April 2018 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH 3/8] [media] v4l: rcar_fdp1: Change platform dependency to ARCH_RENESAS
by Laurent Pinchart 23 Apr '18

23 Apr '18

Hi Geert, On Saturday, 21 April 2018 11:07:11 EEST Laurent Pinchart wrote: > On Friday, 20 April 2018 16:28:29 EEST Geert Uytterhoeven wrote: > > The Renesas Fine Display Processor driver is used on Renesas R-Car SoCs > > only. Since commit 9b5ba0df4ea4f940 ("ARM: shmobile: Introduce > > ARCH_RENESAS") is ARCH_RENESAS a more appropriate platform dependency > > than the legacy ARCH_SHMOBILE, hence use the former. > > > > This will allow to drop ARCH_SHMOBILE on ARM and ARM64 in the near > > future. > > > > Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> > > Acked-by: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> > > How would you like to get this merged ? Unless you would like to merge the whole series in one go, I'll take this in my tree as I have a conflicting patch I would like to submit for v4.18. > > --- > > > > drivers/media/platform/Kconfig | 2 +- > > 1 file changed, 1 insertion(+), 1 deletion(-) > > > > diff --git a/drivers/media/platform/Kconfig > > b/drivers/media/platform/Kconfig index f9235e8f8e962d2e..7ad4725f9d1f9627 > > 100644 > > --- a/drivers/media/platform/Kconfig > > +++ b/drivers/media/platform/Kconfig > > @@ -396,7 +396,7 @@ config VIDEO_SH_VEU > > config VIDEO_RENESAS_FDP1 > > tristate "Renesas Fine Display Processor" > > depends on VIDEO_DEV && VIDEO_V4L2 && HAS_DMA > > - depends on ARCH_SHMOBILE || COMPILE_TEST > > + depends on ARCH_RENESAS || COMPILE_TEST > > depends on (!ARCH_RENESAS && !VIDEO_RENESAS_FCP) || VIDEO_RENESAS_FCP > > select VIDEOBUF2_DMA_CONTIG > > select V4L2_MEM2MEM_DEV -- Regards, Laurent Pinchart

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[alsa-devel] [PATCH 0/4] More COMPILE_TEST patches to build all media drivers on x86_64
by Mauro Carvalho Chehab 23 Apr '18

23 Apr '18

While now all media drivers build with COMPILE_TEST on i386, there are still a few of them that don't build on x86_64. Making them compiling there is just a matter of touching Kconfig files. While here, fix smatch warnings when building the siano driver on big endian architectures. Mauro Carvalho Chehab (4): media: radio: allow building ISA drivers with COMPILE_TEST media: sta2x11_vip: allow build with COMPILE_TEST sound, media: allow building ISA drivers it with COMPILE_TEST media: siano: get rid of __le32/__le16 cast warnings drivers/media/common/siano/smsendian.c | 14 ++++++------ drivers/media/pci/sta2x11/Kconfig | 2 +- drivers/media/radio/Kconfig | 41 +++++++++++++++++++++------------- sound/isa/Kconfig | 9 ++++---- 4 files changed, 39 insertions(+), 27 deletions(-) -- 2.14.3

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[alsa-devel] [PATCH] ALSA: pcm: Return negative delays from SNDRV_PCM_IOCTL_DELAY.
by Jeffery Miller 23 Apr '18

23 Apr '18

The commit c2c86a97175f ("ALSA: pcm: Remove set_fs() in PCM core code") changed SNDRV_PCM_IOCTL_DELAY to return an inconsistent error instead of a negative delay. Originally the call would succeed and return the negative delay. The Chromium OS Audio Server (CRAS) gets confused and hangs when the error is returned instead of the negative delay. Help CRAS avoid the issue by rolling back the behavior to return a negative delay instead of an error. Fixes: c2c86a97175f ("ALSA: pcm: Remove set_fs() in PCM core code") Signed-off-by: Jeffery Miller <jmiller(a)neverware.com> --- This patch changes the SNDRV_PCM_IOCTL_DELAY ioctl return behavior to be similar to what it was in an earlier kernel. I encountered this when upgrading from the 4.4 kernel to 4.14. Chrome Audio Server (cras) does not handle the error return as gracefully as it was handling the successful but negative delays. The intent of this patch is to simply revert the behavior of the call to match what it was previously. It has been applied and tested with the 4.14 kernel. I did confirm this is still an issue and is resolved with the patch on the tiwai/sound.git master branch. sound/core/pcm_compat.c | 7 ++++--- sound/core/pcm_native.c | 23 +++++++++++------------ 2 files changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 0be248543f1e..077008a8e1d5 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -27,10 +27,11 @@ static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream, s32 __user *src) { snd_pcm_sframes_t delay; + int err; - delay = snd_pcm_delay(substream); - if (delay < 0) - return delay; + err = snd_pcm_delay(substream, &delay); + if (err) + return err; if (put_user(delay, src)) return -EFAULT; return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7585444352df..d2ba181c666e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2654,7 +2654,8 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) return err; } -static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) +static int snd_pcm_delay(struct snd_pcm_substream *substream, + snd_pcm_sframes_t *delay) { int err; snd_pcm_sframes_t n = 0; @@ -2664,7 +2665,9 @@ static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) if (!err) n = snd_pcm_calc_delay(substream); snd_pcm_stream_unlock_irq(substream); - return err < 0 ? err : n; + if (!err) + *delay = n; + return err; } static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, @@ -2866,11 +2869,13 @@ static int snd_pcm_common_ioctl(struct file *file, return snd_pcm_hwsync(substream); case SNDRV_PCM_IOCTL_DELAY: { - snd_pcm_sframes_t delay = snd_pcm_delay(substream); + snd_pcm_sframes_t delay; snd_pcm_sframes_t __user *res = arg; + int err; - if (delay < 0) - return delay; + err = snd_pcm_delay(substream, &delay); + if (err) + return err; if (put_user(delay, res)) return -EFAULT; return 0; @@ -2958,13 +2963,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, case SNDRV_PCM_IOCTL_DROP: return snd_pcm_drop(substream); case SNDRV_PCM_IOCTL_DELAY: - { - result = snd_pcm_delay(substream); - if (result < 0) - return result; - *frames = result; - return 0; - } + return snd_pcm_delay(substream, frames); default: return -EINVAL; } -- 2.14.3

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[alsa-devel] [PATCH] ALSA: core: Report audio_tstamp in snd_pcm_sync_ptr
by Takashi Iwai 22 Apr '18

22 Apr '18

From: David Henningsson <diwic(a)ubuntu.com> It looks like a simple mistake that this struct member was forgotten. Audio_tstamp isn't used much, and on some archs (such as x86) this ioctl is not used by default, so that might be the reason why this has slipped for so long. Fixes: 4eeaaeaea1ce ("ALSA: core: add hooks for audio timestamps") Signed-off-by: David Henningsson <diwic(a)ubuntu.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 35ffcce..632228b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2751,6 +2751,7 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, sync_ptr.s.status.hw_ptr = status->hw_ptr; sync_ptr.s.status.tstamp = status->tstamp; sync_ptr.s.status.suspended_state = status->suspended_state; + sync_ptr.s.status.audio_tstamp = status->audio_tstamp; snd_pcm_stream_unlock_irq(substream); if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr))) return -EFAULT; -- 2.7.4

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Re: [alsa-devel] [PATCH 04/15] media: pxa_camera: remove the dmaengine compat need
by Hans Verkuil 22 Apr '18

22 Apr '18

On 04/02/2018 04:26 PM, Robert Jarzmik wrote: > From: Robert Jarzmik <robert.jarzmik(a)renault.com> > > As the pxa architecture switched towards the dmaengine slave map, the > old compatibility mechanism to acquire the dma requestor line number and > priority are not needed anymore. > > This patch simplifies the dma resource acquisition, using the more > generic function dma_request_slave_channel(). > > Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> Acked-by: Hans Verkuil <hans.verkuil(a)cisco.com> Regards, Hans > --- > drivers/media/platform/pxa_camera.c | 22 +++------------------- > 1 file changed, 3 insertions(+), 19 deletions(-) > > diff --git a/drivers/media/platform/pxa_camera.c b/drivers/media/platform/pxa_camera.c > index c71a00736541..4c82d1880753 100644 > --- a/drivers/media/platform/pxa_camera.c > +++ b/drivers/media/platform/pxa_camera.c > @@ -2357,8 +2357,6 @@ static int pxa_camera_probe(struct platform_device *pdev) > .src_maxburst = 8, > .direction = DMA_DEV_TO_MEM, > }; > - dma_cap_mask_t mask; > - struct pxad_param params; > char clk_name[V4L2_CLK_NAME_SIZE]; > int irq; > int err = 0, i; > @@ -2432,34 +2430,20 @@ static int pxa_camera_probe(struct platform_device *pdev) > pcdev->base = base; > > /* request dma */ > - dma_cap_zero(mask); > - dma_cap_set(DMA_SLAVE, mask); > - dma_cap_set(DMA_PRIVATE, mask); > - > - params.prio = 0; > - params.drcmr = 68; > - pcdev->dma_chans[0] = > - dma_request_slave_channel_compat(mask, pxad_filter_fn, > - &params, &pdev->dev, "CI_Y"); > + pcdev->dma_chans[0] = dma_request_slave_channel(&pdev->dev, "CI_Y"); > if (!pcdev->dma_chans[0]) { > dev_err(&pdev->dev, "Can't request DMA for Y\n"); > return -ENODEV; > } > > - params.drcmr = 69; > - pcdev->dma_chans[1] = > - dma_request_slave_channel_compat(mask, pxad_filter_fn, > - &params, &pdev->dev, "CI_U"); > + pcdev->dma_chans[1] = dma_request_slave_channel(&pdev->dev, "CI_U"); > if (!pcdev->dma_chans[1]) { > dev_err(&pdev->dev, "Can't request DMA for Y\n"); > err = -ENODEV; > goto exit_free_dma_y; > } > > - params.drcmr = 70; > - pcdev->dma_chans[2] = > - dma_request_slave_channel_compat(mask, pxad_filter_fn, > - &params, &pdev->dev, "CI_V"); > + pcdev->dma_chans[2] = dma_request_slave_channel(&pdev->dev, "CI_V"); > if (!pcdev->dma_chans[2]) { > dev_err(&pdev->dev, "Can't request DMA for V\n"); > err = -ENODEV; >

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[alsa-devel] [PATCH] ALSA: add more quirks for DSD interfaces
by Daniel Mack 22 Apr '18

22 Apr '18

Based on a downstream patch from Harry ten Berge. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Reported-and-tested-by: wenyi(a)tianyu-wool.com Original-by: Harry ten Berge <htenberge(a)gmail.com> --- sound/usb/quirks.c | 27 +++++++++++++++++++++++++++ 1 file changed, 27 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index acbeb52f6fd6..5681767cc0d5 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1327,20 +1327,47 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { + case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ + case USB_ID(0x20b1, 0x0002): /* Wyred 4 Sound DAC-2 DSD */ + case USB_ID(0x20b1, 0x2004): /* Matrix Audio X-SPDIF 2 */ case USB_ID(0x20b1, 0x3008): /* iFi Audio micro/nano iDSD */ case USB_ID(0x20b1, 0x2008): /* Matrix Audio X-Sabre */ case USB_ID(0x20b1, 0x300a): /* Matrix Audio Mini-i Pro */ case USB_ID(0x22d9, 0x0416): /* OPPO HA-1 */ + case USB_ID(0x22d9, 0x0436): /* OPPO Sonica */ + case USB_ID(0x22d9, 0x0461): /* OPPO UDP-205 */ + case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ + case USB_ID(0x25ce, 0x001f): /* Mytek Brooklyn DAC */ + case USB_ID(0x25ce, 0x0021): /* Mytek Manhattan DAC */ + case USB_ID(0x25ce, 0x8025): /* Mytek Brooklyn DAC+ */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; + case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ + case USB_ID(0x16b0, 0x06b2): /* NuPrime DAC-10 */ + case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ case USB_ID(0x20b1, 0x000a): /* Gustard DAC-X20U */ + case USB_ID(0x20b1, 0x2005): /* Denafrips Ares DAC */ case USB_ID(0x20b1, 0x2009): /* DIYINHK DSD DXD 384kHz USB to I2S/DSD */ case USB_ID(0x20b1, 0x2023): /* JLsounds I2SoverUSB */ + case USB_ID(0x20b1, 0x3021): /* Eastern El. MiniMax Tube DAC Supreme */ case USB_ID(0x20b1, 0x3023): /* Aune X1S 32BIT/384 DSD DAC */ + case USB_ID(0x20b1, 0x302d): /* Unison Research Unico CD Due */ + case USB_ID(0x20b1, 0x3036): /* Holo Springs Level 3 R2R DAC */ + case USB_ID(0x20b1, 0x307b): /* CH Precision C1 DAC */ + case USB_ID(0x20b1, 0x3086): /* Singxer F-1 converter board */ + case USB_ID(0x22d9, 0x0426): /* OPPO HA-2 */ + case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ + case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ + case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ + case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ + case USB_ID(0x6b42, 0x0042): /* MSB Technology */ if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; break; -- 2.14.3

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[alsa-devel] Old firmware version for Dice II ASIC
by Takashi Sakamoto 21 Apr '18

21 Apr '18

Hi Clemens, Recently I get Alesis Multimix 12 FireWire and work for ALSA dice driver to support it. This unit uses ASIC of 'WaveFront DICE II STD' and TCAT extended application protocol is not supported. Unfortunately, the driver in v4.17-rc1 can't handle this unit, due to validation of address sections for global space. $ dmesg snd_dice fw1.0: Sound card registration failed: -19 (sound/firewire/dice/dice-transaction.c) 265 static int get_subaddrs(struct snd_dice *dice) 266 { 267 static const int min_values[10] = { 268 10, 0x64 / 4, 269 10, 0x18 / 4, 270 10, 0x18 / 4, 271 0, 0, 272 0, 0, 273 }; ... 290 err = snd_fw_transaction(dice->unit, TCODE_READ_BLOCK_REQUEST, 291 DICE_PRIVATE_SPACE, pointers, 292 sizeof(__be32) * ARRAY_SIZE(min_values), 0); 293 if (err < 0) 294 goto end; 295 296 for (i = 0; i < ARRAY_SIZE(min_values); ++i) { 297 data = be32_to_cpu(pointers[i]); 298 if (data < min_values[i] || data >= 0x40000) { 299 err = -ENODEV; 300 goto end; 301 } 302 } The condition 'data < min_values[i]' is evaluated as true at a second iteration (i = 1). A quadlet of 0xffffe0000004-7 has 0x000018 and this is less than 0x19 (= 0x64 / 4) in fact. $ ./firewire-request /dev/fw1 read 0xffffe0000000 28 result: 000: 00 00 00 0a 00 00 00 18 00 00 00 22 00 00 00 8a result: 010: 00 00 00 ac 00 00 01 12 00 00 00 00 00 00 00 00 result: 020: 00 00 00 00 00 00 00 00 In line 183 of 'sound/firewire/dice/dice-interface.h', I can see below comments: (sound/firewire/dice/dice-interface.h) 175 #define GLOBAL_SAMPLE_RATE 0x05c ... 181 #define GLOBAL_VERSION 0x060 182 183 /* Some old firmware versions do not have the following global registers: */ ... 188 #define GLOBAL_CLOCK_CAPABILITIES 0x064 But this is not proper in my case because the maximum offset on global sub-address space is 0x05c in quadlet unit. Well, this unit supports 44.1/48.0 kHz and use internal/Rx1 as its sampling clock source according to user manual. This is what in 'check_clock_caps()'. (sound/firewire/dice/dice.c) 96 static int check_clock_caps(struct snd_dice *dice) 97 { 101 /* some very old firmwares don't tell about their clock support */ 102 if (dice->clock_caps > 0) { ... 109 } else { 110 /* this should be supported by any device */ 111 dice->clock_caps = CLOCK_CAP_RATE_44100 | 112 CLOCK_CAP_RATE_48000 | 113 CLOCK_CAP_SOURCE_ARX1 | 114 CLOCK_CAP_SOURCE_INTERNAL; 115 } In my opinion, GLOBAL_VERSION offset belongs to global address space for newer firmware version. Would I request any comment to you for this idea? Regards Takashi Sakamoto

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[alsa-devel] [PATCH 1/5] compress: remove dead code _is_codec_supported()
by Vinod Koul 21 Apr '18

21 Apr '18

The _is_codec_supported() was dead and none using it, so remove this and eliminate unused function warning compress.c:145:13: warning: ‘_is_codec_supported’ defined but not used [-Wunused-function] We can take from git if user appears Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- src/lib/compress.c | 53 ----------------------------------------------------- 1 file changed, 53 deletions(-) diff --git a/src/lib/compress.c b/src/lib/compress.c index 8ae6bbda7a89..934690e39ed0 100644 --- a/src/lib/compress.c +++ b/src/lib/compress.c @@ -142,49 +142,6 @@ static int get_compress_version(struct compress *compress) return version; } -static bool _is_codec_supported(struct compress *compress, struct compr_config *config, - const struct snd_compr_caps *caps) -{ - bool codec = false; - unsigned int i; - - for (i = 0; i < caps->num_codecs; i++) { - if (caps->codecs[i] == config->codec->id) { - /* found the codec */ - codec = true; - break; - } - } - if (codec == false) { - oops(compress, ENXIO, "this codec is not supported"); - return false; - } - - if (config->fragment_size < caps->min_fragment_size) { - oops(compress, EINVAL, "requested fragment size %d is below min supported %d", - config->fragment_size, caps->min_fragment_size); - return false; - } - if (config->fragment_size > caps->max_fragment_size) { - oops(compress, EINVAL, "requested fragment size %d is above max supported %d", - config->fragment_size, caps->max_fragment_size); - return false; - } - if (config->fragments < caps->min_fragments) { - oops(compress, EINVAL, "requested fragments %d are below min supported %d", - config->fragments, caps->min_fragments); - return false; - } - if (config->fragments > caps->max_fragments) { - oops(compress, EINVAL, "requested fragments %d are above max supported %d", - config->fragments, caps->max_fragments); - return false; - } - - /* TODO: match the codec properties */ - return true; -} - static bool _is_codec_type_supported(int fd, struct snd_codec *codec) { struct snd_compr_caps caps; @@ -271,16 +228,6 @@ struct compress *compress_open(unsigned int card, unsigned int device, config->fragments = caps.max_fragments; } -#if 0 - /* FIXME need to turn this On when DSP supports - * and treat in no support case - */ - if (_is_codec_supported(compress, config, &caps) == false) { - oops(compress, errno, "codec not supported\n"); - goto codec_fail; - } -#endif - memcpy(compress->config, config, sizeof(*compress->config)); fill_compress_params(config, &params); -- 2.7.4

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[alsa-devel] [PATCH alsa-lib v4 0/4] alsa-lib: ASoC: topology: Improve hw_configs
by Kirill Marinushkin 20 Apr '18

20 Apr '18

Hello Takashi, This patch series waits since 20th of February, because it modifies UAPI headers and should be done in sync with Linux and ALSA. I see that previously there was no clear understanding, how to do this in sync. We discussed with you on 02 Apr 2018, and you suggested the following steps: Step 1. A person from Intel, who knows the topic (Pierre), adds his "Reviewed-by" or "Acked-by" tags Step 2. I clarify with Mark to apply the Linux part (different patch series) Step 3. After that, I clarify with you to apply the ALSA part (this patch series) The step 1 is done. Old patches are reviewed by Pierre, so they have his "Reviewed-by" tag. New patches are suggested by him, and we agreed on the implementation, so they have his "Acked-by" tag. The step 2 is done. Mark applied the corresponding part to his Linux tree [1]. Now we are on the step 3. If you don't have anything against this patch series, please apply it to ALSA. Best Regards, Kirill [1] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git Kirill Marinushkin (4): ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format() ASoC: topology: Add missing clock gating parameter when parsing hw_configs ASoC: topology: Add definitions for mclk_direction values ASoC: topology: Add alias conf parameter names for hw_configs include/sound/asoc.h | 29 +++++++++-- include/topology.h | 8 +-- src/conf/topology/broadwell/broadwell.conf | 4 +- src/topology/pcm.c | 81 +++++++++++++++++++++++------- 4 files changed, 95 insertions(+), 27 deletions(-) -- 2.13.6

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[alsa-devel] [PATCH][next] ASoC: rt5668: fix incorrect 'and' operator
by Colin King 20 Apr '18

20 Apr '18

From: Colin Ian King <colin.king(a)canonical.com> Currently logical and is being used instead of bitwise and. Fix this. Detected by CoverityScan, CID#1468008 ("Logical vs bitwise operator") Fixes: d59fb2856223 ("ASoC: rt5668: add rt5668B codec driver") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5668.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 52a343f96eb2..3c19d03f2446 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -1194,7 +1194,7 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, int ref, val, reg, idx = -EINVAL; static const int div[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; - val = snd_soc_component_read32(component, RT5668_GPIO_CTRL_1) && + val = snd_soc_component_read32(component, RT5668_GPIO_CTRL_1) & RT5668_GP4_PIN_MASK; if (w->shift == RT5668_PWR_ADC_S1F_BIT && val == RT5668_GP4_PIN_ADCDAT2) -- 2.17.0

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