October 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: dapm: Fix NULL pointer dereference when registering card with widgets
by Jarkko Nikula 03 Oct '14

03 Oct '14

Commit 0bd2ac3dae74 ("ASoC: Remove CODEC pointer from snd_soc_dapm_context") introduced regression to snd_soc_dapm_new_controls() when registering a card with card->dapm_widgets set. Call chain is: snd_soc_register_card() -> snd_soc_instantiate_card() -> snd_soc_dapm_new_controls() -> snd_soc_dapm_new_control() Null pointer dereference occurs since card->dapm context doesn't have associated component. Fix this by setting widget codec pointer conditionally. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 2d06892088d5..3f0b4108f7ca 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3125,7 +3125,8 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } w->dapm = dapm; - w->codec = dapm->component->codec; + if (dapm->component) + w->codec = dapm->component->codec; INIT_LIST_HEAD(&w->sources); INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); -- 2.1.1

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[alsa-devel] [PATCH 1/2] ASoC: tegra: add mic detect gpio to tegra_max98090
by Dylan Reid 03 Oct '14

03 Oct '14

Add an optional mic detect gpio property. If specified in device tree there will be a mic jack created for the given gpio. This will be used by the Tegra-based Chromebooks. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../bindings/sound/nvidia,tegra-audio-max98090.txt | 1 + sound/soc/tegra/tegra_max98090.c | 40 ++++++++++++++++++++++ 2 files changed, 41 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt index 9c7c55c..c949abc 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra-audio-max98090.txt @@ -25,6 +25,7 @@ Required properties: Optional properties: - nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in +- nvidia,mic-det-gpios : The GPIO that detect microphones are plugged in Example: diff --git a/sound/soc/tegra/tegra_max98090.c b/sound/soc/tegra/tegra_max98090.c index b86cd99..01921d7 100644 --- a/sound/soc/tegra/tegra_max98090.c +++ b/sound/soc/tegra/tegra_max98090.c @@ -42,6 +42,7 @@ struct tegra_max98090 { struct tegra_asoc_utils_data util_data; int gpio_hp_det; + int gpio_mic_det; }; static int tegra_max98090_asoc_hw_params(struct snd_pcm_substream *substream, @@ -112,6 +113,22 @@ static struct snd_soc_jack_gpio tegra_max98090_hp_jack_gpio = { .invert = 1, }; +static struct snd_soc_jack tegra_max98090_mic_jack; + +static struct snd_soc_jack_pin tegra_max98090_mic_jack_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, +}; + +static struct snd_soc_jack_gpio tegra_max98090_mic_jack_gpio = { + .name = "Mic detection", + .report = SND_JACK_MICROPHONE, + .debounce_time = 150, + .invert = 1, +}; + static const struct snd_soc_dapm_widget tegra_max98090_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphones", NULL), SND_SOC_DAPM_SPK("Speakers", NULL), @@ -141,6 +158,19 @@ static int tegra_max98090_asoc_init(struct snd_soc_pcm_runtime *rtd) &tegra_max98090_hp_jack_gpio); } + if (gpio_is_valid(machine->gpio_mic_det)) { + snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, + &tegra_max98090_mic_jack); + snd_soc_jack_add_pins(&tegra_max98090_mic_jack, + ARRAY_SIZE(tegra_max98090_mic_jack_pins), + tegra_max98090_mic_jack_pins); + + tegra_max98090_mic_jack_gpio.gpio = machine->gpio_mic_det; + snd_soc_jack_add_gpios(&tegra_max98090_mic_jack, + 1, + &tegra_max98090_mic_jack_gpio); + } + return 0; } @@ -153,6 +183,11 @@ static int tegra_max98090_card_remove(struct snd_soc_card *card) &tegra_max98090_hp_jack_gpio); } + if (gpio_is_valid(machine->gpio_mic_det)) { + snd_soc_jack_free_gpios(&tegra_max98090_mic_jack, 1, + &tegra_max98090_mic_jack_gpio); + } + return 0; } @@ -201,6 +236,11 @@ static int tegra_max98090_probe(struct platform_device *pdev) if (machine->gpio_hp_det == -EPROBE_DEFER) return -EPROBE_DEFER; + machine->gpio_mic_det = + of_get_named_gpio(np, "nvidia,mic-det-gpios", 0); + if (machine->gpio_mic_det == -EPROBE_DEFER) + return -EPROBE_DEFER; + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; -- 2.1.2.330.g565301e

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[alsa-devel] [PATCH] ASoC: sgtl5000: Do a sanity check on SYS_MCLK
by Fabio Estevam 03 Oct '14

03 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> According to the sgtl5000 datasheet the valid range for SYS_MCLK is from 8 to 27 MHz. Add a sanity check prior to enabling SYS_MCLK. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index a604a22..1470437 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1414,6 +1414,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, { struct sgtl5000_priv *sgtl5000; int ret, reg, rev; + unsigned int mclk; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1437,6 +1438,14 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } + /* SGTL5000 SYS_MCLK should be between 8 and 27 MHz */ + mclk = clk_get_rate(sgtl5000->mclk); + if (mclk < 8000000 || mclk > 27000000) { + dev_err(&client->dev, "Invalid SYS_CLK frequency: %u.%03uMHz\n", + mclk / 1000000, mclk / 1000 % 1000); + return -EINVAL; + } + ret = clk_prepare_enable(sgtl5000->mclk); if (ret) return ret; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: sgtl5000: Improve the error message on slave mode setting
by Fabio Estevam 03 Oct '14

03 Oct '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> For sgtl5000 to operate in slave mode it can only work in "Synchronous SYS_MCLK input" mode. In this mode only the following rates can be supported: 256*Fs, 384*Fs, 512*Fs. Improve the error message to give a better indication as to why the clocking failed for slave mode: [ 12.515399] sgtl5000 1-000a: PLL not supported in slave mode [ 12.524124] sgtl5000 1-000a: 233 ratio is not supported. SYS_MCLK needs to be 256, 384 or 512 * fs [ 12.535938] sgtl5000 1-000a: ASoC: can't set sgtl5000 hw params: -22 Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index a604a22..c1936bc 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -626,6 +626,9 @@ static int sgtl5000_set_clock(struct snd_soc_codec *codec, int frame_rate) } else { dev_err(codec->dev, "PLL not supported in slave mode\n"); + dev_err(codec->dev, "%d ratio is not supported. " + "SYS_MCLK needs to be 256, 384 or 512 * fs\n", + sgtl5000->sysclk / sys_fs); return -EINVAL; } } -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rt286: Add depends on I2C
by Bard Liao 03 Oct '14

03 Oct '14

rt286 use I2C as its I/O. So the driver can only available when I2C is selected. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 351bf48..a68d173 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -484,6 +484,7 @@ config SND_SOC_RL6231 config SND_SOC_RT286 tristate + depends on I2C config SND_SOC_RT5631 tristate -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v7 0/2] ASoC: tda998x: add a codec to the HDMI transmitter
by Jean-Francois Moine 03 Oct '14

03 Oct '14

The NXP TDA998x HDMI transmitter may transmit audio to the HDMI link from 2 different sources, I2S and S/PDIF. This patch set adds an interface between the HDMI transmitter and the HDMI CODEC. The interface is used by the TDA998x driver to update the audio constraints from the display characteristics (EDID) and by the audio subsystem to connect the chosen audio source to the HDMI link. v7: - remove the change of the K predivider (Jyri Sarha) - add S24_3LE and S32_LE as possible audio formats (Jyri Sarha) - don't move the struct priv2 definition and use the slave encoder private data as the device private data (Russell King) - remove the useless request_module (Russell King/Mark Brown) - don't lock the HDMI module (Russell King) - use platform_device_unregister to remove the codec (Russell King) v6: - extend the HDMI CODEC instead of using a specific CODEC v5: - use the TDA998x private data instead of a specific area for the CODEC interface - the CODEC is TDA998x specific (Mark Brown) v4: - remove all the TDA998x specific stuff from the CODEC - move the EDID scan from the CODEC to the TDA998x - move the CODEC to sound/soc (Mark Brown) - update the audio_sample_rate from the EDID (Andrew Jackson) v3: fix bad rate (Andrew Jackson) v2: check double stream start (Mark Brown) Jean-Francois Moine (2): ASoC: codecs: Add a transmitter interface to the HDMI CODEC drm/i2c: tda998x: Use the HDMI audio CODEC .../devicetree/bindings/drm/i2c/tda998x.txt | 18 ++ drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 264 ++++++++++++++++++++- include/sound/hdmi.h | 20 ++ sound/soc/codecs/hdmi.c | 176 +++++++++++++- 5 files changed, 460 insertions(+), 19 deletions(-) create mode 100644 include/sound/hdmi.h -- 2.1.1

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[alsa-devel] [BUG] Bad decibel data on C-Media USB sound card
by Allen M. 03 Oct '14

03 Oct '14

Raymond Yau wrote: >Does this number of steps remain the same when your starting volume are not at 151? >Your value 77 seem quite different from alsamixertest recording a sine wave with an amplitude of 1 in Audacity at step 151 shows that the recorded value is amplitude 0.885. Does this indicates some loss? The loopback cable I am using is 2 meters in length, is the resistance of the cable is too high?

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[alsa-devel] [bug] SPDIF (digital-out) not working on NForce4/CK804
by Gilles Hemberg 03 Oct '14

03 Oct '14

Hi all, Sorry to bump in like this, but I was advised to submit a bug-report for the ALSA-drivers on my HTPC: no sound comes through out of my AVR when I select the digital output through SPDIF (TOSLINK). The relevant parts of my HTPC are as follows: * CPU: Athlon64 X2 3800+ * Motherboard: ASUS A8N-SLI Deluxe motherboard (BIOS 1805) NForce4 with CK804 sound (relying on a Realtek ALC850 chip) * RAM: 2x 512 MB DDR-400 * GPU: nVidia Geforce GT240 (which carries a HDMI port and uses the snd- * Receiver: Marantz SR7500 (supporting both Dolby Digital 5.1 and DTS decoding), connected to motherboard's SPDIF TOSLINK port (optical) + JBL 5.1 speaker-set * OpenElec 4.2.0: Kernel 3.16.3, Alsa 1.0.28 Symptoms: Speaker-test does not produce any sound. Command used is as follows: speaker-test -c2 -r 48000 -D hw:0,2 No sound either from other applications (such as XBMC 13.2). SPDIF worked flawlessly on the exact same hardware & connections using Win7/XBMC. Sound does come through the SPDIF when I use hw:0,0 (= analog device). Things tried (none working): * Blacklisting snd-hda-intel (= HDMI from my GPU) to make sure CK804 is the only sound chip visible. * amixer set 'IEC958 Playback AC97-SPSA' Playback 100% unmute * Contacted XBMC developer via IRC for investigation: concluded that problem would be with kernel driver, advised to create this bug-report. * Tried another distro (Ubuntu from XBMCBuntu 13.0). Further details which might be useful: * alsa-info.sh output: http://sprunge.us/fMeR * /proc/asound/card0/ac97#0-0: http://pastebin.com/Q5y1J0Ra * /proc/asound/card0/ac97#0-0+re gs: http://pastebin.com/tBbTCxZg Similar reports found (no working solution): http://mailman.alsa-project.org/pipermail/alsa-devel/2009-September/021336.… http://askubuntu.com/questions/264272/no-sound-output-msi-k8n-neo4-ck804-ac… http://askubuntu.com/questions/462605/new-install-of-ubuntu-14-04-trusty-3-… https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=615225 http://ehc.ac/p/alsa/mailman/message/25973743/ http://wiki.linuxmce.org/index.php/ASUS_A8N-SLI_Deluxe http://forum.xbmc.org/showthread.php?tid=156043&pid=1348176 http://forum.xbmc.org/showthread.php?tid=168608 http://ubuntuforums.org/showthread.php?t=2210288 http://cholla.mmto.org/computers/linux/sound/asus_a8ne.html http://www.nvnews.net/vbulletin/showthread.php?t=80521 ... I am willing to test or provide further details where needed. I am new to Linux, but am a developer myself (C#) so I know how to look up some info... Hoping I haven't offended any of you with this post. THNX, Bye! _____ Gilles Hemberg gilles.hemberg(a)gmail.com

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[alsa-devel] [bug] SPDIF (digital-out) not working on NForce4/CK804
by Gilles Hemberg 03 Oct '14

03 Oct '14

Hi all, Sorry to bump in like this, but I was advised to submit a bug-report for the ALSA-drivers on my HTPC: no sound comes through out of my AVR when I select the digital output through SPDIF (TOSLINK). The relevant parts of my HTPC are as follows: * CPU: Athlon64 X2 3800+ * Motherboard: ASUS A8N-SLI Deluxe motherboard (BIOS 1805) NForce4 with CK804 sound (relying on a Realtek ALC850 chip) * RAM: 2x 512 MB DDR-400 * GPU: nVidia Geforce GT240 (which carries a HDMI port and uses the snd- * Receiver: Marantz SR7500 (supporting both Dolby Digital 5.1 and DTS decoding), connected to motherboard's SPDIF TOSLINK port (optical) + JBL 5.1 speaker-set * OpenElec 4.2.0: Kernel 3.16.3, Alsa 1.0.28 Symptoms: Speaker-test does not produce any sound. Command used is as follows: speaker-test -c2 -r 48000 -D hw:0,2 No sound either from other applications (such as XBMC 13.2). SPDIF worked flawlessly on the exact same hardware & connections using Win7/XBMC. Sound does come through the SPDIF when I use hw:0,0 (= analog device). Things tried (none working): * Blacklisting snd-hda-intel (= HDMI from my GPU) to make sure CK804 is the only sound chip visible. * amixer set 'IEC958 Playback AC97-SPSA' Playback 100% unmute * Contacted XBMC developer via IRC for investigation: concluded that problem would be with kernel driver, advised to create this bug-report. * Tried another distro (Ubuntu from XBMCBuntu 13.0). Further details which might be useful: * alsa-info.sh output: http://sprunge.us/fMeR * /proc/asound/card0/ac97#0-0: http://pastebin.com/Q5y1J0Ra * /proc/asound/card0/ac97#0-0+re gs: http://pastebin.com/tBbTCxZg Similar reports found (no working solution): <http://mailman.alsa-project.org/pipermail/alsa-devel/2009-September/021336.…> http://mailman.alsa-project.org/pipermail/alsa-devel/2009-September/021336.… http://askubuntu.com/questions/264272/no-sound-output-msi-k8n-neo4-ck804-ac… http://askubuntu.com/questions/462605/new-install-of-ubuntu-14-04-trusty-3-… https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=615225 http://ehc.ac/p/alsa/mailman/message/25973743/ http://wiki.linuxmce.org/index.php/ASUS_A8N-SLI_Deluxe http://forum.xbmc.org/showthread.php?tid=156043&pid=1348176 http://forum.xbmc.org/showthread.php?tid=168608 http://ubuntuforums.org/showthread.php?t=2210288 http://cholla.mmto.org/computers/linux/sound/asus_a8ne.html http://www.nvnews.net/vbulletin/showthread.php?t=80521 ... I am willing to test or provide further details where needed. I am new to Linux, but am a developer myself (C#) so I know how to look up some info... Hoping I haven't offended any of you with this post. THNX, Bye! _____ Gilles Hemberg gilles.hemberg(a)gmail.com

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[alsa-devel] [PATCH] ASoC: samsung: fix CDCLK handling
by Daniel Drake 03 Oct '14

03 Oct '14

ODROID is the only platform that uses CDCLK, and right now, CDCLK handling is buggy. If you start pulseaudio on ODROID, audio is broken until reboot (even after killing pulse). This happens because CDCLK gets disabled by i2s.c and never enabled again. pulseaudio does: 1. i2s_startup for playback channel 2. i2s_startup for capture channel 3. i2s_shutdown for capture channel 4. i2s_shutdown for playback channel In step 3 we disable CDCLK even though playback should still be active. In step 4 we do this: u32 mod = readl(i2s->addr + I2SMOD); i2s->cdclk_out = !(mod & MOD_CDCLKCON); and now cdclk_out is always going to be 0, so we'll never turn it back on again. Both this bug and the one that b97c60abf9a tries to fix happened because of the way that CDCLK handling is painfully split between platform and i2s drivers. Simplify the situation and solve the bug with the following approach: - as before, samsung_i2s_dai_probe() gates CDCLK by default (no need for smartq_wm8987 to do this as well) - platform drivers can gate/ungate CDCLK as necessary (currently only odroidx2 needs to do this) - i2s code has no other interaction with CDCLK Signed-off-by: Daniel Drake <drake(a)endlessm.com> --- sound/soc/samsung/i2s.c | 19 ++---------------- sound/soc/samsung/odroidx2_max98090.c | 36 ++++++++++++++++++++++++++--------- sound/soc/samsung/smartq_wm8987.c | 6 ------ 3 files changed, 29 insertions(+), 32 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 9d51347..6654ce6 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -68,8 +68,6 @@ struct i2s_dai { #define DAI_OPENED (1 << 0) /* Dai is opened */ #define DAI_MANAGER (1 << 1) /* Dai is the manager */ unsigned mode; - /* CDCLK pin direction: 0 - input, 1 - output */ - unsigned int cdclk_out:1; /* Driver for this DAI */ struct snd_soc_dai_driver i2s_dai_drv; /* DMA parameters */ @@ -739,9 +737,6 @@ static int i2s_startup(struct snd_pcm_substream *substream, spin_unlock_irqrestore(&lock, flags); - if (!is_opened(other) && i2s->cdclk_out) - i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_OUT); return 0; } @@ -757,24 +752,14 @@ static void i2s_shutdown(struct snd_pcm_substream *substream, i2s->mode &= ~DAI_OPENED; i2s->mode &= ~DAI_MANAGER; - if (is_opened(other)) { + if (is_opened(other)) other->mode |= DAI_MANAGER; - } else { - u32 mod = readl(i2s->addr + I2SMOD); - i2s->cdclk_out = !(mod & MOD_CDCLKCON); - if (other) - other->cdclk_out = i2s->cdclk_out; - } + /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; spin_unlock_irqrestore(&lock, flags); - - /* Gate CDCLK by default */ - if (!is_opened(other)) - i2s_set_sysclk(dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_IN); } static int config_setup(struct i2s_dai *i2s) diff --git a/sound/soc/samsung/odroidx2_max98090.c b/sound/soc/samsung/odroidx2_max98090.c index 278edf9..b700284 100644 --- a/sound/soc/samsung/odroidx2_max98090.c +++ b/sound/soc/samsung/odroidx2_max98090.c @@ -21,20 +21,37 @@ struct odroidx2_drv_data { /* The I2S CDCLK output clock frequency for the MAX98090 codec */ #define MAX98090_MCLK 19200000 +static int odroidx2_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + if (rtd->cpu_dai->active) + return 0; + + return snd_soc_dai_set_sysclk(rtd->cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); +} + +static void odroidx2_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + if (!rtd->cpu_dai->active) + snd_soc_dai_set_sysclk(rtd->cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_IN); +} + +static const struct snd_soc_ops odroidx2_ops = { + .startup = odroidx2_startup, + .shutdown = odroidx2_shutdown, +}; + static int odroidx2_late_probe(struct snd_soc_card *card) { struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; - struct snd_soc_dai *cpu_dai = card->rtd[0].cpu_dai; - int ret; - ret = snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, + return snd_soc_dai_set_sysclk(codec_dai, 0, MAX98090_MCLK, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* Set the cpu DAI configuration in order to use CDCLK */ - return snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_OUT); } static const struct snd_soc_dapm_widget odroidx2_dapm_widgets[] = { @@ -55,6 +72,7 @@ static struct snd_soc_dai_link odroidx2_dai[] = { .codec_dai_name = "HiFi", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, + .ops = &odroidx2_ops, } }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index 9b0ffac..2b5e7c5 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -76,12 +76,6 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - /* Gate the RCLK output on PAD */ - ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); -- 1.9.1

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