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

[PATCH 0/5] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 08 Aug '24

08 Aug '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support" Bard Liao (2): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: SOF: Intel: add initial support for PTL ASoC: Intel: soc-acpi: add PTL match tables include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 121 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 +++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 ++- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 +++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 266 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

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Re: [PATCH v11 07/10] mtd: spi-nor: Add stacked memories support in spi-nor
by Michael Walle 07 Aug '24

07 Aug '24

Hi, > Based on the inputs/suggestions from Tudor, i am planning to add a new > layer between the SPI-NOR and MTD layers to support stacked and parallel > configurations. This new layer will be part of the spi-nor and located in > mtd/spi-nor/ Will AMD submit to maintain this layer? What happens if the maintainer will leave AMD? TBH, personally, I don't like to maintain such a niche feature. I'd really like to see some use cases and performance reports for this, like actual boards (and no evaluation boards don't count). Why wouldn't someone just use an octal flash? And as already mentioned there is also mtdcat, which seems to duplicate some features? -michael

5 13

[PATCH] ASoC: remove bespoke trigger support
by Kuninori Morimoto 06 Aug '24

06 Aug '24

Bespoke trigger support was added when Linux v3.5 by this patch. commit 07bf84aaf736781a283b1bd36eaa911453b14574 ("ASoC: dpcm: Add bespoke trigger()") test-component driver is using it, but this is because it indicates used function for debug/trace purpose. Except test-component driver, bespoke trigger has never been used over 10 years in upstream. We can re-support it if needed in the future, but let's remove it for now, because it just noise in upstream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 5 +--- include/sound/soc-dpcm.h | 1 - sound/soc/generic/test-component.c | 9 ------ sound/soc/soc-dai.c | 20 ------------- sound/soc/soc-pcm.c | 47 +++--------------------------- 5 files changed, 5 insertions(+), 77 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index bbb72ad4c9518..ab4e109fe71d2 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -240,8 +240,6 @@ int snd_soc_pcm_dai_new(struct snd_soc_pcm_runtime *rtd); int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream); int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, int cmd, int rollback); -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd); void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay); @@ -345,8 +343,7 @@ struct snd_soc_dai_ops { */ int (*trigger)(struct snd_pcm_substream *, int, struct snd_soc_dai *); - int (*bespoke_trigger)(struct snd_pcm_substream *, int, - struct snd_soc_dai *); + /* * For hardware based FIFO caused delay reporting. * Optional. diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index ebd24753dd000..773f2db8c31c8 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -58,7 +58,6 @@ enum snd_soc_dpcm_state { enum snd_soc_dpcm_trigger { SND_SOC_DPCM_TRIGGER_PRE = 0, SND_SOC_DPCM_TRIGGER_POST, - SND_SOC_DPCM_TRIGGER_BESPOKE, }; /* diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index e9e5e235a8a65..df2487b700cca 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -181,14 +181,6 @@ static int test_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct return 0; } -static int test_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - mile_stone(dai); - - return 0; -} - static const u64 test_dai_formats = /* * Select below from Sound Card, not auto @@ -228,7 +220,6 @@ static const struct snd_soc_dai_ops test_verbose_ops = { .hw_params = test_dai_hw_params, .hw_free = test_dai_hw_free, .trigger = test_dai_trigger, - .bespoke_trigger = test_dai_bespoke_trigger, .auto_selectable_formats = &test_dai_formats, .num_auto_selectable_formats = 1, }; diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 9e47053419c16..302ca753a1f35 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -685,26 +685,6 @@ int snd_soc_pcm_dai_trigger(struct snd_pcm_substream *substream, return ret; } -int snd_soc_pcm_dai_bespoke_trigger(struct snd_pcm_substream *substream, - int cmd) -{ - struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); - struct snd_soc_dai *dai; - int i, ret; - - for_each_rtd_dais(rtd, i, dai) { - if (dai->driver->ops && - dai->driver->ops->bespoke_trigger) { - ret = dai->driver->ops->bespoke_trigger(substream, - cmd, dai); - if (ret < 0) - return soc_dai_ret(dai, ret); - } - } - - return 0; -} - void snd_soc_pcm_dai_delay(struct snd_pcm_substream *substream, snd_pcm_sframes_t *cpu_delay, snd_pcm_sframes_t *codec_delay) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 4571785736f76..5646525ad7924 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2371,14 +2371,6 @@ static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd) break; } break; - case SND_SOC_DPCM_TRIGGER_BESPOKE: - /* bespoke trigger() - handles both FE and BEs */ - - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd %d\n", - fe->dai_link->name, cmd); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, cmd); - break; default: dev_err(fe->dev, "ASoC: invalid trigger cmd %d for %s\n", cmd, fe->dai_link->name); @@ -2508,26 +2500,12 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int err; dev_dbg(fe->dev, "ASoC: runtime %s close on FE %s\n", stream ? "capture" : "playback", fe->dai_link->name); - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call bespoke trigger - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_STOP); - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd stop\n", - fe->dai_link->name); - - err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); - } + err = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_STOP); dpcm_be_dai_hw_free(fe, stream); @@ -2541,10 +2519,7 @@ static int dpcm_run_update_shutdown(struct snd_soc_pcm_runtime *fe, int stream) static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) { - struct snd_pcm_substream *substream = - snd_soc_dpcm_get_substream(fe, stream); struct snd_soc_dpcm *dpcm; - enum snd_soc_dpcm_trigger trigger = fe->dai_link->trigger[stream]; int ret = 0; dev_dbg(fe->dev, "ASoC: runtime %s open on FE %s\n", @@ -2588,23 +2563,9 @@ static int dpcm_run_update_startup(struct snd_soc_pcm_runtime *fe, int stream) fe->dpcm[stream].state == SND_SOC_DPCM_STATE_STOP) return 0; - if (trigger == SND_SOC_DPCM_TRIGGER_BESPOKE) { - /* call trigger on the frontend - FE takes care of all BE triggers */ - dev_dbg(fe->dev, "ASoC: bespoke trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = snd_soc_pcm_dai_bespoke_trigger(substream, SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } else { - dev_dbg(fe->dev, "ASoC: trigger FE %s cmd start\n", - fe->dai_link->name); - - ret = dpcm_be_dai_trigger(fe, stream, - SNDRV_PCM_TRIGGER_START); - if (ret < 0) - goto hw_free; - } + ret = dpcm_be_dai_trigger(fe, stream, SNDRV_PCM_TRIGGER_START); + if (ret < 0) + goto hw_free; return 0; -- 2.43.0

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[PATCH v6 0/9] iio: adc: dfsdm: add scaling support
by Olivier Moysan 06 Aug '24

06 Aug '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 Changes in v6: - Minor cleanup in DFSDM binding - Fix vref-supply dependency condition in SD modulator binding 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 | 122 +++++++- 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, 592 insertions(+), 72 deletions(-) base-commit: bf829a98bc4bb9076a2fd70f4a69ea7e281863fa -- 2.25.1

2 3

[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

4 12

[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

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[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

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[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

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[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

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