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

[PATCH 00/12] ASoC: Intel: updates for 6.10 - part5
by Pierre-Louis Bossart 01 May '24

01 May '24

This patchset corrects a couple of mistakes corrected, improves snd_soc_card allocation. The new functionality is mostly for SoundWire platforms, with new SKUs for Dell and Acer, and support for the Cirrus Logic bridge/sidecar amplifier topology. Bard Liao (1): ASoC: Intel: sof_sdw: add a space before cfg-amp in components Charles Keepax (2): ASoC: Intel: sof_sdw: Delay update of the codec_conf array ASoC: Intel: sof_sdw: Add callbacks to register sidecar devices Mac Chiang (1): ASoC: Intel: soc-acpi-intel-lnl-match: adds RT714 and RT1318 support Maciej Strozek (1): ASoC: intel: sof_sdw: Add support for cs42l43-cs35l56 sidecar amps Peter Ujfalusi (3): ASoC: Intel: skl_hda_dsp_generic: Allocate snd_soc_card dynamically ASoC: Intel: skl_hda_dsp_generic: Use devm_kasprintf for the components string ASoC: Intel: sof_sdw: Allocate snd_soc_card dynamically Pierre-Louis Bossart (4): ASoC: Intel: soc-acpi: mtl: add Dell SKU 0C64 and 0CC6 ASoC: Intel: soc-acpi: mtl: add support for Acer Swift Go 14 ASoC: Intel: sof-sdw: don't set card long_name ASoC: Intel: sof-sdw: really remove FOUR_SPEAKER quirk sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/Makefile | 1 + sound/soc/intel/boards/bridge_cs35l56.c | 137 ++++++++++++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 1 + sound/soc/intel/boards/skl_hda_dsp_generic.c | 50 +++---- sound/soc/intel/boards/sof_sdw.c | 106 ++++++++------ sound/soc/intel/boards/sof_sdw_common.h | 27 ++++ sound/soc/intel/boards/sof_sdw_cs42l43.c | 14 +- .../intel/common/soc-acpi-intel-lnl-match.c | 52 +++++++ .../intel/common/soc-acpi-intel-mtl-match.c | 83 ++++++++++- 10 files changed, 397 insertions(+), 75 deletions(-) create mode 100644 sound/soc/intel/boards/bridge_cs35l56.c -- 2.40.1

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[PATCH 0/8] ASoC: Intel: avs: PCM code cleanup
by Cezary Rojewski 01 May '24

01 May '24

A set of changes that aims to improve readability of cohesiveness of the pcm code for the avs-driver. Start off with a change that synchronizes DAI open/close - DAIs are started up in ascending order yet their shutdown does not follow the scheme - it is done in the ascending order too, rather than desceding one. This patch is a dependency for the next one in line. To align the HDAudio DAI startup/shutdown with the non-HDAudio equivalents, relocate the code from component to DAI. The reason above is a dependency stems from codec driver requirements - HDAudio code found in sound/pci/hda/ expects substream->runtime->private_data to point to a valid stream (HOST) pointer. With the hard part done, the follow up changes update the existing code to reduce it is complexity - removal of duplicates, renaming of ambiguous functions and adding new fields to DAI-data object so that the number of local variables and casts is reduced. Cezary Rojewski (8): ASoC: pcm: Reverse iterate DAIs when shutting them down ASoC: Intel: avs: Relocate HDA BE DAI specific operations ASoC: Intel: avs: Remove redundancy around DAI shutdown ASoC: Intel: avs: Store pointer to adev in DAI dma_data ASoC: Intel: avs: Remove redundancy around DAI startup ASoC: Intel: avs: Remove redundancy around DAI prepare ASoC: Intel: avs: Store pointer to link_stream in dma_data ASoC: Intel: avs: Clean up hw constraints initialization include/sound/soc.h | 4 + sound/soc/intel/avs/pcm.c | 250 +++++++++++++++++--------------------- sound/soc/soc-pcm.c | 2 +- 3 files changed, 117 insertions(+), 139 deletions(-) -- 2.25.1

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[PATCH 1/1] ALSA: aoa: soundbus: i2sbus: pcm: use 'time_left' variable with wait_for_completion_timeout()
by Wolfram Sang 01 May '24

01 May '24

There is a confusing pattern in the kernel to use a variable named 'timeout' to store the result of wait_for_completion_timeout() 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 self explaining. Fix to the proper variable type 'unsigned long' while here. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- sound/aoa/soundbus/i2sbus/pcm.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 07df5cc0f2d7..98b812ffbde6 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -255,24 +255,24 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, { unsigned long flags; DECLARE_COMPLETION_ONSTACK(done); - long timeout; + unsigned long time_left; spin_lock_irqsave(&i2sdev->low_lock, flags); if (pi->dbdma_ring.stopping) { pi->stop_completion = &done; spin_unlock_irqrestore(&i2sdev->low_lock, flags); - timeout = wait_for_completion_timeout(&done, HZ); + time_left = wait_for_completion_timeout(&done, HZ); spin_lock_irqsave(&i2sdev->low_lock, flags); pi->stop_completion = NULL; - if (timeout == 0) { + if (time_left == 0) { /* timeout expired, stop dbdma forcefully */ printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n"); /* make sure RUN, PAUSE and S0 bits are cleared */ out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); pi->dbdma_ring.stopping = 0; - timeout = 10; + time_left = 10; while (in_le32(&pi->dbdma->status) & ACTIVE) { - if (--timeout <= 0) + if (--time_left <= 0) break; udelay(1); } -- 2.43.0

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topology: Add SND_PCM_RATE_24000 for 24 kHz rate support
by GitHub pull_request - opened 30 Apr '24

30 Apr '24

alsa-project/alsa-lib pull request #396 was opened from yongzhi1: The Low Complexity Communication Codec (LC3) supports 24 kHz sample rate, so add SND_PCM_RATE_24000 to the supported rates. Request URL : https://github.com/alsa-project/alsa-lib/pull/396 Patch URL : https://github.com/alsa-project/alsa-lib/pull/396.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v2] ASoC: sunxi: DMIC: Add controls for adjusting the mic gains
by Joao Schim 30 Apr '24

30 Apr '24

The AllWinner H6 and later SoCs that sport a DMIC block contain a set of registers to control the gain (left + right) of each of the four supported channels. Add ASoC controls for changing each of the stereo channel gains using alsamixer and alike Signed-off-by: Joao Schim <joao(a)schimsalabim.eu> --- sound/soc/sunxi/sun50i-dmic.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index c76628bc86c6..dd32780fb6a4 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -14,6 +14,7 @@ #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/tlv.h> #define SUN50I_DMIC_EN_CTL (0x00) #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) @@ -43,6 +44,17 @@ #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) #define SUN50I_DMIC_CNT (0x2c) #define SUN50I_DMIC_CNT_N (1 << 0) +#define SUN50I_DMIC_D0D1_VOL_CTR (0x30) + #define SUN50I_DMIC_D0D1_VOL_CTR_0R (0) + #define SUN50I_DMIC_D0D1_VOL_CTR_0L (8) + #define SUN50I_DMIC_D0D1_VOL_CTR_1R (16) + #define SUN50I_DMIC_D0D1_VOL_CTR_1L (24) +#define SUN50I_DMIC_D2D3_VOL_CTR (0x34) + #define SUN50I_DMIC_D2D3_VOL_CTR_2R (0) + #define SUN50I_DMIC_D2D3_VOL_CTR_2L (8) + #define SUN50I_DMIC_D2D3_VOL_CTR_3R (16) + #define SUN50I_DMIC_D2D3_VOL_CTR_3L (24) + #define SUN50I_DMIC_HPF_CTRL (0x38) #define SUN50I_DMIC_VERSION (0x50) @@ -273,8 +285,30 @@ static const struct of_device_id sun50i_dmic_of_match[] = { }; MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(sun50i_dmic_vol_scale, -12000, 75, 1); + +static const struct snd_kcontrol_new sun50i_dmic_controls[] = { + + SOC_DOUBLE_TLV("DMIC Channel 0 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_0L, SUN50I_DMIC_D0D1_VOL_CTR_0R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 1 Capture Volume", SUN50I_DMIC_D0D1_VOL_CTR, + SUN50I_DMIC_D0D1_VOL_CTR_1L, SUN50I_DMIC_D0D1_VOL_CTR_1R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 2 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_2L, SUN50I_DMIC_D2D3_VOL_CTR_2R, + 0xFF, 0, sun50i_dmic_vol_scale), + SOC_DOUBLE_TLV("DMIC Channel 3 Capture Volume", SUN50I_DMIC_D2D3_VOL_CTR, + SUN50I_DMIC_D2D3_VOL_CTR_3L, SUN50I_DMIC_D2D3_VOL_CTR_3R, + 0xFF, 0, sun50i_dmic_vol_scale), + + +}; + static const struct snd_soc_component_driver sun50i_dmic_component = { .name = "sun50i-dmic", + .controls = sun50i_dmic_controls, + .num_controls = ARRAY_SIZE(sun50i_dmic_controls), }; static int sun50i_dmic_runtime_suspend(struct device *dev) -- 2.25.1

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[PATCH] ASoC: da7219-aad: fix usage of device_get_named_child_node()
by Pierre-Louis Bossart 30 Apr '24

30 Apr '24

The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid a leaked reference. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/da7219-aad.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c index 6bc068cdcbe2..15e5e3eb592b 100644 --- a/sound/soc/codecs/da7219-aad.c +++ b/sound/soc/codecs/da7219-aad.c @@ -671,8 +671,10 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) return NULL; aad_pdata = devm_kzalloc(dev, sizeof(*aad_pdata), GFP_KERNEL); - if (!aad_pdata) + if (!aad_pdata) { + fwnode_handle_put(aad_np); return NULL; + } aad_pdata->irq = i2c->irq; @@ -753,6 +755,8 @@ static struct da7219_aad_pdata *da7219_aad_fw_to_pdata(struct device *dev) else aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1; + fwnode_handle_put(aad_np); + return aad_pdata; } -- 2.40.1

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[PATCH] ASoC: cs35l56: fix usages of device_get_named_child_node()
by Pierre-Louis Bossart 30 Apr '24

30 Apr '24

The documentation for device_get_named_child_node() mentions this important point: " The caller is responsible for calling fwnode_handle_put() on the returned fwnode pointer. " Add fwnode_handle_put() to avoid leaked references. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/cs35l56.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index dfd703d9e12f..758dfdf9d3ea 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -1361,6 +1361,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 "spk-id-gpios", ACPI_TYPE_PACKAGE, &obj); if (ret) { dev_dbg(cs35l56->base.dev, "Could not get spk-id-gpios package: %d\n", ret); + fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1368,6 +1369,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 if (obj->package.count != 4) { dev_warn(cs35l56->base.dev, "Unexpected spk-id element count %d\n", obj->package.count); + fwnode_handle_put(af01_fwnode); return -ENOENT; } @@ -1382,6 +1384,7 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 */ ret = acpi_dev_add_driver_gpios(adev, cs35l56_af01_spkid_gpios_mapping); if (ret) { + fwnode_handle_put(af01_fwnode); return dev_err_probe(cs35l56->base.dev, ret, "Failed to add gpio mapping to AF01\n"); } @@ -1389,14 +1392,17 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 ret = devm_add_action_or_reset(cs35l56->base.dev, cs35l56_acpi_dev_release_driver_gpios, adev); - if (ret) + if (ret) { + fwnode_handle_put(af01_fwnode); return ret; + } dev_dbg(cs35l56->base.dev, "Added spk-id-gpios mapping to AF01\n"); } desc = fwnode_gpiod_get_index(af01_fwnode, "spk-id", 0, GPIOD_IN, NULL); if (IS_ERR(desc)) { + fwnode_handle_put(af01_fwnode); ret = PTR_ERR(desc); return dev_err_probe(cs35l56->base.dev, ret, "Get GPIO from AF01 failed\n"); } @@ -1405,9 +1411,12 @@ static int cs35l56_try_get_broken_sdca_spkid_gpio(struct cs35l56_private *cs35l5 gpiod_put(desc); if (ret < 0) { + fwnode_handle_put(af01_fwnode); dev_err_probe(cs35l56->base.dev, ret, "Error reading spk-id GPIO\n"); return ret; - } + } + + fwnode_handle_put(af01_fwnode); dev_info(cs35l56->base.dev, "Got spk-id from AF01\n"); -- 2.40.1

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[PATCH] ASoC: meson: cards: select SND_DYNAMIC_MINORS
by Jerome Brunet 30 Apr '24

30 Apr '24

Amlogic sound cards do create a lot of pcm interfaces, possibly more than 8. Some pcm interfaces are internal (like DPCM backends and c2c) and not exposed to userspace. Those interfaces still increase the number passed to snd_find_free_minor(), which eventually exceeds 8 causing -EBUSY error on card registration if CONFIG_SND_DYNAMIC_MINORS=n and the interface is exposed to userspace. select CONFIG_SND_DYNAMIC_MINORS for Amlogic cards to avoid the problem. Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- NOTE: This problem was found while toying with a minimal configuration which disabled CONFIG_SND_DYNAMIC_MINORS and DT overlays. Overlays prepend nodes instead of doing append. This changes the order in which the links are parsed by the card driver, which in turn changes the pcm numbers. The userpace pcm number then got over 8 triggering the problem. sound/soc/meson/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index b93ea33739f2..6458d5dc4902 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -99,6 +99,7 @@ config SND_MESON_AXG_PDM config SND_MESON_CARD_UTILS tristate + select SND_DYNAMIC_MINORS config SND_MESON_CODEC_GLUE tristate -- 2.43.0

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[DNM] common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - edited 30 Apr '24

30 Apr '24

alsa-project/alsa-ucm-conf pull request #413 was edited from ujfalusi: Hi, The PR will introduce a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[DNM] common: pcm: hdmi/sof-hda-dsp/sof-soundwire: Syntax 7 dependent macro to generate the hdmi device sections
by GitHub pull_request - opened 30 Apr '24

30 Apr '24

alsa-project/alsa-ucm-conf pull request #413 was opened from ujfalusi: Hi, The PR will introduce a a generic `HdmiDevice` macro, which can replace duplicated conditions in sof UCM files and potentially used by other devices to generate the needed section for the hdmi PCM devices to allow passthrough. The PR is marked as DNM and is in draft since the change depends on Syntax 7 support, which is only available in git version of alsa-lib, it is not a simple ask for a user (or developer) to use it and the change would break UCM for released alsa-lib. @perexg, is this something that you were suggesting? I think the macro and it's definition is generic enough, I hope that it will be usable for non SOF setups. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/413.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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