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

[PATCH 0/2] ASoC: Intel: soc-acpi: remove unused TigerLake configurations
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

Remove two machine descriptors which were added for development on Intel RVPs but are no longer used or productized. This will not have any impact on end-users. The corresponding cleanup was applied in the SOF topologies. Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: remove unused TGL table with rt5682 only ASoC: Intel: soc-acpi: remove TGL RVP mixed SoundWire/TDM config .../intel/common/soc-acpi-intel-tgl-match.c | 23 ------------------- 1 file changed, 23 deletions(-) -- 2.25.1

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[PATCH] ASoC: wm8524: Do not print probe defer error
by Fabio Estevam 04 Mar '21

04 Mar '21

On an imx8mq-evk the following error is seen: [ 1.375809] wm8524-codec audio-codec: Failed to get mute line: -517 It happens because the codec driver may probe prior to the imx gpio driver, which causes a probe defer. Change to dev_err_probe() to avoid printing this error. Reported-by: Chris Healy <cphealy(a)gmail.com> Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/codecs/wm8524.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8524.c b/sound/soc/codecs/wm8524.c index 4e9ab542f648..81f858f6bd67 100644 --- a/sound/soc/codecs/wm8524.c +++ b/sound/soc/codecs/wm8524.c @@ -227,7 +227,7 @@ static int wm8524_codec_probe(struct platform_device *pdev) wm8524->mute = devm_gpiod_get(&pdev->dev, "wlf,mute", GPIOD_OUT_LOW); if (IS_ERR(wm8524->mute)) { ret = PTR_ERR(wm8524->mute); - dev_err(&pdev->dev, "Failed to get mute line: %d\n", ret); + dev_err_probe(&pdev->dev, ret, "Failed to get mute line\n"); return ret; } -- 2.25.1

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[PATCH v2] ASoC: soc-dai: fix kernel-doc
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/soc/soc-dai.c:167: warning: expecting prototype for snd_soc_xlate_tdm_slot(). Prototype was for snd_soc_xlate_tdm_slot_mask() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/soc-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index cd3bb9a7983f..080fbe053fc5 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -154,7 +154,7 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** - * snd_soc_xlate_tdm_slot - generate tx/rx slot mask. + * snd_soc_xlate_tdm_slot_mask - generate tx/rx slot mask. * @slots: Number of slots in use. * @tx_mask: bitmask representing active TX slots. * @rx_mask: bitmask representing active RX slots. -- 2.25.1

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[PATCH v2] ASoC: Intel: atom: fix kernel-doc
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/soc/intel/atom/sst/sst_loader.c:85: warning: expecting prototype for sst_start_merrifield(). Prototype was for sst_start_mrfld() instead sound/soc/intel/atom/sst/sst_acpi.c:339: warning: expecting prototype for intel_sst_remove(). Prototype was for sst_acpi_remove() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/atom/sst/sst_loader.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 2c1b8a2e3506..3be64430c256 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -328,7 +328,7 @@ static int sst_acpi_probe(struct platform_device *pdev) } /** -* intel_sst_remove - remove function +* sst_acpi_remove - remove function * * @pdev: platform device structure * diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 1c9b0c9ec483..eea889001c24 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -76,7 +76,7 @@ int intel_sst_reset_dsp_mrfld(struct intel_sst_drv *sst_drv_ctx) } /** - * sst_start_merrifield - Start the SST DSP processor + * sst_start_mrfld - Start the SST DSP processor * @sst_drv_ctx: intel_sst_drv context pointer * * This starts the DSP in MERRIFIELD platfroms -- 2.25.1

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[PATCH v2] ASoC: codecs: nau8825: fix kernel-doc
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/soc/codecs/nau8825.c:298: warning: wrong kernel-doc identifier on line: * Ramp up the headphone volume change gradually to target level. sound/soc/codecs/nau8825.c:358: warning: expecting prototype for This func(). Prototype was for nau8825_intlog10_dec3() instead sound/soc/codecs/nau8825.c:411: warning: wrong kernel-doc identifier on line: * computes cross talk suppression sidetone gain. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/codecs/nau8825.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f0cba7b5758b..e19db30c457b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -295,7 +295,7 @@ static inline void nau8825_sema_reset(struct nau8825 *nau8825) } /** - * Ramp up the headphone volume change gradually to target level. + * nau8825_hpvol_ramp - Ramp up the headphone volume change gradually to target level. * * @nau8825: component to register the codec private data with * @vol_from: the volume to start up @@ -347,8 +347,9 @@ static void nau8825_hpvol_ramp(struct nau8825 *nau8825, } /** - * Computes log10 of a value; the result is round off to 3 decimal. This func- - * tion takes reference to dvb-math. The source code locates as the following. + * nau8825_intlog10_dec3 - Computes log10 of a value + * the result is round off to 3 decimal. This function takes reference to + * dvb-math. The source code locates as the following. * Linux/drivers/media/dvb-core/dvb_math.c * @value: input for log10 * @@ -408,7 +409,7 @@ static u32 nau8825_intlog10_dec3(u32 value) } /** - * computes cross talk suppression sidetone gain. + * nau8825_xtalk_sidetone - computes cross talk suppression sidetone gain. * * @sig_org: orignal signal level * @sig_cros: cross talk signal level -- 2.25.1

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[PATCH] ASoC: SOF: Intel: hda: enable async suspend
by Ranjani Sridharan 04 Mar '21

04 Mar '21

The skylake driver disabled async suspend to prevent disabling the DSP before the card was suspended during system suspend. This code was carried over to the SOF driver. But, there is no risk of the DSP getting disabled before the card is suspended with the SOF driver. Therefore, it is safe to enable async suspend and thereby optimize the system resume time. Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 0dc3a8c0f5e3..25b799f0accc 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -614,8 +614,6 @@ static int hda_init_caps(struct snd_sof_dev *sdev) u32 link_mask; int ret = 0; - device_disable_async_suspend(bus->dev); - /* check if dsp is there */ if (bus->ppcap) dev_dbg(sdev->dev, "PP capability, will probe DSP later.\n"); -- 2.25.1

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[PATCH] ALSA: hda: fix kernel-doc warnings
by Pierre-Louis Bossart 04 Mar '21

04 Mar '21

v5.12-rc1 flags new warnings with make W=1, fix missing or broken function descriptors. sound/pci/hda/hda_codec.c:3492: warning: expecting prototype for snd_hda_input_mux_info_info(). Prototype was for snd_hda_input_mux_info() instead sound/pci/hda/hda_codec.c:3521: warning: expecting prototype for snd_hda_input_mux_info_put(). Prototype was for snd_hda_input_mux_put() instead sound/pci/hda/hda_codec.c:3958: warning: expecting prototype for _snd_hda_pin_ctl(). Prototype was for _snd_hda_set_pin_ctl() instead sound/pci/hda/hda_jack.c:223: warning: expecting prototype for snd_hda_set_dirty_all(). Prototype was for snd_hda_jack_set_dirty_all() instead sound/pci/hda/hda_jack.c:309: warning: expecting prototype for snd_hda_jack_detect_enable_mst(). Prototype was for snd_hda_jack_detect_enable_callback_mst() instead sound/pci/hda/hda_generic.c:3933: warning: expecting prototype for snd_dha_gen_add_mute_led_cdev(). Prototype was for snd_hda_gen_add_mute_led_cdev() instead sound/pci/hda/hda_generic.c:4093: warning: expecting prototype for snd_dha_gen_add_micmute_led_cdev(). Prototype was for snd_hda_gen_add_micmute_led_cdev() instead sound/pci/hda/patch_ca0132.c:2357: warning: expecting prototype for Prepare and send the SCP message to DSP(). Prototype was for dspio_scp() instead sound/pci/hda/patch_ca0132.c:2883: warning: expecting prototype for Allocate router ports(). Prototype was for dsp_allocate_router_ports() instead sound/pci/hda/patch_ca0132.c:3202: warning: expecting prototype for Write a block of data into DSP code or data RAM using pre(). Prototype was for dspxfr_one_seg() instead sound/pci/hda/patch_ca0132.c:3397: warning: expecting prototype for data overlay to DSP memories(). Prototype was for dspxfr_image() instead sound/hda/hdac_regmap.c:393: warning: expecting prototype for snd_hdac_regmap_init(). Prototype was for snd_hdac_regmap_exit() instead sound/hda/ext/hdac_ext_controller.c:142: warning: expecting prototype for snd_hdac_ext_bus_get_link_index(). Prototype was for snd_hdac_ext_bus_get_link() instead sound/hda/ext/hdac_ext_stream.c:140: warning: expecting prototype for snd_hdac_ext_linkstream_start(). Prototype was for snd_hdac_ext_link_stream_start() instead Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/hda/ext/hdac_ext_controller.c | 2 +- sound/hda/ext/hdac_ext_stream.c | 2 +- sound/hda/hdac_regmap.c | 2 +- sound/pci/hda/hda_codec.c | 6 +++--- sound/pci/hda/hda_generic.c | 4 ++-- sound/pci/hda/hda_jack.c | 4 ++-- sound/pci/hda/patch_ca0132.c | 9 ++++----- 7 files changed, 14 insertions(+), 15 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index a9bd39b93697..b2df7b4f9227 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -133,7 +133,7 @@ void snd_hdac_link_free_all(struct hdac_bus *bus) EXPORT_SYMBOL_GPL(snd_hdac_link_free_all); /** - * snd_hdac_ext_bus_get_link_index - get link based on codec name + * snd_hdac_ext_bus_get_link - get link based on codec name * @bus: the pointer to HDAC bus object * @codec_name: codec name */ diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c index c4d54a838773..0c005d67fa89 100644 --- a/sound/hda/ext/hdac_ext_stream.c +++ b/sound/hda/ext/hdac_ext_stream.c @@ -133,7 +133,7 @@ void snd_hdac_ext_stream_decouple(struct hdac_bus *bus, EXPORT_SYMBOL_GPL(snd_hdac_ext_stream_decouple); /** - * snd_hdac_ext_linkstream_start - start a stream + * snd_hdac_ext_link_stream_start - start a stream * @stream: HD-audio ext core stream to start */ void snd_hdac_ext_link_stream_start(struct hdac_ext_stream *stream) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d75f31eb9d78..fe3587547cfe 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -386,7 +386,7 @@ int snd_hdac_regmap_init(struct hdac_device *codec) EXPORT_SYMBOL_GPL(snd_hdac_regmap_init); /** - * snd_hdac_regmap_init - Release the regmap from HDA codec + * snd_hdac_regmap_exit - Release the regmap from HDA codec * @codec: the codec object */ void snd_hdac_regmap_exit(struct hdac_device *codec) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9b755062d841..2026f1ccaf5a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3483,7 +3483,7 @@ EXPORT_SYMBOL_GPL(snd_hda_check_amp_list_power); */ /** - * snd_hda_input_mux_info_info - Info callback helper for the input-mux enum + * snd_hda_input_mux_info - Info callback helper for the input-mux enum * @imux: imux helper object * @uinfo: pointer to get/store the data */ @@ -3506,7 +3506,7 @@ int snd_hda_input_mux_info(const struct hda_input_mux *imux, EXPORT_SYMBOL_GPL(snd_hda_input_mux_info); /** - * snd_hda_input_mux_info_put - Put callback helper for the input-mux enum + * snd_hda_input_mux_put - Put callback helper for the input-mux enum * @codec: the HDA codec * @imux: imux helper object * @ucontrol: pointer to get/store the data @@ -3941,7 +3941,7 @@ unsigned int snd_hda_correct_pin_ctl(struct hda_codec *codec, EXPORT_SYMBOL_GPL(snd_hda_correct_pin_ctl); /** - * _snd_hda_pin_ctl - Helper to set pin ctl value + * _snd_hda_set_pin_ctl - Helper to set pin ctl value * @codec: the HDA codec * @pin: referred pin NID * @val: pin control value to set diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5e40944e7342..8b7c5508f368 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3923,7 +3923,7 @@ static void vmaster_update_mute_led(void *private_data, int enabled) } /** - * snd_dha_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED + * snd_hda_gen_add_mute_led_cdev - Create a LED classdev and enable as vmaster mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking */ @@ -4074,7 +4074,7 @@ static int add_micmute_led_hook(struct hda_codec *codec) } /** - * snd_dha_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED + * snd_hda_gen_add_micmute_led_cdev - Create a LED classdev and enable as mic-mute LED * @codec: the HDA codec * @callback: the callback for LED classdev brightness_set_blocking * diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index b8b568046592..ac00866d8032 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -213,7 +213,7 @@ static void jack_detect_update(struct hda_codec *codec, } /** - * snd_hda_set_dirty_all - Mark all the cached as dirty + * snd_hda_jack_set_dirty_all - Mark all the cached as dirty * @codec: the HDA codec * * This function sets the dirty flag to all entries of jack table. @@ -293,7 +293,7 @@ find_callback_from_list(struct hda_jack_tbl *jack, } /** - * snd_hda_jack_detect_enable_mst - enable the jack-detection + * snd_hda_jack_detect_enable_callback_mst - enable the jack-detection * @codec: the HDA codec * @nid: pin NID to enable * @func: callback function to register diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 7e62aed172a9..c966f49fa942 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -2338,7 +2338,7 @@ static int dspio_send_scp_message(struct hda_codec *codec, } /** - * Prepare and send the SCP message to DSP + * dspio_scp - Prepare and send the SCP message to DSP * @codec: the HDA codec * @mod_id: ID of the DSP module to send the command * @src_id: ID of the source @@ -2865,7 +2865,7 @@ static int dsp_dma_stop(struct hda_codec *codec, } /** - * Allocate router ports + * dsp_allocate_router_ports - Allocate router ports * * @codec: the HDA codec * @num_chans: number of channels in the stream @@ -3178,8 +3178,7 @@ static int dspxfr_hci_write(struct hda_codec *codec, } /** - * Write a block of data into DSP code or data RAM using pre-allocated - * DMA engine. + * dspxfr_one_seg - Write a block of data into DSP code or data RAM using pre-allocated DMA engine. * * @codec: the HDA codec * @fls: pointer to a fast load image @@ -3376,7 +3375,7 @@ static int dspxfr_one_seg(struct hda_codec *codec, } /** - * Write the entire DSP image of a DSP code/data overlay to DSP memories + * dspxfr_image - Write the entire DSP image of a DSP code/data overlay to DSP memories * * @codec: the HDA codec * @fls_data: pointer to a fast load image -- 2.25.1

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conf: Remove non-existent SPDIF output on Dell AE515
by GitHub pull_request - opened 03 Mar '21

03 Mar '21

alsa-project/alsa-lib pull request #124 was opened from drjhe: The Dell Professional Sound Bar AE515 shows up in GNOME Settings with a digital output that doesn't physically exist. Add it to the list to suppress this. Request URL : https://github.com/alsa-project/alsa-lib/pull/124 Patch URL : https://github.com/alsa-project/alsa-lib/pull/124.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ALSA: usb: Add Plantronics C320-M USB ctrl msg delay quirk
by John Ernberg 03 Mar '21

03 Mar '21

The microphone in the Plantronics C320-M headset will randomly fail to initialize properly, at least when using Microsoft Teams. Introducing a 20ms delay on the control messages appears to resolve the issue. Link: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/1065 Tested-by: Andreas Kempe <kempe(a)lysator.liu.se> Signed-off-by: John Ernberg <john.ernberg(a)actia.se> --- Note: Microsoft Teams is the only application tests have been performed with and no other kinds of investigations have been performed, like USB packet logging. Originally reported as a possible PulseAudio issue, where it was suggested to try a delay quirk in the kernel. If other tests should be performed, please let me know which tests and I will try to do them. All testing was done on 5.10, patch was then applied on sound.git master for submission I did not register to the mailing lists, please keep me CC on any comments. Thanks! sound/usb/quirks.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index d02dac5fcd40..a31b7db92b1e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1673,6 +1673,14 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) msleep(20); + /* + * Plantronics C320-M needs a delay to avoid random + * microhpone failures. + */ + if (chip->usb_id == USB_ID(0x047f, 0xc025) && + (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) + msleep(20); + /* Zoom R16/24, many Logitech(at least H650e/H570e/BCC950), * Jabra 550a, Kingston HyperX needs a tiny delay here, * otherwise requests like get/set frequency return -- 2.30.0

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[PATCH 0/4] ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec.
by Vitaly Rodionov 03 Mar '21

03 Mar '21

Dell's laptops Inspiron 3500, Inspiron 3501, Inspiron 3505 are using Cirrus Logic CS8409 HDA bridge with CS42L42 companion codec. The CS8409 is a multichannel HD audio routing controller. CS8409 includes support for four channels of digital microphone data and two bidirectional ASPs for up to 32 channels of TDM data or 4 channels of I2S data. The CS8409 is intended to be used with a remote companion codec that implements high performance analog functions in close physical proximity to the end-equipment audio port or speaker driver. The CS42L42 is a low-power audio codec with integrated MIPI SoundWire interface or I2C/I2S/TDM interfaces designed for portable applications. It provides a high-dynamic range, stereo DAC for audio playback and a mono high-dynamic-range ADC for audio capture Stefan Binding (1): ALSA: hda/cirrus: Add Headphone and Headset MIC Volume Control Vitaly Rodionov (3): ALSA: hda/cirrus: Increase AUTO_CFG_MAX_INS from 8 to 18 ALSA: hda/cirrus: Add support for CS8409 HDA bridge and CS42L42 companion codec. ALSA: hda/cirrus: Add jack detect interrupt support from CS42L42 companion codec. sound/pci/hda/hda_auto_parser.h | 2 +- sound/pci/hda/hda_local.h | 2 +- sound/pci/hda/patch_cirrus.c | 1033 +++++++++++++++++++++++++++++++ 3 files changed, 1035 insertions(+), 2 deletions(-) -- 2.25.1

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