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

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

5 23

[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

2 1

[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

3 3

[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

2 1

[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

3 4

[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

2 1

[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

2 3

[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

2 1

[PATCH] ASoC: qcom: Use the defined variable to simplify code
by Tang Bin 04 May '20

04 May '20

Use the defined variable "dev" to make the code cleaner. Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/qcom/lpass-apq8016.c | 9 ++++---- sound/soc/qcom/lpass-cpu.c | 39 ++++++++++++++++------------------ 2 files changed, 22 insertions(+), 26 deletions(-) diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 6575da549..b3610d05b 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -166,28 +166,27 @@ static int apq8016_lpass_init(struct platform_device *pdev) drvdata->pcnoc_mport_clk = devm_clk_get(dev, "pcnoc-mport-clk"); if (IS_ERR(drvdata->pcnoc_mport_clk)) { - dev_err(&pdev->dev, "error getting pcnoc-mport-clk: %ld\n", + dev_err(dev, "error getting pcnoc-mport-clk: %ld\n", PTR_ERR(drvdata->pcnoc_mport_clk)); return PTR_ERR(drvdata->pcnoc_mport_clk); } ret = clk_prepare_enable(drvdata->pcnoc_mport_clk); if (ret) { - dev_err(&pdev->dev, "Error enabling pcnoc-mport-clk: %d\n", - ret); + dev_err(dev, "Error enabling pcnoc-mport-clk: %d\n", ret); return ret; } drvdata->pcnoc_sway_clk = devm_clk_get(dev, "pcnoc-sway-clk"); if (IS_ERR(drvdata->pcnoc_sway_clk)) { - dev_err(&pdev->dev, "error getting pcnoc-sway-clk: %ld\n", + dev_err(dev, "error getting pcnoc-sway-clk: %ld\n", PTR_ERR(drvdata->pcnoc_sway_clk)); return PTR_ERR(drvdata->pcnoc_sway_clk); } ret = clk_prepare_enable(drvdata->pcnoc_sway_clk); if (ret) { - dev_err(&pdev->dev, "Error enabling pcnoc_sway_clk: %d\n", ret); + dev_err(dev, "Error enabling pcnoc_sway_clk: %d\n", ret); return ret; } diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index dbce7e92b..c6d26395f 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -425,12 +425,11 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) dsp_of_node = of_parse_phandle(pdev->dev.of_node, "qcom,adsp", 0); if (dsp_of_node) { - dev_err(&pdev->dev, "DSP exists and holds audio resources\n"); + dev_err(dev, "DSP exists and holds audio resources\n"); return -EBUSY; } - drvdata = devm_kzalloc(&pdev->dev, sizeof(struct lpass_data), - GFP_KERNEL); + drvdata = devm_kzalloc(dev, sizeof(struct lpass_data), GFP_KERNEL); if (!drvdata) return -ENOMEM; platform_set_drvdata(pdev, drvdata); @@ -444,9 +443,9 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "lpass-lpaif"); - drvdata->lpaif = devm_ioremap_resource(&pdev->dev, res); + drvdata->lpaif = devm_ioremap_resource(dev, res); if (IS_ERR((void const __force *)drvdata->lpaif)) { - dev_err(&pdev->dev, "error mapping reg resource: %ld\n", + dev_err(dev, "error mapping reg resource: %ld\n", PTR_ERR((void const __force *)drvdata->lpaif)); return PTR_ERR((void const __force *)drvdata->lpaif); } @@ -455,10 +454,10 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) variant->wrdma_channels + variant->wrdma_channel_start); - drvdata->lpaif_map = devm_regmap_init_mmio(&pdev->dev, drvdata->lpaif, + drvdata->lpaif_map = devm_regmap_init_mmio(dev, drvdata->lpaif, &lpass_cpu_regmap_config); if (IS_ERR(drvdata->lpaif_map)) { - dev_err(&pdev->dev, "error initializing regmap: %ld\n", + dev_err(dev, "error initializing regmap: %ld\n", PTR_ERR(drvdata->lpaif_map)); return PTR_ERR(drvdata->lpaif_map); } @@ -468,10 +467,10 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) for (i = 0; i < variant->num_dai; i++) { dai_id = variant->dai_driver[i].id; - drvdata->mi2s_osr_clk[dai_id] = devm_clk_get(&pdev->dev, + drvdata->mi2s_osr_clk[dai_id] = devm_clk_get(dev, variant->dai_osr_clk_names[i]); if (IS_ERR(drvdata->mi2s_osr_clk[dai_id])) { - dev_warn(&pdev->dev, + dev_warn(dev, "%s() error getting optional %s: %ld\n", __func__, variant->dai_osr_clk_names[i], @@ -480,10 +479,10 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->mi2s_osr_clk[dai_id] = NULL; } - drvdata->mi2s_bit_clk[dai_id] = devm_clk_get(&pdev->dev, + drvdata->mi2s_bit_clk[dai_id] = devm_clk_get(dev, variant->dai_bit_clk_names[i]); if (IS_ERR(drvdata->mi2s_bit_clk[dai_id])) { - dev_err(&pdev->dev, + dev_err(dev, "error getting %s: %ld\n", variant->dai_bit_clk_names[i], PTR_ERR(drvdata->mi2s_bit_clk[dai_id])); @@ -491,41 +490,39 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) } } - drvdata->ahbix_clk = devm_clk_get(&pdev->dev, "ahbix-clk"); + drvdata->ahbix_clk = devm_clk_get(dev, "ahbix-clk"); if (IS_ERR(drvdata->ahbix_clk)) { - dev_err(&pdev->dev, "error getting ahbix-clk: %ld\n", + dev_err(dev, "error getting ahbix-clk: %ld\n", PTR_ERR(drvdata->ahbix_clk)); return PTR_ERR(drvdata->ahbix_clk); } ret = clk_set_rate(drvdata->ahbix_clk, LPASS_AHBIX_CLOCK_FREQUENCY); if (ret) { - dev_err(&pdev->dev, "error setting rate on ahbix_clk: %d\n", - ret); + dev_err(dev, "error setting rate on ahbix_clk: %d\n", ret); return ret; } - dev_dbg(&pdev->dev, "set ahbix_clk rate to %lu\n", + dev_dbg(dev, "set ahbix_clk rate to %lu\n", clk_get_rate(drvdata->ahbix_clk)); ret = clk_prepare_enable(drvdata->ahbix_clk); if (ret) { - dev_err(&pdev->dev, "error enabling ahbix_clk: %d\n", ret); + dev_err(dev, "error enabling ahbix_clk: %d\n", ret); return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, + ret = devm_snd_soc_register_component(dev, &lpass_cpu_comp_driver, variant->dai_driver, variant->num_dai); if (ret) { - dev_err(&pdev->dev, "error registering cpu driver: %d\n", ret); + dev_err(dev, "error registering cpu driver: %d\n", ret); goto err_clk; } ret = asoc_qcom_lpass_platform_register(pdev); if (ret) { - dev_err(&pdev->dev, "error registering platform driver: %d\n", - ret); + dev_err(dev, "error registering platform driver: %d\n", ret); goto err_clk; } -- 2.20.1.windows.1

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[PATCH] Mark the ADAU7118 reset register as volatile.
by chris.d.garren＠gmail.com 04 May '20

04 May '20

From: Chris Garren <cdgarren(a)indesign-llc.com> Without this the previously written value was written to this reg, which caused the different configuration registers to be reset. Signed-off-by: Chris Garren <cdgarren(a)indesign-llc.com> --- sound/soc/codecs/adau7118-i2c.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c index a821136..aa7afb3 100644 --- a/sound/soc/codecs/adau7118-i2c.c +++ b/sound/soc/codecs/adau7118-i2c.c @@ -32,6 +32,12 @@ static const struct reg_default adau7118_reg_defaults[] = { { ADAU7118_REG_RESET, 0x00 }, }; +static bool adau7118_volatile(struct device *dev, unsigned int reg) +{ + return (reg == ADAU7118_REG_RESET); +} + + static const struct regmap_config adau7118_regmap_config = { .reg_bits = 8, .val_bits = 8, @@ -39,6 +45,7 @@ static const struct regmap_config adau7118_regmap_config = { .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), .cache_type = REGCACHE_RBTREE, .max_register = ADAU7118_REG_RESET, + .volatile_reg = adau7118_volatile, }; static int adau7118_probe_i2c(struct i2c_client *i2c, -- 2.7.4

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