July 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH v1] ASoC: meson: axg-fifo: set option to use raw spinlock
by Arseniy Krasnov 06 Aug '24

06 Aug '24

Raw spinlock is needed here, because with enabled PREEMPT_RT, spinlock_t become preemptible, but this regmap lock could be acquired in IRQ handler. Found by lockdep: [ ] ============================= [ ] [ BUG: Invalid wait context ] [ ] 6.9.9-sdkernel #1 Tainted: G O [ ] ----------------------------- [ ] aplay/413 is trying to lock: [ ] ffff000003930018 (axg_fifo:356:(&axg_fifo_regmap_cfg)->lock){....}-{3:3},c [ ] other info that might help us debug this: [ ] context-{2:2} [ ] no locks held by aplay/413. [ ] stack backtrace: [ ] CPU: 0 PID: 413 Comm: aplay Tainted: G O 6.9.9-kernel #1 [ ] Hardware name: SberDevices SberBoom Mini (DT) [ ] Call trace: [ ] dump_backtrace+0x98/0xf0 [ ] show_stack+0x18/0x24 [ ] dump_stack_lvl+0x90/0xd0 [ ] dump_stack+0x18/0x24 [ ] __lock_acquire+0x9dc/0x1f10 [ ] lock_acquire.part.0+0xe8/0x228 [ ] lock_acquire+0x68/0x84 [ ] _raw_spin_lock_irqsave+0x60/0x88 [ ] regmap_lock_spinlock+0x18/0x2c [ ] regmap_read+0x3c/0x78 [ ] axg_fifo_pcm_irq_block+0x4c/0xc8 [ ] __handle_irq_event_percpu+0xa4/0x2f8 [ ] handle_irq_event+0x4c/0xbc [ ] handle_fasteoi_irq+0xa4/0x23c [ ] generic_handle_domain_irq+0x2c/0x44 [ ] gic_handle_irq+0x40/0xc4 [ ] call_on_irq_stack+0x24/0x4c [ ] do_interrupt_handler+0x80/0x84 [ ] el0_interrupt+0x5c/0x124 [ ] __el0_irq_handler_common+0x18/0x24 [ ] el0t_32_irq_handler+0x10/0x1c [ ] el0t_32_irq+0x194/0x198 Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-fifo.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/axg-fifo.c b/sound/soc/meson/axg-fifo.c index ecb3eb7a9723d..a22298f74b35a 100644 --- a/sound/soc/meson/axg-fifo.c +++ b/sound/soc/meson/axg-fifo.c @@ -328,6 +328,7 @@ static const struct regmap_config axg_fifo_regmap_cfg = { .val_bits = 32, .reg_stride = 4, .max_register = FIFO_CTRL2, + .use_raw_spinlock = true, }; int axg_fifo_probe(struct platform_device *pdev) -- 2.30.1

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[PATCH] ASoC: codecs: lpass-macro: fix missing codec version
by Johan Hovold 05 Aug '24

05 Aug '24

Recent changes that started checking the codec version broke audio on the Lenovo ThinkPad X13s: wsa_macro 3240000.codec: Unsupported Codec version (0) wsa_macro 3240000.codec: probe with driver wsa_macro failed with error -22 rx_macro 3200000.rxmacro: Unsupported Codec version (0) rx_macro 3200000.rxmacro: probe with driver rx_macro failed with error -22 Add the missing codec version to the lookup table so that the codec drivers probe successfully. Note that I'm just assuming that this is a 2.0 codec based on the fact that this device uses the older register layout. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Fixes: 727de4fbc546 ("ASoC: codecs: lpass-wsa-macro: Correct support for newer v2.5 version") Signed-off-by: Johan Hovold <johan+linaro(a)kernel.org> --- Cc: regressions(a)lists.linux.dev #regzbot introduced: 378918d59181 sound/soc/codecs/lpass-va-macro.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index b852cc7ffad9..a62ccd09bacd 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1472,6 +1472,8 @@ static void va_macro_set_lpass_codec_version(struct va_macro *va) if ((core_id_0 == 0x01) && (core_id_1 == 0x0F)) version = LPASS_CODEC_VERSION_2_0; + if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && core_id_2 == 0x01) + version = LPASS_CODEC_VERSION_2_0; if ((core_id_0 == 0x02) && (core_id_1 == 0x0E)) version = LPASS_CODEC_VERSION_2_1; if ((core_id_0 == 0x02) && (core_id_1 == 0x0F) && (core_id_2 == 0x50 || core_id_2 == 0x51)) -- 2.44.2

3 2

[PATCH] ASoC: cs43130: Constify snd_soc_component_driver struct
by Christophe JAILLET 02 Aug '24

02 Aug '24

In order to constify `snd_soc_component_driver` struct, duplicate `soc_component_dev_cs43130` into a `soc_component_dev_cs43130_digital` and `soc_component_dev_cs43130_analog`. These 2 new structures share the same .dapm_widgets and .dapm_routes arrays but differ for .num_dapm_widgets and .num_dapm_routes. In the digital case, the last entries are not taken into account. Doing so has several advantages: - `snd_soc_component_driver` can be declared as const to move their declarations to read-only sections. - code in the probe is simpler. There is no need to concatenate some arrays to handle the "analog" case - this saves some memory because all_hp_widgets and analog_hp_routes can be removed. Before : ====== text data bss dec hex filename 53965 8265 4512 66742 104b6 sound/soc/codecs/cs43130.o After : ===== text data bss dec hex filename 54409 7881 64 62354 f392 sound/soc/codecs/cs43130.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested-only. See discussion at [1]. [1]: https://lore.kernel.org/all/ZqNawRmAqBRLIoQq@opensource.cirrus.com/ --- sound/soc/codecs/cs43130.c | 73 +++++++++++++++++--------------------- 1 file changed, 32 insertions(+), 41 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index be4037890fdb..cb4ca80f36d2 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -1415,7 +1415,7 @@ static const char * const bypass_mux_text[] = { static SOC_ENUM_SINGLE_DECL(bypass_enum, SND_SOC_NOPM, 0, bypass_mux_text); static const struct snd_kcontrol_new bypass_ctrl = SOC_DAPM_ENUM("Switch", bypass_enum); -static const struct snd_soc_dapm_widget digital_hp_widgets[] = { +static const struct snd_soc_dapm_widget hp_widgets[] = { SND_SOC_DAPM_MUX("Bypass Switch", SND_SOC_NOPM, 0, 0, &bypass_ctrl), SND_SOC_DAPM_OUTPUT("HPOUTA"), SND_SOC_DAPM_OUTPUT("HPOUTB"), @@ -1447,19 +1447,16 @@ static const struct snd_soc_dapm_widget digital_hp_widgets[] = { CS43130_PDN_HP_SHIFT, 1, cs43130_dac_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD)), -}; -static const struct snd_soc_dapm_widget analog_hp_widgets[] = { +/* Some devices have some extra analog widgets */ +#define NUM_ANALOG_WIDGETS 1 + SND_SOC_DAPM_DAC_E("Analog Playback", NULL, CS43130_HP_OUT_CTL_1, CS43130_HP_IN_EN_SHIFT, 0, cs43130_hpin_event, (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), }; -static struct snd_soc_dapm_widget all_hp_widgets[ - ARRAY_SIZE(digital_hp_widgets) + - ARRAY_SIZE(analog_hp_widgets)]; - -static const struct snd_soc_dapm_route digital_hp_routes[] = { +static const struct snd_soc_dapm_route hp_routes[] = { {"ASPIN PCM", NULL, "ASP PCM Playback"}, {"ASPIN DoP", NULL, "ASP DoP Playback"}, {"XSPIN DoP", NULL, "XSP DoP Playback"}, @@ -1472,15 +1469,12 @@ static const struct snd_soc_dapm_route digital_hp_routes[] = { {"Bypass Switch", "Internal", "HiFi DAC"}, {"HPOUTA", NULL, "Bypass Switch"}, {"HPOUTB", NULL, "Bypass Switch"}, -}; -static const struct snd_soc_dapm_route analog_hp_routes[] = { +/* Some devices have some extra analog routes */ +#define NUM_ANALOG_ROUTES 1 {"Bypass Switch", "Alternative", "Analog Playback"}, }; -static struct snd_soc_dapm_route all_hp_routes[ - ARRAY_SIZE(digital_hp_routes) + - ARRAY_SIZE(analog_hp_routes)]; static const unsigned int cs43130_asp_src_rates[] = { 32000, 44100, 48000, 88200, 96000, 176400, 192000, 352800, 384000 @@ -2398,7 +2392,23 @@ static int cs43130_probe(struct snd_soc_component *component) return 0; } -static struct snd_soc_component_driver soc_component_dev_cs43130 = { +static const struct snd_soc_component_driver soc_component_dev_cs43130_digital = { + .probe = cs43130_probe, + .controls = cs43130_snd_controls, + .num_controls = ARRAY_SIZE(cs43130_snd_controls), + .set_sysclk = cs43130_component_set_sysclk, + .set_pll = cs43130_set_pll, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + /* Don't take into account the ending analog widgets and routes */ + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets) - NUM_ANALOG_WIDGETS, + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes) - NUM_ANALOG_ROUTES, +}; + +static const struct snd_soc_component_driver soc_component_dev_cs43130_analog = { .probe = cs43130_probe, .controls = cs43130_snd_controls, .num_controls = ARRAY_SIZE(cs43130_snd_controls), @@ -2407,6 +2417,10 @@ static struct snd_soc_component_driver soc_component_dev_cs43130 = { .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, + .dapm_widgets = hp_widgets, + .num_dapm_widgets = ARRAY_SIZE(hp_widgets), + .dapm_routes = hp_routes, + .num_dapm_routes = ARRAY_SIZE(hp_routes), }; static const struct regmap_config cs43130_regmap = { @@ -2479,6 +2493,7 @@ static int cs43130_handle_device_data(struct cs43130_private *cs43130) static int cs43130_i2c_probe(struct i2c_client *client) { + const struct snd_soc_component_driver *component_driver; struct cs43130_private *cs43130; int ret; unsigned int reg; @@ -2596,39 +2611,15 @@ static int cs43130_i2c_probe(struct i2c_client *client) switch (cs43130->dev_id) { case CS43130_CHIP_ID: case CS43131_CHIP_ID: - memcpy(all_hp_widgets, digital_hp_widgets, - sizeof(digital_hp_widgets)); - memcpy(all_hp_widgets + ARRAY_SIZE(digital_hp_widgets), - analog_hp_widgets, sizeof(analog_hp_widgets)); - memcpy(all_hp_routes, digital_hp_routes, - sizeof(digital_hp_routes)); - memcpy(all_hp_routes + ARRAY_SIZE(digital_hp_routes), - analog_hp_routes, sizeof(analog_hp_routes)); - - soc_component_dev_cs43130.dapm_widgets = - all_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(all_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - all_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(all_hp_routes); + component_driver = &soc_component_dev_cs43130_analog; break; case CS43198_CHIP_ID: case CS4399_CHIP_ID: - soc_component_dev_cs43130.dapm_widgets = - digital_hp_widgets; - soc_component_dev_cs43130.num_dapm_widgets = - ARRAY_SIZE(digital_hp_widgets); - soc_component_dev_cs43130.dapm_routes = - digital_hp_routes; - soc_component_dev_cs43130.num_dapm_routes = - ARRAY_SIZE(digital_hp_routes); + component_driver = &soc_component_dev_cs43130_digital; break; } - ret = devm_snd_soc_register_component(cs43130->dev, - &soc_component_dev_cs43130, + ret = devm_snd_soc_register_component(cs43130->dev, component_driver, cs43130_dai, ARRAY_SIZE(cs43130_dai)); if (ret < 0) { dev_err(cs43130->dev, -- 2.45.2

2 1

[PATCH 00/12] ALSA/ASoC: use snd_pcm_direction_name()
by Kuninori Morimoto 02 Aug '24

02 Aug '24

Hi Tatasi, Mark Many drivers are selecting strings "playback" / "capture" by own handling, but we have snd_pcm_direction_name() function for it. This patch use it. One note is that snd_pcm_direction_name() will select "Playback" and "Capture", instead of "playback" / "capture". Almost all drivers are using it as dev_dbg() or dev_err() so no problem. But some other drivers are using it as other purpose. It might be issue (?). For example ASoC debugfs dir name will be changed by this patch. Kuninori Morimoto (12): ALSA: pci: pcxhr: use snd_pcm_direction_name() ALSA: pci: rme9652: use snd_pcm_direction_name() ALSA: trace: use snd_pcm_direction_name() ALSA: aloop: use snd_pcm_direction_name() ALSA: pcm_timer: use snd_pcm_direction_name() ASoC: stm: use snd_pcm_direction_name() ASoC: sof: pcm: use snd_pcm_direction_name() ASoC: sof: intel: use snd_pcm_direction_name() ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() ASoC: tegra: use snd_pcm_direction_name() ASoC: soc-pcm: use snd_pcm_direction_name() ASoC: soc-dapm: use snd_pcm_direction_name() include/trace/events/asoc.h | 3 ++- sound/core/pcm_timer.c | 3 +-- sound/drivers/aloop.c | 3 +-- sound/pci/pcxhr/pcxhr_mix22.c | 2 +- sound/pci/rme9652/hdspm.c | 6 ++---- sound/soc/fsl/lpc3xxx-i2s.c | 3 +-- sound/soc/soc-dapm.c | 3 +-- sound/soc/soc-pcm.c | 30 +++++++++++++++--------------- sound/soc/sof/intel/hda-stream.c | 4 +--- sound/soc/sof/pcm.c | 2 +- sound/soc/stm/stm32_i2s.c | 4 ++-- sound/soc/tegra/tegra210_i2s.c | 16 ++++++++-------- 12 files changed, 36 insertions(+), 43 deletions(-) -- 2.43.0

3 14

[PATCH] ASoC: SOF: Remove libraries from topology lookups
by cujomalainey＠chromium.org 01 Aug '24

01 Aug '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Wojciech Macek <wmacek(a)google.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 24ae1d4959be..1c6e035fd313 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static const struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- 2.46.0.rc2.264.g509ed76dc8-goog

3 2

[PATCH] slimbus: generate MODULE_ALIAS() from MODULE_DEVICE_TABLE()
by Masahiro Yamada 01 Aug '24

01 Aug '24

Commit 9e663f4811c6 ("slimbus: core: add support to uevent") added the MODALIAS=slim:* uevent variable, but modpost does not generate the corresponding MODULE_ALIAS(). To support automatic module loading, slimbus drivers still need to manually add MODULE_ALIAS("slim:<manf_id>:<prod_code>:*"), as seen in sound/soc/codecs/wcd9335.c. To automate this, make modpost generate the proper MODULE_ALIAS() from MODULE_DEVICE_TABLE(slim, ). Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- scripts/mod/devicetable-offsets.c | 4 ++++ scripts/mod/file2alias.c | 11 +++++++++++ 2 files changed, 15 insertions(+) diff --git a/scripts/mod/devicetable-offsets.c b/scripts/mod/devicetable-offsets.c index 518200813d4e..9c7b404defbd 100644 --- a/scripts/mod/devicetable-offsets.c +++ b/scripts/mod/devicetable-offsets.c @@ -153,6 +153,10 @@ int main(void) DEVID_FIELD(i3c_device_id, part_id); DEVID_FIELD(i3c_device_id, extra_info); + DEVID(slim_device_id); + DEVID_FIELD(slim_device_id, manf_id); + DEVID_FIELD(slim_device_id, prod_code); + DEVID(spi_device_id); DEVID_FIELD(spi_device_id, name); diff --git a/scripts/mod/file2alias.c b/scripts/mod/file2alias.c index 5d1c61fa5a55..99dce93a4188 100644 --- a/scripts/mod/file2alias.c +++ b/scripts/mod/file2alias.c @@ -960,6 +960,16 @@ static int do_i3c_entry(const char *filename, void *symval, return 1; } +static int do_slim_entry(const char *filename, void *symval, char *alias) +{ + DEF_FIELD(symval, slim_device_id, manf_id); + DEF_FIELD(symval, slim_device_id, prod_code); + + sprintf(alias, "slim:%x:%x:*", manf_id, prod_code); + + return 1; +} + /* Looks like: spi:S */ static int do_spi_entry(const char *filename, void *symval, char *alias) @@ -1555,6 +1565,7 @@ static const struct devtable devtable[] = { {"rpmsg", SIZE_rpmsg_device_id, do_rpmsg_entry}, {"i2c", SIZE_i2c_device_id, do_i2c_entry}, {"i3c", SIZE_i3c_device_id, do_i3c_entry}, + {"slim", SIZE_slim_device_id, do_slim_entry}, {"spi", SIZE_spi_device_id, do_spi_entry}, {"dmi", SIZE_dmi_system_id, do_dmi_entry}, {"platform", SIZE_platform_device_id, do_platform_entry}, -- 2.43.0

2 2

[PATCH 0/3] slimbus: use 'time_left' instead of 'timeout' with wait_for_*() functions
by Wolfram Sang 01 Aug '24

01 Aug '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_*() functions causing patterns like: timeout = wait_for_completion_timeout(...) if (!timeout) return -ETIMEDOUT; with all kinds of permutations. Use 'time_left' as a variable to make the code obvious and self explaining. This is part of a tree-wide series. The rest of the patches can be found here (some parts may still be WIP): git://git.kernel.org/pub/scm/linux/kernel/git/wsa/linux.git i2c/time_left Because these patches are generated, I audit them before sending. This is why I will send series step by step. Build bot is happy with these patches, though. No functional changes intended. Wolfram Sang (3): slimbus: messaging: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ctrl: use 'time_left' variable with wait_for_completion_timeout() slimbus: qcom-ngd-ctrl: use 'time_left' variable with wait_for_completion_timeout() drivers/slimbus/messaging.c | 9 +++++---- drivers/slimbus/qcom-ctrl.c | 7 ++++--- drivers/slimbus/qcom-ngd-ctrl.c | 29 ++++++++++++++++------------- 3 files changed, 25 insertions(+), 20 deletions(-) -- 2.43.0

3 6

[PATCH v2] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 31 Jul '24

31 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- Changes in v2: - v1 was accidentally the backport version for older kernels instead of the one for v6.11 sound/pci/hda/cs35l56_hda.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..f2f623f713d1 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -613,7 +617,7 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) */ if (cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); - else + else if (expose_dsp_controls) add_dsp_controls_required = true; cs35l56->base.fw_patched = false; -- 2.43.0

1 1

[PATCH][LINUX][ALSA][USB AUDIO CLASS 1] wrong channel ids for surround
by Sylvain BERTRAND 31 Jul '24

31 Jul '24

Hi, In the USB audio class 1 specifications, page 33 (3.7.2.3 Audio Channel Cluster Format): Surround is spatially defined as [rear], hence left surround should be SNDRV_CHMAP_RL and not SND_CHMAP_SL (Side). Same for right surround channel. Right? I am not a mailing list subscriber, CC me please. regards, -- Sylvain BERTRAND --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -244,8 +244,8 @@ static struct snd_pcm_chmap_elem *convert_chmap(int channels, unsigned int bits, SNDRV_CHMAP_FR, /* right front */ SNDRV_CHMAP_FC, /* center front */ SNDRV_CHMAP_LFE, /* LFE */ - SNDRV_CHMAP_SL, /* left surround */ - SNDRV_CHMAP_SR, /* right surround */ + SNDRV_CHMAP_RL, /* left surround */ + SNDRV_CHMAP_RR, /* right surround */ SNDRV_CHMAP_FLC, /* left of center */ SNDRV_CHMAP_FRC, /* right of center */ SNDRV_CHMAP_RC, /* surround */

2 3

[PATCH -next] ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate'
by Yue Haibing 31 Jul '24

31 Jul '24

The variable savedbitclkrate is assigned and never used, so can be removed. sound/soc/fsl/lpc3xxx-i2s.c:42:13: warning: variable ‘savedbitclkrate’ set but not used [-Wunused-but-set-variable] Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/soc/fsl/lpc3xxx-i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index af995ca081a3..62ef624d6dd4 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -39,7 +39,7 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 { u32 i2srate; u32 idxx, idyy; - u32 savedbitclkrate, diff, trate, baseclk; + u32 diff, trate, baseclk; /* Adjust rate for sample size (bits) and 2 channels and offset for * divider in clock output @@ -53,14 +53,12 @@ static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, int xbytes, u3 /* Find the best divider */ *clkx = *clky = 0; - savedbitclkrate = 0; diff = ~0; for (idxx = 1; idxx < 0xFF; idxx++) { for (idyy = 1; idyy < 0xFF; idyy++) { trate = (baseclk * idxx) / idyy; if (abs(trate - i2srate) < diff) { diff = abs(trate - i2srate); - savedbitclkrate = trate; *clkx = idxx; *clky = idyy; } -- 2.34.1

3 2

[PATCH] ASoC: rsnd: use pcm_dmaengine code
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd is implementing own DMAEngine code, but we can replace it with pcm_dmaengine code, because these are almost same. Let's use existing and stable code. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 17 --------- sound/soc/sh/rcar/dma.c | 75 ++++------------------------------------ sound/soc/sh/rcar/rsnd.h | 1 - sound/soc/sh/rcar/ssi.c | 2 +- 5 files changed, 9 insertions(+), 87 deletions(-) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 7bddfd5e38d6c..426632996a0a3 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -41,6 +41,7 @@ config SND_SOC_RCAR depends on COMMON_CLK depends on OF select SND_SIMPLE_CARD_UTILS + select SND_DMAENGINE_PCM select REGMAP_MMIO help This option enables R-Car SRU/SCU/SSIU/SSI sound support diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 6bc7027ed4dbf..233128c8ff145 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -660,23 +660,6 @@ static struct rsnd_dai *rsnd_dai_to_rdai(struct snd_soc_dai *dai) return rsnd_rdai_get(priv, dai->id); } -/* - * rsnd_soc_dai functions - */ -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io) -{ - struct snd_pcm_substream *substream = io->substream; - - /* - * this function should be called... - * - * - if rsnd_dai_pointer_update() returns true - * - without spin lock - */ - - snd_pcm_period_elapsed(substream); -} - static void rsnd_dai_stream_init(struct rsnd_dai_stream *io, struct snd_pcm_substream *substream) { diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7b499eee50806..2342bbb6fe92e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -7,6 +7,7 @@ #include <linux/delay.h> #include <linux/of_dma.h> +#include <sound/dmaengine_pcm.h> #include "rsnd.h" /* @@ -22,8 +23,6 @@ struct rsnd_dmaen { struct dma_chan *chan; - dma_cookie_t cookie; - unsigned int dma_len; }; struct rsnd_dmapp { @@ -66,20 +65,6 @@ static struct rsnd_mod mem = { /* * Audio DMAC */ -static void __rsnd_dmaen_complete(struct rsnd_mod *mod, - struct rsnd_dai_stream *io) -{ - if (rsnd_io_is_working(io)) - rsnd_dai_period_elapsed(io); -} - -static void rsnd_dmaen_complete(void *data) -{ - struct rsnd_mod *mod = data; - - rsnd_mod_interrupt(mod, __rsnd_dmaen_complete); -} - static struct dma_chan *rsnd_dmaen_request_channel(struct rsnd_dai_stream *io, struct rsnd_mod *mod_from, struct rsnd_mod *mod_to) @@ -98,13 +83,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dai_stream *io, struct rsnd_priv *priv) { - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - - if (dmaen->chan) - dmaengine_terminate_async(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_STOP); } static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, @@ -120,7 +99,7 @@ static int rsnd_dmaen_cleanup(struct rsnd_mod *mod, * Let's call it under prepare */ if (dmaen->chan) - dma_release_channel(dmaen->chan); + snd_dmaengine_pcm_close_release_chan(io->substream); dmaen->chan = NULL; @@ -153,7 +132,7 @@ static int rsnd_dmaen_prepare(struct rsnd_mod *mod, return -EIO; } - return 0; + return snd_dmaengine_pcm_open(io->substream, dmaen->chan); } static int rsnd_dmaen_start(struct rsnd_mod *mod, @@ -162,12 +141,9 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, { struct rsnd_dma *dma = rsnd_mod_to_dma(mod); struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct snd_pcm_substream *substream = io->substream; struct device *dev = rsnd_priv_to_dev(priv); - struct dma_async_tx_descriptor *desc; struct dma_slave_config cfg = {}; enum dma_slave_buswidth buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; - int is_play = rsnd_io_is_play(io); int ret; /* @@ -195,7 +171,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, } } - cfg.direction = is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM; + cfg.direction = snd_pcm_substream_to_dma_direction(io->substream); cfg.src_addr = dma->src_addr; cfg.dst_addr = dma->dst_addr; cfg.src_addr_width = buswidth; @@ -209,32 +185,7 @@ static int rsnd_dmaen_start(struct rsnd_mod *mod, if (ret < 0) return ret; - desc = dmaengine_prep_dma_cyclic(dmaen->chan, - substream->runtime->dma_addr, - snd_pcm_lib_buffer_bytes(substream), - snd_pcm_lib_period_bytes(substream), - is_play ? DMA_MEM_TO_DEV : DMA_DEV_TO_MEM, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); - - if (!desc) { - dev_err(dev, "dmaengine_prep_slave_sg() fail\n"); - return -EIO; - } - - desc->callback = rsnd_dmaen_complete; - desc->callback_param = rsnd_mod_get(dma); - - dmaen->dma_len = snd_pcm_lib_buffer_bytes(substream); - - dmaen->cookie = dmaengine_submit(desc); - if (dmaen->cookie < 0) { - dev_err(dev, "dmaengine_submit() fail\n"); - return -EIO; - } - - dma_async_issue_pending(dmaen->chan); - - return 0; + return snd_dmaengine_pcm_trigger(io->substream, SNDRV_PCM_TRIGGER_START); } struct dma_chan *rsnd_dma_request_channel(struct device_node *of_node, char *name, @@ -307,19 +258,7 @@ static int rsnd_dmaen_pointer(struct rsnd_mod *mod, struct rsnd_dai_stream *io, snd_pcm_uframes_t *pointer) { - struct snd_pcm_runtime *runtime = rsnd_io_to_runtime(io); - struct rsnd_dma *dma = rsnd_mod_to_dma(mod); - struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); - struct dma_tx_state state; - enum dma_status status; - unsigned int pos = 0; - - status = dmaengine_tx_status(dmaen->chan, dmaen->cookie, &state); - if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { - if (state.residue > 0 && state.residue <= dmaen->dma_len) - pos = dmaen->dma_len - state.residue; - } - *pointer = bytes_to_frames(runtime, pos); + *pointer = snd_dmaengine_pcm_pointer(io->substream); return 0; } diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..dba684e4334b1 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -576,7 +576,6 @@ int rsnd_rdai_ssi_lane_ctrl(struct rsnd_dai *rdai, #define rsnd_rdai_width_get(rdai) \ rsnd_rdai_width_ctrl(rdai, 0) int rsnd_rdai_width_ctrl(struct rsnd_dai *rdai, int width); -void rsnd_dai_period_elapsed(struct rsnd_dai_stream *io); int rsnd_dai_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io, enum rsnd_mod_type type); diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 8d2a86383ae01..b3d4e8ae07eff 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -706,7 +706,7 @@ static void __rsnd_ssi_interrupt(struct rsnd_mod *mod, spin_unlock(&priv->lock); if (elapsed) - rsnd_dai_period_elapsed(io); + snd_pcm_period_elapsed(io->substream); if (stop) snd_pcm_stop_xrun(io->substream); -- 2.43.0

2 1

[PATCH] ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG
by Kuninori Morimoto 31 Jul '24

31 Jul '24

rsnd_mod_confirm_ssi() confirms mod sanity, it should always be confirmed, not only when DEBUG. This patch tidyup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 4 ++-- sound/soc/sh/rcar/rsnd.h | 9 --------- 2 files changed, 2 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index afd69c6eb6544..0f190abf00e75 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -262,7 +262,7 @@ int rsnd_adg_set_src_timesel_gen2(struct rsnd_mod *src_mod, int id = rsnd_mod_id(src_mod); int shift = (id % 2) ? 16 : 0; - rsnd_mod_confirm_src(src_mod); + rsnd_mod_make_sure(src_mod, RSND_MOD_SRC); rsnd_adg_get_timesel_ratio(priv, io, in_rate, out_rate, @@ -291,7 +291,7 @@ static void rsnd_adg_set_ssi_clk(struct rsnd_mod *ssi_mod, u32 val) int shift = (id % 4) * 8; u32 mask = 0xFF << shift; - rsnd_mod_confirm_ssi(ssi_mod); + rsnd_mod_make_sure(ssi_mod, RSND_MOD_SSI); val = val << shift; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index ff294aa2d6407..8cf5d9001f437 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -871,15 +871,6 @@ void rsnd_cmd_remove(struct rsnd_priv *priv); int rsnd_cmd_attach(struct rsnd_dai_stream *io, int id); void rsnd_mod_make_sure(struct rsnd_mod *mod, enum rsnd_mod_type type); -#ifdef DEBUG -#define rsnd_mod_confirm_ssi(mssi) rsnd_mod_make_sure(mssi, RSND_MOD_SSI) -#define rsnd_mod_confirm_src(msrc) rsnd_mod_make_sure(msrc, RSND_MOD_SRC) -#define rsnd_mod_confirm_dvc(mdvc) rsnd_mod_make_sure(mdvc, RSND_MOD_DVC) -#else -#define rsnd_mod_confirm_ssi(mssi) -#define rsnd_mod_confirm_src(msrc) -#define rsnd_mod_confirm_dvc(mdvc) -#endif /* * If you don't need interrupt status debug message, -- 2.43.0

2 1

[PATCH 0/2] ASoC: fsl_micfil: Check the difference for i.MX8 and i.MX9
by Shengjiu Wang 31 Jul '24

31 Jul '24

There are some register difference for i.MX8 and i.MX9 REG_MICFIL_FIFO_CTRL definition is updated. REG_MICFIL_FSYNC_CTRL, REG_MICFIL_VERID, REG_MICFIL_PARAM are added from i.MX9 Shengjiu Wang (2): ASoC: fsl_micfil: Expand the range of FIFO watermark mask ASoC: fsl_micfil: Differentiate register access permission for platforms sound/soc/fsl/fsl_micfil.c | 20 +++++++++++++++----- sound/soc/fsl/fsl_micfil.h | 2 +- 2 files changed, 16 insertions(+), 6 deletions(-) -- 2.34.1

3 4

Re: [PATCH v1] ALSA: hda: cs35l41: Stop creating ALSA Controls for firmware coefficients
by Jaroslav Kysela 31 Jul '24

31 Jul '24

On 30. 07. 24 16:37, Stefan Binding wrote: > Add a kernel parameter to allow coefficients to be exposed as ALSA controls. > > When the CS35L41 loads its firmware, it has a number of controls to > affect its behaviour. Currently, these controls are exposed as ALSA > Controls by default. > > However, nothing in userspace currently uses them, and is unlikely to > do so in the future, therefore we don't need to create ASLA controls > for them. > > These controls can be useful for debug, so we can add a kernel > parameter to re-enable them if necessary. > > Disabling these controls would prevent userspace from trying to read > these controls when the CS35L41 is hibernating, which ordinarily > would result in an error message. This is probably not a right argument to add this code. The codec should be powered up when those controls are accessed or those controls should be cached by the driver. Although the controls have not been used yet, exposing them in this way is not ideal. Could you fix the driver (no I/O errors)? Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 3

[PATCH alsa-lib 0/4] Add API helper functions for creating UMP Endpoint and Blocks
by Takashi Iwai 31 Jul '24

31 Jul '24

Hi, this is a patch set to alsa-lib to add the new API functions for creating a virtual UMP Endpoint and UMP Blocks arbitrarily from a user-space, that is, to create a virtual UMP device. An example program is included in test directory. Takashi === Takashi Iwai (4): ump: Add missing *_set variants for snd_ump_endpoint_info and snd_ump_block_info seq: Add API helper functions for creating UMP Endpoint and Blocks test: Add an example program to create a virtual UMP Endpoint Add test/seq-ump-example to .gitignore .gitignore | 1 + include/seqmid.h | 7 + include/ump.h | 36 ++++- src/rawmidi/ump.c | 304 +++++++++++++++++++++++++++++++++++++++-- src/seq/seq.c | 6 +- src/seq/seq_local.h | 4 + src/seq/seqmid.c | 249 +++++++++++++++++++++++++++++++++ test/Makefile.am | 3 +- test/seq-ump-example.c | 187 +++++++++++++++++++++++++ 9 files changed, 783 insertions(+), 14 deletions(-) create mode 100644 test/seq-ump-example.c -- 2.43.0

2 8

[PATCH v4 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 31 Jul '24

31 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

3 4

Bug in tlv320adc3101 driver
by Terry Sanders 31 Jul '24

31 Jul '24

Hi, There is a bug in the code for the tlv320adc3101 driver when setting up the D value for PLL. The D value is set via I2C using page 0 registers 7 (MSB) and 8 (LSB). The data sheet has the following information for the registers: "Page 0 / Register 7 will be updated when Page 0 / Register 8 is written immediately after Page 0 / Register 7 is written" and: "Page 0 / Register 8 must be written immediately after writing to Page 0 / Register 7" The function that implements this is adc3xxx_setup_pll() in file soc/codecs/tlv320adc3xxx.c line 1172 - 1189. The function is shown below: static void adc3xxx_setup_pll(struct snd_soc_component *component, int div_entry) { int i = div_entry; /* P & R values */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_PR, (adc3xxx_divs[i].pll_p << ADC3XXX_PLLP_SHIFT) | (adc3xxx_divs[i].pll_r << ADC3XXX_PLLR_SHIFT)); /* J value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_J, adc3xxx_divs[i].pll_j & ADC3XXX_PLLJ_MASK); /* D value */ snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_LSB, adc3xxx_divs[i].pll_d & ADC3XXX_PLLD_LSB_MASK); snd_soc_component_write(component, ADC3XXX_PLL_PROG_D_MSB, (adc3xxx_divs[i].pll_d >> 8) & ADC3XXX_PLLD_MSB_MASK); } In the function the LSB (register 8) is written first followed by the MSB (register 7). This is the wrong way around and should be swapped so that the MSB is written first. Regards Terry Sanders

2 1

[PATCH v1] ASoc: tas2781: Move tas2563_dvc_table into a separate Header file
by Shenghao Ding 30 Jul '24

30 Jul '24

Move tas2563_dvc_table into a separate Header file, as only tas2781 codec driver use this table, and hda side codec driver won't use it. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - Create tas2563-tlv.h into include/sound - Move tas2563_dvc_table from tas2781-tlv.h to tas2563-tlv.h - Add "#include <sound/tas2563-tlv.h>" into tas2781-i2c.c --- include/sound/tas2563-tlv.h | 279 +++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 260 ------------------------------ sound/soc/codecs/tas2781-i2c.c | 1 + 3 files changed, 280 insertions(+), 260 deletions(-) create mode 100644 include/sound/tas2563-tlv.h diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h new file mode 100644 index 000000000000..cf6eedb98ad2 --- /dev/null +++ b/include/sound/tas2563-tlv.h @@ -0,0 +1,279 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2563 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2563 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2563 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// + +#ifndef __TAS2563_TLV_H__ +#define __TAS2563_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); + +/* pow(10, db/20) * pow(2,30) */ +static const unsigned char tas2563_dvc_table[][4] = { + { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ + { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ + { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ + { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ + { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ + { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ + { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ + { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ + { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ + { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ + { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ + { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ + { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ + { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ + { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ + { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ + { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ + { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ + { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ + { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ + { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ + { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ + { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ + { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ + { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ + { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ + { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ + { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ + { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ + { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ + { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ + { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ + { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ + { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ + { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ + { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ + { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ + { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ + { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ + { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ + { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ + { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ + { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ + { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ + { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ + { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ + { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ + { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ + { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ + { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ + { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ + { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ + { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ + { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ + { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ + { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ + { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ + { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ + { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ + { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ + { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ + { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ + { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ + { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ + { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ + { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ + { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ + { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ + { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ + { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ + { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ + { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ + { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ + { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ + { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ + { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ + { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ + { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ + { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ + { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ + { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ + { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ + { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ + { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ + { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ + { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ + { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ + { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ + { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ + { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ + { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ + { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ + { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ + { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ + { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ + { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ + { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ + { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ + { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ + { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ + { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ + { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ + { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ + { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ + { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ + { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ + { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ + { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ + { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ + { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ + { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ + { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ + { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ + { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ + { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ + { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ + { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ + { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ + { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ + { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ + { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ + { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ + { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ + { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ + { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ + { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ + { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ + { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ + { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ + { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ + { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ + { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ + { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ + { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ + { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ + { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ + { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ + { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ + { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ + { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ + { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ + { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ + { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ + { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ + { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ + { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ + { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ + { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ + { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ + { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ + { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ + { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ + { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ + { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ + { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ + { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ + { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ + { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ + { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ + { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ + { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ + { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ + { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ + { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ + { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ + { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ + { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ + { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ + { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ + { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ + { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ + { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ + { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ + { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ + { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ + { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ + { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ + { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ + { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ + { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ + { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ + { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ + { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ + { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ + { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ + { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ + { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ + { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ + { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ + { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ + { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ + { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ + { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ + { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ + { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ + { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ + { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ + { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ + { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ + { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ + { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ + { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ + { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ + { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ + { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ + { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ + { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ + { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ + { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ + { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ + { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ + { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ + { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ + { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ + { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ + { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ + { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ + { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ + { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ + { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ + { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ + { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ + { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ + { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ + { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ + { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ + { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ + { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ + { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ + { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ + { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ + { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ + { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ + { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ + { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ + { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ + { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ + { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ + { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ + { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ + { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ + { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ + { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ + { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ + { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ + { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ + { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ + { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ + { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ + { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ + { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ + { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ + { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ + { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ + { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ + { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ +}; +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h index 99c41bfc7827..5b003843752b 100644 --- a/include/sound/tas2781-tlv.h +++ b/include/sound/tas2781-tlv.h @@ -17,265 +17,5 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); -static const DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); -/* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { - { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ - { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ - { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ - { 0X00, 0X00, 0X04, 0X31 }, /* -120.0db */ - { 0X00, 0X00, 0X04, 0X71 }, /* -119.5db */ - { 0X00, 0X00, 0X04, 0XB4 }, /* -119.0db */ - { 0X00, 0X00, 0X04, 0XFC }, /* -118.5db */ - { 0X00, 0X00, 0X05, 0X47 }, /* -118.0db */ - { 0X00, 0X00, 0X05, 0X97 }, /* -117.5db */ - { 0X00, 0X00, 0X05, 0XEC }, /* -117.0db */ - { 0X00, 0X00, 0X06, 0X46 }, /* -116.5db */ - { 0X00, 0X00, 0X06, 0XA5 }, /* -116.0db */ - { 0X00, 0X00, 0X07, 0X0A }, /* -115.5db */ - { 0X00, 0X00, 0X07, 0X75 }, /* -115.0db */ - { 0X00, 0X00, 0X07, 0XE6 }, /* -114.5db */ - { 0X00, 0X00, 0X08, 0X5E }, /* -114.0db */ - { 0X00, 0X00, 0X08, 0XDD }, /* -113.5db */ - { 0X00, 0X00, 0X09, 0X63 }, /* -113.0db */ - { 0X00, 0X00, 0X09, 0XF2 }, /* -112.5db */ - { 0X00, 0X00, 0X0A, 0X89 }, /* -112.0db */ - { 0X00, 0X00, 0X0B, 0X28 }, /* -111.5db */ - { 0X00, 0X00, 0X0B, 0XD2 }, /* -111.0db */ - { 0X00, 0X00, 0X0C, 0X85 }, /* -110.5db */ - { 0X00, 0X00, 0X0D, 0X43 }, /* -110.0db */ - { 0X00, 0X00, 0X0E, 0X0C }, /* -109.5db */ - { 0X00, 0X00, 0X0E, 0XE1 }, /* -109.0db */ - { 0X00, 0X00, 0X0F, 0XC3 }, /* -108.5db */ - { 0X00, 0X00, 0X10, 0XB2 }, /* -108.0db */ - { 0X00, 0X00, 0X11, 0XAF }, /* -107.5db */ - { 0X00, 0X00, 0X12, 0XBC }, /* -107.0db */ - { 0X00, 0X00, 0X13, 0XD8 }, /* -106.5db */ - { 0X00, 0X00, 0X15, 0X05 }, /* -106.0db */ - { 0X00, 0X00, 0X16, 0X44 }, /* -105.5db */ - { 0X00, 0X00, 0X17, 0X96 }, /* -105.0db */ - { 0X00, 0X00, 0X18, 0XFB }, /* -104.5db */ - { 0X00, 0X00, 0X1A, 0X76 }, /* -104.0db */ - { 0X00, 0X00, 0X1C, 0X08 }, /* -103.5db */ - { 0X00, 0X00, 0X1D, 0XB1 }, /* -103.0db */ - { 0X00, 0X00, 0X1F, 0X73 }, /* -102.5db */ - { 0X00, 0X00, 0X21, 0X51 }, /* -102.0db */ - { 0X00, 0X00, 0X23, 0X4A }, /* -101.5db */ - { 0X00, 0X00, 0X25, 0X61 }, /* -101.0db */ - { 0X00, 0X00, 0X27, 0X98 }, /* -100.5db */ - { 0X00, 0X00, 0X29, 0XF1 }, /* -100.0db */ - { 0X00, 0X00, 0X2C, 0X6D }, /* -99.5db */ - { 0X00, 0X00, 0X2F, 0X0F }, /* -99.0db */ - { 0X00, 0X00, 0X31, 0XD9 }, /* -98.5db */ - { 0X00, 0X00, 0X34, 0XCD }, /* -98.0db */ - { 0X00, 0X00, 0X37, 0XEE }, /* -97.5db */ - { 0X00, 0X00, 0X3B, 0X3F }, /* -97.0db */ - { 0X00, 0X00, 0X3E, 0XC1 }, /* -96.5db */ - { 0X00, 0X00, 0X42, 0X79 }, /* -96.0db */ - { 0X00, 0X00, 0X46, 0X6A }, /* -95.5db */ - { 0X00, 0X00, 0X4A, 0X96 }, /* -95.0db */ - { 0X00, 0X00, 0X4F, 0X01 }, /* -94.5db */ - { 0X00, 0X00, 0X53, 0XAF }, /* -94.0db */ - { 0X00, 0X00, 0X58, 0XA5 }, /* -93.5db */ - { 0X00, 0X00, 0X5D, 0XE6 }, /* -93.0db */ - { 0X00, 0X00, 0X63, 0X76 }, /* -92.5db */ - { 0X00, 0X00, 0X69, 0X5B }, /* -92.0db */ - { 0X00, 0X00, 0X6F, 0X99 }, /* -91.5db */ - { 0X00, 0X00, 0X76, 0X36 }, /* -91.0db */ - { 0X00, 0X00, 0X7D, 0X37 }, /* -90.5db */ - { 0X00, 0X00, 0X84, 0XA2 }, /* -90.0db */ - { 0X00, 0X00, 0X8C, 0X7E }, /* -89.5db */ - { 0X00, 0X00, 0X94, 0XD1 }, /* -89.0db */ - { 0X00, 0X00, 0X9D, 0XA3 }, /* -88.5db */ - { 0X00, 0X00, 0XA6, 0XFA }, /* -88.0db */ - { 0X00, 0X00, 0XB0, 0XDF }, /* -87.5db */ - { 0X00, 0X00, 0XBB, 0X5A }, /* -87.0db */ - { 0X00, 0X00, 0XC6, 0X74 }, /* -86.5db */ - { 0X00, 0X00, 0XD2, 0X36 }, /* -86.0db */ - { 0X00, 0X00, 0XDE, 0XAB }, /* -85.5db */ - { 0X00, 0X00, 0XEB, 0XDC }, /* -85.0db */ - { 0X00, 0X00, 0XF9, 0XD6 }, /* -84.5db */ - { 0X00, 0X01, 0X08, 0XA4 }, /* -84.0db */ - { 0X00, 0X01, 0X18, 0X52 }, /* -83.5db */ - { 0X00, 0X01, 0X28, 0XEF }, /* -83.0db */ - { 0X00, 0X01, 0X3A, 0X87 }, /* -82.5db */ - { 0X00, 0X01, 0X4D, 0X2A }, /* -82.0db */ - { 0X00, 0X01, 0X60, 0XE8 }, /* -81.5db */ - { 0X00, 0X01, 0X75, 0XD1 }, /* -81.0db */ - { 0X00, 0X01, 0X8B, 0XF7 }, /* -80.5db */ - { 0X00, 0X01, 0XA3, 0X6E }, /* -80.0db */ - { 0X00, 0X01, 0XBC, 0X48 }, /* -79.5db */ - { 0X00, 0X01, 0XD6, 0X9B }, /* -79.0db */ - { 0X00, 0X01, 0XF2, 0X7E }, /* -78.5db */ - { 0X00, 0X02, 0X10, 0X08 }, /* -78.0db */ - { 0X00, 0X02, 0X2F, 0X51 }, /* -77.5db */ - { 0X00, 0X02, 0X50, 0X76 }, /* -77.0db */ - { 0X00, 0X02, 0X73, 0X91 }, /* -76.5db */ - { 0X00, 0X02, 0X98, 0XC0 }, /* -76.0db */ - { 0X00, 0X02, 0XC0, 0X24 }, /* -75.5db */ - { 0X00, 0X02, 0XE9, 0XDD }, /* -75.0db */ - { 0X00, 0X03, 0X16, 0X0F }, /* -74.5db */ - { 0X00, 0X03, 0X44, 0XDF }, /* -74.0db */ - { 0X00, 0X03, 0X76, 0X76 }, /* -73.5db */ - { 0X00, 0X03, 0XAA, 0XFC }, /* -73.0db */ - { 0X00, 0X03, 0XE2, 0XA0 }, /* -72.5db */ - { 0X00, 0X04, 0X1D, 0X8F }, /* -72.0db */ - { 0X00, 0X04, 0X5B, 0XFD }, /* -71.5db */ - { 0X00, 0X04, 0X9E, 0X1D }, /* -71.0db */ - { 0X00, 0X04, 0XE4, 0X29 }, /* -70.5db */ - { 0X00, 0X05, 0X2E, 0X5A }, /* -70.0db */ - { 0X00, 0X05, 0X7C, 0XF2 }, /* -69.5db */ - { 0X00, 0X05, 0XD0, 0X31 }, /* -69.0db */ - { 0X00, 0X06, 0X28, 0X60 }, /* -68.5db */ - { 0X00, 0X06, 0X85, 0XC8 }, /* -68.0db */ - { 0X00, 0X06, 0XE8, 0XB9 }, /* -67.5db */ - { 0X00, 0X07, 0X51, 0X86 }, /* -67.0db */ - { 0X00, 0X07, 0XC0, 0X8A }, /* -66.5db */ - { 0X00, 0X08, 0X36, 0X21 }, /* -66.0db */ - { 0X00, 0X08, 0XB2, 0XB0 }, /* -65.5db */ - { 0X00, 0X09, 0X36, 0XA1 }, /* -65.0db */ - { 0X00, 0X09, 0XC2, 0X63 }, /* -64.5db */ - { 0X00, 0X0A, 0X56, 0X6D }, /* -64.0db */ - { 0X00, 0X0A, 0XF3, 0X3C }, /* -63.5db */ - { 0X00, 0X0B, 0X99, 0X56 }, /* -63.0db */ - { 0X00, 0X0C, 0X49, 0X48 }, /* -62.5db */ - { 0X00, 0X0D, 0X03, 0XA7 }, /* -62.0db */ - { 0X00, 0X0D, 0XC9, 0X11 }, /* -61.5db */ - { 0X00, 0X0E, 0X9A, 0X2D }, /* -61.0db */ - { 0X00, 0X0F, 0X77, 0XAD }, /* -60.5db */ - { 0X00, 0X10, 0X62, 0X4D }, /* -60.0db */ - { 0X00, 0X11, 0X5A, 0XD5 }, /* -59.5db */ - { 0X00, 0X12, 0X62, 0X16 }, /* -59.0db */ - { 0X00, 0X13, 0X78, 0XF0 }, /* -58.5db */ - { 0X00, 0X14, 0XA0, 0X50 }, /* -58.0db */ - { 0X00, 0X15, 0XD9, 0X31 }, /* -57.5db */ - { 0X00, 0X17, 0X24, 0X9C }, /* -57.0db */ - { 0X00, 0X18, 0X83, 0XAA }, /* -56.5db */ - { 0X00, 0X19, 0XF7, 0X86 }, /* -56.0db */ - { 0X00, 0X1B, 0X81, 0X6A }, /* -55.5db */ - { 0X00, 0X1D, 0X22, 0XA4 }, /* -55.0db */ - { 0X00, 0X1E, 0XDC, 0X98 }, /* -54.5db */ - { 0X00, 0X20, 0XB0, 0XBC }, /* -54.0db */ - { 0X00, 0X22, 0XA0, 0X9D }, /* -53.5db */ - { 0X00, 0X24, 0XAD, 0XE0 }, /* -53.0db */ - { 0X00, 0X26, 0XDA, 0X43 }, /* -52.5db */ - { 0X00, 0X29, 0X27, 0X9D }, /* -52.0db */ - { 0X00, 0X2B, 0X97, 0XE3 }, /* -51.5db */ - { 0X00, 0X2E, 0X2D, 0X27 }, /* -51.0db */ - { 0X00, 0X30, 0XE9, 0X9A }, /* -50.5db */ - { 0X00, 0X33, 0XCF, 0X8D }, /* -50.0db */ - { 0X00, 0X36, 0XE1, 0X78 }, /* -49.5db */ - { 0X00, 0X3A, 0X21, 0XF3 }, /* -49.0db */ - { 0X00, 0X3D, 0X93, 0XC3 }, /* -48.5db */ - { 0X00, 0X41, 0X39, 0XD3 }, /* -48.0db */ - { 0X00, 0X45, 0X17, 0X3B }, /* -47.5db */ - { 0X00, 0X49, 0X2F, 0X44 }, /* -47.0db */ - { 0X00, 0X4D, 0X85, 0X66 }, /* -46.5db */ - { 0X00, 0X52, 0X1D, 0X50 }, /* -46.0db */ - { 0X00, 0X56, 0XFA, 0XE8 }, /* -45.5db */ - { 0X00, 0X5C, 0X22, 0X4E }, /* -45.0db */ - { 0X00, 0X61, 0X97, 0XE1 }, /* -44.5db */ - { 0X00, 0X67, 0X60, 0X44 }, /* -44.0db */ - { 0X00, 0X6D, 0X80, 0X60 }, /* -43.5db */ - { 0X00, 0X73, 0XFD, 0X65 }, /* -43.0db */ - { 0X00, 0X7A, 0XDC, 0XD7 }, /* -42.5db */ - { 0X00, 0X82, 0X24, 0X8A }, /* -42.0db */ - { 0X00, 0X89, 0XDA, 0XAB }, /* -41.5db */ - { 0X00, 0X92, 0X05, 0XC6 }, /* -41.0db */ - { 0X00, 0X9A, 0XAC, 0XC8 }, /* -40.5db */ - { 0X00, 0XA3, 0XD7, 0X0A }, /* -40.0db */ - { 0X00, 0XAD, 0X8C, 0X52 }, /* -39.5db */ - { 0X00, 0XB7, 0XD4, 0XDD }, /* -39.0db */ - { 0X00, 0XC2, 0XB9, 0X65 }, /* -38.5db */ - { 0X00, 0XCE, 0X43, 0X28 }, /* -38.0db */ - { 0X00, 0XDA, 0X7B, 0XF1 }, /* -37.5db */ - { 0X00, 0XE7, 0X6E, 0X1E }, /* -37.0db */ - { 0X00, 0XF5, 0X24, 0XAC }, /* -36.5db */ - { 0X01, 0X03, 0XAB, 0X3D }, /* -36.0db */ - { 0X01, 0X13, 0X0E, 0X24 }, /* -35.5db */ - { 0X01, 0X23, 0X5A, 0X71 }, /* -35.0db */ - { 0X01, 0X34, 0X9D, 0XF8 }, /* -34.5db */ - { 0X01, 0X46, 0XE7, 0X5D }, /* -34.0db */ - { 0X01, 0X5A, 0X46, 0X27 }, /* -33.5db */ - { 0X01, 0X6E, 0XCA, 0XC5 }, /* -33.0db */ - { 0X01, 0X84, 0X86, 0X9F }, /* -32.5db */ - { 0X01, 0X9B, 0X8C, 0X27 }, /* -32.0db */ - { 0X01, 0XB3, 0XEE, 0XE5 }, /* -31.5db */ - { 0X01, 0XCD, 0XC3, 0X8C }, /* -31.0db */ - { 0X01, 0XE9, 0X20, 0X05 }, /* -30.5db */ - { 0X02, 0X06, 0X1B, 0X89 }, /* -30.0db */ - { 0X02, 0X24, 0XCE, 0XB0 }, /* -29.5db */ - { 0X02, 0X45, 0X53, 0X85 }, /* -29.0db */ - { 0X02, 0X67, 0XC5, 0XA2 }, /* -28.5db */ - { 0X02, 0X8C, 0X42, 0X3F }, /* -28.0db */ - { 0X02, 0XB2, 0XE8, 0X55 }, /* -27.5db */ - { 0X02, 0XDB, 0XD8, 0XAD }, /* -27.0db */ - { 0X03, 0X07, 0X36, 0X05 }, /* -26.5db */ - { 0X03, 0X35, 0X25, 0X29 }, /* -26.0db */ - { 0X03, 0X65, 0XCD, 0X13 }, /* -25.5db */ - { 0X03, 0X99, 0X57, 0X0C }, /* -25.0db */ - { 0X03, 0XCF, 0XEE, 0XCF }, /* -24.5db */ - { 0X04, 0X09, 0XC2, 0XB0 }, /* -24.0db */ - { 0X04, 0X47, 0X03, 0XC1 }, /* -23.5db */ - { 0X04, 0X87, 0XE5, 0XFB }, /* -23.0db */ - { 0X04, 0XCC, 0XA0, 0X6D }, /* -22.5db */ - { 0X05, 0X15, 0X6D, 0X68 }, /* -22.0db */ - { 0X05, 0X62, 0X8A, 0XB3 }, /* -21.5db */ - { 0X05, 0XB4, 0X39, 0XBC }, /* -21.0db */ - { 0X06, 0X0A, 0XBF, 0XD4 }, /* -20.5db */ - { 0X06, 0X66, 0X66, 0X66 }, /* -20.0db */ - { 0X06, 0XC7, 0X7B, 0X36 }, /* -19.5db */ - { 0X07, 0X2E, 0X50, 0XA6 }, /* -19.0db */ - { 0X07, 0X9B, 0X3D, 0XF6 }, /* -18.5db */ - { 0X08, 0X0E, 0X9F, 0X96 }, /* -18.0db */ - { 0X08, 0X88, 0XD7, 0X6D }, /* -17.5db */ - { 0X09, 0X0A, 0X4D, 0X2F }, /* -17.0db */ - { 0X09, 0X93, 0X6E, 0XB8 }, /* -16.5db */ - { 0X0A, 0X24, 0XB0, 0X62 }, /* -16.0db */ - { 0X0A, 0XBE, 0X8D, 0X70 }, /* -15.5db */ - { 0X0B, 0X61, 0X88, 0X71 }, /* -15.0db */ - { 0X0C, 0X0E, 0X2B, 0XB0 }, /* -14.5db */ - { 0X0C, 0XC5, 0X09, 0XAB }, /* -14.0db */ - { 0X0D, 0X86, 0XBD, 0X8D }, /* -13.5db */ - { 0X0E, 0X53, 0XEB, 0XB3 }, /* -13.0db */ - { 0X0F, 0X2D, 0X42, 0X38 }, /* -12.5db */ - { 0X10, 0X13, 0X79, 0X87 }, /* -12.0db */ - { 0X11, 0X07, 0X54, 0XF9 }, /* -11.5db */ - { 0X12, 0X09, 0XA3, 0X7A }, /* -11.0db */ - { 0X13, 0X1B, 0X40, 0X39 }, /* -10.5db */ - { 0X14, 0X3D, 0X13, 0X62 }, /* -10.0db */ - { 0X15, 0X70, 0X12, 0XE1 }, /* -9.5db */ - { 0X16, 0XB5, 0X43, 0X37 }, /* -9.0db */ - { 0X18, 0X0D, 0XB8, 0X54 }, /* -8.5db */ - { 0X19, 0X7A, 0X96, 0X7F }, /* -8.0db */ - { 0X1A, 0XFD, 0X13, 0X54 }, /* -7.5db */ - { 0X1C, 0X96, 0X76, 0XC6 }, /* -7.0db */ - { 0X1E, 0X48, 0X1C, 0X37 }, /* -6.5db */ - { 0X20, 0X13, 0X73, 0X9E }, /* -6.0db */ - { 0X21, 0XFA, 0X02, 0XBF }, /* -5.5db */ - { 0X23, 0XFD, 0X66, 0X78 }, /* -5.0db */ - { 0X26, 0X1F, 0X54, 0X1C }, /* -4.5db */ - { 0X28, 0X61, 0X9A, 0XE9 }, /* -4.0db */ - { 0X2A, 0XC6, 0X25, 0X91 }, /* -3.5db */ - { 0X2D, 0X4E, 0XFB, 0XD5 }, /* -3.0db */ - { 0X2F, 0XFE, 0X44, 0X48 }, /* -2.5db */ - { 0X32, 0XD6, 0X46, 0X17 }, /* -2.0db */ - { 0X35, 0XD9, 0X6B, 0X02 }, /* -1.5db */ - { 0X39, 0X0A, 0X41, 0X5F }, /* -1.0db */ - { 0X3C, 0X6B, 0X7E, 0X4F }, /* -0.5db */ - { 0X40, 0X00, 0X00, 0X00 }, /* 0.0db */ - { 0X43, 0XCA, 0XD0, 0X22 }, /* 0.5db */ - { 0X47, 0XCF, 0X26, 0X7D }, /* 1.0db */ - { 0X4C, 0X10, 0X6B, 0XA5 }, /* 1.5db */ - { 0X50, 0X92, 0X3B, 0XE3 }, /* 2.0db */ - { 0X55, 0X58, 0X6A, 0X46 }, /* 2.5db */ - { 0X5A, 0X67, 0X03, 0XDF }, /* 3.0db */ - { 0X5F, 0XC2, 0X53, 0X32 }, /* 3.5db */ - { 0X65, 0X6E, 0XE3, 0XDB }, /* 4.0db */ - { 0X6B, 0X71, 0X86, 0X68 }, /* 4.5db */ - { 0X71, 0XCF, 0X54, 0X71 }, /* 5.0db */ - { 0X78, 0X8D, 0XB4, 0XE9 }, /* 5.5db */ - { 0XFF, 0XFF, 0XFF, 0XFF }, /* 6.0db */ -}; #endif diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e79d613745b4..8a57c045afdd 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -30,6 +30,7 @@ #include <sound/soc.h> #include <sound/tas2781.h> #include <sound/tlv.h> +#include <sound/tas2563-tlv.h> #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> -- 2.34.1

3 4

[PATCH v1] ASoc: tas2781: Add TAS2563 into the Header
by Shenghao Ding 30 Jul '24

30 Jul '24

Add TAS2563 into the Header in case of misunderstanding and add channel No information for error debug in tasdevice_dev_read. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Add TAS2563 into the Header - Add channel no into the log in tasdevice_dev_read for error debug --- sound/soc/codecs/tas2781-comlib.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1fbf4560f5cc..58abbc098a91 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -1,6 +1,6 @@ // SPDX-License-Identifier: GPL-2.0 // -// TAS2781 Common functions for HDA and ASoC Audio drivers +// TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // // Copyright 2023 - 2024 Texas Instruments, Inc. // @@ -64,8 +64,8 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, */ ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); if (ret < 0) { - dev_err(tas_priv->dev, "%s, E=%d\n", - __func__, ret); + dev_err(tas_priv->dev, "%s, E=%d channel:%d\n", + __func__, ret, chn); goto out; } } -- 2.34.1

2 1

[PATCH v3 0/3] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 30 Jul '24

30 Jul '24

This patchset serves to prevent an AB/BA deadlock: thread 0: * (lock A) acquire substream lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) wait for tasklet to finish by calling tasklet_unlock_spin_wait() in tasklet_disable_in_atomic() in ohci_flush_iso_completions() of ohci.c thread 1: * (lock B) enter tasklet * (lock A) attempt to acquire substream lock, waiting for it to be released: snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() in update_pcm_pointers() in process_ctx_payloads() in process_rx_packets() of amdtp-stream.c ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Cc: stable(a)vger.kernel.org Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (3): Revert "ALSA: firewire-lib: obsolete workqueue for period update" Revert "ALSA: firewire-lib: operate for period elapse event in process context" ALSA: firewire-lib: amdtp-stream work queue inline description sound/firewire/amdtp-stream.c | 38 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 25 insertions(+), 14 deletions(-) -- 2.45.2

2 4

[PATCH] ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding
by Krzysztof Kozlowski 30 Jul '24

30 Jul '24

The APQ8016 SBC and MSM8916 QDSP6 sound cards are a bit different from others: they have additional IO muxing address space and pin control. Move them to separate schema, so the original qcom,sm8250.yaml will be easier to manage. New schema is going to grow for other platforms having more of IO muxing address spaces. Cc: Adam Skladowski <a39.skl(a)gmail.com> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../sound/qcom,apq8016-sbc-sndcard.yaml | 205 ++++++++++++++++++ .../bindings/sound/qcom,sm8250.yaml | 137 ------------ 2 files changed, 205 insertions(+), 137 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml new file mode 100644 index 000000000000..6ad451549036 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml @@ -0,0 +1,205 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,apq8016-sbc-sndcard.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm APQ8016 and similar sound cards + +maintainers: + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + - Stephan Gerhold <stephan(a)gerhold.net> + +properties: + compatible: + enum: + - qcom,apq8016-sbc-sndcard + - qcom,msm8916-qdsp6-sndcard + + reg: + items: + - description: Microphone I/O mux register address + - description: Speaker I/O mux register address + + reg-names: + items: + - const: mic-iomux + - const: spkr-iomux + + audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + 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. Valid names could be power supplies, + MicBias of codec and the jacks on the board. + + aux-devs: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + + model: + $ref: /schemas/types.yaml#/definitions/string + description: User visible long sound card name + + pin-switches: + description: List of widget names for which pin switches should be created. + $ref: /schemas/types.yaml#/definitions/string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + +patternProperties: + ".*-dai-link$": + description: + Each subnode represents a dai link. Subnodes of each dai links would be + cpu/codec dais. + + type: object + + properties: + link-name: + description: Indicates dai-link name and PCM stream name. + $ref: /schemas/types.yaml#/definitions/string + maxItems: 1 + + cpu: + description: Holds subnode which indicates cpu dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + platform: + description: Holds subnode which indicates platform dai. + type: object + additionalProperties: false + + properties: + sound-dai: + maxItems: 1 + + codec: + description: Holds subnode which indicates codec dai. + type: object + additionalProperties: false + + properties: + sound-dai: + minItems: 1 + maxItems: 8 + + required: + - link-name + - cpu + + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - model + +additionalProperties: false + +examples: + - | + #include <dt-bindings/sound/qcom,lpass.h> + sound@7702000 { + compatible = "qcom,apq8016-sbc-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "DB410c"; + audio-routing = + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS External1"; + + pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; + pinctrl-names = "default", "sleep"; + + quaternary-dai-link { + link-name = "ADV7533"; + cpu { + sound-dai = <&lpass MI2S_QUATERNARY>; + }; + codec { + sound-dai = <&adv_bridge 0>; + }; + }; + + primary-dai-link { + link-name = "WCD"; + cpu { + sound-dai = <&lpass MI2S_PRIMARY>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + + tertiary-dai-link { + link-name = "WCD-Capture"; + cpu { + sound-dai = <&lpass MI2S_TERTIARY>; + }; + codec { + sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; + }; + }; + }; + + - | + #include <dt-bindings/sound/qcom,q6afe.h> + #include <dt-bindings/sound/qcom,q6asm.h> + sound@7702000 { + compatible = "qcom,msm8916-qdsp6-sndcard"; + reg = <0x07702000 0x4>, <0x07702004 0x4>; + reg-names = "mic-iomux", "spkr-iomux"; + + model = "msm8916"; + widgets = + "Speaker", "Speaker", + "Headphone", "Headphones"; + pin-switches = "Speaker"; + audio-routing = + "Speaker", "Speaker Amp OUT", + "Speaker Amp IN", "HPH_R", + "Headphones", "HPH_L", + "Headphones", "HPH_R", + "AMIC1", "MIC BIAS Internal1", + "AMIC2", "MIC BIAS Internal2", + "AMIC3", "MIC BIAS Internal3"; + aux-devs = <&speaker_amp>; + + pinctrl-names = "default", "sleep"; + pinctrl-0 = <&cdc_pdm_lines_act>; + pinctrl-1 = <&cdc_pdm_lines_sus>; + + mm1-dai-link { + link-name = "MultiMedia1"; + cpu { + sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; + }; + }; + + primary-dai-link { + link-name = "Primary MI2S"; + cpu { + sound-dai = <&q6afedai PRIMARY_MI2S_RX>; + }; + platform { + sound-dai = <&q6routing>; + }; + codec { + sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; + }; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml index c9076dcd44c1..1d3acdc0c733 100644 --- a/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,sm8250.yaml @@ -27,9 +27,7 @@ properties: - qcom,sm8650-sndcard - const: qcom,sm8450-sndcard - enum: - - qcom,apq8016-sbc-sndcard - qcom,apq8096-sndcard - - qcom,msm8916-qdsp6-sndcard - qcom,qcm6490-idp-sndcard - qcom,qcs6490-rb3gen2-sndcard - qcom,qrb5165-rb5-sndcard @@ -58,18 +56,6 @@ properties: $ref: /schemas/types.yaml#/definitions/string description: User visible long sound card name - pin-switches: - description: List of widget names for which pin switches should be created. - $ref: /schemas/types.yaml#/definitions/string-array - - widgets: - description: User specified audio sound widgets. - $ref: /schemas/types.yaml#/definitions/non-unique-string-array - - # Only valid for some compatibles (see allOf if below) - reg: true - reg-names: true - patternProperties: ".*-dai-link$": description: @@ -122,34 +108,6 @@ required: - compatible - model -allOf: - - if: - properties: - compatible: - contains: - enum: - - qcom,apq8016-sbc-sndcard - - qcom,msm8916-qdsp6-sndcard - then: - properties: - reg: - items: - - description: Microphone I/O mux register address - - description: Speaker I/O mux register address - reg-names: - items: - - const: mic-iomux - - const: spkr-iomux - required: - - compatible - - model - - reg - - reg-names - else: - properties: - reg: false - reg-names: false - additionalProperties: false examples: @@ -231,98 +189,3 @@ examples: }; }; }; - - - | - #include <dt-bindings/sound/qcom,lpass.h> - sound@7702000 { - compatible = "qcom,apq8016-sbc-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "DB410c"; - audio-routing = - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS External1"; - - pinctrl-0 = <&cdc_pdm_lines_act &ext_sec_tlmm_lines_act &ext_mclk_tlmm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus &ext_sec_tlmm_lines_sus &ext_mclk_tlmm_lines_sus>; - pinctrl-names = "default", "sleep"; - - quaternary-dai-link { - link-name = "ADV7533"; - cpu { - sound-dai = <&lpass MI2S_QUATERNARY>; - }; - codec { - sound-dai = <&adv_bridge 0>; - }; - }; - - primary-dai-link { - link-name = "WCD"; - cpu { - sound-dai = <&lpass MI2S_PRIMARY>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - - tertiary-dai-link { - link-name = "WCD-Capture"; - cpu { - sound-dai = <&lpass MI2S_TERTIARY>; - }; - codec { - sound-dai = <&lpass_codec 1>, <&wcd_codec 1>; - }; - }; - }; - - - | - #include <dt-bindings/sound/qcom,q6afe.h> - #include <dt-bindings/sound/qcom,q6asm.h> - sound@7702000 { - compatible = "qcom,msm8916-qdsp6-sndcard"; - reg = <0x07702000 0x4>, <0x07702004 0x4>; - reg-names = "mic-iomux", "spkr-iomux"; - - model = "msm8916"; - widgets = - "Speaker", "Speaker", - "Headphone", "Headphones"; - pin-switches = "Speaker"; - audio-routing = - "Speaker", "Speaker Amp OUT", - "Speaker Amp IN", "HPH_R", - "Headphones", "HPH_L", - "Headphones", "HPH_R", - "AMIC1", "MIC BIAS Internal1", - "AMIC2", "MIC BIAS Internal2", - "AMIC3", "MIC BIAS Internal3"; - aux-devs = <&speaker_amp>; - - pinctrl-names = "default", "sleep"; - pinctrl-0 = <&cdc_pdm_lines_act>; - pinctrl-1 = <&cdc_pdm_lines_sus>; - - mm1-dai-link { - link-name = "MultiMedia1"; - cpu { - sound-dai = <&q6asmdai MSM_FRONTEND_DAI_MULTIMEDIA1>; - }; - }; - - primary-dai-link { - link-name = "Primary MI2S"; - cpu { - sound-dai = <&q6afedai PRIMARY_MI2S_RX>; - }; - platform { - sound-dai = <&q6routing>; - }; - codec { - sound-dai = <&lpass_codec 0>, <&wcd_codec 0>; - }; - }; - }; -- 2.43.0

6 6

[PATCH 0/6] ASoC: codecs: wcd93xx/wsa88xx: Correct Soundwire ports mask
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Incorrect mask of Soundwire ports - one bit too long/big - was passed. Theoretically, too wide mask could cause an out of bounds read in sdw_get_slave_dpn_prop() in stream.c, however only in the case of buggy driver, e.g. adding incorrect number of ports via sdw_stream_add_slave(). Not adding Cc-stable because I don't think the issue can lead to real out-of-bounds read. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: wcd937x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd938x-sdw: Correct Soundwire ports mask ASoC: codecs: wcd939x-sdw: Correct Soundwire ports mask ASoC: codecs: wsa881x: Correct Soundwire ports mask ASoC: codecs: wsa883x: Correct Soundwire ports mask ASoC: codecs: wsa884x: Correct Soundwire ports mask sound/soc/codecs/wcd937x-sdw.c | 4 ++-- sound/soc/codecs/wcd938x-sdw.c | 4 ++-- sound/soc/codecs/wcd939x-sdw.c | 4 ++-- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- 6 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240726-asoc-wcd-wsa-swr-ports-genmask-868239f9682f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 8

[PATCH] ALSA: hda: cs35l56: Stop creating ALSA controls for firmware coefficients
by Simon Trimmer 29 Jul '24

29 Jul '24

A number of laptops have gone to market with old firmware versions that export controls that have since been hidden, but we can't just install a newer firmware because the firmware for each product is customized and qualified by the OEM. The issue is that alsactl save and restore has no idea what controls are good to persist which can lead to misconfiguration. As the ALSA controls for the firmware coefficients are not used in normal operation they can all be hidden, but add a kernel parameter so that they can be re-enabled for debugging. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/cs35l56_hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 96d3f13c5abf..1494383b22c9 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -23,6 +23,10 @@ #include "hda_cs_dsp_ctl.h" #include "hda_generic.h" +static bool expose_dsp_controls; +module_param(expose_dsp_controls, bool, 0444); +MODULE_PARM_DESC(expose_dsp_controls, "Expose firmware controls as ALSA controls 0=no (default), 1=yes"); + /* * The cs35l56_hda_dai_config[] reg sequence configures the device as * ASP1_BCLK_FREQ = 3.072 MHz @@ -758,6 +762,9 @@ static int cs35l56_hda_bind(struct device *dev, struct device *master, void *mas cs35l56_hda_create_controls(cs35l56); + if (expose_dsp_controls) + cs35l56_hda_add_dsp_controls(cs35l56); + #if IS_ENABLED(CONFIG_SND_DEBUG) cs35l56->debugfs_root = debugfs_create_dir(dev_name(cs35l56->base.dev), sound_debugfs_root); cs_dsp_init_debugfs(&cs35l56->cs_dsp, cs35l56->debugfs_root); -- 2.43.0

2 2

ASUS RoG Strix Z790-F: SPDIF (PCM) has wrong channels | how to get passthrough working?
by GitHub issues - opened 29 Jul '24

29 Jul '24

alsa-project/alsa-ucm-conf issue #434 was opened from Nightwing0815: Hi there, finally, [after a long journey,](https://github.com/alsa-project/alsa-ucm-conf/issues/350) yesterday I did an update from linuxmint to newest version 22, based on ubuntu 24.04 (I'll attach alsa-info and a system info). Now it's no big problem any more, to get SPDIF working: I just selected the right device and muted all others in alsamixer, except the pcm signal. But hey, there's a new problem: With a speaker-test -c6, I'm not able to hear my rear channels and the lfe channel, where they should be. Rear coming on front speakers... So, what do I have to do? Further, I'm asking for help to configure a pcm signal passthru for dts and (e)ac3 soundfiles. My teufel receiver recognices only a 5.1 stereo source, not an DD5.1 or DTS signal... [alsa-info.txt](https://github.com/user-attachments/files/16417700/alsa-info… [sysinfo.txt](https://github.com/user-attachments/files/16417724/sysinfo.txt) Thanks in advance for getting this problems solved, ToM Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/434 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/7] ASoC: codecs: wsa88xx and wcd93xx: Soundwire port non-functional cleanup
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups to make the code more robust or readable. No functional impact (compiled objects stay the same). Best regards, Krzysztof --- Krzysztof Kozlowski (7): ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum sound/soc/codecs/wcd937x.h | 34 +++++++++++++++++----------------- sound/soc/codecs/wcd938x.c | 18 ------------------ sound/soc/codecs/wcd938x.h | 4 ++-- sound/soc/codecs/wcd939x.h | 6 +++--- sound/soc/codecs/wsa881x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa883x.c | 42 ++++++++++++++++++++++-------------------- sound/soc/codecs/wsa884x.c | 34 ++++++++++++++++++++++------------ 7 files changed, 88 insertions(+), 92 deletions(-) --- base-commit: 66df68b9536e7465aec703733ea91ca39b787ed8 change-id: 20240725-asoc-wsa88xx-port-arrays-c58c96cd888e Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

3 15

[PATCH v2] ASoC: sti: add missing probe entry for player and reader
by Jerome Audu 29 Jul '24

29 Jul '24

This patch addresses a regression in the ASoC STI drivers that was introduced in Linux version 6.6.y. The issue originated from a series of patches (see https://lore.kernel.org/all/87wmy5b0wt.wl-kuninori.morimoto.gx@renesas.com/) that unintentionally omitted necessary probe functions for the player and reader components. Probe function in `sound/soc/sti/sti_uniperif.c:415` is being replaced by another probe function located at `sound/soc/sti/sti_uniperif.c:453`, which should instead be derived from the player and reader components. This patch correctly reinserts the missing probe entries, restoring the intended functionality. Fixes: 9f625f5e6cf9 ("ASoC: sti: merge DAI call back functions into ops") Signed-off-by: Jerome Audu <jau(a)free.fr> --- Changes in v2: - Update changelog - Link to v1: https://lore.kernel.org/r/20240721-sti-audio-fix-v1-1-a8b1ecf61cb4@free.fr --- sound/soc/sti/sti_uniperif.c | 2 +- sound/soc/sti/uniperif.h | 1 + sound/soc/sti/uniperif_player.c | 1 + sound/soc/sti/uniperif_reader.c | 1 + 4 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index ba824f14a39c..a7956e5a4ee5 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -352,7 +352,7 @@ static int sti_uniperiph_resume(struct snd_soc_component *component) return ret; } -static int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai) { struct sti_uniperiph_data *priv = snd_soc_dai_get_drvdata(dai); struct sti_uniperiph_dai *dai_data = &priv->dai_data; diff --git a/sound/soc/sti/uniperif.h b/sound/soc/sti/uniperif.h index 2a5de328501c..74e51f0ff85c 100644 --- a/sound/soc/sti/uniperif.h +++ b/sound/soc/sti/uniperif.h @@ -1380,6 +1380,7 @@ int uni_reader_init(struct platform_device *pdev, struct uniperif *reader); /* common */ +int sti_uniperiph_dai_probe(struct snd_soc_dai *dai); int sti_uniperiph_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index dd9013c47664..6d1ce030963c 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1038,6 +1038,7 @@ static const struct snd_soc_dai_ops uni_player_dai_ops = { .startup = uni_player_startup, .shutdown = uni_player_shutdown, .prepare = uni_player_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_player_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index 065c5f0d1f5f..05ea2b794eb9 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -401,6 +401,7 @@ static const struct snd_soc_dai_ops uni_reader_dai_ops = { .startup = uni_reader_startup, .shutdown = uni_reader_shutdown, .prepare = uni_reader_prepare, + .probe = sti_uniperiph_dai_probe, .trigger = uni_reader_trigger, .hw_params = sti_uniperiph_dai_hw_params, .set_fmt = sti_uniperiph_dai_set_fmt, --- base-commit: 0c3836482481200ead7b416ca80c68a29cfdaabd change-id: 20240721-sti-audio-fix-1374fffb24e7 Best regards, -- Jerome Audu <jau(a)free.fr>

2 1

[RESEND] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 29 Jul '24

29 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

2 1

[PATCH] ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

The wsa_macro_vi_feed_mixer_put() callback for setting VI feedback mixer value could be used for different DAIs (planned in the future CPS DAI), so make the code a bit more generic by using DAI ID from widget->shift, instead of hard-coding it. The get() callback already follows such convention. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-wsa-macro.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 8351f0974e6a..4158657914fb 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2296,36 +2296,37 @@ static int wsa_macro_vi_feed_mixer_put(struct snd_kcontrol *kcontrol, struct wsa_macro *wsa = snd_soc_component_get_drvdata(component); u32 enable = ucontrol->value.integer.value[0]; u32 spk_tx_id = mixer->shift; + u32 dai_id = widget->shift; if (enable) { if (spk_tx_id == WSA_MACRO_TX0 && !test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } if (spk_tx_id == WSA_MACRO_TX1 && !test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { set_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]++; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]++; } } else { if (spk_tx_id == WSA_MACRO_TX0 && test_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX0, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } if (spk_tx_id == WSA_MACRO_TX1 && test_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI])) { + &wsa->active_ch_mask[dai_id])) { clear_bit(WSA_MACRO_TX1, - &wsa->active_ch_mask[WSA_MACRO_AIF_VI]); - wsa->active_ch_cnt[WSA_MACRO_AIF_VI]--; + &wsa->active_ch_mask[dai_id]); + wsa->active_ch_cnt[dai_id]--; } } snd_soc_dapm_mixer_update_power(widget->dapm, kcontrol, enable, NULL); -- 2.43.0

2 1

[PATCH] ASoC: cs42l42: Convert comma to semicolon
by Chen Ni 29 Jul '24

29 Jul '24

Replace a comma between expression statements by a semicolon. Fixes: 90f6a2a20bd2 ("ASoC: cs42l42: Add SoundWire support") Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/codecs/cs42l42-sdw.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 94a66a325303..29891c1f6bec 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -323,15 +323,15 @@ static int cs42l42_sdw_read_prop(struct sdw_slave *peripheral) prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - capture */ - ports[0].num = CS42L42_SDW_CAPTURE_PORT, - ports[0].type = SDW_DPN_FULL, - ports[0].ch_prep_timeout = 10, + ports[0].num = CS42L42_SDW_CAPTURE_PORT; + ports[0].type = SDW_DPN_FULL; + ports[0].ch_prep_timeout = 10; prop->src_dpn_prop = &ports[0]; /* DP2 - playback */ - ports[1].num = CS42L42_SDW_PLAYBACK_PORT, - ports[1].type = SDW_DPN_FULL, - ports[1].ch_prep_timeout = 10, + ports[1].num = CS42L42_SDW_PLAYBACK_PORT; + ports[1].type = SDW_DPN_FULL; + ports[1].ch_prep_timeout = 10; prop->sink_dpn_prop = &ports[1]; return 0; -- 2.25.1

4 3

[PATCH 0/4] ASoC: codecs: wsa88xx: Few cleanups
by Krzysztof Kozlowski 29 Jul '24

29 Jul '24

Hi, Few cleanups around wsa88xx codecs. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant sound/soc/codecs/wsa881x.c | 2 -- sound/soc/codecs/wsa883x.c | 33 ++++++++++++++++++--------------- sound/soc/codecs/wsa884x.c | 5 ++--- 3 files changed, 20 insertions(+), 20 deletions(-) --- base-commit: 0b58e108042b0ed28a71cd7edf5175999955b233 change-id: 20240710-asoc-wsa88xx-version-dd0039265245 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 5

Re: [PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by edmund.raile 29 Jul '24

29 Jul '24

> Thank you for your sending the revised patches, it looks better than the > previous one. However, I have an additional request. Allright, patch v3 it is. > [1] https://git-scm.com/docs/git-revert Should have known git has something like that, how handy! > $ git revert -s b5b519965c4c Yes, 5b5 can be removed via revert, but what is the difference in effect? Just time saving? > $ git revert -s 7ba5ca32fe6e This one I'd like to ask you about: The original inline comment in amdtp-stream.c amdtp_domain_stream_pcm_pointer() ``` // This function is called in software IRQ context of // period_work or process context. // // When the software IRQ context was scheduled by software IRQ // context of IT contexts, queued packets were already handled. // Therefore, no need to flush the queue in buffer furthermore. // // When the process context reach here, some packets will be // already queued in the buffer. These packets should be handled // immediately to keep better granularity of PCM pointer. // // Later, the process context will sometimes schedules software // IRQ context of the period_work. Then, no need to flush the // queue by the same reason as described in the above ``` (let's call the above v1) was replaced with ``` // In software IRQ context, the call causes dead-lock to disable the tasklet // synchronously. ``` on occasion of 7ba5ca32fe6e (let's call this v2). I sought to replace it with ``` // use wq to prevent deadlock between process context spin_lock // of snd_pcm_stream_lock_irq() in snd_pcm_status64() and // softIRQ context spin_lock of snd_pcm_stream_lock_irqsave() // in snd_pcm_period_elapsed() ``` to prevent this issue from occurring again (let's call this v3). Should I include v1, v3 or a combination of v1 and v3 in my next patch? > Just for safe, it is preferable to execute 'scripts/checkpatch.pl' in > kernel tree to check the patchset generated by send-email subcommand[3]. Absolutely should have done so, sorry. Thank you for your patience and guidance, Edmund Raile.

2 1

[PATCH AUTOSEL 5.15 09/13] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 089b6c7994f9a..cfa0c3eaffea7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -258,6 +258,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 10/15] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index cf501baf6f143..9cd85ab19c553 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -387,6 +387,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2 | SOF_SDW_FOUR_SPK), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 09/15] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d03de37e3578c..cf501baf6f143 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -267,6 +267,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 11/17] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index dc144cd7e0e3b..db1dcb9d70466 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -425,6 +425,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 10/17] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 5980fce817976..dc144cd7e0e3b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -286,6 +286,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 15/23] ASoC: Intel: sof_sdw: add quirk for Dell SKU 0B8C
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 92d5b5930e7d55ca07b483490d6298eee828bbe4 ] Jack detection needs to rely on JD2, as most other Dell AlderLake-based devices. Closes: https://github.com/thesofproject/linux/issues/5021 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-4-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 965d46968f95a..0b3b2c88d31ed 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -406,6 +406,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { /* No Jack */ .driver_data = (void *)SOF_SDW_TGL_HDMI, }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0B8C"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 14/23] ASoC: Intel: sof_sdw: fix jack detection on ADL-N variant RVP
by Sasha Levin 28 Jul '24

28 Jul '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 65c90df918205bc84f5448550cde76a54dae5f52 ] Experimental tests show that JD2_100K is required, otherwise the jack is detected always even with nothing plugged-in. To avoid matching with other known quirks the SKU information is used. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240624121119.91552-2-pierre-louis.bossart@linux.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a878b6b3d1948..965d46968f95a 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -277,6 +277,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_BT_OFFLOAD_SSP(2) | SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_SKU, "0000000000070000"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2_100K), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 02/23] ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops
by Sasha Levin 28 Jul '24

28 Jul '24

From: Charles Keepax <ckeepax(a)opensource.cirrus.com> [ Upstream commit 91cdecaba791c74df6da0650e797fe1192cf2700 ] Add quirks for some new Dell laptops using Cirrus amplifiers in a bridge configuration. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://msgid.link/r/20240527193552.165567-11-pierre-louis.bossart@linux.in… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index e41b0d95e0ff7..a878b6b3d1948 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -505,6 +505,22 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE3") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0CE4") + }, + .driver_data = (void *)(SOF_SIDECAR_AMPS), + }, {} }; -- 2.43.0

1 0

[PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 28 Jul '24

28 Jul '24

This patchset serves to prevent a deadlock between process context and softIRQ context: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): ALSA: firewire-lib: restore workqueue for process context ALSA: firewire-lib: prevent deadlock between process and softIRQ context sound/firewire/amdtp-stream.c | 36 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 24 insertions(+), 13 deletions(-) -- 2.45.2

1 2

[PATCH v2 0/2] ALSA: firewire-lib: restore process context workqueue to prevent deadlock
by Edmund Raile 28 Jul '24

28 Jul '24

This patchset serves to prevent a deadlock between process context and softIRQ context: A. In the process context * (lock A) Acquiring spin_lock by snd_pcm_stream_lock_irq() in snd_pcm_status64() * (lock B) Then attempt to enter tasklet B. In the softIRQ context * (lock B) Enter tasklet * (lock A) Attempt to acquire spin_lock by snd_pcm_stream_lock_irqsave() in snd_pcm_period_elapsed() ? tasklet_unlock_spin_wait </NMI> <TASK> ohci_flush_iso_completions firewire_ohci amdtp_domain_stream_pcm_pointer snd_firewire_lib snd_pcm_update_hw_ptr0 snd_pcm snd_pcm_status64 snd_pcm ? native_queued_spin_lock_slowpath </NMI> <IRQ> _raw_spin_lock_irqsave snd_pcm_period_elapsed snd_pcm process_rx_packets snd_firewire_lib irq_target_callback snd_firewire_lib handle_it_packet firewire_ohci context_tasklet firewire_ohci The issue has been reported as a regression of kernel 5.14: Link: https://lore.kernel.org/regressions/kwryofzdmjvzkuw6j3clftsxmoolynljztxqwg7… ("[REGRESSION] ALSA: firewire-lib: snd_pcm_period_elapsed deadlock with Fireface 800") Commit 7ba5ca32fe6e ("ALSA: firewire-lib: operate for period elapse event in process context") removed the process context workqueue from amdtp_domain_stream_pcm_pointer() and update_pcm_pointers() to remove its overhead. Commit b5b519965c4c ("ALSA: firewire-lib: obsolete workqueue for period update") belongs to the same patch series and removed the now-unused workqueue entirely. Though being observed on RME Fireface 800, this issue would affect all Firewire audio interfaces using ohci amdtp + pcm streaming. ALSA streaming, especially under intensive CPU load will reveal this issue the soonest due to issuing more hardIRQs, with time to occurrence ranging from 2 secons to 30 minutes after starting playback. to reproduce the issue: direct ALSA playback to the device: mpv --audio-device=alsa/sysdefault:CARD=Fireface800 Spor-Ignition.flac Time to occurrence: 2s to 30m Likelihood increased by: - high CPU load stress --cpu $(nproc) - switching between applications via workspaces tested with i915 in Xfce PulsaAudio / PipeWire conceal the issue as they run PCM substream without period wakeup mode, issuing less hardIRQs. Backport note: Also applies to and fixes on (tested): 6.10.2, 6.9.12, 6.6.43, 6.1.102, 5.15.164 Edmund Raile (2): ALSA: firewire-lib: restore workqueue for process context ALSA: firewire-lib: prevent deadlock between process and softIRQ context sound/firewire/amdtp-stream.c | 36 ++++++++++++++++++++++------------- sound/firewire/amdtp-stream.h | 1 + 2 files changed, 24 insertions(+), 13 deletions(-) -- 2.45.2

1 2

[PATCH v2 0/4] MSM8953/MSM8976 ASoC support
by Adam Skladowski 28 Jul '24

28 Jul '24

Introduce support for basic sound card setup on MSM8953/MSM8976 platforms, document new compatibles and introduce support for more dais. Most of code is sourced from msm8953-mainline fork over github with some changes implemented by me,some basic changes are mentioned in each patch. [1] - https://lore.kernel.org/lkml/20240723083300.35605-1-krzysztof.kozlowski@lin… Changes since v1 ================ 1. Rebased dt-bindings documentation based on Krzysztof's split patch[1] 2. Resolved clang errors by guarding ret verification inside if 3. Switched quin-iomux to devm_ioremap_resource to not fail on msm8916 Adam Skladowski (1): ASoC: dt-bindings: apq8016-sbc: Add msm8953/msm8976-qdsp6-sndcard Vladimir Lypak (3): ASoC: qcom: apq8016_sbc.c: Add Quinary support ASoC: msm8916-wcd-analog: add cajon and cajon v2 support ASoC: qcom: apq8016_sbc: Add support for msm8953/msm8976 SoC .../sound/qcom,apq8016-sbc-sndcard.yaml | 51 +++++++++-- sound/soc/codecs/msm8916-wcd-analog.c | 63 +++++++++++++- sound/soc/qcom/apq8016_sbc.c | 84 ++++++++++++++++++- 3 files changed, 186 insertions(+), 12 deletions(-) -- 2.45.2

3 9

[PATCH v23 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 26 Jul '24

26 Jul '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Wesley Cheng (32): ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 603 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/xhci-plat.c | 2 + .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 188 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 417 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 684 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1915 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 5705 insertions(+), 29 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

4 59

[PATCH 0/2] {ASoC,media}: constify snd_soc_component_driver struct
by Javier Carrasco 26 Jul '24

26 Jul '24

The `snd_soc_component_driver` struct is never modified after its declaration, and its only direct user `devm_snd_soc_register_component()` expects a constant value anyway. Declare `snd_soc_component_driver` as const to move their declarations to read-only sections. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/cs43130.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/sti-sas.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 14 files changed, 15 insertions(+), 15 deletions(-) --- base-commit: 864b1099d16fc7e332c3ad7823058c65f890486c change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

3 6

[GIT PULL] ASoC fixes for v6.11-merge-window
by Mark Brown 25 Jul '24

25 Jul '24

The following changes since commit c51cba4755609ad97ba97713210c16f043c73224: Fix the unbalanced pm_runtime_enable in wcd937x-sdw (2024-07-12 17:38:18 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-merge-window for you to fetch changes up to ab53dfdcdd1ec8df8729890aefa5b0e3c900afbb: ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components (2024-07-25 14:20:30 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A selection of routine fixes and quirks that came in since the merge window. The fsl-asoc-card change is a fix for systems with multiple cards where updating templates in place leaks data from one card to another. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: tegra: select CONFIG_SND_SIMPLE_CARD_UTILS Curtis Malainey (1): ASoC: Intel: Fix RT5650 SSP lookup Dan Carpenter (1): ASoC: TAS2781: Fix tasdev_load_calibrated_data() Daniel Baluta (1): ASoC: SOF: imx8m: Fix DSP control regmap retrieval Peter Ujfalusi (2): ASoC: SOF: ipc4-topology: Only handle dai_config with HW_PARAMS for ChainDMA ASoC: SOF: ipc4-topology: Preserve the DMA Link ID for ChainDMA on unprepare Pierre-Louis Bossart (2): ASOC: SOF: Intel: hda-loader: only wait for HDaudio IOC for IPC4 devices ASoC: Intel: use soc_intel_is_byt_cr() only when IOSF_MBI is reachable Shengjiu Wang (1): ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components Takashi Iwai (1): ASoC: amd: yc: Support mic on Lenovo Thinkpad E16 Gen 2 Venkata Prasad Potturu (1): ASoC: sof: amd: fix for firmware reload failure in Vangogh platform sound/soc/amd/yc/acp6x-mach.c | 7 ++++ sound/soc/codecs/tas2781-fmwlib.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 46 ++++++++++++++-------- sound/soc/intel/common/soc-acpi-intel-ssp-common.c | 9 +++++ sound/soc/intel/common/soc-intel-quirks.h | 2 +- sound/soc/sof/amd/pci-vangogh.c | 1 - sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/intel/hda-loader.c | 20 ++++++---- sound/soc/sof/intel/hda.c | 17 ++++---- sound/soc/sof/ipc4-topology.c | 18 +++++++-- sound/soc/tegra/Kconfig | 1 + 11 files changed, 85 insertions(+), 40 deletions(-)

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[REGRESSION] ALSA: firewire-lib: heavy digital distortion with Fireface 800
by edmund.raile 25 Jul '24

25 Jul '24

Bisection revealed that the bitcrushing distortion with RME FireFace 800 was caused by 1d717123bb1a7555 ("ALSA: firewire-lib: Avoid -Wflex-array-member-not-at-end warning"). Reverting this commit yields restoration of clear audio output. I will send in a patch reverting this commit for now, soonTM. #regzbot introduced: 1d717123bb1a7555

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[PATCH] ASoC: fsl-asoc-card: Dynamically allocate memory for snd_soc_dai_link_components
by Shengjiu Wang 25 Jul '24

25 Jul '24

The static snd_soc_dai_link_components cause conflict for multiple instances of this generic driver. For example, when there is wm8962 and SPDIF case enabled together, the contaminated snd_soc_dai_link_components will cause another device probe fail. Fixes: 6d174cc4f224 ("ASoC: fsl-asoc-card: merge spdif support from imx-spdif.c") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 46 ++++++++++++++++++++++------------- 1 file changed, 29 insertions(+), 17 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 82df887b3af5..f6c3aeff0d8e 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -306,27 +306,12 @@ static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, return 0; } -SND_SOC_DAILINK_DEFS(hifi, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(hifi_fe, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_DUMMY()), - DAILINK_COMP_ARRAY(COMP_EMPTY())); - -SND_SOC_DAILINK_DEFS(hifi_be, - DAILINK_COMP_ARRAY(COMP_EMPTY()), - DAILINK_COMP_ARRAY(COMP_EMPTY(), COMP_EMPTY())); - static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { /* Default ASoC DAI Link*/ { .name = "HiFi", .stream_name = "HiFi", .ops = &fsl_asoc_card_ops, - SND_SOC_DAILINK_REG(hifi), }, /* DPCM Link between Front-End and Back-End (Optional) */ { @@ -335,7 +320,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { .dpcm_playback = 1, .dpcm_capture = 1, .dynamic = 1, - SND_SOC_DAILINK_REG(hifi_fe), }, { .name = "HiFi-ASRC-BE", @@ -345,7 +329,6 @@ static const struct snd_soc_dai_link fsl_asoc_card_dai[] = { .dpcm_playback = 1, .dpcm_capture = 1, .no_pcm = 1, - SND_SOC_DAILINK_REG(hifi_be), }, }; @@ -637,6 +620,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev[2] = { NULL, NULL }; + struct snd_soc_dai_link_component *dlc; const char *codec_dai_name[2]; const char *codec_dev_name[2]; u32 asrc_fmt = 0; @@ -717,7 +701,35 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) memcpy(priv->dai_link, fsl_asoc_card_dai, sizeof(struct snd_soc_dai_link) * ARRAY_SIZE(priv->dai_link)); + /* + * "Default ASoC DAI Link": 1 cpus, 2 codecs, 1 platforms + * "DPCM Link Front-End": 1 cpus, 1 codecs (dummy), 1 platforms + * "DPCM Link Back-End": 1 cpus, 2 codecs + * totally 10 components + */ + dlc = devm_kcalloc(&pdev->dev, 10, sizeof(*dlc), GFP_KERNEL); + if (!dlc) { + ret = -ENOMEM; + goto asrc_fail; + } + + priv->dai_link[0].cpus = &dlc[0]; + priv->dai_link[0].num_cpus = 1; + priv->dai_link[0].codecs = &dlc[1]; priv->dai_link[0].num_codecs = 1; + priv->dai_link[0].platforms = &dlc[3]; + priv->dai_link[0].num_platforms = 1; + + priv->dai_link[1].cpus = &dlc[4]; + priv->dai_link[1].num_cpus = 1; + priv->dai_link[1].codecs = &dlc[5]; + priv->dai_link[1].num_codecs = 0; /* dummy */ + priv->dai_link[1].platforms = &dlc[6]; + priv->dai_link[1].num_platforms = 1; + + priv->dai_link[2].cpus = &dlc[7]; + priv->dai_link[2].num_cpus = 1; + priv->dai_link[2].codecs = &dlc[8]; priv->dai_link[2].num_codecs = 1; priv->card.dapm_routes = audio_map; -- 2.34.1

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[PATCH v5 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 25 Jul '24

25 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code Changes in v5: - Move iio_backend_read_scale and iio_backend_read_offset to inline and update prototype - Fix channel number & remove legacy binding example in DFSDM binding - Rearrange DFSDM patches to fix build issue in serie - Split SD modulator binding patch. Move reference voltage change in a separate patch Olivier Moysan (9): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator dt-bindings: iio: add vref support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 25 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 124 +++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 293 +++++++++++++++--- drivers/iio/industrialio-backend.c | 101 +++++- include/linux/iio/backend.h | 29 +- 7 files changed, 593 insertions(+), 73 deletions(-) base-commit: 472438c7e0e2261c6737a8321f46ef176eef1c8f -- 2.25.1

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[RFC v2 00/xx] ALSA: ALSA: add snd_pcm_is_playback/capture()
by Kuninori Morimoto 25 Jul '24

25 Jul '24

Hi Iwai-san, Mark Cc Sakamoto-san, Amadeusz, Pierre-Louis It seems we can use _Generic() more simply, v2 RFC is using it. I'm not 100% sure but unfortunately we can't use bit-field with _Generic() in gcc (clang seems possible to handle it ?). pci/ac97 is the only user of bit-field direction. To avoid posting patch-bomb, I will post main patch and some sample patches as RFC. Please review it. Link: https://lore.kernel.org/r/87zfqel1g7.wl-kuninori.morimoto.gx@renesas.com Thank you for your help !! Best regards --- Kuninori Morimoto

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[kernel][usb audio class 1] surround channels mapped to wrong alsa channel ids
by GitHub issues - opened 25 Jul '24

25 Jul '24

alsa-project/alsa-lib issue #403 was opened from sylware: In $linux_src/sound/usb/stream.c, function "convert_chmap", in the "uac1_maps" array, the "left surround" should be SNDRV_CHMAP_RL instead of SNDRV_CHMAP_SL, and the "right surround" should be SNDRV_CHMAP_RR instead of SNDDRV_CHMAP_SR. Issue URL : https://github.com/alsa-project/alsa-lib/issues/403 Repository URL: https://github.com/alsa-project/alsa-lib

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