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

[PATCH 00/10] ASoC: SOF: enable multicore with dynamic pipelines
by Kai Vehmanen 23 Nov '21

23 Nov '21

Hi, initial support for SOF dynamic pipelines was added in commit 5fcdbb2d45df ("ASoC: SOF: Add support for dynamic pipelines"). When a pipeline is marked dynamic in the SOF DSP firmware topology definition (the tplg file kernel loads from filesystem), it means the pipeline resources are not allocated when DSP is booted (at driver probe, or at runtime resume), but rather delayed until the pipeline is actually used. Until now, it has not been possible to use multiple DSP cores in a topology that uses dynamic pipeline capability. This patchset addresses this gap and allows to freely mix dynamic pipelines and multicore usages in topology definitions. The series was originally reviewed in https://github.com/thesofproject/linux/pull/3019 Ranjani Sridharan (10): ASoC: SOF: Intel: hda: expose get_chip_info() ASoC: SOF: Introduce num_cores and ref count per core ASoC: SOF: Add ops for core_get and core_put ASoC: SOF: Intel: TGL: set core_get/put ops ASoC: SOF: Intel: CNL/ICL/APL: set core_get/core_put ops ASoC: SOF: topology: remove sof_load_pipeline_ipc() ASoC: SOF: free widgets in sof_tear_down_pipelines() for static pipelines ASoC: SOF: hda: don't use the core op for power up/power down ASoC: SOF: add support for dynamic pipelines with multi-core ASoC: SOF: Intel: hda: free DAI widget during stop and suspend sound/soc/sof/imx/imx8.c | 1 + sound/soc/sof/imx/imx8m.c | 1 + sound/soc/sof/intel/apl.c | 5 +- sound/soc/sof/intel/bdw.c | 9 +++ sound/soc/sof/intel/byt.c | 9 +++ sound/soc/sof/intel/cnl.c | 5 +- sound/soc/sof/intel/hda-dai.c | 23 +------ sound/soc/sof/intel/hda-dsp.c | 52 +++++++++++++++- sound/soc/sof/intel/hda-loader.c | 24 +++++--- sound/soc/sof/intel/hda.c | 17 ++---- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/icl.c | 5 +- sound/soc/sof/intel/pci-tng.c | 9 +++ sound/soc/sof/intel/shim.h | 7 +++ sound/soc/sof/intel/tgl.c | 46 +++++++++++++- sound/soc/sof/ops.h | 71 ++++++++++++++++------ sound/soc/sof/sof-audio.c | 83 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 6 -- sound/soc/sof/sof-priv.h | 21 +++++-- sound/soc/sof/topology.c | 100 ------------------------------- 20 files changed, 294 insertions(+), 201 deletions(-) base-commit: 626a3dfbdb5d11f92e709680135abf272057ef59 -- 2.33.0

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[PATCH] ASoC: cs42l42: Report initial jack state
by Richard Fitzgerald 23 Nov '21

23 Nov '21

When a jack handler is registered in cs42l42_set_jack() the initial state should be reported if an attached headphone/headset has already been detected. The jack detect sequence takes around 1 second: typically long enough for the machine driver to probe and register the jack handler in time to receive the first report from the interrupt handler. So it is possible on some systems that the correct initial state was seen simply because of lucky timing. Modular builds were more likely to miss the reporting of the initial state. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 4ca239f33737 ("ASoC: cs42l42: Always enable TS_PLUG and TS_UNPLUG interrupts") --- sound/soc/codecs/cs42l42.c | 22 ++++++++++++++++++++++ sound/soc/codecs/cs42l42.h | 2 ++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 0c4303547fd8..43d98bdb5b5b 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -549,8 +549,25 @@ static int cs42l42_set_jack(struct snd_soc_component *component, struct snd_soc_ { struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); + /* Prevent race with interrupt handler */ + mutex_lock(&cs42l42->jack_detect_mutex); cs42l42->jack = jk; + if (jk) { + switch (cs42l42->hs_type) { + case CS42L42_PLUG_CTIA: + case CS42L42_PLUG_OMTP: + snd_soc_jack_report(jk, SND_JACK_HEADSET, SND_JACK_HEADSET); + break; + case CS42L42_PLUG_HEADPHONE: + snd_soc_jack_report(jk, SND_JACK_HEADPHONE, SND_JACK_HEADPHONE); + break; + default: + break; + } + } + mutex_unlock(&cs42l42->jack_detect_mutex); + return 0; } @@ -1618,6 +1635,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) CS42L42_M_DETECT_FT_MASK | CS42L42_M_HSBIAS_HIZ_MASK); + mutex_lock(&cs42l42->jack_detect_mutex); + /* Check auto-detect status */ if ((~masks[5]) & irq_params_table[5].mask) { if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) { @@ -1686,6 +1705,8 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) } } + mutex_unlock(&cs42l42->jack_detect_mutex); + return IRQ_HANDLED; } @@ -2033,6 +2054,7 @@ static int cs42l42_i2c_probe(struct i2c_client *i2c_client, cs42l42->dev = &i2c_client->dev; i2c_set_clientdata(i2c_client, cs42l42); + mutex_init(&cs42l42->jack_detect_mutex); cs42l42->regmap = devm_regmap_init_i2c(i2c_client, &cs42l42_regmap); if (IS_ERR(cs42l42->regmap)) { diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 75ade987d0db..9fff183dce8e 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -12,6 +12,7 @@ #ifndef __CS42L42_H__ #define __CS42L42_H__ +#include <linux/mutex.h> #include <sound/jack.h> #define CS42L42_PAGE_REGISTER 0x00 /* Page Select Register */ @@ -841,6 +842,7 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; + struct mutex jack_detect_mutex; int pll_config; int bclk; u32 sclk; -- 2.11.0

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[PATCH v2 0/3] ASoC: stm32: add pm runtime support
by Olivier Moysan 23 Nov '21

23 Nov '21

Enable support of pm runtime on STM32 SPDIFRX, I2S and DFSDM drivers to allow power state monitoring. Changes in v2: - Move pm runtime enabling before component registration Olivier Moysan (3): ASoC: stm32: i2s: add pm_runtime support ASoC: stm32: dfsdm: add pm_runtime support for audio ASoC: stm32: spdifrx: add pm_runtime support sound/soc/stm/stm32_adfsdm.c | 5 ++++- sound/soc/stm/stm32_i2s.c | 4 ++++ sound/soc/stm/stm32_spdifrx.c | 4 ++++ 3 files changed, 12 insertions(+), 1 deletion(-) -- 2.17.1

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[PATCH v2 0/5] ASoC: SOF: Add PM support for i.MX8/i.MX8X/i.MX8M
by Daniel Baluta 23 Nov '21

23 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for System PM and Runtime PM with SOF for i.MX8 platforms. First patch adds common code to manage clocks, patch 2/3 adds the actual PM support, patch 4/5 implement the DSP start / reset callbacks for i.mx8m. Changes since v1: * rebased on top of Mark's for-next branch Daniel Baluta (5): ASoC: SOF: imx: Add code to manage DSP related clocks ASoC: SOF: imx8: Add runtime PM / System PM support ASoC: SOF: imx8m: Add runtime PM / System PM support ASoC: SOF: imx8m: Implement DSP start ASoC: SOF: imx8m: Implement reset callback sound/soc/sof/imx/imx-common.c | 24 +++++ sound/soc/sof/imx/imx-common.h | 11 ++ sound/soc/sof/imx/imx8.c | 137 +++++++++++++++++++++++ sound/soc/sof/imx/imx8m.c | 191 ++++++++++++++++++++++++++++++++- 4 files changed, 362 insertions(+), 1 deletion(-) -- 2.27.0

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[PATCH 0/5] ASoC: SOF: Add PM support for i.MX8/i.MX8X/i.MX8M
by Daniel Baluta 23 Nov '21

23 Nov '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patch series adds support for System PM and Runtime PM with SOF for i.MX8 platforms. First patch adds common code to manage clocks, patch 2/3 adds the actual PM support, patch 4/5 implement the DSP start / reset callbacks for i.mx8m. Daniel Baluta (5): ASoC: SOF: imx: Add code to manage DSP related clocks ASoC: SOF: imx8: Add runtime PM / System PM support ASoC: SOF: imx8m: Add runtime PM / System PM support ASoC: SOF: imx8m: Implement DSP start ASoC: SOF: imx8m: Implement reset callback sound/soc/sof/imx/imx-common.c | 24 +++++ sound/soc/sof/imx/imx-common.h | 11 ++ sound/soc/sof/imx/imx8.c | 137 +++++++++++++++++++++++ sound/soc/sof/imx/imx8m.c | 191 ++++++++++++++++++++++++++++++++- 4 files changed, 362 insertions(+), 1 deletion(-) -- 2.27.0

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[PATCH 5.16 regression fix] ASoC: soc-acpi: Set mach->id field on comp_ids matches
by Hans de Goede 23 Nov '21

23 Nov '21

Commit dac7cbd55dca ("ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs") and commit 959ae8215a9e ("ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs") simplified the match tables in soc-acpi-intel-byt-match.c and soc-acpi-intel-cht-match.c by merging identical entries using the new .comp_ids snd_soc_acpi_mach field to point a single entry to multiple ACPI HIDs and clearing the previously unique per entry .id field. But various machine drivers from sound/soc/intel/boards rely on mach->id in one or more ways, e.g. some drivers contain the following snippets: adev = acpi_dev_get_first_match_dev(mach->id, NULL, -1); pkg_found = snd_soc_acpi_find_package_from_hid(mach->id, ... if (!strncmp(snd_soc_cards[i].codec_id, mach->id, 8)) { ... All of which are broken by the match table shrinking. Make the snd_soc_acpi_mach.id field non const (the storage for the tables already is non const) and on a comps_ids match copy the matching HID to the id field to fix this. Fixes: dac7cbd55dca ("ASoC: Intel: soc-acpi-byt: shrink tables using compatible IDs") Fixes: 959ae8215a9e ("ASoC: Intel: soc-acpi-cht: shrink tables using compatible IDs") Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- include/sound/soc-acpi.h | 2 +- sound/soc/soc-acpi.c | 4 +++- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 31f4c4f9aeea..ac0893df9c76 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -147,7 +147,7 @@ struct snd_soc_acpi_link_adr { */ /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { - const u8 id[ACPI_ID_LEN]; + u8 id[ACPI_ID_LEN]; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 2ae99b49d3f5..cbd7ea48837b 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -20,8 +20,10 @@ static bool snd_soc_acpi_id_present(struct snd_soc_acpi_mach *machine) if (comp_ids) { for (i = 0; i < comp_ids->num_codecs; i++) { - if (acpi_dev_present(comp_ids->codecs[i], NULL, -1)) + if (acpi_dev_present(comp_ids->codecs[i], NULL, -1)) { + strscpy(machine->id, comp_ids->codecs[i], ACPI_ID_LEN); return true; + } } } -- 2.31.1

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Xonar U7 MKII Analog Multichannel and SPDIF
by GitHub issues - opened 21 Nov '21

21 Nov '21

alsa-project/alsa-lib issue #194 was opened from Gebbx: Hi, Usb sound card Xonar U7 MKII plays multichannel content without issue, but only through aplay `aplay -D hw:2,0 test.wav #6 channels file` in other cases only two channels work (downmixed) I have tried different combinations of settings. Editing asound.conf gives no results too. The digital output also does not work in any combination of settings, but works (only two channels) in the following case: `pcm_slave.u7spdif { pcm "hw:2,1" } pcm.u7spplug { type plug slave u7spdif hint { show on } }` If I understand correctly, in this case it output the decoded signal and it will always be two-channel. I found a topic: https://forum.kodi.tv/showthread.php?tid=253179 So, i have to edit USB-Audio.conf, ``` USB-Audio.pcm.iec958_device { # "NoiseBlaster 3000" 42 "USB Sound Blaster HD" 1 "Xonar U7" 1 "MKII" 1 # my card } ``` but this is not recommended. How can this be done in another way? Example given in the USB-Audio.conf file forces system to reboot. `cards.USB-Audio.pcm.iec958_device."MKII" 1` Thanks! My system: RPI4, LibreELEC 10 Issue URL : https://github.com/alsa-project/alsa-lib/issues/194 Repository URL: https://github.com/alsa-project/alsa-lib

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Where can I find alsa forum?
by GitHub issues - edited 21 Nov '21

21 Nov '21

alsa-project/alsa-lib issue #193 was edited from yangyunfei16: Dear alsa developers: Hello. I am a graduate student in the Institute of information engineering, Chinese Academy of Sciences. We are doing a project to investigate the use of this library by some programmers who use alsa for programming, so we want to ask if there are any websites or forums on which more programmers who use this library communicate and discuss, We want them to do a questionnaire later. Thank you very much for your help! Issue URL : https://github.com/alsa-project/alsa-lib/issues/193 Repository URL: https://github.com/alsa-project/alsa-lib

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project
by GitHub issues - opened 21 Nov '21

21 Nov '21

alsa-project/alsa-lib issue #193 was opened from yangyunfei16: Dear alsa developers: Hello. I am a graduate student in the Institute of information engineering, Chinese Academy of Sciences. We are doing a project to investigate the use of this library by some programmers who use alsa for programming, so we want to ask if there are any websites or forums on which more programmers who use this library communicate and discuss, We want them to do a questionnaire later. Thank you very much for your help! Issue URL : https://github.com/alsa-project/alsa-lib/issues/193 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/2] ASoC: wm8962: Conversion to json-schema and fix
by Geert Uytterhoeven 21 Nov '21

21 Nov '21

Hi all, This patch series converts the Wolfson WM8962 Device Tree binding documentation to json-schema, after fixing an issue in the imx8mn-beacon DTS file. Thanks for your comments! Geert Uytterhoeven (2): arm64: dts: imx: imx8mn-beacon: Drop undocumented clock-names reference ASoC: dt-bindings: wlf,wm8962: Convert to json-schema .../devicetree/bindings/sound/wlf,wm8962.yaml | 118 ++++++++++++++++++ .../devicetree/bindings/sound/wm8962.txt | 43 ------- .../freescale/imx8mn-beacon-baseboard.dtsi | 1 - 3 files changed, 118 insertions(+), 44 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8962.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8962.txt -- 2.25.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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