August 2021 - Alsa-devel - mailman.alsa-project.org

[bug report] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Dan Carpenter 25 Aug '21

25 Aug '21

Hello Kuninori Morimoto, The patch cc64c390b215: "ASoC: rsnd: adg: clearly handle clock error / NULL case" from Aug 20, 2021, leads to the following Smatch static checker warning: sound/soc/sh/rcar/adg.c:396 rsnd_adg_create_null_clk() warn: passing zero to 'ERR_CAST' sound/soc/sh/rcar/adg.c 386 static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, 387 const char * const name, 388 const char *parent) 389 { 390 struct device *dev = rsnd_priv_to_dev(priv); 391 struct clk *clk; 392 393 clk = clk_register_fixed_rate(dev, name, parent, 0, 0); 394 if (IS_ERR_OR_NULL(clk)) { 395 dev_err(dev, "create null clk error\n"); The clk_register_fixed_rate() function doesn't have any documentation, but generally when functions return a mix of error pointers and NULL then the NULL case means the feature is deliberately disabled. So it should not print an error message. Perhaps an info message at most. --> 396 return ERR_CAST(clk); No need to cast this. The types are the same. 397 } 398 399 return clk; 400 } regards, dan carpenter

2 1

[PATCH v2 0/4] Rockchip I2S/TDM controller
by Nicolas Frattaroli 24 Aug '21

24 Aug '21

Changes in v2: - remove ad-hoc writeq and needless (and broken) optimisation in reset assert/deassert. This wouldn't have worked on Big Endian, and would've been pointless on any other platform, as the overhead for saving one write was comparatively big - fix various checkpatch issues - get rid of leftover clk-trcm in schema - set status = "okay" in example in schema instead of "disabled" - change dma-names so rx is first, adjust device trees as necessary - properly reference uint32-array for rx-route and tx-route instead of uint32 - replace trcm-sync with two boolean properties, adjust DT changes accordingly and also get rid of the header file - get rid of rockchip,no-dmaengine. This was only needed for some downstream driver and shouldn't be in the DT - get rid of rockchip,capture-only/playback-only. Rationale being that I have no way to test whether they're needed, and unconditionally setting channels_min to 0 breaks everything - change hclk description in "clocks" Additionally, I have been made aware that registers and code seem to be an extension of the rockchip_i2s driver. This is true, though merging the two seems like more effort than just slapping a few compatibles and properties into the mix. I'll investigate it further in the coming weeks, but would likely still prefer basing any unified driver on mine rather than the other way around, since I think it has better code quality now. Original cover letter: Hello, I come bearing four patches for your consideration. The first of these four patches adds a driver for the Rockchip I2S/TDM controller, used in interfacing between the AHB bus and the I2S bus on some Rockchip SoCs. This allows for audio playback with a matching codec. The controller has three different modes: I2S, I2S/TDM and PCM. It is distinct from the earlier Rockchip I2S controller, and therefore not just an extension of that driver. The driver is based on the downstream version, though various changes have been made to hopefully make it more palatable to upstream. Some needless code duplication has been refactored, and the probe function will no longer let wrong device tree values write nonsense to hardware registers. Properties have been renamed and had their semantics changed. I won't bore you with the details of what downstream did, since that's not what I'm submitting, but the changes are significant enough that I've added myself to the list of authors. The second patch adds the YAML device tree bindings for this, which have been written from scratch by yours truly. Since I didn't like having random integers mean things, I defined them as constants in a header file for the bindings. The third patch adds the i2s1 controller to the rk356x device tree. I didn't add any of the other i2s controllers on that SoC for now as I have no way of testing them; in particular, i2s0 is tied to HDMI, so needs a functioning VOP2 driver to even have a chance of working. The fourth patch makes use of the i2s1 controller to enable analog audio output on the Quartz64 Model A through its RK817 codec. I've tested this to work properly at both 44.1 kHz and 96 kHz, so both mclk_root0 and mclk_root1 are definitely functioning. This is my first kernel contribution, so I most likely did something horribly wrong. That's why I'm more than happy to receive any criticisms and concerns over how the driver is implemented, because I've run out of ideas on how to make it clearly better myself. I'd also like to extend my thanks to Peter Geis, who has been acting as somewhat of a mentor and gave me occasional feedback and ideas during the writing of this patch series. Regards, Nicolas Frattaroli Nicolas Frattaroli (4): ASoC: rockchip: add support for i2s-tdm controller dt-bindings: sound: add rockchip i2s-tdm binding arm64: dts: rockchip: add i2s1 on rk356x arm64: dts: rockchip: add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 193 ++ .../boot/dts/rockchip/rk3566-quartz64-a.dts | 35 +- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 26 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1737 +++++++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 398 ++++ 7 files changed, 2401 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h -- 2.32.0

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[PATCH] [v2] ASoC: Intel: sof_rt5682: Add support for max98390 speaker amp
by Mark Hsieh 24 Aug '21

24 Aug '21

Configure adl_max98390_rt5682 to support the rt5682 headset codec and max98390 speaker BUG=b:191811888 TEST=emerge-brya chromeos-kernel-5_10 Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 59 +++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 12 ++++ sound/soc/intel/boards/sof_rt5682.c | 20 ++++++- .../intel/common/soc-acpi-intel-adl-match.c | 13 ++++ 5 files changed, 104 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 7e29b0d911e2..0017c08c5a62 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -470,6 +470,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1015 select SND_SOC_RT1015P select SND_SOC_RT5682_I2C + select SND_SOC_MAX98390 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI select SND_SOC_INTEL_HDA_DSP_COMMON diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..477883d571ab 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -133,6 +133,65 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { /* For Right */ + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { /* For Left */ + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 2, 16); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 2, 16); + } + } + return 0; +} + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..2458758866f3 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,18 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..dc4966056b7d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,7 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -781,6 +782,13 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98373_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); } @@ -917,7 +925,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); - + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682); if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1043,6 +1052,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.17.1

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alsactl: flush stdout
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-utils pull request #109 was opened from Hello71: Signed-off-by: Alex Xu (Hello71) <alex_y_xu(a)yahoo.ca> Request URL : https://github.com/alsa-project/alsa-utils/pull/109 Patch URL : https://github.com/alsa-project/alsa-utils/pull/109.patch Repository URL: https://github.com/alsa-project/alsa-utils

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Pinebook Pro's audio does not work
by GitHub issues - edited 24 Aug '21

24 Aug '21

alsa-project/alsa-ucm-conf issue #110 was edited from starnight: I have a [Pinebook Pro](https://www.pine64.org/pinebook-pro/) whose audio (including the internal speaker, headphone and internal MIC) does not work by default with general Linux. Here is the [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7037916/… I notice Manjaro ARM has [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… in [pinebookpro-post-install](https://gitlab.manjaro.org/manjaro-arm/packages/c… package. If I install the [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… into `/var/lib/alsa/`, then `alsactl restore`, all of the audio works! However, I guess that **asound.state** is produced as a result, not the original configuration. If this relates to alsa-ucm-conf project, how should I configure the ucm for Pinebook Pro's audio? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/110 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Pinebook Pro's audio does not work
by GitHub issues - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-ucm-conf issue #110 was opened from starnight: I have a Pinebook Pro whose audio (including the internal speaker, headphone and internal MIC) does not work by default with general Linux. Here is the [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/7037916/… I notice Manjaro ARM has [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… in [pinebookpro-post-install](https://gitlab.manjaro.org/manjaro-arm/packages/c… package. If I install the [asound.state](https://gitlab.manjaro.org/manjaro-arm/packages/community/pin… into `/var/lib/alsa/`, then `alsactl restore`, all of the audio works! However, I guess that **asound.state** is produced as a result, not the original configuration. If this relates to alsa-ucm-conf project, how should I configure the ucm for Pinebook Pro's audio? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/110 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Timestamp
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-utils pull request #108 was opened from folkertvanheusden: Added -T / --timestamp (with parameter "realtime", "monotonic" or "raw") which adds a timestamp before each received message when using --dump. Request URL : https://github.com/alsa-project/alsa-utils/pull/108 Patch URL : https://github.com/alsa-project/alsa-utils/pull/108.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

arecord is failing with -V stereo
by sujith kumar reddy 24 Aug '21

24 Aug '21

Hi All, arecord is failing vumeter option -V stereo only. localhost ~ # arecord -Dhw:1,2 -r48000 -c2 -fS32_LE /tmp/test_record.wav -M -d 1 -V stereo Recording WAVE '/tmp/test_record.wav' : Signed 32 bit Little Endian, Rate 48000 Hz, Stereo *** buffer overflow detected ***: terminated Aborted by signal Aborted... Please provide pointers to debug this option. Thanks Sujith

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[PATCH alsa-utils 0/5] aplay: Fix bugs in peak calculations
by Takashi Iwai 24 Aug '21

24 Aug '21

The code in aplay for sample peak calculations and VU-meter representation seems to have a lot of bugs. Let's fix them. Takashi === Takashi Iwai (5): aplay: Fix conversion of unsigned samples in peak calculation aplay: Handle 16bit sample negative overflow in peak calculations aplay: Don't pass most negative integer to abs() in peak calculations aplay: Handle upper bound in peak calculations aplay: Fix out-of-bound access in stereo VU meter drawing aplay/aplay.c | 28 +++++++++++++++++++--------- 1 file changed, 19 insertions(+), 9 deletions(-) -- 2.26.2

1 5

rawmidi: hide the frame structure using a decoder function
by GitHub pull_request - opened 24 Aug '21

24 Aug '21

alsa-project/alsa-lib pull request #173 was opened from perexg: The frame structure should be hidden. Remove it and add snd_rawmidi_decode_frame0() which returns the timestamp and the midi chunk pointer to the application. BugLink: https://github.com/alsa-project/alsa-lib/issues/172 Request URL : https://github.com/alsa-project/alsa-lib/pull/173 Patch URL : https://github.com/alsa-project/alsa-lib/pull/173.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0