May 2020 - Alsa-devel - mailman.alsa-project.org

[RFC] DPCM for Tegra
by Sameer Pujar 06 May '20

06 May '20

Hi Mark, Earlier I had sent Tegra ASoC series [0] for review and writing back to follow up on the same. Background ========== There were following concerns on [0]: - Usage of mixer control overrides in each driver for PCM parameters. - Exposure of routing controls to the user. The comments are mostly captured in [1] and [2]. There was a suggestion to use DPCM for the Tegra audio requirements. Note: As of now, below does not cover specific solution for propogation of audio configurations or PCM parameters (like sample rate, sample size and channels). DPCM Testing ============ Since then I was looking into internals of DPCM and was trying to get it working with Tegra. I was able to get following things working. - Audio playback/capture over I2S/DMIC/DSPK. - Audio resampling use case. - Mixing of two audio streams with resampler in the path. Objective was to understand and get audio working with DPCM. Please note that I used simple-card DPCM driver for a quick testing on top of the above Tegra Audio series. I had to tweak the simple-card driver a little, but currently keeping it out of the scope of current discussion. At a high level Tegra Audio HW is depicted as below. | Front End PCMs | SoC DSP | Back End DAIs | ************* ADMAIF<0> <------------> * * <----DAI<0>-----> I2S * * ADMAIF<1> <------------> * * <----DAI<1>-----> DMIC * XBAR * ADMAIF<2> <------------> * * <----DAI<2>-----> DSPK * * ADMAIF<N> <------------> * * <----DAI<3>-----> SFC (Resampler) * * * * <----DAI<4>-----> MIXER * * * * <----DAI<N>-----> ... ************* Note: ----- * XBAR is just a cross bar interconnecting one component to another. Specific switch needs to be programmed for audio data to flow from one component to another. * SFC or Mixer are separate HW blocks and separate ASoC drivers are written for these. These drivers were not sent earlier as part of initial upstream series [0]. Follow up queries ================= Based on the above experience I do have few follow up queries and request for your inputs on this. a) Can I use a DAPM Mux control to activate a BE path? This in turn can program required switch in XBAR. This is needed for following reasons: - For an open platform like Jetson, we want to give maximum flexibility for a user to customize their audio paths. Number of connected components and order of these can vary depending on a use case. - Allow re-use of audio components across multiple use cases. For example, number of SFC instances are lesser than PCM playback or capture devices. b) I have modelled SFC and MIXER as backends. Is this allowed? This was done to include SFC or MIXER HW components as part of the sound card and use like below in one of the audio use cases. ADMAIF1(FE) --> SFC(BE1) --> I2S(BE2) ... OR ADMAIF2(FE) --> SFC(BE1) --> I2S(BE2) ... I used following workaround to connect multiple BE components. With this I can see PCM callbacks happen for all BE DAIs along the DAPM path. The obective was to connect multiple components together and (a) was used to connect one component to another. Each "-->" here connects two components and it is a switch in XBAR. --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e256d43..ee7af55 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1494,7 +1494,7 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, /* get number of valid DAI paths and their widgets */ paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, - dpcm_end_walk_at_be); + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- c) Hostless mode did NOT work: - Following audio path was intended to be tested: I2S1 --> SFC --> I2S2 - [3] offers two options: * CODEC<->CODEC: If I were to use a separate DAI link for each BE to BE connection, then it will result in a similar design what we have currently. * Hostless: I did not come across references for this. (Any references in this regard will be helpful) May be the current Tegra ASoC design is more suitable for component model as you had previously mentioned. I wanted to understand if above, especially (a) and (b), are acceptable in this regard or if there are better options to interconnect multiple ASoC components. Looking forward for your feedback. Thanks, Sameer. References ========== [0] http://patchwork.ozlabs.org/project/linux-tegra/list/?series=159664&archive… [1] http://patchwork.ozlabs.org/project/linux-tegra/patch/1582180492-25297-6-gi… [2] http://patchwork.ozlabs.org/project/linux-tegra/patch/1582180492-25297-4-gi… [3] https://www.kernel.org/doc/html/v5.6/sound/soc/dpcm.html

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[hints] arecord -L show entries which are not capture pcms
by GitHub issues - reopened 06 May '20

06 May '20

alsa-project/alsa-lib issue #39 was reopened from sylware: arecord -L will show pcms which are not capture pcms (for instance front speakers, surroundX). Don't know if it is fix-able. Wait for topology/ucm port of classic drivers? Issue URL : https://github.com/alsa-project/alsa-lib/issues/39 Repository URL: https://github.com/alsa-project/alsa-lib

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RE: [PATCH 0/1] Fiji GPU audio register timeout when in BACO state
by Deucher, Alexander 06 May '20

06 May '20

[AMD Public Use] > -----Original Message----- > From: Nicholas Johnson <nicholas.johnson-opensource(a)outlook.com.au> > Sent: Sunday, April 26, 2020 12:02 PM > To: linux-kernel(a)vger.kernel.org > Cc: Deucher, Alexander <Alexander.Deucher(a)amd.com>; Koenig, Christian > <Christian.Koenig(a)amd.com>; Zhou, David(ChunMing) > <David1.Zhou(a)amd.com>; Nicholas Johnson <nicholas.johnson- > opensource(a)outlook.com.au> > Subject: [PATCH 0/1] Fiji GPU audio register timeout when in BACO state > > Hi all, > > Since Linux v5.7-rc1 / commit 4fdda2e66de0 ("drm/amdgpu/runpm: enable > runpm on baco capable VI+ asics"), my AMD R9 Nano has been using runpm / > BACO. You can tell visually when it sleeps, because the fan on the graphics > card is switched off to save power. It did not spin down the fan in v5.6.x. > > This is great (I love it), except that when it is sleeping, the PCIe audio function > of the GPU has issues if anything tries to access it. You get dmesg errors such > as these: > > snd_hda_intel 0000:08:00.1: spurious response 0x0:0x0, last cmd=0x170500 > snd_hda_intel 0000:08:00.1: azx_get_response timeout, switching to polling > mode: last cmd=0x001f0500 snd_hda_intel 0000:08:00.1: No response from > codec, disabling MSI: last cmd=0x001f0500 snd_hda_intel 0000:08:00.1: No > response from codec, resetting bus: last cmd=0x001f0500 > snd_hda_codec_hdmi hdaudioC1D0: Unable to sync register 0x2f0d00. -11 > > The above is with the Fiji XT GPU at 0000:08:00.0 in a Thunderbolt enclosure > (not that Thunderbolt should affect it, but I feel I should mention it just in > case). I dropped a lot of duplicate dmesg lines, as some of them repeated a > lot of times before the driver gave up. > > I offer this patch to disable runpm for Fiji while a fix is found, if you decide > that is the best approach. Regardless, I will gladly test any patches you come > up with instead and confirm that the above issue has been fixed. > > I cannot tell if any other GPUs are affected. The only other cards to which I > have access are a couple of AMD R9 280X (Tahiti XT), which use radeon driver > instead of amdgpu driver. Adding a few more people. Do you know what is accessing the audio? The audio should have a dependency on the GPU device. The GPU won't enter runtime pm until the audio has entered runtime pm and vice versa on resume. Please attach a copy of your dmesg output and lspci output. Alex

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[PATCH] ASoC: amd :High hw_level while simultaneous capture
by Akshu Agrawal 06 May '20

06 May '20

Simultaneous capture on dmic and headset mic is having issue with high hw_level being reported. Issue Can be reproduced by: arecord -D hw:2,0 -f dat -d 60 /tmp/test0 & arecord -D hw:2,2 -f dat -d 60 /tmp/test1 & cat /proc/asound/card2/pcm?c/sub0/status Actual issue is : When we open one capture stream on one instance lets say I2S_SP and then once again if we open other capture on other instance lets say I2S_BT while first capture is in progress and when we try to read the status of both running instances by below command cat /proc/asound/card2/pcm?c/sub0/status we observe that avail_max is being doubled on first opened capture(I2S_SP in the example). This is because our previous implementation was like when any instance is opened it gets initialized in dma_open irrespective of on what instance it called open. For example: First I2S_SP called opened it initializes both SP/BT capture streams irrespective of on which instance the stream opened.next time I2S_BT called opened and it initializes both SP/BT this corrupts the behaviour . So with this patch the stream gets initialized only on specific instance when ever it gets opened calls hw_params. This rectifies the issue. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 37 +++++++++++++++++++---------- 1 file changed, 24 insertions(+), 13 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index e362f0bc9e46..a36c5cb848cd 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -241,14 +241,6 @@ static int acp3x_dma_open(struct snd_soc_component *component, adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_stream = substream; - adata->i2ssp_play_stream = substream; - } else { - adata->capture_stream = substream; - adata->i2ssp_capture_stream = substream; - } - i2s_data->acp3x_base = adata->acp3x_base; runtime->private_data = i2s_data; return ret; @@ -263,23 +255,42 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; struct acp3x_platform_info *pinfo; + struct i2s_dev_data *adata; u64 size; prtd = substream->private_data; card = prtd->card; pinfo = snd_soc_card_get_drvdata(card); + adata = dev_get_drvdata(component->dev); rtd = substream->runtime->private_data; if (!rtd) return -EINVAL; - if (pinfo) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (pinfo) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { rtd->i2s_instance = pinfo->play_i2s_instance; - else + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + adata->play_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_play_stream = substream; + } + } else { rtd->i2s_instance = pinfo->cap_i2s_instance; - else + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + adata->capture_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_capture_stream = substream; + } + } + } else { pr_err("pinfo failed\n"); - + } size = params_buffer_bytes(params); rtd->dma_addr = substream->dma_buffer.addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); -- 2.17.1

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[PATCH v2 1/2] dt-bindings: sound: lpass-cpu: Document DAI subnodes
by Stephan Gerhold 06 May '20

06 May '20

The lpass-cpu driver now allows configuring the MI2S SD lines by defining subnodes for one of the DAIs. Document this in the device tree bindings. Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Changes in v2: - Clarify number of entries for qcom,playback/capture-sd-lines - Suggest more generic node names (dai@...) for children DAI device nodes v1: https://lore.kernel.org/alsa-devel/20200406135608.126171-1-stephan@gerhold.… --- .../bindings/sound/qcom,lpass-cpu.txt | 25 +++++++++++++++++++ 1 file changed, 25 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt index 21c648328be9..32c2cdb3d32f 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.txt @@ -30,6 +30,8 @@ Required properties: - reg : Must contain an address for each entry in reg-names. - reg-names : A list which must include the following entries: * "lpass-lpaif" +- #address-cells : Must be 1 +- #size-cells : Must be 0 @@ -37,6 +39,20 @@ Optional properties: - qcom,adsp : Phandle for the audio DSP node +By default, the driver uses up to 4 MI2S SD lines, for a total of 8 channels. +The SD lines to use can be configured by adding subnodes for each of the DAIs. + +Required properties for each DAI (represented by a subnode): +- reg : Must be one of the DAI IDs + (usually part of dt-bindings header) +- qcom,playback-sd-lines: List of serial data lines to use for playback + Each SD line should be represented by a number from 0-3. +- qcom,capture-sd-lines : List of serial data lines to use for capture + Each SD line should be represented by a number from 0-3. + +Note that adding a subnode changes the default to "no lines configured", +so both playback and capture lines should be configured when a subnode is added. + Example: lpass@28100000 { @@ -51,4 +67,13 @@ lpass@28100000 { reg = <0x28100000 0x10000>; reg-names = "lpass-lpaif"; qcom,adsp = <&adsp>; + + #address-cells = <1>; + #size-cells = <0>; + + /* Optional to set different MI2S SD lines */ + dai@3 { + reg = <MI2S_QUATERNARY>; + qcom,playback-sd-lines = <0 1>; + }; }; -- 2.26.2

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[PATCH] ALSA: hda: Return true,false for return type bool
by Jason Yan 06 May '20

06 May '20

Fix the following coccicheck warning: include/sound/hdaudio.h:210:73-74: WARNING: return of 0/1 in function 'snd_hdac_is_in_pm' with return type bool include/sound/hdaudio.h:211:76-77: WARNING: return of 0/1 in function 'snd_hdac_is_power_on' with return type bool Signed-off-by: Jason Yan <yanaijie(a)huawei.com> --- include/sound/hdaudio.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index d365297b3698..10945963d727 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -207,8 +207,8 @@ static inline int snd_hdac_power_down_pm(struct hdac_device *codec) { return 0; static inline int snd_hdac_keep_power_up(struct hdac_device *codec) { return 0; } static inline void snd_hdac_enter_pm(struct hdac_device *codec) {} static inline void snd_hdac_leave_pm(struct hdac_device *codec) {} -static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return 0; } -static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return 1; } +static inline bool snd_hdac_is_in_pm(struct hdac_device *codec) { return false; } +static inline bool snd_hdac_is_power_on(struct hdac_device *codec) { return true; } #endif /* -- 2.21.1

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[PATCH] ASoC: amd :High hw_level while simultaneous capture
by Akshu Agrawal 05 May '20

05 May '20

From: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Simultaneous capture on dmic and headset mic is having issue with high hw_level being reported. Issue Can be reproduced by: arecord -D hw:2,0 -f dat -d 60 /tmp/test0 & arecord -D hw:2,2 -f dat -d 60 /tmp/test1 & cat /proc/asound/card2/pcm?c/sub0/status Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 37 +++++++++++++++++++---------- 1 file changed, 24 insertions(+), 13 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index e362f0bc9e46..a36c5cb848cd 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -241,14 +241,6 @@ static int acp3x_dma_open(struct snd_soc_component *component, adata->i2ssp_play_stream && !adata->i2ssp_capture_stream) rv_writel(1, adata->acp3x_base + mmACP_EXTERNAL_INTR_ENB); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_stream = substream; - adata->i2ssp_play_stream = substream; - } else { - adata->capture_stream = substream; - adata->i2ssp_capture_stream = substream; - } - i2s_data->acp3x_base = adata->acp3x_base; runtime->private_data = i2s_data; return ret; @@ -263,23 +255,42 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, struct snd_soc_pcm_runtime *prtd; struct snd_soc_card *card; struct acp3x_platform_info *pinfo; + struct i2s_dev_data *adata; u64 size; prtd = substream->private_data; card = prtd->card; pinfo = snd_soc_card_get_drvdata(card); + adata = dev_get_drvdata(component->dev); rtd = substream->runtime->private_data; if (!rtd) return -EINVAL; - if (pinfo) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (pinfo) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { rtd->i2s_instance = pinfo->play_i2s_instance; - else + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + adata->play_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_play_stream = substream; + } + } else { rtd->i2s_instance = pinfo->cap_i2s_instance; - else + switch (rtd->i2s_instance) { + case I2S_BT_INSTANCE: + adata->capture_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + adata->i2ssp_capture_stream = substream; + } + } + } else { pr_err("pinfo failed\n"); - + } size = params_buffer_bytes(params); rtd->dma_addr = substream->dma_buffer.addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); -- 2.17.1

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[alsa-devel] [RESEND PATCH] ASoC: wm8524: Add support S32_LE
by Shengjiu Wang 05 May '20

05 May '20

Allow 32bit sample with this codec. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8524.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 91e3d1570c45..4e9ab542f648 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -159,7 +159,9 @@ static int wm8524_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define WM8524_RATES SNDRV_PCM_RATE_8000_192000 -#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define WM8524_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8524_dai_ops = { .startup = wm8524_startup, -- 2.21.0

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[PATCH 1/2] ALSA: hda: Use dev_to_hdac_dev macro
by Kai-Heng Feng 05 May '20

05 May '20

Use dev_to_hdac_dev() instead of container_of(). No functional change intended. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/hda/ext/hdac_ext_bus.c | 2 +- sound/hda/hdac_device.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 73bfa71845f6..d0a604c939df 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -62,7 +62,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_exit); static void default_release(struct device *dev) { - snd_hdac_ext_bus_device_exit(container_of(dev, struct hdac_device, dev)); + snd_hdac_ext_bus_device_exit(dev_to_hdac_dev(dev)); } /** diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index e3119f5cb0d5..333220f0f8af 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -20,7 +20,7 @@ static int get_codec_vendor_name(struct hdac_device *codec); static void default_release(struct device *dev) { - snd_hdac_device_exit(container_of(dev, struct hdac_device, dev)); + snd_hdac_device_exit(dev_to_hdac_dev(dev)); } /** -- 2.17.1

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[PATCH] ASoC: rt1015: Add condition to prevent SoC providing bclk in ratio of 50 times of sample rate.
by jack.yu＠realtek.com 04 May '20

04 May '20

From: Jack Yu <jack.yu(a)realtek.com> Add condition to prevent SoC providing bclk in ratio of 50 times of sample rate. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 27 +++++++++++++++++++++++++++ sound/soc/codecs/rt1015.h | 1 + 2 files changed, 28 insertions(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index bb310bc7febd..5eb07a430ae3 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -780,6 +780,14 @@ static int rt1015_set_component_pll(struct snd_soc_component *component, freq_out == rt1015->pll_out) return 0; + if (source == RT1015_PLL_S_BCLK) { + if (rt1015->bclk_ratio == 0) { + dev_err(component->dev, + "Can not support bclk ratio as 0.\n"); + return -EINVAL; + } + } + switch (source) { case RT1015_PLL_S_MCLK: snd_soc_component_update_bits(component, RT1015_CLK2, @@ -819,12 +827,30 @@ static int rt1015_set_component_pll(struct snd_soc_component *component, return 0; } +static int rt1015_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s ratio=%d\n", __func__, ratio); + + rt1015->bclk_ratio = ratio; + + if (ratio == 50) { + dev_dbg(component->dev, "Unsupport bclk ratio\n"); + return -EINVAL; + } + + return 0; +} + static int rt1015_probe(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); rt1015->component = component; + rt1015->bclk_ratio = 0; snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, 0x061c); return 0; @@ -844,6 +870,7 @@ static void rt1015_remove(struct snd_soc_component *component) static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, + .set_bclk_ratio = rt1015_set_bclk_ratio, }; static struct snd_soc_dai_driver rt1015_dai[] = { diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index ef3745a4faae..6fbe802082c4 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -362,6 +362,7 @@ struct rt1015_priv { int sysclk_src; int lrck; int bclk; + int bclk_ratio; int id; int pll_src; int pll_in; -- 2.25.1

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