July 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/3] ASoC: topology: Add support for BE DAIs
by mengdong.lin＠linux.intel.com 05 Aug '16

05 Aug '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is topology ABI udpate in this series. Topology core can check size of each ABI objects to detect version mismatch between user space and kernel. This series adds support for BE (Back End) DAIs: - Define the type and ABI struct for Backend DAIs. - The topology core can configure an existing BE DAI or create new BE DAIs. The user space series "topology: Add support for BE DAIs" is also submitted. Is it possible to meet the timeline of v4.8? There will be 2 pairs of kernel & user space series for PCM & BE links later. Mengdong Lin (3): ASoC: topology: ABI - Add the types for BE DAI ASoC: topology: Add support for configuring existing BE DAIs ASoC: topology: Able to create BE DAIs include/sound/soc-topology.h | 1 + include/uapi/sound/asoc.h | 34 +++++++++- sound/soc/soc-core.c | 3 +- sound/soc/soc-topology.c | 156 ++++++++++++++++++++++++++++++++++++++++++- 4 files changed, 189 insertions(+), 5 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH 00/12] ASoC: Intel: Skylake: Add support for library load
by Vinod Koul 05 Aug '16

05 Aug '16

This patch series adds support for library loading. A module maybe part of base firmware or in a different library. A library may have one or more modules. The names of library to be loaded are passed as topology manifest data. The library list is loaded after base FW during dsp init. So we add support library load IPC and the usage in broxton ops for loading libs. To load a list of lib(s) coming from topology, we need to split DSP init, into object creation and fw_init. The fw_init is invoked from soc-probe after topology is initialized. Now fw is initialized later, so module UUID query was move to runtime as the firmware is not available at topology initialization. While at it, also fix the parsing UUIDs only once for SKL and BXT ops and one comment code style. Lastly this series adds support for specifying processing domain for modules. Kranthi G (1): ASoC: Intel: Skylake: add support for tplg manifest load Ramesh Babu (2): ASoC: Intel: Skylake: Add library loading IPCs ASoC: Intel: Skylake: Add library loading support Senthilnathan Veppur (2): ASoC: Intel: Skylake: add additional args to module parsing ASoC: Intel: Skylake: Add module processing domain support Vinod Koul (7): ASoC: Intel: Skylake: Check list empty while getting module info ASoC: Intel: Skylake: Move modules query to runtime ASoC: Intel: Skylake: modify skl_get_dsp_ops() ASoC: Intel: Skylake: split fw and dsp initialization ASoC: Intel: Skylake: Parse UUIDs once ASoC: Intel: Bxt: Parse UUIDs once ASoC: Intel: Skylake: Fix a comment style sound/soc/intel/skylake/bxt-sst.c | 114 ++++++++++++++++++++++++--- sound/soc/intel/skylake/skl-messages.c | 39 +++++---- sound/soc/intel/skylake/skl-pcm.c | 20 +++++ sound/soc/intel/skylake/skl-sst-dsp.h | 7 +- sound/soc/intel/skylake/skl-sst-ipc.c | 29 +++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 12 ++- sound/soc/intel/skylake/skl-sst-utils.c | 14 +++- sound/soc/intel/skylake/skl-sst.c | 43 ++++++---- sound/soc/intel/skylake/skl-topology.c | 55 +++++++++++-- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl-tplg-interface.h | 15 +++- sound/soc/intel/skylake/skl.h | 2 + 12 files changed, 297 insertions(+), 55 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/3] DesignWare HDMI I2S suport
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi Mark, Thierry, Russell These are DesignWare HDMI I2S support patches. It will use ALSA SoC hdmi-codec driver, but we can't use it as-is. So, 2), 3) patches modify hdmi-codec style. Kuninori Morimoto (3): 1) drm: bridge: add DesignWare HDMI I2S audio support 2) ASoC: hdmi-codec: callback function will be called with private data 3) ASoC: hdmi-codec: enable multi probe for same device drivers/gpu/drm/bridge/Kconfig | 8 ++ drivers/gpu/drm/bridge/Makefile | 1 + drivers/gpu/drm/bridge/dw-hdmi-audio.h | 7 ++ drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c | 123 +++++++++++++++++++++++++++++ drivers/gpu/drm/bridge/dw-hdmi.c | 22 +++++- drivers/gpu/drm/bridge/dw-hdmi.h | 21 +++++ include/sound/hdmi-codec.h | 13 +-- sound/soc/codecs/hdmi-codec.c | 81 ++++++++++++++++--- 8 files changed, 259 insertions(+), 17 deletions(-) create mode 100644 drivers/gpu/drm/bridge/dw-hdmi-i2s-audio.c -- 1.9.1

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[alsa-devel] Question about struct snd_soc_dai() :: cpu_dai->codec
by Kuninori Morimoto 05 Aug '16

05 Aug '16

Hi ALSA SoC My current headache is ALSA SoC's each modules (= Card/Codec/CPU/Platform) doesn't care about "unbind/rmmod". For example, if someone unbinded/rmmoded "Codec", Card or other modules doesn't know about it. Thus, user can continue to use this sound card, and kernel will be Oops. So, I would like to solve this issue, and for this purpose, now I'm investigating about ALSA SoC structures. It is now super complex, and I noticed that it is having duplicated struct members. If possible, I would like to cleanup this issue. During this investigating, I noticed 1 strange operation ${LINUX}/sound/soc/soc-core.c :: snd_soc_runtime_set_dai_fmt int snd_soc_runtime_set_dai_fmt(xxx) { ... /* Flip the polarity for the "CPU" end of a CODEC<->CODEC link */ if (cpu_dai->codec) { ... } ... } struct snd_soc_dai { ... /* parent platform/codec */ struct snd_soc_codec *codec; struct snd_soc_component *component; ... } According to snd_soc_dai, *codec is its "parent". I wonder does "cpu_dai" really can have "codec" parent ?? I think this is not correct, and we can remove this operation ?

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[alsa-devel] [PATCH] sound: Fix empty menuconfig SND_ARM
by Eugeniu Rosca 03 Aug '16

03 Aug '16

commit 3c8f7710c1c4 ("ASoC: fix broken pxa SoC support") has removed the SND_ARM dependency from SND_PXA2XX_LIB and SND_PXA2XX_LIB_AC97, by moving these config entries outside of the "if SND_ARM ... endif" construct. However, by placing these 2 symbols right between the SND_ARM menuconfig definition and the first SND_ARM menu entry, the side effect is that the SND_ARM menu becomes empty and all the config entries caught between "if SND_ARM ... endif" no more belong to menuconfig SND_ARM, but to its parent (menuconfig SND). Fix this. Signed-off-by: Eugeniu Rosca <rosca.eugeniu(a)gmail.com> --- sound/arm/Kconfig | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index e040621..65171f6 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -9,14 +9,6 @@ menuconfig SND_ARM Drivers that are implemented on ASoC can be found in "ALSA for SoC audio support" section. -config SND_PXA2XX_LIB - tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 - select SND_DMAENGINE_PCM - -config SND_PXA2XX_LIB_AC97 - bool - if SND_ARM config SND_ARMAACI @@ -42,3 +34,10 @@ config SND_PXA2XX_AC97 endif # SND_ARM +config SND_PXA2XX_LIB + tristate + select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 + select SND_DMAENGINE_PCM + +config SND_PXA2XX_LIB_AC97 + bool -- 2.9.2

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[alsa-devel] Improving status timestamp accuracy
by Alan Young 02 Aug '16

02 Aug '16

I am looking at ways to get more accurate status timestamp information for various SoC drivers. The data that is obtained by snd_pcm_status(). One route would be to implement the more accurate timestamp mechanisms that currently are only available for HDA and Skylake (which I think is the SoC version of HDA). Looking at the code however, I think that may be unnecessary, at least for my purposes. It may not actually be practical in many cases. A call to snd_pcm_status() result in snd_pcm_update_hw_ptr0() being called. This gets the current output position (pos) via substream->ops->pointer(substream) and then makes all the other calculations based on the result. In theory, the result of substream->ops->pointer() could be sample accurate but in practice it is very unlikely to be better than period accurate. In fact, if I read it right, it will just about always be accurate to the point of the last period interrupt. Even when a DMA driver claims support of DMA_RESIDUE_GRANULARITY_BURST, it is often the case that the actual granularity is a period. The consequence of all that is that, for most drivers, the accuracy of a status report is period time. The result values, tstamp & audio_tstamp, are calculated using the current time and the pos estimate from above. snd_pcm_update_hw_ptr0() is also called when there is a DMA interrupt. At that time the calculate results will be accurate, or at worst consistently inaccurate (there could be a constant offset). It would be useful if a snd_pcm_status() call would simply return the results from the point of the last interrupt, and not try to estimate a current value based on the inaccurate substream->ops->pointer() result. It could either: (a) return the result from the time of the last interrupt, in which case tstamp would be in the past and driver_tstamp would be now; or (b) update audio_tstamp based on the elapsed time since it was recorded. (b) effectively abandons the idea that a current position report will be accurate outside of an interrupt callback but, even if it is, doing so is unlikely to result in any loss of accuracy in practice (assuming a drift of better than 40ppm and period time < 100ms). Any comments on either of these approaches? I guess (b) is more compatible with the current model. Alan.

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[alsa-devel] [PATCH] ASoC: cs53l30: Do not ignore errors if mclk is specified
by Nicolin Chen 01 Aug '16

01 Aug '16

When the clock is specified, there could be other errors besides the EPROBE_DEFER so don't ignore them. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 2c0d9c4..227c556 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -999,8 +999,8 @@ static int cs53l30_i2c_probe(struct i2c_client *client, /* Check if MCLK provided */ cs53l30->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs53l30->mclk)) { - if (PTR_ERR(cs53l30->mclk) == -EPROBE_DEFER) { - ret = -EPROBE_DEFER; + if (PTR_ERR(cs53l30->mclk) != -ENOENT) { + ret = PTR_ERR(cs53l30->mclk); goto error; } /* Otherwise mark the mclk pointer to NULL */ -- 2.1.4

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[alsa-devel] [PATCH] ASoC: da7219: Make more efficient use of MCLK within driver
by Adam Thomson 01 Aug '16

01 Aug '16

Currently, if the driver has control of MCLK then it remains enabled as long as the codec is in STANDBY or above. The MCLK is only really required in STANDBY when a 3-pole jack is inserted and the HP detect procedure is required to run. This patch updates the code to enable/disable the MCLK when moving between the STANDBY and PREPARE bias level, and when a 3-pole jack is inserted and HP detection is required, thus saving power at all other times. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219-aad.c | 18 +++++++++++++++++- sound/soc/codecs/da7219.c | 18 +++++++++++------- 2 files changed, 28 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index f0057cd..4e369a1 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -13,6 +13,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/clk.h> #include <linux/i2c.h> #include <linux/property.h> #include <linux/pm_wakeirq.h> @@ -115,12 +116,23 @@ static void da7219_aad_hptest_work(struct work_struct *work) u16 tonegen_freq_hptest; u8 accdet_cfg8; - int report = 0; + int report = 0, ret = 0; /* Lock DAPM and any Kcontrols that are affected by this test */ snd_soc_dapm_mutex_lock(dapm); mutex_lock(&da7219->lock); + /* Ensure MCLK is available for HP test procedure */ + if (da7219->mclk) { + ret = clk_prepare_enable(da7219->mclk); + if (ret) { + dev_err(codec->dev, "Failed to enable mclk - %d\n", ret); + mutex_unlock(&da7219->lock); + snd_soc_dapm_mutex_unlock(dapm); + return; + } + } + /* Bypass cache so it saves current settings */ regcache_cache_bypass(da7219->regmap, true); @@ -250,6 +262,10 @@ static void da7219_aad_hptest_work(struct work_struct *work) snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK, DA7219_HP_R_AMP_OE_MASK); + /* Remove MCLK, if previously enabled */ + if (da7219->mclk) + clk_disable_unprepare(da7219->mclk); + mutex_unlock(&da7219->lock); snd_soc_dapm_mutex_unlock(dapm); diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 50ea943..737e914 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1508,11 +1508,10 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: break; - case SND_SOC_BIAS_STANDBY: - if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { - /* MCLK */ + case SND_SOC_BIAS_PREPARE: + /* Enable MCLK for transition to ON state */ + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_STANDBY) { if (da7219->mclk) { ret = clk_prepare_enable(da7219->mclk); if (ret) { @@ -1521,11 +1520,19 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, return ret; } } + } + break; + case SND_SOC_BIAS_STANDBY: + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Master bias */ snd_soc_update_bits(codec, DA7219_REFERENCES, DA7219_BIAS_EN_MASK, DA7219_BIAS_EN_MASK); + } else { + /* Remove MCLK */ + if (da7219->mclk) + clk_disable_unprepare(da7219->mclk); } break; case SND_SOC_BIAS_OFF: @@ -1534,9 +1541,6 @@ static int da7219_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, DA7219_REFERENCES, DA7219_BIAS_EN_MASK, 0); - /* MCLK */ - if (da7219->mclk) - clk_disable_unprepare(da7219->mclk); break; } -- 1.9.3

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[alsa-devel] [PATCH v2] ASoC: rt5659: Add mclk controls
by Nicolin Chen 01 Aug '16

01 Aug '16

The codec driver should control the mclk. So this patch adds this support. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changlog: v1->v2 * Only set the mclk pointer to NULL when getting an ENOENT error code (i.e. no clock is specified) -- corrected by Lars-Peter Documentation/devicetree/bindings/sound/rt5659.txt | 3 +++ sound/soc/codecs/rt5659.c | 24 ++++++++++++++++++++++ sound/soc/codecs/rt5659.h | 1 + 3 files changed, 28 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/rt5659.txt b/Documentation/devicetree/bindings/sound/rt5659.txt index 5f79e7f..1766e05 100644 --- a/Documentation/devicetree/bindings/sound/rt5659.txt +++ b/Documentation/devicetree/bindings/sound/rt5659.txt @@ -12,6 +12,9 @@ Required properties: Optional properties: +- clocks: The phandle of the master clock to the CODEC +- clock-names: Should be "mclk" + - realtek,in1-differential - realtek,in3-differential - realtek,in4-differential diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 1b30914..42c183f 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -9,6 +9,7 @@ * published by the Free Software Foundation. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -3565,7 +3566,9 @@ static int rt5659_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) static int rt5659_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); struct rt5659_priv *rt5659 = snd_soc_codec_get_drvdata(codec); + int ret; switch (level) { case SND_SOC_BIAS_PREPARE: @@ -3582,6 +3585,17 @@ static int rt5659_set_bias_level(struct snd_soc_codec *codec, RT5659_PWR_FV1 | RT5659_PWR_FV2); break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + ret = clk_prepare_enable(rt5659->mclk); + if (ret) { + dev_err(codec->dev, + "failed to enable MCLK: %d\n", ret); + return ret; + } + } + break; + case SND_SOC_BIAS_OFF: regmap_update_bits(rt5659->regmap, RT5659_PWR_DIG_1, RT5659_PWR_LDO, 0); @@ -3591,6 +3605,7 @@ static int rt5659_set_bias_level(struct snd_soc_codec *codec, RT5659_PWR_MB | RT5659_PWR_VREF2); regmap_update_bits(rt5659->regmap, RT5659_DIG_MISC, RT5659_DIG_GATE_CTRL, 0); + clk_disable_unprepare(rt5659->mclk); break; default: @@ -4020,6 +4035,15 @@ static int rt5659_i2c_probe(struct i2c_client *i2c, regmap_write(rt5659->regmap, RT5659_RESET, 0); + /* Check if MCLK provided */ + rt5659->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(rt5659->mclk)) { + if (PTR_ERR(rt5659->mclk) != -ENOENT) + return PTR_ERR(rt5659->mclk); + /* Otherwise mark the mclk pointer to NULL */ + rt5659->mclk = NULL; + } + rt5659_calibrate(rt5659); /* line in diff mode*/ diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index d31c9e5..d69b0eb 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -1796,6 +1796,7 @@ struct rt5659_priv { struct gpio_desc *gpiod_reset; struct snd_soc_jack *hs_jack; struct delayed_work jack_detect_work; + struct clk *mclk; int sysclk; int sysclk_src; -- 2.1.4

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Add clocking option
by Stefan Müller-Klieser 01 Aug '16

01 Aug '16

The AHCLKX clock output of the McASP can be used as audio MCLK, even if the McASP runs in slave mode. Add a DT binding to make this option available in a simple-card setup. Signed-off-by: Stefan Müller-Klieser <s.mueller-klieser(a)phytec.de> --- Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt | 4 ++++ include/linux/platform_data/davinci_asp.h | 5 +++++ sound/soc/davinci/davinci-mcasp.c | 8 ++++++++ 3 files changed, 17 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt index 46bc982..96d6244 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt @@ -39,6 +39,10 @@ Optional properties: please refer to pinctrl-bindings.txt - fck_parent : Should contain a valid clock name which will be used as parent for the McASP fck +- ahclkx-en: In case the MCASP runs in slave mode, e.g. i2s SCK and WS slave, + we can still provide the MCLK to a codec PLL using the AHCLKX + transmit clock output. This setup can be activated by using this + property. Clock master mode remains unaffected. Example: diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h index 85ad68f..c878836 100644 --- a/include/linux/platform_data/davinci_asp.h +++ b/include/linux/platform_data/davinci_asp.h @@ -44,6 +44,11 @@ struct davinci_mcasp_pdata { int clk_input_pin; /* + * Activate AHCLKX clock output in slave mode. + */ + bool ahclkx_en; + + /* * This flag works when both clock and FS are outputs for the cpu * and makes clock more accurate (FS is not symmetrical and the * clock is very fast. diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 237dc67..aba888e 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -94,6 +94,7 @@ struct davinci_mcasp { int sysclk_freq; bool bclk_master; + bool ahclkx_en; /* McASP FIFO related */ u8 txnumevt; @@ -605,6 +606,10 @@ static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id, mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); + } else if (mcasp->ahclkx_en) { + mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); + mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, AHCLKX); } else { mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE); @@ -1610,6 +1615,8 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( if (ret >= 0) pdata->op_mode = val; + pdata->ahclkx_en = of_property_read_bool(np, "ahclkx-en"); + ret = of_property_read_u32(np, "tdm-slots", &val); if (ret >= 0) { if (val < 2 || val > 32) { @@ -1851,6 +1858,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->version = pdata->version; mcasp->txnumevt = pdata->txnumevt; mcasp->rxnumevt = pdata->rxnumevt; + mcasp->ahclkx_en = pdata->ahclkx_en; mcasp->dev = &pdev->dev; -- 1.9.1

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