May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH AUTOSEL 6.3 23/37] ASoC: Intel: avs: Fix avs_path_module::instance_id size
by Sasha Levin 31 May '23

31 May '23

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 320f4d868b83a804e3a4bd61a5b7d0f1db66380e ] All IPCs using instance_id use 8 bit value. Original commit used 16 bit value because FW reports possible max value in 16 bit field, but in practice FW limits the value to 8 bits. Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-7-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/avs.h | 4 ++-- sound/soc/intel/avs/dsp.c | 4 ++-- sound/soc/intel/avs/path.h | 2 +- sound/soc/intel/avs/probes.c | 2 +- 4 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/avs/avs.h b/sound/soc/intel/avs/avs.h index d7fccdcb9c167..0cf38c9e768e7 100644 --- a/sound/soc/intel/avs/avs.h +++ b/sound/soc/intel/avs/avs.h @@ -283,8 +283,8 @@ void avs_release_firmwares(struct avs_dev *adev); int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, u8 core_id, u8 domain, void *param, u32 param_size, - u16 *instance_id); -void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id, + u8 *instance_id); +void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id, u8 ppl_instance_id, u8 core_id); int avs_dsp_create_pipeline(struct avs_dev *adev, u16 req_size, u8 priority, bool lp, u16 attributes, u8 *instance_id); diff --git a/sound/soc/intel/avs/dsp.c b/sound/soc/intel/avs/dsp.c index b881100d3e02a..aa03af4473e94 100644 --- a/sound/soc/intel/avs/dsp.c +++ b/sound/soc/intel/avs/dsp.c @@ -225,7 +225,7 @@ static int avs_dsp_put_core(struct avs_dev *adev, u32 core_id) int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, u8 core_id, u8 domain, void *param, u32 param_size, - u16 *instance_id) + u8 *instance_id) { struct avs_module_entry mentry; bool was_loaded = false; @@ -272,7 +272,7 @@ int avs_dsp_init_module(struct avs_dev *adev, u16 module_id, u8 ppl_instance_id, return ret; } -void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u16 instance_id, +void avs_dsp_delete_module(struct avs_dev *adev, u16 module_id, u8 instance_id, u8 ppl_instance_id, u8 core_id) { struct avs_module_entry mentry; diff --git a/sound/soc/intel/avs/path.h b/sound/soc/intel/avs/path.h index 197222c5e008e..657f7b093e805 100644 --- a/sound/soc/intel/avs/path.h +++ b/sound/soc/intel/avs/path.h @@ -37,7 +37,7 @@ struct avs_path_pipeline { struct avs_path_module { u16 module_id; - u16 instance_id; + u8 instance_id; union avs_gtw_attributes gtw_attrs; struct avs_tplg_module *template; diff --git a/sound/soc/intel/avs/probes.c b/sound/soc/intel/avs/probes.c index 70a94201d6a56..275928281c6c6 100644 --- a/sound/soc/intel/avs/probes.c +++ b/sound/soc/intel/avs/probes.c @@ -18,7 +18,7 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id { struct avs_probe_cfg cfg = {{0}}; struct avs_module_entry mentry; - u16 dummy; + u8 dummy; avs_get_module_entry(adev, &AVS_PROBE_MOD_UUID, &mentry); -- 2.39.2

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[PATCH AUTOSEL 6.3 22/37] ASoC: Intel: avs: Account for UID of ACPI device
by Sasha Levin 31 May '23

31 May '23

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 836855100b87b4dd7a82546131779dc255c18b67 ] Configurations with multiple codecs attached to the platform are supported but only if each from the set is different. Add new field representing the 'Unique ID' so that codecs that share Vendor and Part IDs can be differentiated and thus enabling support for such configurations. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-acpi.h | 1 + sound/soc/intel/avs/board_selection.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index b38fd25c57295..528279056b3ab 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr { /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { u8 id[ACPI_ID_LEN]; + const char *uid; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index b2823c2107f77..60f8fb0bff95b 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -443,7 +443,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev) } for (mach = boards->machs; mach->id[0]; mach++) { - if (!acpi_dev_present(mach->id, NULL, -1)) + if (!acpi_dev_present(mach->id, mach->uid, -1)) continue; if (mach->machine_quirk) -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 19/37] ASoC: soc-pcm: test if a BE can be prepared
by Sasha Levin 31 May '23

31 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ] In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 1e7d09556fe3e..b7bc1865b9e4a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -123,6 +123,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7958c9defd492..1db82501fec18 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2417,6 +2417,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3057,3 +3060,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.39.2

1 0

[PATCH AUTOSEL 6.3 15/37] ASoC: lpass: Fix for KASAN use_after_free out of bounds
by Sasha Levin 31 May '23

31 May '23

From: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> [ Upstream commit 75e5fab7db0cecb6e16b22c34608f0b40a4c7cd1 ] When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com Link: https://lore.kernel.org/r/20230511112532.16106-1-quic_visr@quicinc.com Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 589c490a8c487..88d8fa58ec1eb 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.39.2

1 0

[PATCH] ASoC: nau8825: Add the management of headset detection for power saving
by David Lin 31 May '23

31 May '23

The patch is to manage HSD feature for power saving. The detail is to disable HSD feature after the headset detection is done. When the jack is inserted, the HSD feature will be enabled again. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index cc3e18207c42..f6dd84b32e0b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1704,6 +1704,10 @@ static void nau8825_setup_auto_irq(struct nau8825 *nau8825) { struct regmap *regmap = nau8825->regmap; + /* Enable HSD function */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE, NAU8825_HSD_AUTO_MODE); + /* Enable headset jack type detection complete interruption and * jack ejection interruption. */ @@ -1955,6 +1959,9 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0xf); + /* Disable HSD function */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, NAU8825_HSD_AUTO_MODE, 0); + /* Leaving HPOL/R grounded after jack insert by default. They will be * ungrounded as part of the widget power up sequence at the beginning * of playback to reduce pop. -- 2.25.1

2 1

[PATCH] dirvers/gpio: remove redundant reset gpio suffix
by Lizhe 31 May '23

31 May '23

no need to add the suffix "gpio" or "gpios" after "reset", as they will be automatically added, see of_find_gpio(). Signed-off-by: Lizhe <sensor1010(a)163.com> --- sound/soc/codecs/cs35l34.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs35l34.c b/sound/soc/codecs/cs35l34.c index b3f98023e6a7..cbf0f2f4f9df 100644 --- a/sound/soc/codecs/cs35l34.c +++ b/sound/soc/codecs/cs35l34.c @@ -1061,7 +1061,7 @@ static int cs35l34_i2c_probe(struct i2c_client *i2c_client) dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); cs35l34->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, - "reset-gpios", GPIOD_OUT_LOW); + "reset", GPIOD_OUT_LOW); if (IS_ERR(cs35l34->reset_gpio)) { ret = PTR_ERR(cs35l34->reset_gpio); goto err_regulator; -- 2.34.1

2 1

[PATCH v2 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 May '23

31 May '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Change notes are included with each patch, most of the changes are trivial, the notable ones are moving the IRQs out of irqchip and into the MFD, and moving the DT binding to sound. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 320 +++ MAINTAINERS | 5 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 86 + drivers/mfd/cs42l43-sdw.c | 213 ++ drivers/mfd/cs42l43.c | 1141 +++++++++ drivers/mfd/cs42l43.h | 23 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 31 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 279 ++ include/linux/mfd/cs42l43-regs.h | 1172 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 951 +++++++ sound/soc/codecs/cs42l43-sdw.c | 75 + sound/soc/codecs/cs42l43.c | 2275 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 126 + 26 files changed, 7513 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

5 14

[PATCH v5 0/3] Add TDM audio on StarFive JH7110
by Walker Chen 31 May '23

31 May '23

This patchset adds TDM audio driver for the StarFive JH7110 SoC. The first patch adds device tree binding for TDM module. The second patch adds tdm driver support for JH7110 SoC. The last patch adds device tree node and pins configuration of tdm to JH7110 dts. The series has been tested on the VisionFive 2 board by plugging an audio expansion board. For more information of audio expansion board, you can take a look at the following webpage: https://wiki.seeedstudio.com/ReSpeaker_2_Mics_Pi_HAT/ The last patch should be applied after the following patch: https://lore.kernel.org/all/20230322094820.24738-5-walker.chen@starfivetech… Changes since v4: - Rebased on Linux 6.4-rc2. - Adjusted code indentation in jh7110_tdm_syncdiv(). - Removed '__maybe_unused' for jh7110_tdm_system_suspend() and jh7110_tdm_system_resume(). - Removed initialization for variable 'ret' in jh7110_tdm_hw_params(). - Return immedially when entering unsurpported case in jh7110_tdm_set_dai_fmt(). - Check the return value with IS_ERR_OR_NULL() for the call to devm_reset_control_array_get_exclusive(). - Dropped the jh7110_tdm_clk_enable() in probe and rely on runtime PM check if CONFIG_PM is enabled. - Used RUNTIME_PM_OPS() instead of SET_RUNTIME_PM_OPS() and SYSTEM_SLEEP_PM_OPS() instead of SET_SYSTEM_SLEEP_PM_OPS() in structure jh7110_tdm_pm_ops. Changes since v3: - Modified the subject of patch [1/3] to reflect which subsystem it is. - Use the set of functions with 'clk_bulk_' to get/enable/disable clocks. - Dropped the overlay from patch [3/3]. - Dropped the redundant macro 'CONFIG_PM' and 'CONFIG_PM_SLEEP' around suspend() and resume(). Changes since v2: - Use dt-overlay to describe sound card because need to plug the audio expansion board into the VisionFive2 board. - Modified the coding style for driver. - Moved assignment of stop_dma_first to startup function of dai_driver. - Dropped some useless macro definition. - Use loops to get/enable/disable clocks. Changes since v1: - Rebased on Linux 6.3-rc4. - Added the dts file dedicated to describe audio card. - Added the item for JH7110 audio board to the dt-binding of StarFive SoC-based boards. --- v4: https://lore.kernel.org/all/20230511091549.28003-1-walker.chen@starfivetech… v3: https://lore.kernel.org/all/20230506090116.9206-1-walker.chen@starfivetech.… v2: https://lore.kernel.org/all/20230420024118.22677-1-walker.chen@starfivetech… v1: https://lore.kernel.org/all/20230329153320.31390-1-walker.chen@starfivetech… Walker Chen (3): ASoC: dt-bindings: Add TDM controller bindings for StarFive JH7110 ASoC: starfive: Add JH7110 TDM driver riscv: dts: starfive: add the node and pins configuration for tdm .../bindings/sound/starfive,jh7110-tdm.yaml | 98 +++ MAINTAINERS | 6 + .../jh7110-starfive-visionfive-2.dtsi | 40 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 21 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/starfive/Kconfig | 15 + sound/soc/starfive/Makefile | 2 + sound/soc/starfive/jh7110_tdm.c | 679 ++++++++++++++++++ 9 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-tdm.yaml create mode 100644 sound/soc/starfive/Kconfig create mode 100644 sound/soc/starfive/Makefile create mode 100644 sound/soc/starfive/jh7110_tdm.c base-commit: f1fcbaa18b28dec10281551dfe6ed3a3ed80e3d6 prerequisite-patch-id: 398744c61913c76a35754de867c4f820ca7a8d99 prerequisite-patch-id: 07986c6e98ce85d3ee84f38c780e72145f098fc8 prerequisite-patch-id: 1babe83d6bf999bad17584dc595480f9070a5369 prerequisite-patch-id: 63c1f70ca2bcf827429542e60d74411b7666ceff prerequisite-patch-id: 9fbb7ad1dd258bb8ff5946c4a0e59de4bfd82a04 prerequisite-patch-id: a7f61973ccb6bc425daa8a86c048fd0f5a915b76 prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: 584c256c9acb52ee2773d0c81c3f4977fc18155a prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 -- 2.17.1

4 10

speaker-test: allow controlling how long the noise is heard by the user
by GitHub issues - opened 30 May '23

30 May '23

alsa-project/alsa-utils issue #219 was opened from jidanni: $ speaker-test --nloops 1 makes a pink noise for about one second. On the man page mention how to make this into say a half second, as some users would like to listen to shorter test sounds, and no matter if the user does ``` $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 5 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 50 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 500 $ speaker-test --device hdmi:HDMI -c 2 --nloops 1 --period 5000 ``` what the user hears is exactly the same, therefore --period is not what controls this item. I am saying that some users can confirm their speakers work without listing to a full second of pink noise, and thus would like a way to listen to a shorter length. As far as ``` -b | --buffer TIME Use buffer size of TIME microseconds. When 0 is given, use the maximal buffer size. The default value is 0. -p | --period TIME Use period size of TIME microseconds. When 0 is given, the pe‐ riods given by -P option is used. The default value is 0. ``` are concerned, they must be documenting something completely different. Debian Package: alsa-utils Version: 1.2.8-1 Issue URL : https://github.com/alsa-project/alsa-utils/issues/219 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 0/3] ASoC: topology: allow for partial dailink stream_name match
by Pierre-Louis Bossart 30 May '23

30 May '23

The topology file and the machine driver rely on common definitions for the dailink stream_name. To avoid any backwards-compatibility problems, the machine driver stream names are set in stone and cannot be modified. This is problematic when we try to name some of the topology widgets after the stream_name, since the widget name is limited to 44 characters tools/include/uapi/sound/asound.h:#define SNDRV_CTL_ELEM_ID_NAME_MAXLEN 44 Existing examples include "Analog Playback and Capture" for HDaudio dailinks, which leaves less than 20 chars to identify widgets/controls with a meaningful name. Since the 44-char limit is part of the UAPI definitions, we assumed there is no way to increase it. This patchset suggests instead a partial match which allows topology files to use a shorter stream_name, which in turn allows for self-explanatory widget names that comply with the 44-char limit. This should not break any existing setup but with the introduction of a partial match new dailinks should be named carefully to avoid confusions between e.g. 'link1' and 'link10'. The last patch fixes such an issue in the 'nocodec' test topology used by Intel. Ranjani Sridharan (3): ASoC: topology: Allow partial matching when finding DAI link ASoC: SOF: topology: Use partial match for connecting DAI link and DAI widget ASoC: SOF: Intel: HDA: Limit the number of dai drivers for nocodec mode sound/soc/soc-topology.c | 6 +++--- sound/soc/sof/intel/hda.c | 6 +++++- sound/soc/sof/intel/hda.h | 4 +++- sound/soc/sof/topology.c | 2 +- 4 files changed, 12 insertions(+), 6 deletions(-) -- 2.37.2

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