fix ELD constraints for (E)AC3, DTS(-HD) and MLP formats
by Matthias Reichl 25 Jun '23

25 Jun '23

The SADs of compressed formats contain the channel and sample rate info of the audio data inside the compressed stream, but when building constraints we must use the rates and channels used to transport the compressed streams. eg 48kHz 6ch EAC3 needs to be transmitted as a 2ch 192kHz stream. This patch fixes the constraints for the common AC3 and DTS formats, the constraints for the less common MPEG, DSD etc formats are copied directly from the info in the SADs as before as I don't have the specs and equipment to test those. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/core/pcm_drm_eld.c | 73 ++++++++++++++++++++++++++++++++++++++-- 1 file changed, 70 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_drm_eld.c b/sound/core/pcm_drm_eld.c index 4b5faae5d16e5..07075071972dd 100644 --- a/sound/core/pcm_drm_eld.c +++ b/sound/core/pcm_drm_eld.c @@ -2,11 +2,25 @@ /* * PCM DRM helpers */ +#include <linux/bitfield.h> #include <linux/export.h> +#include <linux/hdmi.h> #include <drm/drm_edid.h> #include <sound/pcm.h> #include <sound/pcm_drm_eld.h> +#define SAD0_CHANNELS_MASK GENMASK(2, 0) /* max number of channels - 1 */ +#define SAD0_FORMAT_MASK GENMASK(6, 3) /* audio format */ + +#define SAD1_RATE_MASK GENMASK(6, 0) /* bitfield of supported rates */ +#define SAD1_RATE_32000_MASK BIT(0) +#define SAD1_RATE_44100_MASK BIT(1) +#define SAD1_RATE_48000_MASK BIT(2) +#define SAD1_RATE_88200_MASK BIT(3) +#define SAD1_RATE_96000_MASK BIT(4) +#define SAD1_RATE_176400_MASK BIT(5) +#define SAD1_RATE_192000_MASK BIT(6) + static const unsigned int eld_rates[] = { 32000, 44100, @@ -17,9 +31,62 @@ static const unsigned int eld_rates[] = { 192000, }; +static unsigned int map_rate_families(const u8 *sad, + unsigned int mask_32000, + unsigned int mask_44100, + unsigned int mask_48000) +{ + unsigned int rate_mask = 0; + + if (sad[1] & SAD1_RATE_32000_MASK) + rate_mask |= mask_32000; + if (sad[1] & (SAD1_RATE_44100_MASK | SAD1_RATE_88200_MASK | SAD1_RATE_176400_MASK)) + rate_mask |= mask_44100; + if (sad[1] & (SAD1_RATE_48000_MASK | SAD1_RATE_96000_MASK | SAD1_RATE_192000_MASK)) + rate_mask |= mask_48000; + return rate_mask; +} + +static unsigned int sad_rate_mask(const u8 *sad) +{ + switch (FIELD_GET(SAD0_FORMAT_MASK, sad[0])) { + case HDMI_AUDIO_CODING_TYPE_PCM: + return sad[1] & SAD1_RATE_MASK; + case HDMI_AUDIO_CODING_TYPE_AC3: + case HDMI_AUDIO_CODING_TYPE_DTS: + return map_rate_families(sad, + SAD1_RATE_32000_MASK, + SAD1_RATE_44100_MASK, + SAD1_RATE_48000_MASK); + case HDMI_AUDIO_CODING_TYPE_EAC3: + case HDMI_AUDIO_CODING_TYPE_DTS_HD: + case HDMI_AUDIO_CODING_TYPE_MLP: + return map_rate_families(sad, + 0, + SAD1_RATE_176400_MASK, + SAD1_RATE_192000_MASK); + default: + /* TODO adjust for other compressed formats as well */ + return sad[1] & SAD1_RATE_MASK; + } +} + static unsigned int sad_max_channels(const u8 *sad) { - return 1 + (sad[0] & 7); + switch (FIELD_GET(SAD0_FORMAT_MASK, sad[0])) { + case HDMI_AUDIO_CODING_TYPE_PCM: + return 1 + FIELD_GET(SAD0_CHANNELS_MASK, sad[0]); + case HDMI_AUDIO_CODING_TYPE_AC3: + case HDMI_AUDIO_CODING_TYPE_DTS: + case HDMI_AUDIO_CODING_TYPE_EAC3: + return 2; + case HDMI_AUDIO_CODING_TYPE_DTS_HD: + case HDMI_AUDIO_CODING_TYPE_MLP: + return 8; + default: + /* TODO adjust for other compressed formats as well */ + return 1 + FIELD_GET(SAD0_CHANNELS_MASK, sad[0]); + } } static int eld_limit_rates(struct snd_pcm_hw_params *params, @@ -42,7 +109,7 @@ static int eld_limit_rates(struct snd_pcm_hw_params *params, * requested number of channels. */ if (c->min <= max_channels) - rate_mask |= sad[1]; + rate_mask |= sad_rate_mask(sad); } } @@ -70,7 +137,7 @@ static int eld_limit_channels(struct snd_pcm_hw_params *params, rate_mask |= BIT(i); for (i = drm_eld_sad_count(eld); i > 0; i--, sad += 3) - if (rate_mask & sad[1]) + if (rate_mask & sad_rate_mask(sad)) t.max = max(t.max, sad_max_channels(sad)); } -- 2.39.2

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[PATCH v2 RESEND] ASoC: fsl MPC52xx drivers require PPC_BESTCOMM
by Randy Dunlap 25 Jun '23

25 Jun '23

Both SND_MPC52xx_SOC_PCM030 and SND_MPC52xx_SOC_EFIKA select SND_SOC_MPC5200_AC97. The latter symbol depends on PPC_BESTCOMM, so the 2 former symbols should also depend on PPC_BESTCOMM since "select" does not follow any dependency chains. This prevents a kconfig warning and build errors: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [m]: - SND_MPC52xx_SOC_PCM030 [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=m] && SOUND [=y] && !UML && SND [=m] && SND_SOC [=m] && SND_POWERPC_SOC [=m] && PPC_EFIKA [=y] ERROR: modpost: "mpc5200_audio_dma_destroy" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! ERROR: modpost: "mpc5200_audio_dma_create" [sound/soc/fsl/mpc5200_psc_ac97.ko] undefined! Fixes: 40d9ec14e7e1 ("ASoC: remove BROKEN from Efika and pcm030 fabric drivers") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> --- v2: use correct email address for Mark Brown. sound/soc/fsl/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff -- a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -243,7 +243,7 @@ config SND_SOC_MPC5200_AC97 config SND_MPC52xx_SOC_PCM030 tristate "SoC AC97 Audio support for Phytec pcm030 and WM9712" - depends on PPC_MPC5200_SIMPLE + depends on PPC_MPC5200_SIMPLE && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_WM9712 help @@ -252,7 +252,7 @@ config SND_MPC52xx_SOC_PCM030 config SND_MPC52xx_SOC_EFIKA tristate "SoC AC97 Audio support for bbplan Efika and STAC9766" - depends on PPC_EFIKA + depends on PPC_EFIKA && PPC_BESTCOMM select SND_SOC_MPC5200_AC97 select SND_SOC_STAC9766 help

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