March 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH] ASoC: core: Fix use-after-free after deferred card registration
by Guenter Roeck 26 Mar '19

26 Mar '19

If snd_soc_register_card() fails because one of its links fails to instantiate with -EPROBE_DEFER, and the to-be-registered link is a legacy link, a subsequent retry will trigger a use-after-free and quite often a system crash. Example: byt-max98090 byt-max98090: ASoC: failed to init link Baytrail Audio byt-max98090 byt-max98090: snd_soc_register_card failed -517 .... BUG: KASAN: use-after-free in snd_soc_init_platform+0x233/0x312 Read of size 8 at addr ffff888067c43070 by task kworker/1:1/23 snd_soc_init_platform() allocates memory attached to the card device. This memory is released when the card device is released. However, the pointer to the memory (dai_link->platforms) is only cleared from soc_cleanup_platform(), which is called from soc_cleanup_card_resources(), but not if snd_soc_register_card() fails early. Add the missing call to soc_cleanup_platform() in the error handling code of snd_soc_register_card() to fix the problem. Fixes: 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error") Cc: Curtis Malainey <cujomalainey(a)chromium.org> Signed-off-by: Guenter Roeck <linux(a)roeck-us.net> --- sound/soc/soc-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 93d316d5bf8e..6bf9884d0863 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2799,6 +2799,7 @@ int snd_soc_register_card(struct snd_soc_card *card) if (ret) { dev_err(card->dev, "ASoC: failed to init link %s\n", link->name); + soc_cleanup_platform(card); mutex_unlock(&client_mutex); return ret; } -- 2.7.4

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[alsa-devel] [RESEND] [-, JACK, plugin, 1/1] configure: use /etc instead of /etc
by Alex Ivanov 26 Mar '19

26 Mar '19

Could you please test this change again? I need it for my maintainer work. Thank you! Use $sysconfdir as prefix for ALSA_LCONF_DIR by default. Otherwise install fails on non-FHS distros. Signed-off-by: Alex Ivanov <gnidorah(a)ya.ru> --- configure.ac | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index 5b80585..1be4a70 100644 --- a/configure.ac +++ b/configure.ac @@ -237,7 +237,12 @@ AC_ARG_WITH(alsalconfdir, [path where ALSA local add-on config files are stored]), alsalconfdir="$withval", alsalconfdir="") if test -z "$alsalconfdir"; then - alsalconfdir="/etc/alsa/conf.d" + eval dir="$sysconfdir" + case "$dir" in + /*) ;; + *) dir="$dir" + esac + alsalconfdir="$dir/alsa/conf.d" fi AC_DEFINE_UNQUOTED(ALSA_LCONF_DIR, "$alsalconfdir", [directory containing local ALSA add-on config files]) ALSA_LCONF_DIR="$alsalconfdir" -- 2.19.2

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[alsa-devel] [asoc:for-5.2 84/86] include/linux/dynamic_debug.h:80:13: error: initializer element is not constant
by kbuild test robot 26 Mar '19

26 Mar '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: b86ef5367761603df5f66ce08fb014b991f6b51d commit: be1df61cf06efb355c90702e46b8d46f055acb4e [84/86] ASoC: fsl: Add Audio Mixer CPU DAI driver config: sparc64-allmodconfig (attached as .config) compiler: sparc64-linux-gnu-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout be1df61cf06efb355c90702e46b8d46f055acb4e # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=sparc64 All errors (new ones prefixed by >>): In file included from include/linux/printk.h:330:0, from include/linux/kernel.h:15, from include/linux/clk.h:16, from sound/soc//fsl/fsl_audmix.c:8: sound/soc//fsl/fsl_audmix.c: In function 'fsl_audmix_state_trans': >> include/linux/dynamic_debug.h:80:13: error: initializer element is not constant .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc//fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ include/linux/dynamic_debug.h:80:13: note: (near initialization for '__UNIQUE_ID_ddebug14.format') .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc//fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ -- In file included from include/linux/printk.h:330:0, from include/linux/kernel.h:15, from include/linux/clk.h:16, from sound/soc/fsl/fsl_audmix.c:8: sound/soc/fsl/fsl_audmix.c: In function 'fsl_audmix_state_trans': >> include/linux/dynamic_debug.h:80:13: error: initializer element is not constant .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc/fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ include/linux/dynamic_debug.h:80:13: note: (near initialization for '__UNIQUE_ID_ddebug14.format') .format = (fmt), \ ^ include/linux/dynamic_debug.h:116:2: note: in expansion of macro 'DEFINE_DYNAMIC_DEBUG_METADATA' DEFINE_DYNAMIC_DEBUG_METADATA(id, fmt); \ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:136:2: note: in expansion of macro '__dynamic_func_call' __dynamic_func_call(__UNIQUE_ID(ddebug), fmt, func, ##__VA_ARGS__) ^~~~~~~~~~~~~~~~~~~ include/linux/dynamic_debug.h:150:2: note: in expansion of macro '_dynamic_func_call' _dynamic_func_call(fmt,__dynamic_dev_dbg, \ ^~~~~~~~~~~~~~~~~~ include/linux/device.h:1495:2: note: in expansion of macro 'dynamic_dev_dbg' dynamic_dev_dbg(dev, dev_fmt(fmt), ##__VA_ARGS__) ^~~~~~~~~~~~~~~ sound/soc/fsl/fsl_audmix.c:93:3: note: in expansion of macro 'dev_dbg' dev_dbg(comp->dev, prm.msg); ^~~~~~~ vim +80 include/linux/dynamic_debug.h ffa10cb4 Jason Baron 2011-08-11 68 b9075fa9 Joe Perches 2011-10-31 69 extern __printf(3, 4) 906d2015 Joe Perches 2014-09-24 70 void __dynamic_netdev_dbg(struct _ddebug *descriptor, ffa10cb4 Jason Baron 2011-08-11 71 const struct net_device *dev, b9075fa9 Joe Perches 2011-10-31 72 const char *fmt, ...); ffa10cb4 Jason Baron 2011-08-11 73 2bdde670 Rasmus Villemoes 2019-03-07 74 #define DEFINE_DYNAMIC_DEBUG_METADATA(name, fmt) \ c0d2af63 Joe Perches 2012-10-18 75 static struct _ddebug __aligned(8) \ 07613b0b Jason Baron 2011-10-04 76 __attribute__((section("__verbose"))) name = { \ 07613b0b Jason Baron 2011-10-04 77 .modname = KBUILD_MODNAME, \ 07613b0b Jason Baron 2011-10-04 78 .function = __func__, \ 07613b0b Jason Baron 2011-10-04 79 .filename = __FILE__, \ 07613b0b Jason Baron 2011-10-04 @80 .format = (fmt), \ 07613b0b Jason Baron 2011-10-04 81 .lineno = __LINE__, \ 07613b0b Jason Baron 2011-10-04 82 .flags = _DPRINTK_FLAGS_DEFAULT, \ 2bdde670 Rasmus Villemoes 2019-03-07 83 _DPRINTK_KEY_INIT \ 07613b0b Jason Baron 2011-10-04 84 } 07613b0b Jason Baron 2011-10-04 85 :::::: The code at line 80 was first introduced by commit :::::: 07613b0b5ef8570033aa806d1731dce599862223 dynamic_debug: consolidate repetitive struct _ddebug descriptor definitions :::::: TO: Jason Baron <jbaron(a)redhat.com> :::::: CC: Greg Kroah-Hartman <gregkh(a)suse.de> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] snd_pcm_writei 0 frames
by Giuliano Zannetti - ART S.p.A. 26 Mar '19

26 Mar '19

Hi, I've the following configuration in my asound.conf: |-----| - - > |-----| - - > |-----| - - > |-----| - - > |-----| - - > |----------| |plug | | | - - > | | - - > | | - - > | | - - > | (output) | |-----| - - > | | - - > | | - - > | | - - > | | - - > | hw | | | - - > | | - - > | | - - > | | - - > | | | | | | |route| - - > |dmix | - - > | | | | | | | | - - > | | - - > | | | | | | | | - - > | | - - > | | | | | | |-----| - - > |-----| - - > |----------| |route| |multi| | | - - > | | | | - - > | | - - > |----| - - > |----| - - > |----------| | | - - > | | - - > | | - - > | | - - > |(loopback)| |-----| - - > | | - - > |plug| - - > |dmix| - - > | hw | |-----| - - > |----| - - > |----| - - > |----------| As you can see I'm using a loopback interface. I open the pcm plug in the block mode to write. writtenFrames = snd_pcm_writei(handle, bufferToWrite, bufferToWriteSize); Using the configuration above I have several "writes" with 0 written frames. It seems like the snd_pcm_writei does not hold until data will be written. In general, 0 written frames sounds weird to me, because snd_pcm_writei should block itself if no space is available in the circular buffer right? So, my questions are two: 1. Writing 0 frames is an expected behaviour (regardless of my configuration in the asound.conf)? In what scenarios can I have that? 2. How my configuration could trigger this behaviour? Best Regards Giuliano Zannetti

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[alsa-devel] [PATCH] ALSA: emux: Add support of loading GUS-patch
by Takashi Iwai 26 Mar '19

26 Mar '19

It's a feature request for the ancient sutff, but it's still valid; the loading of a GUS-patch isn't available via hwdep device although it's supported over OSS sequencer. The only missing piece is the call of snd_soundfont_load_guspatch() in synth emux hwdep code. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/synth/emux/emux_hwdep.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index d9fcae071b47..fae48d108b97 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -39,6 +39,11 @@ snd_emux_hwdep_load_patch(struct snd_emux *emu, void __user *arg) if (copy_from_user(&patch, arg, sizeof(patch))) return -EFAULT; + if (patch.key == GUS_PATCH) + return snd_soundfont_load_guspatch(emu->sflist, arg, + patch.len + sizeof(patch), + TMP_CLIENT_ID); + if (patch.type >= SNDRV_SFNT_LOAD_INFO && patch.type <= SNDRV_SFNT_PROBE_DATA) { err = snd_soundfont_load(emu->sflist, arg, patch.len + sizeof(patch), TMP_CLIENT_ID); -- 2.16.4

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[alsa-devel] Common Mode Voltage Setting
by Annaliese McDermond 26 Mar '19

26 Mar '19

I have an architectural question regarding some work I’m doing. We have a product that hooks the Raspberry Pi to various amateur radio rigs via a sound card interface. The input characteristics of those radios vary widely in the amount of signal they need to drive the proper RF output and such. One of the setting s I’ve implemented to try to help our customers control this is changing the Common Mode voltage on the tlv320. There are a couple of options, but the gist is that one will allow higher output levels without clipping. We need to keep the flexibility of having the lower setting because there are some radios that require as little drive as possible. My question is really surrounding where the appropriate setting for this would be. I could put it in the device tree, but that requires adding entries, pushing this past the DT folks and it becomes something that a user has to reboot to change settings on. I think it probably more appropriately goes as a mixer control so that it can be changed on the fly without reboot and is a dynamic parameter. I just want to make sure there’s no strong preference here on the “right way” it should be done or if I’m missing a way to set this that’s a “none of the above.” Thanks! -- Annaliese McDermond nh6z(a)nh6z.net

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[alsa-devel] NHLT Specification
by Iker Perez del Palomar 26 Mar '19

26 Mar '19

Hello, My name is Iker Perez, I am contacting you because at the moment I am working with NHLT at my company, Codethink. This is the first time we have been involved with it, so we have been trying to find a specification or a document that could help us understand it better. We tried the Intel website [1] but it hasn't got the document it states is linked to it. I was wondering if someone could point us toward any documentation you are aware of? Thank you, Regards, Iker -- Iker Perez del Palomar, Software Engineer Codethink Ltd 35 Dale St, Manchester M1 2HF http://www.codethink.co.uk/ Manchester, M1 2JW, United Kingdom We respect your privacy. See https://www.codethink.co.uk/privacy.html

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[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Only enable with common clock
by Mark Brown 26 Mar '19

26 Mar '19

Some architectures do not yet support the common clock API at all but the tlv320aic32x4 driver now requires it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e99320c79b8..419142111b6d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -180,8 +180,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC31XX if I2C - select SND_SOC_TLV320AIC32X4_I2C if I2C - select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER + select SND_SOC_TLV320AIC32X4_I2C if I2C && COMMON_CLK + select SND_SOC_TLV320AIC32X4_SPI if SPI_MASTER && COMMON_CLK select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C select SND_SOC_TLV320DAC33 if I2C @@ -1110,11 +1110,13 @@ config SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_I2C tristate "Texas Instruments TLV320AIC32x4 audio CODECs - I2C" depends on I2C + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC32X4_SPI tristate "Texas Instruments TLV320AIC32x4 audio CODECs - SPI" depends on SPI_MASTER + depends on COMMON_CLK select SND_SOC_TLV320AIC32X4 config SND_SOC_TLV320AIC3X -- 2.20.1

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[alsa-devel] Applied "ASoC: Intel: Skylake: enable S24_LE format support" to the asoc tree
by Mark Brown 26 Mar '19

26 Mar '19

The patch ASoC: Intel: Skylake: enable S24_LE format support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 36e075ce74ec4e261a638bf09d10b3348ca4d883 Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Sat, 23 Mar 2019 18:40:10 +0530 Subject: [PATCH] ASoC: Intel: Skylake: enable S24_LE format support To enable S24_LE format, sample_type in topology fw has to be set to 1. But sample_type defined in topology firmware configuration is not getting reflected in the dsp param. This patch sets sample_type in base config so that the sample type defined in the topology firmware is reflected in the dsp params. This issues was uncovered while debugging the S24_LE format which require the MSB byte in 32 bit word to be skipped. Setting sample_type in topology firmware to 1 helps to skip MSB byte word. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 28c4806b196a..4bf70b4429f0 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -483,6 +483,7 @@ static void skl_set_base_module_format(struct skl_sst *ctx, base_cfg->audio_fmt.bit_depth = format->bit_depth; base_cfg->audio_fmt.valid_bit_depth = format->valid_bit_depth; base_cfg->audio_fmt.ch_cfg = format->ch_cfg; + base_cfg->audio_fmt.sample_type = format->sample_type; dev_dbg(ctx->dev, "bit_depth=%x valid_bd=%x ch_config=%x\n", format->bit_depth, format->valid_bit_depth, -- 2.20.1

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