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

[alsa-devel] [PATCH] ALSA: usb-audio: fix spelling mistake "Frequence" -> "Frequency"
by Colin King 02 Oct '20

02 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in equalizer name fields, fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer_us16x08.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c index 26ed23b18b77..4f594f9552a8 100644 --- a/sound/usb/mixer_us16x08.c +++ b/sound/usb/mixer_us16x08.c @@ -1119,7 +1119,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ Low Frequence", + .name = "EQ Low Frequency", }, { /* EQ mid low gain */ .kcontrol_new = &snd_us16x08_eq_gain_ctl, @@ -1133,7 +1133,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQLOWMIDFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidLow Frequence", + .name = "EQ MidLow Frequency", }, { /* EQ mid low Q */ .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, @@ -1154,7 +1154,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQHIGHMIDFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ MidHigh Frequence", + .name = "EQ MidHigh Frequency", }, { /* EQ mid high Q */ .kcontrol_new = &snd_us16x08_eq_mid_width_ctl, @@ -1175,7 +1175,7 @@ static const struct snd_us16x08_control_params eq_controls[] = { .control_id = SND_US16X08_ID_EQHIGHFREQ, .type = USB_MIXER_U8, .num_channels = 16, - .name = "EQ High Frequence", + .name = "EQ High Frequency", }, }; -- 2.19.1

3 3

[PATCH V2] ASoC: Intel: boards: Use FS as nau8825 sysclk in nau88125_* machine
by Radoslaw Biernacki 08 Sep '20

08 Sep '20

This single fix address two issues on machines with nau88125: 1) Audio distortion, due to lack of required clock rate on MCLK line 2) Loud audible "pops" on headphones if there is no sysclk during nau8825 playback power up sequence Explanation for: 1) Due to Skylake HW limitation, MCLK pin can only output 24MHz clk rate (it can be only connected to XTAL parent clk). The BCLK pin can be driven by dividers and therefore FW is able to set it to rate required by chosen audio format. According to nau8825 datasheet, 256*FS sysclk gives the best audio quality and the only way to achieve this (taking into account the above limitations) its to regenerate the MCLK from BCLK on nau8825 side by FFL. Without required clk rate, audio is distorted by added harmonics. 2) Currently Skylake does not output MCLK/FS when the back-end DAI op hw_param is called, so we cannot switch to MCLK/FS in hw_param. This patch reduces pop by letting nau8825 keep using its internal VCO clock during widget power up sequence, until SNDRV_PCM_TRIGGER_START when MCLK/FS is available. Once device resumes, the system will only enable power sequence for playback without doing hardware parameter, audio format, and PLL configure. In the mean time, the jack detecion sequence has changed PLL parameters and switched to internal clock. Thus, the playback signal distorted without correct PLL parameters. That is why we need to configure the PLL again in SNDRV_PCM_TRIGGER_RESUME case. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Radoslaw Biernacki <rad(a)semihalf.com> --- .../soc/intel/boards/skl_nau88l25_max98357a.c | 72 +++++++++++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++++++----- 2 files changed, 107 insertions(+), 38 deletions(-) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index d7b8154c43a4..0f3cea1342d1 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/jack.h> #include <sound/pcm.h> @@ -47,7 +48,7 @@ enum { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; @@ -60,14 +61,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -292,24 +286,40 @@ static const struct snd_soc_ops skylake_nau8825_fe_ops = { .startup = skl_fe_startup, }; -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + int ret = 0; - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + msleep(20); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, @@ -630,10 +640,34 @@ static int skylake_card_late_probe(struct snd_soc_card *card) return hdac_hdmi_jack_port_init(component, &card->dapm); } +static int __maybe_unused skylake_nau8825_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + dev_dbg(codec_dai->dev, "playback_active:%d playback_widget->active:%d codec_dai->rate:%d\n", + codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], + codec_dai->playback_widget->active, + codec_dai->rate); + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825max", .owner = THIS_MODULE, + .resume_post = skylake_nau8825_resume_post, .dai_link = skylake_dais, .num_links = ARRAY_SIZE(skylake_dais), .controls = skylake_controls, diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 4b317bcf6ea0..be44a40067f0 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/platform_device.h> +#include <linux/delay.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -57,7 +58,7 @@ static const struct snd_kcontrol_new skylake_controls[] = { }; static int platform_clock_control(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *k, int event) + struct snd_kcontrol *k, int event) { struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_card *card = dapm->card; @@ -70,14 +71,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } - if (SND_SOC_DAPM_EVENT_ON(event)) { - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); - if (ret < 0) { - dev_err(card->dev, "set sysclk err = %d\n", ret); - return -EIO; - } - } else { + if (!SND_SOC_DAPM_EVENT_ON(event)) { ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_INTERNAL, 0, SND_SOC_CLOCK_IN); if (ret < 0) { @@ -85,6 +79,7 @@ static int platform_clock_control(struct snd_soc_dapm_widget *w, return -EIO; } } + return ret; } @@ -344,24 +339,40 @@ static int skylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -static int skylake_nau8825_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) +static int skylake_nau8825_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, - NAU8825_CLK_MCLK, 24000000, SND_SOC_CLOCK_IN); + int ret = 0; - if (ret < 0) - dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ret = snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "can't set FS clock %d\n", ret); + break; + } + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + break; + case SNDRV_PCM_TRIGGER_RESUME: + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, runtime->rate, + runtime->rate * 256); + if (ret < 0) + dev_err(codec_dai->dev, "can't set FLL: %d\n", ret); + msleep(20); + break; + } return ret; } -static const struct snd_soc_ops skylake_nau8825_ops = { - .hw_params = skylake_nau8825_hw_params, +static struct snd_soc_ops skylake_nau8825_ops = { + .trigger = skylake_nau8825_trigger, }; static const unsigned int channels_dmic[] = { @@ -671,10 +682,34 @@ static int skylake_card_late_probe(struct snd_soc_card *card) return hdac_hdmi_jack_port_init(component, &card->dapm); } +static int __maybe_unused skylake_nau8825_resume_post(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai; + + codec_dai = snd_soc_card_get_codec_dai(card, SKL_NUVOTON_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found\n"); + return -EIO; + } + + dev_dbg(codec_dai->dev, "playback_active:%d playback_widget->active:%d codec_dai->rate:%d\n", + codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK], + codec_dai->playback_widget->active, + codec_dai->rate); + + if (codec_dai->stream_active[SNDRV_PCM_STREAM_PLAYBACK] && + codec_dai->playback_widget->active) + snd_soc_dai_set_sysclk(codec_dai, NAU8825_CLK_FLL_FS, 0, + SND_SOC_CLOCK_IN); + + return 0; +} + /* skylake audio machine driver for SPT + NAU88L25 */ static struct snd_soc_card skylake_audio_card = { .name = "sklnau8825adi", .owner = THIS_MODULE, + .resume_post = skylake_nau8825_resume_post, .dai_link = skylake_dais, .num_links = ARRAY_SIZE(skylake_dais), .controls = skylake_controls, -- 2.17.1

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[PATCH v3 0/7] Add H6 I2S support
by Clément Péron 04 Sep '20

04 Sep '20

Hi, This is a sequel of Marcus Cooper serie[0], where remarks made by Maxime have been fixed. I have tested it on my Beelink GS1 board. Thanks, Clement 0: https://lore.kernel.org/patchwork/cover/1139949/ Changes since v2 (thanks Maxime): - Add details in commit log about sign extend sample - Only set FIFO regs as volatile in regmap - Missing a space (detected by checkpatch) Changes since v1: - Fix missing header in set sign extend sample Jernej Skrabec (3): dt-bindings: ASoC: sun4i-i2s: Add H6 compatible ASoC: sun4i-i2s: Add support for H6 I2S arm64: dts: sun50i-h6: Add HDMI audio to H6 DTSI Marcus Cooper (4): ASoC: sun4i-i2s: Adjust LRCLK width ASoC: sun4i-i2s: Set sign extend sample ASoc: sun4i-i2s: Add 20 and 24 bit support ASoC: sun4i-i2s: Adjust regmap settings .../sound/allwinner,sun4i-a10-i2s.yaml | 2 + arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 31 ++ sound/soc/sunxi/sun4i-i2s.c | 265 +++++++++++++++++- 3 files changed, 296 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] Attempting to understand odd snd-usb-audio code and behavior
by Keith A. Milner 21 Aug '20

21 Aug '20

Over the last few weeks I've been trying to dig through the snd-usb-audio code to try to get an understanding of it. I've used ftrace and also a local version which I have heavily polluted with kprintf(KERN_INFO statements so I can track what it's doing in practice. I've been playing with this with some USB audio interfaces I have some which work, and some which don't, and trying to understand the differences and, ultimately, what I can do to make the non-functioning ones work. I've got to the point where I have a fairly good understanding of some parts, but I'm struggling to understand why the code is behaving as it does in practice. My observation when plugging the USB device in is that, as expected, it tries to build the audio device, and add components to it including attaching capture and playback streams. It seems to go through this phase without too many surprises, and we get to the standard dmesg output: usbcore: registered new interface driver snd-usb-audio After this, it gets confusing. It seems to open and close the PCMs (e.g, snd_usb_capture_open) several times in a row. Each time it sets up the hardware info, checks and sets the format and then closes the PCM. I don't see anything obviously different between most of these cycles. Here's an example my logging output (hopefully this will make sense): Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Setting up hw info Aug 22 20:04:16 kerneldev kernel: [ 923.140789] Checking format: Aug 22 20:04:16 kerneldev kernel: [ 923.140790] Interface number = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] alt setting index = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] format type = 1 Aug 22 20:04:16 kerneldev kernel: [ 923.140790] frame size = 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140791] Setting pt=125 ptmin=125 Aug 22 20:04:16 kerneldev kernel: [ 923.140792] Checking for unusual bit rates Aug 22 20:04:16 kerneldev kernel: [ 923.140794] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.140921] Called: pcm.c - snd_usb_hw_params for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Setting format: Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Bytes per period = 8816 Aug 22 20:04:16 kerneldev kernel: [ 923.140945] Frames per period = 1102 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Periods per buffer = 4 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Channels = 2 Aug 22 20:04:16 kerneldev kernel: [ 923.140946] Rate = 44100 Aug 22 20:04:16 kerneldev kernel: [ 923.141279] Called: endpoint.c - snd_usb_add_endpoint of type 0 with direction 1 Aug 22 20:04:16 kerneldev kernel: [ 923.141292] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141293] Called: pcm.c - configure_endpoint Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] Setting endpoint params for endpoint num 142 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] channels=2 Aug 22 20:04:16 kerneldev kernel: [ 923.141294] period bytes=8816 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] frames per period=1102 Aug 22 20:04:16 kerneldev kernel: [ 923.141295] periods per buffer=4 Aug 22 20:04:16 kerneldev kernel: [ 923.141296] Called: endpoint.c - data_ep_set_params for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.141301] Called: pcm.c - snd_usb_pcm_prepare for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142768] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142772] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142774] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.142830] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.142836] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142837] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:04:16 kerneldev kernel: [ 923.142846] Called: pcm.c - snd_usb_capture_close for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146760] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146834] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146838] Called: init.c - snd_card_file_remove Aug 22 20:04:16 kerneldev kernel: [ 923.146841] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146845] Called: init.c - snd_card_file_add Aug 22 20:04:16 kerneldev kernel: [ 923.146848] Called: pcm.c - snd_usb_capture_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Called: pcm.c - snd_usb_pcm_open for pcm substream 0 Aug 22 20:04:16 kerneldev kernel: [ 923.146849] Setting up hw info It seems to do this several times, before the behavior changing slightly, as it now starts to try to set the endpoint parameters and seems to leave the stream open. However, it then tries to open the stream and set the endpoint parameters again resulting in errors: Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Called: endpoint.c - snd_usb_endpoint_set_params for endpoint 142, type 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] Setting endpoint params for endpoint num 142 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] channels=2 Aug 22 20:22:00 kerneldev kernel: [ 1987.452945] period bytes=1768 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] frames per period=221 Aug 22 20:22:00 kerneldev kernel: [ 1987.452946] periods per buffer=593 Aug 22 20:22:00 kerneldev kernel: [ 1987.452947] usb 1-1: Unable to change format on ep #8e: already in use Maybe this is driven by hotplug or something, but it seems to be performing a lot of pointless activities. Eventually it seems to try to open the PCM playback or capture stream (with some devices it opens playback, with others capture) and then checks the pointer: Aug 22 20:22:00 kerneldev kernel: [ 1987.470194] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470198] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470656] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470657] Pointer is 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:00 kerneldev kernel: [ 1987.470660] Pointer is 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473567] Called: pcm.c - snd_usb_pcm_pointer for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473569] Pointer is 0 The above is from a non-working device. For a working device the pointer increments. Eventually this activity ends: Aug 22 20:22:05 kerneldev kernel: [ 1992.473575] Called: pcm.c - snd_usb_substream_playback_trigger for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473576] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473578] Called: endpoint.c - snd_usb_endpoint_stop for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.473602] Called: pcm.c - snd_usb_hw_free for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503287] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 142, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503288] Called: endpoint.c - snd_usb_endpoint_deactivate for endpoint 13, type 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503323] Called: pcm.c - snd_usb_playback_close for pcm substream 0 Aug 22 20:22:05 kerneldev kernel: [ 1992.503656] Called: init.c - snd_card_file_remove This seems as if it might be some sort of test of the PCMs, to make sure they are working. Certainly in non-working devices the pcm_pointer stays at zero and in working devices it increments. It also seems to try harder with some devices than others, even though the device descriptors seem very similar. Can anyone explain any of this to me, particularly why it's exhibiting this behaviour. I've uploaded one of the logs from a non-working device to Pastebin if anyone wants to look at a full example: https://pastebin.com/uBZEuQ57 Also, is this the best way to approach this, or are there other tools/techniques that can help? I've experimented with ftrace and that was useful, but didn't give the level of detail I wanted. Cheers, Keith

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[PATCH v4 1/2] dt-bindings: tas2562: Convert the tas2562 binding to yaml
by Dan Murphy 28 Jul '20

28 Jul '20

Convert the TAS2562 text file to yaml format. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2562.txt | 34 --------- .../devicetree/bindings/sound/tas2562.yaml | 69 +++++++++++++++++++ 2 files changed, 69 insertions(+), 34 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2562.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2562.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt deleted file mode 100644 index 94796b547184..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ /dev/null @@ -1,34 +0,0 @@ -Texas Instruments TAS2562 Smart PA - -The TAS2562 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: - - #address-cells - Should be <1>. - - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562", "ti,tas2563". - - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - - ti,imon-slot-no:- TDM TX current sense time slot. - -Optional properties: -- interrupt-parent: phandle to the interrupt controller which provides - the interrupt. -- interrupts: (GPIO) interrupt to which the chip is connected. -- shut-down: GPIO used to control the state of the device. - -Examples: -tas2562@4c { - #address-cells = <1>; - #size-cells = <0>; - compatible = "ti,tas2562"; - reg = <0x4c>; - - interrupt-parent = <&gpio1>; - interrupts = <14>; - - shut-down = <&gpio1 15 0>; - ti,imon-slot-no = <0>; -}; - diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml new file mode 100644 index 000000000000..d2d3401bf251 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml @@ -0,0 +1,69 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2562.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2562 Smart PA + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TAS2562 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. + +properties: + compatible: + enum: + - ti,tas2562 + - ti,tas2563 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + shut-down-gpio: + description: GPIO used to control the state of the device. + deprecated: true + + shutdown-gpio: + description: GPIO used to control the state of the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2562"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + shutdown-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + }; + }; + -- 2.26.2

2 5

[PATCH 1/4] ASoC: tas2562: Add right and left channel slot programming
by Dan Murphy 23 Jul '20

23 Jul '20

Add programming for the tdm slots for the right and left. This also requires configuring the RX/TX offsets for the DAI format type. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 71 +++++++++++++++++++++++++++++--------- sound/soc/codecs/tas2562.h | 5 ++- 2 files changed, 59 insertions(+), 17 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d26e30a2948c..1d3c381aeefe 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -175,7 +175,37 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, { struct snd_soc_component *component = dai->component; struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); - int ret = 0; + int left_slot, right_slot; + int slots_cfg; + int ret; + + if (!tx_mask) { + dev_err(component->dev, "tx masks must not be 0\n"); + return -EINVAL; + } + + if (slots == 1) { + if (tx_mask != 1) + return -EINVAL; + + left_slot = 0; + right_slot = 0; + } else { + left_slot = __ffs(tx_mask); + tx_mask &= ~(1 << left_slot); + if (tx_mask == 0) { + right_slot = left_slot; + } else { + right_slot = __ffs(tx_mask); + tx_mask &= ~(1 << right_slot); + } + } + + slots_cfg = (right_slot << TAS2562_RIGHT_SLOT_SHIFT) | left_slot; + + ret = snd_soc_component_write(component, TAS2562_TDM_CFG3, slots_cfg); + if (ret < 0) + return ret; switch (slot_width) { case 16: @@ -208,6 +238,18 @@ static int tas2562_set_dai_tdm_slot(struct snd_soc_dai *dai, if (ret < 0) return ret; + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG5, + TAS2562_TDM_CFG5_VSNS_SLOT_MASK, + tas2562->v_sense_slot); + if (ret < 0) + return ret; + + ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG6, + TAS2562_TDM_CFG6_ISNS_SLOT_MASK, + tas2562->i_sense_slot); + if (ret < 0) + return ret; + return 0; } @@ -285,7 +327,8 @@ 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; + u8 asi_cfg_1 = 0; + u8 tdm_rx_start_slot = 0; int ret; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { @@ -307,27 +350,23 @@ static int tas2562_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) 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): + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_DSP_B: tdm_rx_start_slot = 0; break; - default: - dev_err(tas2562->dev, "DAI Format is not found, fmt=0x%x\n", - fmt); - ret = -EINVAL; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + tdm_rx_start_slot = 1; break; + default: + dev_err(tas2562->dev, + "DAI Format is not found, fmt=0x%x\n", fmt); + return -EINVAL; } ret = snd_soc_component_update_bits(component, TAS2562_TDM_CFG1, - TAS2562_TDM_CFG1_RX_OFFSET_MASK, - tdm_rx_start_slot); - + TAS2562_RX_OFF_MASK, (tdm_rx_start_slot << 1)); if (ret < 0) return ret; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 28e75fc431d0..18209f397921 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -34,6 +34,10 @@ #define TAS2562_TDM_DET TAS2562_REG(0, 0x11) #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) +#define TAS2562_RX_OFF_MASK GENMASK(5, 1) +#define TAS2562_TX_OFF_MASK GENMASK(3, 1) +#define TAS2562_RIGHT_SLOT_SHIFT 4 + /* Page 2 */ #define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) #define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) @@ -49,7 +53,6 @@ #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) -- 2.26.2

3 7

[PATCH] dt-bindings: sound: sgtl5000: Convert to json-schema
by Thierry Reding 22 Jul '20

22 Jul '20

From: Thierry Reding <treding(a)nvidia.com> Convert the Freescale SGTL5000 device tree bindings from free-form text format to json-schema. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- .../devicetree/bindings/sound/sgtl5000.txt | 60 ---------- .../devicetree/bindings/sound/sgtl5000.yaml | 103 ++++++++++++++++++ .../devicetree/bindings/trivial-devices.yaml | 2 - 3 files changed, 103 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.txt create mode 100644 Documentation/devicetree/bindings/sound/sgtl5000.yaml diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt deleted file mode 100644 index 9d9ff5184939..000000000000 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ /dev/null @@ -1,60 +0,0 @@ -* Freescale SGTL5000 Stereo Codec - -Required properties: -- compatible : "fsl,sgtl5000". - -- reg : the I2C address of the device - -- #sound-dai-cells: must be equal to 0 - -- clocks : the clock provider of SYS_MCLK - -- VDDA-supply : the regulator provider of VDDA - -- VDDIO-supply: the regulator provider of VDDIO - -Optional properties: - -- VDDD-supply : the regulator provider of VDDD - -- micbias-resistor-k-ohms : the bias resistor to be used in kOhms - The resistor can take values of 2k, 4k or 8k. - If set to 0 it will be off. - If this node is not mentioned or if the value is unknown, then - micbias resistor is set to 4K. - -- micbias-voltage-m-volts : the bias voltage to be used in mVolts - The voltage can take values from 1.25V to 3V by 250mV steps - If this node is not mentioned or the value is unknown, then - the value is set to 1.25V. - -- lrclk-strength: the LRCLK pad strength. Possible values are: -0, 1, 2 and 3 as per the table below: - -VDDIO 1.8V 2.5V 3.3V -0 = Disable -1 = 1.66 mA 2.87 mA 4.02 mA -2 = 3.33 mA 5.74 mA 8.03 mA -3 = 4.99 mA 8.61 mA 12.05 mA - -- sclk-strength: the SCLK pad strength. Possible values are: -0, 1, 2 and 3 as per the table below: - -VDDIO 1.8V 2.5V 3.3V -0 = Disable -1 = 1.66 mA 2.87 mA 4.02 mA -2 = 3.33 mA 5.74 mA 8.03 mA -3 = 4.99 mA 8.61 mA 12.05 mA - -Example: - -sgtl5000: codec@a { - compatible = "fsl,sgtl5000"; - reg = <0x0a>; - #sound-dai-cells = <0>; - clocks = <&clks 150>; - micbias-resistor-k-ohms = <2>; - micbias-voltage-m-volts = <2250>; - VDDA-supply = <&reg_3p3v>; - VDDIO-supply = <&reg_3p3v>; -}; diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.yaml b/Documentation/devicetree/bindings/sound/sgtl5000.yaml new file mode 100644 index 000000000000..4f29b63c54d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sgtl5000.yaml @@ -0,0 +1,103 @@ +# SPDX-License-Identifier: GPL-2.0-only +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/sgtl5000.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale SGTL5000 Stereo Codec + +maintainers: + - Fabio Estevam <festevam(a)gmail.com> + +properties: + compatible: + const: fsl,sgtl5000 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + items: + - description: the clock provider of SYS_MCLK + + VDDA-supply: + description: the regulator provider of VDDA + + VDDIO-supply: + description: the regulator provider of VDDIO + + VDDD-supply: + description: the regulator provider of VDDD + + micbias-resistor-k-ohms: + description: The bias resistor to be used in kOhms. The resistor can take + values of 2k, 4k or 8k. If set to 0 it will be off. If this node is not + mentioned or if the value is unknown, then micbias resistor is set to + 4k. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 2, 4, 8 ] + + micbias-voltage-m-volts: + description: The bias voltage to be used in mVolts. The voltage can take + values from 1.25V to 3V by 250mV steps. If this node is not mentioned + or the value is unknown, then the value is set to 1.25V. + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 1250, 1500, 1750, 2000, 2250, 2500, 2750, 3000 ] + + lrclk-strength: + description: | + The LRCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + + sclk-strength: + description: | + The SCLK pad strength. Possible values are: 0, 1, 2 and 3 as per the + table below: + + VDDIO 1.8V 2.5V 3.3V + 0 = Disable + 1 = 1.66 mA 2.87 mA 4.02 mA + 2 = 3.33 mA 5.74 mA 8.03 mA + 3 = 4.99 mA 8.61 mA 12.05 mA + $ref: "/schemas/types.yaml#/definitions/uint32" + enum: [ 0, 1, 2, 3 ] + +required: + - compatible + - reg + - "#sound-dai-cells" + - clocks + - VDDA-supply + - VDDIO-supply + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; + #sound-dai-cells = <0>; + clocks = <&clks 150>; + micbias-resistor-k-ohms = <2>; + micbias-voltage-m-volts = <2250>; + VDDA-supply = <&reg_3p3v>; + VDDIO-supply = <&reg_3p3v>; + }; + }; +... diff --git a/Documentation/devicetree/bindings/trivial-devices.yaml b/Documentation/devicetree/bindings/trivial-devices.yaml index 4165352a590a..b7e94fe8643f 100644 --- a/Documentation/devicetree/bindings/trivial-devices.yaml +++ b/Documentation/devicetree/bindings/trivial-devices.yaml @@ -80,8 +80,6 @@ properties: - fsl,mpl3115 # MPR121: Proximity Capacitive Touch Sensor Controller - fsl,mpr121 - # SGTL5000: Ultra Low-Power Audio Codec - - fsl,sgtl5000 # G751: Digital Temperature Sensor and Thermal Watchdog with Two-Wire Interface - gmt,g751 # Infineon IR38064 Voltage Regulator -- 2.24.1

2 1

[PATCH 1/2] dt-bindings: tas2770: Convert tas2770 binding to yaml
by Dan Murphy 22 Jul '20

22 Jul '20

Convert the tas2770 binding to yaml format. Add in the reset-gpio to the binding as it is in the code but not documented in the binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- .../devicetree/bindings/sound/tas2770.txt | 37 --------- .../devicetree/bindings/sound/tas2770.yaml | 76 +++++++++++++++++++ 2 files changed, 76 insertions(+), 37 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tas2770.txt create mode 100644 Documentation/devicetree/bindings/sound/tas2770.yaml diff --git a/Documentation/devicetree/bindings/sound/tas2770.txt b/Documentation/devicetree/bindings/sound/tas2770.txt deleted file mode 100644 index ede6bb3d9637..000000000000 --- a/Documentation/devicetree/bindings/sound/tas2770.txt +++ /dev/null @@ -1,37 +0,0 @@ -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>; - }; - diff --git a/Documentation/devicetree/bindings/sound/tas2770.yaml b/Documentation/devicetree/bindings/sound/tas2770.yaml new file mode 100644 index 000000000000..8c667fd37a57 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2770.yaml @@ -0,0 +1,76 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019-20 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: "http://devicetree.org/schemas/sound/tas2770.yaml#" +$schema: "http://devicetree.org/meta-schemas/core.yaml#" + +title: Texas Instruments TAS2770 Smart PA + +maintainers: + - Shi Fu <shifu0704(a)thundersoft.com> + +description: | + 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. + +properties: + compatible: + enum: + - ti,tas2770 + + reg: + maxItems: 1 + description: | + I2C address of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpio: + description: GPIO used to reset the device. + + interrupts: + maxItems: 1 + + ti,imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX current sense time slot. + + ti,vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + description: TDM TX voltage sense time slot. + + ti,asi-format: + $ref: /schemas/types.yaml#/definitions/uint32 + description: Sets TDM RX capture edge. + enum: + - 0 # Rising edge + - 1 # Falling edge + + '#sound-dai-cells': + const: 1 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tas2770"; + reg = <0x4c>; + #sound-dai-cells = <1>; + interrupt-parent = <&gpio1>; + interrupts = <14>; + reset-gpio = <&gpio1 15 0>; + ti,imon-slot-no = <0>; + ti,vmon-slot-no = <2>; + }; + }; + -- 2.26.2

3 3

[PATCH 0/4] ASoC: Fix dailink checks for DPCM
by Pierre-Louis Bossart 17 Jul '20

17 Jul '20

We've had a couple of changes that introduce regressions with the multi-cpu DAI solutions, and while trying to fix them we found additional inconsistencies that should also go to stable branches. Bard Liao (1): ASoC: core: only convert non DPCM link to DPCM link Pierre-Louis Bossart (3): ASoC: soc-pcm: dpcm: fix playback/capture checks ASoC: Intel: boards: replace capture_only by dpcm_capture ASoC: SOF: nocodec: conditionally set dpcm_capture/dpcm_playback flags sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 4 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/soc-core.c | 22 ++++++++-- sound/soc/soc-pcm.c | 44 ++++++++++++++----- sound/soc/sof/nocodec.c | 6 ++- 7 files changed, 62 insertions(+), 20 deletions(-) base-commit: 8a9144c1cf523221b37dd3393827253c91fcbf54 -- 2.20.1

4 25

[PATCH v4 00/23] Add support for Tegra210 Audio
by Sameer Pujar 17 Jul '20

17 Jul '20

Overview ======== Audio Processing Engine (APE) comprises of Audio DMA (ADMA) and Audio Hub (AHUB) unit. AHUB is a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. * Add YAML DT binding documentation for above mentioned modules. * Helper function for ACIF programming is exposed for Tegra210 and later. * Add ASoC driver components for each of the above modules. * Add DT entries for above components for Tegra210, Tegra186 and Tegra194. * Enable these components for Jetson Nano/TX1/TX2/Xavier * Enhance simple-card DPCM driver to suit Tegra Audio applications with few changes in core. * To begin with, enable sound card support for Tegra210 based platforms like Jetson Nano/TX1. Sound card support for platforms based on Tegra186 and later will be subsequently added which can re-use all of the above components. Changelog ========= v3 -> v4 -------- * [1/23] "ASoC: dt-bindings: tegra: Add DT bindings for Tegra210" - Removed multiple examples and retained one example per doc - Fixed as per inputs on the previous series - Tested bindings with 'make dt_binding_check/dtbs_check' * [2/23] "ASoC: tegra: Add support for CIF programming" - No change * Common changes (for patch [3/10] to [7/10]) - Mixer control overrides, for PCM parameters (rate, channel, bits), in each driver are dropped. - Updated routing as per DPCM usage - Minor changes related to formatting * New changes (patch [8/23] to [18/23] and patch [23/23]) - Based on discussions in following threads DPCM is used for Tegra Audio. https://lkml.org/lkml/2020/2/20/91 https://lkml.org/lkml/2020/4/30/519 - The simple-card driver is used for Tegra Audio and accordingly some enhancements are made in simple-card and core drivers. - Patch [8/23] to [18/23] are related to simple-card and core changes. - Patch [23/23] adds sound card support to realize complete audio path. This is based on simple-card driver with proposed enhancements. - Re-ordered patches depending on above v2 -> v3 -------- * [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB - Updated licence - Removed redundancy w.r.t items/const/enum - Added constraints wherever needed with "pattern" property * [2/10] "ASoC: tegra: add support for CIF programming" - Removed tegra_cif.c - Instead added inline helper function in tegra_cif.h * common changes (for patch [3/10] to [7/10]) - Replace LATE system calls with Normal sleep - Remove explicit RPM suspend in driver remove() call - Use devm_kzalloc() instead of devm_kcalloc() for single element - Replace 'ret' with 'err' for better reading - Consistent error printing style across drivers - Minor formating fixes * [8/10] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later" (New patch) - Enables ACONNECT and AHUB components. With this AHUB and components are registered with ASoC core. v1 -> v2 -------- * [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB" - no changes * [2/9] "ASoC: tegra: add support for CIF programming" - removed CIF programming changes for legacy chips. - this patch now exposes helper function for CIF programming, which can be used on Tegra210 later. - later tegra_cif.c can be extended for legacy chips as well. - updated commit message accordingly * [3/9] "ASoC: tegra: add Tegra210 based DMIC driver" - removed unnecessary initialization of 'ret' in probe() * [4/9] "ASoC: tegra: add Tegra210 based I2S driver" - removed unnecessary initialization of 'ret' in probe() - fixed indentation - added consistent bracing for if-else clauses - updated 'rx_fifo_th' type to 'unsigned int' - used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h * [5/9] "ASoC: tegra: add Tegra210 based AHUB driver" - used of_device_get_match_data() to get 'soc_data' and removed explicit of_match_device() - used devm_platform_ioremap_resource() and removed explicit platform_get_resource() - fixed indentation for devm_snd_soc_register_component() - updated commit message - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [6/9] "ASoC: tegra: add Tegra186 based DSPK driver" - removed unnecessary initialization of 'ret' in probe() - updated 'max_th' to 'unsigned int' - shortened lengthy macro names to avoid wrapping in tegra186_dspk_wr_reg() and to be consistent * [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver" - used of_device_get_match_data() and removed explicit of_match_device() - used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [8/9] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * common changes for patch [3/9] to [7/9] - sorted headers in alphabetical order - moved MODULE_DEVICE_TABLE() right below *_of_match table - removed macro DRV_NAME - removed explicit 'owner' field from platform_driver structure - added 'const' to snd_soc_dai_ops structure ================== Sameer Pujar (23): ASoC: dt-bindings: tegra: Add DT bindings for Tegra210 ASoC: tegra: Add support for CIF programming ASoC: tegra: Add Tegra210 based DMIC driver ASoC: tegra: Add Tegra210 based I2S driver ASoC: tegra: Add Tegra210 based AHUB driver ASoC: tegra: Add Tegra186 based DSPK driver ASoC: tegra: Add Tegra210 based ADMAIF driver ASoC: soc-core: Fix component name_prefix parsing ASoC: simple-card: Use of_node and DAI names for DAI link names ASoC: simple-card: Wrong daifmt for CPU end of DPCM DAI link ASoC: simple-card: Loop over all children for 'mclk-fs' ASoC: simple-card: Support DPCM DAI link with multiple Codecs ASoC: simple-card: DPCM DAI link direction as per DAI capability ASoC: soc-core: Probe auxiliary component before others ASoC: soc-core: Identify 'no_pcm' DAI links for DPCM ASoC: soc-pcm: Get all BEs along DAPM path ASoC: dt-bindings: simple-card: Add compatible for component chaining ASoC: simple-card: Add support for component chaining arm64: defconfig: Build AHUB component drivers arm64: defconfig: Enable CONFIG_TEGRA210_ADMA arm64: tegra: Add DT binding for AHUB components arm64: tegra: Enable AHUB components on few Tegra platforms arm64: tegra: Add support for APE sound card on Jetson Nano and TX1 .../bindings/sound/nvidia,tegra186-dspk.yaml | 88 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 116 +++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 144 ++++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 88 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 106 +++ .../devicetree/bindings/sound/simple-card.yaml | 1 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 +++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 136 +++- arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 85 +++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 219 +++++- arch/arm64/configs/defconfig | 8 + include/sound/simple_card.h | 5 + include/sound/soc.h | 3 + sound/soc/generic/simple-card.c | 96 ++- sound/soc/soc-core.c | 43 +- sound/soc/soc-dai.c | 1 + sound/soc/soc-pcm.c | 3 +- sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra186_dspk.c | 427 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 70 ++ sound/soc/tegra/tegra210_admaif.c | 845 +++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 162 ++++ sound/soc/tegra/tegra210_ahub.c | 578 ++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 100 +++ sound/soc/tegra/tegra210_dmic.c | 440 +++++++++++ sound/soc/tegra/tegra210_dmic.h | 82 ++ sound/soc/tegra/tegra210_i2s.c | 780 +++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 126 +++ sound/soc/tegra/tegra_cif.h | 65 ++ sound/soc/tegra/tegra_pcm.c | 235 +++++- sound/soc/tegra/tegra_pcm.h | 21 +- 35 files changed, 5657 insertions(+), 36 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

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