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[alsa-devel] [PATCH] ASoC: sof: include types.h at header.h
by Kuninori Morimoto 16 Oct '19

16 Oct '19

Content-Transfer-Encoding: 8bit Without <types.h> we will get these error linux/include/sound/sof/header.h:125:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:136:2: error: unknown type name ‘uint32_t’uint32_t size; linux/include/sound/sof/header.h:137:2: error: unknown type name ‘uint32_t’uint32_t cmd; ... linux/include/sound/sof/dai-imx.h:18:2: error: unknown type name ‘uint16_t’uint16_t reserved1; linux/include/sound/sof/dai-imx.h:30:2: error: unknown type name ‘uint16_t’uint16_t tdm_slot_width; linux/include/sound/sof/dai-imx.h:31:2: error: unknown type name ‘uint16_t’uint16_t reserved2; Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/sof/header.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index 10f00c0..332143f 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -9,6 +9,7 @@ #ifndef __INCLUDE_SOUND_SOF_HEADER_H__ #define __INCLUDE_SOUND_SOF_HEADER_H__ +#include <linux/types.h> #include <uapi/sound/sof/abi.h> /** \addtogroup sof_uapi uAPI -- 2.7.4

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Re: [alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by Mark Brown 16 Oct '19

16 Oct '19

On Wed, Oct 16, 2019 at 11:28:47AM +0000, Shuming [范書銘] wrote: > > These are new DT bindings and need to be added to the binding document. > Sure, could I send another patch for it? Yes, the DT bindings should go in a separate patch anyway.

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Re: [alsa-devel] [PATCH] ASoC: rt1011: Read and apply r0 and temperature device property
by Cheng-yi Chiang 16 Oct '19

16 Oct '19

On Wed, Oct 16, 2019 at 7:29 PM Shuming [范書銘] <shumingf(a)realtek.com> wrote: > > > On Wed, Oct 16, 2019 at 04:58:45PM +0800, shumingf(a)realtek.com wrote: > > > > > +static int rt1011_parse_dp(struct rt1011_priv *rt1011, struct device > > > +*dev) { > > > + device_property_read_u32(dev, "realtek,temperature_calib", > > > + &rt1011->temperature_calib); > > > + device_property_read_u32(dev, "realtek,r0_calib", > > > + &rt1011->r0_calib); > > > > These are new DT bindings and need to be added to the binding document. > > Sure, could I send another patch for it? > > > ------Please consider the environment before printing this e-mail. Hi Shuming, Yes, you will need a patch before this one to add the binding documentation. Thanks!

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[alsa-devel] [PATCH 0/2] soundwire: Kconfig/build improvements
by Pierre-Louis Bossart 16 Oct '19

16 Oct '19

The Intel kbuild test reported issues with COMPILE_TEST or cross-compilation when SOF is enabled, fix. This has no functional impact. Pierre-Louis Bossart (2): soundwire: intel: add missing headers for cross-compilation soundwire: intel: remove X86 dependency drivers/soundwire/Kconfig | 2 +- drivers/soundwire/intel.c | 1 + drivers/soundwire/intel_init.c | 1 + 3 files changed, 3 insertions(+), 1 deletion(-) -- 2.20.1

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[alsa-devel] [PATCH 0/2] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 16 Oct '19

16 Oct '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the for-next. v3 changes: 1. https://patchwork.kernel.org/patch/11164283/ and https://patchwork.kernel.org/patch/11164305/ has been merged. 2. change the name of mtk_wdt_compatible to mtk_wdt_data 3. remove toprgu_reset struct and use mtk_wdt_dev instead 4. get the value of sw_rst_num from .h file 5. adddd mt2712-resets.h for mt2712 6. improve commit message v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on 3. patchwork list: https://patchwork.kernel.org/cover/11164285/ https://patchwork.kernel.org/patch/11164295/ https://patchwork.kernel.org/patch/11164299/ https://patchwork.kernel.org/patch/11164283/ https://patchwork.kernel.org/patch/11164305/ v1 changes: 1. patchwork list: https://patchwork.kernel.org/cover/11164173/ https://patchwork.kernel.org/patch/11164181/ https://patchwork.kernel.org/patch/11164185/ https://patchwork.kernel.org/patch/11164187/ https://patchwork.kernel.org/patch/11164175/ yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../devicetree/bindings/watchdog/mtk-wdt.txt | 10 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 111 +++++++++++++++++- .../reset-controller/mt2712-resets.h | 22 ++++ .../reset-controller/mt8183-resets.h | 15 +++ 5 files changed, 155 insertions(+), 4 deletions(-) create mode 100644 include/dt-bindings/reset-controller/mt2712-resets.h -- 2.18.0

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[alsa-devel] [PATCH] ASoC: pcm3168a: Use fixup instead of constraint for channels and formats
by Peter Ujfalusi 16 Oct '19

16 Oct '19

The snd_pcm_hw_constraint_minmax() works fine when a single codec is connected to a single CPU DAI, but in multicodec or DPCM setup the constraints placed by the driver will apply to the whole PCM stream (FE included) and thus prevents more than 8 playback channels for example. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, I have a board which uses two pcm3168a codec connected to one McASP port with parallel serializers. One 16 channel (2x 8 channel) playback and one 12 (2x 6 channel) capture should be possible as I use only one CPU DAI + platform. However no matter how I tried to set things up ALSA blocked more than 8 channel playback in simple multicodec or DPCM setup. It turned out that the constraint set by the codec on the BE is actually applies to the FE as well as they both use the same runtime. The codec still have different constraints depending on the interface format, so I replaced the constraint with a fixup function to make sure that the codec will not advertise non supported formats. Is this a know feature? Another thing which I have came across is the lack of support for parallel data pins in ASoC core. For example the soc-pcm only checks the tdm slots to fix up the channels supported by the codec, but when we have multiple pins in use and they are operate in parallel, then the number of slots must be multiplied by the data pins in use. I have wip patches for this, adding int snd_soc_dai_set_data_pins(struct snd_soc_dai *dai, unsigned int tx_data_pins, unsigned int rx_data_pins); and then using the information in soc-pcm, but need to check further to see if there are other places this might be needed. Regards, Peter sound/soc/codecs/pcm3168a.c | 98 +++++++++++++++---------------------- 1 file changed, 40 insertions(+), 58 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 88b75695fbf7..708dac27feff 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -62,6 +62,7 @@ struct pcm3168a_priv { unsigned long sysclk; struct pcm3168a_io_params io_params[2]; + struct snd_soc_dai_driver dai_drv[2]; }; static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" }; @@ -314,6 +315,37 @@ static int pcm3168a_set_dai_sysclk(struct snd_soc_dai *dai, return 0; } +static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; + unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; + + if (!pcm3168a->io_params[dai->id].fmt) + return; + + if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { + /* S16_LE is only supported in RIGHT_J mode */ + formats |= SNDRV_PCM_FMTBIT_S16_LE; + + /* + * If multi DIN/DOUT is not selected, RIGHT_J can only support + * two channels (no TDM support) + */ + if (pcm3168a->io_params[dai->id].tdm_slots != 2) + channel_max = 2; + } + + if (dai->id == PCM3168A_DAI_DAC) { + dai->driver->playback.channels_max = channel_max; + dai->driver->playback.formats = formats; + } else { + dai->driver->capture.channels_max = channel_max; + dai->driver->capture.formats = formats; + } +} + static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) { struct snd_soc_component *component = dai->component; @@ -376,6 +408,8 @@ static int pcm3168a_set_dai_fmt(struct snd_soc_dai *dai, unsigned int format) regmap_update_bits(pcm3168a->regmap, reg, mask, fmt << shift); + pcm3168a_update_fixup_pcm_stream(dai); + return 0; } @@ -409,6 +443,8 @@ static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, else io_params->tdm_mask = rx_mask; + pcm3168a_update_fixup_pcm_stream(dai); + return 0; } @@ -530,63 +566,7 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return 0; } -static int pcm3168a_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_component *component = dai->component; - struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); - unsigned int sample_min; - unsigned int channel_max; - unsigned int channel_maxs[] = { - 8, /* DAC */ - 6 /* ADC */ - }; - - /* - * Available Data Bits - * - * RIGHT_J : 24 / 16 - * LEFT_J : 24 - * I2S : 24 - * - * TDM available - * - * I2S - * LEFT_J - */ - switch (pcm3168a->io_params[dai->id].fmt) { - case PCM3168A_FMT_RIGHT_J: - sample_min = 16; - channel_max = 2; - break; - case PCM3168A_FMT_LEFT_J: - case PCM3168A_FMT_I2S: - case PCM3168A_FMT_DSP_A: - case PCM3168A_FMT_DSP_B: - sample_min = 24; - channel_max = channel_maxs[dai->id]; - break; - default: - sample_min = 24; - channel_max = 2; - } - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - sample_min, 32); - - /* Allow all channels in multi DIN/DOUT mode */ - if (pcm3168a->io_params[dai->id].tdm_slots == 2) - channel_max = channel_maxs[dai->id]; - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_CHANNELS, - 2, channel_max); - - return 0; -} static const struct snd_soc_dai_ops pcm3168a_dai_ops = { - .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, @@ -776,8 +756,10 @@ int pcm3168a_probe(struct device *dev, struct regmap *regmap) pm_runtime_enable(dev); pm_runtime_idle(dev); - ret = devm_snd_soc_register_component(dev, &pcm3168a_driver, pcm3168a_dais, - ARRAY_SIZE(pcm3168a_dais)); + memcpy(pcm3168a->dai_drv, pcm3168a_dais, sizeof(pcm3168a->dai_drv)); + ret = devm_snd_soc_register_component(dev, &pcm3168a_driver, + pcm3168a->dai_drv, + ARRAY_SIZE(pcm3168a->dai_drv)); if (ret) { dev_err(dev, "failed to register component: %d\n", ret); goto err_regulator; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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Re: [alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio interface (Re: Alsa-devel Digest, Vol 152, Issue 150)
by Hin-Tak Leung 15 Oct '19

15 Oct '19

> Date: Tue, 15 Oct 2019 00:10:28 +1030 > From: "Geoffrey D. Bennett" <g(a)b4.vu> > To: wwp <subscript(a)free.fr> > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio > interface > Message-ID: <20191014134028.GA11018(a)b4.vu> > Content-Type: text/plain; charset=us-ascii > Hi wwp, > On Sun, Oct 13, 2019 at 10:20:30AM +0200, wwp wrote: > > Hello, > > > > I've recently bought a Focusrite Scarlett 18i8 3rd Gen audio interface, > > great device, but in CentOS 7, with either the available 3.x or 4.x > > kernels, > ... > > Of course, adding the 0x1235:8214 USB ID to the quirks just doesn't > > work. I presume there's more work like the patch submitted here (I saw > > the archives) in the "[PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 > > mixer interface" mail thread? Have you tried treating it as a 18i8 2nd Gen device first, at least? FWIW, I had adapted Geoffrey's work as a DKMS module: https://github.com/HinTak/sound-usb-dkms But you'll still need to be running as recent as a 5.2.x release kernel to use it. It just makes it easier to play with the code without recompiling the whole kernel. HTH. 3.x and 4.x is definitely way too old!

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[alsa-devel] Focusrite Scarlett 18i8 3rd Gen audio interface
by wwp 15 Oct '19

15 Oct '19

Hello, I've recently bought a Focusrite Scarlett 18i8 3rd Gen audio interface, great device, but in CentOS 7, with either the available 3.x or 4.x kernels, the device just appears in the available Alsa devices, but no mixer is available, the device direct hardware monitor output is also not working (it is, by default, and works well in Windows, but I presume that w/ no mixer support in Alsa, it's expected that nothing works). I'm able to bound I/O ports in QJackCtl but nothing goes out of the device. Of course, adding the 0x1235:8214 USB ID to the quirks just doesn't work. I presume there's more work like the patch submitted here (I saw the archives) in the "[PATCH RFC V2] ALSA: usb-audio: Scarlett Gen 2 mixer interface" mail thread? Do you it is worth trying applying that 2nd Gen patch to one of the v4 kernel available for CentOS 7, adding the 3rd Gen quirks and see how it behaves? Or is there already progress on supporting the 3rd Gen one too? Regards, -- wwp https://useplaintext.email/

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[alsa-devel] [PATCH] soundwire: slave: Fix unused function warning on !ACPI
by Michal Suchanek 15 Oct '19

15 Oct '19

Fixes the following warning on !ACPI systems: drivers/soundwire/slave.c:16:12: warning: ‘sdw_slave_add’ defined but not used [-Wunused-function] static int sdw_slave_add(struct sdw_bus *bus, ^~~~~~~~~~~~~ Signed-off-by: Michal Suchanek <msuchanek(a)suse.de> --- drivers/soundwire/slave.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/slave.c b/drivers/soundwire/slave.c index f39a5815e25d..34c7e65831d1 100644 --- a/drivers/soundwire/slave.c +++ b/drivers/soundwire/slave.c @@ -6,6 +6,7 @@ #include <linux/soundwire/sdw_type.h> #include "bus.h" +#if IS_ENABLED(CONFIG_ACPI) static void sdw_slave_release(struct device *dev) { struct sdw_slave *slave = dev_to_sdw_dev(dev); @@ -60,7 +61,6 @@ static int sdw_slave_add(struct sdw_bus *bus, return ret; } -#if IS_ENABLED(CONFIG_ACPI) /* * sdw_acpi_find_slaves() - Find Slave devices in Master ACPI node * @bus: SDW bus instance -- 2.22.0

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[alsa-devel] [PATCH] soundwire: qcom: fix devm_platform_ioremap_resource.cocci warnings
by Julia Lawall 15 Oct '19

15 Oct '19

From: kbuild test robot <lkp(a)intel.com> Use devm_platform_ioremap_resource helper which wraps platform_get_resource() and devm_ioremap_resource() together. Generated by: scripts/coccinelle/api/devm_platform_ioremap_resource.cocci Fixes: 4a789cff1848 ("soundwire: qcom: add support for SoundWire controller") CC: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Signed-off-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- url: https://github.com/0day-ci/linux/commits/Srinivas-Kandagatla/soundwire-Add-… :::::: branch date: 9 hours ago :::::: commit date: 9 hours ago qcom.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -784,7 +784,6 @@ static int qcom_swrm_probe(struct platfo struct sdw_master_prop *prop; struct sdw_bus_params *params; struct qcom_swrm_ctrl *ctrl; - struct resource *res; int ret; u32 val; @@ -801,8 +800,7 @@ static int qcom_swrm_probe(struct platfo } else { ctrl->reg_read = qcom_swrm_mmio_reg_read; ctrl->reg_write = qcom_swrm_mmio_reg_write; - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - ctrl->base = devm_ioremap_resource(dev, res); + ctrl->base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(ctrl->base)) return PTR_ERR(ctrl->base); }

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[alsa-devel] [PATCH v2] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 15 Oct '19

15 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 94 ++++++++++++++++++++++ 2 files changed, 95 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..5e0e7db 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -320,6 +320,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH select SND_SOC_RT5514_SPI select SND_SOC_MAX98927 select SND_SOC_HDAC_HDMI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for RT5663 + RT5514 + MAX98927. diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..dc09a85 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,61 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +139,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +156,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +708,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +724,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

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[alsa-devel] [PATCH] ASoC: simple-card: fixup simple_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index fc9c753..10b82bf 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -149,7 +149,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -179,7 +179,7 @@ static int simple_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: audio-graph: fixup graph_dai_link_of_dpcm() comment
by Kuninori Morimoto 15 Oct '19

15 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> The comment is wrong. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 6007e63..9ad35d9 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -232,7 +232,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, if (li->cpu) { int is_single_links = 0; - /* BE is dummy */ + /* Codec is dummy */ codecs->of_node = NULL; codecs->dai_name = "snd-soc-dummy-dai"; codecs->name = "snd-soc-dummy"; @@ -263,7 +263,7 @@ static int graph_dai_link_of_dpcm(struct asoc_simple_priv *priv, } else { struct snd_soc_codec_conf *cconf; - /* FE is dummy */ + /* CPU is dummy */ cpus->of_node = NULL; cpus->dai_name = "snd-soc-dummy-dai"; cpus->name = "snd-soc-dummy"; -- 2.7.4

3 2

[alsa-devel] [PATCH v2] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Remove one extra blank line also Regards, Peter sound/soc/codecs/pcm3168a.c | 3 --- 1 file changed, 3 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..313500ab36df 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,9 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; - if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ formats |= SNDRV_PCM_FMTBIT_S16_LE; -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: pcm3168a: Fix serial mode dependent format support
by Peter Ujfalusi 15 Oct '19

15 Oct '19

fmt 0 is perfectly valid (PCM3168A_FMT_I2S). Remove the return in case fmt == 0. Fixes: ("ASoC: pcm3168a: Use fixup instead of constraint for channels and formats") Reported-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 708dac27feff..df8395b361d4 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -322,8 +322,6 @@ static void pcm3168a_update_fixup_pcm_stream(struct snd_soc_dai *dai) u64 formats = SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE; unsigned int channel_max = dai->id == PCM3168A_DAI_DAC ? 8 : 6; - if (!pcm3168a->io_params[dai->id].fmt) - return; if (pcm3168a->io_params[dai->id].fmt == PCM3168A_FMT_RIGHT_J) { /* S16_LE is only supported in RIGHT_J mode */ -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

1 1

[alsa-devel] [PATCH v3] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 15 Oct '19

15 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v2 to v3: - Remove spinlock around setting clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index a15fbf71b9d7..af060162b0af 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -2054,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); } /* not for DVI mode */ -- 2.23.0.581.g78d2f28ef7-goog

2 1

[alsa-devel] [RFC PATCH 0/2] Introduce for-dpcm DT property
by Daniel Baluta 15 Oct '19

15 Oct '19

We need to be able to create DPCM links even if we have a single CPU DAI or just a dummy CPU DAI. Daniel Baluta (2): ASoC: simple-card: Introduce force-dpcm DT property ASoC: simple-card: Add documentation for force-dpcm property .../devicetree/bindings/sound/simple-card.txt | 1 + include/sound/simple_card_utils.h | 4 +++ sound/soc/generic/simple-card-utils.c | 17 +++++++++++++ sound/soc/generic/simple-card.c | 25 +++++++++++++++++-- 4 files changed, 45 insertions(+), 2 deletions(-) -- 2.17.1

4 7

Re: [alsa-devel] [PATCH 2/2] ASoC: simple_card_utils.h: Fix potential multiple redefinition error
by Kuninori Morimoto 15 Oct '19

15 Oct '19

Hi > asoc_simple_debug_info and asoc_simple_debug_dai must be static > otherwise we might a compilation error if the compiler decides > not to inline the given function. > > Fixes: 0580dde59438686d ("ASoC: simple-card-utils: add asoc_simple_debug_info()") > Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> > --- Acked-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Thank you for your help !! Best regards --- Kuninori Morimoto

1 0

[alsa-devel] [PATCH] ASoC: Intel: eve: Enable mclk and ssp sclk early
by Brent Lu 14 Oct '19

14 Oct '19

From: Naveen M <naveen.m(a)intel.com> rt5663 and rt5514 needs mclk/sclk early to synchronize its internal clocks. Signed-off-by: Naveen M <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Signed-off-by: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 1 + .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 100 +++++++++++++++++++++ 2 files changed, 101 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7a..d5f167e 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -315,6 +315,7 @@ config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST depends on SPI + select SND_SOC_INTEL_SKYLAKE_SSP_CLK select SND_SOC_RT5663 select SND_SOC_RT5514 select SND_SOC_RT5514_SPI diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 74dda87..7b6e70b 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -22,6 +22,9 @@ #include "../../codecs/rt5514.h" #include "../../codecs/rt5663.h" #include "../../codecs/hdac_hdmi.h" +#include <linux/clk.h> +#include <linux/clk-provider.h> +#include <linux/clkdev.h> #define KBL_REALTEK_CODEC_DAI "rt5663-aif" #define KBL_REALTEK_DMIC_CODEC_DAI "rt5514-aif1" @@ -50,6 +53,8 @@ struct kbl_codec_private { struct snd_soc_jack kabylake_headset; struct list_head hdmi_pcm_list; struct snd_soc_jack kabylake_hdmi[2]; + struct clk *mclk; + struct clk *sclk; }; enum { @@ -71,6 +76,67 @@ static const struct snd_kcontrol_new kabylake_controls[] = { SOC_DAPM_PIN_SWITCH("DMIC"), }; +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct kbl_codec_private *priv = snd_soc_card_get_drvdata(card); + int ret = 0; + + /* + * MCLK/SCLK need to be ON early for a successful synchronization of + * codec internal clock. And the clocks are turned off during + * POST_PMD after the stream is stopped. + */ + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (__clk_is_enabled(priv->mclk)) + return 0; + + /* Enable MCLK */ + ret = clk_set_rate(priv->mclk, 24000000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for mclk, err: %d\n", + ret); + return ret; + } + + ret = clk_prepare_enable(priv->mclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable mclk, err: %d\n", ret); + return ret; + } + + /* Enable SCLK */ + ret = clk_set_rate(priv->sclk, 3072000); + if (ret < 0) { + dev_err(card->dev, "Can't set rate for sclk, err: %d\n", + ret); + clk_disable_unprepare(priv->mclk); + return ret; + } + + ret = clk_prepare_enable(priv->sclk); + if (ret < 0) { + dev_err(card->dev, "Can't enable sclk, err: %d\n", ret); + clk_disable_unprepare(priv->mclk); + } + break; + case SND_SOC_DAPM_POST_PMD: + if (!__clk_is_enabled(priv->mclk)) + return 0; + + clk_disable_unprepare(priv->mclk); + clk_disable_unprepare(priv->sclk); + break; + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), @@ -79,11 +145,15 @@ static const struct snd_soc_dapm_widget kabylake_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_SPK("HDMI1", NULL), SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), }; static const struct snd_soc_dapm_route kabylake_map[] = { /* Headphones */ + { "Headphone Jack", NULL, "Platform Clock" }, { "Headphone Jack", NULL, "HPOL" }, { "Headphone Jack", NULL, "HPOR" }, @@ -92,6 +162,7 @@ static const struct snd_soc_dapm_route kabylake_map[] = { { "Right Spk", NULL, "Right BE_OUT" }, /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, { "IN1P", NULL, "Headset Mic" }, { "IN1N", NULL, "Headset Mic" }, @@ -643,6 +714,7 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; + int ret = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -658,6 +730,34 @@ static int kabylake_audio_probe(struct platform_device *pdev) dmic_constraints = mach->mach_params.dmic_num == 2 ? &constraints_dmic_2ch : &constraints_dmic_channels; + ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); + if (IS_ERR(ctx->mclk)) { + ret = PTR_ERR(ctx->mclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_mclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", + ret); + return ret; + } + + ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); + if (IS_ERR(ctx->sclk)) { + ret = PTR_ERR(ctx->sclk); + if (ret == -ENOENT) { + dev_info(&pdev->dev, + "Failed to get ssp1_sclk, defer probe\n"); + return -EPROBE_DEFER; + } + + dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", + ret); + return ret; + } + return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.7.4

3 7

[alsa-devel] [PATCH -next] ASoC: SOF: Fix randbuild error
by YueHaibing 14 Oct '19

14 Oct '19

When LEDS_TRIGGER_AUDIO is m and SND_SOC_SOF is y, sound/soc/sof/control.o: In function `snd_sof_switch_put': control.c:(.text+0x587): undefined reference to `ledtrig_audio_set' control.c:(.text+0x593): undefined reference to `ledtrig_audio_set' Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 5d43001ae436 ("ASoC: SOF: acpi led support for switch controls") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/control.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index 41551e8..2c4abd4 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -36,10 +36,12 @@ static void update_mute_led(struct snd_sof_control *scontrol, scontrol->led_ctl.led_value = temp; +#if IS_REACHABLE(CONFIG_LEDS_TRIGGER_AUDIO) if (!scontrol->led_ctl.direction) ledtrig_audio_set(LED_AUDIO_MUTE, temp ? LED_OFF : LED_ON); else ledtrig_audio_set(LED_AUDIO_MICMUTE, temp ? LED_OFF : LED_ON); +#endif } static inline u32 mixer_to_ipc(unsigned int value, u32 *volume_map, int size) -- 2.7.4

3 2

Re: [alsa-devel] linux-next: Tree for Oct 14 (sound/soc/sof/)
by Randy Dunlap 14 Oct '19

14 Oct '19

On 10/13/19 11:47 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20191011: > on i386: ld: sound/soc/sof/control.o: in function `snd_sof_switch_put': control.c:(.text+0x49a): undefined reference to `ledtrig_audio_set' ld: control.c:(.text+0x4d1): undefined reference to `ledtrig_audio_set' when CONFIG_LEDS_TRIGGER_AUDIO=m CONFIG_SND_SOC_SOF=y This code in <linux/leds.h> does not handle the config combo above: #if IS_ENABLED(CONFIG_LEDS_TRIGGER_AUDIO) enum led_brightness ledtrig_audio_get(enum led_audio type); void ledtrig_audio_set(enum led_audio type, enum led_brightness state); #else static inline enum led_brightness ledtrig_audio_get(enum led_audio type) { return LED_OFF; } static inline void ledtrig_audio_set(enum led_audio type, enum led_brightness state) { } #endif -- ~Randy

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - edited 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was edited from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: `#inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] TypeError: 'device' is an invalid keyword argument for this function
by GitHub issues - opened 14 Oct '19

14 Oct '19

alsa-project/alsa-lib issue #13 was opened from paddygoat: Hello again! I upgraded Alsa lib to 1.1.7 and tested ok. A small error as below when running in Jupyter Notebook, Python 3.6:: ` #inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NONBLOCK) # UltraMic384K 16bit r0, USB Audio inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1)` --------------------------------------------------------------------------- TypeError Traceback (most recent call last) <ipython-input-6-947bfb637aff> in <module> 16 modelName="randomforestPaddyFalse" 17 modelType="randomforest" ---> 18 recordAnalyzeAudio(10, outputWavFile, 2.0, modelName, modelType) <ipython-input-5-8ddddfa5b724> in recordAnalyzeAudio(duration, outputWavFile, midTermBufferSizeSec, modelName, modelType) 87 # UltraMic384K 16bit r0, USB Audio 88 ---> 89 inp = alsaaudio.PCM(type = alsaaudio.PCM_CAPTURE, mode=alsaaudio.PCM_NORMAL, device='UltraMic384K 16bit r0') 90 # class alsaaudio.PCM(type=PCM_PLAYBACK, mode=PCM_NORMAL, device='default', cardindex=-1) 91 inp.setchannels(2) TypeError: 'device' is an invalid keyword argument for this function Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/13 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH v8 0/3] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 14 Oct '19

14 Oct '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v7: - Added support to dai based on Discussion on mailing list. - Moved readable array to proper switch statement. - Remove unnecessary locks which can be added once tsens is supported. - Suffix gain to Volume as suggested by Mark. Srinivas Kandagatla (3): dt-bindings: ASoC: Add WSA881x bindings dt-bindings: ASoC: WSA881x: Add missing #sound-dai-cells ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 68 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1153 +++++++++++++++++ 4 files changed, 1233 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

2 4

[alsa-devel] [PATCH] ASoC: stm32: spdifrx: retry synchronization in sync state
by Olivier Moysan 14 Oct '19

14 Oct '19

When STM32 SPDIFRX is in sync state, allow multiple synchro attempts, instead of exiting on first unsuccessful trial. This is useful when spdif signal is not immediately available on input. This also allows Pulseaudio to check iec capture device availability when no signal is present. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_spdifrx.c | 18 ++++++++++++++++-- 1 file changed, 16 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index cd4b235fce57..3fd28ee01675 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -351,6 +351,8 @@ static int stm32_spdifrx_start_sync(struct stm32_spdifrx_data *spdifrx) SPDIFRX_CR_CUMSK | SPDIFRX_CR_PTMSK | SPDIFRX_CR_RXSTEO; cr_mask = cr; + cr |= SPDIFRX_CR_NBTRSET(SPDIFRX_NBTR_63); + cr_mask |= SPDIFRX_CR_NBTR_MASK; cr |= SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); cr_mask |= SPDIFRX_CR_SPDIFEN_MASK; ret = regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, @@ -666,7 +668,7 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) struct snd_pcm_substream *substream = spdifrx->substream; struct platform_device *pdev = spdifrx->pdev; unsigned int cr, mask, sr, imr; - unsigned int flags; + unsigned int flags, sync_state; int err = 0, err_xrun = 0; regmap_read(spdifrx->regmap, STM32_SPDIFRX_SR, &sr); @@ -726,11 +728,23 @@ static irqreturn_t stm32_spdifrx_isr(int irq, void *devid) } if (err) { - /* SPDIFRX in STATE_STOP. Disable SPDIFRX to clear errors */ + regmap_read(spdifrx->regmap, STM32_SPDIFRX_CR, &cr); + sync_state = FIELD_GET(SPDIFRX_CR_SPDIFEN_MASK, cr) && + SPDIFRX_SPDIFEN_SYNC; + + /* SPDIFRX is in STATE_STOP. Disable SPDIFRX to clear errors */ cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_DISABLE); regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, SPDIFRX_CR_SPDIFEN_MASK, cr); + /* If SPDIFRX was in STATE_SYNC, retry synchro */ + if (sync_state) { + cr = SPDIFRX_CR_SPDIFENSET(SPDIFRX_SPDIFEN_SYNC); + regmap_update_bits(spdifrx->regmap, STM32_SPDIFRX_CR, + SPDIFRX_CR_SPDIFEN_MASK, cr); + return IRQ_HANDLED; + } + if (substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_DISCONNECTED); -- 2.17.1

2 1

[alsa-devel] [PATCH -next] ASoC: adau7118: Fix Kconfig warning without CONFIG_I2C
by YueHaibing 14 Oct '19

14 Oct '19

When building a kernel without CONFIG_I2C, Kconfig warns: WARNING: unmet direct dependencies detected for REGMAP_I2C Depends on [n]: I2C [=n] Selected by [y]: - SND_SOC_ADAU7118_I2C [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] Add missing I2C dependency to SND_SOC_ADAU7118_I2C to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: ca514c0f12b0 ("ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d196752..6a28741 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -416,6 +416,7 @@ config SND_SOC_ADAU7118_HW config SND_SOC_ADAU7118_I2C tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + depends on I2C select SND_SOC_ADAU7118 select REGMAP_I2C help -- 2.7.4

2 1

[alsa-devel] [PATCH -next] ASoC: fsl_mqs: fix old-style function declaration
by YueHaibing 14 Oct '19

14 Oct '19

gcc warn about this: sound/soc/fsl/fsl_mqs.c:146:1: warning: static is not at beginning of declaration [-Wold-style-declaration] Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_mqs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index f7fc44e..0c813a4 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -143,7 +143,7 @@ static void fsl_mqs_shutdown(struct snd_pcm_substream *substream, MQS_EN_MASK, 0); } -const static struct snd_soc_component_driver soc_codec_fsl_mqs = { +static const struct snd_soc_component_driver soc_codec_fsl_mqs = { .idle_bias_on = 1, .non_legacy_dai_naming = 1, }; -- 2.7.4

4 5

[alsa-devel] [PATCH 0/4] ASoC: SOF: fix static analysis warnings
by Pierre-Louis Bossart 14 Oct '19

14 Oct '19

Recent SOF commits generate warnings reported e.g. by cppcheck. Unfortunately those tools are too verbose to be used by our CI, so there's always a delay in detecting such issues. Pierre-Louis Bossart (4): ASoC: SOF: Intel: byt: fix operator precedence warnings ASoC: SOF: Intel: bdw: fix operator precedence warnings ASoC: SOF: topology: remove always-true redundant test ASoC: SOF: topology: check errors when parsing LED tokens sound/soc/sof/intel/bdw.c | 16 ++++++++-------- sound/soc/sof/intel/byt.c | 16 ++++++++-------- sound/soc/sof/topology.c | 12 ++++++++---- 3 files changed, 24 insertions(+), 20 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH v2 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 14 Oct '19

14 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Changes from v1: - fix a compile error and make kbuild bot happy https://mailman.alsa-project.org/pipermail/alsa-devel/2019-October/156315.h… Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1125 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1292 insertions(+), 326 deletions(-) -- 2.23.0.581.g78d2f28ef7-goog

4 18

[alsa-devel] [PATCH] firmware: vpd: Add an interface to read VPD value
by Cheng-Yi Chiang 14 Oct '19

14 Oct '19

Add an interface for other driver to query VPD value. This will be used for ASoC machine driver to query calibration data stored in VPD for smart amplifier speaker resistor calibration. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/firmware/google/vpd.c | 16 ++++++++++++++++ include/linux/firmware/google/google_vpd.h | 18 ++++++++++++++++++ 2 files changed, 34 insertions(+) create mode 100644 include/linux/firmware/google/google_vpd.h diff --git a/drivers/firmware/google/vpd.c b/drivers/firmware/google/vpd.c index db0812263d46..71e9d2da63be 100644 --- a/drivers/firmware/google/vpd.c +++ b/drivers/firmware/google/vpd.c @@ -65,6 +65,22 @@ static ssize_t vpd_attrib_read(struct file *filp, struct kobject *kobp, info->bin_attr.size); } +int vpd_attribute_read_value(bool ro, const char *key, + char **value, u32 value_len) +{ + struct vpd_attrib_info *info; + struct vpd_section *sec = ro ? &ro_vpd : &rw_vpd; + + list_for_each_entry(info, &sec->attribs, list) { + if (strcmp(info->key, key) == 0) { + *value = kstrndup(info->value, value_len, GFP_KERNEL); + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL(vpd_attribute_read_value); + /* * vpd_section_check_key_name() * diff --git a/include/linux/firmware/google/google_vpd.h b/include/linux/firmware/google/google_vpd.h new file mode 100644 index 000000000000..6f1160f28af8 --- /dev/null +++ b/include/linux/firmware/google/google_vpd.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Google VPD interface. + * + * Copyright 2019 Google Inc. + */ + +/* Interface for reading VPD value on Chrome platform. */ + +#ifndef __GOOGLE_VPD_H +#define __GOOGLE_VPD_H + +#include <linux/types.h> + +int vpd_attribute_read_value(bool ro, const char *key, + char **value, u32 value_len); + +#endif /* __GOOGLE_VPD_H */ -- 2.23.0.581.g78d2f28ef7-goog

6 10

[alsa-devel] [PATCH] ASoC: rt1011: export r0 and temperature config
by Shuming [范書銘] 14 Oct '19

14 Oct '19

In chromebook case, the machine driver will get the r0 calibration data and temperature from VPD. Therefore, the codec exports r0 and temperature config API for it. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 41 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt1011.h | 5 +++++ 2 files changed, 46 insertions(+) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index a92a0bacd812..a992bb2224b2 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1325,6 +1325,47 @@ static int rt1011_r0_load_info(struct snd_kcontrol *kcontrol, .put = rt1011_r0_load_mode_put \ } +int rt1011_r0_config(struct snd_soc_component *component, unsigned int r0) +{ + struct rt1011_priv *rt1011 = + snd_soc_component_get_drvdata(component); + struct device *dev; + unsigned int r0_integer, r0_factor, format; + + if (!component->card->instantiated) + return 0; + + if (!r0) + return -EINVAL; + + dev = regmap_get_device(rt1011->regmap); + rt1011->r0_reg = r0; + + format = 2147483648U; /* 2^24 * 128 */ + r0_integer = format / rt1011->r0_reg / 128; + r0_factor = ((format / rt1011->r0_reg * 100) / 128) + - (r0_integer * 100); + dev_dbg(dev, "New r0 resistance about %d.%02d ohm, reg=0x%X\n", + r0_integer, r0_factor, rt1011->r0_reg); + + if (rt1011->r0_reg) + rt1011_r0_load(rt1011); + + return 0; +} +EXPORT_SYMBOL_GPL(rt1011_r0_config); + +int rt1011_temperature_config(struct snd_soc_component *component, + unsigned int temp) +{ + snd_soc_component_update_bits(component, + RT1011_STP_INITIAL_RESISTANCE_TEMP, 0x3ff, + (temp << 2)); + + return 0; +} +EXPORT_SYMBOL_GPL(rt1011_temperature_config); + static const struct snd_kcontrol_new rt1011_snd_controls[] = { /* I2S Data In Selection */ SOC_ENUM("DIN Source", rt1011_din_source_enum), diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..b99df334be14 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -670,4 +670,9 @@ struct rt1011_priv { int recv_spk_mode; }; +int rt1011_r0_config(struct snd_soc_component *component, unsigned int r0); +int rt1011_temperature_config(struct snd_soc_component *component, + unsigned int temp); + + #endif /* end of _RT1011_H_ */ -- 2.23.0

4 9

[alsa-devel] [PATCH RESEND] ASoC: rt1011: set tx/rx slots from tx/rx_mask in TDM case
by Shuming [范書銘] 14 Oct '19

14 Oct '19

The TX/RX slot configuration use tx/rx_mask which requested by the machine driver. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1011.c | 97 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/rt1011.h | 23 ++++++++++ 2 files changed, 115 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1011.c b/sound/soc/codecs/rt1011.c index be1e276e3631..a36144de11cf 100644 --- a/sound/soc/codecs/rt1011.c +++ b/sound/soc/codecs/rt1011.c @@ -1805,8 +1805,8 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, struct snd_soc_component *component = dai->component; struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); - unsigned int val = 0, tdm_en = 0; - int ret = 0; + unsigned int val = 0, tdm_en = 0, rx_slotnum, tx_slotnum; + int ret = 0, first_bit, last_bit; snd_soc_dapm_mutex_lock(dapm); if (rx_mask || tx_mask) @@ -1850,6 +1850,91 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } + /* Rx slot configuration */ + rx_slotnum = hweight_long(rx_mask); + first_bit = find_next_bit((unsigned long *)&rx_mask, 32, 0); + if (rx_slotnum > 1 || rx_slotnum == 0) { + ret = -EINVAL; + dev_dbg(component->dev, "too many rx slots or zero slot\n"); + goto _set_tdm_err_; + } + if (rx_slotnum) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 6) { + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_L_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + (first_bit << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + ((first_bit+1) + << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + } else { + snd_soc_component_update_bits(component, + RT1011_CROSS_BQ_SET_1, RT1011_MONO_LR_SEL_MASK, + RT1011_MONO_R_CHANNEL); + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_4, + RT1011_TDM_I2S_TX_L_DAC1_1_MASK | + RT1011_TDM_I2S_TX_R_DAC1_1_MASK, + ((first_bit-1) + << RT1011_TDM_I2S_TX_L_DAC1_1_SFT) | + (first_bit << RT1011_TDM_I2S_TX_R_DAC1_1_SFT)); + } + } + + /* Tx slot configuration */ + tx_slotnum = hweight_long(tx_mask); + first_bit = find_next_bit((unsigned long *)&tx_mask, 32, 0); + last_bit = find_last_bit((unsigned long *)&tx_mask, 32); + if (tx_slotnum > 2 || (last_bit-first_bit) > 1) { + ret = -EINVAL; + dev_dbg(component->dev, "too many tx slots or tx slot location error\n"); + goto _set_tdm_err_; + } + + if (tx_slotnum == 1) { + snd_soc_component_update_bits(component, RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, first_bit); + if (first_bit == 1) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC1_1_MASK, + RT1011_TDM_I2S_RX_ADC1_1_LL); + else if (first_bit == 3) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC2_1_MASK, + RT1011_TDM_I2S_RX_ADC2_1_LL); + else if (first_bit == 5) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC3_1_MASK, + RT1011_TDM_I2S_RX_ADC3_1_LL); + else if (first_bit == 7) + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_3, + RT1011_TDM_I2S_RX_ADC4_1_MASK, + RT1011_TDM_I2S_RX_ADC4_1_LL); + } else if (tx_slotnum == 2) { + if (first_bit == 0 || first_bit == 2 || + first_bit == 4 || first_bit == 6) { + snd_soc_component_update_bits(component, + RT1011_TDM1_SET_2, + RT1011_TDM_I2S_DOCK_ADCDAT_LEN_1_MASK | + RT1011_TDM_ADCDAT1_DATA_LOCATION, + RT1011_TDM_I2S_DOCK_ADCDAT_2CH | first_bit); + } else { + ret = -EINVAL; + dev_dbg(component->dev, + "tx slot location should be paired and start from slot0/2/4/6\n"); + goto _set_tdm_err_; + } + } + snd_soc_component_update_bits(component, RT1011_TDM1_SET_1, RT1011_I2S_CH_TX_MASK | RT1011_I2S_CH_RX_MASK | RT1011_I2S_CH_TX_LEN_MASK | RT1011_I2S_CH_RX_LEN_MASK, val); @@ -1860,10 +1945,12 @@ static int rt1011_set_tdm_slot(struct snd_soc_dai *dai, RT1011_TDM_I2S_DOCK_EN_1_MASK, tdm_en); snd_soc_component_update_bits(component, RT1011_TDM2_SET_2, RT1011_TDM_I2S_DOCK_EN_2_MASK, tdm_en); - snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, - RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, - RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); + if (tx_slotnum) + snd_soc_component_update_bits(component, RT1011_TDM_TOTAL_SET, + RT1011_ADCDAT1_PIN_CONFIG | RT1011_ADCDAT2_PIN_CONFIG, + RT1011_ADCDAT1_OUTPUT | RT1011_ADCDAT2_OUTPUT); +_set_tdm_err_: snd_soc_dapm_mutex_unlock(dapm); return ret; } diff --git a/sound/soc/codecs/rt1011.h b/sound/soc/codecs/rt1011.h index 2d65983f3d0f..3f7dea1191ad 100644 --- a/sound/soc/codecs/rt1011.h +++ b/sound/soc/codecs/rt1011.h @@ -460,6 +460,23 @@ #define RT1011_TDM_I2S_DOCK_EN_1_MASK (0x1 << 3) #define RT1011_TDM_I2S_DOCK_EN_1_SFT 3 #define RT1011_TDM_I2S_DOCK_EN_1 (0x1 << 3) +#define RT1011_TDM_ADCDAT1_DATA_LOCATION (0x7 << 0) + +/* TDM1 Setting-3 (0x0118) */ +#define RT1011_TDM_I2S_RX_ADC1_1_MASK (0x3 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_MASK (0x3 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_MASK (0x3 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_MASK (0x3 << 0) +#define RT1011_TDM_I2S_RX_ADC1_1_LL (0x2 << 6) +#define RT1011_TDM_I2S_RX_ADC2_1_LL (0x2 << 4) +#define RT1011_TDM_I2S_RX_ADC3_1_LL (0x2 << 2) +#define RT1011_TDM_I2S_RX_ADC4_1_LL (0x2 << 0) + +/* TDM1 Setting-4 (0x011a) */ +#define RT1011_TDM_I2S_TX_L_DAC1_1_MASK (0x7 << 12) +#define RT1011_TDM_I2S_TX_R_DAC1_1_MASK (0x7 << 8) +#define RT1011_TDM_I2S_TX_L_DAC1_1_SFT 12 +#define RT1011_TDM_I2S_TX_R_DAC1_1_SFT 8 /* TDM2 Setting-2 (0x0120) */ #define RT1011_TDM_I2S_DOCK_ADCDAT_LEN_2_MASK (0x7 << 13) @@ -585,6 +602,12 @@ #define RT1011_STP_T0_EN_BIT 6 #define RT1011_STP_T0_EN (0x1 << 6) +/* Cross Biquad Setting-1 (0x0702) */ +#define RT1011_MONO_LR_SEL_MASK (0x3 << 5) +#define RT1011_MONO_L_CHANNEL (0x0 << 5) +#define RT1011_MONO_R_CHANNEL (0x1 << 5) +#define RT1011_MONO_LR_MIX_CHANNEL (0x2 << 5) + /* ClassD Internal Setting-1 (0x1300) */ #define RT1011_DRIVER_READY_SPK (0x1 << 12) #define RT1011_DRIVER_READY_SPK_BIT 12 -- 2.23.0

2 2

Re: [alsa-devel] [PATCH -next] ASoC: atmel: select SND_ATMEL_SOC_DMA for SND_ATMEL_SOC_SSC
by Alexandre Belloni 12 Oct '19

12 Oct '19

On 12/10/2019 10:42:30+0800, Mao Wenan wrote: > If SND_ATMEL_SOC_SSC_PDC=y and SND_ATMEL_SOC_SSC_DMA=m, > below errors can be found: > sound/soc/atmel/atmel_ssc_dai.o: In function > `atmel_ssc_set_audio': > atmel_ssc_dai.c:(.text+0x6fe): undefined reference to > `atmel_pcm_dma_platform_register' > make: *** [vmlinux] Error 1 > > After commit 18291410557f ("ASoC: atmel: enable > SOC_SSC_PDC and SOC_SSC_DMA in Kconfig"), SND_ATMEL_SOC_DMA > and SND_ATMEL_SOC_SSC are selected by SND_ATMEL_SOC_SSC_DMA, > SND_ATMEL_SOC_SSC is also selected by SND_ATMEL_SOC_SSC_PDC, > the results are SND_ATMEL_SOC_DMA=m but SND_ATMEL_SOC_SSC=y, > so the errors happen. > > This patch make SND_ATMEL_SOC_SSC select SND_ATMEL_SOC_DMA. > > Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") > Signed-off-by: Mao Wenan <maowenan(a)huawei.com> > --- > sound/soc/atmel/Kconfig | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig > index f118c22..2938f6b 100644 > --- a/sound/soc/atmel/Kconfig > +++ b/sound/soc/atmel/Kconfig > @@ -19,6 +19,7 @@ config SND_ATMEL_SOC_DMA > > config SND_ATMEL_SOC_SSC > tristate > + select SND_ATMEL_SOC_DMA > This is not the solution because this doesn't allow to compile out DMA and use only PDC. I think Arnd already submitted a proper patch. > config SND_ATMEL_SOC_SSC_PDC > tristate "SoC PCM DAI support for AT91 SSC controller using PDC" > -- > 2.7.4 > -- Alexandre Belloni, Bootlin Embedded Linux and Kernel engineering https://bootlin.com

1 0

[alsa-devel] [PATCH v3 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 11 Oct '19

11 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- Changes in v2: * Use C++ coment style on the entire comment in the .c files; * Do not use ternery operator; * Do not use `devm_regulator_get_optional` as the device always requires power; * Let DAPM suspend/resume the device. Changes in v3: * no changes. MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 586 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 770 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index 55199ef7fa74..bf2c71f90eb6 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1002,6 +1002,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ae213c6392ec 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -396,6 +398,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73b2d5982dcb..2118407a1dd5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -305,6 +308,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..45a5d2dcc0f2 --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw +// driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..a8211362fe82 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..bf5a5d75f81a --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,586 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + st->iovdd = devm_regulator_get(st->dev, "IOVDD"); + if (IS_ERR(st->iovdd)) { + dev_err(st->dev, "Could not get iovdd: %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + return devm_add_action_or_reset(st->dev, adau7118_regulator_disable, + st); +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

4 7

[alsa-devel] [PATCH 00/21] ASoC: soc-core cleanup - step 4
by Kuninori Morimoto 11 Oct '19

11 Oct '19

Hi Mark Cc Pierre-Louis These are step4 of soc-core cleanup. These related to soc-topology. If my understanding and my code are correct, there is no effect to topology side, but I can't test. I'm not sure who can test these, but I guess Pierre-Louis knows ? Pierre-Louis, can you help it ? Kuninori Morimoto (21): ASoC: soc-core: remove for_each_rtdcom_safe() ASoC: soc-core: add for_each_rtd_components() and replace ASoC: soc-core: move soc_init_dai_link() ASoC: soc-core: rename soc_init_dai_link() to soc_dai_link_sanity_check() ASoC: soc-core: remove duplicated soc_is_dai_link_bound() ASoC: soc-core: call soc_bind_dai_link() under snd_soc_add_dai_link() ASoC: soc-core: add soc_unbind_dai_link() ASoC: soc-core: snd_soc_unbind_card() cleanup ASoC: soc-core: remove unneeded snd_soc_tplg_component_remove() ASoC: soc-core: move snd_soc_lookup_component() ASoC: soc-core: add snd_soc_del_component() ASoC: soc-core: use snd_soc_lookup_component() at snd_soc_unregister_component() ASoC: soc-core: move snd_soc_register_dai() ASoC: soc-core: have legacy_dai_naming at snd_soc_register_dai() ASoC: soc-core: move snd_soc_unregister_dais() ASoC: soc-core: add snd_soc_unregister_dai() ASoC: soc-core: don't call snd_soc_dapm_new_dai_widgets() at snd_soc_register_dai() ASoC: soc-core: use mutex_lock() at snd_soc_add_component() ASoC: soc-core: call snd_soc_register_dai() from snd_soc_register_dais() ASoC: soc-core: remove topology specific operation ASoC: soc.h: dobj is used only when SND_SOC_TOPOLOGY include/sound/soc.h | 25 ++- sound/soc/soc-component.c | 43 +--- sound/soc/soc-compress.c | 52 ++--- sound/soc/soc-core.c | 542 ++++++++++++++++++++-------------------------- sound/soc/soc-pcm.c | 49 ++--- sound/soc/soc-topology.c | 32 ++- 6 files changed, 320 insertions(+), 423 deletions(-) -- 2.7.4 Thank you for your help !! Best regards --- Kuninori Morimoto

2 41

[alsa-devel] [PATCH v2 0/5] soundwire: Add support to Qualcomm SoundWire master
by Srinivas Kandagatla 11 Oct '19

11 Oct '19

Thanks for reviewing the RFC patchset. Here is new patchset addressing all the comments from RFC. This patchset adds support for Qualcomm SoundWire Master Controller found in most of Qualcomm SoCs and WCD audio codecs. This driver along with WCD934x codec and WSA881x Class-D Smart Speaker Amplifier drivers is on DragonBoard DB845c based of SDM845 SoC. WCD934x and WSA881x patches will be posted soon. SoundWire controller on SDM845 is integrated in WCD934x audio codec via SlimBus interface. Currently this driver is very minimal and only supports PDM. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Test and add PCM support. Thanks, srini Changes since RFC: - updated bindings as suggested to take care of more bus parameters. - fixed error code of snd_soc_dai_get_sdw_stream() dummy function. - Cleaned up driver to handle read/writes in same way without special casing. - removed unused defines Srinivas Kandagatla (4): soundwire: stream: make stream name a const pointer ASoC: core: add support to snd_soc_dai_get_sdw_stream() dt-bindings: soundwire: add bindings for Qcom controller soundwire: qcom: add support for SoundWire controller Vinod Koul (1): soundwire: Add compute_params callback .../bindings/soundwire/qcom,sdw.txt | 167 ++++ drivers/soundwire/Kconfig | 9 + drivers/soundwire/Makefile | 4 + drivers/soundwire/qcom.c | 919 ++++++++++++++++++ drivers/soundwire/stream.c | 12 +- include/linux/soundwire/sdw.h | 6 +- include/sound/soc-dai.h | 10 + 7 files changed, 1124 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/soundwire/qcom,sdw.txt create mode 100644 drivers/soundwire/qcom.c -- 2.21.0

6 40

[alsa-devel] [PATCH 20/36] ASoC: samsung: s3c2412-i2s: avoid hardcoded S3C2410_PA_IIS
by Arnd Bergmann 11 Oct '19

11 Oct '19

The constant requires indirectly including a machine header file, but it's not actually used any more since commit 87b132bc0315 ("ASoC: samsung: s3c24{xx,12}-i2s: port to use generic dmaengine API"), so remove it completely. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/s3c-i2s-v2.c | 3 +-- sound/soc/samsung/s3c-i2s-v2.h | 3 +-- sound/soc/samsung/s3c2412-i2s.c | 2 +- 3 files changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 7e196b599be1..6a254fdaa6fb 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -616,8 +616,7 @@ int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate); int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base) + struct s3c_i2sv2_info *i2s) { struct device *dev = dai->dev; unsigned int iismod; diff --git a/sound/soc/samsung/s3c-i2s-v2.h b/sound/soc/samsung/s3c-i2s-v2.h index fe42b77999fd..8c6fc0d3d77e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.h +++ b/sound/soc/samsung/s3c-i2s-v2.h @@ -83,8 +83,7 @@ extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info, * @base: The base address for the registers. */ extern int s3c_i2sv2_probe(struct snd_soc_dai *dai, - struct s3c_i2sv2_info *i2s, - unsigned long base); + struct s3c_i2sv2_info *i2s); /** * s3c_i2sv2_cleanup - cleanup resources allocated in s3c_i2sv2_probe diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 787a3f6e9f24..5bf571762e89 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -49,7 +49,7 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) snd_soc_dai_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, &s3c2412_i2s_pcm_stereo_in); - ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s); if (ret) return ret; -- 2.20.0

2 1

[alsa-devel] Counting links in simple-card
by Daniel Baluta 11 Oct '19

11 Oct '19

Hi Morimoto-san, In simple-card.c simple_get_dais_count function there is this comment: * ex1) * CPU0 --- Codec0 * CPU1 --- Codec1 * CPU2 -----/ * CPU3 --- Codec2 * * => 5 links = 2xCPU-Codec + 2xCPU-dummy + 1xdummy-Codec * => 7 DAIs = 4xCPU + 3xCodec * => 1 ccnf = 1xdummy-Codec So, I don't understand exactly how did you get the formula for links. AFAICT there should be 3 x CPU-Codec + 1 x CPU-dummy. thanks, Daniel.

2 7

[alsa-devel] [PATCH v2] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 10 Oct '19

10 Oct '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - Added type and description and removed number of items from 'model' property. Best regards, Maciej Falkowski --- .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2-audio.yaml | 84 +++++++++++++++++++ 2 files changed, 84 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..62b5b9a1dec0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,84 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are + the WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + +required: + - compatible + - model + - audio-codec + - i2s-controller + - audio-amplifier + - samsung,audio-routing + - mic-bias-gpios + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

3 2

[alsa-devel] [PATCH 19/36] ARM: s3c: rx1950: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-rx1950.c | 16 +++++++ sound/soc/samsung/rx1950_uda1380.c | 72 +++++++++-------------------- 2 files changed, 39 insertions(+), 49 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-rx1950.c b/arch/arm/mach-s3c24xx/mach-rx1950.c index 29f9b345a531..9f3b0f18cd36 100644 --- a/arch/arm/mach-s3c24xx/mach-rx1950.c +++ b/arch/arm/mach-s3c24xx/mach-rx1950.c @@ -724,6 +724,20 @@ static struct i2c_board_info rx1950_i2c_devices[] = { }, }; +static struct gpiod_lookup_table rx1950_audio_gpio_table = { + .dev_id = "rx1950-audio", + .table = { + GPIO_LOOKUP("GPIOG", 12, "hp-gpio", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOA", 1, "speaker-power", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device rx1950_audio = { + .name = "rx1950-audio", + .id = -1, +}; + static struct platform_device *rx1950_devices[] __initdata = { &s3c2410_device_dclk, &s3c_device_lcd, @@ -742,6 +756,7 @@ static struct platform_device *rx1950_devices[] __initdata = { &power_supply, &rx1950_battery, &rx1950_leds, + &rx1950_audio, }; static void __init rx1950_map_io(void) @@ -809,6 +824,7 @@ static void __init rx1950_init_machine(void) gpio_direction_output(S3C2410_GPJ(6), 0); pwm_add_table(rx1950_pwm_lookup, ARRAY_SIZE(rx1950_pwm_lookup)); + gpiod_add_lookup_table(&rx1950_audio_gpio_table); platform_add_devices(rx1950_devices, ARRAY_SIZE(rx1950_devices)); i2c_register_board_info(0, rx1950_i2c_devices, diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 4b247e91ae5b..cf96eaaf6b4e 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -12,16 +12,13 @@ // Vasily Khoruzhick <anarsoul(a)gmail.com> #include <linux/types.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <sound/soc.h> #include <sound/jack.h> -#include <mach/gpio-samsung.h> #include "regs-iis.h" -#include <asm/mach-types.h> - #include "s3c24xx-i2s.h" static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd); @@ -58,7 +55,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { [0] = { - .gpio = S3C2410_GPG(12), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,8 +119,6 @@ static struct snd_soc_card rx1950_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static struct platform_device *s3c24xx_snd_device; - static int rx1950_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -134,13 +128,15 @@ static int rx1950_startup(struct snd_pcm_substream *substream) &hw_rates); } +struct gpio_desc *gpiod_speaker_power; + static int rx1950_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C2410_GPA(1), 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C2410_GPA(1), 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -214,57 +210,35 @@ static int rx1950_uda1380_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int __init rx1950_init(void) +static int rx1950_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_rx1950()) - return -ENODEV; + struct device *dev = &pdev->dev; /* configure some gpios */ - ret = gpio_request(S3C2410_GPA(1), "speaker-power"); - if (ret) - goto err_gpio; - - ret = gpio_direction_output(S3C2410_GPA(1), 0); - if (ret) - goto err_gpio_conf; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_plat_alloc; - } - - platform_set_drvdata(s3c24xx_snd_device, &rx1950_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) { - platform_device_put(s3c24xx_snd_device); - goto err_plat_add; + gpiod_speaker_power = devm_gpiod_get(dev, "speaker-power", GPIOD_OUT_LOW); + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "cannot get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - return 0; - -err_plat_add: -err_plat_alloc: -err_gpio_conf: - gpio_free(S3C2410_GPA(1)); + hp_jack_gpios[0].gpiod_dev = dev; + rx1950_asoc.dev = dev; -err_gpio: - return ret; + return devm_snd_soc_register_card(dev, &rx1950_asoc); } -static void __exit rx1950_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C2410_GPA(1)); -} +struct platform_driver rx1950_audio = { + .driver = { + .name = "rx1950-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = rx1950_probe, +}; -module_init(rx1950_init); -module_exit(rx1950_exit); +module_platform_driver(rx1950_audio); /* Module information */ MODULE_AUTHOR("Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC RX1950"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:rx1950-audio"); -- 2.20.0

1 0

[alsa-devel] [PATCH 18/36] ARM: s3c: gta02-audio: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-gta02.c | 17 ++++++ sound/soc/samsung/neo1973_wm8753.c | 85 +++++++++--------------------- 2 files changed, 43 insertions(+), 59 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-gta02.c b/arch/arm/mach-s3c24xx/mach-gta02.c index 594901f3b8e5..526fd0933289 100644 --- a/arch/arm/mach-s3c24xx/mach-gta02.c +++ b/arch/arm/mach-s3c24xx/mach-gta02.c @@ -15,6 +15,7 @@ #include <linux/delay.h> #include <linux/timer.h> #include <linux/init.h> +#include <linux/gpio/machine.h> #include <linux/gpio.h> #include <linux/gpio_keys.h> #include <linux/workqueue.h> @@ -474,6 +475,20 @@ static struct platform_device gta02_buttons_device = { }, }; +static struct gpiod_lookup_table gta02_audio_gpio_table = { + .dev_id = "neo1973-audio", + .table = { + GPIO_LOOKUP("GPIOJ", 2, "amp-shut", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOJ", 1, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device gta02_audio = { + .name = "neo1973-audio", + .id = -1, +}; + static void __init gta02_map_io(void) { s3c24xx_init_io(gta02_iodesc, ARRAY_SIZE(gta02_iodesc)); @@ -498,6 +513,7 @@ static struct platform_device *gta02_devices[] __initdata = { &gta02_buttons_device, &s3c_device_adc, &s3c_device_ts, + &gta02_audio, }; static void gta02_poweroff(void) @@ -524,6 +540,7 @@ static void __init gta02_machine_init(void) i2c_register_board_info(0, gta02_i2c_devs, ARRAY_SIZE(gta02_i2c_devs)); + gpiod_add_lookup_table(&gta02_audio_gpio_table); platform_add_devices(gta02_devices, ARRAY_SIZE(gta02_devices)); pm_power_off = gta02_poweroff; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 38f536bafa09..b41e3dc297e7 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -11,14 +11,11 @@ #include <linux/module.h> #include <linux/platform_device.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <sound/soc.h> -#include <mach/gpio-samsung.h> -#include <asm/mach-types.h> #include "regs-iis.h" - #include "../codecs/wm8753.h" #include "s3c24xx-i2s.h" @@ -166,6 +163,7 @@ static struct snd_soc_ops neo1973_voice_ops = { .hw_free = neo1973_voice_hw_free, }; +static struct gpio_desc *gpiod_hp_in, *gpiod_amp_shut; static int gta02_speaker_enabled; static int lm4853_set_spk(struct snd_kcontrol *kcontrol, @@ -173,7 +171,7 @@ static int lm4853_set_spk(struct snd_kcontrol *kcontrol, { gta02_speaker_enabled = ucontrol->value.integer.value[0]; - gpio_set_value(S3C2410_GPJ(2), !gta02_speaker_enabled); + gpiod_set_value(gpiod_hp_in, !gta02_speaker_enabled); return 0; } @@ -188,7 +186,7 @@ static int lm4853_get_spk(struct snd_kcontrol *kcontrol, static int lm4853_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *k, int event) { - gpio_set_value(S3C2410_GPJ(1), SND_SOC_DAPM_EVENT_OFF(event)); + gpiod_set_value(gpiod_amp_shut, SND_SOC_DAPM_EVENT_OFF(event)); return 0; } @@ -308,13 +306,8 @@ static struct snd_soc_codec_conf neo1973_codec_conf[] = { }, }; -static const struct gpio neo1973_gta02_gpios[] = { - { S3C2410_GPJ(2), GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" }, - { S3C2410_GPJ(1), GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" }, -}; - static struct snd_soc_card neo1973 = { - .name = "neo1973", + .name = "neo1973gta02", .owner = THIS_MODULE, .dai_link = neo1973_dai, .num_links = ARRAY_SIZE(neo1973_dai), @@ -332,62 +325,36 @@ static struct snd_soc_card neo1973 = { .fully_routed = true, }; -static struct platform_device *neo1973_snd_device; - -static int __init neo1973_init(void) +static int neo1973_probe(struct platform_device *pdev) { - int ret; - - if (!machine_is_neo1973_gta02()) - return -ENODEV; + struct device *dev = &pdev->dev; - if (machine_is_neo1973_gta02()) { - neo1973.name = "neo1973gta02"; - neo1973.num_aux_devs = 1; - - ret = gpio_request_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - if (ret) - return ret; + gpiod_hp_in = devm_gpiod_get(dev, "hp", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_hp_in)) { + dev_err(dev, "missing gpio %s\n", "hp"); + return PTR_ERR(gpiod_hp_in); } - - neo1973_snd_device = platform_device_alloc("soc-audio", -1); - if (!neo1973_snd_device) { - ret = -ENOMEM; - goto err_gpio_free; + gpiod_amp_shut = devm_gpiod_get(dev, "amp-shut", GPIOD_OUT_HIGH); + if (IS_ERR(gpiod_amp_shut)) { + dev_err(dev, "missing gpio %s\n", "amp-shut"); + return PTR_ERR(gpiod_amp_shut); } - platform_set_drvdata(neo1973_snd_device, &neo1973); - ret = platform_device_add(neo1973_snd_device); - - if (ret) - goto err_put_device; - - return 0; - -err_put_device: - platform_device_put(neo1973_snd_device); -err_gpio_free: - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } - return ret; + neo1973.dev = dev; + return devm_snd_soc_register_card(dev, &neo1973); } -module_init(neo1973_init); - -static void __exit neo1973_exit(void) -{ - platform_device_unregister(neo1973_snd_device); - if (machine_is_neo1973_gta02()) { - gpio_free_array(neo1973_gta02_gpios, - ARRAY_SIZE(neo1973_gta02_gpios)); - } -} -module_exit(neo1973_exit); +struct platform_driver neo1973_audio = { + .driver = { + .name = "neo1973-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = neo1973_probe, +}; +module_platform_driver(neo1973_audio); /* Module information */ MODULE_AUTHOR("Graeme Gregory, graeme(a)openmoko.org, www.openmoko.org"); MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:neo1973-audio"); -- 2.20.0

1 0

[alsa-devel] [PATCH 17/36] ARM: s3c: h1940-audio: turn into platform driver
by Arnd Bergmann 10 Oct '19

10 Oct '19

Avoid machine specific headers by using a gpio lookup table combined with a platform_driver for this board. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- arch/arm/mach-s3c24xx/mach-h1940.c | 18 ++++++++ sound/soc/samsung/h1940_uda1380.c | 71 ++++++++++-------------------- 2 files changed, 41 insertions(+), 48 deletions(-) diff --git a/arch/arm/mach-s3c24xx/mach-h1940.c b/arch/arm/mach-s3c24xx/mach-h1940.c index 74d6b68e91c7..922d53bacbdf 100644 --- a/arch/arm/mach-s3c24xx/mach-h1940.c +++ b/arch/arm/mach-s3c24xx/mach-h1940.c @@ -475,6 +475,22 @@ static struct gpiod_lookup_table h1940_mmc_gpio_table = { }, }; +static struct gpiod_lookup_table h1940_audio_gpio_table = { + .dev_id = "h1940-audio", + .table = { + GPIO_LOOKUP("H1940_LATCH", + H1940_LATCH_AUDIO_POWER - H1940_LATCH_GPIO(0), + "speaker-power", GPIO_ACTIVE_HIGH), + GPIO_LOOKUP("GPIOG", 4, "hp", GPIO_ACTIVE_HIGH), + { }, + }, +}; + +static struct platform_device h1940_audio = { + .name = "h1940-audio", + .id = -1, +}; + static struct pwm_lookup h1940_pwm_lookup[] = { PWM_LOOKUP("samsung-pwm", 0, "pwm-backlight", NULL, 36296, PWM_POLARITY_NORMAL), @@ -652,6 +668,7 @@ static struct platform_device *h1940_devices[] __initdata = { &s3c_device_ts, &power_supply, &h1940_battery, + &h1940_audio, }; static void __init h1940_map_io(void) @@ -691,6 +708,7 @@ static void __init h1940_init(void) s3c24xx_fb_set_platdata(&h1940_fb_info); gpiod_add_lookup_table(&h1940_mmc_gpio_table); + gpiod_add_lookup_table(&h1940_audio_gpio_table); s3c24xx_mci_set_platdata(&h1940_mmc_cfg); s3c24xx_udc_set_platdata(&h1940_udc_cfg); s3c24xx_ts_set_platdata(&h1940_ts_cfg); diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index a95c34e53a2b..5fc001b4effd 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -15,9 +15,6 @@ #include <sound/jack.h> #include "regs-iis.h" -#include <asm/mach-types.h> - -#include <mach/gpio-samsung.h> #include "s3c24xx-i2s.h" static const unsigned int rates[] = { @@ -31,6 +28,8 @@ static const struct snd_pcm_hw_constraint_list hw_rates = { .list = rates, }; +static struct gpio_desc *gpiod_speaker_power; + static struct snd_soc_jack hp_jack; static struct snd_soc_jack_pin hp_jack_pins[] = { @@ -47,7 +46,6 @@ static struct snd_soc_jack_pin hp_jack_pins[] = { static struct snd_soc_jack_gpio hp_jack_gpios[] = { { - .gpio = S3C2410_GPG(4), .name = "hp-gpio", .report = SND_JACK_HEADPHONE, .invert = 1, @@ -123,9 +121,9 @@ static int h1940_spk_power(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { if (SND_SOC_DAPM_EVENT_ON(event)) - gpio_set_value(S3C_GPIO_END + 9, 1); + gpiod_set_value(gpiod_speaker_power, 1); else - gpio_set_value(S3C_GPIO_END + 9, 0); + gpiod_set_value(gpiod_speaker_power, 0); return 0; } @@ -151,8 +149,6 @@ static const struct snd_soc_dapm_route audio_map[] = { {"VINM", NULL, "Mic Jack"}, }; -static struct platform_device *s3c24xx_snd_device; - static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) { snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, @@ -194,55 +190,34 @@ static struct snd_soc_card h1940_asoc = { .num_dapm_routes = ARRAY_SIZE(audio_map), }; -static int __init h1940_init(void) +static int h1940_probe(struct platform_device *pdev) { - int ret; + struct device *dev = &pdev->dev; - if (!machine_is_h1940()) - return -ENODEV; + h1940_asoc.dev = dev; + hp_jack_gpios[0].gpiod_dev = dev; + gpiod_speaker_power = devm_gpiod_get(&pdev->dev, "speaker-power", + GPIOD_OUT_LOW); - /* configure some gpios */ - ret = gpio_request(S3C_GPIO_END + 9, "speaker-power"); - if (ret) - goto err_out; - - ret = gpio_direction_output(S3C_GPIO_END + 9, 0); - if (ret) - goto err_gpio; - - s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); - if (!s3c24xx_snd_device) { - ret = -ENOMEM; - goto err_gpio; + if (IS_ERR(gpiod_speaker_power)) { + dev_err(dev, "Could not get gpio\n"); + return PTR_ERR(gpiod_speaker_power); } - platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); - ret = platform_device_add(s3c24xx_snd_device); - - if (ret) - goto err_plat; - - return 0; - -err_plat: - platform_device_put(s3c24xx_snd_device); -err_gpio: - gpio_free(S3C_GPIO_END + 9); - -err_out: - return ret; -} - -static void __exit h1940_exit(void) -{ - platform_device_unregister(s3c24xx_snd_device); - gpio_free(S3C_GPIO_END + 9); + return devm_snd_soc_register_card(dev, &h1940_asoc); } -module_init(h1940_init); -module_exit(h1940_exit); +static struct platform_driver h1940_audio_driver = { + .driver = { + .name = "h1940-audio", + .pm = &snd_soc_pm_ops, + }, + .probe = h1940_probe, +}; +module_platform_driver(h1940_audio_driver); /* Module information */ MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC H1940"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:h1940-audio"); -- 2.20.0

1 0

[alsa-devel] Record at 384 KHz
by GitHub issues - opened 10 Oct '19

10 Oct '19

alsa-project/alsa-lib issue #12 was opened from paddygoat: Tried everything I can think of to record at 384. How do I do it? Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/12 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH 0/9] ASoC: SOF updates for Intel/i.MX
by Pierre-Louis Bossart 10 Oct '19

10 Oct '19

Small set of updates for Intel and NXP platforms. On the Intel side, the code is now aligned with the HDaudio legacy with the sync_write. OEMs reported concerns with mute LEDs not supported with SOF/UCM, we can now tie a topology-defined control with a mute LED. Note that the large set of differences in the topology is largely due to code move, the new functionality is minimal in size. On the i.MX side, add topology support for ESAI parameters. This reflects on-going work on the firmware side. And last add debug module parameters for ACPI and PCI modules. For now the only flag used is for PM runtime, e.g. to measure the suspend-resume latencies of child devices without dealing with DSP reinitialization or keep the parent active while testing the SoundWire clock-stop-mode. Daniel Baluta (2): ASoC: SOF: imx: Describe ESAI parameters to be sent to DSP ASoC: SOF: imx: Read ESAI parameters and send them to DSP Jaska Uimonen (3): ASoC: SOF: enable dual control for pga AsoC: SOF: refactor control load code ASoC: SOF: acpi led support for switch controls Kai Vehmanen (2): ASoC: SOF: enable sync_write in hdac_bus Revert "ASoC: SOF: Force polling mode on CFL and CNL" Pierre-Louis Bossart (2): ASoC: SOF: acpi: add debug module param ASoC: SOF: pci: add debug module param include/sound/sof/dai-imx.h | 34 +++ include/sound/sof/dai.h | 2 + include/uapi/sound/sof/abi.h | 2 +- include/uapi/sound/sof/tokens.h | 7 +- sound/soc/sof/control.c | 32 +++ sound/soc/sof/intel/hda.c | 8 +- sound/soc/sof/pcm.c | 8 + sound/soc/sof/sof-acpi-dev.c | 12 +- sound/soc/sof/sof-pci-dev.c | 12 +- sound/soc/sof/sof-priv.h | 9 + sound/soc/sof/topology.c | 401 +++++++++++++++++++------------- 11 files changed, 354 insertions(+), 173 deletions(-) create mode 100644 include/sound/sof/dai-imx.h -- 2.20.1

2 20

[alsa-devel] Applied "ASoC: tas2562: Introduce the TAS2562 amplifier" to the asoc tree
by Mark Brown 10 Oct '19

10 Oct '19

The patch ASoC: tas2562: Introduce the TAS2562 amplifier has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From c173dba44c2d2ec2adaa4e607e9729b74236bd57 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Tue, 8 Oct 2019 13:15:17 -0500 Subject: [PATCH] ASoC: tas2562: Introduce the TAS2562 amplifier Introduce the Texas Instruments TAS2562 amplifier driver with I/V sense for loud speaker applications. http://www.ti.com/lit/gpn/tas2562 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20191008181517.5332-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2562.c | 590 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 85 ++++++ 4 files changed, 682 insertions(+) create mode 100644 sound/soc/codecs/tas2562.c create mode 100644 sound/soc/codecs/tas2562.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ae213c6392ec..f4747ebc251e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -181,6 +181,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_STI_SAS select SND_SOC_TAS2552 if I2C + select SND_SOC_TAS2562 if I2C select SND_SOC_TAS2770 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C @@ -1133,6 +1134,10 @@ config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" depends on I2C +config SND_SOC_TAS2562 + tristate "Texas Instruments TAS2562 Mono Audio amplifier" + depends on I2C + config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 2118407a1dd5..ddfd07071925 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ snd-soc-max98504-objs := max98504.o snd-soc-simple-amplifier-objs := simple-amplifier.o snd-soc-tpa6130a2-objs := tpa6130a2.o snd-soc-tas2552-objs := tas2552.o +snd-soc-tas2562-objs := tas2562.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -481,6 +482,7 @@ obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_STI_SAS) += snd-soc-sti-sas.o obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o +obj-$(CONFIG_SND_SOC_TAS2562) += snd-soc-tas2562.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c new file mode 100644 index 000000000000..729acd874c48 --- /dev/null +++ b/sound/soc/codecs/tas2562.c @@ -0,0 +1,590 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for the Texas Instruments TAS2562 CODEC +// Copyright (C) 2019 Texas Instruments Inc. + + +#include <linux/module.h> +#include <linux/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> +#include <linux/delay.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "tas2562.h" + +#define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FORMAT_S32_LE) + +struct tas2562_data { + struct snd_soc_component *component; + struct gpio_desc *sdz_gpio; + struct regmap *regmap; + struct device *dev; + struct i2c_client *client; + int v_sense_slot; + int i_sense_slot; +}; + +static int tas2562_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct tas2562_data *tas2562 = + snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_ACTIVE); + break; + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_PREPARE: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_MUTE); + break; + case SND_SOC_BIAS_OFF: + snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_SHUTDOWN); + break; + + default: + dev_err(tas2562->dev, + "wrong power level setting %d\n", level); + return -EINVAL; + } + + return 0; +} + +static int tas2562_set_samplerate(struct tas2562_data *tas2562, int samplerate) +{ + int samp_rate; + int ramp_rate; + + switch (samplerate) { + case 7350: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ; + break; + case 8000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ; + break; + case 14700: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ; + break; + case 16000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ; + break; + case 22050: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ; + break; + case 24000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ; + break; + case 29400: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ; + break; + case 32000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ; + break; + case 44100: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ; + break; + case 48000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ; + break; + case 88200: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ; + break; + case 96000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ; + break; + case 176400: + ramp_rate = TAS2562_TDM_CFG0_RAMPRATE_44_1; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ; + break; + case 192000: + ramp_rate = 0; + samp_rate = TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ; + break; + default: + dev_info(tas2562->dev, "%s, unsupported sample rate, %d\n", + __func__, samplerate); + return -EINVAL; + } + + snd_soc_component_update_bits(tas2562->component, TAS2562_TDM_CFG0, + TAS2562_TDM_CFG0_RAMPRATE_MASK, ramp_rate); + snd_soc_component_update_bits(tas2562->component, TAS2562_TDM_CFG0, + TAS2562_TDM_CFG0_SAMPRATE_MASK, samp_rate); + + return 0; +} + +static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret = 0; + + switch (slot_width) { + case 16: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_16B); + break; + case 24: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_24B); + break; + case 32: + ret = snd_soc_component_update_bits(component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXLEN_MASK, + TAS2562_TDM_CFG2_RXLEN_32B); + break; + + case 0: + /* Do not change slot width */ + break; + default: + dev_err(tas2562->dev, "slot width not supported"); + ret = -EINVAL; + } + + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) +{ + int ret; + + switch (bitwidth) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_16B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_24B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 4; + break; + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG2, + TAS2562_TDM_CFG2_RXWLEN_MASK, + TAS2562_TDM_CFG2_RXWLEN_32B); + tas2562->v_sense_slot = tas2562->i_sense_slot + 4; + break; + + default: + dev_info(tas2562->dev, "Not supported params format\n"); + } + + ret = snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG5, + TAS2562_TDM_CFG5_VSNS_EN | TAS2562_TDM_CFG5_VSNS_SLOT_MASK, + TAS2562_TDM_CFG5_VSNS_EN | tas2562->v_sense_slot); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(tas2562->component, + TAS2562_TDM_CFG6, + TAS2562_TDM_CFG6_ISNS_EN | TAS2562_TDM_CFG6_ISNS_SLOT_MASK, + TAS2562_TDM_CFG6_ISNS_EN | tas2562->i_sense_slot); + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + + ret = tas2562_set_bitwidth(tas2562, params_format(params)); + if (ret) { + dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + return ret; + } + + ret = tas2562_set_samplerate(tas2562, params_rate(params)); + if (ret) + dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + + return ret; +} + +static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + u8 tdm_rx_start_slot = 0, asi_cfg_1 = 0; + int ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + asi_cfg_1 = 0; + break; + case SND_SOC_DAIFMT_IB_NF: + asi_cfg_1 |= TAS2562_TDM_CFG1_RX_FALLING; + break; + default: + dev_err(tas2562->dev, "ASI format Inverse is not found\n"); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, + TAS2562_TDM_CFG1_RX_EDGE_MASK, + asi_cfg_1); + if (ret < 0) { + dev_err(tas2562->dev, "Failed to set RX edge\n"); + return ret; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case (SND_SOC_DAIFMT_I2S): + case (SND_SOC_DAIFMT_DSP_A): + case (SND_SOC_DAIFMT_DSP_B): + tdm_rx_start_slot = BIT(1); + break; + case (SND_SOC_DAIFMT_LEFT_J): + tdm_rx_start_slot = 0; + break; + default: + dev_err(tas2562->dev, "DAI Format is not found, fmt=0x%x\n", + fmt); + ret = -EINVAL; + break; + } + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, + TAS2562_TDM_CFG1_RX_OFFSET_MASK, + tdm_rx_start_slot); + + if (ret < 0) + return ret; + + return 0; +} + +static int tas2562_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_component *component = dai->component; + + return snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + mute ? TAS2562_MUTE : 0); +} + +static int tas2562_codec_probe(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + + tas2562->component = component; + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 1); + + ret = snd_soc_component_update_bits(component, TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, TAS2562_MUTE); + if (ret < 0) + return ret; + + return 0; +} + +#ifdef CONFIG_PM +static int tas2562_suspend(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(tas2562->regmap, true); + regcache_mark_dirty(tas2562->regmap); + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 0); + + return 0; +} + +static int tas2562_resume(struct snd_soc_component *component) +{ + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + if (tas2562->sdz_gpio) + gpiod_set_value_cansleep(tas2562->sdz_gpio, 1); + + regcache_cache_only(tas2562->regmap, false); + + return regcache_sync(tas2562->regmap); +} +#else +#define tas2562_suspend NULL +#define tas2562_resume NULL +#endif + +static const char * const tas2562_ASI1_src[] = { + "I2C offset", "Left", "Right", "LeftRightDiv2", +}; + +static SOC_ENUM_SINGLE_DECL(tas2562_ASI1_src_enum, TAS2562_TDM_CFG2, 4, + tas2562_ASI1_src); + +static const struct snd_kcontrol_new tas2562_asi1_mux = + SOC_DAPM_ENUM("ASI1 Source", tas2562_ASI1_src_enum); + +static int tas2562_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + break; + default: + break; + } + + return 0; +} + +static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); + +static const struct snd_kcontrol_new isense_switch = + SOC_DAPM_SINGLE("Switch", TAS2562_PWR_CTRL, TAS2562_ISENSE_POWER_EN, + 1, 1); + +static const struct snd_kcontrol_new vsense_switch = + SOC_DAPM_SINGLE("Switch", TAS2562_PWR_CTRL, TAS2562_VSENSE_POWER_EN, + 1, 1); + +static const struct snd_kcontrol_new tas2562_snd_controls[] = { + SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, + tas2562_dac_tlv), +}; + +static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), + SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), + SND_SOC_DAPM_SWITCH("VSENSE", TAS2562_PWR_CTRL, 2, 1, &vsense_switch), + SND_SOC_DAPM_SIGGEN("VMON"), + SND_SOC_DAPM_SIGGEN("IMON"), + SND_SOC_DAPM_OUTPUT("OUT"), +}; + +static const struct snd_soc_dapm_route tas2562_audio_map[] = { + {"ASI1 Sel", "I2C offset", "ASI1"}, + {"ASI1 Sel", "Left", "ASI1"}, + {"ASI1 Sel", "Right", "ASI1"}, + {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, + { "DAC", NULL, "DAC IN" }, + { "OUT", NULL, "DAC" }, + {"ISENSE", "Switch", "IMON"}, + {"VSENSE", "Switch", "VMON"}, +}; + +static const struct snd_soc_component_driver soc_component_dev_tas2562 = { + .probe = tas2562_codec_probe, + .suspend = tas2562_suspend, + .resume = tas2562_resume, + .set_bias_level = tas2562_set_bias_level, + .controls = tas2562_snd_controls, + .num_controls = ARRAY_SIZE(tas2562_snd_controls), + .dapm_widgets = tas2562_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tas2562_dapm_widgets), + .dapm_routes = tas2562_audio_map, + .num_dapm_routes = ARRAY_SIZE(tas2562_audio_map), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct snd_soc_dai_ops tas2562_speaker_dai_ops = { + .hw_params = tas2562_hw_params, + .set_fmt = tas2562_set_dai_fmt, + .set_tdm_slot = tas2562_set_dai_tdm_slot, + .digital_mute = tas2562_mute, +}; + +static struct snd_soc_dai_driver tas2562_dai[] = { + { + .name = "tas2562-amplifier", + .id = 0, + .playback = { + .stream_name = "ASI1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, + .ops = &tas2562_speaker_dai_ops, + }, +}; + +static const struct regmap_range_cfg tas2562_ranges[] = { + { + .range_min = 0, + .range_max = 5 * 128, + .selector_reg = TAS2562_PAGE_CTRL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +static const struct reg_default tas2562_reg_defaults[] = { + { TAS2562_PAGE_CTRL, 0x00 }, + { TAS2562_SW_RESET, 0x00 }, + { TAS2562_PWR_CTRL, 0x0e }, + { TAS2562_PB_CFG1, 0x20 }, + { TAS2562_TDM_CFG0, 0x09 }, + { TAS2562_TDM_CFG1, 0x02 }, +}; + +static const struct regmap_config tas2562_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = 5 * 128, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = tas2562_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2562_reg_defaults), + .ranges = tas2562_ranges, + .num_ranges = ARRAY_SIZE(tas2562_ranges), +}; + +static int tas2562_parse_dt(struct tas2562_data *tas2562) +{ + struct device *dev = tas2562->dev; + int ret = 0; + + tas2562->sdz_gpio = devm_gpiod_get_optional(dev, "shut-down-gpio", + GPIOD_OUT_HIGH); + if (IS_ERR(tas2562->sdz_gpio)) { + if (PTR_ERR(tas2562->sdz_gpio) == -EPROBE_DEFER) { + tas2562->sdz_gpio = NULL; + return -EPROBE_DEFER; + } + } + + ret = fwnode_property_read_u32(dev->fwnode, "ti,imon-slot-no", + &tas2562->i_sense_slot); + if (ret) + dev_err(dev, "Looking up %s property failed %d\n", + "ti,imon-slot-no", ret); + + return ret; +} + +static int tas2562_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev = &client->dev; + struct tas2562_data *data; + int ret; + + data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->client = client; + data->dev = &client->dev; + + tas2562_parse_dt(data); + + data->regmap = devm_regmap_init_i2c(client, &tas2562_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(dev, "failed to allocate register map: %d\n", ret); + return ret; + } + + dev_set_drvdata(&client->dev, data); + + return devm_snd_soc_register_component(dev, &soc_component_dev_tas2562, + tas2562_dai, + ARRAY_SIZE(tas2562_dai)); + +} + +static const struct i2c_device_id tas2562_id[] = { + { "tas2562", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2562_id); + +static const struct of_device_id tas2562_of_match[] = { + { .compatible = "ti,tas2562", }, + { }, +}; +MODULE_DEVICE_TABLE(of, tas2562_of_match); + +static struct i2c_driver tas2562_i2c_driver = { + .driver = { + .name = "tas2562", + .of_match_table = of_match_ptr(tas2562_of_match), + }, + .probe = tas2562_probe, + .id_table = tas2562_id, +}; + +module_i2c_driver(tas2562_i2c_driver); + +MODULE_AUTHOR("Dan Murphy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2562 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h new file mode 100644 index 000000000000..60f2bb1d198b --- /dev/null +++ b/sound/soc/codecs/tas2562.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * tas2562.h - ALSA SoC Texas Instruments TAS2562 Mono Audio Amplifier + * + * Copyright (C) 2019 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + */ + +#ifndef __TAS2562_H__ +#define __TAS2562_H__ + +#define TAS2562_PAGE_CTRL 0x00 + +#define TAS2562_REG(page, reg) ((page * 128) + reg) + +#define TAS2562_SW_RESET TAS2562_REG(0, 0x01) +#define TAS2562_PWR_CTRL TAS2562_REG(0, 0x02) +#define TAS2562_PB_CFG1 TAS2562_REG(0, 0x03) +#define TAS2562_MISC_CFG1 TAS2562_REG(0, 0x04) +#define TAS2562_MISC_CFG2 TAS2562_REG(0, 0x05) + +#define TAS2562_TDM_CFG0 TAS2562_REG(0, 0x06) +#define TAS2562_TDM_CFG1 TAS2562_REG(0, 0x07) +#define TAS2562_TDM_CFG2 TAS2562_REG(0, 0x08) +#define TAS2562_TDM_CFG3 TAS2562_REG(0, 0x09) +#define TAS2562_TDM_CFG4 TAS2562_REG(0, 0x0a) +#define TAS2562_TDM_CFG5 TAS2562_REG(0, 0x0b) +#define TAS2562_TDM_CFG6 TAS2562_REG(0, 0x0c) +#define TAS2562_TDM_CFG7 TAS2562_REG(0, 0x0d) +#define TAS2562_TDM_CFG8 TAS2562_REG(0, 0x0e) +#define TAS2562_TDM_CFG9 TAS2562_REG(0, 0x0f) +#define TAS2562_TDM_CFG10 TAS2562_REG(0, 0x10) +#define TAS2562_TDM_DET TAS2562_REG(0, 0x11) +#define TAS2562_REV_ID TAS2562_REG(0, 0x7d) + +/* Page 2 */ +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) + +#define TAS2562_RESET BIT(0) + +#define TAS2562_MODE_MASK 0x3 +#define TAS2562_ACTIVE 0x0 +#define TAS2562_MUTE 0x1 +#define TAS2562_SHUTDOWN 0x2 + +#define TAS2562_TDM_CFG1_RX_EDGE_MASK BIT(0) +#define TAS2562_TDM_CFG1_RX_FALLING 1 +#define TAS2562_TDM_CFG1_RX_OFFSET_MASK GENMASK(4, 0) + +#define TAS2562_TDM_CFG0_RAMPRATE_MASK BIT(5) +#define TAS2562_TDM_CFG0_RAMPRATE_44_1 BIT(5) +#define TAS2562_TDM_CFG0_SAMPRATE_MASK GENMASK(3, 1) +#define TAS2562_TDM_CFG0_SAMPRATE_7305_8KHZ 0x0 +#define TAS2562_TDM_CFG0_SAMPRATE_14_7_16KHZ 0x1 +#define TAS2562_TDM_CFG0_SAMPRATE_22_05_24KHZ 0x2 +#define TAS2562_TDM_CFG0_SAMPRATE_29_4_32KHZ 0x3 +#define TAS2562_TDM_CFG0_SAMPRATE_44_1_48KHZ 0x4 +#define TAS2562_TDM_CFG0_SAMPRATE_88_2_96KHZ 0x5 +#define TAS2562_TDM_CFG0_SAMPRATE_176_4_192KHZ 0x6 + +#define TAS2562_TDM_CFG2_RIGHT_JUSTIFY BIT(6) + +#define TAS2562_TDM_CFG2_RXLEN_MASK GENMASK(0, 1) +#define TAS2562_TDM_CFG2_RXLEN_16B 0x0 +#define TAS2562_TDM_CFG2_RXLEN_24B BIT(0) +#define TAS2562_TDM_CFG2_RXLEN_32B BIT(1) + +#define TAS2562_TDM_CFG2_RXWLEN_MASK GENMASK(3, 2) +#define TAS2562_TDM_CFG2_RXWLEN_16B 0x0 +#define TAS2562_TDM_CFG2_RXWLEN_20B BIT(2) +#define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) +#define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) + +#define TAS2562_VSENSE_POWER_EN BIT(2) +#define TAS2562_ISENSE_POWER_EN BIT(3) + +#define TAS2562_TDM_CFG5_VSNS_EN BIT(6) +#define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) + +#define TAS2562_TDM_CFG6_ISNS_EN BIT(6) +#define TAS2562_TDM_CFG6_ISNS_SLOT_MASK GENMASK(5, 0) + +#endif /* __TAS2562_H__ */ -- 2.20.1

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[alsa-devel] Question about DPCM FE vs BE
by Kuninori Morimoto 10 Oct '19

10 Oct '19

Hi ALSA ML In my understanding, DPCM needs FE and BE. And, one FE can have multiple BE, and one BE can have multiple FE. My question this time is that one snd_soc_pcm_runtime can be both FE and BE in the same time (= Sometimes FE, sometimes BE) ?? In my understanding, it never happen. But, is this correct ? I'm asking because do we need .be_clients/.fe_clients ? If one pcm_runtime can't be FE / BE in the same time, just .clients is enough I think. static int dpcm_be_connect(...) { ... - list_add(&dpcm->list_be, &fe->dpcm[stream].be_clients); - list_add(&dpcm->list_fe, &be->dpcm[stream].fe_clients); + list_add(&dpcm->list_be, &fe->dpcm[stream].clients); + list_add(&dpcm->list_fe, &be->dpcm[stream].clients); ... } Thank you for your help !! Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v7 0/2] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 10 Oct '19

10 Oct '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 and Lenovo YOGA C630 Laptop based on SDM850 with WSA8815 speaker amplifiers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v6: - Added reviewed-by tag from Pierre and Rob - Removed udelay argument as suggested by Pierre. - Added device id for WSA8815 - rebased on sound-next Srinivas Kandagatla (2): dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 62 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1113 +++++++++++++++++ 4 files changed, 1187 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: add missing MIX2 path for RX1/2
by Srinivas Kandagatla 09 Oct '19

09 Oct '19

This patch adds missing MIX2 path on RX1/2 which take IIR1 and IIR2 as inputs. Without this patch sound card fails to intialize with below warning: ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX2 MIX2 INP1 ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX2 MIX2 INP1 Reported-by: Stephan Gerhold <stephan(a)gerhold.net> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Tested-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-digital.c | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9fa5d44fdc79..58b2468fb2a7 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -243,6 +243,10 @@ static const char *const rx_mix1_text[] = { "ZERO", "IIR1", "IIR2", "RX1", "RX2", "RX3" }; +static const char * const rx_mix2_text[] = { + "ZERO", "IIR1", "IIR2" +}; + static const char *const dec_mux_text[] = { "ZERO", "ADC1", "ADC2", "ADC3", "DMIC1", "DMIC2" }; @@ -270,6 +274,16 @@ static const struct soc_enum rx3_mix1_inp_enum[] = { SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX3_B2_CTL, 0, 6, rx_mix1_text), }; +/* RX1 MIX2 */ +static const struct soc_enum rx_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX1_B3_CTL, + 0, 3, rx_mix2_text); + +/* RX2 MIX2 */ +static const struct soc_enum rx2_mix2_inp1_chain_enum = + SOC_ENUM_SINGLE(LPASS_CDC_CONN_RX2_B3_CTL, + 0, 3, rx_mix2_text); + /* DEC */ static const struct soc_enum dec1_mux_enum = SOC_ENUM_SINGLE( LPASS_CDC_CONN_TX_B1_CTL, 0, 6, dec_mux_text); @@ -309,6 +323,10 @@ static const struct snd_kcontrol_new rx3_mix1_inp2_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP2 Mux", rx3_mix1_inp_enum[1]); static const struct snd_kcontrol_new rx3_mix1_inp3_mux = SOC_DAPM_ENUM( "RX3 MIX1 INP3 Mux", rx3_mix1_inp_enum[2]); +static const struct snd_kcontrol_new rx1_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX1 MIX2 INP1 Mux", rx_mix2_inp1_chain_enum); +static const struct snd_kcontrol_new rx2_mix2_inp1_mux = SOC_DAPM_ENUM( + "RX2 MIX2 INP1 Mux", rx2_mix2_inp1_chain_enum); /* Digital Gain control -38.4 dB to +38.4 dB in 0.3 dB steps */ static const DECLARE_TLV_DB_SCALE(digital_gain, -3840, 30, 0); @@ -740,6 +758,10 @@ static const struct snd_soc_dapm_widget msm8916_wcd_digital_dapm_widgets[] = { &rx3_mix1_inp2_mux), SND_SOC_DAPM_MUX("RX3 MIX1 INP3", SND_SOC_NOPM, 0, 0, &rx3_mix1_inp3_mux), + SND_SOC_DAPM_MUX("RX1 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx1_mix2_inp1_mux), + SND_SOC_DAPM_MUX("RX2 MIX2 INP1", SND_SOC_NOPM, 0, 0, + &rx2_mix2_inp1_mux), SND_SOC_DAPM_MUX("CIC1 MUX", SND_SOC_NOPM, 0, 0, &cic1_mux), SND_SOC_DAPM_MUX("CIC2 MUX", SND_SOC_NOPM, 0, 0, &cic2_mux), -- 2.21.0

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[alsa-devel] [PATCH 0/2] ASoC: simple_card_utils.h: Fix two potential compilation errors
by Daniel Baluta 09 Oct '19

09 Oct '19

For this to happen we symbol DEBUG to be defined. Daniel Baluta (2): ASoC: simple_card_utils.h: Add missing include ASoC: Fix potential multiple redefinition error include/sound/simple_card_utils.h | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) -- 2.17.1

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[alsa-devel] [RFC PATCH] ASoC: soc-dapm: Invalidate DAPM path during dapm addition of routes
by Srinivas Kandagatla 09 Oct '19

09 Oct '19

From: Sudheer Papothi <spapothi(a)codeaurora.org> During sound card registration, dapm adds routes of codec and other component paths, but the invalidation of the widgets in these paths will happen only when the sound card is instantiated. As these routes are added before sound card instantiation, these widgets are not invalidated until a playback or recording usecase is started. Audio playback or recording usecase is not started in the case of codec loopback. So, if codec loopback is performed just after soundcard registration, then the widgets are not powered up as those widgets are not invalidated, results into codec loopback failure. Change is to remove the sound card instantiation check condition in dapm add paths, so widgets get invalidated whenever they are added. Signed-off-by: Sudheer Papothi <spapothi(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-dapm.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b6378f025836..13544f7c850b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2862,8 +2862,7 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(widgets[dir], "Route added"); } - if (dapm->card->instantiated && path->connect) - dapm_path_invalidate(path); + dapm_path_invalidate(path); return 0; err: -- 2.21.0

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[alsa-devel] [RFC PATCH] ASoC: soc-dapm: Skip suspending widgets with ignore flag
by Srinivas Kandagatla 09 Oct '19

09 Oct '19

From: Sudheer Papothi <spapothi(a)codeaurora.org> For wigdets which have set the suspend ignore flag asoc framework shouldn't mark them as dirty when ASoC suspend function is called. This change adds check to skip suspending the widgets with the flag set. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Sudheer Papothi <spapothi(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/soc-dapm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 13544f7c850b..4ecfd32e59b8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -303,6 +303,8 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card) mutex_lock(&card->dapm_mutex); list_for_each_entry(w, &card->widgets, list) { + if (w->ignore_suspend) + continue; if (w->is_ep) { dapm_mark_dirty(w, "Rechecking endpoints"); if (w->is_ep & SND_SOC_DAPM_EP_SINK) -- 2.21.0

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[alsa-devel] [PATCH 0/1] [for 4.14] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 09 Oct '19

09 Oct '19

This backport is for 4.14-stable. Oleksandr Suvorov (1): ASoC: sgtl5000: Improve VAG power and mute control sound/soc/codecs/sgtl5000.c | 234 +++++++++++++++++++++++++++++++----- 1 file changed, 203 insertions(+), 31 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/1] [for 4.4/4.9] VAG power control improvement for sgtl5000 codec
by Oleksandr Suvorov 09 Oct '19

09 Oct '19

This is a backport to stable kernel versions 4.4 and 4.9. Oleksandr Suvorov (1): ASoC: sgtl5000: Improve VAG power and mute control sound/soc/codecs/sgtl5000.c | 234 +++++++++++++++++++++++++++++++----- 1 file changed, 203 insertions(+), 31 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ASoC: msm8916-wcd-digital: Remove broken MIX2 DAPM routes
by Stephan Gerhold 09 Oct '19

09 Oct '19

Since 5.3-rc1, the kernel (correctly) complains about missing widgets when loading msm8916-wcd-digital: ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR1 -> IIR1 -> RX2 MIX2 INP1 ASoC: no sink widget found for RX1 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX1 MIX2 INP1 ASoC: no sink widget found for RX2 MIX2 INP1 ASoC: Failed to add route IIR2 -> IIR2 -> RX2 MIX2 INP1 This happens because RX1/2 MIX2 INP1 is not actually supported by msm8916-wcd-digital for some reason. There were some traces of it in the initial patch, but the last remnants were removed in commit fecd5c09ddf8 ("ASoC: msm8916-wcd-digital: remove few unused variables"). On 5.3 everything seemingly worked even with that error, but on 5.4-rc1 the sound card is no longer successfully registered: qcom-apq8016-sbc 7702000.sound: ASoC: failed to instantiate card -19 This happens because a check for the return code of snd_soc_dapm_add_routes() was added in commit daa480bde6b3 ("ASoC: soc-core: tidyup for snd_soc_dapm_add_routes()"). Avoid the error by removing the routes that reference the missing widgets. Fixes: ef8a4757a6db ("ASoC: msm8916-wcd-digital: Add sidetone support") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/codecs/msm8916-wcd-digital.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9fa5d44fdc79..36d8f619e298 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1022,11 +1022,6 @@ static const struct snd_soc_dapm_route msm8916_wcd_digital_audio_map[] = { {"RX3 MIX1 INP2", "IIR1", "IIR1"}, {"RX3 MIX1 INP2", "IIR2", "IIR2"}, - {"RX1 MIX2 INP1", "IIR1", "IIR1"}, - {"RX2 MIX2 INP1", "IIR1", "IIR1"}, - {"RX1 MIX2 INP1", "IIR2", "IIR2"}, - {"RX2 MIX2 INP1", "IIR2", "IIR2"}, - {"IIR1", NULL, "IIR1 INP1 MUX"}, {"IIR1 INP1 MUX", "DEC1", "DEC1 MUX"}, {"IIR1 INP1 MUX", "DEC2", "DEC2 MUX"}, -- 2.23.0

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[alsa-devel] [PATCH] ASoC: jz4740: Remove unused match variable
by Stephen Boyd 09 Oct '19

09 Oct '19

After commit 67ad656bdd70 ("ASoC: jz4740: Use of_device_get_match_data()"), the match local variable is unused and the compiler rightly warns. sound/soc/jz4740/jz4740-i2s.c: In function 'jz4740_i2s_dev_probe': sound/soc/jz4740/jz4740-i2s.c:500:29: warning: unused variable 'match' [-Wunused-variable] 500 | const struct of_device_id *match; Drop it. Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Paul Cercueil <paul(a)crapouillou.net> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Frank Rowand <frowand.list(a)gmail.com> Cc: <alsa-devel(a)alsa-project.org> Fixes: 67ad656bdd70 ("ASoC: jz4740: Use of_device_get_match_data()") Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- Can also be squashed. Sorry I missed this warning in the compile log. sound/soc/jz4740/jz4740-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index d2dab4d24b87..38d48d101783 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -497,7 +497,6 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) struct jz4740_i2s *i2s; struct resource *mem; int ret; - const struct of_device_id *match; i2s = devm_kzalloc(&pdev->dev, sizeof(*i2s), GFP_KERNEL); if (!i2s) base-commit: 442630f691a1537b7e0cc35e3d580222077549cb -- Sent by a computer through tubes

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[alsa-devel] [PATCH v3] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 09 Oct '19

09 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 7 +- sound/hda/intel-dsp-config.c | 247 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 48 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 321 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..3f06377a1808 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,247 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <linux/dmi.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST (1<<0) +#define FLAG_SOF (1<<1) +#define FLAG_SOF_ONLY_IF_DMIC (1<<16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table - the order of entries is important! + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..7e262678abeb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Force this driver to be used and bypass the DSP driver selection"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2060,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

3 7

[alsa-devel] [PATCH v4] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 09 Oct '19

09 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems. The examples are taken from the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 +++++ sound/hda/Kconfig | 10 +- sound/hda/Makefile | 4 +- sound/hda/intel-dsp-config.c | 249 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +- sound/pci/hda/hda_intel.c | 49 ++---- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 +-- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 + 11 files changed, 321 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..e3903e8f2d97 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,5 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o intel-nhlt.o +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..d9f6d9e872b4 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,249 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <linux/bits.h> +#include <linux/dmi.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/intel-dsp-config.h> +#include <sound/intel-nhlt.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=legacy, 2=SST, 3=SOF)"); + +#define FLAG_SST BIT(0) +#define FLAG_SOF BIT(1) +#define FLAG_SOF_ONLY_IF_DMIC BIT(16) + +struct config_entry { + u32 flags; + u16 device; + const struct dmi_system_id *dmi_table; +}; + +/* + * configuration table + * - the order of similar PCI ID entries is important! + * - the first successful match will win + */ +static const struct config_entry config_table[] = { +/* Cometlake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_LP) + { + /* prefer SST */ + .flags = FLAG_SST, + .device = 0x02c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + { + .flags = FLAG_SOF, + .device = 0x02c8, + }, +#endif +/* Cometlake-H */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_CML_H) + { + .flags = FLAG_SST, + .device = 0x06c8, + }, +#elif IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + { + .flags = FLAG_SOF, + .device = 0x06c8, + }, +#endif +/* Merrifield */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_MERRIFIELD) + { + .flags = FLAG_SOF, + .device = 0x119a, + }, +#endif +/* Broxton-T */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x1a98, + }, +#endif +/* Geminilake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + { + .flags = FLAG_SOF, + .device = 0x3198, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Google Chromebooks", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_GLK) + { + .flags = FLAG_SST, + .device = 0x3198, + }, +#endif +/* Icelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ICELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x34c8, + }, +#endif +/* Elkhart Lake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_ELKHARTLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x4b55, + }, +#endif +/* Appololake (Broxton-P) */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + { + .flags = FLAG_SOF, + .device = 0x5a98, + .dmi_table = (const struct dmi_system_id []) { + { + .ident = "Up Squared", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AAEON"), + DMI_MATCH(DMI_BOARD_NAME, "UP-APL01"), + } + }, + {} + } + }, +#endif +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + { + .flags = FLAG_SST, + .device = 0x5a98, + }, +#endif +/* Cannonlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9dc8, + }, +#endif +/* Sunrise Point-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_SKYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d70, + }, +#endif +/* Kabylake-LP */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_KABYLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0x9d71, + }, +#endif +/* Tigerlake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_TIGERLAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa0c8, + }, +#endif +/* Coffelake */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC, + .device = 0xa348, + }, +#endif +}; + +static const struct config_entry *snd_intel_dsp_find_config + (struct pci_dev *pci, const struct config_entry *table, u32 len) +{ + u16 device; + + device = pci->device; + for (; len > 0; len--, table++) { + if (table->device != device) + continue; + if (table->dmi_table && !dmi_check_system(table->dmi_table)) + continue; + return table; + } + return NULL; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + const struct config_entry *cfg; + + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* find the configuration for the specific device */ + cfg = snd_intel_dsp_find_config(pci, config_table, ARRAY_SIZE(config_table)); + if (!cfg) + return SND_INTEL_DSP_DRIVER_ANY; + + if (cfg->flags & FLAG_SOF) { + if (cfg->flags & FLAG_SOF_ONLY_IF_DMIC) { + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + } else { + return SND_INTEL_DSP_DRIVER_SOF; + } + } + + if (cfg->flags & FLAG_SST) + return SND_INTEL_DSP_DRIVER_SST; + + return SND_INTEL_DSP_DRIVER_LEGACY; +} +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..a916963822e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool dsp_driver = 1; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,9 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(dsp_driver, bool, 0444); +MODULE_PARM_DESC(dsp_driver, "Allow DSP driver selection (bypass this driver) " + "(0=off, 1=on) (default=1)"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2021,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2038,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: soc-pcm: fixup dpcm_prune_paths() loop continue
by Kuninori Morimoto 09 Oct '19

09 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> dpcm_prune_paths() is checking widget at 2 parts. (A) is for CPU, (B) is for Codec. If we focus to (A) part, continue at (a) is for (1) loop. But, if we focus to (B) part, continue at (b) is for (2) loop, not for (1). This is bug. This patch fixup this issue. static int dpcm_prune_paths(...) { ... (1) for_each_dpcm_be(fe, stream, dpcm) { ... ^ widget = dai_get_widget(...); | (A) if (widget && widget_in_list(...)) | (a) continue; v ^ (2) for_each_rtd_codec_dai(...) { | widget = dai_get_widget(...); (B) | if (widget && widget_in_list(...)) v (b) continue; } ... Fixes: 2e5894d73789 ("ASoC: pcm: Add support for DAI multicodec") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-pcm.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 6706b91..5d973ac 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1443,6 +1443,7 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget *widget; struct snd_soc_dai *dai; int prune = 0; + int do_prune; /* Destroy any old FE <--> BE connections */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1456,13 +1457,16 @@ static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* is there a valid CODEC DAI widget for this BE */ + do_prune = 1; for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) - continue; + do_prune = 0; } + if (!do_prune) + continue; dev_dbg(fe->dev, "ASoC: pruning %s BE %s for %s\n", stream ? "capture" : "playback", -- 2.7.4

1 0

[alsa-devel] Is this bug ? dpcm_prune_paths()
by Kuninori Morimoto 09 Oct '19

09 Oct '19

Hi ALSA ML. I'm checking soc-pcm.c, and I noticed strange code at dpcm_prune_paths(). static int dpcm_prune_paths(...) { ... /* Destroy any old FE <--> BE connections */ (1) for_each_dpcm_be(fe, stream, dpcm) { unsigned int i; - /* is there a valid CPU DAI widget for this BE */ | widget = dai_get_widget(dpcm->be->cpu_dai, stream); | (A) /* prune the BE if it's no longer in our active list */ | if (widget && widget_in_list(list, widget)) |(a) continue; - | /* is there a valid CODEC DAI widget for this BE */ |(2) for_each_rtd_codec_dai(dpcm->be, i, dai) { | widget = dai_get_widget(dai, stream); (B) | /* prune the BE if it's no longer in our active list */ | if (widget && widget_in_list(list, widget)) |(b) continue; - } - ... | dpcm->state = ... (C) dpcm->be->dpcm[stream].runtime_update = ... | prune++; - } ... } In my understanding, (A) part is for CPU, and (B) part is for Codec. Guessing from (a), I think it want to skip setup (C) if CPU widget exist and matches to list. If so, I guess (b) is maybe bug ? This continue is for (2) loop, thus, it is totally do nothing now. But maybe it want to be continue for (1) loop ? Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[alsa-devel] [PATCH -next] ASoc: tas2770: Fix build error without GPIOLIB
by YueHaibing 08 Oct '19

08 Oct '19

If GPIOLIB is not set, building fails: sound/soc/codecs/tas2770.c: In function tas2770_reset: sound/soc/codecs/tas2770.c:38:3: error: implicit declaration of function gpiod_set_value_cansleep; did you mean gpio_set_value_cansleep? [-Werror=implicit-function-declaration] gpiod_set_value_cansleep(tas2770->reset_gpio, 0); ^~~~~~~~~~~~~~~~~~~~~~~~ gpio_set_value_cansleep sound/soc/codecs/tas2770.c: In function tas2770_i2c_probe: sound/soc/codecs/tas2770.c:749:24: error: implicit declaration of function devm_gpiod_get_optional; did you mean devm_regulator_get_optional? [-Werror=implicit-function-declaration] tas2770->reset_gpio = devm_gpiod_get_optional(tas2770->dev, ^~~~~~~~~~~~~~~~~~~~~~~ devm_regulator_get_optional sound/soc/codecs/tas2770.c:751:13: error: GPIOD_OUT_HIGH undeclared (first use in this function); did you mean GPIOF_INIT_HIGH? GPIOD_OUT_HIGH); ^~~~~~~~~~~~~~ GPIOF_INIT_HIGH Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 1a476abc723e ("tas2770: add tas2770 smart PA kernel driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac957..d639f17 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1108,6 +1108,7 @@ config SND_SOC_TAS2552 config SND_SOC_TAS2770 tristate "Texas Instruments TAS2770 speaker amplifier" depends on I2C + select GPIOLIB config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" -- 2.7.4

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[alsa-devel] [PATCH v2 0/4] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 08 Oct '19

08 Oct '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the following patch: 1. this series add support reset-controller in infra [v5,2/2] clk: reset: Modify reset-controller driver https://patchwork.kernel.org/patch/11060419/ v2 changes: 1. remove "WIP" that in the title of patches 2. add hyper link for the patch that depends on Jiaxin Yu (2): dt-bindings: medaitek: mt8183: add property "resets" && "reset-names" ASoC: mt8183: fix audio playback slowly after playback during bootup yong.liang (2): dt-bindings: mediatek: mt8183: Add #reset-cells watchdog: mtk_wdt: mt8183: Add reset controller .../bindings/sound/mt8183-afe-pcm.txt | 6 + .../devicetree/bindings/watchdog/mtk-wdt.txt | 9 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 110 +++++++++++++++++- .../reset-controller/mt8183-resets.h | 13 +++ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 15 +++ 6 files changed, 150 insertions(+), 4 deletions(-) -- 2.18.0

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[alsa-devel] [PATCH 00/44] ASoC: don't use snd_pcm_ops
by Kuninori Morimoto 08 Oct '19

08 Oct '19

Hi Mark Current snd_soc_component_driver is using snd_pcm_ops. But, it is mainly for ALSA, thus, it not good much to ALSA SoC. For example .open case (1)(2), each callback can't know it is called for which component. Thus, it need to use using snd_soc_rtdcom_lookup() (3). --- ALSA SoC --- ... if (component->driver->ops && component->driver->ops->open) (1) return component->driver->ops->open(substream); ... --- driver --- (2) static int xxx_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; (3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..); ... } It works today, but, will not work in the future if we support multi CPU/Codec/Platform, because 1 rtd might have multiple components which have same driver name. In other words snd_soc_rtdcom_lookup() doesn't work correctly in the future. To solve this issue, each callback needs to be called with component. This patch copies each snd_pcm_ops member under component driver, and having "component" as parameter. --- ALSA SoC --- ... if (component->driver->open) => return component->driver->open(component, substream); ... --- driver --- => static int xxx_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { ... } Kuninori Morimoto (44): ASoC: soc-core: merge snd_pcm_ops member to component driver ASoC: soc-core: add new pcm_construct/pcm_destruct ASoC: soc-core: add snd_soc_pcm_lib_ioctl() ASoC: intel: sst-haswell-pcm: remove snd_pcm_ops ASoC: intel: skl-pcm: remove snd_pcm_ops ASoC: intel: sst-mfld-platform-pcm: remove snd_pcm_ops ASoC: intel: sst-baytrail-pcm: remove snd_pcm_ops ASoC: amd: acp3x-pcm-dma: remove snd_pcm_ops ASoC: amd: acp-pcm-dma: remove snd_pcm_ops ASoC: atmel: atmel-pcm-pdc: remove snd_pcm_ops ASoC: au1x: dbdma2: remove snd_pcm_ops ASoC: au1x: dma: remove snd_pcm_ops ASoC: xtensa: xtfpga-i2s: remove snd_pcm_ops ASoC: xilinx: xlnx_formatter_pcm: remove snd_pcm_ops ASoC: uniphier: aio-dma: remove snd_pcm_ops ASoC: txx9: txx9aclc: remove snd_pcm_ops ASoC: stm: stm32_adfsdm: remove snd_pcm_ops ASoC: sprd: sprd-pcm-dma: remove snd_pcm_ops ASoC: sof: pcm: remove snd_pcm_ops ASoC: sh: rsnd: remove snd_pcm_ops ASoC: sh: fsi: remove snd_pcm_ops ASoC: sh: dma-sh7760: remove snd_pcm_ops ASoC: sh: siu_pcm: remove snd_pcm_ops ASoC: samsung: idma: remove snd_pcm_ops ASoC: qcom: lpass-platform: remove snd_pcm_ops ASoC: qcom: q6routing: remove snd_pcm_ops ASoC: qcom: q6asm-dai: remove snd_pcm_ops ASoC: pxa: mmp-pcm: remove snd_pcm_ops ASoC: pxa: remove snd_pcm_ops ASoC: meson: remove snd_pcm_ops ASoC: mediatek: mtk-btcvsd: remove snd_pcm_ops ASoC: mediatek: remove snd_pcm_ops ASoC: kirkwood: kirkwood-dma: remove snd_pcm_ops ASoC: dwc: dwc-pcm: remove snd_pcm_ops ASoC: rt5514-spi: remove snd_pcm_ops ASoC: bcm: cygnus-pcm: remove snd_pcm_ops ASoC: fsl: imx-pcm-fiq: remove snd_pcm_ops ASoC: fsl: fsl_dma: remove snd_pcm_ops ASoC: fsl: mpc5200_dma: remove snd_pcm_ops ASoC: fsl: fsl_asrc_dma: remove snd_pcm_ops ASoC: fsl: fsl_dma: don't use snd_soc_rtdcom_lookup() ASoC: soc-generic-dmaengine-pcm: remove snd_pcm_ops ASoC: soc-utils: remove snd_pcm_ops ASoC: soc-component: remove snd_pcm_ops from component driver include/sound/pxa2xx-lib.h | 26 +++++- include/sound/soc-component.h | 41 ++++++++- include/sound/soc.h | 5 + sound/arm/pxa2xx-pcm-lib.c | 80 +++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 63 ++++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 54 +++++------ sound/soc/atmel/atmel-pcm-pdc.c | 48 +++++----- sound/soc/au1x/dbdma2.c | 64 +++++++------ sound/soc/au1x/dma.c | 63 +++++++------ sound/soc/bcm/cygnus-pcm.c | 56 +++++------ sound/soc/codecs/rt5514-spi.c | 41 +++++---- sound/soc/dwc/dwc-pcm.c | 48 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 56 +++++------ sound/soc/fsl/fsl_dma.c | 53 +++++------ sound/soc/fsl/imx-pcm-fiq.c | 56 ++++++----- sound/soc/fsl/mpc5200_dma.c | 51 ++++++----- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 30 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 52 +++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 62 +++++++------ sound/soc/intel/skylake/skl-pcm.c | 67 +++++++++----- sound/soc/kirkwood/kirkwood-dma.c | 50 +++++----- .../soc/mediatek/common/mtk-afe-platform-driver.c | 28 +++--- .../soc/mediatek/common/mtk-afe-platform-driver.h | 10 +- sound/soc/mediatek/common/mtk-btcvsd.c | 76 ++++++--------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++- sound/soc/meson/axg-fifo.c | 56 +++++------ sound/soc/meson/axg-fifo.h | 20 +++- sound/soc/meson/axg-frddr.c | 24 ++++- sound/soc/meson/axg-toddr.c | 24 ++++- sound/soc/pxa/mmp-pcm.c | 62 ++++++++----- sound/soc/pxa/pxa-ssp.c | 14 ++- sound/soc/pxa/pxa2xx-ac97.c | 14 ++- sound/soc/pxa/pxa2xx-i2s.c | 14 ++- sound/soc/pxa/pxa2xx-pcm.c | 14 ++- sound/soc/qcom/lpass-platform.c | 70 +++++++------- sound/soc/qcom/qdsp6/q6asm-dai.c | 71 +++++++------- sound/soc/qcom/qdsp6/q6routing.c | 14 +-- sound/soc/samsung/idma.c | 58 ++++++------ sound/soc/sh/dma-sh7760.c | 46 +++++----- sound/soc/sh/fsi.c | 31 ++++--- sound/soc/sh/rcar/core.c | 23 +++-- sound/soc/sh/siu_pcm.c | 44 +++++---- sound/soc/soc-component.c | 76 +++++++-------- sound/soc/soc-core.c | 12 +++ sound/soc/soc-generic-dmaengine-pcm.c | 102 ++++++++++----------- sound/soc/soc-pcm.c | 11 +-- sound/soc/soc-utils.c | 11 +-- sound/soc/sof/pcm.c | 88 +++++++++--------- sound/soc/sprd/sprd-pcm-dma.c | 74 +++++++-------- sound/soc/stm/stm32_adfsdm.c | 42 +++++---- sound/soc/txx9/txx9aclc.c | 48 +++++----- sound/soc/uniphier/aio-dma.c | 51 ++++++----- sound/soc/xilinx/xlnx_formatter_pcm.c | 54 +++++------ sound/soc/xtensa/xtfpga-i2s.c | 30 +++--- 55 files changed, 1311 insertions(+), 1089 deletions(-) -- 2.7.4

2 89

[alsa-devel] [PATCH v2] firmware: vpd: Add an interface to read VPD value
by Cheng-Yi Chiang 08 Oct '19

08 Oct '19

Add an interface for other driver to query VPD value. This will be used for ASoC machine driver to query calibration data stored in VPD for smart amplifier speaker resistor calibration. The example usage in ASoC machine driver is like: #define DSM_CALIB_KEY "dsm_calib" static int load_calibration_data(struct cml_card_private *ctx) { char *data = NULL; int ret; u32 value_len; /* Read calibration data from VPD. */ ret = vpd_attribute_read(1, DSM_CALIB_KEY, (u8 **)&data, &value_len); /* Parsing of this string...*/ } Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- Change from v1 to v2: - Use kmemdup to copy data. - Set value_len according to bin_attr.size. - Check return value of kmemdup. - Rename the function to vpd_attribute_read. - Add docstrings for the function. - Returns -ENOENT when the key is not found. - Use EXPORT_SYMBOL_GPL. Note: The user of this API is in ASoC machine driver cml_rt1011_rt5682. It is pending on the initial machine driver change https://patchwork.kernel.org/patch/11161145/ and the codec driver change to provide API to do calibration. https://patchwork.kernel.org/patch/11179237/ The draft patch of machine driver change which uses this API is at https://chromium-review.googlesource.com/c/chromiumos/third_party/kernel/+/… drivers/firmware/google/vpd.c | 31 ++++++++++++++++++++++ include/linux/firmware/google/google_vpd.h | 18 +++++++++++++ 2 files changed, 49 insertions(+) create mode 100644 include/linux/firmware/google/google_vpd.h diff --git a/drivers/firmware/google/vpd.c b/drivers/firmware/google/vpd.c index db0812263d46..c2be0e756402 100644 --- a/drivers/firmware/google/vpd.c +++ b/drivers/firmware/google/vpd.c @@ -65,6 +65,37 @@ static ssize_t vpd_attrib_read(struct file *filp, struct kobject *kobp, info->bin_attr.size); } +/** + * vpd_attribute_read - Read VPD value for a key. + * @ro: True for RO section. False for RW section. + * @key: A string for key. + * @value: Where to write the VPD value on success. The caller + * must free the memory. + * @value_len: The length of value in bytes. + * + * Returns 0 on success, -ENOENT when the key is not found, and + * -ENOMEM when failed to allocate memory for value. + */ +int vpd_attribute_read(bool ro, const char *key, + u8 **value, u32 *value_len) +{ + struct vpd_attrib_info *info; + struct vpd_section *sec = ro ? &ro_vpd : &rw_vpd; + + list_for_each_entry(info, &sec->attribs, list) { + if (strcmp(info->key, key) == 0) { + *value = kmemdup(info->value, info->bin_attr.size, + GFP_KERNEL); + if (!*value) + return -ENOMEM; + *value_len = info->bin_attr.size; + return 0; + } + } + return -ENOENT; +} +EXPORT_SYMBOL_GPL(vpd_attribute_read); + /* * vpd_section_check_key_name() * diff --git a/include/linux/firmware/google/google_vpd.h b/include/linux/firmware/google/google_vpd.h new file mode 100644 index 000000000000..4364eaa4e1e3 --- /dev/null +++ b/include/linux/firmware/google/google_vpd.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Google VPD interface. + * + * Copyright 2019 Google Inc. + */ + +/* Interface for reading VPD value on Chrome platform. */ + +#ifndef __GOOGLE_VPD_H +#define __GOOGLE_VPD_H + +#include <linux/types.h> + +int vpd_attribute_read(bool ro, const char *key, + u8 **value, u32 *value_len); + +#endif /* __GOOGLE_VPD_H */ -- 2.23.0.581.g78d2f28ef7-goog

2 2

[alsa-devel] [RFC PATCH 00/12] soundwire/SOF: updated interfaces, functional integration
by Pierre-Louis Bossart 08 Oct '19

08 Oct '19

This series builds on the 'soundwire: add Master device support, GreyBus style' RFC'. It provides enhancements to the stream callbacks, a split initialization. Most of the SOF patches were already submitted in an earlier RFC, and feedback on the parameters was taken into account. The main change here are the API changes with a split between ACPI scan, probe, startup steps. Known limits: a) Power management (regular suspend/resume and pm_runtime) is not supported for now as we need to run additional checks on hardware. This will be provided as a separate series. b) during validation checks on CML/ICL, initialization and playback/capture worked fine, but we observed a reproducible system freeze while doing load/unload tests, so likely an initialization missing or a leak to be fixed. Comments and feedback welcome. Pierre-Louis Bossart (7): ASoC: soc-acpi: add link_mask field ASoC: SOF: support alternate list of machines ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: Intel: add build support for SoundWire ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect Rander Wang (5): soundwire: intel: update stream callbacks for hwparams/free stream operations soundwire: intel: add prepare support in sdw dai driver soundwire: intel: add trigger support in sdw dai driver soundwire: intel: add sdw_stream_setup helper for .startup callback soundwire: intel: free all resources on hw_free() drivers/soundwire/intel.c | 181 ++++++++++++++++++- drivers/soundwire/intel_init.c | 2 +- include/linux/soundwire/sdw_intel.h | 40 ++++- include/sound/soc-acpi.h | 2 + include/sound/sof.h | 3 + include/sound/sof/dai-intel.h | 18 +- sound/soc/sof/intel/Kconfig | 23 +++ sound/soc/sof/intel/hda-loader.c | 9 + sound/soc/sof/intel/hda.c | 261 +++++++++++++++++++++++++++- sound/soc/sof/intel/hda.h | 36 ++++ 10 files changed, 546 insertions(+), 29 deletions(-) -- 2.20.1

2 14

[alsa-devel] [PATCH v2 1/3] ASoC: tas2770: Fix snd_soc_update_bits error handling
by Dan Murphy 07 Oct '19

07 Oct '19

According the documentation for snd_soc_update_bits the API will return a 1 if the update was successful with a value change, a 0 if the update was successful with no value change or a negative if the command just failed. So the value of return in the driver needs to be checked for being less then 0 or the caller may indicate failure when the value actually changed. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Missed one update_bit return issue. sound/soc/codecs/tas2770.c | 46 +++++++++++++++++++++++++------------- 1 file changed, 30 insertions(+), 16 deletions(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index dbbb21fe0548..361d0bba72b3 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -82,7 +82,8 @@ static int tas2770_codec_suspend(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) + + if (ret < 0) return ret; return 0; @@ -96,8 +97,9 @@ static int tas2770_codec_resume(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - if (ret) - return -EINVAL; + + if (ret < 0) + return ret; return 0; } @@ -149,7 +151,10 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, } end: - return ret; + if (ret < 0) + return ret; + + return 0; } static const struct snd_kcontrol_new isense_switch = @@ -199,7 +204,10 @@ static int tas2770_mute(struct snd_soc_dai *dai, int mute) TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -252,7 +260,10 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) tas2770->i_sense_slot); end: - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) @@ -344,9 +355,11 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) } end: - if (!ret) - tas2770->sampling_rate = samplerate; - return ret; + if (ret < 0) + return ret; + + tas2770->sampling_rate = samplerate; + return 0; } static int tas2770_hw_params(struct snd_pcm_substream *substream, @@ -401,7 +414,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_RX_MASK, asi_cfg_1); - if (ret) + if (ret < 0) return ret; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { @@ -426,7 +439,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_MASK, (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); - if (ret) + if (ret < 0) return ret; value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); @@ -472,12 +485,12 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_30_MASK, (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); - if (ret) + if (ret < 0) return ret; ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_RXS_MASK, (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); - if (ret) + if (ret < 0) return ret; switch (slot_width) { @@ -511,10 +524,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } - if (!ret) - tas2770->slot_width = slot_width; + if (ret < 0) + return ret; - return ret; + tas2770->slot_width = slot_width; + return 0; } static struct snd_soc_dai_ops tas2770_dai_ops = { -- 2.22.0.214.g8dca754b1e

2 5

[alsa-devel] [PATCH 0/2] [RFC] ASoC: topology: fix access beyond data
by Guennadi Liakhovetski 07 Oct '19

07 Oct '19

This is likely incomplete, I haven't reviewed the whole of the topology parsing code, and I haven't tested this yet - neither for actually being a problem without these patches nor for fixing them. This is a preview to give everybody a chance to showt out if I misunderstood something and there isn't actually a problem to be fixed. Thanks Guennadi Guennadi Liakhovetski (2): ASoC: topology: protect against accessing beyond loaded topology data ASoC: topology: don't access beyond topology data sound/soc/soc-topology.c | 37 +++++++++++++++++++++++++++++-------- 1 file changed, 29 insertions(+), 8 deletions(-) -- 1.9.3

2 5

[alsa-devel] what's the kernel policy WRT firmware parsing security?
by Guennadi Liakhovetski 07 Oct '19

07 Oct '19

Hi, I decided to have a look at whether the ALSA topology parsing is bullet proof against malformed topology files. It seems not quite to be the case. At least I seem to have found a possibility of crashing the kernel by a malformed topology file. I haven't tested it, so, maybe I'm wrong. In principle, firmware files can only be written by root, and if you have root access to the system, it's anyway doomed. Is this the approach and we aren't really trying to make topology parsing 100% safe, or do we want to fix any such possible parsing issues? Thanks Guennadi

2 2

[alsa-devel] [PATCH 0/1] ALSA: hda: add fallback to polling to hdac_bus_get_response()
by Kai Vehmanen 07 Oct '19

07 Oct '19

Hey all, while debugging issues with some Intel platforms related to display audio codec probe (see https://lists.freedesktop.org/archives/intel-gfx/2019-October/214621.html ), I found a discrepancy in behaviour between snd-hda-intel and SOF, despite using the same snd-hda-codec-hdmi as the codec driver. The specific problem I was debugging appears in a stress test (designed to uncover the above display driver issue) where driver-unload, s3-suspend, resume and driver-reload is done in a loop and repeated for hundreds of iterations. When using SOF, I would get occasional probe fail due to a missing HDA irq. The AZX snd_hda_intel driver nicely survives this test. The explanation seems to be differences in the hdac get_response() implementation. While the specific issue could be solved with other means, the git history shows a number of rare issues with HDA codecs where polling has helped. It would seem best to align the logic with the AZX driver implementation that has seen much more usage over the years. This will benefit SOF and any other users of the HDAC library. Kai Vehmanen (1): ALSA: hda: add fallback to polling to hdac_bus_get_response() sound/hda/hdac_controller.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) -- 2.17.1

2 3

[alsa-devel] [PATCH v2 RESEND] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 07 Oct '19

07 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v1 to v2: - Use audio_lock to protect audio clock. - Fix the patch title. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..749d8e4c535b 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,17 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + unsigned long flags; + + spin_lock_irqsave(&hdmi->audio_lock, flags); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + spin_unlock_irqrestore(&hdmi->audio_lock, flags); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2056,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

2 2

[alsa-devel] [PATCH 07/10] ASoC: jz4740: Use of_device_get_match_data()
by Stephen Boyd 07 Oct '19

07 Oct '19

This probe function is only called if the device is backed by a DT node, so switch this call to of_device_get_match_data() to reduce code size and simplify a bit. This also avoids needing to reference a potentially undefined variable because of_device_get_match_data() doesn't need to know anything beyond the struct device to find the match table. Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Paul Cercueil <paul(a)crapouillou.net> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Frank Rowand <frowand.list(a)gmail.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Stephen Boyd <swboyd(a)chromium.org> --- Please ack or pick for immediate merge so the last patch can be merged. sound/soc/jz4740/jz4740-i2s.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index 13408de34055..d2dab4d24b87 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -503,9 +503,8 @@ static int jz4740_i2s_dev_probe(struct platform_device *pdev) if (!i2s) return -ENOMEM; - match = of_match_device(jz4740_of_matches, &pdev->dev); - if (match) - i2s->version = (enum jz47xx_i2s_version)match->data; + i2s->version = + (enum jz47xx_i2s_version)of_device_get_match_data(&pdev->dev); mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); i2s->base = devm_ioremap_resource(&pdev->dev, mem); -- Sent by a computer through tubes

2 1

[alsa-devel] [PATCH] ASoC: core: Fix pcm code debugfs error
by Hans de Goede 07 Oct '19

07 Oct '19

We can have 2 dcpm-s with the same backend and frontend name (capture + playback pair), this causes the following debugfs error on Intel Bay Trail systems: [ 298.969049] debugfs: Directory 'SSP2-Codec' with parent 'Baytrail Audio Port' already present! This commit adds a ":playback" or ":capture" postfix to the debugfs dir name fixing this. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/soc-pcm.c | 13 ++++++++++--- 1 file changed, 10 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..2e723012eb52 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1146,6 +1146,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, { struct snd_soc_dpcm *dpcm; unsigned long flags; + char *name; /* only add new dpcms */ for_each_dpcm_be(fe, stream, dpcm) { @@ -1171,9 +1172,15 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, stream ? "<-" : "->", be->dai_link->name); #ifdef CONFIG_DEBUG_FS - dpcm->debugfs_state = debugfs_create_dir(be->dai_link->name, - fe->debugfs_dpcm_root); - debugfs_create_u32("state", 0644, dpcm->debugfs_state, &dpcm->state); + name = kasprintf(GFP_KERNEL, "%s:%s", be->dai_link->name, + stream ? "capture" : "playback"); + if (name) { + dpcm->debugfs_state = debugfs_create_dir(name, + fe->debugfs_dpcm_root); + debugfs_create_u32("state", 0644, dpcm->debugfs_state, + &dpcm->state); + kfree(name); + } #endif return 1; } -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_mqs: Fix error handling in probe
by Dan Carpenter 07 Oct '19

07 Oct '19

There are several problems in the error handling in fsl_mqs_probe(). 1) "ret" isn't initialized on some paths. GCC has a feature which warns about uninitialized variables but the code initializes "ret" to zero at the start of the function so the checking is turned off. 2) "gpr_np" is a pointer so initializing it to zero is confusing and generates a Sparse warning. 3) of_parse_phandle() doesn't return error pointers on error, it returns NULL. 4) If devm_snd_soc_register_component() fails then the function should free the "gpr_np". Fixes: 9e28f6532c61 ("ASoC: fsl_mqs: Add MQS component driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/fsl/fsl_mqs.c | 27 +++++++++++++++------------ 1 file changed, 15 insertions(+), 12 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1619a553514..f5f2f659bcbf 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -179,10 +179,10 @@ static const struct regmap_config fsl_mqs_regmap_config = { static int fsl_mqs_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; - struct device_node *gpr_np = 0; + struct device_node *gpr_np = NULL; struct fsl_mqs *mqs_priv; void __iomem *regs; - int ret = 0; + int ret; mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); if (!mqs_priv) @@ -199,17 +199,16 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (mqs_priv->use_gpr) { gpr_np = of_parse_phandle(np, "gpr", 0); - if (IS_ERR(gpr_np)) { + if (!gpr_np) { dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); - ret = PTR_ERR(gpr_np); - goto out; + return -EINVAL; } mqs_priv->regmap = syscon_node_to_regmap(gpr_np); if (IS_ERR(mqs_priv->regmap)) { dev_err(&pdev->dev, "failed to get gpr regmap\n"); ret = PTR_ERR(mqs_priv->regmap); - goto out; + goto err_free_gpr_np; } } else { regs = devm_platform_ioremap_resource(pdev, 0); @@ -230,7 +229,7 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (IS_ERR(mqs_priv->ipg)) { dev_err(&pdev->dev, "failed to get the clock: %ld\n", PTR_ERR(mqs_priv->ipg)); - goto out; + return PTR_ERR(mqs_priv->ipg); } } @@ -238,17 +237,21 @@ static int fsl_mqs_probe(struct platform_device *pdev) if (IS_ERR(mqs_priv->mclk)) { dev_err(&pdev->dev, "failed to get the clock: %ld\n", PTR_ERR(mqs_priv->mclk)); - goto out; + ret = PTR_ERR(mqs_priv->mclk); + goto err_free_gpr_np; } dev_set_drvdata(&pdev->dev, mqs_priv); pm_runtime_enable(&pdev->dev); - return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, &fsl_mqs_dai, 1); -out: - if (!IS_ERR(gpr_np)) - of_node_put(gpr_np); + if (ret) + goto err_free_gpr_np; + return 0; + +err_free_gpr_np: + of_node_put(gpr_np); return ret; } -- 2.20.1

2 1

[alsa-devel] [PATCH -next] ASoC: fsl_mqs: remove set but not used variable 'bclk'
by YueHaibing 07 Oct '19

07 Oct '19

Fixes gcc '-Wunused-but-set-variable' warning: sound/soc/fsl/fsl_mqs.c: In function fsl_mqs_hw_params: sound/soc/fsl/fsl_mqs.c:54:6: warning: variable bclk set but not used [-Wunused-but-set-variable] It is never used, so can be removed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/fsl_mqs.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index c1619a5..7b9cab3 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -51,10 +51,9 @@ static int fsl_mqs_hw_params(struct snd_pcm_substream *substream, struct fsl_mqs *mqs_priv = snd_soc_component_get_drvdata(component); unsigned long mclk_rate; int div, res; - int bclk, lrclk; + int lrclk; mclk_rate = clk_get_rate(mqs_priv->mclk); - bclk = snd_soc_params_to_bclk(params); lrclk = params_rate(params); /* -- 2.7.4

2 1

[alsa-devel] tegra30 tdm audio support
by Ben Dooks 07 Oct '19

07 Oct '19

This adds support for tegra30 tdm audio and fixes a bug in the fifo handling on rx when non-32 bit aligned data is used. v2: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock v3: - cleanup fsync patch - fix rebase issue with tdm patch

3 19

[alsa-devel] [PATCH] ALSA: hdac: clear link output stream mapping
by Pierre-Louis Bossart 07 Oct '19

07 Oct '19

From: Rander Wang <rander.wang(a)linux.intel.com> Fix potential DMA hang upon starting playback on devices in HDA mode on Intel platforms (Gemini Lake/Whiskey Lake/Comet Lake/Ice Lake). It doesn't affect platforms before Gemini Lake or any Intel device in non-HDA mode. The reset value for the LOSDIV register is all output streams valid. Clear this register to invalidate non-existent streams when the bus is powered up. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/hda_register.h | 3 +++ sound/hda/ext/hdac_ext_controller.c | 5 +++++ 2 files changed, 8 insertions(+) diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 0fd39295b426..057d2a2d0bd0 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -264,6 +264,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_ML_LOUTPAY 0x20 #define AZX_REG_ML_LINPAY 0x30 +/* bit0 is reserved, with BIT(1) mapping to stream1 */ +#define ML_LOSIDV_STREAM_MASK 0xFFFE + #define ML_LCTL_SCF_MASK 0xF #define AZX_MLCTL_SPA (0x1 << 16) #define AZX_MLCTL_CPA (0x1 << 23) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 211ca85acd8c..cfab60d88c92 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -270,6 +270,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, ret = snd_hdac_ext_bus_link_power_up(link); + /* + * clear the register to invalidate all the output streams + */ + snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, + ML_LOSIDV_STREAM_MASK, 0); /* * wait for 521usec for codec to report status * HDA spec section 4.3 - Codec Discovery -- 2.20.1

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360
by Kai-Heng Feng 07 Oct '19

07 Oct '19

Headphone on XPS 9350/9360 produces a background white noise. The The noise level somehow correlates with "Headphone Mic Boost", when it sets to 1 the noise disappears. However, doing this has a side effect, which also decreases the overall headphone volume so I didn't send the patch upstream. The noise was bearable back then, but after commit 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") the noise exacerbates to a point it starts hurting ears. So let's use the workaround to set "Headphone Mic Boost" to 1 and lock it so it's not touchable by userspace. Fixes: 717f43d81afc ("ALSA: hda/realtek - Update headset mode for ALC256") BugLink: https://bugs.launchpad.net/bugs/1654448 BugLink: https://bugs.launchpad.net/bugs/1845810 Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 24 +++++++++++++++++++++--- 1 file changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b000b36ac3c6..b5c225a56b98 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5358,6 +5358,17 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec, } } +static void alc256_fixup_dell_xps_13_headphone_noise2(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_INPUT, 0, HDA_AMP_VOLMASK, 1); + snd_hda_override_wcaps(codec, 0x1a, get_wcaps(codec, 0x1a) & ~AC_WCAP_IN_AMP); +} + static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -5822,6 +5833,7 @@ enum { ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, ALC275_FIXUP_DELL_XPS, ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE, + ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2, ALC293_FIXUP_LENOVO_SPK_NOISE, ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, @@ -6558,6 +6570,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_dell_xps_13_headphone_noise2, + .chained = true, + .chain_id = ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE + }, [ALC293_FIXUP_LENOVO_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -7001,17 +7019,17 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x0706, "Dell Inspiron 7559", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x0738, "Dell Precision 5820", ALC269_FIXUP_NO_SHUTUP), - SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x075c, "Dell XPS 27 7760", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE2), SND_PCI_QUIRK(0x1028, 0x084b, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x084e, "Dell", ALC274_FIXUP_DELL_AIO_LINEOUT_VERB), SND_PCI_QUIRK(0x1028, 0x0871, "Dell Precision 3630", ALC255_FIXUP_DELL_HEADSET_MIC), -- 2.17.1

2 1

[alsa-devel] [RESEND TRIVIAL] sound: Fix Kconfig indentation
by Krzysztof Kozlowski 07 Oct '19

07 Oct '19

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> Acked-by: Mark Brown <broonie(a)kernel.org> --- sound/core/Kconfig | 28 ++++++++++---------- sound/drivers/Kconfig | 20 +++++++------- sound/firewire/Kconfig | 6 ++--- sound/isa/Kconfig | 18 ++++++------- sound/mips/Kconfig | 12 ++++----- sound/pci/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 14 +++++----- sound/soc/codecs/Kconfig | 32 +++++++++++------------ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 48 +++++++++++++++++----------------- sound/soc/pxa/Kconfig | 16 ++++++------ sound/soc/qcom/Kconfig | 20 +++++++------- sound/soc/samsung/Kconfig | 8 +++--- sound/soc/sof/imx/Kconfig | 12 ++++----- sound/soc/sof/intel/Kconfig | 22 ++++++++-------- sound/soc/xilinx/Kconfig | 20 +++++++------- sound/soc/zte/Kconfig | 12 ++++----- sound/usb/Kconfig | 32 +++++++++++------------ 18 files changed, 162 insertions(+), 162 deletions(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4ee79ad6ae22..4044c42d8595 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -72,11 +72,11 @@ config SND_PCM_OSS config SND_PCM_OSS_PLUGINS bool "OSS PCM (digital audio) API - Include plugin system" depends on SND_PCM_OSS - default y + default y help - If you disable this option, the ALSA's OSS PCM API will not - support conversion of channels, formats and rates. It will - behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. config SND_PCM_TIMER bool "PCM timer interface" if EXPERT @@ -128,13 +128,13 @@ config SND_SUPPORT_OLD_API or older). config SND_PROC_FS - bool "Sound Proc FS Support" if EXPERT - depends on PROC_FS - default y - help - Say 'N' to disable Sound proc FS, which may reduce code size about - 9KB on x86_64 platform. - If unsure say Y. + bool "Sound Proc FS Support" if EXPERT + depends on PROC_FS + default y + help + Say 'N' to disable Sound proc FS, which may reduce code size about + 9KB on x86_64 platform. + If unsure say Y. config SND_VERBOSE_PROCFS bool "Verbose procfs contents" @@ -142,8 +142,8 @@ config SND_VERBOSE_PROCFS default y help Say Y here to include code for verbose procfs contents (provides - useful information to developers when a problem occurs). On the - other side, it makes the ALSA subsystem larger. + useful information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. config SND_VERBOSE_PRINTK bool "Verbose printk" @@ -164,7 +164,7 @@ config SND_DEBUG_VERBOSE depends on SND_DEBUG help Say Y here to enable extra-verbose debugging messages. - + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. So, say Y only if you are ready to be annoyed. diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 09932cc98e9d..15d6d46acf9c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART - tristate - select SND_RAWMIDI + tristate + select SND_RAWMIDI config SND_OPL3_LIB tristate @@ -90,16 +90,16 @@ config SND_DUMMY will be called snd-dummy. config SND_ALOOP - tristate "Generic loopback driver (PCM)" - select SND_PCM - help - Say 'Y' or 'M' to include support for the PCM loopback device. + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. This module returns played samples back to the user space using the standard ALSA PCM device. The devices are routed 0->1 and - 1->0, where first number is the playback PCM device and second + 1->0, where first number is the playback PCM device and second number is the capture device. Module creates two PCM devices and configured number of substreams (see the pcm_substreams module - parameter). + parameter). The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% @@ -142,12 +142,12 @@ config SND_MTS64 select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with - additional SMPTE Timecode capabilities for the parallel port. + additional SMPTE Timecode capabilities for the parallel port. Say 'Y' to include support for this device. To compile this driver as a module, chose 'M' here: the module - will be called snd-mts64. + will be called snd-mts64. config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index b0a904cdb932..995c2cefc222 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -77,7 +77,7 @@ config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB select SND_HWDEP - help + help Say Y here to include support for FireWire devices based on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware: * Edirol FA-66/FA-101 @@ -111,8 +111,8 @@ config SND_BEBOB * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro - To compile this driver as a module, choose M here: the module - will be called snd-bebob. + To compile this driver as a module, choose M here: the module + will be called snd-bebob. config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..6ffa48dd5983 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -2,22 +2,22 @@ # ALSA ISA drivers config SND_WSS_LIB - tristate - select SND_PCM + tristate + select SND_PCM select SND_TIMER config SND_SB_COMMON - tristate + tristate config SND_SB8_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON config SND_SB16_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON menuconfig SND_ISA bool "ISA sound devices" diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 8a33402fd415..b497b803c834 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -14,15 +14,15 @@ config SND_SGI_O2 tristate "SGI O2 Audio" depends on SGI_IP32 select SND_PCM - help - Sound support for the SGI O2 Workstation. + help + Sound support for the SGI O2 Workstation. config SND_SGI_HAL2 - tristate "SGI HAL2 Audio" - depends on SGI_HAS_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 select SND_PCM - help - Sound support for the SGI Indy and Indigo2 Workstation. + help + Sound support for the SGI Indy and Indigo2 Workstation. endif # SND_MIPS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..93bc9bef7641 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -217,7 +217,7 @@ config SND_CMIPCI will be called snd-cmipci. config SND_OXYGEN_LIB - tristate + tristate config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2333efac758a..8039a8febefa 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -33,13 +33,13 @@ config SND_EP93XX_SOC_AC97 select SND_SOC_AC97_BUS config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. config SND_EP93XX_SOC_SIMONE tristate "SoC Audio support for Simplemachines Sim.One board" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ed8ba18d93d3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -258,16 +258,16 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - help - Normally ASoC codec drivers are only built if a machine driver which - uses them is also built since they are only usable with a machine - driver. Selecting this option will allow these drivers to be built - without an explicit machine driver for test and development purposes. + help + Normally ASoC codec drivers are only built if a machine driver which + uses them is also built since they are only usable with a machine + driver. Selecting this option will allow these drivers to be built + without an explicit machine driver for test and development purposes. Support for the bus types used to access the codecs to be built must be selected separately. - If unsure select "N". + If unsure select "N". config SND_SOC_88PM860X tristate @@ -571,8 +571,8 @@ config SND_SOC_CS42XX8_I2C # Cirrus Logic CS43130 HiFi DAC config SND_SOC_CS43130 - tristate "Cirrus Logic CS43130 CODEC" - depends on I2C + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" @@ -644,19 +644,19 @@ config SND_SOC_L3 tristate config SND_SOC_DA7210 - tristate + tristate config SND_SOC_DA7213 - tristate + tristate config SND_SOC_DA7218 tristate config SND_SOC_DA7219 - tristate + tristate config SND_SOC_DA732X - tristate + tristate config SND_SOC_DA9055 tristate @@ -718,7 +718,7 @@ config SND_SOC_INNO_RK3036 select REGMAP_MMIO config SND_SOC_ISABELLE - tristate + tristate config SND_SOC_LM49453 tristate @@ -989,7 +989,7 @@ config SND_SOC_RT5640 tristate config SND_SOC_RT5645 - tristate + tristate config SND_SOC_RT5651 tristate @@ -1225,7 +1225,7 @@ config SND_SOC_UDA134X tristate config SND_SOC_UDA1380 - tristate + tristate depends on I2C config SND_SOC_WCD9335 @@ -1353,7 +1353,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate config SND_SOC_WM8955 tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..597516c0c192 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -113,7 +113,7 @@ config SND_SOC_INTEL_SKYLAKE select SND_SOC_INTEL_CNL select SND_SOC_INTEL_CFL help - This is a backwards-compatible option to select all devices + This is a backwards-compatible option to select all devices supported by the Intel SST/Skylake driver. This option is no longer recommended and will be deprecated when the SOF driver is introduced. Distributions should explicitly diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7ab4a5f..882ff36a7c9c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -3,13 +3,13 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help - Intel ASoC Machine Drivers. If you have a Intel machine that - has an audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y - Note that the answer to this question doesn't directly affect the - kernel: saying N will just cause the configurator to skip all - the questions about Intel ASoC machine drivers. + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH @@ -114,11 +114,11 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 - help - This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5672 audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" @@ -311,20 +311,20 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + depends on SPI + select SND_SOC_RT5663 + select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI + select SND_SOC_MAX98927 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5663 + RT5514 + MAX98927. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" @@ -393,7 +393,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 213d4dab0346..295cfffa4646 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -190,14 +190,14 @@ config SND_PXA2XX_SOC_MAGICIAN HTC Magician. config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" @@ -205,7 +205,7 @@ config SND_PXA2XX_SOC_IMOTE2 select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help - Say Y if you want to add support for SoC audio on the + Say Y if you want to add support for SoC audio on the IMote 2. config SND_MMP_SOC_BROWNSTONE diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 60086858e920..6530d2462a9e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -3,8 +3,8 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help - Say Y or M if you want to add support to use audio devices - in Qualcomm Technologies SOC-based platforms. + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. config SND_SOC_LPASS_CPU tristate @@ -30,17 +30,17 @@ config SND_SOC_STORM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help - Say Y or M if you want add support for SoC audio on the - Qualcomm Technologies IPQ806X-based Storm board. + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help - Support for Qualcomm Technologies LPASS audio block in - APQ8016 SOC-based systems. - Say Y if you want to use audio devices on MI2S. + Support for Qualcomm Technologies LPASS audio block in + APQ8016 SOC-based systems. + Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON tristate @@ -93,9 +93,9 @@ config SND_SOC_MSM8996 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help - Support for Qualcomm Technologies LPASS audio block in - APQ8096 SoC-based systems. - Say Y if you want to use audio device on this SoCs + Support for Qualcomm Technologies LPASS audio block in + APQ8096 SoC-based systems. + Say Y if you want to use audio device on this SoCs config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..9304177de78a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -195,10 +195,10 @@ config SND_SOC_ODROID Say Y here to enable audio support for the Odroid XU3/XU4. config SND_SOC_ARNDALE_RT5631_ALC5631 - tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_RT5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on I2C + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..6315fba8ce71 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -5,9 +5,9 @@ config SND_SOC_SOF_IMX_TOPLEVEL depends on ARM64|| COMPILE_TEST depends on SND_SOC_SOF_OF help - This adds support for Sound Open Firmware for NXP i.MX platforms. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX platforms. + Say Y if you have such a device. + If unsure select "N". if SND_SOC_SOF_IMX_TOPLEVEL @@ -16,8 +16,8 @@ config SND_SOC_SOF_IMX8 depends on IMX_SCU depends on IMX_DSP help - This adds support for Sound Open Firmware for NXP i.MX8 platforms - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX8 platforms + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index d62f51d33be1..56f16e4543c0 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_PCI config SND_SOC_SOF_INTEL_HIFI_EP_IPC tristate help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_INTEL_ATOM_HIFI_EP @@ -217,31 +217,31 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the Tigerlake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tigerlake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_ELKHARTLAKE_SUPPORT bool "SOF support for ElkhartLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the ElkhartLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the ElkhartLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 69973179ef15..1d3586b68db7 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,15 +9,15 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" - help - Select this option to enable Xilinx audio formatter - support. This provides DMA platform device support for - audio functionality. + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" - help - Select this option to enable Xilinx SPDIF Audio. - This provides playback and capture of SPDIF audio in - AES format. + tristate "Audio support for the the Xilinx SPDIF" + help + Select this option to enable Xilinx SPDIF Audio. + This provides playback and capture of SPDIF audio in + AES format. diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a7842e4b791c..a23d4f13ca19 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -18,9 +18,9 @@ config ZX_I2S ZTE ZX I2S interface config ZX_TDM - tristate "ZTE ZX TDM Driver Support" - depends on COMMON_CLK - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to the - ZTE ZX TDM interface + tristate "ZTE ZX TDM Driver Support" + depends on COMMON_CLK + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to the + ZTE ZX TDM interface diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e2c53a0841da..059242f15d75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -107,24 +107,24 @@ config SND_USB_US122L will be called snd-usb-us122l. config SND_USB_6FIRE - tristate "TerraTec DMX 6Fire USB" - select FW_LOADER - select BITREVERSE - select SND_RAWMIDI - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for TerraTec 6fire DMX USB interface. - - You will need firmware files in order to be able to use the device - after it has been coldstarted. An install script for the firmware - and further help can be found at - http://sixfireusb.sourceforge.net + tristate "TerraTec DMX 6Fire USB" + select FW_LOADER + select BITREVERSE + select SND_RAWMIDI + select SND_PCM + select SND_VMASTER + help + Say Y here to include support for TerraTec 6fire DMX USB interface. + + You will need firmware files in order to be able to use the device + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net config SND_USB_HIFACE - tristate "M2Tech hiFace USB-SPDIF driver" - select SND_PCM - help + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help Select this option to include support for M2Tech hiFace USB-SPDIF interface. -- 2.17.1

2 1

[alsa-devel] [PATCH v2] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 06 Oct '19

06 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems like proposed in the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 34 ++++++++++ sound/hda/Kconfig | 10 ++- sound/hda/Makefile | 7 ++- sound/hda/intel-dsp-config.c | 105 +++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +--- sound/pci/hda/hda_intel.c | 48 +++++--------- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 ++---- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 ++ 11 files changed, 179 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..c36622bee3f8 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,34 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..f331e2216fe0 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,105 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/module.h> +#include <linux/pci.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=noDSP, 2=legacy, 3=SST, 4=SOF)"); + +static const u16 sof_skl_table[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + 0x02c8, /* Cometlake-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + 0x06c8, /* Cometlake-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + 0x3198, /* Geminilake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + 0x9dc8, /* Cannonlake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + 0xa348, /* Coffelake */ +#endif +}; + +static int snd_intel_dsp_check_device(u16 device, const u16 *table, u32 len) +{ + for (; len > 0; len--, table++) { + if (*table == device) + return 1; + } + return 0; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + if (snd_intel_dsp_check_device(pci->device, sof_skl_table, ARRAY_SIZE(sof_skl_table))) { + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_LEGACY; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* Prefer SST driver for Broxton-P (Appololake) */ +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) || IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + if (pci->device == 0x5a98) +#if IS_ENABLED(CONFIG_SND_SOC_INTEL_APL) + return SND_INTEL_DSP_DRIVER_SST; +#else + return SND_INTEL_DSP_DRIVER_SOF; +#endif +#endif + + /* DMIC check for Skylake+ */ + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + + return SND_INTEL_DSP_DRIVER_ANY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..7e262678abeb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Force this driver to be used and bypass the DSP driver selection"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,16 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2060,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

2 2

[alsa-devel] [PATCH -next] ASoC: tas2770: fix platform_no_drv_owner.cocci warnings
by YueHaibing 06 Oct '19

06 Oct '19

Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/tas2770.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index 9da88ccb1d51..dbbb21fe0548 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -793,7 +793,6 @@ MODULE_DEVICE_TABLE(of, tas2770_of_match); static struct i2c_driver tas2770_i2c_driver = { .driver = { .name = "tas2770", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tas2770_of_match), }, .probe = tas2770_i2c_probe,

1 0

[alsa-devel] [PATCH 00/10] ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: support WoV
by Tzung-Bi Shih 05 Oct '19

05 Oct '19

This series makes mt6358, cros_ec_codec, and mt8183-mt6358-ts3a227-max98357 support WoV (wake on voice). The first 3 commits are some cleanups and refactors. It looks like breaking the existing interface. But please be noticed that, the cros_ec_codec has not used by any real device yet. The refactor is very necessary to keep the style consistent and for easier to further extend and maintain. platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX The 4th commit extends the feature offered from EC codec. platform/chrome: cros_ec: add common commands for EC codec The 5th commit changes the behavior of setting and getting DMIC gains. ASoC: cros_ec_codec: read max DMIC gain from EC codec The 6th and 7th commit make cros_ec_codec support WoV. ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV The 8th commit sets necessary registers on mt6358 to support WoV. ASoC: mediatek: mt6358: support WoV The last 2 commit make machine driver mt8183-mt6358-ts3a227-max98357 support WoV if ec-codec is in DTS. ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV Tzung-Bi Shih (10): platform/chrome: cros_ec: remove unused EC feature ASoC: cros_ec_codec: refactor I2S RX ASoC: cros_ec_codec: extract DMIC EC command from I2S RX platform/chrome: cros_ec: add common commands for EC codec ASoC: cros_ec_codec: read max DMIC gain from EC codec ASoC: dt-bindings: cros_ec_codec: add SHM bindings ASoC: cros_ec_codec: support WoV ASoC: mediatek: mt6358: support WoV ASoC: dt-bindings: mt8183: add ec-codec ASoC: mediatek: mt8183: support WoV .../bindings/sound/google,cros-ec-codec.txt | 24 +- .../sound/mt8183-mt6358-ts3a227-max98357.txt | 3 + drivers/platform/chrome/cros_ec_trace.c | 5 +- .../linux/platform_data/cros_ec_commands.h | 285 ++++- sound/soc/codecs/cros_ec_codec.c | 1125 +++++++++++++---- sound/soc/codecs/mt6358.c | 105 ++ sound/soc/mediatek/Kconfig | 1 + .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 70 +- 8 files changed, 1292 insertions(+), 326 deletions(-) -- 2.23.0.581.g78d2f28ef7-goog

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[alsa-devel] alsa compliance test: H340 (USB audio) playback / capture rate asymmetry bug?
by Hin-Tak Leung 05 Oct '19

05 Oct '19

I am using Takashi Iwai's tree grafted onto mainline as DKMS ( as in https://github.com/HinTak/sound-usb-dkms ). Running the alsa compliance test commit f6167eb77d038b5b7a0d39645e7f2ae7fee6fdc0 (origin/stabilize-12464.B, origin/release-R78-12499.B) Capture all passed, but play back failed a couple, regarding the sample rate. It is a small head set with stereo head phone and a mic. ID 046d:0a38 Logitech, Inc. Headset H340 The question is, for such a cheap headset, why would the playback rate and the capture rate to be different? For many applications/ usages, with a device that's most capture and playback capable, we would like the rates to agree - both pass, or both fail in the same direction? Is this a hardware or software issue? Or, somebody suggested, I haven't looked, issue with the alsa compliance test itself, possibly regarding frame counts of usb devices? detail below. 5 passed, 0 failed Device Information Name: hw:CARD=H340 Stream: CAPTURE Format: ['S16_LE'] Channels range: [2, 2] Rate: [44100] Period_size range: [45, 131072] Buffer_size range: [90, 262144] Test Params Set channels 2: pass Set format S16_LE: pass Set rate 44100: pass Test Rates Set rate 44100: pass Test All Pairs Set channels 2, format S16_LE, rate 44100: pass 3 passed, 2 failed Device Information Name: hw:CARD=H340 Stream: PLAYBACK Format: ['S16_LE'] Channels range: [2, 2] Rate: [44100] Period_size range: [45, 131072] Buffer_size range: [90, 262144] Test Params Set channels 2: pass Set format S16_LE: pass Set rate 44100: pass Test Rates Set rate 44100: fail - Expected rate is 44100.000000, measure 44092.918362, difference 7.081638 > threshold 4.410000 Test All Pairs Set channels 2, format S16_LE, rate 44100: fail - Expected rate is 44100.000000, measure 44093.049192, difference 6.950808 > threshold 4.410000

4 7

[alsa-devel] [RFC PATCH 0/2] Alter the trigger order for FE/BE DAI's based on command
by Ranjani Sridharan 04 Oct '19

04 Oct '19

Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST determine the order in which FE DAI and BE DAI are triggered. In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is triggered before the BE DAI and in the case of SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before the FE DAI. And this order remains the same irrespective of the trigger command. In the case of the SOF driver, during playback, the FW expects the BE DAI to be triggered before the FE DAI during the START trigger. The BE DAI trigger handles the starting of Link DMA and so it must be started before the FE DAI is started to prevent xruns during pause/release. This can be addressed by setting the trigger order for the FE dai link to SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger, the FW expects the FE DAI to be triggered before the BE DAI. Retaining the same order during the START and STOP commands, results in FW error as the DAI component in the FW is still active. The issue can be fixed by mirroring the trigger order of FE and BE DAI's during the START and STOP trigger. So, with the trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME and the BE DAI will be triggered first during the STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands and the FE DAI will be triggered first during the SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands. My first thought was to use a BESPOKE trigger for the SOF driver but looking more into the implementation of the bespoke trigger in soc_pcm_bespoke_trigger() didnt indicate it had much to do with the ordering of the FE/BE DAI's but rather just use the bespoke trigger callbacks in the DAI driver. More details on the SOF issue can be found in: Github Issue: https://github.com/thesofproject/linux/issues/1160 Ranjani Sridharan (2): ASoC: pcm: update FE/BE trigger order based on the command ASoC: SOF: topology: set trigger order for FE DAI link sound/soc/soc-pcm.c | 99 +++++++++++++++++++++++++++++----------- sound/soc/sof/topology.c | 4 ++ 2 files changed, 76 insertions(+), 27 deletions(-) -- 2.17.1

3 5

[alsa-devel] [PATCH] ASoC: tas2770: Fix snd_soc_update_bits error handling
by Dan Murphy 04 Oct '19

04 Oct '19

According the documentation for snd_soc_update_bits the API will return a 1 if the update was successful with a value change, a 0 if the update was successful with no value change or a negative if the command just failed. So the value of return in the driver needs to be checked for being less then 0 or the caller may indicate failure when the value actually changed. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- arch/arm/configs/omap2plus_defconfig | 1 + sound/soc/codecs/tas2770.c | 44 ++++++++++++++++++---------- 2 files changed, 30 insertions(+), 15 deletions(-) diff --git a/arch/arm/configs/omap2plus_defconfig b/arch/arm/configs/omap2plus_defconfig index 64eb896907bf..8a57185112a2 100644 --- a/arch/arm/configs/omap2plus_defconfig +++ b/arch/arm/configs/omap2plus_defconfig @@ -395,6 +395,7 @@ CONFIG_SND_SOC_OMAP_ABE_TWL6040=m CONFIG_SND_SOC_OMAP_HDMI=m CONFIG_SND_SOC_CPCAP=m CONFIG_SND_SOC_TLV320AIC23_I2C=m +CONFIG_SND_SOC_TAS2770=m CONFIG_SND_SIMPLE_CARD=m CONFIG_SND_AUDIO_GRAPH_CARD=m CONFIG_HID_GENERIC=m diff --git a/sound/soc/codecs/tas2770.c b/sound/soc/codecs/tas2770.c index dbbb21fe0548..774c8f2833c8 100644 --- a/sound/soc/codecs/tas2770.c +++ b/sound/soc/codecs/tas2770.c @@ -82,7 +82,8 @@ static int tas2770_codec_suspend(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_SHUTDOWN); - if (ret) + + if (ret < 0) return ret; return 0; @@ -96,8 +97,9 @@ static int tas2770_codec_resume(struct snd_soc_component *component) TAS2770_PWR_CTRL, TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - if (ret) - return -EINVAL; + + if (ret < 0) + return ret; return 0; } @@ -149,7 +151,10 @@ static int tas2770_dac_event(struct snd_soc_dapm_widget *w, } end: - return ret; + if (ret < 0) + return ret; + + return 0; } static const struct snd_kcontrol_new isense_switch = @@ -199,7 +204,10 @@ static int tas2770_mute(struct snd_soc_dai *dai, int mute) TAS2770_PWR_CTRL_MASK, TAS2770_PWR_CTRL_ACTIVE); - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) @@ -252,7 +260,10 @@ static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) tas2770->i_sense_slot); end: - return ret; + if (ret < 0) + return ret; + + return 0; } static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) @@ -344,9 +355,11 @@ static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) } end: - if (!ret) - tas2770->sampling_rate = samplerate; - return ret; + if (ret < 0) + return ret; + + tas2770->sampling_rate = samplerate; + return 0; } static int tas2770_hw_params(struct snd_pcm_substream *substream, @@ -426,7 +439,7 @@ static int tas2770_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG1, TAS2770_TDM_CFG_REG1_MASK, (tdm_rx_start_slot << TAS2770_TDM_CFG_REG1_51_SHIFT)); - if (ret) + if (ret < 0) return ret; value = snd_soc_component_read32(component, TAS2770_TDM_CFG_REG3); @@ -472,12 +485,12 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_30_MASK, (left_slot << TAS2770_TDM_CFG_REG3_30_SHIFT)); - if (ret) + if (ret < 0) return ret; ret = snd_soc_component_update_bits(component, TAS2770_TDM_CFG_REG3, TAS2770_TDM_CFG_REG3_RXS_MASK, (right_slot << TAS2770_TDM_CFG_REG3_RXS_SHIFT)); - if (ret) + if (ret < 0) return ret; switch (slot_width) { @@ -511,10 +524,11 @@ static int tas2770_set_dai_tdm_slot(struct snd_soc_dai *dai, ret = -EINVAL; } - if (!ret) - tas2770->slot_width = slot_width; + if (ret < 0) + return ret; - return ret; + tas2770->slot_width = slot_width; + return 0; } static struct snd_soc_dai_ops tas2770_dai_ops = { -- 2.22.0.214.g8dca754b1e

2 2

[alsa-devel] [PATCH] ASoC: wm_adsp: Don't generate kcontrols without READ flags
by Charles Keepax 04 Oct '19

04 Oct '19

From: Stuart Henderson <stuarth(a)opensource.cirrus.com> User space always expects to be able to read ALSA controls, so ensure no kcontrols are generated without an appropriate READ flag. In the case of a read of such a control zeros will be returned. Signed-off-by: Stuart Henderson <stuarth(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 85396d920e0a9..9b8bb7bbe945d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1259,8 +1259,7 @@ static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) } if (in) { - if (in & WMFW_CTL_FLAG_READABLE) - out |= rd; + out |= rd; if (in & WMFW_CTL_FLAG_WRITEABLE) out |= wr; if (in & WMFW_CTL_FLAG_VOLATILE) -- 2.11.0

2 1

[alsa-devel] [PATCH v4 1/3] ASoC: samsung: arndale: Simplify DAI link initialization
by Sylwester Nawrocki 04 Oct '19

04 Oct '19

There is only one DAI link so we can drop an unnecessary loop statement. Use card->dai_link in place of direct static arndale_rt5631_dai[] array dereference as a prerequisite for adding support for other CODECs. Unnecessary assignment of dai_link->codecs->name to NULL is removed. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Based on for-next branch from git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound Changes since v2: - none. Changes since v1: - rebased due to reordering patches in the series --- sound/soc/samsung/arndale_rt5631.c | 42 +++++++++++++----------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index fd8c6642fb0d..004c84fafad9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -88,36 +88,30 @@ static void arndale_put_of_nodes(struct snd_soc_card *card) static int arndale_audio_probe(struct platform_device *pdev) { - int n, ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &arndale_rt5631; + struct snd_soc_dai_link *dai_link; + int ret; card->dev = &pdev->dev; + dai_link = card->dai_link; + + dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0); + if (!dai_link->cpus->of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } - for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { - if (!arndale_rt5631_dai[n].cpus->dai_name) { - arndale_rt5631_dai[n].cpus->of_node = of_parse_phandle(np, - "samsung,audio-cpu", n); - - if (!arndale_rt5631_dai[n].cpus->of_node) { - dev_err(&pdev->dev, - "Property 'samsung,audio-cpu' missing or invalid\n"); - return -EINVAL; - } - } - if (!arndale_rt5631_dai[n].platforms->name) - arndale_rt5631_dai[n].platforms->of_node = - arndale_rt5631_dai[n].cpus->of_node; - - arndale_rt5631_dai[n].codecs->name = NULL; - arndale_rt5631_dai[n].codecs->of_node = of_parse_phandle(np, - "samsung,audio-codec", n); - if (!arndale_rt5631_dai[0].codecs->of_node) { - dev_err(&pdev->dev, + if (!dai_link->platforms->name) + dai_link->platforms->of_node = dai_link->cpus->of_node; + + dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0); + if (!dai_link->codecs->of_node) { + dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err_put_of_nodes; - } + ret = -EINVAL; + goto err_put_of_nodes; } ret = devm_snd_soc_register_card(card->dev, card); -- 2.17.1

2 5

[alsa-devel] [asoc:for-5.5 32/32] sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 04 Oct '19

04 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 1a476abc723e644248dba975b71122fcf878703b [32/32] tas2770: add tas2770 smart PA kernel driver If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> coccinelle warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2770.c:796:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 2

[alsa-devel] [PATCH 0/6][resend] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 04 Oct '19

04 Oct '19

Hi Mark Some soc-core cleanup patch step3 are accepted, but not all. These are remains. Kuninori Morimoto (6): ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() sound/soc/soc-core.c | 109 +++++++++++++++++++++++++++------------------------ sound/soc/soc-ops.c | 11 +----- 2 files changed, 59 insertions(+), 61 deletions(-) -- 2.7.4 Thank you for your help !! Best regards --- Kuninori Morimoto

3 18

[alsa-devel] [PATCH] ASoc: rockchip: i2s: Fix RPM imbalance
by Robin Murphy 04 Oct '19

04 Oct '19

If rockchip_pcm_platform_register() fails, e.g. upon deferring to wait for an absent DMA channel, we return without disabling RPM, which makes subsequent re-probe attempts scream with errors about the unbalanced enable. Don't do that. Fixes: ebb75c0bdba2 ("ASoC: rockchip: i2s: Adjust devm usage") Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> --- sound/soc/rockchip/rockchip_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index af2d5a6124c8..61c984f10d8e 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -677,7 +677,7 @@ static int rockchip_i2s_probe(struct platform_device *pdev) ret = rockchip_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - return ret; + goto err_suspend; } return 0; -- 2.21.0.dirty

2 1

[alsa-devel] [PATCH] ASoC: rsnd: Document r8a774b1 bindings
by Biju Das 04 Oct '19

04 Oct '19

Document SoC specific bindings for RZ/G2N (r8a774b1) SoC. Signed-off-by: Biju Das <biju.das(a)bp.renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 5c52182..797fd03 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -268,6 +268,7 @@ Required properties: - "renesas,rcar_sound-r8a7745" (RZ/G1E) - "renesas,rcar_sound-r8a77470" (RZ/G1C) - "renesas,rcar_sound-r8a774a1" (RZ/G2M) + - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - "renesas,rcar_sound-r8a7778" (R-Car M1A) - "renesas,rcar_sound-r8a7779" (R-Car H1) -- 2.7.4

2 1

Re: [alsa-devel] Custom Kernel driver to open and use another ALSA PCM
by Kai Vehmanen 04 Oct '19

04 Oct '19

Hi, (routing back to alsa-devel, Ed's alsa-devel subscription is still pending so he can't sent to the list, forwarding with his permission) On Fri, 4 Oct 2019, ed nwave wrote: > The example used of a voice changing app was just an illustration of how > I need to connect "my driver" directly to a sound card driver that's > all. I understand the user-space comment entirely. The actual > objective is one common driver ( my PCM driver ) being used by 4 music > applications where I can dynamically switch the output PCM without > restarting the applications or the applications even knowing the output > cards not real. so far this is pretty much directly from original design goals of JACK. But ok, then you mention this: > I must have direct access to audio hardware i.e. hw:1,0 > for applications like RoonReady/ RAAT. So Pulseaudio's out. I require > the ability to Now this can be a problem of course if you cannot modify the apps. The you would have to route this via alsa-lib plugins. > follow ( like and MPD playlist ). As I understand Jack is set for a > certain rate and format at the start when connecting and does not update > rates automatically if the rate of the songs change; can jack connect > directly to hw and handle DSD rates?. This is all part of a headless I've been out of the loop for some years now, but no, I don't think JACK supports these. But you are anyways customizing quite a lot of SW, so you could e.g. consider extending JACK and/or some other audio server for this purpose, or just write a new one. > Other requirements are my applications must have no plugins or dmix and > connect directly to hardware i.e. hw:1,0 - due to rate demands, no > remixing. Another requirement is the latency required by one company to > attain multi-room streaming being about 1ms max, this is for > certification. Finally there's the need to change bit rate settings I think reaching the latency requirement (and not doing mixing) can certainly be done in user-space, so I still read the "need to access harwdare directly" more as "existing user-space components are not suitable for my needs" than a strict requirement for doing this in kernel. But alas, I get your context and I understand these constraints are real for you as well. Just wanted to make sure you have taken good enough look at user-space options. > All that's left for me to do is to connect pipe the data directly to the > driver for the my sound card. Question is can it be done? And if so, > where do I even start? How can I import snd_usb_audio and open it so it > appears a the typical snd_card struct? The longrunning trend has been to push stuff like this to user-space, so you will not find much examples. I haven't done this myself, so I'll let other people make recommendations. I'd start by looking at ALSA OSS emulation and the virtualized drivers we have (like sound/xen/). Br, Kai

1 0

Re: [alsa-devel] kernel freeze regression relating to imx-sdma
by Kevin Groeneveld 04 Oct '19

04 Oct '19

Hi Lucas, Thanks for the feedback. >-----Original Message----- >From: Lucas Stach [mailto:l.stach@pengutronix.de] >Sent: October 3, 2019 5:26 PM >To: Kevin Groeneveld; dmaengine(a)vger.kernel.org >Subject: Re: kernel freeze regression relating to imx-sdma > >Hi Kevin, > >Am Donnerstag, den 03.10.2019, 21:11 +0000 schrieb Kevin Groeneveld: >> We have recently come across an issue which can cause the linux kernel to >> freeze on an iMX6 based system. After bisecting the issue I have found the >> issue is related to the following imx-sdma commits: >> >> 64068853bc77786d1a28abb4087d6a3e93aedbe2 dmaengine: imx-sdma: use GFP_NOWAIT for dma descriptor allocations >> b8603d2a5795c42f78998e70dc792336e0dc20c9 dmaengine: imx-sdma: implement channel termination via worker >> ebb853b1bd5f659b92c71dc6a9de44cfc37c78c0 Revert "dmaengine: imx-sdma: alloclate bd memory from dma pool" >> c06abca69218ac42fa58d1ba7a7b0d9bab5f1b18 Revert "dmaengine: imx-sdma: Use GFP_NOWAIT for dma allocations" >> >> If I revert these four commits the freeze does not happen. The issue is still >> present up to at least v5.3. >> >> The problem seems to be related to overruns when doing an SPDIF capture, >> particularly when there is not actually a valid SPDIF input to the iMX6. When >> the issue occurs the following is (usually) printed on the console (using a v5.0 >> based kernel): >> >> [ 1251.944877] rcu: INFO: rcu_sched detected stalls on CPUs/tasks: >> [ 1251.950823] rcu: (detected by 0, t=2102 jiffies, g=220425, q=0) >> [ 1251.956866] rcu: All QSes seen, last rcu_sched kthread activity 2102 (95117-93015), jiffies_till_next_fqs=1, root ->qsmask 0x0 >> [ 1251.968266] arecord S 0 1916 1908 0x00000002 >> [ 1251.973816] [<8010dcec>] (unwind_backtrace) from [<8010a4c4>] (show_stack+0x10/0x14) >> [ 1251.981596] [<8010a4c4>] (show_stack) from [<8016b314>] (rcu_check_callbacks+0x57c/0x5d8) >> [ 1251.989806] [<8016b314>] (rcu_check_callbacks) from [<8016fe84>] (update_process_times+0x30/0x5c) >> [ 1251.998709] [<8016fe84>] (update_process_times) from [<8017e960>] (tick_sched_timer+0x44/0x94) >> [ 1252.007353] [<8017e960>] (tick_sched_timer) from [<80170a84>] (__hrtimer_run_queues+0xd8/0x160) >> [ 1252.016081] [<80170a84>] (__hrtimer_run_queues) from [<8017103c>] (hrtimer_interrupt+0xdc/0x284) >> [ 1252.024891] [<8017103c>] (hrtimer_interrupt) from [<8010d6c0>] (twd_handler+0x2c/0x38) >> [ 1252.032839] [<8010d6c0>] (twd_handler) from [<80160b34>] (handle_percpu_devid_irq+0x70/0x11c) >> [ 1252.041401] [<80160b34>] (handle_percpu_devid_irq) from [<8015c234>] (generic_handle_irq+0x18/0x28) >> [ 1252.050480] [<8015c234>] (generic_handle_irq) from [<8015c814>] (__handle_domain_irq+0xa0/0xb4) >> [ 1252.059211] [<8015c814>] (__handle_domain_irq) from [<8039098c>] (gic_handle_irq+0x58/0x90) >> [ 1252.067590] [<8039098c>] (gic_handle_irq) from [<80101a0c>] (__irq_svc+0x6c/0x90) >> [ 1252.075085] Exception stack(0x848afe30 to 0x848afe78) >> [ 1252.080155] fe20: 8decf200 00000000 00000000 80b84ac8 >> [ 1252.088354] fe40: 00000000 8decf200 892f7800 00001589 892f7800 00000000 00000000 848ae000 >> [ 1252.096549] fe60: 8d8cf000 848afe80 805721e8 8056aae4 60000013 ffffffff >> [ 1252.103200] [<80101a0c>] (__irq_svc) from [<8056aae4>] (__snd_pcm_stream_unlock_mode+0xc0/0xd8) >> [ 1252.111925] [<8056aae4>] (__snd_pcm_stream_unlock_mode) from [<00000000>] ( (null)) >> [ 1252.119695] rcu: rcu_sched kthread starved for 2102 jiffies! g220425 f0x2 RCU_GP_WAIT_FQS(5) ->state=0x0 ->cpu=0 >> [ 1252.129877] rcu: RCU grace-period kthread stack dump: >> [ 1252.134938] rcu_sched R running task 0 10 2 0x00000000 >> [ 1252.142034] [<806eee5c>] (__schedule) from [<806eef80>] (schedule+0x90/0xa0) >> [ 1252.149116] [<806eef80>] (schedule) from [<806f2244>] (schedule_timeout+0x1f0/0x238) >> [ 1252.156891] [<806f2244>] (schedule_timeout) from [<8016940c>] (rcu_gp_kthread+0x550/0x8f0) >> [ 1252.165183] [<8016940c>] (rcu_gp_kthread) from [<8013a358>] (kthread+0x110/0x128) >> [ 1252.172693] [<8013a358>] (kthread) from [<801010e8>] (ret_from_fork+0x14/0x2c) >> [ 1252.179925] Exception stack(0x8f877fb0 to 0x8f877ff8) >> [ 1252.184991] 7fa0: 00000000 00000000 00000000 00000000 >> [ 1252.193187] 7fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 >> [ 1252.201380] 7fe0: 00000000 00000000 00000000 00000000 00000013 00000000 >> >> I suspect the issue may not actually be a bug with imx-sdma but a >> race condition elsewhere that is exposed by the four mentioned >> commits. >> >> I did notice that when an SPDIF overrun occurs >> dmaengine_terminate_async is called but dmaengine_synchronize is >> not. Should these calls always be paired? Or is it okay to call >> dmaengine_terminate_async without dmaengine_synchronize in some >> cases? > >Yes, it seems this uncovered a bug in the SPDIF driver. Before those >commits the dmaengine_terminate_async in the SDMA driver wasn't really >async, now it is. The API requires that you call dmaengine_synchronize >eventually after the terminate_async before starting any new operations >on the dmaengine. It doesn't necessarily need to be in the same >codepath, but the the async terminate always needs to be paired up with >a synchronize. I don't think the bug is limited to SPDIF. The calls to dmaengine_terminate_async are in sound/core/pcm_dmaengine.c. When the ALSA device is closed it does call dmaengine_synchronize. But when there is an overrun it does not. It was commit bc0e7345168c0f7483d2d1da86285d89136417cd (by Lars-Peter Clausen) where the call to dmaengine_synchronize was added for closing an ALSA device. It seems the overrun case was overlooked? I think maybe I should copy the ALSA list on this as well. I have also found another manifestation of the imx-sdma change which may be useful for others trying to debug this. If I run the following command: arecord -Dhw:4,0 -fS24_LE -c2 -r44100 --buffer-size 100 > /dev/null With kernel v4.20-rc5 I get an endless stream of overrun messages. With kernel v4.20-rc6 (which includes the four imx-sdma commits mentioned above) I get a few overrun messages but then arecord exits with "arecord: pcm_read:2096: read error: Input/output error" Best regards, Kevin

1 0

[alsa-devel] Custom Kernel driver to open and use another ALSA PCM
by ed nwave 04 Oct '19

04 Oct '19

Hello, A little bit of an unorthodox request but I need my custom PCM driver to connect directly to another PCM driver if possible. I have an application that already sees my custom PCM driver as the connected sound card, but I also need to pipe the PCM data within my driver after altering it out to another PCM of my choice ( my actual sound card ). As an example, my PCM driver acts as a sound card connected to a microphone capture application, within the my custom PCM I've added various "dalek" and other voice changing alterations and now I need to output this stream into my actual sound card. The sound card I need my driver to import or connect to is a DAC over USB and I believe the loaded kernel module is snd_sub_audio which I can see via lsmod. It's PCM hw:1,0. So is this possible? An if so, how? Even the most basic information would be appreciated. Thanks.

2 1

[alsa-devel] Making alsamixer configurable, adding mouse support + minor fixes ...
by GitHub pull_request - opened 03 Oct '19

03 Oct '19

alsa-project/alsa-utils pull request #28 was opened from braph: ... continued from PR #26 ... Request URL : https://github.com/alsa-project/alsa-utils/pull/28 Patch URL : https://github.com/alsa-project/alsa-utils/pull/28.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH v2] FROMLIST: drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 03 Oct '19

03 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- Change from v1 to v2: - Use audio_lock to protect audio clock. drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..749d8e4c535b 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,17 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + unsigned long flags; + + spin_lock_irqsave(&hdmi->audio_lock, flags); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + spin_unlock_irqrestore(&hdmi->audio_lock, flags); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2056,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

1 0

Re: [alsa-devel] [External] Re: alsa-lib: Add ucm support for whiskeylake sof-skl_hda_card audio
by Pierre-Louis Bossart 03 Oct '19

03 Oct '19

On 9/19/19 9:54 AM, Mark Pearson wrote: >> >> Indeed UCM is required for all cases where SOF and PulseAudio are used. >> >> Our thinking was however to add this UCM file to the new repository outside >> of alsa-lib [1]. There is an on-going thread started by Jaroslav to move those >> files and relicense them as BSD-3-Clause [2] >> >> [1] >> https://mailman.alsa-project.org/pipermail/alsa-devel/2019- >> July/153257.html >> [2] >> https://mailman.alsa-project.org/pipermail/alsa-devel/2019- >> September/155246.html > > Thanks Pierre. > > Do we have any approximate timelines on when and how this will happen? (I'm new to this) > > One of my main aims is that we have a customer using Debian and one of our platforms that require this change - I need to make sure I understand how this would roll out and what actions they need to take in the meantime if it's not going to be available in Debian. I think the first order would be to have the file cleaned-up, with its Intel origin clearly stated with a signed-off-by tag. Then once this is done, the Debian package creation needs to be handled (using either the ALSA repo or the cloned version on SOF GitHub). I don't have any experience with Debian packages so can't really comment on the effort it would take.

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[alsa-devel] [PATCH v2 00/10] Basic sound support for Arndale board / wm8994 updates
by Sylwester Nawrocki 02 Oct '19

02 Oct '19

This patch series adds basic audio support for Exynos5250 SoC based Arndale board, the Bluetooth receiver source and HDMI output are not covered yet. There is also one fix for wm8994 driver related to WM1811 CODEC and wm8994 updates to handle MCLK clocks, similar to patches: ae1ea48c5c59 ("ASoC: arizona: Add gating for source clocks of the FLLs") 7a4413d0dc96 ("ASoC: arizona: Add gating for clock when used for direct MCLK") The main changes since v1: - Fixed bug in "ASoC: wm8994: Add support for MCLKn clock gating" patch (thanks to Charles), - dropped the MFD part patch and moved getting of clocks to the CODEC driver, - Added missing Kconfig entries to patch "ASoC: samsung: arndale: Add support for WM1811 CODEC", - Added a patch renaming the driver and a patch for exynos_defconfig. Sylwester Nawrocki (10): ASoC: wm8994: Do not register inapplicable controls for WM1811 ASoC: samsung: arndale: Add missing OF node dereferencing ASoC: wm8994: Add support for setting MCLK clock rate ASoC: wm8994: Add support for MCLKn clock gating ASoC: samsung: arndale: Simplify DAI link initialization ASoC: dt-bindings: Document "samsung,arndale-wm1811" compatible ASoC: samsung: arndale: Add support for WM1811 CODEC ASoC: samsung: Rename Arndale card driver ARM: dts: arndale: Add audio support (WM1811 CODEC boards) ARM: exynos_defconfig: Enable Arndale audio driver .../devicetree/bindings/sound/arndale.txt | 5 +- arch/arm/boot/dts/exynos5250-arndale.dts | 27 ++- arch/arm/configs/exynos_defconfig | 1 + sound/soc/codecs/wm8994.c | 199 +++++++++++++--- sound/soc/codecs/wm8994.h | 10 +- sound/soc/samsung/Kconfig | 12 +- sound/soc/samsung/Makefile | 4 +- sound/soc/samsung/arndale.c | 217 ++++++++++++++++++ sound/soc/samsung/arndale_rt5631.c | 138 ----------- 9 files changed, 438 insertions(+), 175 deletions(-) create mode 100644 sound/soc/samsung/arndale.c delete mode 100644 sound/soc/samsung/arndale_rt5631.c -- 2.17.1

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[alsa-devel] [PATCH] ALSA: hda: add Intel DSP configuration / probe code
by Jaroslav Kysela 02 Oct '19

02 Oct '19

For distributions, we need one place where we can decide which driver will be activated for the auto-configation of the Intel's HDA hardware with DSP. Actually, we cover three drivers: * Legacy HDA * Intel SST * Intel Sound Open Firmware (SOF) All those drivers registers similar PCI IDs, so the first driver probed from the PCI stack can win. But... it is not guaranteed that the correct driver wins. This commit changes Intel's NHLT ACPI module to a common DSP probe module for the Intel's hardware. All above sound drivers calls this code. The user can force another behaviour using the module parameter 'dsp_driver' located in the 'snd-intel-dspcfg' module. This change allows to add specific dmi checks for the specific systems like proposed in the pull request: https://github.com/thesofproject/linux/pull/927 Tested on Lenovo Carbon X1 7th gen. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/intel-dsp-config.h | 35 ++++++++++++ sound/hda/Kconfig | 10 +++- sound/hda/Makefile | 7 ++- sound/hda/intel-dsp-config.c | 97 ++++++++++++++++++++++++++++++++ sound/hda/intel-nhlt.c | 3 - sound/pci/hda/Kconfig | 11 +--- sound/pci/hda/hda_intel.c | 49 +++++----------- sound/soc/intel/Kconfig | 2 +- sound/soc/intel/skylake/skl.c | 19 ++----- sound/soc/sof/intel/Kconfig | 2 +- sound/soc/sof/sof-pci-dev.c | 6 ++ 11 files changed, 173 insertions(+), 68 deletions(-) create mode 100644 include/sound/intel-dsp-config.h create mode 100644 sound/hda/intel-dsp-config.c diff --git a/include/sound/intel-dsp-config.h b/include/sound/intel-dsp-config.h new file mode 100644 index 000000000000..700f86282a83 --- /dev/null +++ b/include/sound/intel-dsp-config.h @@ -0,0 +1,35 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * intel-dsp-config.h - Intel DSP config + * + * Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + */ + +#ifndef __INTEL_DSP_CONFIG_H__ +#define __INTEL_DSP_CONFIG_H__ + +struct pci_dev; + +enum { + SND_INTEL_DSP_DRIVER_ANY = 0, + SND_INTEL_DSP_DRIVER_NODSP, + SND_INTEL_DSP_DRIVER_LEGACY, + SND_INTEL_DSP_DRIVER_SST, + SND_INTEL_DSP_DRIVER_SOF, + SND_INTEL_DSP_DRIVER_LAST = SND_INTEL_DSP_DRIVER_SOF +}; + +#if IS_ENABLED(CONFIG_SND_INTEL_DSP_CONFIG) + +int snd_intel_dsp_driver_probe(struct pci_dev *pci); + +#else + +static inline int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + return SND_INTEL_DSP_DRIVER_ANY; +} + +#endif + +#endif diff --git a/sound/hda/Kconfig b/sound/hda/Kconfig index 3d33fc1757ba..b0c88fe040ee 100644 --- a/sound/hda/Kconfig +++ b/sound/hda/Kconfig @@ -34,6 +34,12 @@ config SND_HDA_PREALLOC_SIZE via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. config SND_INTEL_NHLT - tristate + bool # this config should be selected only for Intel ACPI platforms. - # A fallback is provided so that the code compiles in all cases. \ No newline at end of file + # A fallback is provided so that the code compiles in all cases. + +config SND_INTEL_DSP_CONFIG + tristate + select SND_INTEL_NHLT if ACPI + # this config should be selected only for Intel DSP platforms. + # A fallback is provided so that the code compiles in all cases. diff --git a/sound/hda/Makefile b/sound/hda/Makefile index 8560f6ef1b19..c593253c9829 100644 --- a/sound/hda/Makefile +++ b/sound/hda/Makefile @@ -14,5 +14,8 @@ obj-$(CONFIG_SND_HDA_CORE) += snd-hda-core.o #extended hda obj-$(CONFIG_SND_HDA_EXT_CORE) += ext/ -snd-intel-nhlt-objs := intel-nhlt.o -obj-$(CONFIG_SND_INTEL_NHLT) += snd-intel-nhlt.o +snd-intel-dspcfg-objs := intel-dsp-config.o +ifneq ($(CONFIG_ACPI),) + snd-intel-dspcfg-objs += intel-nhlt.o +endif +obj-$(CONFIG_SND_INTEL_DSP_CONFIG) += snd-intel-dspcfg.o diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c new file mode 100644 index 000000000000..1abbf74ba643 --- /dev/null +++ b/sound/hda/intel-dsp-config.c @@ -0,0 +1,97 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2019 Jaroslav Kysela <perex(a)perex.cz> + +#include <sound/core.h> +#include <linux/pci.h> +#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> + +static int dsp_driver; + +module_param(dsp_driver, int, 0444); +MODULE_PARM_DESC(dsp_driver, "Force the DSP driver for Intel DSP (0=auto, 1=noDSP, 2=legacy, 3=SST, 4=SOF)"); + +static const u16 sof_skl_table[] = { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + 0x02c8, /* Cometlake-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + 0x06c8, /* Cometlake-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + 0x3198, /* Geminilake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + 0x5a98, /* Broxton-P (Appololake) */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + 0x9dc8, /* Cannonlake */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + 0xa348, /* Coffelake */ +#endif +}; + +static int snd_intel_dsp_check_device(u16 device, const u16 *table, u32 len) +{ + for (; len > 0; len--, table++) { + if (*table == device) + return 1; + } + return 0; +} + +static int snd_intel_dsp_check_dmic(struct pci_dev *pci) +{ + struct nhlt_acpi_table *nhlt; + int ret = 0; + + if (snd_intel_dsp_check_device(pci->device, sof_skl_table, ARRAY_SIZE(sof_skl_table))) { + nhlt = intel_nhlt_init(&pci->dev); + if (nhlt) { + if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) + ret = 1; + intel_nhlt_free(nhlt); + } + } + return ret; +} + +int snd_intel_dsp_driver_probe(struct pci_dev *pci) +{ + if (dsp_driver > 0 && dsp_driver <= SND_INTEL_DSP_DRIVER_LAST) + return dsp_driver; + + /* Intel vendor only */ + if (snd_BUG_ON(pci->vendor != 0x8086)) + return SND_INTEL_DSP_DRIVER_ANY; + + /* + * detect DSP by checking class/subclass/prog-id information + * class=04 subclass 03 prog-if 00: no DSP, use legacy driver + * class=04 subclass 01 prog-if 00: DSP is present + * (and may be required e.g. for DMIC or SSP support) + * class=04 subclass 03 prog-if 80: use DSP or legacy mode + */ + if (pci->class == 0x040300) + return SND_INTEL_DSP_DRIVER_NODSP; + if (pci->class != 0x040100 && pci->class != 0x040380) { + dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, selecting HDA legacy driver\n", pci->class); + return SND_INTEL_DSP_DRIVER_LEGACY; + } + + dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); + + /* DMIC check for Skylake+ */ + if (snd_intel_dsp_check_dmic(pci)) { + dev_info(&pci->dev, "Digital mics found on Skylake+ platform, using SOF driver\n"); + return SND_INTEL_DSP_DRIVER_SOF; + } + + return SND_INTEL_DSP_DRIVER_ANY; +} + +EXPORT_SYMBOL_GPL(snd_intel_dsp_driver_probe); + +MODULE_LICENSE("GPL v2"); +MODULE_DESCRIPTION("Intel DSP config driver"); diff --git a/sound/hda/intel-nhlt.c b/sound/hda/intel-nhlt.c index daede96f28ee..097ff6c10099 100644 --- a/sound/hda/intel-nhlt.c +++ b/sound/hda/intel-nhlt.c @@ -102,6 +102,3 @@ int intel_nhlt_get_dmic_geo(struct device *dev, struct nhlt_acpi_table *nhlt) return dmic_geo; } EXPORT_SYMBOL_GPL(intel_nhlt_get_dmic_geo); - -MODULE_LICENSE("GPL v2"); -MODULE_DESCRIPTION("Intel NHLT driver"); diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index dae47a45b2b8..bd48335d09d7 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -12,7 +12,7 @@ config SND_HDA_INTEL tristate "HD Audio PCI" depends on SND_PCI select SND_HDA - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help Say Y here to include support for Intel "High Definition Audio" (Azalia) and its compatible devices. @@ -23,15 +23,6 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. -config SND_HDA_INTEL_DETECT_DMIC - bool "DMIC detection and probe abort" - depends on SND_HDA_INTEL - help - Say Y to detect digital microphones on SKL+ devices. DMICs - cannot be handled by the HDaudio legacy driver and are - currently only supported by the SOF driver. - If unsure say N. - config SND_HDA_TEGRA tristate "NVIDIA Tegra HD Audio" depends on ARCH_TEGRA diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 240f4ca76391..c76c2deea47c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,7 +46,7 @@ #include <sound/initval.h> #include <sound/hdaudio.h> #include <sound/hda_i915.h> -#include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include <linux/vgaarb.h> #include <linux/vga_switcheroo.h> #include <linux/firmware.h> @@ -124,7 +124,7 @@ static char *patch[SNDRV_CARDS]; static bool beep_mode[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = CONFIG_SND_HDA_INPUT_BEEP_MODE}; #endif -static bool dmic_detect = IS_ENABLED(CONFIG_SND_HDA_INTEL_DETECT_DMIC); +static bool no_dsp_driver; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Intel HD audio interface."); @@ -159,8 +159,8 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -module_param(dmic_detect, bool, 0444); -MODULE_PARM_DESC(dmic_detect, "DMIC detect on SKL+ platforms"); +module_param(no_dsp_driver, bool, 0444); +MODULE_PARM_DESC(no_dsp_driver, "Do not return from probe when another DSP driver claims device"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -2020,25 +2020,6 @@ static const struct hda_controller_ops pci_hda_ops = { .position_check = azx_position_check, }; -static int azx_check_dmic(struct pci_dev *pci, struct azx *chip) -{ - struct nhlt_acpi_table *nhlt; - int ret = 0; - - if (chip->driver_type == AZX_DRIVER_SKL && - pci->class != 0x040300) { - nhlt = intel_nhlt_init(&pci->dev); - if (nhlt) { - if (intel_nhlt_get_dmic_geo(&pci->dev, nhlt)) { - ret = -ENODEV; - dev_info(&pci->dev, "Digital mics found on Skylake+ platform, aborting probe\n"); - } - intel_nhlt_free(nhlt); - } - } - return ret; -} - static int azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -2056,6 +2037,17 @@ static int azx_probe(struct pci_dev *pci, return -ENOENT; } + /* + * stop probe if another Intel's DSP driver should be activated + */ + if (!no_dsp_driver) { + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_NODSP && + err != SND_INTEL_DSP_DRIVER_LEGACY) + return -ENODEV; + } + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) { @@ -2069,17 +2061,6 @@ static int azx_probe(struct pci_dev *pci, card->private_data = chip; hda = container_of(chip, struct hda_intel, chip); - /* - * stop probe if digital microphones detected on Skylake+ platform - * with the DSP enabled. This is an opt-in behavior defined at build - * time or at run-time with a module parameter - */ - if (dmic_detect) { - err = azx_check_dmic(pci, chip); - if (err < 0) - goto out_free; - } - pci_set_drvdata(pci, card); err = register_vga_switcheroo(chip); diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..9ad89d56092b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..58ba3e9469ba 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -27,6 +27,7 @@ #include <sound/hda_i915.h> #include <sound/hda_codec.h> #include <sound/intel-nhlt.h> +#include <sound/intel-dsp-config.h> #include "skl.h" #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" @@ -987,22 +988,10 @@ static int skl_probe(struct pci_dev *pci, switch (skl_pci_binding) { case SND_SKL_PCI_BIND_AUTO: - /* - * detect DSP by checking class/subclass/prog-id information - * class=04 subclass 03 prog-if 00: no DSP, use legacy driver - * class=04 subclass 01 prog-if 00: DSP is present - * (and may be required e.g. for DMIC or SSP support) - * class=04 subclass 03 prog-if 80: use DSP or legacy mode - */ - if (pci->class == 0x040300) { - dev_info(&pci->dev, "The DSP is not enabled on this platform, aborting probe\n"); + err = snd_intel_dsp_driver_probe(pci); + if (err != SND_INTEL_DSP_DRIVER_ANY && + err != SND_INTEL_DSP_DRIVER_SST) return -ENODEV; - } - if (pci->class != 0x040100 && pci->class != 0x040380) { - dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); - return -ENODEV; - } - dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, aborting probe\n"); diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..8a5d5c0f95f2 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT if ACPI + select SND_INTEL_DSP_CONFIG help This option is not user-selectable but automagically handled by 'select' statements at a higher level diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index d66412a77873..3a9e0e2a150d 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/pci.h> #include <linux/pm_runtime.h> +#include <sound/intel-dsp-config.h> #include <sound/soc-acpi.h> #include <sound/soc-acpi-intel-match.h> #include <sound/sof.h> @@ -277,6 +278,11 @@ static int sof_pci_probe(struct pci_dev *pci, const struct snd_sof_dsp_ops *ops; int ret; + ret = snd_intel_dsp_driver_probe(pci); + if (ret != SND_INTEL_DSP_DRIVER_ANY && + ret != SND_INTEL_DSP_DRIVER_SOF) + return -ENODEV; + dev_dbg(&pci->dev, "PCI DSP detected"); /* get ops for platform */ -- 2.20.1

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[alsa-devel] [PATCH 1/2] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Maciej Falkowski 02 Oct '19

02 Oct '19

Convert Samsung I2S controller to newer dt-schema format Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ------------- .../bindings/sound/samsung-i2s.yaml | 119 ++++++++++++++++++ 2 files changed, 119 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..59dc76035cb4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,119 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sangbeom Kim <sbkim73(a)samsung.com> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - "samsung,s3c6410-i2s" + - "samsung,s5pv210-i2s" + - "samsung,exynos5420-i2s" + - "samsung,exynos7-i2s" + - "samsung,exynos7-i2s1" + + reg: + maxItems: 1 + + dmas: + description: list of DMA controller phandle and DMA request line ordered pairs. + + dma-names: + description: | + identifier string for each DMA request line in the dmas property. + These strings correspond 1:1 with the ordered pairs in dmas. + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: + - const: iis + - const: i2s_opclk0 + - items: + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + description: | + "iis" is the i2s bus clock. + For i2s1 and i2s2 - "iis", "i2s_opclk0" + For i2s0 - "iis", "i2s_opclk0", "i2s_opclk1" + + "#clock-cells": + const: 1 + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10 + &pdma0 9 + &pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss 0>, // EXYNOS_I2S_BUS + <&clock_audss 0>, // EXYNOS_I2S_BUS + <&clock_audss 0>; // EXYNOS_SCLK_I2S + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

5 19

[alsa-devel] Period size setting is ignored when dmix is used
by Andrea Narciso - ART S.p.A. 02 Oct '19

02 Oct '19

I would like to set different period size (or time) values for different ALSA applications, in order to optimize latency and processor load based on specific application needs. For example, a telephony application will use a shorter period size/time than a media player application, so that low latency does not affect the quality of the call. I found out that if dmix plugin is used in the ALSA path, period size/time settings are completely ignored and forced to default values (supposedly read from the ALSA driver of the HW card). For example, not using dmix, period size value is correctly accepted (but of course I cannot play audio from multiple applications at the same time): aplay -vv --period-size=4096 -D plug:hw <WAV file name> Plug PCM: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 ... period_size : 4096 ... Adding dmix instead, I can play audio from multiple applications at the same time, but period size is ignored and I cannot therefore set different latencies for different applications: aplay -vv --period-size=4096 -D plug:dmix <WAV file name> Plug PCM: Hardware PCM card 0 'HDA Intel' device 0 subdevice 0 ... period_size : 1024 ... Am I missing something? Is there a way to implement mixing + different latency settings in ALSA applications, without using an audio server? Thanks Andrea

2 1

[alsa-devel] [PATCH] ASoC: atmel: fix atmel_ssc_set_audio link failure
by Arnd Bergmann 02 Oct '19

02 Oct '19

The ssc audio driver can call into both pdc and dma backends. With the latest rework, the logic to do this in a safe way avoiding link errors was removed, bringing back link errors that were fixed long ago in commit 061981ff8cc8 ("ASoC: atmel: properly select dma driver state") such as sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio': atmel_ssc_dai.c:(.text+0xac): undefined reference to `atmel_pcm_pdc_platform_register' Fix it this time using Makefile hacks and a comment to prevent this from accidentally getting removed again rather than Kconfig hacks. Fixes: 18291410557f ("ASoC: atmel: enable SOC_SSC_PDC and SOC_SSC_DMA in Kconfig") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/atmel/Kconfig | 4 ++-- sound/soc/atmel/Makefile | 10 ++++++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index f118c229ed82..25c31bf64936 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -10,11 +10,11 @@ config SND_ATMEL_SOC if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC - tristate + bool depends on HAS_DMA config SND_ATMEL_SOC_DMA - tristate + bool select SND_SOC_GENERIC_DMAENGINE_PCM config SND_ATMEL_SOC_SSC diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 1f6890ed3738..c7d2989791be 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -6,8 +6,14 @@ snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o snd-soc-atmel-i2s-objs := atmel-i2s.o snd-soc-mchp-i2s-mcc-objs := mchp-i2s-mcc.o -obj-$(CONFIG_SND_ATMEL_SOC_PDC) += snd-soc-atmel-pcm-pdc.o -obj-$(CONFIG_SND_ATMEL_SOC_DMA) += snd-soc-atmel-pcm-dma.o +# pdc and dma need to both be built-in if any user of +# ssc is built-in. +ifdef CONFIG_SND_ATMEL_SOC_PDC +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-pdc.o +endif +ifdef CONFIG_SND_ATMEL_SOC_DMA +obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel-pcm-dma.o +endif obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o obj-$(CONFIG_SND_ATMEL_SOC_I2S) += snd-soc-atmel-i2s.o obj-$(CONFIG_SND_MCHP_SOC_I2S_MCC) += snd-soc-mchp-i2s-mcc.o -- 2.20.0

4 6

[alsa-devel] [PATCH v3 RESEND 1/3] ASoC: samsung: arndale: Simplify DAI link initialization
by Sylwester Nawrocki 02 Oct '19

02 Oct '19

There is only one DAI link so we can drop an unnecessary loop statement. Use card->dai_link in place of direct static arndale_rt5631_dai[] array dereference as a prerequisite for adding support for other CODECs. Unnecessary assignment of dai_link->codecs->name to NULL is removed. Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Acked-by: Krzysztof Kozlowski <krzk(a)kernel.org> Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v1: - rebased due to reordering patches in the series --- sound/soc/samsung/arndale_rt5631.c | 42 +++++++++++++----------------- 1 file changed, 18 insertions(+), 24 deletions(-) diff --git a/sound/soc/samsung/arndale_rt5631.c b/sound/soc/samsung/arndale_rt5631.c index fd8c6642fb0d..004c84fafad9 100644 --- a/sound/soc/samsung/arndale_rt5631.c +++ b/sound/soc/samsung/arndale_rt5631.c @@ -88,36 +88,30 @@ static void arndale_put_of_nodes(struct snd_soc_card *card) static int arndale_audio_probe(struct platform_device *pdev) { - int n, ret; struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &arndale_rt5631; + struct snd_soc_dai_link *dai_link; + int ret; card->dev = &pdev->dev; + dai_link = card->dai_link; + + dai_link->cpus->of_node = of_parse_phandle(np, "samsung,audio-cpu", 0); + if (!dai_link->cpus->of_node) { + dev_err(&pdev->dev, + "Property 'samsung,audio-cpu' missing or invalid\n"); + return -EINVAL; + } - for (n = 0; np && n < ARRAY_SIZE(arndale_rt5631_dai); n++) { - if (!arndale_rt5631_dai[n].cpus->dai_name) { - arndale_rt5631_dai[n].cpus->of_node = of_parse_phandle(np, - "samsung,audio-cpu", n); - - if (!arndale_rt5631_dai[n].cpus->of_node) { - dev_err(&pdev->dev, - "Property 'samsung,audio-cpu' missing or invalid\n"); - return -EINVAL; - } - } - if (!arndale_rt5631_dai[n].platforms->name) - arndale_rt5631_dai[n].platforms->of_node = - arndale_rt5631_dai[n].cpus->of_node; - - arndale_rt5631_dai[n].codecs->name = NULL; - arndale_rt5631_dai[n].codecs->of_node = of_parse_phandle(np, - "samsung,audio-codec", n); - if (!arndale_rt5631_dai[0].codecs->of_node) { - dev_err(&pdev->dev, + if (!dai_link->platforms->name) + dai_link->platforms->of_node = dai_link->cpus->of_node; + + dai_link->codecs->of_node = of_parse_phandle(np, "samsung,audio-codec", 0); + if (!dai_link->codecs->of_node) { + dev_err(&pdev->dev, "Property 'samsung,audio-codec' missing or invalid\n"); - ret = -EINVAL; - goto err_put_of_nodes; - } + ret = -EINVAL; + goto err_put_of_nodes; } ret = devm_snd_soc_register_card(card->dev, card); -- 2.17.1

4 8

[alsa-devel] [PATCH v2 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- Changes in v2: * Use C++ coment style on the entire comment in the .c files; * Do not use ternery operator; * Do not use `devm_regulator_get_optional` as the device always requires power; * Let DAPM suspend/resume the device. MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 586 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 770 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index 296de2b51c83..8ba47cac8e83 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1002,6 +1002,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bcac95785493..ae213c6392ec 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -396,6 +398,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 73b2d5982dcb..2118407a1dd5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -305,6 +308,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..45a5d2dcc0f2 --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw +// driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..a8211362fe82 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..bf5a5d75f81a --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,586 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver +// +// Copyright 2019 Analog Devices Inc. + +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + if (ret < 0) + return ret; + + return 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + st->iovdd = devm_regulator_get(st->dev, "IOVDD"); + if (IS_ERR(st->iovdd)) { + dev_err(st->dev, "Could not get iovdd: %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + return devm_add_action_or_reset(st->dev, adau7118_regulator_disable, + st); +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] [PATCH 1/2] ASOC: Add ADAU7118 8 Channel PDM-to-I2S/TDM Converter driver
by Nuno Sá 02 Oct '19

02 Oct '19

This patch adds support for the 8 channel PDM-to-I2S/TDM converter. The ADAU7118 converts four stereo pulse density modulation (PDM) bitstreams into one pulse code modulation (PCM) output stream. The source for the PDM data can be eight microphones or other PDM sources. The PCM audio data is output on a serial audio interface port in either inter-IC serial (I2S) or time domain multiplexed (TDM) format. Signed-off-by: Nuno Sá <nuno.sa(a)analog.com> --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 28 ++ sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau7118-hw.c | 43 +++ sound/soc/codecs/adau7118-i2c.c | 82 +++++ sound/soc/codecs/adau7118.c | 621 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau7118.h | 24 ++ 7 files changed, 805 insertions(+) create mode 100644 sound/soc/codecs/adau7118-hw.c create mode 100644 sound/soc/codecs/adau7118-i2c.c create mode 100644 sound/soc/codecs/adau7118.c create mode 100644 sound/soc/codecs/adau7118.h diff --git a/MAINTAINERS b/MAINTAINERS index a50e97a63bc8..77c920559d82 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -985,6 +985,7 @@ F: drivers/media/i2c/adv7842* ANALOG DEVICES INC ASOC CODEC DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> +M: Nuno Sá <nuno.sa(a)analog.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a5346299..006b0686776c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -34,6 +34,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADAU7002 + select SND_SOC_ADAU7118_I2C if I2C + select SND_SOC_ADAU7118_HW select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4118 if I2C @@ -392,6 +394,32 @@ config SND_SOC_ADAU1977_I2C config SND_SOC_ADAU7002 tristate "Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter" +config SND_SOC_ADAU7118 + tristate + +config SND_SOC_ADAU7118_HW + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - HW Mode" + select SND_SOC_ADAU7118 + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter. In this mode, the device works in standalone mode which + means that there is no bus to comunicate with it. Stereo mode is not + supported in this mode. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-hw. + +config SND_SOC_ADAU7118_I2C + tristate "Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM Converter - I2C" + select SND_SOC_ADAU7118 + select REGMAP_I2C + help + Enable support for the Analog Devices ADAU7118 8 Channel PDM-to-I2S/TDM + Converter over I2C. This gives full support over the device. + + To compile this driver as a module, choose M here: the module + will be called snd-soc-adau7118-i2c. + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8cf4325..974dc1eda797 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,9 @@ snd-soc-adau1977-objs := adau1977.o snd-soc-adau1977-spi-objs := adau1977-spi.o snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adau7002-objs := adau7002.o +snd-soc-adau7118-objs := adau7118.o +snd-soc-adau7118-i2c-objs := adau7118-i2c.o +snd-soc-adau7118-hw-objs := adau7118-hw.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -301,6 +304,9 @@ obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU7002) += snd-soc-adau7002.o +obj-$(CONFIG_SND_SOC_ADAU7118) += snd-soc-adau7118.o +obj-$(CONFIG_SND_SOC_ADAU7118_I2C) += snd-soc-adau7118-i2c.o +obj-$(CONFIG_SND_SOC_ADAU7118_HW) += snd-soc-adau7118-hw.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o diff --git a/sound/soc/codecs/adau7118-hw.c b/sound/soc/codecs/adau7118-hw.c new file mode 100644 index 000000000000..499dd48ea01b --- /dev/null +++ b/sound/soc/codecs/adau7118-hw.c @@ -0,0 +1,43 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter Standalone Hw + * driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/platform_device.h> + +#include "adau7118.h" + +static int adau7118_probe_hw(struct platform_device *pdev) +{ + return adau7118_probe(&pdev->dev, NULL, true); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct platform_device_id adau7118_id[] = { + { .name = "adau7118" }, + { } +}; +MODULE_DEVICE_TABLE(platform, adau7118_id); + +static struct platform_driver adau7118_driver_hw = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_hw, + .id_table = adau7118_id, +}; +module_platform_driver(adau7118_driver_hw); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver for standalone hw mode"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118-i2c.c b/sound/soc/codecs/adau7118-i2c.c new file mode 100644 index 000000000000..1fd0b0287699 --- /dev/null +++ b/sound/soc/codecs/adau7118-i2c.c @@ -0,0 +1,82 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "adau7118.h" + +static const struct reg_default adau7118_reg_defaults[] = { + { ADAU7118_REG_VENDOR_ID, 0x41 }, + { ADAU7118_REG_DEVICE_ID1, 0x71 }, + { ADAU7118_REG_DEVICE_ID2, 0x18 }, + { ADAU7118_REG_REVISION_ID, 0x00 }, + { ADAU7118_REG_ENABLES, 0x3F }, + { ADAU7118_REG_DEC_RATIO_CLK_MAP, 0xC0 }, + { ADAU7118_REG_HPF_CONTROL, 0xD0 }, + { ADAU7118_REG_SPT_CTRL1, 0x41 }, + { ADAU7118_REG_SPT_CTRL2, 0x00 }, + { ADAU7118_REG_SPT_CX(0), 0x01 }, + { ADAU7118_REG_SPT_CX(1), 0x11 }, + { ADAU7118_REG_SPT_CX(2), 0x21 }, + { ADAU7118_REG_SPT_CX(3), 0x31 }, + { ADAU7118_REG_SPT_CX(4), 0x41 }, + { ADAU7118_REG_SPT_CX(5), 0x51 }, + { ADAU7118_REG_SPT_CX(6), 0x61 }, + { ADAU7118_REG_SPT_CX(7), 0x71 }, + { ADAU7118_REG_DRIVE_STRENGTH, 0x2a }, + { ADAU7118_REG_RESET, 0x00 }, +}; + +static const struct regmap_config adau7118_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .reg_defaults = adau7118_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau7118_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .max_register = ADAU7118_REG_RESET, +}; + +static int adau7118_probe_i2c(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *map; + + map = devm_regmap_init_i2c(i2c, &adau7118_regmap_config); + if (IS_ERR(map)) { + dev_err(&i2c->dev, "Failed to init regmap %ld\n", PTR_ERR(map)); + return PTR_ERR(map); + } + + return adau7118_probe(&i2c->dev, map, false); +} + +static const struct of_device_id adau7118_of_match[] = { + { .compatible = "adi,adau7118" }, + {} +}; +MODULE_DEVICE_TABLE(of, adau7118_of_match); + +static const struct i2c_device_id adau7118_id[] = { + {"adau7118", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, adau7118_id); + +static struct i2c_driver adau7118_driver = { + .driver = { + .name = "adau7118", + .of_match_table = adau7118_of_match, + }, + .probe = adau7118_probe_i2c, + .id_table = adau7118_id, +}; +module_i2c_driver(adau7118_driver); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver over I2C"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.c b/sound/soc/codecs/adau7118.c new file mode 100644 index 000000000000..d6b90e31042f --- /dev/null +++ b/sound/soc/codecs/adau7118.c @@ -0,0 +1,621 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Analog Devices ADAU7118 8 channel PDM-to-I2S/TDM Converter driver + * + * Copyright 2019 Analog Devices Inc. + */ +#include <linux/bitfield.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "adau7118.h" + +#define ADAU7118_DEC_RATIO_MASK GENMASK(1, 0) +#define ADAU7118_DEC_RATIO(x) FIELD_PREP(ADAU7118_DEC_RATIO_MASK, x) +#define ADAU7118_CLK_MAP_MASK GENMASK(7, 4) +#define ADAU7118_SLOT_WIDTH_MASK GENMASK(5, 4) +#define ADAU7118_SLOT_WIDTH(x) FIELD_PREP(ADAU7118_SLOT_WIDTH_MASK, x) +#define ADAU7118_TRISTATE_MASK BIT(6) +#define ADAU7118_TRISTATE(x) FIELD_PREP(ADAU7118_TRISTATE_MASK, x) +#define ADAU7118_DATA_FMT_MASK GENMASK(3, 1) +#define ADAU7118_DATA_FMT(x) FIELD_PREP(ADAU7118_DATA_FMT_MASK, x) +#define ADAU7118_SAI_MODE_MASK BIT(0) +#define ADAU7118_SAI_MODE(x) FIELD_PREP(ADAU7118_SAI_MODE_MASK, x) +#define ADAU7118_LRCLK_BCLK_POL_MASK GENMASK(1, 0) +#define ADAU7118_LRCLK_BCLK_POL(x) \ + FIELD_PREP(ADAU7118_LRCLK_BCLK_POL_MASK, x) +#define ADAU7118_SPT_SLOT_MASK GENMASK(7, 4) +#define ADAU7118_SPT_SLOT(x) FIELD_PREP(ADAU7118_SPT_SLOT_MASK, x) +#define ADAU7118_FULL_SOFT_R_MASK BIT(1) +#define ADAU7118_FULL_SOFT_R(x) FIELD_PREP(ADAU7118_FULL_SOFT_R_MASK, x) + +struct adau7118_data { + struct regmap *map; + struct device *dev; + struct regulator *iovdd; + struct regulator *dvdd; + u32 slot_width; + u32 slots; + bool hw_mode; + bool right_j; +}; + +/* Input Enable */ +static const struct snd_kcontrol_new adau7118_dapm_pdm_control[4] = { + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 0, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 1, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 2, 1, 0), + SOC_DAPM_SINGLE("Capture Switch", ADAU7118_REG_ENABLES, 3, 1, 0), +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_sw[] = { + /* Input Enable Switches */ + SND_SOC_DAPM_SWITCH("PDM0", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[0]), + SND_SOC_DAPM_SWITCH("PDM1", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[1]), + SND_SOC_DAPM_SWITCH("PDM2", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[2]), + SND_SOC_DAPM_SWITCH("PDM3", SND_SOC_NOPM, 0, 0, + &adau7118_dapm_pdm_control[3]), + + /* PDM Clocks */ + SND_SOC_DAPM_SUPPLY("PDM_CLK0", ADAU7118_REG_ENABLES, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PDM_CLK1", ADAU7118_REG_ENABLES, 5, 0, NULL, 0), + + /* Output channels */ + SND_SOC_DAPM_AIF_OUT("AIF1TX1", "Capture", 0, ADAU7118_REG_SPT_CX(0), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX2", "Capture", 0, ADAU7118_REG_SPT_CX(1), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX3", "Capture", 0, ADAU7118_REG_SPT_CX(2), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX4", "Capture", 0, ADAU7118_REG_SPT_CX(3), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX5", "Capture", 0, ADAU7118_REG_SPT_CX(4), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX6", "Capture", 0, ADAU7118_REG_SPT_CX(5), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX7", "Capture", 0, ADAU7118_REG_SPT_CX(6), + 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX8", "Capture", 0, ADAU7118_REG_SPT_CX(7), + 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_sw[] = { + { "PDM0", "Capture Switch", "PDM_DAT0" }, + { "PDM1", "Capture Switch", "PDM_DAT1" }, + { "PDM2", "Capture Switch", "PDM_DAT2" }, + { "PDM3", "Capture Switch", "PDM_DAT3" }, + { "AIF1TX1", NULL, "PDM0" }, + { "AIF1TX2", NULL, "PDM0" }, + { "AIF1TX3", NULL, "PDM1" }, + { "AIF1TX4", NULL, "PDM1" }, + { "AIF1TX5", NULL, "PDM2" }, + { "AIF1TX6", NULL, "PDM2" }, + { "AIF1TX7", NULL, "PDM3" }, + { "AIF1TX8", NULL, "PDM3" }, + { "Capture", NULL, "PDM_CLK0" }, + { "Capture", NULL, "PDM_CLK1" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets_hw[] = { + SND_SOC_DAPM_AIF_OUT("AIF1TX", "Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route adau7118_routes_hw[] = { + { "AIF1TX", NULL, "PDM_DAT0" }, + { "AIF1TX", NULL, "PDM_DAT1" }, + { "AIF1TX", NULL, "PDM_DAT2" }, + { "AIF1TX", NULL, "PDM_DAT3" }, +}; + +static const struct snd_soc_dapm_widget adau7118_widgets[] = { + SND_SOC_DAPM_INPUT("PDM_DAT0"), + SND_SOC_DAPM_INPUT("PDM_DAT1"), + SND_SOC_DAPM_INPUT("PDM_DAT2"), + SND_SOC_DAPM_INPUT("PDM_DAT3"), +}; + +static int adau7118_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int chan, ret; + + dev_dbg(st->dev, "Set channel map, %d", tx_num); + + for (chan = 0; chan < tx_num; chan++) { + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CX(chan), + ADAU7118_SPT_SLOT_MASK, + ADAU7118_SPT_SLOT(tx_slot[chan])); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set format, fmt:%d\n", fmt); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(0)); + break; + case SND_SOC_DAIFMT_LEFT_J: + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + ADAU7118_DATA_FMT(1)); + break; + case SND_SOC_DAIFMT_RIGHT_J: + st->right_j = true; + break; + default: + dev_err(st->dev, "Invalid format %d", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + if (ret < 0) + return ret; + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(0); + break; + case SND_SOC_DAIFMT_NB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(2); + break; + case SND_SOC_DAIFMT_IB_NF: + regval = ADAU7118_LRCLK_BCLK_POL(1); + break; + case SND_SOC_DAIFMT_IB_IF: + regval = ADAU7118_LRCLK_BCLK_POL(3); + break; + default: + dev_err(st->dev, "Invalid Inv mask %d", + fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL2, + ADAU7118_LRCLK_BCLK_POL_MASK, + regval); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret; + + dev_dbg(st->dev, "Set tristate, %d\n", tristate); + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_TRISTATE_MASK, + ADAU7118_TRISTATE(tristate)); + + return ret < 0 ? ret : 0; +} + +static int adau7118_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + int ret = 0; + u32 regval; + + dev_dbg(st->dev, "Set tdm, slots:%d width:%d\n", slots, slot_width); + + switch (slot_width) { + case 32: + regval = ADAU7118_SLOT_WIDTH(0); + break; + case 24: + regval = ADAU7118_SLOT_WIDTH(2); + break; + case 16: + regval = ADAU7118_SLOT_WIDTH(1); + break; + default: + dev_err(st->dev, "Invalid slot width:%d\n", slot_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SLOT_WIDTH_MASK, regval); + if (ret < 0) + return ret; + + st->slot_width = slot_width; + st->slots = slots; + + return 0; +} + +static int adau7118_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct adau7118_data *st = + snd_soc_component_get_drvdata(dai->component); + u32 data_width = params_width(params), slots_width; + int ret; + u32 regval; + + if (!st->slots) { + /* set stereo mode */ + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_SAI_MODE_MASK, + ADAU7118_SAI_MODE(0)); + if (ret < 0) + return ret; + + slots_width = 32; + } else { + slots_width = st->slot_width; + } + + if (data_width > slots_width) { + dev_err(st->dev, "Invalid data_width:%d, slots_width:%d", + data_width, slots_width); + return -EINVAL; + } + + if (st->right_j) { + switch (slots_width - data_width) { + case 8: + /* delay bclck by 8 */ + regval = ADAU7118_DATA_FMT(2); + break; + case 12: + /* delay bclck by 12 */ + regval = ADAU7118_DATA_FMT(3); + break; + case 16: + /* delay bclck by 16 */ + regval = ADAU7118_DATA_FMT(4); + break; + default: + dev_err(st->dev, + "Cannot set right_j setting, slot_w:%d, data_w:%d\n", + slots_width, data_width); + return -EINVAL; + } + + ret = snd_soc_component_update_bits(dai->component, + ADAU7118_REG_SPT_CTRL1, + ADAU7118_DATA_FMT_MASK, + regval); + if (ret < 0) + return ret; + } + + return 0; +} + +static int adau7118_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + int ret = 0; + + dev_dbg(st->dev, "Set bias level %d\n", level); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) { + if (!st->iovdd) + return 0; + + /* power on */ + ret = regulator_enable(st->iovdd); + if (ret) + return ret; + + /* there's no timing constraints before enabling dvdd */ + ret = regulator_enable(st->dvdd); + if (ret) { + regulator_disable(st->iovdd); + return ret; + } + + if (st->hw_mode) + return 0; + + regcache_cache_only(st->map, false); + /* sync cache */ + ret = snd_soc_component_cache_sync(component); + } + break; + case SND_SOC_BIAS_OFF: + /* power off */ + if (!st->iovdd) + return 0; + + ret = regulator_disable(st->dvdd); + if (ret) + return ret; + + ret = regulator_disable(st->iovdd); + if (ret) + return ret; + + if (st->hw_mode) + return 0; + + /* cache only */ + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + + break; + } + + return ret; +} + +static int adau7118_component_probe(struct snd_soc_component *component) +{ + struct adau7118_data *st = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret = 0; + + if (st->hw_mode) { + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_hw, + ARRAY_SIZE(adau7118_widgets_hw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_hw, + ARRAY_SIZE(adau7118_routes_hw)); + } else { + snd_soc_component_init_regmap(component, st->map); + ret = snd_soc_dapm_new_controls(dapm, adau7118_widgets_sw, + ARRAY_SIZE(adau7118_widgets_sw)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, adau7118_routes_sw, + ARRAY_SIZE(adau7118_routes_sw)); + } + + return ret; +} + +#ifdef CONFIG_PM +static int adau7118_suspend(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); +} + +static int adau7118_resume(struct snd_soc_component *component) +{ + return snd_soc_component_force_bias_level(component, + SND_SOC_BIAS_STANDBY); +} +#else +#define adau7118_suspend NULL +#define adau7118_resume NULL +#endif + +static const struct snd_soc_dai_ops adau7118_ops = { + .hw_params = adau7118_hw_params, + .set_channel_map = adau7118_set_channel_map, + .set_fmt = adau7118_set_fmt, + .set_tdm_slot = adau7118_set_tdm_slot, + .set_tristate = adau7118_set_tristate, +}; + +static struct snd_soc_dai_driver adau7118_dai = { + .name = "adau7118-hifi-capture", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 8, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | + SNDRV_PCM_FMTBIT_S20_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 192000, + .sig_bits = 24, + }, +}; + +static const struct snd_soc_component_driver adau7118_component_driver = { + .probe = adau7118_component_probe, + .set_bias_level = adau7118_set_bias_level, + .suspend = adau7118_suspend, + .resume = adau7118_resume, + .dapm_widgets = adau7118_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau7118_widgets), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static void adau7118_regulator_disable(void *data) +{ + struct adau7118_data *st = data; + int ret; + /* + * If we fail to disable DVDD, don't bother in trying IOVDD. We + * actually don't want to be left in the situation where DVDD + * is enabled and IOVDD is disabled. + */ + ret = regulator_disable(st->dvdd); + if (ret) + return; + + regulator_disable(st->iovdd); +} + +static int adau7118_regulator_setup(struct adau7118_data *st) +{ + int ret = 0; + + st->iovdd = devm_regulator_get_optional(st->dev, "IOVDD"); + if (!IS_ERR(st->iovdd)) { + /* + * Only get dvdd if iovdd is present. Furthermore, + * if iovdd is present, dvdd must also be given. The two + * supplies should be controlled together since dvdd should + * not be applied before iovdd. + */ + st->dvdd = devm_regulator_get(st->dev, "DVDD"); + if (IS_ERR(st->dvdd)) { + dev_err(st->dev, "Could not get dvdd: %ld\n", + PTR_ERR(st->dvdd)); + return PTR_ERR(st->dvdd); + } + /* just assume the device is in reset */ + if (!st->hw_mode) { + regcache_mark_dirty(st->map); + regcache_cache_only(st->map, true); + } + + ret = devm_add_action_or_reset(st->dev, + adau7118_regulator_disable, st); + } else { + if (PTR_ERR(st->iovdd) == -ENODEV) { + st->iovdd = NULL; + } else { + dev_err(st->dev, "Failed to get IOVDD %ld\n", + PTR_ERR(st->iovdd)); + return PTR_ERR(st->iovdd); + } + } + + return ret; +} + +static int adau7118_parset_dt(const struct adau7118_data *st) +{ + int ret; + u32 dec_ratio = 0; + /* 4 inputs */ + u32 clk_map[4], regval; + + if (st->hw_mode) + return 0; + + ret = device_property_read_u32(st->dev, "adi,decimation-ratio", + &dec_ratio); + if (!ret) { + switch (dec_ratio) { + case 64: + regval = ADAU7118_DEC_RATIO(0); + break; + case 32: + regval = ADAU7118_DEC_RATIO(1); + break; + case 16: + regval = ADAU7118_DEC_RATIO(2); + break; + default: + dev_err(st->dev, "Invalid dec ratio: %u", dec_ratio); + return -EINVAL; + } + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_DEC_RATIO_MASK, regval); + if (ret) + return ret; + } + + ret = device_property_read_u32_array(st->dev, "adi,pdm-clk-map", + clk_map, ARRAY_SIZE(clk_map)); + if (!ret) { + int pdm; + u32 _clk_map = 0; + + for (pdm = 0; pdm < ARRAY_SIZE(clk_map); pdm++) + _clk_map |= (clk_map[pdm] << (pdm + 4)); + + ret = regmap_update_bits(st->map, + ADAU7118_REG_DEC_RATIO_CLK_MAP, + ADAU7118_CLK_MAP_MASK, _clk_map); + if (ret) + return ret; + } + + return 0; +} + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode) +{ + struct adau7118_data *st; + int ret; + + st = devm_kzalloc(dev, sizeof(*st), GFP_KERNEL); + if (!st) + return -ENOMEM; + + st->dev = dev; + st->hw_mode = hw_mode; + dev_set_drvdata(dev, st); + + if (!hw_mode) { + st->map = map; + adau7118_dai.ops = &adau7118_ops; + /* + * Perform a full soft reset. This will set all register's + * with their reset values. + */ + ret = regmap_update_bits(map, ADAU7118_REG_RESET, + ADAU7118_FULL_SOFT_R_MASK, + ADAU7118_FULL_SOFT_R(1)); + if (ret) + return ret; + } + + ret = adau7118_parset_dt(st); + if (ret) + return ret; + + ret = adau7118_regulator_setup(st); + if (ret) + return ret; + + return devm_snd_soc_register_component(dev, + &adau7118_component_driver, + &adau7118_dai, 1); +} +EXPORT_SYMBOL_GPL(adau7118_probe); + +MODULE_AUTHOR("Nuno Sa <nuno.sa(a)analog.com>"); +MODULE_DESCRIPTION("ADAU7118 8 channel PDM-to-I2S/TDM Converter driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau7118.h b/sound/soc/codecs/adau7118.h new file mode 100644 index 000000000000..c65679a4dff1 --- /dev/null +++ b/sound/soc/codecs/adau7118.h @@ -0,0 +1,24 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef _LINUX_ADAU7118_H +#define _LINUX_ADAU7118_H + +struct regmap; +struct device; + +/* register map */ +#define ADAU7118_REG_VENDOR_ID 0x00 +#define ADAU7118_REG_DEVICE_ID1 0x01 +#define ADAU7118_REG_DEVICE_ID2 0x02 +#define ADAU7118_REG_REVISION_ID 0x03 +#define ADAU7118_REG_ENABLES 0x04 +#define ADAU7118_REG_DEC_RATIO_CLK_MAP 0x05 +#define ADAU7118_REG_HPF_CONTROL 0x06 +#define ADAU7118_REG_SPT_CTRL1 0x07 +#define ADAU7118_REG_SPT_CTRL2 0x08 +#define ADAU7118_REG_SPT_CX(num) (0x09 + (num)) +#define ADAU7118_REG_DRIVE_STRENGTH 0x11 +#define ADAU7118_REG_RESET 0x12 + +int adau7118_probe(struct device *dev, struct regmap *map, bool hw_mode); + +#endif -- 2.23.0

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[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 02 Oct '19

02 Oct '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

2 3

[alsa-devel] [PATCH 00/44] ASoC: don't use snd_pcm_ops
by Kuninori Morimoto 02 Oct '19

02 Oct '19

Hi Mark Current snd_soc_component_driver is using snd_pcm_ops. But, it is mainly for ALSA, thus, it not good much to ALSA SoC. For example .open case (1)(2), each callback can't know it is called for which component. Thus, it need to use using snd_soc_rtdcom_lookup() (3). --- ALSA SoC --- ... if (component->driver->ops && component->driver->ops->open) (1) return component->driver->ops->open(substream); ... --- driver --- (2) static int xxx_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; (3) struct snd_soc_component *component = snd_soc_rtdcom_lookup(..); ... } It works today, but, will not work in the future if we support multi CPU/Codec/Platform, because 1 rtd might have multiple components which have same driver name. In other words snd_soc_rtdcom_lookup() doesn't work correctly in the future. To solve this issue, each callback needs to be called with component. This patch copies each snd_pcm_ops member under component driver, and having "component" as parameter. --- ALSA SoC --- ... if (component->driver->open) => return component->driver->open(component, substream); ... --- driver --- => static int xxx_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { ... } Kuninori Morimoto (44): ASoC: soc-core: merge snd_pcm_ops member to component driver ASoC: soc-core: add new pcm_construct/pcm_destruct ASoC: soc-core: add snd_soc_pcm_lib_ioctl() ASoC: intel: sst-haswell-pcm: remove snd_pcm_ops ASoC: intel: skl-pcm: remove snd_pcm_ops ASoC: intel: sst-mfld-platform-pcm: remove snd_pcm_ops ASoC: intel: sst-baytrail-pcm: remove snd_pcm_ops ASoC: amd: acp3x-pcm-dma: remove snd_pcm_ops ASoC: amd: acp-pcm-dma: remove snd_pcm_ops ASoC: atmel: atmel-pcm-pdc: remove snd_pcm_ops ASoC: au1x: dbdma2: remove snd_pcm_ops ASoC: au1x: dma: remove snd_pcm_ops ASoC: xtensa: xtfpga-i2s: remove snd_pcm_ops ASoC: xilinx: xlnx_formatter_pcm: remove snd_pcm_ops ASoC: uniphier: aio-dma: remove snd_pcm_ops ASoC: txx9: txx9aclc: remove snd_pcm_ops ASoC: stm: stm32_adfsdm: remove snd_pcm_ops ASoC: sprd: sprd-pcm-dma: remove snd_pcm_ops ASoC: sof: pcm: remove snd_pcm_ops ASoC: sh: rsnd: remove snd_pcm_ops ASoC: sh: fsi: remove snd_pcm_ops ASoC: sh: dma-sh7760: remove snd_pcm_ops ASoC: sh: siu_pcm: remove snd_pcm_ops ASoC: samsung: idma: remove snd_pcm_ops ASoC: qcom: lpass-platform: remove snd_pcm_ops ASoC: qcom: q6routing: remove snd_pcm_ops ASoC: qcom: q6asm-dai: remove snd_pcm_ops ASoC: pxa: mmp-pcm: remove snd_pcm_ops ASoC: pxa: remove snd_pcm_ops ASoC: meson: remove snd_pcm_ops ASoC: mediatek: mtk-btcvsd: remove snd_pcm_ops ASoC: mediatek: remove snd_pcm_ops ASoC: kirkwood: kirkwood-dma: remove snd_pcm_ops ASoC: dwc: dwc-pcm: remove snd_pcm_ops ASoC: rt5514-spi: remove snd_pcm_ops ASoC: bcm: cygnus-pcm: remove snd_pcm_ops ASoC: fsl: imx-pcm-fiq: remove snd_pcm_ops ASoC: fsl: fsl_dma: remove snd_pcm_ops ASoC: fsl: mpc5200_dma: remove snd_pcm_ops ASoC: fsl: fsl_asrc_dma: remove snd_pcm_ops ASoC: fsl: fsl_dma: don't use snd_soc_rtdcom_lookup() ASoC: soc-generic-dmaengine-pcm: remove snd_pcm_ops ASoC: soc-utils: remove snd_pcm_ops ASoC: soc-component: remove snd_pcm_ops from component driver include/sound/pxa2xx-lib.h | 26 +++++- include/sound/soc-component.h | 41 ++++++++- include/sound/soc.h | 5 + sound/arm/pxa2xx-pcm-lib.c | 80 +++++++++++++--- sound/soc/amd/acp-pcm-dma.c | 63 ++++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 54 +++++------ sound/soc/atmel/atmel-pcm-pdc.c | 48 +++++----- sound/soc/au1x/dbdma2.c | 64 +++++++------ sound/soc/au1x/dma.c | 63 +++++++------ sound/soc/bcm/cygnus-pcm.c | 56 +++++------ sound/soc/codecs/rt5514-spi.c | 41 +++++---- sound/soc/dwc/dwc-pcm.c | 48 +++++----- sound/soc/fsl/fsl_asrc_dma.c | 56 +++++------ sound/soc/fsl/fsl_dma.c | 53 +++++------ sound/soc/fsl/imx-pcm-fiq.c | 56 ++++++----- sound/soc/fsl/mpc5200_dma.c | 51 ++++++----- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 30 +++--- sound/soc/intel/baytrail/sst-baytrail-pcm.c | 52 +++++------ sound/soc/intel/haswell/sst-haswell-pcm.c | 62 +++++++------ sound/soc/intel/skylake/skl-pcm.c | 67 +++++++++----- sound/soc/kirkwood/kirkwood-dma.c | 50 +++++----- .../soc/mediatek/common/mtk-afe-platform-driver.c | 28 +++--- .../soc/mediatek/common/mtk-afe-platform-driver.h | 10 +- sound/soc/mediatek/common/mtk-btcvsd.c | 76 ++++++--------- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 11 ++- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 11 ++- sound/soc/meson/axg-fifo.c | 56 +++++------ sound/soc/meson/axg-fifo.h | 20 +++- sound/soc/meson/axg-frddr.c | 24 ++++- sound/soc/meson/axg-toddr.c | 24 ++++- sound/soc/pxa/mmp-pcm.c | 62 ++++++++----- sound/soc/pxa/pxa-ssp.c | 14 ++- sound/soc/pxa/pxa2xx-ac97.c | 14 ++- sound/soc/pxa/pxa2xx-i2s.c | 14 ++- sound/soc/pxa/pxa2xx-pcm.c | 14 ++- sound/soc/qcom/lpass-platform.c | 70 +++++++------- sound/soc/qcom/qdsp6/q6asm-dai.c | 71 +++++++------- sound/soc/qcom/qdsp6/q6routing.c | 14 +-- sound/soc/samsung/idma.c | 58 ++++++------ sound/soc/sh/dma-sh7760.c | 46 +++++----- sound/soc/sh/fsi.c | 31 ++++--- sound/soc/sh/rcar/core.c | 23 +++-- sound/soc/sh/siu_pcm.c | 44 +++++---- sound/soc/soc-component.c | 76 +++++++-------- sound/soc/soc-core.c | 12 +++ sound/soc/soc-generic-dmaengine-pcm.c | 102 ++++++++++----------- sound/soc/soc-pcm.c | 11 +-- sound/soc/soc-utils.c | 11 +-- sound/soc/sof/pcm.c | 88 +++++++++--------- sound/soc/sprd/sprd-pcm-dma.c | 74 +++++++-------- sound/soc/stm/stm32_adfsdm.c | 42 +++++---- sound/soc/txx9/txx9aclc.c | 48 +++++----- sound/soc/uniphier/aio-dma.c | 51 ++++++----- sound/soc/xilinx/xlnx_formatter_pcm.c | 54 +++++------ sound/soc/xtensa/xtfpga-i2s.c | 30 +++--- 55 files changed, 1311 insertions(+), 1089 deletions(-) -- 2.7.4

1 0

[alsa-devel] [asoc:for-5.5 32/32] sound/soc/codecs/tas2770.c:211:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 1a476abc723e644248dba975b71122fcf878703b [32/32] tas2770: add tas2770 smart PA kernel driver reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout 1a476abc723e644248dba975b71122fcf878703b make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/tas2770.c:211:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2770.c:218:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/soc/codecs/tas2770.c:225:14: sparse: sparse: restricted snd_pcm_format_t degrades to integer >> sound/soc/codecs/tas2770.c:361:58: sparse: sparse: incorrect type in argument 2 (different base types) @@ expected int bitwidth @@ got restricted snd_pcm_formint bitwidth @@ >> sound/soc/codecs/tas2770.c:361:58: sparse: expected int bitwidth >> sound/soc/codecs/tas2770.c:361:58: sparse: got restricted snd_pcm_format_t vim +211 sound/soc/codecs/tas2770.c 204 205 static int tas2770_set_bitwidth(struct tas2770_priv *tas2770, int bitwidth) 206 { 207 int ret; 208 struct snd_soc_component *component = tas2770->component; 209 210 switch (bitwidth) { > 211 case SNDRV_PCM_FORMAT_S16_LE: 212 ret = snd_soc_component_update_bits(component, 213 TAS2770_TDM_CFG_REG2, 214 TAS2770_TDM_CFG_REG2_RXW_MASK, 215 TAS2770_TDM_CFG_REG2_RXW_16BITS); 216 tas2770->v_sense_slot = tas2770->i_sense_slot + 2; 217 break; 218 case SNDRV_PCM_FORMAT_S24_LE: 219 ret = snd_soc_component_update_bits(component, 220 TAS2770_TDM_CFG_REG2, 221 TAS2770_TDM_CFG_REG2_RXW_MASK, 222 TAS2770_TDM_CFG_REG2_RXW_24BITS); 223 tas2770->v_sense_slot = tas2770->i_sense_slot + 4; 224 break; 225 case SNDRV_PCM_FORMAT_S32_LE: 226 ret = snd_soc_component_update_bits(component, 227 TAS2770_TDM_CFG_REG2, 228 TAS2770_TDM_CFG_REG2_RXW_MASK, 229 TAS2770_TDM_CFG_REG2_RXW_32BITS); 230 tas2770->v_sense_slot = tas2770->i_sense_slot + 4; 231 break; 232 233 default: 234 return -EINVAL; 235 } 236 237 tas2770->channel_size = bitwidth; 238 239 ret = snd_soc_component_update_bits(component, 240 TAS2770_TDM_CFG_REG5, 241 TAS2770_TDM_CFG_REG5_VSNS_MASK | 242 TAS2770_TDM_CFG_REG5_50_MASK, 243 TAS2770_TDM_CFG_REG5_VSNS_ENABLE | 244 tas2770->v_sense_slot); 245 if (ret) 246 goto end; 247 ret = snd_soc_component_update_bits(component, 248 TAS2770_TDM_CFG_REG6, 249 TAS2770_TDM_CFG_REG6_ISNS_MASK | 250 TAS2770_TDM_CFG_REG6_50_MASK, 251 TAS2770_TDM_CFG_REG6_ISNS_ENABLE | 252 tas2770->i_sense_slot); 253 254 end: 255 return ret; 256 } 257 258 static int tas2770_set_samplerate(struct tas2770_priv *tas2770, int samplerate) 259 { 260 int ret; 261 struct snd_soc_component *component = tas2770->component; 262 263 switch (samplerate) { 264 case 48000: 265 ret = snd_soc_component_update_bits(component, 266 TAS2770_TDM_CFG_REG0, 267 TAS2770_TDM_CFG_REG0_SMP_MASK, 268 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 269 if (ret) 270 goto end; 271 ret = snd_soc_component_update_bits(component, 272 TAS2770_TDM_CFG_REG0, 273 TAS2770_TDM_CFG_REG0_31_MASK, 274 TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); 275 if (ret) 276 goto end; 277 break; 278 case 44100: 279 ret = snd_soc_component_update_bits(component, 280 TAS2770_TDM_CFG_REG0, 281 TAS2770_TDM_CFG_REG0_SMP_MASK, 282 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 283 if (ret) 284 goto end; 285 ret = snd_soc_component_update_bits(component, 286 TAS2770_TDM_CFG_REG0, 287 TAS2770_TDM_CFG_REG0_31_MASK, 288 TAS2770_TDM_CFG_REG0_31_44_1_48KHZ); 289 if (ret) 290 goto end; 291 break; 292 case 96000: 293 ret = snd_soc_component_update_bits(component, 294 TAS2770_TDM_CFG_REG0, 295 TAS2770_TDM_CFG_REG0_SMP_MASK, 296 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 297 if (ret) 298 goto end; 299 ret = snd_soc_component_update_bits(component, 300 TAS2770_TDM_CFG_REG0, 301 TAS2770_TDM_CFG_REG0_31_MASK, 302 TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); 303 break; 304 case 88200: 305 ret = snd_soc_component_update_bits(component, 306 TAS2770_TDM_CFG_REG0, 307 TAS2770_TDM_CFG_REG0_SMP_MASK, 308 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 309 if (ret) 310 goto end; 311 ret = snd_soc_component_update_bits(component, 312 TAS2770_TDM_CFG_REG0, 313 TAS2770_TDM_CFG_REG0_31_MASK, 314 TAS2770_TDM_CFG_REG0_31_88_2_96KHZ); 315 break; 316 case 19200: 317 ret = snd_soc_component_update_bits(component, 318 TAS2770_TDM_CFG_REG0, 319 TAS2770_TDM_CFG_REG0_SMP_MASK, 320 TAS2770_TDM_CFG_REG0_SMP_48KHZ); 321 if (ret) 322 goto end; 323 ret = snd_soc_component_update_bits(component, 324 TAS2770_TDM_CFG_REG0, 325 TAS2770_TDM_CFG_REG0_31_MASK, 326 TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); 327 if (ret) 328 goto end; 329 break; 330 case 17640: 331 ret = snd_soc_component_update_bits(component, 332 TAS2770_TDM_CFG_REG0, 333 TAS2770_TDM_CFG_REG0_SMP_MASK, 334 TAS2770_TDM_CFG_REG0_SMP_44_1KHZ); 335 if (ret) 336 goto end; 337 ret = snd_soc_component_update_bits(component, 338 TAS2770_TDM_CFG_REG0, 339 TAS2770_TDM_CFG_REG0_31_MASK, 340 TAS2770_TDM_CFG_REG0_31_176_4_192KHZ); 341 break; 342 default: 343 ret = -EINVAL; 344 } 345 346 end: 347 if (!ret) 348 tas2770->sampling_rate = samplerate; 349 return ret; 350 } 351 352 static int tas2770_hw_params(struct snd_pcm_substream *substream, 353 struct snd_pcm_hw_params *params, 354 struct snd_soc_dai *dai) 355 { 356 struct snd_soc_component *component = dai->component; 357 struct tas2770_priv *tas2770 = 358 snd_soc_component_get_drvdata(component); 359 int ret; 360 > 361 ret = tas2770_set_bitwidth(tas2770, params_format(params)); 362 if (ret < 0) 363 goto end; 364 365 366 ret = tas2770_set_samplerate(tas2770, params_rate(params)); 367 368 end: 369 return ret; 370 } 371 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-5.5 29/32] sound/core/pcm_dmaengine.c:429:50: sparse: sparse: restricted snd_pcm_format_t degrades to integer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: e957204e732bc2916a241dc61dd7dd14e9a98350 [29/32] ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout e957204e732bc2916a241dc61dd7dd14e9a98350 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/core/pcm_dmaengine.c:429:50: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:55: sparse: sparse: restricted snd_pcm_format_t degrades to integer sound/core/pcm_dmaengine.c:429:79: sparse: sparse: restricted snd_pcm_format_t degrades to integer vim +429 sound/core/pcm_dmaengine.c 371 372 /** 373 * snd_dmaengine_pcm_refine_runtime_hwparams - Refine runtime hw params 374 * @substream: PCM substream 375 * @dma_data: DAI DMA data 376 * @hw: PCM hw params 377 * @chan: DMA channel to use for data transfers 378 * 379 * Returns 0 on success, a negative error code otherwise. 380 * 381 * This function will query DMA capability, then refine the pcm hardware 382 * parameters. 383 */ 384 int snd_dmaengine_pcm_refine_runtime_hwparams( 385 struct snd_pcm_substream *substream, 386 struct snd_dmaengine_dai_dma_data *dma_data, 387 struct snd_pcm_hardware *hw, 388 struct dma_chan *chan) 389 { 390 struct dma_slave_caps dma_caps; 391 u32 addr_widths = BIT(DMA_SLAVE_BUSWIDTH_1_BYTE) | 392 BIT(DMA_SLAVE_BUSWIDTH_2_BYTES) | 393 BIT(DMA_SLAVE_BUSWIDTH_4_BYTES); 394 snd_pcm_format_t i; 395 int ret = 0; 396 397 if (!hw || !chan || !dma_data) 398 return -EINVAL; 399 400 ret = dma_get_slave_caps(chan, &dma_caps); 401 if (ret == 0) { 402 if (dma_caps.cmd_pause && dma_caps.cmd_resume) 403 hw->info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; 404 if (dma_caps.residue_granularity <= DMA_RESIDUE_GRANULARITY_SEGMENT) 405 hw->info |= SNDRV_PCM_INFO_BATCH; 406 407 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) 408 addr_widths = dma_caps.dst_addr_widths; 409 else 410 addr_widths = dma_caps.src_addr_widths; 411 } 412 413 /* 414 * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep 415 * hw.formats set to 0, meaning no restrictions are in place. 416 * In this case it's the responsibility of the DAI driver to 417 * provide the supported format information. 418 */ 419 if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) 420 /* 421 * Prepare formats mask for valid/allowed sample types. If the 422 * dma does not have support for the given physical word size, 423 * it needs to be masked out so user space can not use the 424 * format which produces corrupted audio. 425 * In case the dma driver does not implement the slave_caps the 426 * default assumption is that it supports 1, 2 and 4 bytes 427 * widths. 428 */ > 429 for (i = SNDRV_PCM_FORMAT_FIRST; i <= SNDRV_PCM_FORMAT_LAST; i++) { 430 int bits = snd_pcm_format_physical_width(i); 431 432 /* 433 * Enable only samples with DMA supported physical 434 * widths 435 */ 436 switch (bits) { 437 case 8: 438 case 16: 439 case 24: 440 case 32: 441 case 64: 442 if (addr_widths & (1 << (bits / 8))) 443 hw->formats |= pcm_format_to_bits(i); 444 break; 445 default: 446 /* Unsupported types */ 447 break; 448 } 449 } 450 451 return ret; 452 } 453 EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_refine_runtime_hwparams); 454 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [asoc:for-5.5 2/32] sound/soc/fsl/fsl_mqs.c:182:38: sparse: sparse: Using plain integer as NULL pointer
by kbuild test robot 01 Oct '19

01 Oct '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 head: 1a476abc723e644248dba975b71122fcf878703b commit: 9e28f6532c611c0c3fa759d2101aba9f0d41e860 [2/32] ASoC: fsl_mqs: Add MQS component driver reproduce: # apt-get install sparse # sparse version: v0.6.1-rc1-37-gd466a02-dirty git checkout 9e28f6532c611c0c3fa759d2101aba9f0d41e860 make ARCH=x86_64 allmodconfig make C=1 CF='-fdiagnostic-prefix -D__CHECK_ENDIAN__' If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> sparse warnings: (new ones prefixed by >>) >> sound/soc/fsl/fsl_mqs.c:182:38: sparse: sparse: Using plain integer as NULL pointer vim +182 sound/soc/fsl/fsl_mqs.c 178 179 static int fsl_mqs_probe(struct platform_device *pdev) 180 { 181 struct device_node *np = pdev->dev.of_node; > 182 struct device_node *gpr_np = 0; 183 struct fsl_mqs *mqs_priv; 184 void __iomem *regs; 185 int ret = 0; 186 187 mqs_priv = devm_kzalloc(&pdev->dev, sizeof(*mqs_priv), GFP_KERNEL); 188 if (!mqs_priv) 189 return -ENOMEM; 190 191 /* On i.MX6sx the MQS control register is in GPR domain 192 * But in i.MX8QM/i.MX8QXP the control register is moved 193 * to its own domain. 194 */ 195 if (of_device_is_compatible(np, "fsl,imx8qm-mqs")) 196 mqs_priv->use_gpr = false; 197 else 198 mqs_priv->use_gpr = true; 199 200 if (mqs_priv->use_gpr) { 201 gpr_np = of_parse_phandle(np, "gpr", 0); 202 if (IS_ERR(gpr_np)) { 203 dev_err(&pdev->dev, "failed to get gpr node by phandle\n"); 204 ret = PTR_ERR(gpr_np); 205 goto out; 206 } 207 208 mqs_priv->regmap = syscon_node_to_regmap(gpr_np); 209 if (IS_ERR(mqs_priv->regmap)) { 210 dev_err(&pdev->dev, "failed to get gpr regmap\n"); 211 ret = PTR_ERR(mqs_priv->regmap); 212 goto out; 213 } 214 } else { 215 regs = devm_platform_ioremap_resource(pdev, 0); 216 if (IS_ERR(regs)) 217 return PTR_ERR(regs); 218 219 mqs_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev, 220 "core", 221 regs, 222 &fsl_mqs_regmap_config); 223 if (IS_ERR(mqs_priv->regmap)) { 224 dev_err(&pdev->dev, "failed to init regmap: %ld\n", 225 PTR_ERR(mqs_priv->regmap)); 226 return PTR_ERR(mqs_priv->regmap); 227 } 228 229 mqs_priv->ipg = devm_clk_get(&pdev->dev, "core"); 230 if (IS_ERR(mqs_priv->ipg)) { 231 dev_err(&pdev->dev, "failed to get the clock: %ld\n", 232 PTR_ERR(mqs_priv->ipg)); 233 goto out; 234 } 235 } 236 237 mqs_priv->mclk = devm_clk_get(&pdev->dev, "mclk"); 238 if (IS_ERR(mqs_priv->mclk)) { 239 dev_err(&pdev->dev, "failed to get the clock: %ld\n", 240 PTR_ERR(mqs_priv->mclk)); 241 goto out; 242 } 243 244 dev_set_drvdata(&pdev->dev, mqs_priv); 245 pm_runtime_enable(&pdev->dev); 246 247 return devm_snd_soc_register_component(&pdev->dev, &soc_codec_fsl_mqs, 248 &fsl_mqs_dai, 1); 249 out: 250 if (!IS_ERR(gpr_np)) 251 of_node_put(gpr_np); 252 253 return ret; 254 } 255 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH 00/13] ASoC: SOF: fixes and improvements
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

No new features in this patchset but fixes and improvements identified during validation/product hardening/SoundWire development. I realize we are at the end of the merge window so will let Takashi/Mark decide if this series can be provided as fixes for v5.4 or queued for later. Kai Vehmanen (2): ASoC: SOF: pcm: fix resource leak in hw_free ASoC: SOF: Intel: hda: fix warnings during FW load Keyon Jie (1): ASoC: SOF: topology: fix parse fail issue for byte/bool tuple types Liam Girdwood (3): ASoC: SOF: ipc: retain DSP context after FW exception. ASoC: SOF: Intel: initialise and verify FW crash dump data. ASoC: SOF: Intel: Add context data to any IPC timeout. Marcin Rajwa (2): ASoC: SOF: ipc: introduce no_stream_position in sof_ipc_stream_params struct ASoC: SOF: Intel: hda: fix reset of host_period_bytes Pan Xiuli (1): ASoC: SOF: pcm: harden PCM STOP sequence Pierre-Louis Bossart (2): ASoC: SOF: loader: fix kernel oops on firmware boot failure ASoC: SOF: trace: move to opt-in with Kconfig and module parameter Ranjani Sridharan (2): ASoC: SOF: core: check for mandatory fw_ready op during SOF probe ASoC: SOF: Intel: hda: Disable DMI L1 entry during capture include/sound/sof/stream.h | 4 +-- sound/soc/sof/Kconfig | 16 ++++++++++++ sound/soc/sof/core.c | 28 +++++++++++++++----- sound/soc/sof/debug.c | 16 ++++++++++++ sound/soc/sof/intel/Kconfig | 10 +++++++ sound/soc/sof/intel/bdw.c | 29 +++++++++++++++++++- sound/soc/sof/intel/byt.c | 29 +++++++++++++++++++- sound/soc/sof/intel/hda-ctrl.c | 12 +++------ sound/soc/sof/intel/hda-loader.c | 1 + sound/soc/sof/intel/hda-pcm.c | 15 ++++++++--- sound/soc/sof/intel/hda-stream.c | 45 +++++++++++++++++++++++++++----- sound/soc/sof/intel/hda.c | 7 +++++ sound/soc/sof/intel/hda.h | 5 +++- sound/soc/sof/ipc.c | 10 +------ sound/soc/sof/loader.c | 4 +-- sound/soc/sof/pcm.c | 35 +++++++++++++++++-------- sound/soc/sof/sof-priv.h | 17 ++++++++---- sound/soc/sof/topology.c | 4 ++- sound/soc/sof/trace.c | 17 +++++++++++- 19 files changed, 245 insertions(+), 59 deletions(-) -- 2.20.1

2 26

[alsa-devel] [PATCH] ASoC: wm_adsp: Fix theoretical NULL pointer for alg_region
by Charles Keepax 01 Oct '19

01 Oct '19

From: Li Xu <li.xu(a)cirrus.com> Fix potential NULL pointer dereference for alg_region in wm_adsp_buffer_parse_legacy. In practice this can never happen as loading the firmware should have failed at the wm_adsp2_setup_algs stage, however probably better for the code to be robust against future changes and this is more helpful for static analysis. Signed-off-by: Li Xu <li.xu(a)cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ae28d9907c30f..85396d920e0a9 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3697,11 +3697,16 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) u32 xmalg, addr, magic; int i, ret; + alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); + if (!alg_region) { + adsp_err(dsp, "No algorithm region found\n"); + return -EINVAL; + } + buf = wm_adsp_buffer_alloc(dsp); if (!buf) return -ENOMEM; - alg_region = wm_adsp_find_alg_region(dsp, WMFW_ADSP2_XM, dsp->fw_id); xmalg = dsp->ops->sys_config_size / sizeof(__be32); addr = alg_region->base + xmalg + ALG_XM_FIELD(magic); -- 2.11.0

2 1

[alsa-devel] [PATCH] ASoC: madera: Add support for using MCLK3
by Charles Keepax 01 Oct '19

01 Oct '19

Some Madera CODECs have a third MCLK pin allow this to be sourced for the FLLs and system clocks. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/madera.h | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/madera.h b/sound/soc/codecs/madera.h index 1f3e8e230cf25..6d8938a3fb647 100644 --- a/sound/soc/codecs/madera.h +++ b/sound/soc/codecs/madera.h @@ -27,6 +27,7 @@ #define MADERA_FLL_SRC_NONE -1 #define MADERA_FLL_SRC_MCLK1 0 #define MADERA_FLL_SRC_MCLK2 1 +#define MADERA_FLL_SRC_MCLK3 2 #define MADERA_FLL_SRC_SLIMCLK 3 #define MADERA_FLL_SRC_FLL1 4 #define MADERA_FLL_SRC_FLL2 5 @@ -51,6 +52,7 @@ #define MADERA_CLK_SRC_MCLK1 0x0 #define MADERA_CLK_SRC_MCLK2 0x1 +#define MADERA_CLK_SRC_MCLK3 0x2 #define MADERA_CLK_SRC_FLL1 0x4 #define MADERA_CLK_SRC_FLL2 0x5 #define MADERA_CLK_SRC_FLL3 0x6 -- 2.11.0

2 1

[alsa-devel] [PATCH 0/4] ASoC: Intel: boards: improvements for RT5651/82-based platforms
by Pierre-Louis Bossart 01 Oct '19

01 Oct '19

RT5651/RT5682 boards are used for SOF validation. This patchset corrects problems identified during module load/reload tests and an error for DMIC support in some configurations. These fixes are not critical and can be queued for v5.5 Bard Liao (1): ASoC: intel: sof_rt5682: use separate route map for dmic Jaska Uimonen (3): ASoC: intel: bytcr_rt5651: add null check to support_button_press ASoC: rt5682: add NULL handler to set_jack function ASoC: intel: sof_rt5682: add remove function to disable jack sound/soc/codecs/rt5651.c | 3 ++ sound/soc/codecs/rt5682.c | 12 +++++-- sound/soc/intel/boards/sof_rt5682.c | 51 ++++++++++++++++++++++++++--- 3 files changed, 60 insertions(+), 6 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH] ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency
by Arnd Bergmann 01 Oct '19

01 Oct '19

CONFIG_SND_SOC_SOF_IMX depends on CONFIG_SND_SOC_SOF, but is in turn referenced by the sof-of-dev driver. This creates a reverse dependency that manifests in a link error when CONFIG_SND_SOC_SOF_OF is built-in but CONFIG_SND_SOC_SOF_IMX=m: sound/soc/sof/sof-of-dev.o:(.data+0x118): undefined reference to `sof_imx8_ops' Make the latter a 'bool' symbol and change the Makefile so the imx8 driver is compiled the same way as the driver using it. A nicer way would be to reverse the layering and move all the imx specific bits of sof-of-dev.c into the imx driver itself, which can then call into the common code. Doing this would need more testing and can be done if we add another driver like the first one. Fixes: f4df4e4042b0 ("ASoC: SOF: imx8: Fix COMPILE_TEST error") Fixes: 202acc565a1f ("ASoC: SOF: imx: Add i.MX8 HW support") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/sof/imx/Kconfig | 2 +- sound/soc/sof/imx/Makefile | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..a3891654a1fc 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -12,7 +12,7 @@ config SND_SOC_SOF_IMX_TOPLEVEL if SND_SOC_SOF_IMX_TOPLEVEL config SND_SOC_SOF_IMX8 - tristate "SOF support for i.MX8" + bool "SOF support for i.MX8" depends on IMX_SCU depends on IMX_DSP help diff --git a/sound/soc/sof/imx/Makefile b/sound/soc/sof/imx/Makefile index 6ef908e8c807..9e8f35df0ff2 100644 --- a/sound/soc/sof/imx/Makefile +++ b/sound/soc/sof/imx/Makefile @@ -1,4 +1,6 @@ # SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) snd-sof-imx8-objs := imx8.o -obj-$(CONFIG_SND_SOC_SOF_IMX8) += snd-sof-imx8.o +ifdef CONFIG_SND_SOC_SOF_IMX8 +obj-$(CONFIG_SND_SOC_SOF_OF) += snd-sof-imx8.o +endif -- 2.20.0

3 5

[alsa-devel] [PATCH] ASoC: max98373: check for device node before parsing
by Sathyanarayana Nujella 01 Oct '19

01 Oct '19

Below Oops is caused in a system which uses ACPI instead of device node: of_get_named_gpiod_flags: can't parse 'maxim,reset-gpio' property of node '(null)[0]' BUG: kernel NULL pointer dereference, address: 0000000000000010 This patch avoids NULL pointer deferencing by adding a check before parsing and initializes to make reset-gpio pin as invalid. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Jairaj Arava <jairaj.arava(a)intel.com> --- sound/soc/codecs/max98373.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/max98373.c b/sound/soc/codecs/max98373.c index e609abcf3220..eb709d528259 100644 --- a/sound/soc/codecs/max98373.c +++ b/sound/soc/codecs/max98373.c @@ -901,16 +901,20 @@ static void max98373_slot_config(struct i2c_client *i2c, max98373->i_slot = value & 0xF; else max98373->i_slot = 1; - - max98373->reset_gpio = of_get_named_gpio(dev->of_node, + if (dev->of_node) { + max98373->reset_gpio = of_get_named_gpio(dev->of_node, "maxim,reset-gpio", 0); - if (!gpio_is_valid(max98373->reset_gpio)) { - dev_err(dev, "Looking up %s property in node %s failed %d\n", - "maxim,reset-gpio", dev->of_node->full_name, - max98373->reset_gpio); + if (!gpio_is_valid(max98373->reset_gpio)) { + dev_err(dev, "Looking up %s property in node %s failed %d\n", + "maxim,reset-gpio", dev->of_node->full_name, + max98373->reset_gpio); + } else { + dev_dbg(dev, "maxim,reset-gpio=%d", + max98373->reset_gpio); + } } else { - dev_dbg(dev, "maxim,reset-gpio=%d", - max98373->reset_gpio); + /* this makes reset_gpio as invalid */ + max98373->reset_gpio = -1; } if (!device_property_read_u32(dev, "maxim,spkfb-slot-no", &value)) -- 2.7.4

2 1

[alsa-devel] [PATCH V2 1/2] ASoC: fsl_mqs: add DT binding documentation
by Shengjiu Wang 01 Oct '19

01 Oct '19

Add the DT binding documentation for NXP MQS driver Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 -refine the comments for properties .../devicetree/bindings/sound/fsl,mqs.txt | 36 +++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.txt b/Documentation/devicetree/bindings/sound/fsl,mqs.txt new file mode 100644 index 000000000000..40353fc30255 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.txt @@ -0,0 +1,36 @@ +fsl,mqs audio CODEC + +Required properties: + - compatible : Must contain one of "fsl,imx6sx-mqs", "fsl,codec-mqs" + "fsl,imx8qm-mqs", "fsl,imx8qxp-mqs". + - clocks : A list of phandles + clock-specifiers, one for each entry in + clock-names + - clock-names : "mclk" - must required. + "core" - required if compatible is "fsl,imx8qm-mqs", it + is for register access. + - gpr : A phandle of General Purpose Registers in IOMUX Controller. + Required if compatible is "fsl,imx6sx-mqs". + +Required if compatible is "fsl,imx8qm-mqs": + - power-domains: A phandle of PM domain provider node. + - reg: Offset and length of the register set for the device. + +Example: + +mqs: mqs { + compatible = "fsl,imx6sx-mqs"; + gpr = <&gpr>; + clocks = <&clks IMX6SX_CLK_SAI1>; + clock-names = "mclk"; + status = "disabled"; +}; + +mqs: mqs@59850000 { + compatible = "fsl,imx8qm-mqs"; + reg = <0x59850000 0x10000>; + clocks = <&clk IMX8QM_AUD_MQS_IPG>, + <&clk IMX8QM_AUD_MQS_HMCLK>; + clock-names = "core", "mclk"; + power-domains = <&pd_mqs0>; + status = "disabled"; +}; -- 2.21.0

4 6

[alsa-devel] [PATCH v2] ASoC: Intel: cht_bsw_rt5645: Add quirk for boards using pmc_plt_clk_0
by Sam McNally 01 Oct '19

01 Oct '19

As of commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL"), the cht_bsw_rt5645 driver needs to enable the clock it's using for the codec's mclk. It does this from commit 7735bce05a9c ("ASoC: Intel: boards: use devm_clk_get() unconditionally"), enabling pmc_plt_clk_3. However, Strago family Chromebooks use pmc_plt_clk_0 for the codec mclk, resulting in white noise with some digital microphones. Add a DMI-based quirk for Strago family Chromebooks to use pmc_plt_clk_0 instead - mirroring the changes made to cht_bsw_max98090_ti in commit a182ecd3809c ("ASoC: intel: cht_bsw_max98090_ti: Add quirk for boards using pmc_plt_clk_0") and making use of the existing dmi_check_system() call and related infrastructure added in commit 22af29114eb4 ("ASoC: Intel: cht-bsw-rt5645: add quirks for SSP0/AIF1/AIF2 routing"). Signed-off-by: Sam McNally <sammc(a)chromium.org> --- Changes in v2: - Added to the description references to more related commits (similar cht_bsw_max98090_ti clock quirks inspiring this change and and the cht_bsw_rt5645 quirks setup) sound/soc/intel/boards/cht_bsw_rt5645.c | 26 +++++++++++++++++++------ 1 file changed, 20 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 8879c3be29d5..c68a5b85a4a0 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -48,6 +48,7 @@ struct cht_mc_private { #define CHT_RT5645_SSP2_AIF2 BIT(16) /* default is using AIF1 */ #define CHT_RT5645_SSP0_AIF1 BIT(17) #define CHT_RT5645_SSP0_AIF2 BIT(18) +#define CHT_RT5645_PMC_PLT_CLK_0 BIT(19) static unsigned long cht_rt5645_quirk = 0; @@ -59,6 +60,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP0_AIF1 enabled"); if (cht_rt5645_quirk & CHT_RT5645_SSP0_AIF2) dev_info(dev, "quirk SSP0_AIF2 enabled"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + dev_info(dev, "quirk PMC_PLT_CLK_0 enabled"); } static int platform_clock_control(struct snd_soc_dapm_widget *w, @@ -226,15 +229,21 @@ static int cht_aif1_hw_params(struct snd_pcm_substream *substream, return 0; } -/* uncomment when we have a real quirk static int cht_rt5645_quirk_cb(const struct dmi_system_id *id) { cht_rt5645_quirk = (unsigned long)id->driver_data; return 1; } -*/ static const struct dmi_system_id cht_rt5645_quirk_table[] = { + { + /* Strago family Chromebooks */ + .callback = cht_rt5645_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Intel_Strago"), + }, + .driver_data = (void *)CHT_RT5645_PMC_PLT_CLK_0, + }, { }, }; @@ -526,6 +535,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) int dai_index = 0; int ret_val = 0; int i; + const char *mclk_name; drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_KERNEL); if (!drv) @@ -662,11 +672,15 @@ static int snd_cht_mc_probe(struct platform_device *pdev) if (ret_val) return ret_val; - drv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); + if (cht_rt5645_quirk & CHT_RT5645_PMC_PLT_CLK_0) + mclk_name = "pmc_plt_clk_0"; + else + mclk_name = "pmc_plt_clk_3"; + + drv->mclk = devm_clk_get(&pdev->dev, mclk_name); if (IS_ERR(drv->mclk)) { - dev_err(&pdev->dev, - "Failed to get MCLK from pmc_plt_clk_3: %ld\n", - PTR_ERR(drv->mclk)); + dev_err(&pdev->dev, "Failed to get MCLK from %s: %ld\n", + mclk_name, PTR_ERR(drv->mclk)); return PTR_ERR(drv->mclk); } -- 2.23.0.237.gc6a4ce50a0-goog

3 2

[alsa-devel] [PATCH v3 1/2] ASoC: dt-bindings: Convert Allwinner A10 codec to a schema
by Maxime Ripard 01 Oct '19

01 Oct '19

From: Maxime Ripard <maxime.ripard(a)bootlin.com> The Allwinner SoCs have an embedded audio codec that is supported in Linux, with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that controller over to a YAML schemas. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v2: - Change the audio-routing values to an enum, and enforce boundaries on the size - Add restrictions to the possible values of audio-routing Changes from v1: - Fix subject prefix --- .../sound/allwinner,sun4i-a10-codec.yaml | 262 ++++++++++++++++++ .../devicetree/bindings/sound/sun4i-codec.txt | 94 ------- 2 files changed, 262 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml new file mode 100644 index 000000000000..faa75b91c072 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-codec.yaml @@ -0,0 +1,262 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-codec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 Codec Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + enum: + - allwinner,sun4i-a10-codec + - allwinner,sun6i-a31-codec + - allwinner,sun7i-a20-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: codec + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + + allwinner,audio-routing: + description: |- + A list of the connections between audio components. Each entry + is a pair of strings, the first being the connection's sink, the + second being the connection's source. + allOf: + - $ref: /schemas/types.yaml#definitions/non-unique-string-array + - minItems: 2 + maxItems: 18 + enum: + # Audio Pins on the SoC + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + + # Microphone Biases from the SoC + - HBIAS + - MBIAS + + # Board Connectors + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + allwinner,codec-analog-controls: + $ref: /schemas/types.yaml#/definitions/phandle + description: Phandle to the codec analog controls in the PRCM + + allwinner,pa-gpios: + description: GPIO to enable the external amplifier + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +allOf: + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + - allwinner,sun8i-a23-codec + - allwinner,sun8i-h3-codec + - allwinner,sun8i-v3s-codec + + then: + if: + properties: + compatible: + const: allwinner,sun6i-a31-codec + + then: + required: + - resets + - allwinner,audio-routing + + else: + required: + - resets + - allwinner,audio-routing + - allwinner,codec-analog-controls + + - if: + properties: + compatible: + enum: + - allwinner,sun6i-a31-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - MIC3 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-a23-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-h3-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - LINEIN + - LINEOUT + - MIC1 + - MIC2 + - HBIAS + - MBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + + - if: + properties: + compatible: + enum: + - allwinner,sun8i-v3s-codec + + then: + properties: + allwinner,audio-routing: + enum: + - HP + - HPCOM + - MIC1 + - HBIAS + - Headphone + - Headset Mic + - Line In + - Line Out + - Mic + - Speaker + +additionalProperties: false + +examples: + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun7i-a20-codec"; + reg = <0x01c22c00 0x40>; + interrupts = <0 30 4>; + clocks = <&apb0_gates 0>, <&codec_clk>; + clock-names = "apb", "codec"; + dmas = <&dma 0 19>, <&dma 0 19>; + dma-names = "rx", "tx"; + }; + + - | + codec@1c22c00 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun6i-a31-codec"; + reg = <0x01c22c00 0x98>; + interrupts = <0 29 4>; + clocks = <&ccu 61>, <&ccu 135>; + clock-names = "apb", "codec"; + resets = <&ccu 42>; + dmas = <&dma 15>, <&dma 15>; + dma-names = "rx", "tx"; + allwinner,audio-routing = + "Headphone", "HP", + "Speaker", "LINEOUT", + "LINEIN", "Line In", + "MIC1", "MBIAS", + "MIC1", "Mic", + "MIC2", "HBIAS", + "MIC2", "Headset Mic"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-codec.txt b/Documentation/devicetree/bindings/sound/sun4i-codec.txt deleted file mode 100644 index 66579bbd3294..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-codec.txt +++ /dev/null @@ -1,94 +0,0 @@ -* Allwinner A10 Codec - -Required properties: -- compatible: must be one of the following compatibles: - - "allwinner,sun4i-a10-codec" - - "allwinner,sun6i-a31-codec" - - "allwinner,sun7i-a20-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- reg: must contain the registers location and length -- interrupts: must contain the codec interrupt -- dmas: DMA channels for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry - in clock-names. -- clock-names: should contain the following: - - "apb": the parent APB clock for this controller - - "codec": the parent module clock - -Optional properties: -- allwinner,pa-gpios: gpio to enable external amplifier - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-codec" - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- resets: phandle to the reset control for this device -- allwinner,audio-routing: A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. Valid names include: - - Audio pins on the SoC: - "HP" - "HPCOM" - "LINEIN" (not on sun8i-v3s) - "LINEOUT" (not on sun8i-a23 or sun8i-v3s) - "MIC1" - "MIC2" (not on sun8i-v3s) - "MIC3" (sun6i-a31 only) - - Microphone biases from the SoC: - "HBIAS" - "MBIAS" (not on sun8i-v3s) - - Board connectors: - "Headphone" - "Headset Mic" - "Line In" - "Line Out" - "Mic" - "Speaker" - -Required properties for the following compatibles: - - "allwinner,sun8i-a23-codec" - - "allwinner,sun8i-h3-codec" - - "allwinner,sun8i-v3s-codec" -- allwinner,codec-analog-controls: A phandle to the codec analog controls - block in the PRCM. - -Example: -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun7i-a20-codec"; - reg = <0x01c22c00 0x40>; - interrupts = <0 30 4>; - clocks = <&apb0_gates 0>, <&codec_clk>; - clock-names = "apb", "codec"; - dmas = <&dma 0 19>, <&dma 0 19>; - dma-names = "rx", "tx"; -}; - -codec: codec@1c22c00 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun6i-a31-codec"; - reg = <0x01c22c00 0x98>; - interrupts = <GIC_SPI 29 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&ccu CLK_APB1_CODEC>, <&ccu CLK_CODEC>; - clock-names = "apb", "codec"; - resets = <&ccu RST_APB1_CODEC>; - dmas = <&dma 15>, <&dma 15>; - dma-names = "rx", "tx"; - allwinner,audio-routing = - "Headphone", "HP", - "Speaker", "LINEOUT", - "LINEIN", "Line In", - "MIC1", "MBIAS", - "MIC1", "Mic", - "MIC2", "HBIAS", - "MIC2", "Headset Mic"; -}; -- 2.21.0

4 4

[alsa-devel] [PATCH] ASoC: xlnx: Use devm_platform_ioremap_resource() in xlnx_formatter_pcm_probe()
by Markus Elfring 01 Oct '19

01 Oct '19

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Thu, 19 Sep 2019 17:27:57 +0200 Simplify this function implementation by using a known wrapper function. This issue was detected by using the Coccinelle software. Signed-off-by: Markus Elfring <elfring(a)users.sourceforge.net> --- sound/soc/xilinx/xlnx_formatter_pcm.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 48970efe7838..fb652e73abeb 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -564,7 +564,6 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) int ret; u32 val; struct xlnx_pcm_drv_data *aud_drv_data; - struct resource *res; struct device *dev = &pdev->dev; aud_drv_data = devm_kzalloc(dev, sizeof(*aud_drv_data), GFP_KERNEL); @@ -584,13 +583,7 @@ static int xlnx_formatter_pcm_probe(struct platform_device *pdev) return ret; } - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "audio formatter node:addr to resource failed\n"); - ret = -ENXIO; - goto clk_err; - } - aud_drv_data->mmio = devm_ioremap_resource(dev, res); + aud_drv_data->mmio = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(aud_drv_data->mmio)) { dev_err(dev, "audio formatter ioremap failed\n"); ret = PTR_ERR(aud_drv_data->mmio); -- 2.23.0

2 1

[alsa-devel] [PATCH v2] ASoC: soc-component.h: remove GPL explanation from header
by Kuninori Morimoto 01 Oct '19

01 Oct '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, thus, GPL explanation is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark - this is v2 of "[PATCH] ASoC: soc-component.h: tidyup for soc-component.h" - Subject was renamed - keep "soc.h" - do nothing to header-check include/sound/soc-component.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..1a832695 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,10 +3,6 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: dt-bindings: mt8183: add missing update
by Tzung-Bi Shih 01 Oct '19

01 Oct '19

Headset codec is optional. Add missing update to DT binding document. Fixes: a962a809e5e4 ("ASoC: mediatek: mt8183: make headset codec optional") Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../bindings/sound/mt8183-mt6358-ts3a227-max98357.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt index d6d5207fa996..17ff3892f439 100644 --- a/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt +++ b/Documentation/devicetree/bindings/sound/mt8183-mt6358-ts3a227-max98357.txt @@ -2,9 +2,11 @@ MT8183 with MT6358, TS3A227 and MAX98357 CODECS Required properties: - compatible : "mediatek,mt8183_mt6358_ts3a227_max98357" -- mediatek,headset-codec: the phandles of ts3a227 codecs - mediatek,platform: the phandle of MT8183 ASoC platform +Optional properties: +- mediatek,headset-codec: the phandles of ts3a227 codecs + Example: sound { -- 2.23.0.351.gc4317032e6-goog

2 1

[alsa-devel] [PATCH v2 00/14] ASoC: soc-core cleanup - step 3
by Kuninori Morimoto 01 Oct '19

01 Oct '19

Hi Mark These are v2 of soc-core.c cleanup (maybe) step 3. These try to clenaup around kmalloc/kfree related code. Basically these do nothing from technical point of view. I added +1 extra patch as [00/14] Kuninori Morimoto (14): ASoC: soc-core: move soc_free_pcm_runtime() ASoC: soc-core: merge soc_add_pcm_runtime() into soc_new_pcm_runtime() ASoC: soc-core: call list_del(&rtd->list) at soc_free_pcm_runtime() ASoC: soc-core: create rtd->codec_dais first ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() ASoC: soc-core: merge soc_free_pcm_runtime() and soc_rtd_free() ASoC: soc-core: tidyup soc_new_pcm_runtime() alloc order ASoC: soc-core: remove snd_soc_rtdcom_del_all() ASoC: soc-core: use devm_kzalloc() for rtd ASoC: soc-core: remove soc_remove_dai_links() ASoC: soc-core: add soc_setup_card_name() ASoC: soc-core: use devm_xxx for component related resource ASoC: soc-core: setup card->rtd_num at snd_soc_instantiate_card() ASoC: soc-ops: use snd_soc_card_get_kcontrol() at snd_soc_limit_volume() include/sound/soc.h | 1 - sound/soc/soc-core.c | 287 ++++++++++++++++++++++++++------------------------- sound/soc/soc-ops.c | 11 +- 3 files changed, 151 insertions(+), 148 deletions(-) -- 2.7.4

2 22

[alsa-devel] [PATCH v7] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 01 Oct '19

01 Oct '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: rt5663: clean up indentation issues
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There are two break statements that are indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5663.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2943692f66ed..e6c1ec6c426e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3644,7 +3644,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_PWR_ANLG_1, RT5663_LDO1_DVO_MASK | RT5663_AMP_HP_MASK, RT5663_LDO1_DVO_0_9V | RT5663_AMP_HP_3X); - break; + break; case CODEC_VER_0: regmap_update_bits(rt5663->regmap, RT5663_DIG_MISC, RT5663_DIG_GATE_CTRL_MASK, RT5663_DIG_GATE_CTRL_EN); @@ -3663,7 +3663,7 @@ static int rt5663_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5663->regmap, RT5663_TDM_2, RT5663_DATA_SWAP_ADCDAT1_MASK, RT5663_DATA_SWAP_ADCDAT1_LL); - break; + break; default: dev_err(&i2c->dev, "%s:Unknown codec type\n", __func__); } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: wcd9335: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is an if statement that is indented one level too deeply, remove the extraneous tabs. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/wcd9335.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index f318403133e9..f11ffa28683b 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -2837,11 +2837,11 @@ static int wcd9335_codec_enable_dec(struct snd_soc_dapm_widget *w, TX_HPF_CUT_OFF_FREQ_MASK) >> 5; snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x10); snd_soc_component_update_bits(comp, dec_cfg_reg, 0x08, 0x00); - if (hpf_coff_freq != CF_MIN_3DB_150HZ) { - snd_soc_component_update_bits(comp, dec_cfg_reg, - TX_HPF_CUT_OFF_FREQ_MASK, - hpf_coff_freq << 5); - } + if (hpf_coff_freq != CF_MIN_3DB_150HZ) { + snd_soc_component_update_bits(comp, dec_cfg_reg, + TX_HPF_CUT_OFF_FREQ_MASK, + hpf_coff_freq << 5); + } break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(comp, tx_vol_ctl_reg, 0x10, 0x00); -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: stm32: sai: clean up indentation issue
by Colin King 01 Oct '19

01 Oct '19

From: Colin Ian King <colin.king(a)canonical.com> There is a statement that is indented one level too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/stm/stm32_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index ef4273361d0d..e20267504b16 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -100,7 +100,7 @@ static int stm32_sai_sync_conf_provider(struct stm32_sai_data *sai, int synco) dev_err(&sai->pdev->dev, "%pOFn%s already set as sync provider\n", sai->pdev->dev.of_node, prev_synco == STM_SAI_SYNC_OUT_A ? "A" : "B"); - stm32_sai_pclk_disable(&sai->pdev->dev); + stm32_sai_pclk_disable(&sai->pdev->dev); return -EINVAL; } -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: core: Clarify usage of ignore_machine
by Daniel Baluta 01 Oct '19

01 Oct '19

For a sound card ignore_machine means that existing FEs links should be ignored and existing BEs links should be overridden with some information from the matching component driver. Current code make some confusions about this so fix it! Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 88978a3036c4..e32a45f6bd88 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1853,7 +1853,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) for_each_component(component) { - /* does this component override FEs ? */ + /* does this component override BEs ? */ if (!component->driver->ignore_machine) continue; @@ -1874,7 +1874,7 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) continue; } - dev_info(card->dev, "info: override FE DAI link %s\n", + dev_info(card->dev, "info: override BE DAI link %s\n", card->dai_link[i].name); /* override platform component */ -- 2.17.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: adau1761: Add PGA Slew time control
by Alexandru Ardelean 01 Oct '19

01 Oct '19

From: Lars-Peter Clausen <lars(a)metafoo.de> The PGA Slew Time control allows to configure the rate with which the PGA gain control ramps up/down to the target setting. The PGA slew control is done via the ALC Control 0 register. There are 2 bits on that reg, that control PGA slew time, while the other bits control parts of the ALC (automatic level control) block. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- sound/soc/codecs/adau1761.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index bef3e9e74c26..428bf95ea648 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -29,6 +29,7 @@ #define ADAU1761_REC_MIXER_RIGHT1 0x400d #define ADAU1761_LEFT_DIFF_INPUT_VOL 0x400e #define ADAU1761_RIGHT_DIFF_INPUT_VOL 0x400f +#define ADAU1761_ALC_CTRL0 0x4011 #define ADAU1761_PLAY_LR_MIXER_LEFT 0x4020 #define ADAU1761_PLAY_MIXER_LEFT0 0x401c #define ADAU1761_PLAY_MIXER_LEFT1 0x401d @@ -72,6 +73,7 @@ static const struct reg_default adau1761_reg_defaults[] = { { ADAU1761_REC_MIXER_RIGHT0, 0x00 }, { ADAU1761_REC_MIXER_RIGHT1, 0x00 }, { ADAU1761_LEFT_DIFF_INPUT_VOL, 0x00 }, + { ADAU1761_ALC_CTRL0, 0x00 }, { ADAU1761_RIGHT_DIFF_INPUT_VOL, 0x00 }, { ADAU1761_PLAY_LR_MIXER_LEFT, 0x00 }, { ADAU1761_PLAY_MIXER_LEFT0, 0x00 }, @@ -148,6 +150,21 @@ static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_capture_bias_enum, ADAU17X1_REC_POWER_MGMT, 1, 0x3, adau1761_bias_select_text, adau1761_bias_select_values); +static const unsigned int adau1761_pga_slew_time_values[] = { + 3, 0, 1, 2, +}; + +static const char * const adau1761_pga_slew_time_text[] = { + "Off", + "24 ms", + "48 ms", + "96 ms", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(adau1761_pga_slew_time_enum, + ADAU1761_ALC_CTRL0, 6, 0x3, adau1761_pga_slew_time_text, + adau1761_pga_slew_time_values); + static const struct snd_kcontrol_new adau1761_jack_detect_controls[] = { SOC_SINGLE("Speaker Auto-mute Switch", ADAU1761_DIGMIC_JACKDETECT, 4, 1, 0), @@ -162,6 +179,8 @@ static const struct snd_kcontrol_new adau1761_differential_mode_controls[] = { SOC_DOUBLE_R_TLV("PGA Boost Capture Volume", ADAU1761_REC_MIXER_LEFT1, ADAU1761_REC_MIXER_RIGHT1, 3, 2, 0, adau1761_pga_boost_tlv), + + SOC_ENUM("PGA Capture Slew Time", adau1761_pga_slew_time_enum), }; static const struct snd_kcontrol_new adau1761_single_mode_controls[] = { @@ -633,6 +652,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg) case ADAU1761_DEJITTER: case ADAU1761_CLK_ENABLE0: case ADAU1761_CLK_ENABLE1: + case ADAU1761_ALC_CTRL0: return true; default: break; -- 2.20.1

2 7

[alsa-devel] [PATCH v2] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 01 Oct '19

01 Oct '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - Rebased on top 'ASoC: soc-component: fix a couple missing error assignments' Changes since rfc: - Fixed missing error assignments (copy pasted the error) - Document the trigger order in the commit message Regards, Peter sound/soc/soc-pcm.c | 64 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 61 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a1b99ac57d9e..66910500e3b6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1074,6 +1074,42 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

3 2

[alsa-devel] [PATCH] drm/bridge: dw-hdmi: Restore audio when setting a mode
by Cheng-Yi Chiang 01 Oct '19

01 Oct '19

From: Daniel Kurtz <djkurtz(a)chromium.org> When setting a new display mode, dw_hdmi_setup() calls dw_hdmi_enable_video_path(), which disables all hdmi clocks, including the audio clock. We should only (re-)enable the audio clock if audio was already enabled when setting the new mode. Without this patch, on RK3288, there will be HDMI audio on some monitors if i2s was played to headphone when the monitor was plugged. ACER H277HU and ASUS PB278 are two of the monitors showing this issue. Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index aa7efd4da1c8..c60e951122c9 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -1982,6 +1982,15 @@ static void hdmi_disable_overflow_interrupts(struct dw_hdmi *hdmi) HDMI_IH_MUTE_FC_STAT2); } +static void dw_hdmi_audio_restore(struct dw_hdmi *hdmi) +{ + mutex_lock(&hdmi->audio_mutex); + + hdmi_enable_audio_clk(hdmi, hdmi->audio_enable); + + mutex_unlock(&hdmi->audio_mutex); +} + static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) { int ret; @@ -2045,7 +2054,7 @@ static int dw_hdmi_setup(struct dw_hdmi *hdmi, struct drm_display_mode *mode) /* HDMI Initialization Step E - Configure audio */ hdmi_clk_regenerator_update_pixel_clock(hdmi); - hdmi_enable_audio_clk(hdmi, true); + dw_hdmi_audio_restore(hdmi); } /* not for DVI mode */ -- 2.23.0.444.g18eeb5a265-goog

1 0

[alsa-devel] [PATCH 1/5] ASoC: amd: Registering device endpoints using MFD framework
by Ravulapati Vishnu vardhan rao 30 Sep '19

30 Sep '19

Removed platform based endpoint registering. Now Registering DMA and multiple I2S BT playback endpoint devices automatically by using MFD framework. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Reviewed-by: Vijendar Mukunda <vijendar.mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x.h | 54 ++++++++------- sound/soc/amd/raven/pci-acp3x.c | 148 +++++++++++++++++++++++++++++----------- 2 files changed, 139 insertions(+), 63 deletions(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index 4f2cadd..3a27fc8 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -7,33 +7,41 @@ #include "chip_offset_byte.h" -#define ACP3x_PHY_BASE_ADDRESS 0x1240000 -#define ACP3x_I2S_MODE 0 -#define ACP3x_REG_START 0x1240000 -#define ACP3x_REG_END 0x1250200 -#define I2S_MODE 0x04 -#define BT_TX_THRESHOLD 26 -#define BT_RX_THRESHOLD 25 -#define ACP3x_POWER_ON 0x00 -#define ACP3x_POWER_ON_IN_PROGRESS 0x01 -#define ACP3x_POWER_OFF 0x02 -#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 +#define ACP3x_DEVS 3 +#define ACP3x_PHY_BASE_ADDRESS 0x1240000 +#define ACP3x_I2S_MODE 0 +#define ACP3x_REG_START 0x1240000 +#define ACP3x_REG_END 0x1250200 +#define ACP3x_I2STDM_REG_START 0x1242400 +#define ACP3x_I2STDM_REG_END 0x1242410 +#define ACP3x_BT_TDM_REG_START 0x1242800 +#define ACP3x_BT_TDM_REG_END 0x1242810 +#define I2S_MODE 0x04 +#define I2S_RX_THRESHOLD 27 +#define I2S_TX_THRESHOLD 28 +#define BT_TX_THRESHOLD 26 +#define BT_RX_THRESHOLD 25 +#define ACP_ERR_INTR_MASK 29 +#define ACP3x_POWER_ON 0x00 +#define ACP3x_POWER_ON_IN_PROGRESS 0x01 +#define ACP3x_POWER_OFF 0x02 +#define ACP3x_POWER_OFF_IN_PROGRESS 0x03 #define ACP3x_SOFT_RESET__SoftResetAudDone_MASK 0x00010001 #define ACP_SRAM_PTE_OFFSET 0x02050000 -#define PAGE_SIZE_4K_ENABLE 0x2 +#define PAGE_SIZE_4K_ENABLE 0x2 #define MEM_WINDOW_START 0x4000000 -#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 -#define CAPTURE_FIFO_ADDR_OFFSET 0x500 - -#define PLAYBACK_MIN_NUM_PERIODS 2 -#define PLAYBACK_MAX_NUM_PERIODS 8 -#define PLAYBACK_MAX_PERIOD_SIZE 16384 -#define PLAYBACK_MIN_PERIOD_SIZE 4096 -#define CAPTURE_MIN_NUM_PERIODS 2 -#define CAPTURE_MAX_NUM_PERIODS 8 -#define CAPTURE_MAX_PERIOD_SIZE 16384 -#define CAPTURE_MIN_PERIOD_SIZE 4096 +#define PLAYBACK_FIFO_ADDR_OFFSET 0x400 +#define CAPTURE_FIFO_ADDR_OFFSET 0x500 + +#define PLAYBACK_MIN_NUM_PERIODS 2 +#define PLAYBACK_MAX_NUM_PERIODS 8 +#define PLAYBACK_MAX_PERIOD_SIZE 16384 +#define PLAYBACK_MIN_PERIOD_SIZE 4096 +#define CAPTURE_MIN_NUM_PERIODS 2 +#define CAPTURE_MAX_NUM_PERIODS 8 +#define CAPTURE_MAX_PERIOD_SIZE 16384 +#define CAPTURE_MIN_PERIOD_SIZE 4096 #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index facec24..c0b8fad 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -9,23 +9,49 @@ #include <linux/io.h> #include <linux/platform_device.h> #include <linux/interrupt.h> +#include <linux/mfd/core.h> +#include <linux/pm_runtime.h> +#include <sound/pcm.h> #include "acp3x.h" +struct i2s_platform_data { + unsigned int cap; + int channel; + u32 snd_rates; +}; struct acp3x_dev_data { + struct device *parent; + struct mfd_cell *cell; + struct resource *res; void __iomem *acp3x_base; bool acp3x_audio_mode; - struct resource *res; struct platform_device *pdev; }; + +static struct device *get_mfd_cell_dev(const char *device_name, int r) +{ + char auto_dev_name[25]; + struct device *dev; + + snprintf(auto_dev_name, sizeof(auto_dev_name), + "%s.%d.auto", device_name, r); + dev = bus_find_device_by_name(&platform_bus_type, + NULL, auto_dev_name); + dev_info(dev, "device %s added\n", auto_dev_name); + return dev; +} + static int snd_acp3x_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { int ret; - u32 addr, val; + uint64_t addr; + int val, i, r; struct acp3x_dev_data *adata; - struct platform_device_info pdevinfo; + struct device *dev; + struct i2s_platform_data *i2s_pdata; unsigned int irqflags; if (pci_enable_device(pci)) { @@ -63,47 +89,82 @@ static int snd_acp3x_probe(struct pci_dev *pci, } pci_set_master(pci); pci_set_drvdata(pci, adata); - + adata->parent = &pci->dev; val = rv_readl(adata->acp3x_base + mmACP_I2S_PIN_CONFIG); switch (val) { case I2S_MODE: adata->res = devm_kzalloc(&pci->dev, - sizeof(struct resource) * 2, - GFP_KERNEL); - if (!adata->res) { + sizeof(struct resource) * 4, + GFP_KERNEL); + adata->cell = devm_kzalloc(&pci->dev, + sizeof(struct mfd_cell) * 3, + GFP_KERNEL); + if (!adata->cell) { ret = -ENOMEM; goto unmap_mmio; } - adata->res[0].name = "acp3x_i2s_iomem"; - adata->res[0].flags = IORESOURCE_MEM; - adata->res[0].start = addr; - adata->res[0].end = addr + (ACP3x_REG_END - ACP3x_REG_START); - - adata->res[1].name = "acp3x_i2s_irq"; - adata->res[1].flags = IORESOURCE_IRQ; - adata->res[1].start = pci->irq; - adata->res[1].end = pci->irq; - - adata->acp3x_audio_mode = ACP3x_I2S_MODE; - - memset(&pdevinfo, 0, sizeof(pdevinfo)); - pdevinfo.name = "acp3x_rv_i2s"; - pdevinfo.id = 0; - pdevinfo.parent = &pci->dev; - pdevinfo.num_res = 2; - pdevinfo.res = adata->res; - pdevinfo.data = &irqflags; - pdevinfo.size_data = sizeof(irqflags); - - adata->pdev = platform_device_register_full(&pdevinfo); - if (IS_ERR(adata->pdev)) { - dev_err(&pci->dev, "cannot register %s device\n", - pdevinfo.name); - ret = PTR_ERR(adata->pdev); - goto unmap_mmio; + i2s_pdata = devm_kzalloc(&pci->dev, + sizeof(struct i2s_platform_data) * 3, + GFP_KERNEL); + if (i2s_pdata == NULL) { + kfree(adata->res); + kfree(adata->cell); + return -ENOMEM; } + adata->res[0].name = "acp3x_i2s_iomem"; + adata->res[0].flags = IORESOURCE_MEM; + adata->res[0].start = addr; + adata->res[0].end = addr + + (ACP3x_REG_END - ACP3x_REG_START); + i2s_pdata[0].cap = 0; + i2s_pdata[0].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[1].name = "acp3x_i2s_sp_play_cap"; + adata->res[1].flags = IORESOURCE_MEM; + adata->res[1].start = addr + ACP3x_I2STDM_REG_START; + adata->res[1].end = addr + ACP3x_I2STDM_REG_END; + i2s_pdata[1].cap = 0; + i2s_pdata[1].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[2].name = "acp3x_i2s_bt_play_cap"; + adata->res[2].flags = IORESOURCE_MEM; + adata->res[2].start = addr + ACP3x_BT_TDM_REG_START; + adata->res[2].end = addr + ACP3x_BT_TDM_REG_END; + i2s_pdata[2].cap = 0; + i2s_pdata[2].snd_rates = SNDRV_PCM_RATE_8000_96000; + + adata->res[3].name = "acp3x_i2s_irq"; + adata->res[3].flags = IORESOURCE_IRQ; + adata->res[3].start = pci->irq; + adata->res[3].end = adata->res[3].start; + + adata->acp3x_audio_mode = ACP3x_I2S_MODE; + + adata->cell[0].name = "acp3x_rv_i2s_dma"; + adata->cell[0].num_resources = 4; + adata->cell[0].resources = &adata->res[0]; + adata->cell[0].platform_data = &irqflags; + adata->cell[0].pdata_size = sizeof(irqflags); + + adata->cell[1].name = "acp3x_i2s_playcap"; + adata->cell[1].num_resources = 1; + adata->cell[1].resources = &adata->res[1]; + adata->cell[1].platform_data = &i2s_pdata[0]; + adata->cell[1].pdata_size = + sizeof(struct i2s_platform_data); + + adata->cell[2].name = "acp3x_i2s_playcap"; + adata->cell[2].num_resources = 1; + adata->cell[2].resources = &adata->res[2]; + adata->cell[2].platform_data = &i2s_pdata[1]; + adata->cell[2].pdata_size = + sizeof(struct i2s_platform_data); + r = mfd_add_hotplug_devices(adata->parent, adata->cell, 3); + for (i = 0; i < 3 ; i++) + dev = get_mfd_cell_dev(adata->cell[i].name, i); break; + default: dev_err(&pci->dev, "Invalid ACP audio mode : %d\n", val); ret = -ENODEV; @@ -112,23 +173,29 @@ static int snd_acp3x_probe(struct pci_dev *pci, return 0; unmap_mmio: - pci_disable_msi(pci); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); + /*ignore device status and return driver probe error*/ + return -ENODEV; release_regions: pci_release_regions(pci); + /*ignore device status and return driver probe error*/ + return -ENODEV; disable_pci: pci_disable_device(pci); - - return ret; + /*ignore device status and return driver probe error*/ + return -ENODEV; } static void snd_acp3x_remove(struct pci_dev *pci) { struct acp3x_dev_data *adata = pci_get_drvdata(pci); - - platform_device_unregister(adata->pdev); + mfd_remove_devices(adata->parent); + kfree(adata->res); + kfree(adata->cell); iounmap(adata->acp3x_base); - pci_disable_msi(pci); pci_release_regions(pci); pci_disable_device(pci); @@ -151,6 +218,7 @@ static struct pci_driver acp3x_driver = { module_pci_driver(acp3x_driver); +MODULE_AUTHOR("Vishnuvardhanrao.Ravulapati(a)amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu(a)amd.com"); MODULE_DESCRIPTION("AMD ACP3x PCI driver"); MODULE_LICENSE("GPL v2"); -- 2.7.4

4 9

Re: [alsa-devel] [PATCH v2] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Pierre-Louis Bossart 28 Sep '19

28 Sep '19

On 9/26/19 9:55 PM, Navid Emamdoost wrote: > On Wed, Sep 25, 2019 at 12:05:28PM -0500, Pierre-Louis Bossart wrote: >> On 9/25/19 11:19 AM, Navid Emamdoost wrote: >>> In snd_skl_parse_uuids if allocation for module->instance_id fails, the >>> allocated memory for module shoulde be released. I changes the >>> allocation for module to use devm_kzalloc to be resource_managed >>> allocation and avoid the release in error path. >> >> if you use devm_, don't you need to fix the error path as well then, I see a >> kfree(uuid) in skl_freeup_uuid_list(). >> >> I am not very familiar with this code but the error seems to be that the >> list_add_tail() is called after the module->instance_id is allocated, so >> there is a risk that the module allocated earlier is not freed (since it's >> not yet added to the list). Freeing the module as done in patch 1 works, >> using devm_ without fixing the error path does not seem correct to me. >> > Thanks for the feedback, then it's your call if you can accept patch 1 as > fix. Cezary, it's really your call. > Navid. >>> >>> Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> >>> --- >>> Changes in v2: >>> - Changed the allocation for module from kzalloc to devm_kzalloc >>> --- >>> sound/soc/intel/skylake/skl-sst-utils.c | 2 +- >>> 1 file changed, 1 insertion(+), 1 deletion(-) >>> >>> diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c >>> index d43cbf4a71ef..ac37f04b0eea 100644 >>> --- a/sound/soc/intel/skylake/skl-sst-utils.c >>> +++ b/sound/soc/intel/skylake/skl-sst-utils.c >>> @@ -284,7 +284,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, >>> */ >>> for (i = 0; i < num_entry; i++, mod_entry++) { >>> - module = kzalloc(sizeof(*module), GFP_KERNEL); >>> + module = devm_kzalloc(ctx->dev, sizeof(*module), GFP_KERNEL); >>> if (!module) { >>> ret = -ENOMEM; >>> goto free_uuid_list; >>> >>

4 5

[alsa-devel] [PATCH v5] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 27 Sep '19

27 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v5: - Removed '#clock-cells' property from required properties - Added deprecated property to 'clock-output-names' and removed corresponding comment. Best regards, Maciej Falkowski --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 136 ++++++++++++++++++ 2 files changed, 136 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..b9502b82fa9d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,136 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for i2s0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for i2s1 and i2s2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clk. i2s0 has internal mux to select the source + of root clk and i2s1 and i2s2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: + deprecated: true + oneOf: + - items: # for i2s0 + - const: i2s_cdclk0 + - items: # for i2s1 + - const: i2s_cdclk1 + - items: # for i2s2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

4 3

[alsa-devel] [PATCH] ASoC: Skylake SST driver - blacklist the PCI device IDs for the auto probe
by Jaroslav Kysela 27 Sep '19

27 Sep '19

There are basically three drivers for the PCI devices for the recent Intel hardware with the build-in DSPs. The legacy HDA driver has dmic_detect module option for the auto detection of the platforms with the digital microphone. Because the SOF driver is preferred, just skip PCI probe in the Skylake SST driver when the PCI device ID clashes by default. The user can override the auto behaviour with the pci_binding module parameter. Boot log from Lenovo Carbon X1 (7th gen) with the default settings: snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, aborting probe snd_soc_skl 0000:00:1f.3: SOF driver is preferred on this platform, aborting probe sof-audio-pci 0000:00:1f.3: warning: No matching ASoC machine driver found sof-audio-pci 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040380 .... Perhaps, it may be more wise to create one shared module and all three drivers should call the driver detection routine(s) from one place. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/skylake/skl.c | 37 +++++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 141dbbf975ac..cace55ca8d55 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -38,6 +38,39 @@ static int skl_pci_binding; module_param_named(pci_binding, skl_pci_binding, int, 0444); MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); +/* + * + */ +static int skl_sof_support(struct pci_dev *pci) +{ + /* the SOF driver has same PCI IDs */ + if (pci->vendor == 0x8086) { + switch (pci->device) { +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_LP) + case 0x02c8: /* CML-LP */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMETLAKE_H) + case 0x06c8: /* CML-H */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_GEMINILAKE) + case 0x3198: /* GLK */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_APOLLOLAKE) + case 0x5a98: /* BXT-P */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_CANNONLAKE) + case 0x9dc8: /* CNL */ +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COFFEELAKE) + case 0xa348: /* CFL */ +#endif + case 0x0000: /* a dummy value when no SOF driver enabled */ + return 1; + } + } + return 0; +} + /* * initialize the PCI registers */ @@ -1002,6 +1035,10 @@ static int skl_probe(struct pci_dev *pci, dev_err(&pci->dev, "Unknown PCI class/subclass/prog-if information (0x%06x) found, aborting probe\n", pci->class); return -ENODEV; } + if (skl_sof_support(pci)) { + dev_info(&pci->dev, "SOF driver is preferred on this platform, aborting probe\n"); + return -ENODEV; + } dev_info(&pci->dev, "DSP detected with PCI class/subclass/prog-if info 0x%06x\n", pci->class); break; case SND_SKL_PCI_BIND_LEGACY: -- 2.20.1

4 10

[alsa-devel] [PATCH] ASoC: amd: acp3x: clean up an indentation issue
by Colin King 27 Sep '19

27 Sep '19

From: Colin Ian King <colin.king(a)canonical.com> There is a return statement that is indented too deeply, remove the extraneous tab. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index bc4dfafdfcd1..ea57088d50ce 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -631,7 +631,7 @@ static int acp3x_audio_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!res) { dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); - return -ENODEV; + return -ENODEV; } adata = devm_kzalloc(&pdev->dev, sizeof(*adata), GFP_KERNEL); -- 2.20.1

3 2

[alsa-devel] sof-skl_hda: remove volume setting in enable/disable sequences
by GitHub pull_request - opened 27 Sep '19

27 Sep '19

alsa-project/alsa-ucm-conf pull request #2 was opened from juimonen: If the volume setting is defined in enable sequence you will always get that exact volume when switching outputs in some user space software like Pulseaudio. Pulseaudio will itself keep track of the volume previously set. Signed-off-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/2 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/2.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] Playback volumes
by GitHub pull_request - opened 27 Sep '19

27 Sep '19

alsa-project/alsa-ucm-conf pull request #1 was opened from juimonen: Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/1 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/1.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[alsa-devel] [PATCH] ASoC: bdw-rt5677: channel constraint support
by Brent Lu 27 Sep '19

27 Sep '19

BDW boards using this machine driver supports only stereo capture and playback. Implement a constraint to enforce it. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/bdw-rt5677.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/intel/boards/bdw-rt5677.c b/sound/soc/intel/boards/bdw-rt5677.c index 4a4d335..a312b55 100644 --- a/sound/soc/intel/boards/bdw-rt5677.c +++ b/sound/soc/intel/boards/bdw-rt5677.c @@ -22,6 +22,8 @@ #include "../../codecs/rt5677.h" +#define DUAL_CHANNEL 2 + struct bdw_rt5677_priv { struct gpio_desc *gpio_hp_en; struct snd_soc_component *component; @@ -245,6 +247,36 @@ static int bdw_rt5677_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int bdw_fe_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + /* + * On this platform for PCM device we support, + * stereo + */ + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + + return 0; +} + +static const struct snd_soc_ops bdw_rt5677_fe_ops = { + .startup = bdw_fe_startup, +}; + /* broadwell digital audio interface glue - connects codec <--> CPU */ SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); @@ -273,6 +305,7 @@ static struct snd_soc_dai_link bdw_rt5677_dais[] = { }, .dpcm_capture = 1, .dpcm_playback = 1, + .ops = &bdw_rt5677_fe_ops, SND_SOC_DAILINK_REG(fe, dummy, platform), }, -- 2.7.4

4 9

[alsa-devel] [PATCH 0/4] ASoC: mt8183: fix audio playback slowly after playback
by Jiaxin Yu 27 Sep '19

27 Sep '19

This series patches add reset controller for MT8183, and audio will use it in machine driver during bootup, they depend on the following patch: 1. this series add support reset-controller in infra [v5,2/2] clk: reset: Modify reset-controller driver *** BLURB HERE *** Jiaxin Yu (2): WIP: dt-bindings: medaitek: mt8183: add property "resets" && "reset-names" WIP: ASoC: mt8183: fix audio playback slowly after playback during bootup yong.liang (2): WIP: dt-bindings: mediatek: mt8183: Add #reset-cells WIP: watchdog: mtk_wdt: mt8183: Add reset controller .../bindings/sound/mt8183-afe-pcm.txt | 6 + .../devicetree/bindings/watchdog/mtk-wdt.txt | 9 +- drivers/watchdog/Kconfig | 1 + drivers/watchdog/mtk_wdt.c | 110 +++++++++++++++++- .../reset-controller/mt8183-resets.h | 13 +++ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 15 +++ 6 files changed, 150 insertions(+), 4 deletions(-) -- 2.18.0

1 4

[alsa-devel] [PATCH] ALSA: hda - Force polling mode on CML for fixing codec communication
by Sathyanarayana Nujella 26 Sep '19

26 Sep '19

We observed the same issue as reported by commit a8d7bde23e7130686b7662 ("ALSA: hda - Force polling mode on CFL for fixing codec communication") Apply the same fix to CML. Signed-off-by: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Signed-off-by: Joyce Toh <joyce.toh(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91e71be42fa4..e6997a5bf098 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -370,6 +370,7 @@ enum { #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) #define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) +#define IS_CML(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x02c8) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -1754,7 +1755,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, azx_bus(chip)->dma_type = SNDRV_DMA_TYPE_DEV_UC; /* Workaround for a communication error on CFL (bko#199007) and CNL */ - if (IS_CFL(pci) || IS_CNL(pci)) + if (IS_CFL(pci) || IS_CNL(pci) || IS_CML(pci)) azx_bus(chip)->polling_mode = 1; if (chip->driver_type == AZX_DRIVER_NVIDIA) { -- 2.7.4

4 3

[alsa-devel] [PATCH] ASoC: topology: Fix a signedness bug in soc_tplg_dapm_widget_create()
by Dan Carpenter 26 Sep '19

26 Sep '19

The "template.id" variable is an enum and in this context GCC will treat it as an unsigned int so it can never be less than zero. Fixes: 8a9782346dcc ("ASoC: topology: Add topology core") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index aa9a1fca46fa..0fd032914a31 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1582,7 +1582,7 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, /* map user to kernel widget ID */ template.id = get_widget_id(le32_to_cpu(w->id)); - if (template.id < 0) + if ((int)template.id < 0) return template.id; /* strings are allocated here, but used and freed by the widget */ -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: amd: Missing Initialization of IRQFLAGS
by Ravulapati Vishnu vardhan rao 26 Sep '19

26 Sep '19

Fix for missing initialization of IRQFLAGS in ACP-PCI driver and Missing Macro of ACP3x_DEVS. Follow up to IDb33df346 Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index 627798a..3887ea0 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -219,7 +219,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, sizeof(struct resource) * 4, GFP_KERNEL); adata->cell = devm_kzalloc(&pci->dev, - sizeof(struct mfd_cell) * 3, + sizeof(struct mfd_cell) * ACP3x_DEVS, GFP_KERNEL); if (!adata->cell) { ret = -ENOMEM; @@ -260,6 +260,7 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->res[3].flags = IORESOURCE_IRQ; adata->res[3].start = pci->irq; adata->res[3].end = adata->res[3].start; + irqflags = 0; adata->acp3x_audio_mode = ACP3x_I2S_MODE; @@ -282,8 +283,9 @@ static int snd_acp3x_probe(struct pci_dev *pci, adata->cell[2].platform_data = &i2s_pdata[1]; adata->cell[2].pdata_size = sizeof(struct i2s_platform_data); - r = mfd_add_hotplug_devices(adata->parent, adata->cell, 3); - for (i = 0; i < 3 ; i++) + r = mfd_add_hotplug_devices(adata->parent, adata->cell, + ACP3x_DEVS); + for (i = 0; i < ACP3x_DEVS ; i++) dev = get_mfd_cell_dev(adata->cell[i].name, i); break; -- 2.7.4

2 1

[alsa-devel] [PATCH] dt-bindings: sound: Convert Samsung Exynos Odroid XU3/XU4 audio complex to dt-schema
by Marek Szyprowski 26 Sep '19

26 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec to newer dt-schema format. 'clocks' property is unneeded in the bindings and is left undefined in 'properties'. 'samsung,audio-widgets' and 'samsung,audio-routing' are optional from driver perspective and they are set as unrequired. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,odroid.txt | 54 ----------- .../bindings/sound/samsung,odroid.yaml | 91 +++++++++++++++++++ 2 files changed, 91 insertions(+), 54 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroid.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt deleted file mode 100644 index e9da2200e173..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ /dev/null @@ -1,54 +0,0 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec - -Required properties: - - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), - "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), - "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - - model - the user-visible name of this sound complex - - clocks - should contain entries matching clock names in the clock-names - property - - samsung,audio-widgets - this property specifies off-codec audio elements - like headphones or speakers, for details see widgets.txt - - samsung,audio-routing - a list of the connections between audio - components; each entry is a pair of strings, the first being the - connection's sink, the second being the connection's source; - valid names for sources and sinks are the MAX98090's pins (as - documented in its binding), and the jacks on the board - - For Odroid X2: - "Headphone Jack", "Mic Jack", "DMIC" - - For Odroid U3, XU3: - "Headphone Jack", "Speakers" - - For Odroid XU4: - no entries - -Required sub-nodes: - - - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S - controller - - 'codec' subnode with a 'sound-dai' property containing list of phandles - to the CODEC nodes, first entry must be corresponding to the MAX98090 - CODEC and the second entry must be the phandle of the HDMI IP block node - -Example: - -sound { - compatible = "hardkernel,odroid-xu3-audio"; - model = "Odroid-XU3"; - samsung,audio-routing = - "Headphone Jack", "HPL", - "Headphone Jack", "HPR", - "IN1", "Mic Jack", - "Mic Jack", "MICBIAS"; - - cpu { - sound-dai = <&i2s0 0>; - }; - codec { - sound-dai = <&hdmi>, <&max98090>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.yaml b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml new file mode 100644 index 000000000000..db6d3ea3180e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.yaml @@ -0,0 +1,91 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,odroid.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + oneOf: + - const: hardkernel,odroid-xu3-audio + + - const: hardkernel,odroid-xu4-audio + deprecated: true + + - const: samsung,odroid-xu3-audio + deprecated: true + + - const: samsung,odroid-xu4-audio + deprecated: true + + model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + + cpu: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: phandles to the I2S controllers + + codec: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles to the CODEC nodes, + first entry must be corresponding to the MAX98090 CODEC and + the second entry must be the phandle of the HDMI IP block node. + + samsung,audio-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + This property specifies off-codec audio elements + like headphones or speakers, for details see widgets.txt + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio + components; each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source; + valid names for sources and sinks are the MAX98090's pins (as + documented in its binding), and the jacks on the board. + For Odroid X2: "Headphone Jack", "Mic Jack", "DMIC" + For Odroid U3, XU3: "Headphone Jack", "Speakers" + For Odroid XU4: no entries + +required: + - compatible + - model + - cpu + - codec + +examples: + - | + sound { + compatible = "hardkernel,odroid-xu3-audio"; + model = "Odroid-XU3"; + samsung,audio-routing = + "Headphone Jack", "HPL", + "Headphone Jack", "HPR", + "IN1", "Mic Jack", + "Mic Jack", "MICBIAS"; + + cpu { + sound-dai = <&i2s0 0>; + }; + + codec { + sound-dai = <&hdmi>, <&max98090>; + }; + }; + -- 2.17.1

2 1

[alsa-devel] [PATCH] dt-bindings: sound: Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to dt-schema
by Marek Szyprowski 26 Sep '19

26 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung Exynos5433 TM2(E) audio complex with WM5110 codec to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- .../bindings/sound/samsung,tm2-audio.txt | 42 ---------- .../bindings/sound/samsung,tm2-audio.yaml | 83 +++++++++++++++++++ 2 files changed, 83 insertions(+), 42 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt deleted file mode 100644 index f5ccc12ddc00..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.txt +++ /dev/null @@ -1,42 +0,0 @@ -Samsung Exynos5433 TM2(E) audio complex with WM5110 codec - -Required properties: - - - compatible : "samsung,tm2-audio" - - model : the user-visible name of this sound complex - - audio-codec : the first entry should be phandle of the wm5110 audio - codec node, as described in ../mfd/arizona.txt; - the second entry should be phandle of the HDMI - transmitter node - - i2s-controller : the list of phandle and argument tuples pointing to - I2S controllers, the first entry should be I2S0 and - the second one I2S1 - - audio-amplifier : the phandle of the MAX98504 amplifier - - samsung,audio-routing : a list of the connections between audio components; - each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source; valid names for sources and sinks are the - WM5110's and MAX98504's pins and the jacks on the - board: HP, SPK, Main Mic, Sub Mic, Third Mic, - Headset Mic - - mic-bias-gpios : GPIO pin that enables the Main Mic bias regulator - - -Example: - -sound { - compatible = "samsung,tm2-audio"; - audio-codec = <&wm5110>, <&hdmi>; - i2s-controller = <&i2s0 0>, <&i2s1 0>; - audio-amplifier = <&max98504>; - mic-bias-gpios = <&gpr3 2 0>; - model = "wm5110"; - samsung,audio-routing = - "HP", "HPOUT1L", - "HP", "HPOUT1R", - "SPK", "SPKOUT", - "SPKOUT", "HPOUT2L", - "SPKOUT", "HPOUT2R", - "Main Mic", "MICBIAS2", - "IN1R", "Main Mic"; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml new file mode 100644 index 000000000000..377f8cbe17b8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,tm2-audio.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung,tm2-audio.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung Exynos SoC Exynos5433 TM2(E) audio complex with WM5110 codec + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + const: samsung,tm2-audio + + model: + maxItems: 1 + + audio-codec: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: | + phandle of the wm5110 audio codec node, + as described in ../mfd/arizona.txt; + - description: phandle of the HDMI transmitter node. + + i2s-controller: + allOf: + - $ref: /schemas/types.yaml#/definitions/phandle-array + - items: + - description: phandle of the I2S0. + - description: phandle of the I2S1. + + audio-amplifier: + $ref: /schemas/types.yaml#/definitions/phandle + description: phandle of the MAX98504 amplifier. + + samsung,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + List of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's + source. Valid names for sources and sinks are + the WM5110's and MAX98504's pins and the jacks on the + board: HP, SPK, Main Mic, Sub Mic, Third Mic, Headset Mic. + + mic-bias-gpios: + description: GPIO pin that enables the Main Mic bias regulator. + maxItems: 1 + +required: + - compatible + - model + - audio-codec + - i2s-controller + - audio-amplifier + - samsung,audio-routing + - mic-bias-gpios + +additionalProperties: false + +examples: + - | + sound { + compatible = "samsung,tm2-audio"; + model = "wm5110"; + audio-codec = <&wm5110>, <&hdmi>; + i2s-controller = <&i2s0 0>, <&i2s1 0>; + audio-amplifier = <&max98504>; + mic-bias-gpios = <&gpr3>; + samsung,audio-routing = + "HP", "HPOUT1L", + "HP", "HPOUT1R", + "SPK", "SPKOUT", + "SPKOUT", "HPOUT2L", + "SPKOUT", "HPOUT2R", + "Main Mic", "MICBIAS2", + "IN1R", "Main Mic"; + }; + -- 2.17.1

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Re: [alsa-devel] [PATCH V2 1/2] string: Add stracpy and stracpy_pad mechanisms
by Rasmus Villemoes 26 Sep '19

26 Sep '19

On 26/09/2019 10.25, Stephen Kitt wrote: > Le 26/09/2019 09:29, Rasmus Villemoes a écrit : >> On 26/09/2019 02.01, Stephen Kitt wrote: >>> Le 25/09/2019 23:50, Andrew Morton a écrit : >>>> On Tue, 23 Jul 2019 06:51:36 -0700 Joe Perches <joe(a)perches.com> wrote: >>>> >> >> Please don't. At least not for the cases where the source is a string >> literal - that just gives worse code generation (because gcc doesn't >> know anything about strscpy or strlcpy), and while a run-time (silent) >> truncation is better than a run-time buffer overflow, wouldn't it be >> even better with a build time error? > > Yes, that was the plan once Joe's patch gets merged (if it does), and my > patch was only an example of using stracpy, as a step on the road. I was > intending to follow up with a patch converting stracpy to something like > https://www.openwall.com/lists/kernel-hardening/2019/07/06/14 > > __FORTIFY_INLINE ssize_t strscpy(char *dest, const char *src, size_t count) > { > size_t dest_size = __builtin_object_size(dest, 0); > size_t src_size = __builtin_object_size(src, 0); > if (__builtin_constant_p(count) && > __builtin_constant_p(src_size) && > __builtin_constant_p(dest_size) && Eh? Isn't the output of __builtin_object_size() by definition a compile-time constant - whatever the compiler happens to know about the object size (or a sentinel 0 or -1 depending on the type argument)? > > #define stracpy(dest, src) \ > ({ \ > size_t count = ARRAY_SIZE(dest); \ > size_t dest_size = __builtin_object_size(dest, 0); \ > size_t src_size = __builtin_object_size(src, 0); \ > BUILD_BUG_ON(!(__same_type(dest, char[]) || \ > __same_type(dest, unsigned char[]) || \ > __same_type(dest, signed char[]))); \ > \ > (__builtin_constant_p(count) && \ > __builtin_constant_p(src_size) && \ > __builtin_constant_p(dest_size) && \ > src_size <= count && \ > src_size <= dest_size && \ > src[src_size - 1] == '\0') ? \ > (((size_t) strcpy(dest, src)) & 0) + src_size - 1 \ > : \ > strscpy(dest, src, count); \ > }) > > and both of these get optimised to movs when copying a constant string > which fits in the target. But does it catch the case of overflowing a char[] member in a struct passed by reference at build time? I'm surprised that __builtin_object_size(dest, 0) seems to be (size_t)-1, when dest is s->name (with struct s { char name[4]; };). So I'm not very confident that any of the fancy fortify logic actually works here - we _really_ should have some Kbuild infrastructure for saying "this .c file should not compile" so we can test that the fortifications actually work in the simple and common cases. > I was going at this from the angle of improving the existing APIs and > their resulting code. I'm not against stracpy() as a wrapper for strscpy(), but we should make sure that whenever we can fail at build time (i.e., both source and dst lengths known), we do - and in that case also let the compiler optimize the copy (not only to do the immediate movs, but that also gives it wider opportunity to remove it completely as dead stores if the surrounding code ends up dead - with a call to some strscpy(), gcc cannot eliminate that). If stracpy() can be made sufficiently magic that it fails at build time for the string literal cases, fine, let's not add yet another API. Otherwise, I think the static_strcpy() is a much more readable and reliable API for those cases. If I'm reading your stracpy() macro correctly, you're explicitly requesting a run-time truncation (the src_size <= dest_size check causing as to fall back to strscpy) rather than failing at build time. Rasmus

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Re: [alsa-devel] [PATCH V2 1/2] string: Add stracpy and stracpy_pad mechanisms
by Rasmus Villemoes 26 Sep '19

26 Sep '19

On 26/09/2019 02.01, Stephen Kitt wrote: > Le 25/09/2019 23:50, Andrew Morton a écrit : >> On Tue, 23 Jul 2019 06:51:36 -0700 Joe Perches <joe(a)perches.com> wrote: >> >>> Several uses of strlcpy and strscpy have had defects because the >>> last argument of each function is misused or typoed. >>> >>> Add macro mechanisms to avoid this defect. >>> >>> stracpy (copy a string to a string array) must have a string >>> array as the first argument (dest) and uses sizeof(dest) as the >>> count of bytes to copy. >>> >>> These mechanisms verify that the dest argument is an array of >>> char or other compatible types like u8 or s8 or equivalent. >>> >>> A BUILD_BUG is emitted when the type of dest is not compatible. >>> >> >> I'm still reluctant to merge this because we don't have code in -next >> which *uses* it. You did have a patch for that against v1, I believe? >> Please dust it off and send it along? > > Joe had a Coccinelle script to mass-convert strlcpy and strscpy. > Here's a different patch which converts some of ALSA's strcpy calls to > stracpy: Please don't. At least not for the cases where the source is a string literal - that just gives worse code generation (because gcc doesn't know anything about strscpy or strlcpy), and while a run-time (silent) truncation is better than a run-time buffer overflow, wouldn't it be even better with a build time error? Something like /** static_strcpy - run-time version of static initialization * * @d: destination array, must be array of char of known size * @s: source, must be a string literal * * This is a simple wrapper for strcpy(d, s), which checks at build-time that the strcpy() won't overflow. In most cases (for short strings), gcc won't even emit a call to strcpy but rather just do a few immediate stores into the destination, e.g. 0: c7 07 66 6f 6f 00 movl $0x6f6f66,(%rdi) * for strcpy(d->name, "foo"). */ #define static_strcpy(d, s) ({ \ static_assert(__same_type(d, char[]), "destination must be char array"); \ static_assert(__same_type(s, const char[], "source must be a string literal"); \ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ strcpy(d, s); \ }) The "" s "" trick guarantees that s is a string literal - it probably doesn't give a very nice error message, but that's why I added the redundant type comparison to a const char[] which should hopefully give a better clue. Rasmus PS: Yes, we already have a fortified strcpy(), but for some reason it doesn't catch the common case where we're populating a char[] member of some struct. So this diff --git a/lib/vsprintf.c b/lib/vsprintf.c index e78017a3e1bd..3b0c5ae9f49e 100644 --- a/lib/vsprintf.c +++ b/lib/vsprintf.c @@ -3420,3 +3420,14 @@ int sscanf(const char *buf, const char *fmt, ...) return i; } EXPORT_SYMBOL(sscanf); + +struct s { char name[4]; }; +char buf[4]; +void f3(void) +{ + strcpy(buf, "long"); +} +void f4(struct s *s) +{ + strcpy(s->name, "long"); +} with CONFIG_FORTIFY_SOURCE=y complains about f3(), but f4() is just fine... PPS: A quick test of static_strcpy(): // ss.c #include <errno.h> #include <string.h> #include <assert.h> #define __same_type(x, y) __builtin_types_compatible_p(typeof(x), typeof(y)) #define static_strcpy(d, s) ({ \ static_assert(__same_type(d, char[]), "destination must be char array"); \ static_assert(__same_type(s, const char[]), "source must be a string literal"); \ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ strcpy(d, s); \ }) struct s { char name[4]; }; void f0(struct s *s) { static_strcpy(s->name, "foo"); } #if 1 void f1(struct s *s) { static_strcpy(s->name, strerror(EIO)); } void f2(struct s *s) { static_strcpy(s->name, "long"); } void f3(char *d) { static_strcpy(d, "foo"); } void f4(char *d) { static_strcpy(d, strerror(EIO)); } #endif // gcc -Wall -O2 -o ss.o -c ss.c In file included from ss.c:3:0: ss.c: In function ‘f1’: ss.c:9:3: error: static assertion failed: "source must be a string literal" static_assert(__same_type(s, const char[]), "source must be a string literal"); \ ^ ss.c:18:24: note: in expansion of macro ‘static_strcpy’ void f1(struct s *s) { static_strcpy(s->name, strerror(EIO)); } ^~~~~~~~~~~~~ ss.c:18:47: error: expected ‘)’ before ‘strerror’ void f1(struct s *s) { static_strcpy(s->name, strerror(EIO)); } ^ ss.c:10:40: note: in definition of macro ‘static_strcpy’ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ ^ In file included from ss.c:3:0: ss.c: In function ‘f2’: ss.c:10:3: error: static assertion failed: "source does not fit in destination" static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ ^ ss.c:19:24: note: in expansion of macro ‘static_strcpy’ void f2(struct s *s) { static_strcpy(s->name, "long"); } ^~~~~~~~~~~~~ ss.c: In function ‘f3’: ss.c:8:3: error: static assertion failed: "destination must be char array" static_assert(__same_type(d, char[]), "destination must be char array"); \ ^ ss.c:20:20: note: in expansion of macro ‘static_strcpy’ void f3(char *d) { static_strcpy(d, "foo"); } ^~~~~~~~~~~~~ ss.c: In function ‘f4’: ss.c:8:3: error: static assertion failed: "destination must be char array" static_assert(__same_type(d, char[]), "destination must be char array"); \ ^ ss.c:21:20: note: in expansion of macro ‘static_strcpy’ void f4(char *d) { static_strcpy(d, strerror(EIO)); } ^~~~~~~~~~~~~ ss.c:9:3: error: static assertion failed: "source must be a string literal" static_assert(__same_type(s, const char[]), "source must be a string literal"); \ ^ ss.c:21:20: note: in expansion of macro ‘static_strcpy’ void f4(char *d) { static_strcpy(d, strerror(EIO)); } ^~~~~~~~~~~~~ ss.c:21:37: error: expected ‘)’ before ‘strerror’ void f4(char *d) { static_strcpy(d, strerror(EIO)); } ^ ss.c:10:40: note: in definition of macro ‘static_strcpy’ static_assert(sizeof(d) >= sizeof("" s ""), "source does not fit in destination"); \ ^

1 0

Re: [alsa-devel] [PATCH V5 4/4] ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
by S.j. Wang 26 Sep '19

26 Sep '19

Hi > Just a small concern... > > On Thu, Sep 26, 2019 at 09:29:51AM +0800, Shengjiu Wang wrote: > > static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) > > { > > + > > + release_pair = false; > > + ret = snd_soc_set_runtime_hwparams(substream, > > + &snd_imx_hardware); > > This set_runtime_hwparams() always returns 0 for now, but if one day it > changes and it fails here, kfree() will be still ignored although the startup() > gets error-out. > > We could avoid this if we continue to ignore the return value like the > current code. Or we may check ret at kfree() also? I like to ignore the return value. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH V5 0/4] update supported sample format
by Shengjiu Wang 26 Sep '19

26 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. changes in v5 - free asrc pair when error happens include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 66 ++++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 200 insertions(+), 87 deletions(-) -- 2.21.0

2 5

Re: [alsa-devel] [PATCH v2] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Pierre-Louis Bossart 25 Sep '19

25 Sep '19

On 9/25/19 11:19 AM, Navid Emamdoost wrote: > In snd_skl_parse_uuids if allocation for module->instance_id fails, the > allocated memory for module shoulde be released. I changes the > allocation for module to use devm_kzalloc to be resource_managed > allocation and avoid the release in error path. if you use devm_, don't you need to fix the error path as well then, I see a kfree(uuid) in skl_freeup_uuid_list(). I am not very familiar with this code but the error seems to be that the list_add_tail() is called after the module->instance_id is allocated, so there is a risk that the module allocated earlier is not freed (since it's not yet added to the list). Freeing the module as done in patch 1 works, using devm_ without fixing the error path does not seem correct to me. > > Signed-off-by: Navid Emamdoost <navid.emamdoost(a)gmail.com> > --- > Changes in v2: > - Changed the allocation for module from kzalloc to devm_kzalloc > --- > sound/soc/intel/skylake/skl-sst-utils.c | 2 +- > 1 file changed, 1 insertion(+), 1 deletion(-) > > diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c > index d43cbf4a71ef..ac37f04b0eea 100644 > --- a/sound/soc/intel/skylake/skl-sst-utils.c > +++ b/sound/soc/intel/skylake/skl-sst-utils.c > @@ -284,7 +284,7 @@ int snd_skl_parse_uuids(struct sst_dsp *ctx, const struct firmware *fw, > */ > > for (i = 0; i < num_entry; i++, mod_entry++) { > - module = kzalloc(sizeof(*module), GFP_KERNEL); > + module = devm_kzalloc(ctx->dev, sizeof(*module), GFP_KERNEL); > if (!module) { > ret = -ENOMEM; > goto free_uuid_list; >

1 0

[alsa-devel] [PATCH] [MT6660] Mediatek Smart Amplifier Driver
by richtek.jeff.chang＠gmail.com 25 Sep '19

25 Sep '19

From: Richtek Jeff Chang <richtek.jeff.chang(a)gmail.com> Signed-off-by: Richtek Jeff Chang <richtek.jeff.chang(a)gmail.com> --- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mt6660.c | 802 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mt6660.h | 68 ++++ 4 files changed, 885 insertions(+) create mode 100644 sound/soc/codecs/mt6660.c create mode 100644 sound/soc/codecs/mt6660.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9f89a53..3407436 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -120,6 +120,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ML26124 if I2C select SND_SOC_MT6351 if MTK_PMIC_WRAP select SND_SOC_MT6358 if MTK_PMIC_WRAP + select SND_SOC_MT6660 if I2C select SND_SOC_NAU8540 if I2C select SND_SOC_NAU8810 if I2C select SND_SOC_NAU8822 if I2C @@ -1444,6 +1445,18 @@ config SND_SOC_MT6358 Enable support for the platform which uses MT6358 as external codec device. +config SND_SOC_MT6660 + tristate "MediaTek MT6660 Speaker amplifier" + depends on I2C + select CRC32 + select CRYPTO_SHA256 + select CRYPTO_RSA + help + MediaTek MT6660 is a smart power amplifier which contain + speaker protection, multi-band DRC, equalizer functions. + Select N if you don't have MT6660 on board. + Select M to build this as module. + config SND_SOC_NAU8540 tristate "Nuvoton Technology Corporation NAU85L40 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b4bb8c..e6a8054 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -117,6 +117,7 @@ snd-soc-msm8916-analog-objs := msm8916-wcd-analog.o snd-soc-msm8916-digital-objs := msm8916-wcd-digital.o snd-soc-mt6351-objs := mt6351.o snd-soc-mt6358-objs := mt6358.o +snd-soc-mt6660-objs := mt6660.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o snd-soc-nau8822-objs := nau8822.o @@ -398,6 +399,7 @@ obj-$(CONFIG_SND_SOC_MSM8916_WCD_ANALOG) +=snd-soc-msm8916-analog.o obj-$(CONFIG_SND_SOC_MSM8916_WCD_DIGITAL) +=snd-soc-msm8916-digital.o obj-$(CONFIG_SND_SOC_MT6351) += snd-soc-mt6351.o obj-$(CONFIG_SND_SOC_MT6358) += snd-soc-mt6358.o +obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c new file mode 100644 index 0000000..b82cf03 --- /dev/null +++ b/sound/soc/codecs/mt6660.c @@ -0,0 +1,802 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/version.h> +#include <linux/err.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/pcm_params.h> +#include <linux/debugfs.h> + +#include "mt6660.h" + +union mt6660_multi_byte_data { + u32 data_u32; + u16 data_u16; + u8 data_u8; + u8 data[4]; +}; + +struct codec_reg_val { + u32 addr; + u32 mask; + u32 data; +}; + +struct reg_size_table { + u32 addr; + u8 size; +}; + +static const struct reg_size_table mt6660_reg_size_table[] = { + { MT6660_REG_HPF1_COEF, 4 }, + { MT6660_REG_HPF2_COEF, 4 }, + { MT6660_REG_TDM_CFG3, 2 }, + { MT6660_REG_RESV17, 2 }, + { MT6660_REG_RESV23, 2 }, + { MT6660_REG_SIGMAX, 2 }, + { MT6660_REG_DEVID, 2}, + { MT6660_REG_TDM_CFG3, 2}, + { MT6660_REG_HCLIP_CTRL, 2}, + { MT6660_REG_DA_GAIN, 2}, +}; + +static int mt6660_get_reg_size(uint32_t addr) +{ + int i = 0; + + for (i = 0; i < ARRAY_SIZE(mt6660_reg_size_table); i++) { + if (mt6660_reg_size_table[i].addr == addr) + return mt6660_reg_size_table[i].size; + } + return 1; +} + +static int32_t mt6660_i2c_update_bits(struct mt6660_chip *chip, + uint32_t addr, uint32_t mask, uint32_t data) +{ + int ret = 0; + uint32_t value; + int size = mt6660_get_reg_size(addr); + union mt6660_multi_byte_data mdata; + + memcpy(&mdata, &data, sizeof(uint32_t)); + + mutex_lock(&chip->io_lock); + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, addr, size, (u8 *)&value); + if (ret < 0) { + mutex_unlock(&chip->io_lock); + return ret; + } + switch (size) { + case 1: + value &= ~mask; + value |= (mdata.data_u8 & mask); + break; + case 2: + value = be16_to_cpu(value); + value &= ~mask; + value |= (mdata.data_u16 & mask); + value = be16_to_cpu(value); + break; + case 4: + value = be32_to_cpu(value); + value &= ~mask; + value |= (mdata.data_u32 & mask); + value = be32_to_cpu(value); + break; + default: + dev_err(chip->dev, "%s Invalid bytes\n", __func__); + break; + } + + ret = i2c_smbus_write_i2c_block_data( + chip->i2c, addr, size, (u8 *)&value); + if (ret < 0) { + mutex_unlock(&chip->io_lock); + return ret; + } + mutex_unlock(&chip->io_lock); + + return 0; +} + +static int mt6660_i2c_read(struct mt6660_chip *chip, unsigned int reg) +{ + int size = mt6660_get_reg_size(reg); + int i = 0, ret = 0; + u8 data[4] = {0}; + u32 reg_data = 0; + + ret = i2c_smbus_read_i2c_block_data(chip->i2c, reg, size, data); + if (ret < 0) + return ret; + for (i = 0; i < size; i++) { + reg_data <<= 8; + reg_data |= data[i]; + } + return reg_data; +} + +static unsigned int mt6660_component_io_read( + struct snd_soc_component *component, unsigned int reg) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + return mt6660_i2c_read(chip, reg); +} + +static int mt6660_component_io_write(struct snd_soc_component *component, + unsigned int reg, unsigned int data) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int size = mt6660_get_reg_size(reg); + u8 reg_data[4] = {0}; + int i = 0; + + for (i = 0; i < size; i++) + reg_data[size - i - 1] = (data >> (8 * i)) & 0xff; + + return i2c_smbus_write_i2c_block_data(chip->i2c, reg, size, reg_data); +} + +static const int mt6660_dump_table[] = { + MT6660_REG_DEVID, + MT6660_REG_SYSTEM_CTRL, + MT6660_REG_IRQ_STATUS1, + MT6660_REG_SERIAL_CFG1, + MT6660_REG_DATAO_SEL, + MT6660_REG_TDM_CFG3, + MT6660_REG_HPF_CTRL, + MT6660_REG_HPF1_COEF, + MT6660_REG_HPF2_COEF, + MT6660_REG_PATH_BYPASS, + MT6660_REG_WDT_CTRL, + MT6660_REG_HCLIP_CTRL, + MT6660_REG_VOL_CTRL, + MT6660_REG_SPS_CTRL, + MT6660_REG_SIGMAX, + MT6660_REG_CALI_T0, + MT6660_REG_BST_CTRL, + MT6660_REG_PROTECTION_CFG, + MT6660_REG_DA_GAIN, + MT6660_REG_AUDIO_IN2_SEL, + MT6660_REG_SIG_GAIN, + MT6660_REG_PLL_CFG1, + MT6660_REG_DRE_CTRL, + MT6660_REG_DRE_THDMODE, + MT6660_REG_DRE_CORASE, + MT6660_REG_PWM_CTRL, + MT6660_REG_DC_PROTECT_CTRL, + MT6660_REG_ADC_USB_MODE, + MT6660_REG_INTERNAL_CFG, + MT6660_REG_RESV0, + MT6660_REG_RESV1, + MT6660_REG_RESV2, + MT6660_REG_RESV3, + MT6660_REG_RESV7, + MT6660_REG_RESV10, + MT6660_REG_RESV11, + MT6660_REG_RESV16, + MT6660_REG_RESV17, + MT6660_REG_RESV19, + MT6660_REG_RESV21, + MT6660_REG_RESV23, + MT6660_REG_RESV31, + MT6660_REG_RESV40, +}; + +static const struct codec_reg_val e4_reg_inits[] = { + { MT6660_REG_WDT_CTRL, 0x80, 0x00 }, + { MT6660_REG_SPS_CTRL, 0x01, 0x01 }, + { MT6660_REG_AUDIO_IN2_SEL, 0x1c, 0x04 }, + { MT6660_REG_RESV11, 0x0c, 0x00 }, + { MT6660_REG_RESV31, 0x03, 0x03 }, + { MT6660_REG_RESV40, 0x01, 0x00 }, + { MT6660_REG_RESV0, 0x44, 0x04 }, + { MT6660_REG_RESV19, 0xff, 0x82 }, + { MT6660_REG_RESV17, 0x7777, 0x7273 }, + { MT6660_REG_RESV16, 0x07, 0x03 }, + { MT6660_REG_DRE_CORASE, 0xe0, 0x20 }, + { MT6660_REG_ADDA_CLOCK, 0xff, 0x70 }, + { MT6660_REG_RESV21, 0xff, 0x20 }, + { MT6660_REG_DRE_THDMODE, 0xff, 0x40 }, + { MT6660_REG_RESV23, 0xffff, 0x17f8 }, + { MT6660_REG_PWM_CTRL, 0xff, 0x15 }, + { MT6660_REG_ADC_USB_MODE, 0xff, 0x00 }, + { MT6660_REG_PROTECTION_CFG, 0xff, 0x1d }, + { MT6660_REG_HPF1_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_HPF2_COEF, 0xffffffff, 0x7fdb7ffe }, + { MT6660_REG_SIG_GAIN, 0xff, 0x58 }, + { MT6660_REG_RESV6, 0xff, 0xce }, + { MT6660_REG_SIGMAX, 0xffff, 0x7fff }, + { MT6660_REG_DA_GAIN, 0xffff, 0x0116 }, + { MT6660_REG_TDM_CFG3, 0x1800, 0x0800 }, + { MT6660_REG_DRE_CTRL, 0x1f, 0x07 }, +}; + +static int mt6660_i2c_init_setting(struct mt6660_chip *chip) +{ + int i, len, ret; + const struct codec_reg_val *init_table; + + init_table = e4_reg_inits; + len = ARRAY_SIZE(e4_reg_inits); + + for (i = 0; i < len; i++) { + ret = mt6660_i2c_update_bits(chip, init_table[i].addr, + init_table[i].mask, init_table[i].data); + if (ret < 0) + return ret; + } + return 0; +} + +static int mt6660_chip_power_on( + struct snd_soc_component *component, int on_off) +{ + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = 0; + unsigned int val; + + dev_dbg(component->dev, "%s: on_off = %d\n", __func__, on_off); + mutex_lock(&chip->var_lock); + if (on_off) { + if (chip->pwr_cnt == 0) { + ret = mt6660_i2c_update_bits(chip, + MT6660_REG_SYSTEM_CTRL, 0x01, 0x00); + val = mt6660_i2c_read(chip, MT6660_REG_IRQ_STATUS1); + dev_info(chip->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + } + chip->pwr_cnt++; + } else { + chip->pwr_cnt--; + if (chip->pwr_cnt == 0) { + ret = mt6660_i2c_update_bits(chip, + MT6660_REG_SYSTEM_CTRL, 0x01, 0xff); + } + if (chip->pwr_cnt < 0) { + dev_warn(chip->dev, "not paired on/off\n"); + chip->pwr_cnt = 0; + } + } + mutex_unlock(&chip->var_lock); + if (ret < 0) + pr_err("%s ret = %d\n", __func__, ret); + return ret; +} + +static int mt6660_component_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + int ret; + unsigned int val; + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + if (dapm->bias_level == level) { + dev_warn(component->dev, "%s: repeat level change\n", __func__); + return 0; + } + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level != SND_SOC_BIAS_OFF) + break; + dev_dbg(component->dev, "exit low power mode\n"); + ret = mt6660_chip_power_on(component, 1); + if (ret < 0) { + dev_err(component->dev, "power on fail\n"); + return ret; + } + break; + case SND_SOC_BIAS_OFF: + dev_dbg(component->dev, "enter low power mode\n"); + val = mt6660_i2c_read(chip, MT6660_REG_IRQ_STATUS1); + dev_info(component->dev, + "%s reg0x05 = 0x%x\n", __func__, val); + ret = mt6660_chip_power_on(component, 0); + if (ret < 0) { + dev_err(component->dev, "power off fail\n"); + return ret; + } + break; + default: + return -EINVAL; + } + dapm->bias_level = level; + dev_dbg(component->dev, "c bias_level = %d\n", level); + return 0; +} + +static int mt6660_component_probe(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + + ret = mt6660_component_set_bias_level(component, SND_SOC_BIAS_STANDBY); + if (ret < 0) { + dev_err(component->dev, "config bias standby fail\n"); + return ret; + } + + ret = mt6660_component_set_bias_level(component, SND_SOC_BIAS_OFF); + if (ret < 0) { + dev_err(component->dev, "config bias off fail\n"); + return ret; + } + chip->component = component; + return 0; +} + +static void mt6660_component_remove(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "%s++\n", __func__); + chip->component = NULL; + dev_dbg(component->dev, "%s--\n", __func__); +} + +static int mt6660_codec_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + usleep_range(1000, 1100); + break; + } + return 0; +} + +static int mt6660_codec_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + dev_dbg(component->dev, + "%s: before classd turn on\n", __func__); + /* config to adaptive mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x03); + if (ret < 0) { + dev_err(component->dev, "config mode adaptive fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMU: + /* voltage sensing enable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x04); + if (ret < 0) { + dev_err(component->dev, + "enable voltage sensing fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_PRE_PMD: + /* voltage sensing disable */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV7, 0x04, 0x00); + if (ret < 0) { + dev_err(component->dev, + "disable voltage sensing fail\n"); + return ret; + } + /* pop-noise improvement 1 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x10); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 1 fail\n"); + return ret; + } + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(component->dev, + "%s: after classd turn off\n", __func__); + /* pop-noise improvement 2 */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_RESV10, 0x10, 0x00); + if (ret < 0) { + dev_err(component->dev, + "pop-noise improvement 2 fail\n"); + return ret; + } + /* config to off mode */ + ret = snd_soc_component_update_bits(component, + MT6660_REG_BST_CTRL, 0x03, 0x00); + if (ret < 0) { + dev_err(component->dev, "config mode off fail\n"); + return ret; + } + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget mt6660_component_dapm_widgets[] = { + SND_SOC_DAPM_DAC_E("DAC", NULL, MT6660_REG_PLL_CFG1, + 0, 1, mt6660_codec_dac_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("VI ADC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA("PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV_E("ClassD", MT6660_REG_SYSTEM_CTRL, 2, 0, + NULL, 0, mt6660_codec_classd_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SPK("SPK", NULL), +}; + +static const struct snd_soc_dapm_route mt6660_component_dapm_routes[] = { + { "DAC", NULL, "aif_playback"}, + { "PGA", NULL, "DAC"}, + { "ClassD", NULL, "PGA"}, + { "SPK", NULL, "ClassD"}, + { "VI ADC", NULL, "ClassD"}, + { "aif_capture", NULL, "VI ADC"}, +}; + +static int mt6660_component_put_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + int ret, put_ret = 0; + + ret = mt6660_chip_power_on(component, 1); + if (ret < 0) + dev_err(component->dev, "%s: pwr on fail\n", __func__); + put_ret = snd_soc_put_volsw(kcontrol, ucontrol); + if (ret < 0) + return ret; + ret = mt6660_chip_power_on(component, 0); + if (ret < 0) + dev_err(component->dev, "%s: pwr off fail\n", __func__); + return put_ret; +} + +static int mt6660_component_get_volsw(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct mt6660_chip *chip = (struct mt6660_chip *) + snd_soc_component_get_drvdata(component); + int ret = -EINVAL; + + if (!strcmp(kcontrol->id.name, "Chip_Rev")) { + ucontrol->value.integer.value[0] = chip->chip_rev & 0x0f; + ret = 0; + } + return ret; +} + +static const DECLARE_TLV_DB_SCALE(vol_ctl_tlv, -1155, 5, 0); + +static const struct snd_kcontrol_new mt6660_component_snd_controls[] = { + SOC_SINGLE_EXT_TLV("Volume_Ctrl", MT6660_REG_VOL_CTRL, 0, 255, + 1, snd_soc_get_volsw, mt6660_component_put_volsw, + vol_ctl_tlv), + SOC_SINGLE_EXT("WDT_Enable", MT6660_REG_WDT_CTRL, 7, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Hard_Clip_Enable", MT6660_REG_HCLIP_CTRL, 8, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Clip_Enable", MT6660_REG_SPS_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("BoostMode", MT6660_REG_BST_CTRL, 0, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DRE_Enable", MT6660_REG_DRE_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("DC_Protect_Enable", + MT6660_REG_DC_PROTECT_CTRL, 3, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("I2SLRS", MT6660_REG_DATAO_SEL, 6, 3, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("I2SDOLS", MT6660_REG_DATAO_SEL, 3, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("I2SDORS", MT6660_REG_DATAO_SEL, 0, 7, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + /* for debug purpose */ + SOC_SINGLE_EXT("HPF_AUD_IN_EN", MT6660_REG_HPF_CTRL, 0, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("AUD_LOOP_BACK", MT6660_REG_PATH_BYPASS, 4, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("Mute_Enable", MT6660_REG_SYSTEM_CTRL, 1, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("CS_Comp_Disable", MT6660_REG_PATH_BYPASS, 2, 1, 0, + snd_soc_get_volsw, mt6660_component_put_volsw), + SOC_SINGLE_EXT("T0_SEL", MT6660_REG_CALI_T0, 0, 7, 0, + snd_soc_get_volsw, NULL), + SOC_SINGLE_EXT("Chip_Rev", SND_SOC_NOPM, 0, 16, 0, + mt6660_component_get_volsw, NULL), +}; + + +static const struct snd_soc_component_driver mt6660_component_driver = { + .probe = mt6660_component_probe, + .remove = mt6660_component_remove, + + .read = mt6660_component_io_read, + .write = mt6660_component_io_write, + + .controls = mt6660_component_snd_controls, + .num_controls = ARRAY_SIZE(mt6660_component_snd_controls), + .dapm_widgets = mt6660_component_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(mt6660_component_dapm_widgets), + .dapm_routes = mt6660_component_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(mt6660_component_dapm_routes), + + .set_bias_level = mt6660_component_set_bias_level, + .idle_bias_on = false, /* idle_bias_off = true */ +}; + +static int mt6660_component_aif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(dai->component); + int ret = 0; + + dev_dbg(dai->dev, "%s\n", __func__); + if (dapm->bias_level == SND_SOC_BIAS_OFF) + ret = mt6660_component_set_bias_level(dai->component, + SND_SOC_BIAS_STANDBY); + return ret; +} + +static int mt6660_component_aif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *dai) +{ + int word_len = params_physical_width(hw_params); + int aud_bit = params_width(hw_params); + u16 reg_data = 0; + int ret = 0; + + dev_dbg(dai->dev, "%s: ++\n", __func__); + dev_dbg(dai->dev, "format: 0x%08x\n", params_format(hw_params)); + dev_dbg(dai->dev, "rate: 0x%08x\n", params_rate(hw_params)); + dev_dbg(dai->dev, "word_len: %d, aud_bit: %d\n", word_len, aud_bit); + if (word_len > 32 || word_len < 16) { + dev_err(dai->dev, "not supported word length\n"); + return -ENOTSUPP; + } + switch (aud_bit) { + case 16: + reg_data = 3; + break; + case 18: + reg_data = 2; + break; + case 20: + reg_data = 1; + break; + case 24: + case 32: + reg_data = 0; + break; + default: + return -ENOTSUPP; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_SERIAL_CFG1, 0xc0, (reg_data << 6)); + if (ret < 0) { + dev_err(dai->dev, "config aud bit fail\n"); + return ret; + } + ret = snd_soc_component_update_bits(dai->component, + MT6660_REG_TDM_CFG3, 0x3f0, word_len << 4); + if (ret < 0) { + dev_err(dai->dev, "config word len fail\n"); + return ret; + } + dev_dbg(dai->dev, "%s: --\n", __func__); + return 0; +} + +static const struct snd_soc_dai_ops mt6660_component_aif_ops = { + .startup = mt6660_component_aif_startup, + .hw_params = mt6660_component_aif_hw_params, +}; + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_soc_dai_driver mt6660_codec_dai = { + .name = "mt6660-aif", + .playback = { + .stream_name = "aif_playback", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + .capture = { + .stream_name = "aif_capture", + .channels_min = 1, + .channels_max = 2, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, + /* dai properties */ + .symmetric_rates = 1, + .symmetric_channels = 1, + .symmetric_samplebits = 1, + /* dai operations */ + .ops = &mt6660_component_aif_ops, +}; + +static inline int mt6660_chip_id_check(struct i2c_client *i2c) +{ + u8 id[2] = {0}; + int ret = 0; + + i2c_smbus_write_byte_data(i2c, 0x03, 0x00); + ret = i2c_smbus_read_i2c_block_data(i2c, MT6660_REG_DEVID, 2, id); + if (ret < 0) + return ret; + ret = (id[0] << 8) + id[1]; + ret &= 0x0ff0; + if (ret != 0x00e0 && ret != 0x01e0) + return -ENODEV; + i2c_smbus_write_byte_data(i2c, 0x03, 0x01); + return 0; +} + +static inline int _mt6660_chip_sw_reset(struct mt6660_chip *chip) +{ + i2c_smbus_write_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL, 0x80); + msleep(30); + return 0; +} + +static inline int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) +{ + u8 reg_data = 0; + int ret = 0; + + ret = i2c_smbus_read_byte_data(chip->i2c, MT6660_REG_SYSTEM_CTRL); + if (ret < 0) + return ret; + reg_data = (u8)ret; + if (on_off) + reg_data &= (~0x01); + else + reg_data |= 0x01; + return i2c_smbus_write_byte_data( + chip->i2c, MT6660_REG_SYSTEM_CTRL, reg_data); +} + +static inline int _mt6660_read_chip_revision(struct mt6660_chip *chip) +{ + u8 reg_data[2] = {0}; + int ret = 0; + + ret = i2c_smbus_read_i2c_block_data( + chip->i2c, MT6660_REG_DEVID, 2, reg_data); + if (ret < 0) { + dev_err(chip->dev, "get chip revision fail\n"); + return ret; + } + chip->chip_rev = reg_data[1]; + return 0; +} + +int mt6660_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) +{ + struct mt6660_chip *chip = NULL; + int ret = 0; + + ret = mt6660_chip_id_check(client); + if (ret < 0) { + dev_err(&client->dev, "chip id check fail\n"); + return ret; + } + chip = devm_kzalloc(&client->dev, sizeof(*chip), GFP_KERNEL); + if (!chip) + return -ENOMEM; + chip->i2c = client; + chip->dev = &client->dev; + mutex_init(&chip->var_lock); + mutex_init(&chip->io_lock); + i2c_set_clientdata(client, chip); + + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 1 fail\n"); + goto probe_fail; + } + /* chip reset first */ + ret = _mt6660_chip_sw_reset(chip); + if (ret < 0) { + dev_err(chip->dev, "chip reset fail\n"); + goto probe_fail; + } + /* chip power on */ + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(chip->dev, "chip power on 2 fail\n"); + goto probe_fail; + } + ret = _mt6660_read_chip_revision(chip); + if (ret < 0) { + dev_err(chip->dev, "read chip revision fail\n"); + goto probe_fail; + } + ret = mt6660_i2c_init_setting(chip); + if (ret < 0) { + dev_err(chip->dev, "chip i2c init setting fail\n"); + goto probe_fail; + } + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(chip->dev, "chip power off fail\n"); + goto probe_fail; + } + return snd_soc_register_component(chip->dev, + &mt6660_component_driver, &mt6660_codec_dai, 1); +probe_fail: + mutex_destroy(&chip->var_lock); + return ret; +} + +int mt6660_i2c_remove(struct i2c_client *client) +{ + struct mt6660_chip *chip = i2c_get_clientdata(client); + + dev_dbg(chip->dev, "%s++\n", __func__); + snd_soc_unregister_component(chip->dev); + mutex_destroy(&chip->var_lock); + dev_dbg(chip->dev, "%s--\n", __func__); + return 0; +} + +static const struct of_device_id __maybe_unused mt6660_of_id[] = { + { .compatible = "mediatek,mt6660",}, + {}, +}; +MODULE_DEVICE_TABLE(of, mt6660_of_id); + +static const struct i2c_device_id mt6660_i2c_id[] = { + {"mt6660", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, mt6660_i2c_id); + +static struct i2c_driver mt6660_i2c_driver = { + .driver = { + .name = "mt6660", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(mt6660_of_id), + }, + .probe = mt6660_i2c_probe, + .remove = mt6660_i2c_remove, + .id_table = mt6660_i2c_id, +}; +module_i2c_driver(mt6660_i2c_driver); + +MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com"); +MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); +MODULE_LICENSE("GPL"); +MODULE_VERSION("1.0.6_G"); diff --git a/sound/soc/codecs/mt6660.h b/sound/soc/codecs/mt6660.h new file mode 100644 index 0000000..4d5cc16 --- /dev/null +++ b/sound/soc/codecs/mt6660.h @@ -0,0 +1,68 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Copyright (c) 2019 MediaTek Inc. + */ + +#ifndef __SND_SOC_MT6660_H +#define __SND_SOC_MT6660_H + +#include <linux/mutex.h> + +struct mt6660_chip { + struct i2c_client *i2c; + struct device *dev; + struct snd_soc_component *component; + struct platform_device *param_dev; + struct mutex var_lock; + struct mutex io_lock; + u16 chip_rev; + int pwr_cnt; +}; + +#define MT6660_REG_DEVID (0x00) +#define MT6660_REG_SYSTEM_CTRL (0x03) +#define MT6660_REG_IRQ_STATUS1 (0x05) +#define MT6660_REG_ADDA_CLOCK (0x07) +#define MT6660_REG_SERIAL_CFG1 (0x10) +#define MT6660_REG_DATAO_SEL (0x12) +#define MT6660_REG_TDM_CFG3 (0x15) +#define MT6660_REG_HPF_CTRL (0x18) +#define MT6660_REG_HPF1_COEF (0x1A) +#define MT6660_REG_HPF2_COEF (0x1B) +#define MT6660_REG_PATH_BYPASS (0x1E) +#define MT6660_REG_WDT_CTRL (0x20) +#define MT6660_REG_HCLIP_CTRL (0x24) +#define MT6660_REG_VOL_CTRL (0x29) +#define MT6660_REG_SPS_CTRL (0x30) +#define MT6660_REG_SIGMAX (0x33) +#define MT6660_REG_CALI_T0 (0x3F) +#define MT6660_REG_BST_CTRL (0x40) +#define MT6660_REG_PROTECTION_CFG (0x46) +#define MT6660_REG_DA_GAIN (0x4c) +#define MT6660_REG_AUDIO_IN2_SEL (0x50) +#define MT6660_REG_SIG_GAIN (0x51) +#define MT6660_REG_PLL_CFG1 (0x60) +#define MT6660_REG_DRE_CTRL (0x68) +#define MT6660_REG_DRE_THDMODE (0x69) +#define MT6660_REG_DRE_CORASE (0x6B) +#define MT6660_REG_PWM_CTRL (0x70) +#define MT6660_REG_DC_PROTECT_CTRL (0x74) +#define MT6660_REG_ADC_USB_MODE (0x7c) +#define MT6660_REG_INTERNAL_CFG (0x88) +#define MT6660_REG_RESV0 (0x98) +#define MT6660_REG_RESV1 (0x99) +#define MT6660_REG_RESV2 (0x9A) +#define MT6660_REG_RESV3 (0x9B) +#define MT6660_REG_RESV6 (0xA2) +#define MT6660_REG_RESV7 (0xA3) +#define MT6660_REG_RESV10 (0xB0) +#define MT6660_REG_RESV11 (0xB1) +#define MT6660_REG_RESV16 (0xB6) +#define MT6660_REG_RESV17 (0xB7) +#define MT6660_REG_RESV19 (0xB9) +#define MT6660_REG_RESV21 (0xBB) +#define MT6660_REG_RESV23 (0xBD) +#define MT6660_REG_RESV31 (0xD3) +#define MT6660_REG_RESV40 (0xE0) + +#endif /* __SND_SOC_MT6660_H */ -- 2.7.4

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[alsa-devel] About soc_check_tplg_fes
by Daniel Baluta 25 Sep '19

25 Sep '19

Hi all, In function soc_check_tplg_fes (sound/soc/soc-core.c) there is this code: for_each_card_prelinks(card, i, dai_link) { /* ignore this FE */ if (dai_link->dynamic) { dai_link->ignore = true; continue; } dev_info(card->dev, "info: override FE DAI link %s\n", card->dai_link[i].name); So, we ignore any FE DAI link but then the info message says the contrary. Any idea? thanks, Daniel.

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[alsa-devel] [PATCH v3 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 25 Sep '19

25 Sep '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 473 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 500 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- Changes from V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 Changes from V2: Reordered acpi match table as MAX98357A is a non I2C device. Hence probe order might fail. 1.9.1

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[alsa-devel] [RFC] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 25 Sep '19

25 Sep '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Start DMA first followed by CPU DAI (currently used sequence) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Stop CPU DAI first followed by DMA Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, on a new TI platform (j721e) where we can disable the McASP FIFO (the CPU dai) since the UDMA + PDMA provides the buffering I have started to see error interrupts right after pcm_trigger:STOP and in rare cases even on PAUSE that McASP underruns. I was also able to reproduce the same issue on am335x board, but it is much harder to trigger it. With this patch the underrun after trigger:STOP is gone. If I think about the issue, I'm not sure why it was not noticed before as the behavior makes sense: we stop the DMA first then we stop the CPU DAI. If between the DMA stop and DAI stop we would need a sample in the DAI (which is still running) then for sure we will underrun in the HW (or overrun in case of capture). When I run the ALSA conformance test [1] it is easier to trigger. Not sure if anyone else have seen such underrun/overrun when stopping a stream, but the fact that I have seen it with both UDMA+PDMA and EDMA on different platforms makes me wonder if the issue can be seen on other platforms as well. [1] https://chromium.googlesource.com/chromiumos/platform/audiotest/+/master/al… Regards, Peter --- sound/soc/soc-pcm.c | 66 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 62 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..c96430e70752 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,24 +1056,60 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + for_each_rtd_codec_dai(rtd, i, codec_dai) { ret = snd_soc_dai_trigger(codec_dai, substream, cmd); if (ret < 0) return ret; } - for_each_rtdcom(rtd, rtdcom) { - component = rtdcom->component; + snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; - ret = snd_soc_component_trigger(component, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: soc-pcm: Use different sequence for start/stop trigger
by Peter Ujfalusi 25 Sep '19

25 Sep '19

On stream stop currently we stop the DMA first followed by the CPU DAI. This can cause underflow (playback) or overflow (capture) on the DAI side as the DMA is no longer feeding data while the DAI is still active. It can be observed easily if the DAI side does not have FIFO (or it is disabled) to survive the time while the DMA is stopped, but still can happen on relatively slow CPUs when relatively high sampling rate is used: the FIFO is drained between the time the DMA is stopped and the DAI is stopped. It can only fixed by using different sequence within trigger for 'stop' and 'start': case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: Trigger order: dai_link, DMA, CPU DAI then the codec case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: Trigger order: codec, CPU DAI, DMA then dai_link Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/soc-pcm.c | 66 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 62 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..41e081e95108 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1047,7 +1047,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) return 0; } -static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +static int soc_pcm_trigger_start(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; @@ -1056,8 +1056,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) struct snd_soc_dai *codec_dai; int i, ret; - for_each_rtd_codec_dai(rtd, i, codec_dai) { - ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (rtd->dai_link->ops->trigger) { + ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) return ret; } @@ -1070,10 +1070,46 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - snd_soc_dai_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_pcm_trigger_stop(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component; + struct snd_soc_rtdcom_list *rtdcom; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; + int i, ret; + + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_trigger(codec_dai, substream, cmd); + if (ret < 0) + return ret; + } + + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); + if (ret < 0) + return ret; + + for_each_rtdcom(rtd, rtdcom) { + component = rtdcom->component; + + ret = snd_soc_component_trigger(component, substream, cmd); + if (ret < 0) + return ret; + } + if (rtd->dai_link->ops->trigger) { ret = rtd->dai_link->ops->trigger(substream, cmd); if (ret < 0) @@ -1083,6 +1119,28 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = soc_pcm_trigger_start(substream, cmd); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = soc_pcm_trigger_stop(substream, cmd); + break; + default: + return -EINVAL; + } + + return ret; +} + static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, int cmd) { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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Re: [alsa-devel] [PATCH] ASoC: Intel: Skylake: prevent memory leak in snd_skl_parse_uuids
by Andy Shevchenko 25 Sep '19

25 Sep '19

On Tue, Sep 24, 2019 at 11:08:38PM -0500, Navid Emamdoost wrote: > In snd_skl_parse_uuids if allocation for module->instance_id fails, the > allocated memory for module shoulde be released. Since it's using devm_*() for the second allocation it would be cleaner to either switch it to regular kcalloc() or to switch the first one to devm_kzalloc(), whatever suits better. > module->instance_id = devm_kzalloc(ctx->dev, size, GFP_KERNEL); > if (!module->instance_id) { > + kfree(module); -- With Best Regards, Andy Shevchenko

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[alsa-devel] [PATCH 0/3] Make pcm_ioplug check lock status before locking (fixes pcm_jack lockups)
by Ben Russell 25 Sep '19

25 Sep '19

This is my first time contributing a patch to a mailing list. If I've made a mess, please let me know so I can learn how to avoid it in future. The purpose of this patchset is to fix a specific common lockup in the pcm_jack plugin (from alsa-plugins). I'm not sure if this is the right approach to take, but at the very least it should make the pcm_ioplug code a bit more resilient. The problem is this: When using the pcm_jack plugin, if a program attempts to play audio using the SND_PCM_FORMAT_FLOAT format, said program locks up. This should be enough to reproduce the bug: pcm.rawjack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } } pcm.!default { type plug slave.pcm "rawjack" } What's happening is that several snd_pcm_ioplug_* functions assume that the pcm mutex is locked already. It then proceeds to unlock the mutex, call a function, and then relock the mutex. When the mutex isn't locked already, the initial unlock results in a silently ignored pthread error, and the lock results in the program eventually deadlocking as it doesn't expect the mutex to be held at that point. Patch 2 modifies pcm_ioplug to check if the mutex is held before doing the unlock-act-lock sequence, and if the mutex is not held then it skips the unlock and lock stages. This depends on Patch 1, which adds a snd_pcm_is_locked function to give the state of the mutex. Patch 3 is completely optional. It adds assertions which make sure that all uses of snd_pcm_lock/snd_pcm_unlock are correct. On one hand this will likely result in crashes in some of the less refined parts of the code. On the other hand, when that happens, you'll know which parts need a bit more love. I know it was useful for finding this issue in the first place. These patches fix the problems I am having, but if you have a more suitable approach to fixing this problem then please let me know. Regards, Ben R Ben Russell (3): pcm_local: Add snd_pcm_is_locked pcm_ioplug: Don't unlock+lock if we're not locked pcm_local: assert() when using mutexes incorrectly src/pcm/pcm_ioplug.c | 50 +++++++++++++++++++++++----------- src/pcm/pcm_local.h | 64 ++++++++++++++++++++++++++++++++++++++------ 2 files changed, 91 insertions(+), 23 deletions(-) -- 2.23.0

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[alsa-devel] [PATCH v2 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 25 Sep '19

25 Sep '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 473 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 500 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- Changes from V1: Addressed comments on ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 1.9.1

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Re: [alsa-devel] [PATCH V4 4/4] ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
by S.j. Wang 25 Sep '19

25 Sep '19

Hi > On Tue, Sep 24, 2019 at 06:52:35PM +0800, Shengjiu Wang wrote: > > There is error "aplay: pcm_write:2023: write error: Input/output error" > > on i.MX8QM/i.MX8QXP platform for S24_3LE format. > > > > In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit > > sample, but we didn't add any constraint, that cause issues. > > > > So we need to query the caps of dma, then update the hw parameters > > according to the caps. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 4 +-- > > sound/soc/fsl/fsl_asrc.h | 3 ++ > > sound/soc/fsl/fsl_asrc_dma.c | 59 > > +++++++++++++++++++++++++++++++----- > > 3 files changed, 56 insertions(+), 10 deletions(-) > > > > @@ -270,12 +268,17 @@ static int fsl_asrc_dma_hw_free(struct > > snd_pcm_substream *substream) > > > > static int fsl_asrc_dma_startup(struct snd_pcm_substream *substream) > > { > > + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; > > struct snd_soc_pcm_runtime *rtd = substream->private_data; > > struct snd_pcm_runtime *runtime = substream->runtime; > > struct snd_soc_component *component = > snd_soc_rtdcom_lookup(rtd, > > DRV_NAME); > > + struct snd_dmaengine_dai_dma_data *dma_data; > > struct device *dev = component->dev; > > struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); > > struct fsl_asrc_pair *pair; > > + struct dma_chan *tmp_chan = NULL; > > + u8 dir = tx ? OUT : IN; > > + int ret = 0; > > > > pair = kzalloc(sizeof(struct fsl_asrc_pair), GFP_KERNEL); > > Sorry, I didn't catch it previously. We would need to release this memory > also for all error-out paths, as the code doesn't have any error-out routine, > prior to applying this change. > > > if (!pair) > > @@ -285,11 +288,51 @@ static int fsl_asrc_dma_startup(struct > > snd_pcm_substream *substream) > > > + /* Request a dummy pair, which will be released later. > > + * Request pair function needs channel num as input, for this > > + * dummy pair, we just request "1" channel temporary. > > + */ > > "temporary" => "temporarily" > > > + ret = fsl_asrc_request_pair(1, pair); > > + if (ret < 0) { > > + dev_err(dev, "failed to request asrc pair\n"); > > + return ret; > > + } > > + > > + /* Request a dummy dma channel, which will be release later. */ > > "release" => "released" Ok, will update them. Best regards Wang shengjiu

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[alsa-devel] [PATCH V4 0/4] update supported sample format
by Shengjiu Wang 24 Sep '19

24 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams changes in v4 - update according to Nicolin's comments. include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 59 ++++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 61 ++------------------ 6 files changed, 193 insertions(+), 87 deletions(-) -- 2.21.0

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[alsa-devel] Making alsamixer configurable, adding mouse support + minor fixes
by GitHub pull_request - opened 24 Sep '19

24 Sep '19

alsa-project/alsa-utils pull request #26 was opened from braph: Hello, I'm using the alsamixer, but I am missing mouse support and the ability to configure it. I implemented those things for myself now, maybe it would also be useful for others. There are some commits that would even fit in the original alsamixer code without my rather heavy modifications, these are: 6eb0bc8f0848567f302e94ea5ee3a6d7c6966e65: _alsamixer: remove redundant sprintf() calls_ bef5b723a43f310ba73bcaa52f3da74ef5e08dde: _alsamixer: use mvwhline/mvwvline for drawing lines_ ae0604c93aa4bacbb678eb8ec832ad9e17c8b56a: _alsamixer: device_name.c: recognize DEL (ASCII 127) as backspace_ And maybe 485f3b0da1ffead2f88cffd9f2203516000415b5: _alsamixer: fix window_size_changed()_ Regards, Benjamin Request URL : https://github.com/alsa-project/alsa-utils/pull/26 Patch URL : https://github.com/alsa-project/alsa-utils/pull/26.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] alsa-ucm-conf repository - initial import
by Jaroslav Kysela 24 Sep '19

24 Sep '19

Hello all, I just made an initial import to to alsa-ucm-conf repository which is in the 'import' branch: https://github.com/alsa-project/alsa-ucm-conf/tree/import I do not have the BSD-3-Clause licence acknowledgements from some minor contributors. I do not think that it's a problem. I would expect that if we do not have a disagreement, it is fine. The question is, if we need automake/autoconf tools to package those files. It seems to me that it might be clever to keep this ASIS for now. Users/packagers can just do 'cp' to the appropriate directory. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH] ASoC: samsung: i2s: Add clocks' macros descriptions
by Marek Szyprowski 24 Sep '19

24 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> To increase macro readability added descriptions to clocks macros. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- include/dt-bindings/sound/samsung-i2s.h | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/include/dt-bindings/sound/samsung-i2s.h b/include/dt-bindings/sound/samsung-i2s.h index 77545f14c379..fd752475c762 100644 --- a/include/dt-bindings/sound/samsung-i2s.h +++ b/include/dt-bindings/sound/samsung-i2s.h @@ -2,8 +2,13 @@ #ifndef _DT_BINDINGS_SAMSUNG_I2S_H #define _DT_BINDINGS_SAMSUNG_I2S_H -#define CLK_I2S_CDCLK 0 -#define CLK_I2S_RCLK_SRC 1 -#define CLK_I2S_RCLK_PSR 2 +/* the CDCLK (CODECLKO) gate clock */ +#define CLK_I2S_CDCLK 0 + +/* the RCLKSRC mux clock (corresponding to RCLKSRC bit in IISMOD register) */ +#define CLK_I2S_RCLK_SRC 1 + +/* the RCLK prescaler divider clock (corresponding to the IISPSR register) */ +#define CLK_I2S_RCLK_PSR 2 #endif /* _DT_BINDINGS_SAMSUNG_I2S_H */ -- 2.17.1

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[alsa-devel] Propagating audio properties along the audio path
by Marc Gonzalez 24 Sep '19

24 Sep '19

Hello everyone, Disclaimer: I've never worked in the sound/ layer, and it is possible that some of my questions are silly or obvious. Basically, I'm trying to implement some form of eARC(*) on an arm64 SoC. (*) enhanced Audio Return Channel (from HDMI 2.1) The setup looks like this: A = Some kind of audio source, typically a TV or game console B = The arm64 SoC, equipped with some nice speakers HDMI A ------> B If we look inside B, we actually have B1 = an eARC receiver (input = HDMI, output = I2S) B2 = an audio DSP (input = I2S, output = speakers) I2S ? B1 -----> B2 -----> speakers If I read the standard right, B is supposed to advertise which audio formats it supports, and A is supposed to pick "the best". For the sake of argument, let's say A picks "PCM, 48 kHz, 8 channels, 16b". At some point, B receives audio packets, parses the Channel Status, and determines that A is sending "PCM, 48 kHz, 8 channels, 16b". The driver then configures the I2S link, and forwards the audio stream over I2S to the DSP. QUESTION_1: How is the DSP supposed to "learn" the properties of the audio stream? (AFAIU, they're not embedded in the data, so there must be some side-channel?) I assume the driver of B1 is supposed to propagate the info to the driver of B2? (Via some call-backs? By calling a function in B2?) QUESTION_2: Does it ever make sense for B2 to ask B1 to change the audio properties? (Not sure if B1 is even allowed to renegotiate.) Regards.

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[alsa-devel] [PATCH 0/2] Add CML M/C for RT5682 and RT1011
by sathya.prakash.m.r＠intel.com 24 Sep '19

24 Sep '19

From: Sathya Prakash M R <sathya.prakash.m.r(a)intel.com> The series adds CML m/c driver with RT5682 on SSP0 and RT1011 on SSP1. The RT1011 is smart AMP which supports 2 Whoofers and 2 tweeter support. RT5682 is headset codec supporting playback and record. The m/c supports HDMI playback and DMIC record. Naveen Manohar (2): ASoC: Intel: boards: Add CML m/c using RT1011 and RT5682 ASoC: Intel: Add acpi match for rt1011 based m/c driver sound/soc/intel/boards/Kconfig | 18 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cml_rt1011_rt5682.c | 475 ++++++++++++++++++++++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 7 + 4 files changed, 502 insertions(+) create mode 100644 sound/soc/intel/boards/cml_rt1011_rt5682.c -- V1: Acked-by:pierre-louis.bossart@linux.intel.com at https://github.com/thesofproject/linux/pull/1161 1.9.1

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[alsa-devel] [PATCH v4] dt-bindings: sound: Convert Samsung I2S controller to dt-schema
by Marek Szyprowski 24 Sep '19

24 Sep '19

From: Maciej Falkowski <m.falkowski(a)samsung.com> Convert Samsung I2S controller to newer dt-schema format. Signed-off-by: Maciej Falkowski <m.falkowski(a)samsung.com> Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v4: - Removed description of i2s device nodes' clocks from 'clocks' property. - Added 'clock-output-names' property. - Added description of clock names. - Added '#clock-cells' property to required properties - Description of the provided clocks moved to samsung-i2s.h in separate patch Best regards, Maciej Falkowski --- .../devicetree/bindings/sound/samsung-i2s.txt | 84 ----------- .../bindings/sound/samsung-i2s.yaml | 136 ++++++++++++++++++ 2 files changed, 136 insertions(+), 84 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.txt b/Documentation/devicetree/bindings/sound/samsung-i2s.txt deleted file mode 100644 index a88cb00fa096..000000000000 --- a/Documentation/devicetree/bindings/sound/samsung-i2s.txt +++ /dev/null @@ -1,84 +0,0 @@ -* Samsung I2S controller - -Required SoC Specific Properties: - -- compatible : should be one of the following. - - samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. - - samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with - secondary fifo, s/w reset control and internal mux for root clk src. - - samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for - playback, stereo channel capture, secondary fifo using internal - or external dma, s/w reset control, internal mux for root clk src - and 7.1 channel TDM support for playback. TDM (Time division multiplexing) - is to allow transfer of multiple channel audio data on single data line. - - samsung,exynos7-i2s: with all the available features of exynos5 i2s, - exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo - with only external dma and more no.of root clk sampling frequencies. - - samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports - stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with - slightly modified bit offsets. - -- reg: physical base address of the controller and length of memory mapped - region. -- dmas: list of DMA controller phandle and DMA request line ordered pairs. -- dma-names: identifier string for each DMA request line in the dmas property. - These strings correspond 1:1 with the ordered pairs in dmas. -- clocks: Handle to iis clock and RCLK source clk. -- clock-names: - i2s0 uses some base clocks from CMU and some are from audio subsystem internal - clock controller. The clock names for i2s0 should be "iis", "i2s_opclk0" and - "i2s_opclk1" as shown in the example below. - i2s1 and i2s2 uses clocks from CMU. The clock names for i2s1 and i2s2 should - be "iis" and "i2s_opclk0". - "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources of the root - clk. i2s0 has internal mux to select the source of root clk and i2s1 and i2s2 - doesn't have any such mux. -- #clock-cells: should be 1, this property must be present if the I2S device - is a clock provider in terms of the common clock bindings, described in - ../clock/clock-bindings.txt. -- clock-output-names (deprecated): from the common clock bindings, names of - the CDCLK I2S output clocks, suggested values are "i2s_cdclk0", "i2s_cdclk1", - "i2s_cdclk3" for the I2S0, I2S1, I2S2 devices respectively. - -There are following clocks available at the I2S device nodes: - CLK_I2S_CDCLK - the CDCLK (CODECLKO) gate clock, - CLK_I2S_RCLK_PSR - the RCLK prescaler divider clock (corresponding to the - IISPSR register), - CLK_I2S_RCLK_SRC - the RCLKSRC mux clock (corresponding to RCLKSRC bit in - IISMOD register). - -Refer to the SoC datasheet for availability of the above clocks. -The CLK_I2S_RCLK_PSR and CLK_I2S_RCLK_SRC clocks are usually only available -in the IIS Multi Audio Interface. - -Note: Old DTs may not have the #clock-cells property and then not use the I2S -node as a clock supplier. - -Optional SoC Specific Properties: - -- samsung,idma-addr: Internal DMA register base address of the audio - sub system(used in secondary sound source). -- pinctrl-0: Should specify pin control groups used for this controller. -- pinctrl-names: Should contain only one value - "default". -- #sound-dai-cells: should be 1. - - -Example: - -i2s0: i2s@3830000 { - compatible = "samsung,s5pv210-i2s"; - reg = <0x03830000 0x100>; - dmas = <&pdma0 10 - &pdma0 9 - &pdma0 8>; - dma-names = "tx", "rx", "tx-sec"; - clocks = <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_I2S_BUS>, - <&clock_audss EXYNOS_SCLK_I2S>; - clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; - #clock-cells = <1>; - samsung,idma-addr = <0x03000000>; - pinctrl-names = "default"; - pinctrl-0 = <&i2s0_bus>; - #sound-dai-cells = <1>; -}; diff --git a/Documentation/devicetree/bindings/sound/samsung-i2s.yaml b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml new file mode 100644 index 000000000000..7d2750c26f11 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung-i2s.yaml @@ -0,0 +1,136 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/samsung-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Samsung SoC I2S controller + +maintainers: + - Krzysztof Kozlowski <krzk(a)kernel.org> + - Sylwester Nawrocki <s.nawrocki(a)samsung.com> + +properties: + compatible: + description: | + samsung,s3c6410-i2s: for 8/16/24bit stereo I2S. + + samsung,s5pv210-i2s: for 8/16/24bit multichannel(5.1) I2S with + secondary fifo, s/w reset control and internal mux for root clk src. + + samsung,exynos5420-i2s: for 8/16/24bit multichannel(5.1) I2S for + playback, stereo channel capture, secondary fifo using internal + or external dma, s/w reset control, internal mux for root clk src + and 7.1 channel TDM support for playback. TDM (Time division multiplexing) + is to allow transfer of multiple channel audio data on single data line. + + samsung,exynos7-i2s: with all the available features of exynos5 i2s. + exynos7 I2S has 7.1 channel TDM support for capture, secondary fifo + with only external dma and more no.of root clk sampling frequencies. + + samsung,exynos7-i2s1: I2S1 on previous samsung platforms supports + stereo channels. exynos7 i2s1 upgraded to 5.1 multichannel with + slightly modified bit offsets. + enum: + - samsung,s3c6410-i2s + - samsung,s5pv210-i2s + - samsung,exynos5420-i2s + - samsung,exynos7-i2s + - samsung,exynos7-i2s1 + + reg: + maxItems: 1 + + dmas: + minItems: 2 + maxItems: 3 + + dma-names: + oneOf: + - items: + - const: tx + - const: rx + - items: + - const: tx + - const: rx + - const: tx-sec + + clocks: + minItems: 1 + maxItems: 3 + + clock-names: + oneOf: + - items: + - const: iis + - items: # for i2s0 + - const: iis + - const: i2s_opclk0 + - const: i2s_opclk1 + - items: # for i2s1 and i2s2 + - const: iis + - const: i2s_opclk0 + description: | + "iis" is the i2s bus clock and i2s_opclk0, i2s_opclk1 are sources + of the root clk. i2s0 has internal mux to select the source + of root clk and i2s1 and i2s2 doesn't have any such mux. + + "#clock-cells": + const: 1 + + clock-output-names: # deprecated + oneOf: + - items: # for i2s0 + - const: i2s_cdclk0 + - items: # for i2s1 + - const: i2s_cdclk1 + - items: # for i2s2 + - const: i2s_cdclk2 + description: Names of the CDCLK I2S output clocks. + + samsung,idma-addr: + $ref: /schemas/types.yaml#/definitions/uint32 + description: | + Internal DMA register base address of the audio + sub system(used in secondary sound source). + + pinctrl-0: + description: Should specify pin control groups used for this controller. + + pinctrl-names: + const: default + + "#sound-dai-cells": + const: 1 + +required: + - compatible + - reg + - dmas + - dma-names + - clocks + - clock-names + - "#clock-cells" + +examples: + - | + #include <dt-bindings/clock/exynos-audss-clk.h> + + i2s0: i2s@3830000 { + compatible = "samsung,s5pv210-i2s"; + reg = <0x03830000 0x100>; + dmas = <&pdma0 10>, + <&pdma0 9>, + <&pdma0 8>; + dma-names = "tx", "rx", "tx-sec"; + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_SCLK_I2S>; + clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; + #clock-cells = <1>; + samsung,idma-addr = <0x03000000>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s0_bus>; + #sound-dai-cells = <1>; + }; + -- 2.17.1

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add DSD support for EVGA NU Audio
by Jussi Laako 24 Sep '19

24 Sep '19

EVGA NU Audio is actually a USB audio device on a PCIexpress card, with it's own USB controller. It supports both PCM and DSD. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 5fd4ccce452d..fbfde996fee7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1658,6 +1658,7 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case 0x25ce: /* Mytek devices */ case 0x278b: /* Rotel? */ case 0x2ab6: /* T+A devices */ + case 0x3842: /* EVGA */ case 0xc502: /* HiBy devices */ if (fp->dsd_raw) return SNDRV_PCM_FMTBIT_DSD_U32_BE; -- 2.17.1

2 1

[alsa-devel] [PATCH alsa-lib] pcm: chmap: Fix memory leak at snd_pcm_set_chmap()
by Takashi Iwai 24 Sep '19

24 Sep '19

snd_pcm_set_chmap() leaks the memory returned from snd_pcm_get_chmap() without releasing. Add the missing free() call as well as a slight code refactoring. Reported-by: Conrad Jones BugLink: https://github.com/alsa-project/alsa-lib/pull/11 Fixes: d20e24e5d161 ("chmap: Always succeed setting the map to what it already is") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 178d43875f00..d6bf31ac7ae3 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -8099,8 +8099,13 @@ snd_pcm_chmap_t *snd_pcm_get_chmap(snd_pcm_t *pcm) */ int snd_pcm_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map) { - const snd_pcm_chmap_t *oldmap = snd_pcm_get_chmap(pcm); - if (oldmap && chmap_equal(oldmap, map)) + const snd_pcm_chmap_t *oldmap; + int nochange; + + oldmap = snd_pcm_get_chmap(pcm); + nochange = (oldmap && chmap_equal(oldmap, map)); + free((void *)oldmap); + if (nochange) return 0; if (!pcm->ops->set_chmap) -- 2.16.4

1 0

[alsa-devel] TDM audio support and sample size updates for tegra30
by Ben Dooks 24 Sep '19

24 Sep '19

A patch series for TDM audio, non 16bit sample size and a small fifo fix for tegra30 series SoC. Changes since v1: - moved edge-control and data-offset to the set_fmt callbacks - fixed dev_dbg in set_tdm callback - fixed dev_dbg message contents in set_tdm callback - changed fsync width to be permanently 1 clock

3 11

Re: [alsa-devel] [PATCH V3 4/4] ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8
by S.j. Wang 24 Sep '19

24 Sep '19

Hi > > One issue for error-out and some nit-pickings inline. Thanks. > > On Thu, Sep 19, 2019 at 08:11:42PM +0800, Shengjiu Wang wrote: > > There is error "aplay: pcm_write:2023: write error: Input/output error" > > on i.MX8QM/i.MX8QXP platform for S24_3LE format. > > > > In i.MX8QM/i.MX8QXP, the DMA is EDMA, which don't support 24bit > > sample, but we didn't add any constraint, that cause issues. > > > > So we need to query the caps of dma, then update the hw parameters > > according to the caps. > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/fsl_asrc.c | 4 +-- > > sound/soc/fsl/fsl_asrc.h | 3 +++ > > sound/soc/fsl/fsl_asrc_dma.c | 52 > > +++++++++++++++++++++++++++++++----- > > 3 files changed, 50 insertions(+), 9 deletions(-) > > > > @@ -276,6 +274,11 @@ static int fsl_asrc_dma_startup(struct > snd_pcm_substream *substream) > > struct device *dev = component->dev; > > struct fsl_asrc *asrc_priv = dev_get_drvdata(dev); > > struct fsl_asrc_pair *pair; > > + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; > > + u8 dir = tx ? OUT : IN; > > + struct dma_chan *tmp_chan; > > + struct snd_dmaengine_dai_dma_data *dma_data; > > Nit: would it be possible to reorganize these a bit? Usually we put struct > things together unless there is a dependency, similar to > fsl_asrc_dma_hw_params(). > > > @@ -285,9 +288,44 @@ static int fsl_asrc_dma_startup(struct > > snd_pcm_substream *substream) > > > > runtime->private_data = pair; > > > > + /* Request a temp pair, which is release in the end */ > > Nit: "which will be released later" or "and will release it later"? And could > we use a work like "dummy"? Or at least I would love to see the comments > explaining the parameter "1" > in the function call below. > > > + ret = fsl_asrc_request_pair(1, pair); > > + if (ret < 0) { > > + dev_err(dev, "failed to request asrc pair\n"); > > + return ret; > > + } > > + > > + tmp_chan = fsl_asrc_get_dma_channel(pair, dir); > > + if (!tmp_chan) { > > + dev_err(dev, "can't get dma channel\n"); > > Could we align with other error messages using "failed to"? > > > + ret = snd_soc_set_runtime_hwparams(substream, > &snd_imx_hardware); > > + if (ret) > > + return ret; > > + > [...] > > + dma_release_channel(tmp_chan); > > + fsl_asrc_release_pair(pair); > > I think we need an "out:" here for those error-out routines to goto. > Otherwise, it'd be a pair leak? > > > + > > Could we drop this? There is a blank line below already :) > Will update them. Best regards Wang Shengjiu

1 0

[alsa-devel] [PATCH] ASoC: soc-component: fix a couple missing error assignments
by Dan Carpenter 23 Sep '19

23 Sep '19

There were a couple places where the return value wasn't assigned so the error handling wouldn't trigger. Fixes: 5c0769af4caf ("ASoC: soc-dai: add snd_soc_dai_bespoke_trigger()") Fixes: 95aef3553384 ("ASoC: soc-dai: add snd_soc_dai_trigger()") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e163dde5eab1..a1b99ac57d9e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1070,7 +1070,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - snd_soc_dai_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; @@ -1097,7 +1097,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, return ret; } - snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); + ret = snd_soc_dai_bespoke_trigger(cpu_dai, substream, cmd); if (ret < 0) return ret; -- 2.20.1

2 1

[alsa-devel] [PATCH trivial] sound: Fix Kconfig indentation
by Krzysztof Kozlowski 23 Sep '19

23 Sep '19

Adjust indentation from spaces to tab (+optional two spaces) as in coding style with command like: $ sed -e 's/^ /\t/' -i */Kconfig Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/core/Kconfig | 28 ++++++++++---------- sound/drivers/Kconfig | 20 +++++++------- sound/firewire/Kconfig | 6 ++--- sound/isa/Kconfig | 18 ++++++------- sound/mips/Kconfig | 12 ++++----- sound/pci/Kconfig | 2 +- sound/soc/cirrus/Kconfig | 14 +++++----- sound/soc/codecs/Kconfig | 32 +++++++++++------------ sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Kconfig | 48 +++++++++++++++++----------------- sound/soc/pxa/Kconfig | 16 ++++++------ sound/soc/qcom/Kconfig | 20 +++++++------- sound/soc/samsung/Kconfig | 8 +++--- sound/soc/sof/imx/Kconfig | 12 ++++----- sound/soc/sof/intel/Kconfig | 22 ++++++++-------- sound/soc/xilinx/Kconfig | 20 +++++++------- sound/soc/zte/Kconfig | 12 ++++----- sound/usb/Kconfig | 32 +++++++++++------------ 18 files changed, 162 insertions(+), 162 deletions(-) diff --git a/sound/core/Kconfig b/sound/core/Kconfig index 4ee79ad6ae22..4044c42d8595 100644 --- a/sound/core/Kconfig +++ b/sound/core/Kconfig @@ -72,11 +72,11 @@ config SND_PCM_OSS config SND_PCM_OSS_PLUGINS bool "OSS PCM (digital audio) API - Include plugin system" depends on SND_PCM_OSS - default y + default y help - If you disable this option, the ALSA's OSS PCM API will not - support conversion of channels, formats and rates. It will - behave like most of new OSS/Free drivers in 2.4/2.6 kernels. + If you disable this option, the ALSA's OSS PCM API will not + support conversion of channels, formats and rates. It will + behave like most of new OSS/Free drivers in 2.4/2.6 kernels. config SND_PCM_TIMER bool "PCM timer interface" if EXPERT @@ -128,13 +128,13 @@ config SND_SUPPORT_OLD_API or older). config SND_PROC_FS - bool "Sound Proc FS Support" if EXPERT - depends on PROC_FS - default y - help - Say 'N' to disable Sound proc FS, which may reduce code size about - 9KB on x86_64 platform. - If unsure say Y. + bool "Sound Proc FS Support" if EXPERT + depends on PROC_FS + default y + help + Say 'N' to disable Sound proc FS, which may reduce code size about + 9KB on x86_64 platform. + If unsure say Y. config SND_VERBOSE_PROCFS bool "Verbose procfs contents" @@ -142,8 +142,8 @@ config SND_VERBOSE_PROCFS default y help Say Y here to include code for verbose procfs contents (provides - useful information to developers when a problem occurs). On the - other side, it makes the ALSA subsystem larger. + useful information to developers when a problem occurs). On the + other side, it makes the ALSA subsystem larger. config SND_VERBOSE_PRINTK bool "Verbose printk" @@ -164,7 +164,7 @@ config SND_DEBUG_VERBOSE depends on SND_DEBUG help Say Y here to enable extra-verbose debugging messages. - + Let me repeat: it enables EXTRA-VERBOSE DEBUGGING messages. So, say Y only if you are ready to be annoyed. diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 09932cc98e9d..15d6d46acf9c 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only config SND_MPU401_UART - tristate - select SND_RAWMIDI + tristate + select SND_RAWMIDI config SND_OPL3_LIB tristate @@ -90,16 +90,16 @@ config SND_DUMMY will be called snd-dummy. config SND_ALOOP - tristate "Generic loopback driver (PCM)" - select SND_PCM - help - Say 'Y' or 'M' to include support for the PCM loopback device. + tristate "Generic loopback driver (PCM)" + select SND_PCM + help + Say 'Y' or 'M' to include support for the PCM loopback device. This module returns played samples back to the user space using the standard ALSA PCM device. The devices are routed 0->1 and - 1->0, where first number is the playback PCM device and second + 1->0, where first number is the playback PCM device and second number is the capture device. Module creates two PCM devices and configured number of substreams (see the pcm_substreams module - parameter). + parameter). The loopback device allows time sychronization with an external timing source using the time shift universal control (+-20% @@ -142,12 +142,12 @@ config SND_MTS64 select SND_RAWMIDI help The ESI Miditerminal 4140 is a 4 In 4 Out MIDI Interface with - additional SMPTE Timecode capabilities for the parallel port. + additional SMPTE Timecode capabilities for the parallel port. Say 'Y' to include support for this device. To compile this driver as a module, chose 'M' here: the module - will be called snd-mts64. + will be called snd-mts64. config SND_SERIAL_U16550 tristate "UART16550 serial MIDI driver" diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index b0a904cdb932..995c2cefc222 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -77,7 +77,7 @@ config SND_BEBOB tristate "BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware" select SND_FIREWIRE_LIB select SND_HWDEP - help + help Say Y here to include support for FireWire devices based on BridgeCo DM1000/DM1100/DM1500 with BeBoB firmware: * Edirol FA-66/FA-101 @@ -111,8 +111,8 @@ config SND_BEBOB * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro - To compile this driver as a module, choose M here: the module - will be called snd-bebob. + To compile this driver as a module, choose M here: the module + will be called snd-bebob. config SND_FIREWIRE_DIGI00X tristate "Digidesign Digi 002/003 family support" diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index b690ed937cbe..6ffa48dd5983 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -2,22 +2,22 @@ # ALSA ISA drivers config SND_WSS_LIB - tristate - select SND_PCM + tristate + select SND_PCM select SND_TIMER config SND_SB_COMMON - tristate + tristate config SND_SB8_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON config SND_SB16_DSP - tristate - select SND_PCM - select SND_SB_COMMON + tristate + select SND_PCM + select SND_SB_COMMON menuconfig SND_ISA bool "ISA sound devices" diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig index 8a33402fd415..b497b803c834 100644 --- a/sound/mips/Kconfig +++ b/sound/mips/Kconfig @@ -14,15 +14,15 @@ config SND_SGI_O2 tristate "SGI O2 Audio" depends on SGI_IP32 select SND_PCM - help - Sound support for the SGI O2 Workstation. + help + Sound support for the SGI O2 Workstation. config SND_SGI_HAL2 - tristate "SGI HAL2 Audio" - depends on SGI_HAS_HAL2 + tristate "SGI HAL2 Audio" + depends on SGI_HAS_HAL2 select SND_PCM - help - Sound support for the SGI Indy and Indigo2 Workstation. + help + Sound support for the SGI Indy and Indigo2 Workstation. endif # SND_MIPS diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 7630f808d087..93bc9bef7641 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -217,7 +217,7 @@ config SND_CMIPCI will be called snd-cmipci. config SND_OXYGEN_LIB - tristate + tristate config SND_OXYGEN tristate "C-Media 8786, 8787, 8788 (Oxygen)" diff --git a/sound/soc/cirrus/Kconfig b/sound/soc/cirrus/Kconfig index 2333efac758a..8039a8febefa 100644 --- a/sound/soc/cirrus/Kconfig +++ b/sound/soc/cirrus/Kconfig @@ -33,13 +33,13 @@ config SND_EP93XX_SOC_AC97 select SND_SOC_AC97_BUS config SND_EP93XX_SOC_SNAPPERCL15 - tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" - depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C - select SND_EP93XX_SOC_I2S - select SND_SOC_TLV320AIC23_I2C - help - Say Y or M here if you want to add support for I2S audio on the - Bluewater Systems Snapper CL15 module. + tristate "SoC Audio support for Bluewater Systems Snapper CL15 module" + depends on SND_EP93XX_SOC && MACH_SNAPPER_CL15 && I2C + select SND_EP93XX_SOC_I2S + select SND_SOC_TLV320AIC23_I2C + help + Say Y or M here if you want to add support for I2S audio on the + Bluewater Systems Snapper CL15 module. config SND_EP93XX_SOC_SIMONE tristate "SoC Audio support for Simplemachines Sim.One board" diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 229cc89f8c5a..ef9d73b89623 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -257,16 +257,16 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) - help - Normally ASoC codec drivers are only built if a machine driver which - uses them is also built since they are only usable with a machine - driver. Selecting this option will allow these drivers to be built - without an explicit machine driver for test and development purposes. + help + Normally ASoC codec drivers are only built if a machine driver which + uses them is also built since they are only usable with a machine + driver. Selecting this option will allow these drivers to be built + without an explicit machine driver for test and development purposes. Support for the bus types used to access the codecs to be built must be selected separately. - If unsure select "N". + If unsure select "N". config SND_SOC_88PM860X tristate @@ -570,8 +570,8 @@ config SND_SOC_CS42XX8_I2C # Cirrus Logic CS43130 HiFi DAC config SND_SOC_CS43130 - tristate "Cirrus Logic CS43130 CODEC" - depends on I2C + tristate "Cirrus Logic CS43130 CODEC" + depends on I2C config SND_SOC_CS4341 tristate "Cirrus Logic CS4341 CODEC" @@ -643,19 +643,19 @@ config SND_SOC_L3 tristate config SND_SOC_DA7210 - tristate + tristate config SND_SOC_DA7213 - tristate + tristate config SND_SOC_DA7218 tristate config SND_SOC_DA7219 - tristate + tristate config SND_SOC_DA732X - tristate + tristate config SND_SOC_DA9055 tristate @@ -717,7 +717,7 @@ config SND_SOC_INNO_RK3036 select REGMAP_MMIO config SND_SOC_ISABELLE - tristate + tristate config SND_SOC_LM49453 tristate @@ -988,7 +988,7 @@ config SND_SOC_RT5640 tristate config SND_SOC_RT5645 - tristate + tristate config SND_SOC_RT5651 tristate @@ -1220,7 +1220,7 @@ config SND_SOC_UDA134X tristate config SND_SOC_UDA1380 - tristate + tristate depends on I2C config SND_SOC_WCD9335 @@ -1348,7 +1348,7 @@ config SND_SOC_WM8904 depends on I2C config SND_SOC_WM8940 - tristate + tristate config SND_SOC_WM8955 tristate diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 01c99750212a..597516c0c192 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -113,7 +113,7 @@ config SND_SOC_INTEL_SKYLAKE select SND_SOC_INTEL_CNL select SND_SOC_INTEL_CFL help - This is a backwards-compatible option to select all devices + This is a backwards-compatible option to select all devices supported by the Intel SST/Skylake driver. This option is no longer recommended and will be deprecated when the SOF driver is introduced. Distributions should explicitly diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 5c27f7ab4a5f..882ff36a7c9c 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -3,13 +3,13 @@ menuconfig SND_SOC_INTEL_MACH bool "Intel Machine drivers" depends on SND_SOC_INTEL_SST_TOPLEVEL || SND_SOC_SOF_INTEL_TOPLEVEL help - Intel ASoC Machine Drivers. If you have a Intel machine that - has an audio controller with a DSP and I2S or DMIC port, then - enable this option by saying Y + Intel ASoC Machine Drivers. If you have a Intel machine that + has an audio controller with a DSP and I2S or DMIC port, then + enable this option by saying Y - Note that the answer to this question doesn't directly affect the - kernel: saying N will just cause the configurator to skip all - the questions about Intel ASoC machine drivers. + Note that the answer to this question doesn't directly affect the + kernel: saying N will just cause the configurator to skip all + the questions about Intel ASoC machine drivers. if SND_SOC_INTEL_MACH @@ -114,11 +114,11 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH depends on X86_INTEL_LPSS || COMPILE_TEST select SND_SOC_ACPI select SND_SOC_RT5670 - help - This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell - platforms with RT5672 audio codec. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" @@ -311,20 +311,20 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927_MACH If unsure select "N". config SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH - tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" + tristate "KBL with RT5663, RT5514 and MAX98927 in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST - depends on SPI - select SND_SOC_RT5663 - select SND_SOC_RT5514 - select SND_SOC_RT5514_SPI - select SND_SOC_MAX98927 - select SND_SOC_HDAC_HDMI - help - This adds support for ASoC Onboard Codec I2S machine driver. This will - create an alsa sound card for RT5663 + RT5514 + MAX98927. - Say Y or m if you have such a device. This is a recommended option. - If unsure select "N". + depends on SPI + select SND_SOC_RT5663 + select SND_SOC_RT5514 + select SND_SOC_RT5514_SPI + select SND_SOC_MAX98927 + select SND_SOC_HDAC_HDMI + help + This adds support for ASoC Onboard Codec I2S machine driver. This will + create an alsa sound card for RT5663 + RT5514 + MAX98927. + Say Y or m if you have such a device. This is a recommended option. + If unsure select "N". config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH tristate "KBL with DA7219 and MAX98357A in I2S Mode" @@ -393,7 +393,7 @@ config SND_SOC_INTEL_SKL_HDA_DSP_GENERIC_MACH help This adds support for ASoC machine driver for Intel platforms SKL/KBL/BXT/APL with iDisp, HDA audio codecs. - Say Y or m if you have such a device. This is a recommended option. + Say Y or m if you have such a device. This is a recommended option. If unsure select "N". endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 213d4dab0346..295cfffa4646 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -190,14 +190,14 @@ config SND_PXA2XX_SOC_MAGICIAN HTC Magician. config SND_PXA2XX_SOC_MIOA701 - tristate "SoC Audio support for MIO A701" - depends on SND_PXA2XX_SOC && MACH_MIOA701 + tristate "SoC Audio support for MIO A701" + depends on SND_PXA2XX_SOC && MACH_MIOA701 depends on AC97_BUS=n - select SND_PXA2XX_SOC_AC97 - select SND_SOC_WM9713 - help - Say Y if you want to add support for SoC audio on the - MIO A701. + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9713 + help + Say Y if you want to add support for SoC audio on the + MIO A701. config SND_PXA2XX_SOC_IMOTE2 tristate "SoC Audio support for IMote 2" @@ -205,7 +205,7 @@ config SND_PXA2XX_SOC_IMOTE2 select SND_PXA2XX_SOC_I2S select SND_SOC_WM8940 help - Say Y if you want to add support for SoC audio on the + Say Y if you want to add support for SoC audio on the IMote 2. config SND_MMP_SOC_BROWNSTONE diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 60086858e920..6530d2462a9e 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -3,8 +3,8 @@ config SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST help - Say Y or M if you want to add support to use audio devices - in Qualcomm Technologies SOC-based platforms. + Say Y or M if you want to add support to use audio devices + in Qualcomm Technologies SOC-based platforms. config SND_SOC_LPASS_CPU tristate @@ -30,17 +30,17 @@ config SND_SOC_STORM select SND_SOC_LPASS_IPQ806X select SND_SOC_MAX98357A help - Say Y or M if you want add support for SoC audio on the - Qualcomm Technologies IPQ806X-based Storm board. + Say Y or M if you want add support for SoC audio on the + Qualcomm Technologies IPQ806X-based Storm board. config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" depends on SND_SOC_QCOM select SND_SOC_LPASS_APQ8016 help - Support for Qualcomm Technologies LPASS audio block in - APQ8016 SOC-based systems. - Say Y if you want to use audio devices on MI2S. + Support for Qualcomm Technologies LPASS audio block in + APQ8016 SOC-based systems. + Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON tristate @@ -93,9 +93,9 @@ config SND_SOC_MSM8996 select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON help - Support for Qualcomm Technologies LPASS audio block in - APQ8096 SoC-based systems. - Say Y if you want to use audio device on this SoCs + Support for Qualcomm Technologies LPASS audio block in + APQ8096 SoC-based systems. + Say Y if you want to use audio device on this SoCs config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 638983123d8f..9304177de78a 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -195,10 +195,10 @@ config SND_SOC_ODROID Say Y here to enable audio support for the Odroid XU3/XU4. config SND_SOC_ARNDALE_RT5631_ALC5631 - tristate "Audio support for RT5631(ALC5631) on Arndale Board" - depends on I2C - select SND_SAMSUNG_I2S - select SND_SOC_RT5631 + tristate "Audio support for RT5631(ALC5631) on Arndale Board" + depends on I2C + select SND_SAMSUNG_I2S + select SND_SOC_RT5631 config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" diff --git a/sound/soc/sof/imx/Kconfig b/sound/soc/sof/imx/Kconfig index 5acae75f5750..6315fba8ce71 100644 --- a/sound/soc/sof/imx/Kconfig +++ b/sound/soc/sof/imx/Kconfig @@ -5,9 +5,9 @@ config SND_SOC_SOF_IMX_TOPLEVEL depends on ARM64|| COMPILE_TEST depends on SND_SOC_SOF_OF help - This adds support for Sound Open Firmware for NXP i.MX platforms. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX platforms. + Say Y if you have such a device. + If unsure select "N". if SND_SOC_SOF_IMX_TOPLEVEL @@ -16,8 +16,8 @@ config SND_SOC_SOF_IMX8 depends on IMX_SCU depends on IMX_DSP help - This adds support for Sound Open Firmware for NXP i.MX8 platforms - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for NXP i.MX8 platforms + Say Y if you have such a device. + If unsure select "N". endif ## SND_SOC_SOF_IMX_IMX_TOPLEVEL diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 479ba249e219..18c41fdc47c8 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -36,7 +36,7 @@ config SND_SOC_SOF_INTEL_PCI config SND_SOC_SOF_INTEL_HIFI_EP_IPC tristate help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_INTEL_ATOM_HIFI_EP @@ -217,31 +217,31 @@ config SND_SOC_SOF_COMETLAKE_H_SUPPORT config SND_SOC_SOF_TIGERLAKE_SUPPORT bool "SOF support for Tigerlake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the Tigerlake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the Tigerlake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_TIGERLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_ELKHARTLAKE_SUPPORT bool "SOF support for ElkhartLake" help - This adds support for Sound Open Firmware for Intel(R) platforms - using the ElkhartLake processors. - Say Y if you have such a device. - If unsure select "N". + This adds support for Sound Open Firmware for Intel(R) platforms + using the ElkhartLake processors. + Say Y if you have such a device. + If unsure select "N". config SND_SOC_SOF_ELKHARTLAKE tristate select SND_SOC_SOF_HDA_COMMON help - This option is not user-selectable but automagically handled by + This option is not user-selectable but automagically handled by 'select' statements at a higher level config SND_SOC_SOF_HDA_COMMON diff --git a/sound/soc/xilinx/Kconfig b/sound/soc/xilinx/Kconfig index 69973179ef15..1d3586b68db7 100644 --- a/sound/soc/xilinx/Kconfig +++ b/sound/soc/xilinx/Kconfig @@ -9,15 +9,15 @@ config SND_SOC_XILINX_I2S encapsulates PCM in AES format and sends AES data. config SND_SOC_XILINX_AUDIO_FORMATTER - tristate "Audio support for the the Xilinx audio formatter" - help - Select this option to enable Xilinx audio formatter - support. This provides DMA platform device support for - audio functionality. + tristate "Audio support for the the Xilinx audio formatter" + help + Select this option to enable Xilinx audio formatter + support. This provides DMA platform device support for + audio functionality. config SND_SOC_XILINX_SPDIF - tristate "Audio support for the the Xilinx SPDIF" - help - Select this option to enable Xilinx SPDIF Audio. - This provides playback and capture of SPDIF audio in - AES format. + tristate "Audio support for the the Xilinx SPDIF" + help + Select this option to enable Xilinx SPDIF Audio. + This provides playback and capture of SPDIF audio in + AES format. diff --git a/sound/soc/zte/Kconfig b/sound/soc/zte/Kconfig index a7842e4b791c..a23d4f13ca19 100644 --- a/sound/soc/zte/Kconfig +++ b/sound/soc/zte/Kconfig @@ -18,9 +18,9 @@ config ZX_I2S ZTE ZX I2S interface config ZX_TDM - tristate "ZTE ZX TDM Driver Support" - depends on COMMON_CLK - select SND_SOC_GENERIC_DMAENGINE_PCM - help - Say Y or M if you want to add support for codecs attached to the - ZTE ZX TDM interface + tristate "ZTE ZX TDM Driver Support" + depends on COMMON_CLK + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for codecs attached to the + ZTE ZX TDM interface diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e2c53a0841da..059242f15d75 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -107,24 +107,24 @@ config SND_USB_US122L will be called snd-usb-us122l. config SND_USB_6FIRE - tristate "TerraTec DMX 6Fire USB" - select FW_LOADER - select BITREVERSE - select SND_RAWMIDI - select SND_PCM - select SND_VMASTER - help - Say Y here to include support for TerraTec 6fire DMX USB interface. - - You will need firmware files in order to be able to use the device - after it has been coldstarted. An install script for the firmware - and further help can be found at - http://sixfireusb.sourceforge.net + tristate "TerraTec DMX 6Fire USB" + select FW_LOADER + select BITREVERSE + select SND_RAWMIDI + select SND_PCM + select SND_VMASTER + help + Say Y here to include support for TerraTec 6fire DMX USB interface. + + You will need firmware files in order to be able to use the device + after it has been coldstarted. An install script for the firmware + and further help can be found at + http://sixfireusb.sourceforge.net config SND_USB_HIFACE - tristate "M2Tech hiFace USB-SPDIF driver" - select SND_PCM - help + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help Select this option to include support for M2Tech hiFace USB-SPDIF interface. -- 2.17.1

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v5.4
by Mark Brown 23 Sep '19

23 Sep '19

The following changes since commit fca11622d600228bec405456f41590155b3a3eca: ASoC: sdm845: remove unneeded semicolon (2019-09-15 10:29:51 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.4-rc1 for you to fetch changes up to 147162f575152db80000fb3de26358264768ee9f: ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies (2019-09-20 12:06:48 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.4 A small smattering of ASoC fixes for v5.4 - nothing too exciting here, all small standalone things. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies Bard Liao (1): ASoC: core: delete component->card_list in soc_remove_component only Bard liao (1): ASoC: core: use list_del_init and move it back to soc_cleanup_component Gregory CLEMENT (1): ASoC: atmel_ssc_dai: Remove wrong spinlock usage Kuninori Morimoto (1): ASoC: rsnd: do error check after rsnd_channel_normalization() Mihai Serban (1): ASoC: fsl_sai: Fix noise when using EDMA Peter Ujfalusi (1): ASoC: pcm3168a: The codec does not support S32_LE sound/soc/atmel/atmel_ssc_dai.c | 12 ++---------- sound/soc/atmel/atmel_ssc_dai.h | 1 - sound/soc/codecs/pcm3168a.c | 3 +-- sound/soc/fsl/fsl_sai.c | 15 +++++++++++++++ sound/soc/fsl/fsl_sai.h | 1 + sound/soc/sh/rcar/ssi.c | 10 +++++----- sound/soc/soc-core.c | 2 +- sound/soc/ti/Kconfig | 11 +++++++++-- 8 files changed, 34 insertions(+), 21 deletions(-)

2 1

[alsa-devel] [PATCH V3 0/4] update supported sample format
by Shengjiu Wang 21 Sep '19

21 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_refine_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch changes in v3 - Fix build report by kbuild test robot <lkp(a)intel.com> - change snd_dmaengine_pcm_set_runtime_hwparams to snd_dmaengine_pcm_refine_runtime_hwparams include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 52 ++++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 62 +++----------------- 6 files changed, 188 insertions(+), 86 deletions(-) -- 2.21.0

2 7

[alsa-devel] [PATCH 1/2] PCI: Add a helper to check Power Resource Requirements _PR3 existence
by Kai-Heng Feng 20 Sep '19

20 Sep '19

A driver may want to know the existence of _PR3, to choose different runtime suspend behavior. A user will be add in next patch. This is mostly the same as nouveau_pr3_present(). Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- drivers/pci/pci.c | 20 ++++++++++++++++++++ include/linux/pci.h | 2 ++ 2 files changed, 22 insertions(+) diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c index 1b27b5af3d55..776af15b92c2 100644 --- a/drivers/pci/pci.c +++ b/drivers/pci/pci.c @@ -5856,6 +5856,26 @@ int pci_set_vga_state(struct pci_dev *dev, bool decode, return 0; } +bool pci_pr3_present(struct pci_dev *pdev) +{ + struct pci_dev *parent_pdev = pci_upstream_bridge(pdev); + struct acpi_device *parent_adev; + + if (acpi_disabled) + return false; + + if (!parent_pdev) + return false; + + parent_adev = ACPI_COMPANION(&parent_pdev->dev); + if (!parent_adev) + return false; + + return parent_adev->power.flags.power_resources && + acpi_has_method(parent_adev->handle, "_PR3"); +} +EXPORT_SYMBOL_GPL(pci_pr3_present); + /** * pci_add_dma_alias - Add a DMA devfn alias for a device * @dev: the PCI device for which alias is added diff --git a/include/linux/pci.h b/include/linux/pci.h index 82e4cd1b7ac3..9b6f7b67fac9 100644 --- a/include/linux/pci.h +++ b/include/linux/pci.h @@ -2348,9 +2348,11 @@ struct irq_domain *pci_host_bridge_acpi_msi_domain(struct pci_bus *bus); void pci_msi_register_fwnode_provider(struct fwnode_handle *(*fn)(struct device *)); +bool pci_pr3_present(struct pci_dev *pdev); #else static inline struct irq_domain * pci_host_bridge_acpi_msi_domain(struct pci_bus *bus) { return NULL; } +static bool pci_pr3_present(struct pci_dev *pdev) { return false; } #endif #ifdef CONFIG_EEH -- 2.17.1

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[alsa-devel] [PATCH v7 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 20 Sep '19

20 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c https://patchwork.kernel.org/patch/11047447 has been picked up by Mark Brown in ASoC tree for-5.4 branch. Changes from v6 to v7: 1. rockchip_max98090: Use phandle of HDMI from DTS to find codec_dai. With this we don't need to set fixed id for the created hdmi-audio-codec device. Cheng-Yi Chiang (4): drm: bridge: dw-hdmi: Report connector status using callback ASoC: rockchip-max98090: Add description for rockchip,hdmi-codec ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../bindings/sound/rockchip-max98090.txt | 2 + .../boot/dts/rk3288-veyron-analog-audio.dtsi | 1 + .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 11 ++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rockchip_max98090.c | 149 ++++++++++++++---- 7 files changed, 182 insertions(+), 29 deletions(-) -- 2.23.0.237.gc6a4ce50a0-goog

2 6

[alsa-devel] [PATCH] ASoC: core: use list_del_init and move it back to soc_cleanup_component
by Bard liao 20 Sep '19

20 Sep '19

commit a0a4bf57a977 ("ASoC: core: delete component->card_list in soc_remove_component only") was trying to fix a kernel oops when list_del was called twice without re-init the list. Use list_del_init() can solve it, too. Besides, it will be more readable if we cleanup all component related resource at soc_cleanup_component(). Suggested-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-core.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index aff4b4bf4d07..88978a3036c4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,6 +978,7 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del_init(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -990,7 +991,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - list_del(&component->card_list); + soc_cleanup_component(component); } -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: pcm3168a: The codec does not support S32_LE
by Peter Ujfalusi 20 Sep '19

20 Sep '19

24 bits is supported in all modes and 16 bit only when the codec is slave and the DAI is set to RIGHT_J. Remove the unsupported sample format. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index 50ed86d45c26..88b75695fbf7 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -21,8 +21,7 @@ #define PCM3168A_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) #define PCM3168A_FMT_I2S 0x0 #define PCM3168A_FMT_LEFT_J 0x1 -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: ti: fix SND_SOC_DM365_VOICE_CODEC dependencies
by Arnd Bergmann 20 Sep '19

20 Sep '19

SND_SOC_DM365_VOICE_CODEC is a 'bool' option in a choice statement, meaning it cannot be set to =m, but it selects two other drivers that we may want to be loadable modules after all: WARNING: unmet direct dependencies detected for SND_SOC_CQ0093VC Depends on [m]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] Selected by [y]: - SND_SOC_DM365_VOICE_CODEC [=y] && <choice> Selected by [m]: - SND_SOC_ALL_CODECS [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMPILE_TEST [=y] Add an intermediate symbol that sets SND_SOC_CQ0093VC and MFD_DAVINCI_VOICECODEC to =m if SND_SOC=m. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/ti/Kconfig | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig index 87a9b9dd4e98..29f61053ab62 100644 --- a/sound/soc/ti/Kconfig +++ b/sound/soc/ti/Kconfig @@ -200,11 +200,18 @@ config SND_SOC_DM365_AIC3X_CODEC config SND_SOC_DM365_VOICE_CODEC bool "Voice Codec - CQ93VC" - select MFD_DAVINCI_VOICECODEC - select SND_SOC_CQ0093VC help Say Y if you want to add support for SoC On-chip voice codec endchoice +config SND_SOC_DM365_VOICE_CODEC_MODULE + def_tristate y + depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC + select MFD_DAVINCI_VOICECODEC + select SND_SOC_CQ0093VC + help + The is an internal symbol needed to ensure that the codec + and MFD driver can be built as loadable modules if necessary. + endmenu -- 2.20.0

2 1

[alsa-devel] [PATCH] ASoC: soc-component.h: tidyup for soc-component.h
by Kuninori Morimoto 20 Sep '19

20 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc-component.h already has SPDX License, GPL explanation is not needed. And it is included from soc.h only. No need to header-test. This patch tidyup these. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark this is for for-5.4 include/Kbuild | 1 + include/sound/soc-component.h | 6 ------ 2 files changed, 1 insertion(+), 6 deletions(-) diff --git a/include/Kbuild b/include/Kbuild index c38f0d4..c6909b5 100644 --- a/include/Kbuild +++ b/include/Kbuild @@ -1052,6 +1052,7 @@ header-test- += sound/soc-acpi-intel-match.h header-test- += sound/soc-dai.h header-test- += sound/soc-dapm.h header-test- += sound/soc-dpcm.h +header-test- += sound/soc-component.h header-test- += sound/sof/control.h header-test- += sound/sof/dai-intel.h header-test- += sound/sof/dai.h diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 5d80b2e..a869199 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -3,16 +3,10 @@ * soc-component.h * * Copyright (c) 2019 Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. */ #ifndef __SOC_COMPONENT_H #define __SOC_COMPONENT_H -#include <sound/soc.h> - /* * Component probe and remove ordering levels for components with runtime * dependencies. -- 2.7.4

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[alsa-devel] RfC / [PATCH] Jack sense support for Medion E4254
by Jan-Marek Glogowski 20 Sep '19

20 Sep '19

Hi everybody, I recently bought the laptop. It has one jack for attaching a headset. Jack sense is announced by the HDA config but not working. Except for some positioning, and color information, the PIN config seemed to be sensible, and everything I tried to override, didn't result in a working jack. Manually unmuting the jack using hda-analyser resulted in working output, so generally the config seemed to work. Reading the patch-realtek code, if found ALC256_FIXUP_ASUS_HEADSET_MODE, which somehow changed the pin 0x19 to be associated with the 0x21 headset out pin, resulting in this dmesg change: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 And for whatever reason jack sense is now working, at least for some normal output-only headset (currently I don't have one included a mic with me, so actually can't test that mic volume works correctly). I was just wondering, if my pin values are actually correct, because they still define different association, as I understand it, but the ASUS fix even works with these wrong channels /associations 2 and 4. So the attached patch works for me, but I'm still wondering, if this is the correct solution. FWIW, jack sense works fine in Windows. Anything I might be missing here? Jan-Marek ------- >From 0ccc5f07d8737690bd2df6d88a5af0f1f36d0e40 Mon Sep 17 00:00:00 2001 From: Jan-Marek Glogowski <glogow(a)fbihome.de> Date: Sun, 15 Sep 2019 16:57:28 +0200 Subject: [PATCH] ALSA: hda/realtek: PCI quirk for Medion E4254 The laptop has a combined jack to attach headsets on the right. The BIOS encodes them as two different colored jacks at the front, but otherwise it seems to be configured ok. But any adaption of the pins config on its own doesn't fix the jack detection to work in Linux. Still Windows works correct. This is somehow fixed by chaining ALC256_FIXUP_ASUS_HEADSET_MODE, which seems to register the microphone jack as a headset part and also results in fixing jack sensing, visible in dmesg as: -snd_hda_codec_realtek hdaudioC0D0: Mic=0x19 +snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 Signed-off-by: Jan-Marek Glogowski <glogow(a)fbihome.de> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c1ddfd2fac52..448e0146dab2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5800,6 +5800,7 @@ enum { ALC256_FIXUP_ASUS_MIC_NO_PRESENCE, ALC299_FIXUP_PREDATOR_SPK, ALC294_FIXUP_ASUS_INTSPK_HEADSET_MIC, + ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -6850,6 +6851,16 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x04a11040 }, + { 0x21, 0x04211020 }, + { } + }, + .chained = true, + .chain_id = ALC256_FIXUP_ASUS_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7113,6 +7124,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ + SND_PCI_QUIRK(0x10ec, 0x118c, "Medion EE4254 MD62100", ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. @@ -7280,6 +7292,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC225_FIXUP_HEADSET_JACK, .name = "alc-headset-jack"}, {.id = ALC295_FIXUP_CHROME_BOOK, .name = "alc-chrome-book"}, {.id = ALC299_FIXUP_PREDATOR_SPK, .name = "predator-spk"}, + {.id = ALC256_FIXUP_MEDION_HEADSET_NO_PRESENCE, .name = "alc256-medion-headset"}, {} }; #define ALC225_STANDARD_PINS \ -- 2.20.1

2 3

[alsa-devel] [PATCH] ALSA: hda - Apply AMD controller workaround for Raven platform
by Takashi Iwai 20 Sep '19

20 Sep '19

It's reported that the garbled sound on HP Envy x360 13z-ag000 (Ryzen Laptop) is fixed by the same workaround applied to other AMD chips. Update the driver_data entry for Raven (1022:15e3) to use the newly introduced preset, AZX_DCAPS_PRESET_AMD_SB. Since it already contains AZX_DCAPS_PM_RUNTIME, we can drop that bit, too. Reported-and-tested-by: Dennis Padiernos <depadiernos(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 91e71be42fa4..240f4ca76391 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2485,8 +2485,7 @@ static const struct pci_device_id azx_ids[] = { AZX_DCAPS_PM_RUNTIME }, /* AMD Raven */ { PCI_DEVICE(0x1022, 0x15e3), - .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_SB | - AZX_DCAPS_PM_RUNTIME }, + .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_AMD_SB }, /* ATI HDMI */ { PCI_DEVICE(0x1002, 0x0002), .driver_data = AZX_DRIVER_ATIHDMI_NS | AZX_DCAPS_PRESET_ATI_HDMI_NS }, -- 2.16.4

1 0

[alsa-devel] alsa-lib: Add ucm support for whiskeylake sof-skl_hda_card audio
by Elimar Riesebieter 19 Sep '19

19 Sep '19

Hi ALSA devs, please notice the attached suggestion from Lenovo to support sound for whiskeylake sof-skl_hda_card audio. Thanks Elimar -- You cannot propel yourself forward by patting yourself on the back.

4 3

[alsa-devel] [PATCH v1 0/9] Basic sound support for Arndale board / wm8994 updates
by Sylwester Nawrocki 19 Sep '19

19 Sep '19

This patch series adds basic audio support for Exynos5250 SoC based Arndale board, the Bluetooth receiver source and HDMI output are not covered yet. There is also one fix for wm8994 driver related to WM1811 CODEC and wm8994 updates to handle MCLK clocks, similar to patches: ae1ea48c5c59 ("ASoC: arizona: Add gating for source clocks of the FLLs") 7a4413d0dc96 ("ASoC: arizona: Add gating for clock when used for direct MCLK") Sylwester Nawrocki (9): ASoC: wm8994: Do not register inapplicable controls for WM1811 mfd: wm8994: Add support for MCLKn clock control ASoC: wm8994: Add support for setting MCLKn clock rate ASoC: wm8994: Add support for MCLKn clock gating ASoC: samsung: arndale: Simplify DAI link initialization ASoC: dt-bindings: Document "samsung,arndale-wm1811" compatible ASoC: samsung: arndale: Add support for WM1811 CODEC ASoC: samsung: arndale: Add missing OF node dereferencing ARM: dts: arndale: Add audio support (WM1811 CODEC boards) .../devicetree/bindings/sound/arndale.txt | 5 +- arch/arm/boot/dts/exynos5250-arndale.dts | 27 ++- drivers/mfd/wm8994-core.c | 9 + include/linux/mfd/wm8994/core.h | 9 + sound/soc/codecs/wm8994.c | 164 +++++++++++++++--- sound/soc/samsung/arndale_rt5631.c | 155 +++++++++++++---- 6 files changed, 306 insertions(+), 63 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v6] dt-bindings: ASoC: Add tas2770 smart PA dt bindings
by shifu0704＠thundersoft.com 19 Sep '19

19 Sep '19

From: Frank Shi <shifu0704(a)thundersoft.com> Add tas2770 smart PA dt bindings Signed-off-by: Frank Shi <shifu0704(a)thundersoft.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 ++++++++++++++++++++++ 1 file changed, 37 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt new file mode 100644 index 0000000..ede6bb3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.txt @@ -0,0 +1,37 @@ +Texas Instruments TAS2770 Smart PA + +The TAS2770 is a mono, digital input Class-D audio amplifier optimized for +efficiently driving high peak power into small loudspeakers. +Integrated speaker voltage and current sense provides for +real time monitoring of loudspeaker behavior. + +Required properties: + + - compatible: - Should contain "ti,tas2770". + - reg: - The i2c address. Should contain <0x4c>, <0x4d>,<0x4e>, or <0x4f>. + - #address-cells - Should be <1>. + - #size-cells - Should be <0>. + - ti,asi-format: - Sets TDM RX capture edge. 0->Rising; 1->Falling. + - ti,imon-slot-no:- TDM TX current sense time slot. + - ti,vmon-slot-no:- TDM TX voltage sense time slot. + +Optional properties: + +- interrupt-parent: the phandle to the interrupt controller which provides + the interrupt. +- interrupts: interrupt specification for data-ready. + +Examples: + + tas2770@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #address-cells = <1>; + #size-cells = <0>; + interrupt-parent = <&msm_gpio>; + interrupts = <97 0>; + ti,asi-format = <0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + -- 2.7.4

2 2

[alsa-devel] DPCM: suspicious code in dpcm_prune_paths()
by Guennadi Liakhovetski 19 Sep '19

19 Sep '19

Hi, This code in dpcm_prune_paths() seems to have no effect: /* is there a valid CODEC DAI widget for this BE */ for_each_rtd_codec_dai(dpcm->be, i, dai) { widget = dai_get_widget(dai, stream); /* prune the BE if it's no longer in our active list */ if (widget && widget_in_list(list, widget)) continue; } Neither of the variables, that are modified in this loop are used outside of it. Was the "continue" maybe meant to continue the encompassing loop? Thanks Guennadi

2 1

[alsa-devel] [PATCH v6 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 19 Sep '19

19 Sep '19

This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v5: - updated slave bindings with proper licencing and fixed up examples. Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 62 + .../soundwire/soundwire-controller.yaml | 82 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1345 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

3 9

Re: [alsa-devel] [PATCH V2 3/4] ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_set_runtime_hwparams
by S.j. Wang 19 Sep '19

19 Sep '19

Hi > > When set the runtime hardware parameters, we may need to query the > capability of DMA to complete the parameters. > > This patch is to Extract this operation from > dmaengine_pcm_set_runtime_hwparams function to a separate function > snd_dmaengine_pcm_set_runtime_hwparams, that other components > which need this feature can call this function. > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > --- kbuild test robot report compile issue, will resend v3 after fixing. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ASoC: core: delete component->card_list in soc_remove_component only
by Bard liao 19 Sep '19

19 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> We add component->card_list in the end of soc_probe_component(). In other words, component->card_list will not be added if there is an error in the soc_probe_component() function. So we can't delete component->card_list in the error handling of soc_probe_component(). Fixes: 22d1423187e5 ("ASoC: soc-core: add soc_cleanup_component()") Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35f48e9c5ead..aff4b4bf4d07 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -978,7 +978,6 @@ static void soc_cleanup_component(struct snd_soc_component *component) /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); - list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); component->card = NULL; @@ -991,7 +990,7 @@ static void soc_remove_component(struct snd_soc_component *component) return; snd_soc_component_remove(component); - + list_del(&component->card_list); soc_cleanup_component(component); } -- 2.17.1

5 12

[alsa-devel] [RFC PATCH 0/9] soundwire: add Master device support, GreyBus style
by Pierre-Louis Bossart 18 Sep '19

18 Sep '19

The current use of platform devices leads to limitations in terms of API and problematic reference counts flagged during the review of sysfs patches. As suggested by Greg KH, this patch series introduces a 'Master device' and gets rid of platform devices. The code is inspired by GreyBus, e.g. 'gb_hd_driver' is translated for the SoundWire context as 'sdw_md_driver' and likewise 'gb_host_device' is translated as 'sdw_master_device'. There are differences in the way devices are created, using device_register instead of the multiple steps used in GreyBus. All I know about GreyBus is that it's based on Unipro, so the code updates are based on a first-order analysis, it could very well be I missed concepts while trying to reverse-engineer the code in staging/greybus/. I focused on the concepts mainly and it's also possible that the dual-license for new files or plan-vanilla EXPORT_SYMBOL are not appropriate for all the changes in this series. Please don't take errors as deliberate intent to work-around GPL but rather as an indicator of maturity of the code only developed in the last few days. These changes make it possible to provide new callbacks, e.g splitting the bus startup from the initializations in probe, which is very much desired on Intel platforms to detect the machine driver as early as possible before the hardware is fully powered/enable. These patches will be provided as a separate RFC later today to illustrate the benefit of these changes. To make the code more consistent the first series of patches are renames. I did not rename 'sdw_slave' or 'module_sdw_driver' to avoid compatibility issues since there are codec drivers almost ready for integration. I don't have any specific opinion on if/when additional renames should be done, as long as there is a means to clearly identify what is specific to a SoundWire Master I am fine. Feedback and reviews welcome. Bard Liao (1): soundwire: add device driver to sdw_md_driver Pierre-Louis Bossart (8): soundwire: renames to prepare support for master drivers/devices soundwire: rename dev_to_sdw_dev macro soundwire: rename drv_to_sdw_slave_driver macro soundwire: bus_type: rename sdw_drv_ to sdw_slave_drv soundwire: intel: rename res field as link_res soundwire: add support for sdw_slave_type soundwire: add initial definitions for sdw_master_device soundwire: intel: remove platform devices and provide new interface drivers/base/regmap/regmap-sdw.c | 4 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/bus.c | 2 +- drivers/soundwire/bus_type.c | 60 +++--- drivers/soundwire/intel.c | 117 ++++++----- drivers/soundwire/intel.h | 22 +-- drivers/soundwire/intel_init.c | 293 +++++++++++++++++++--------- drivers/soundwire/master.c | 80 ++++++++ drivers/soundwire/slave.c | 9 +- include/linux/soundwire/sdw.h | 39 +++- include/linux/soundwire/sdw_intel.h | 86 +++++++- include/linux/soundwire/sdw_type.h | 34 +++- 12 files changed, 551 insertions(+), 197 deletions(-) create mode 100644 drivers/soundwire/master.c -- 2.20.1

2 17

[alsa-devel] [PATCH] ASoC: rsnd: do error check after rsnd_channel_normalization()
by Kuninori Morimoto 18 Sep '19

18 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSI need to use rsnd_channel_normalization() for TDM-split mode, thus, channel check need to do after that. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark This is for v5.4 sound/soc/sh/rcar/ssi.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f6a7466..fc5d089 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -286,6 +286,11 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_ssi_is_multi_slave(mod, io)) return 0; + if (rsnd_runtime_is_tdm_split(io)) + chan = rsnd_io_converted_chan(io); + + chan = rsnd_channel_normalization(chan); + if (ssi->usrcnt > 0) { if (ssi->rate != rate) { dev_err(dev, "SSI parent/child should use same rate\n"); @@ -300,11 +305,6 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, return 0; } - if (rsnd_runtime_is_tdm_split(io)) - chan = rsnd_io_converted_chan(io); - - chan = rsnd_channel_normalization(chan); - main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

2 1

[alsa-devel] [PATCH V2 0/4] update supported sample format
by Shengjiu Wang 18 Sep '19

18 Sep '19

This patch serial is to update the supported format for fsl_asrc and fix some format issue. Shengjiu Wang (4): ASoC: fsl_asrc: Use in(out)put_format instead of in(out)put_word_width ASoC: fsl_asrc: update supported sample format ASoC: pcm_dmaengine: Extract snd_dmaengine_pcm_set_runtime_hwparams ASoC: fsl_asrc: Fix error with S24_3LE format bitstream in i.MX8 changes in v2 - extract snd_dmaengine_pcm_set_runtime_hwparams in one separate path. - 4th patch depends on 3rd patch include/sound/dmaengine_pcm.h | 5 ++ sound/core/pcm_dmaengine.c | 83 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_asrc.c | 65 ++++++++++++++------- sound/soc/fsl/fsl_asrc.h | 7 ++- sound/soc/fsl/fsl_asrc_dma.c | 48 +++++++++++++--- sound/soc/soc-generic-dmaengine-pcm.c | 62 ++------------------ 6 files changed, 181 insertions(+), 89 deletions(-) -- 2.21.0

2 5

[alsa-devel] RFC: TDM mode support on the tegra30
by Ben Dooks 18 Sep '19

18 Sep '19

This is an initial set that adds support for both the TDM and for formats other than 8/16bit audio through the audio system. It also fixes a minor issue with the FIFO stopping order. I have further patches following on which deal with selection of clocks to allow the use of the tegra's audio unit to run as clock slave (all current examples have the tegra as clock master).

6 33

[alsa-devel] Licence change for your alsa-lib UCM commits (Marcin Juszkiewicz: 5 total)
by Jaroslav Kysela 18 Sep '19

18 Sep '19

As discussed, we would like to move the UCM configuration files from the alsa-lib repository to new alsa-ucm-conf repository with the licence change from LGPL-2.1 to BSD-3-Clause. I would like to check, if you agree with this licence change. Please, answer to this e-mail and write your agreement / disagreement (keep CC to the alsa-devel mailing list for the archiving purposes). Thank you for your time and co-operation. Reference: https://mailman.alsa-project.org/pipermail/alsa-devel/2019-July/153257.html List of your commit(s): f07c14b08280fb461b7f070d55a625e1f5e88a1e conf/ucm: DAISY-I2S: add Samsung ARM Chromebook UCM config from ChromeOS 5b019ea310e6a5d6ff8933bcbca1f3f3471be33b conf/ucm: PandaBoard: add UCM config from Ubuntu 9d0f93af0b35bfdcbd3d758d98715508b69d6008 conf/ucm: PandaBoardES: add UCM config from Ubuntu 7db247be11381bef6e38a41f1d51fa78a99350b5 conf/ucm: SDP4430: add UCM config from Ubuntu 52de910868f56ee7e6b07a1a2ad6a54913f745d0 conf/ucm: tegraalc5632: add UCM config from Ubuntu --- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

15 19

[alsa-devel] 5.1 to stereo using libasound
by Eric Desjardins 18 Sep '19

18 Sep '19

Hi, I am playing a mkv with 5.1 channels from a mkv. I configure libasound for 6 channels and I feed them interleave audio using snd_pcm_writei. I use the default pulse audio device. So far, nothing is wrong. However, I do not ear things correctly. I listen this on my stereo speaker. There is no mixdown to stereo and I was wondering how to do that. I tried to use chmap api but anything I try on the default device returns an error. I was wondering what was the correct way to do that? Using the mixing api? How do we detect that my device output stereo if chmap doesn't work? I am on a ubuntu 18.04 linux box. Thanks, Eric

2 3

[alsa-devel] [PATCH v6 0/4] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. The dependent change on hdmi-codec.c https://patchwork.kernel.org/patch/11047447 has been picked up by Mark Brown in ASoC tree for-5.4 branch. Changes from v5 to v6: 1. Remove the patch for sound/soc/codecs/hdmi-codec.c because it is accepted. 2. Rebase the rest of patches based on drm-misc-next tree. Cheng-Yi Chiang (4): drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 6 files changed, 153 insertions(+), 27 deletions(-) -- 2.23.0.237.gc6a4ce50a0-goog

2 6

[alsa-devel] [PATCH v2] ASoC: atmel_ssc_dai: Remove wrong spinlock usage
by Gregory CLEMENT 18 Sep '19

18 Sep '19

A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Reviewed-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Signed-off-by: Gregory CLEMENT <gregory.clement(a)bootlin.com> --- Changelog: v1 -> v2: - Removed the spinlock from the atmel_ssc_info struct - Added the Reviewed-by flag form Alex sound/soc/atmel/atmel_ssc_dai.c | 12 ++---------- sound/soc/atmel/atmel_ssc_dai.h | 1 - 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6f89483ac88c..786b48ae4905 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); - if (ssc_p->dir_mask & dir_mask) { - spin_unlock_irq(&ssc_p->lock); + if (ssc_p->dir_mask & dir_mask) return -EBUSY; - } + ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +349,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +362,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index ae764cb541c7..3470b966e449 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -93,7 +93,6 @@ struct atmel_ssc_state { struct atmel_ssc_info { char *name; struct ssc_device *ssc; - spinlock_t lock; /* lock for dir_mask */ unsigned short dir_mask; /* 0=unused, 1=playback, 2=capture */ unsigned short initialized; /* true if SSC has been initialized */ unsigned short daifmt; -- 2.23.0

2 1

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: Remove wrong spinlock usage
by Gregory CLEMENT 18 Sep '19

18 Sep '19

A potential bug was reported in the email "[BUG] atmel_ssc_dai: a possible sleep-in-atomic bug in atmel_ssc_shutdown"[1] Indeed in the function atmel_ssc_shutdown() free_irq() was called in a critical section protected by spinlock. However this spinlock is only used in atmel_ssc_shutdown() and atmel_ssc_startup() functions. After further analysis, it occurred that the call to these function are already protected by mutex used on the calling functions. Then we can remove the spinlock which will fix this bug as a side effect. Thanks to this patch the following message disappears: "BUG: sleeping function called from invalid context at kernel/locking/mutex.c:909" [1]: https://www.spinics.net/lists/alsa-devel/msg71286.html Signed-off-by: Gregory CLEMENT <gregory.clement(a)bootlin.com> --- sound/soc/atmel/atmel_ssc_dai.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6f89483ac88c..365957e86419 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -116,19 +116,16 @@ static struct atmel_pcm_dma_params ssc_dma_params[NUM_SSC_DEVICES][2] = { static struct atmel_ssc_info ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc1", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, { .name = "ssc2", - .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = SSC_DIR_MASK_UNUSED, .initialized = 0, }, @@ -317,13 +314,11 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, snd_soc_dai_set_dma_data(dai, substream, dma_params); - spin_lock_irq(&ssc_p->lock); if (ssc_p->dir_mask & dir_mask) { spin_unlock_irq(&ssc_p->lock); return -EBUSY; } ssc_p->dir_mask |= dir_mask; - spin_unlock_irq(&ssc_p->lock); return 0; } @@ -355,7 +350,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dir_mask = 1 << dir; - spin_lock_irq(&ssc_p->lock); ssc_p->dir_mask &= ~dir_mask; if (!ssc_p->dir_mask) { if (ssc_p->initialized) { @@ -369,7 +363,6 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, ssc_p->cmr_div = ssc_p->tcmr_period = ssc_p->rcmr_period = 0; ssc_p->forced_divider = 0; } - spin_unlock_irq(&ssc_p->lock); /* Shutdown the SSC clock. */ pr_debug("atmel_ssc_dai: Stopping clock\n"); -- 2.23.0

2 2

[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 18 Sep '19

18 Sep '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

1 0

[alsa-devel] [PATCH v5 0/5] Add HDMI jack support on RK3288
by Cheng-Yi Chiang 18 Sep '19

18 Sep '19

This patch series supports HDMI jack reporting on RK3288, which uses DRM dw-hdmi driver and hdmi-codec codec driver. The previous discussion about reporting jack status using hdmi-notifier and drm_audio_component is at https://lore.kernel.org/patchwork/patch/1083027/ The new approach is to use a callback mechanism that is specific to hdmi-codec. Changes from v4 to v5: - synopsys/Kconfig: Remove the incorrect dependency change in v4. - rockchip/Kconfig: Add dependency of hdmi-codec when it is really need for jack support. Cheng-Yi Chiang (5): ASoC: hdmi-codec: Add an op to set callback function for plug event drm: bridge: dw-hdmi: Report connector status using callback drm: dw-hdmi-i2s: Use fixed id for codec device ASoC: rockchip_max98090: Add dai_link for HDMI ASoC: rockchip_max98090: Add HDMI jack support .../drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 13 +- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 41 ++++++- include/drm/bridge/dw_hdmi.h | 4 + include/sound/hdmi-codec.h | 17 +++ sound/soc/codecs/hdmi-codec.c | 46 +++++++ sound/soc/rockchip/Kconfig | 3 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 3 +- sound/soc/rockchip/rockchip_max98090.c | 116 ++++++++++++++---- 8 files changed, 216 insertions(+), 27 deletions(-) -- 2.22.0.510.g264f2c817a-goog

4 12

[alsa-devel] alsa acting up
by liebrecht＠grossmann-venter.com 18 Sep '19

18 Sep '19

See images in links. 1) "http://grossmann-venter.com/visitors/ug/alsa/alsa-pulse-181vsl_unplug.png" 2) "http://grossmann-venter.com/visitors/ug/alsa/alsa-dont recognize1818VSLpng" 1) As soon as pulseaudio is started Alsa says 1818VSL is unplugged....? 2) Clearly Presonus 1818VSL is found as lsusb output says, but alsa has no clue about it. How do I get alsa to recognise 181VSL as it is found by lsusb so it is presnet on the system. It used to work fine but alsa just stopped working with 1818VSL recently for no reason. I know this is a developers group, but please at least answer where I can get help. Alsa help is absolutely non-existing. I also posted a bug with no response in another thread. Thanks.

1 0