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

[PATCH] ALSA: hda - fix the 'Capture Switch' value change notifications
by Jaroslav Kysela 12 Aug '21

12 Aug '21

The original code in the cap_put_caller() function does not handle correctly the positive values returned from the passed function for multiple iterations. It means that the change notifications may be lost. Fixes: 352f7f914ebb ("ALSA: hda - Merge Realtek parser code to generic parser") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213851 Cc: <stable(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/hda_generic.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index e97d00585e8e..481d8f8d3396 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3460,7 +3460,7 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, struct hda_gen_spec *spec = codec->spec; const struct hda_input_mux *imux; struct nid_path *path; - int i, adc_idx, err = 0; + int i, adc_idx, ret, err = 0; imux = &spec->input_mux; adc_idx = kcontrol->id.index; @@ -3470,9 +3470,13 @@ static int cap_put_caller(struct snd_kcontrol *kcontrol, if (!path || !path->ctls[type]) continue; kcontrol->private_value = path->ctls[type]; - err = func(kcontrol, ucontrol); - if (err < 0) + ret = func(kcontrol, ucontrol); + if (ret < 0) { + err = ret; break; + } + if (ret > 0) + err = 1; } mutex_unlock(&codec->control_mutex); if (err >= 0 && spec->cap_sync_hook) -- 2.31.1

2 1

Filtering/manipulating MIDI messages
by GitHub issues - opened 12 Aug '21

12 Aug '21

alsa-project/alsa-utils issue #106 was opened from neilbags: Hi, I'm using aconnect to send MIDI data from one USB MIDI device to another. I need to filter the messages, so that only one MIDI channel gets sent through, or ideally, send data that comes in on one channel from device A out to device B on a different MIDI channel. Is there any way to do this just with the command line tools? Issue URL : https://github.com/alsa-project/alsa-utils/issues/106 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3 00/27] ALSA: hda/cirrus: Split generic cirrus HDA codecs and CS8490 bridge into separate modules.
by Vitaly Rodionov 11 Aug '21

11 Aug '21

This series of patches splits generic cirrus HDA codecs and CS8490 bridge into separate modules, adds support for multiple companion codecs connected to CS8409, and also adds support for new DELL HW platform. CS8409 part is not really a HDA codec, it is a HDA bridge where companion codecs (up to 16) can be attached. With growing number of supported configurations and platforms, patch_cirrus is getting less and less transparent and maintainable. So, the logical step is to separate generic Cirrus HDA codecs support and Cirrus HDA bridge support. Lots of improvements to existing functionality, code clean-up and refactoring, remove duplicated/redundant code, improve I2C functions etc. Add support for new DELL HW platform with 2 CS42L42 codecs for front and rear jacks. Lucas Tanure (12): ALSA: hda/cirrus: Move CS8409 HDA bridge to separate module ALSA: hda/cs8409: Move arrays of configuration to a new file ALSA: hda/cs8409: Disable unsolicited response for the first boot ALSA: hda/cs8409: Prevent I2C access during suspend time ALSA: hda/cs8409: Generalize volume controls ALSA: hda/cs8409: Dont disable I2C clock between consecutive accesses ALSA: hda/cs8409: Avoid setting the same I2C address for every access ALSA: hda/cs8409: Avoid re-setting the same page as the last access ALSA: hda/cs8409: Support i2c bulk read/write functions ALSA: hda/cs8409: Separate CS8409, CS42L42 and project functions ALSA: hda/cs8409: Move codec properties to its own struct ALSA: hda/cs8409: Add support for dolphin Stefan Binding (15): ALSA: hda/cs8409: Use enums for register names and coefficients ALSA: hda/cs8409: Mask all CS42L42 interrupts on initialization ALSA: hda/cs8409: Reduce HS pops/clicks for Cyborg ALSA: hda/cs8409: Disable unnecessary Ring Sense for Cyborg/Warlock/Bullseye ALSA: hda/cs8409: Disable unsolicited responses during suspend ALSA: hda/cs8409: Mask CS42L42 wake events ALSA: hda/cs8409: Simplify CS42L42 jack detect. ALSA: hda/cs8409: Support multiple sub_codecs for Suspend/Resume/Unsol events ALSA: hda/cs8409: Add Support to disable jack type detection for CS42L42 ALSA: hda/cs8409: Enable Full Scale Volume for Line Out Codec on Dolphin ALSA: hda/cs8409: Set fixed sample rate of 48kHz for CS42L42 ALSA: hda/cs8409: Use timeout rather than retries for I2C transaction waits ALSA: hda/cs8409: Remove unnecessary delays ALSA: hda/cs8409: Follow correct CS42L42 power down sequence for suspend ALSA: hda/cs8409: Unmute/Mute codec when stream starts/stops sound/pci/hda/Kconfig | 10 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_cirrus.c | 1074 ----------------------- sound/pci/hda/patch_cs8409-tables.c | 560 ++++++++++++ sound/pci/hda/patch_cs8409.c | 1253 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.h | 369 ++++++++ 6 files changed, 2194 insertions(+), 1074 deletions(-) create mode 100644 sound/pci/hda/patch_cs8409-tables.c create mode 100644 sound/pci/hda/patch_cs8409.c create mode 100644 sound/pci/hda/patch_cs8409.h -- 2.25.1

2 29

[PATCH v3 00/20] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 11 Aug '21

11 Aug '21

This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simpified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +---------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> | bedais | | | +---------+ +---------+ +---------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ Thanks, srini Changes since v2: - seperated dsp lpass ip specific bindings out of dsp bindings - seperated audio ports and clocks code out of Q6AFE - fixed various cppcheck warnings. - Reuse existing code and most of the hardware specific bits from old drivers - added i2s module support - Addressed most of the comments from Pierre. - converted most of the existing bindings to yaml format. Srinivas Kandagatla (20): soc: dt-bindings: qcom: apr: convert to yaml soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: replace q6afe bindings with q6dsp-audio-ports ASoC: dt-bindings: replace q6afe-clk with q6dsp audio clk bindings ASoC: dt-bindings: replace q6asm with q6dsp audio stream bindings ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-dai compatible ASoC: dt-bindings: q6dsp: add q6apm-bedai compatible ASoC: dt-bindings: q6dsp-clocks: add q6prm clocks compatible ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add bedai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 209 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 201 --- .../devicetree/bindings/sound/qcom,q6asm.txt | 70 - .../sound/qcom,q6dsp-audio-clocks.yaml | 74 ++ .../sound/qcom,q6dsp-audio-ports.yaml | 202 +++ .../sound/qcom,q6dsp-audio-stream.yaml | 118 ++ drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 286 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6afe.h | 204 +-- .../sound/qcom,q6dsp-audio-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 203 +++ sound/soc/qcom/Kconfig | 26 + sound/soc/qcom/Makefile | 1 + sound/soc/qcom/qdsp6/Makefile | 19 +- sound/soc/qcom/qdsp6/audioreach.c | 1166 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 690 ++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-bedai.c | 266 ++++ sound/soc/qcom/qdsp6/q6apm-dai.c | 504 +++++++ sound/soc/qcom/qdsp6/q6apm.c | 919 +++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 155 +++ sound/soc/qcom/qdsp6/q6dsp-audio-clocks.c | 185 +++ sound/soc/qcom/qdsp6/q6dsp-audio-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-audio-ports.c | 625 +++++++++ sound/soc/qcom/qdsp6/q6dsp-audio-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 256 ++++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1114 ++++++++++++++++ 33 files changed, 7514 insertions(+), 1506 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.txt delete mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-audio-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-audio-ports.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-audio-stream.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-audio-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-bedai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-audio-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-audio-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-audio-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-audio-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

5 40

[PATCH] usb-audio: Input source control - digidesign mbox
by Damien Zammit 11 Aug '21

11 Aug '21

This adds a second mixer control to Digidesign Mbox to select between Analog/SPDIF capture. Users will note that selecting the SPDIF input source automatically switches the clock mode to sync to SPDIF, which is a feature of the hardware. (You can change the clock source back to internal if you want to capture from spdif but not sync to its clock although this mode is probably not recommended). Unfortunately, starting the stream resets both modes to Internally clocked and Analog input mode. --- sound/usb/mixer_quirks.c | 197 +++++++++++++++++++++++++++++++++++++++++------ 1 file changed, 175 insertions(+), 22 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7..ef771bb3e 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -596,31 +596,53 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) /* Digidesign Mbox 1 clock source switch (internal/spdif) */ -static int snd_mbox1_switch_get(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_clk_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct snd_usb_audio *chip = list->mixer->chip; + unsigned char buff[3]; + + /* Read clock source */ + snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + + if (!buff[0] && !buff[1] && !buff[2]) { + /* SPDIF synced */ + kctl->private_value = 1; + } else { + /* Internally clocked */ + kctl->private_value = 0; + } + ucontrol->value.enumerated.item[0] = kctl->private_value; return 0; } -static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) +static int snd_mbox1_clk_switch_update(struct usb_mixer_interface *mixer, int val) { struct snd_usb_audio *chip = mixer->chip; int err; unsigned char buff[3]; + unsigned char source[1]; err = snd_usb_lock_shutdown(chip); if (err < 0) return err; - /* Prepare for magic command to toggle clock source */ + /* Read input source */ err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), 0x81, USB_DIR_IN | USB_TYPE_CLASS | - USB_RECIP_INTERFACE, 0x00, 0x500, buff, 1); + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); if (err < 0) goto err; + + /* Read clock source */ err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), 0x81, USB_DIR_IN | @@ -644,13 +666,15 @@ static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) buff[0] = buff[1] = buff[2] = 0x00; } - /* Send the magic command to toggle the clock source */ + /* Toggle clock source */ err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x1, USB_TYPE_CLASS | USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); if (err < 0) goto err; + + /* Read clock source */ err = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0), 0x81, USB_DIR_IN | @@ -671,8 +695,8 @@ static int snd_mbox1_switch_update(struct usb_mixer_interface *mixer, int val) return err; } -static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, - struct snd_ctl_elem_value *ucontrol) +static int snd_mbox1_clk_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) { struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); struct usb_mixer_interface *mixer = list->mixer; @@ -685,12 +709,12 @@ static int snd_mbox1_switch_put(struct snd_kcontrol *kctl, return 0; kctl->private_value = new_val; - err = snd_mbox1_switch_update(mixer, new_val); + err = snd_mbox1_clk_switch_update(mixer, new_val); return err < 0 ? err : 1; } -static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) +static int snd_mbox1_clk_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) { static const char *const texts[2] = { "Internal", @@ -700,27 +724,156 @@ static int snd_mbox1_switch_info(struct snd_kcontrol *kcontrol, return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); } -static int snd_mbox1_switch_resume(struct usb_mixer_elem_list *list) +static int snd_mbox1_clk_switch_resume(struct usb_mixer_elem_list *list) +{ + return snd_mbox1_clk_switch_update(list->mixer, list->kctl->private_value); +} + +/* Digidesign Mbox 1 input source switch (analog/spdif) */ + +static int snd_mbox1_src_switch_get(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct snd_usb_audio *chip = list->mixer->chip; + unsigned char source[1]; + + /* Read input source */ + snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + + kctl->private_value = source[0] - 1; + ucontrol->value.enumerated.item[0] = kctl->private_value; + return 0; +} + +static int snd_mbox1_src_switch_update(struct usb_mixer_interface *mixer, int val) +{ + struct snd_usb_audio *chip = mixer->chip; + int err; + unsigned char buff[3]; + unsigned char source[1]; + + err = snd_usb_lock_shutdown(chip); + if (err < 0) + return err; + + /* Read input source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + if (err < 0) + goto err; + + /* 2 possibilities: ANALOG Source -> 0x01 + * S/PDIF Source -> 0x02 + * Setting the input source to S/PDIF resets the clock source to S/PDIF + */ + source[0] = (val + 1) & 0xff; + + /* Toggle the input source */ + err = snd_usb_ctl_msg(chip->dev, + usb_sndctrlpipe(chip->dev, 0), 0x1, + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + if (err < 0) + goto err; + + /* Read input source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_INTERFACE, 0x00, 0x500, source, 1); + if (err < 0) + goto err; + + /* Read clock source */ + err = snd_usb_ctl_msg(chip->dev, + usb_rcvctrlpipe(chip->dev, 0), 0x81, + USB_DIR_IN | + USB_TYPE_CLASS | + USB_RECIP_ENDPOINT, 0x100, 0x81, buff, 3); + if (err < 0) + goto err; +err: + snd_usb_unlock_shutdown(chip); + return err; +} + +static int snd_mbox1_src_switch_put(struct snd_kcontrol *kctl, + struct snd_ctl_elem_value *ucontrol) +{ + struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl); + struct usb_mixer_interface *mixer = list->mixer; + int err; + bool cur_val, new_val; + + cur_val = kctl->private_value; + new_val = ucontrol->value.enumerated.item[0]; + if (cur_val == new_val) + return 0; + + kctl->private_value = new_val; + err = snd_mbox1_src_switch_update(mixer, new_val); + return err < 0 ? err : 1; +} + +static int snd_mbox1_src_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *const texts[2] = { + "Analog", + "S/PDIF" + }; + + return snd_ctl_enum_info(uinfo, 1, ARRAY_SIZE(texts), texts); +} + +static int snd_mbox1_src_switch_resume(struct usb_mixer_elem_list *list) { - return snd_mbox1_switch_update(list->mixer, list->kctl->private_value); + return snd_mbox1_src_switch_update(list->mixer, list->kctl->private_value); } -static const struct snd_kcontrol_new snd_mbox1_switch = { +static const struct snd_kcontrol_new snd_mbox1_clk_switch = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Clock Source", .index = 0, .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, - .info = snd_mbox1_switch_info, - .get = snd_mbox1_switch_get, - .put = snd_mbox1_switch_put, + .info = snd_mbox1_clk_switch_info, + .get = snd_mbox1_clk_switch_get, + .put = snd_mbox1_clk_switch_put, + .private_value = 0 +}; + +static const struct snd_kcontrol_new snd_mbox1_src_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .index = 1, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_mbox1_src_switch_info, + .get = snd_mbox1_src_switch_get, + .put = snd_mbox1_src_switch_put, .private_value = 0 }; -static int snd_mbox1_create_sync_switch(struct usb_mixer_interface *mixer) +static int snd_mbox1_controls_create(struct usb_mixer_interface *mixer) { - return add_single_ctl_with_resume(mixer, 0, - snd_mbox1_switch_resume, - &snd_mbox1_switch, NULL); + int err; + err = add_single_ctl_with_resume(mixer, 0, + snd_mbox1_clk_switch_resume, + &snd_mbox1_clk_switch, NULL); + if (err < 0) + return err; + + return add_single_ctl_with_resume(mixer, 1, + snd_mbox1_src_switch_resume, + &snd_mbox1_src_switch, NULL); } /* Native Instruments device quirks */ @@ -3029,7 +3182,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) break; case USB_ID(0x0dba, 0x1000): /* Digidesign Mbox 1 */ - err = snd_mbox1_create_sync_switch(mixer); + err = snd_mbox1_controls_create(mixer); break; case USB_ID(0x17cc, 0x1011): /* Traktor Audio 6 */ -- 2.13.1

2 1

[PATCH 00/12] ASoC: cs42l42: Series of bugfixes and improvements
by Richard Fitzgerald 11 Aug '21

11 Aug '21

Various bugfixes and improvements to the cs42l42 codec driver. Clocking bugfixes: - Remove any use of MCLK > 12.288 MHz because it isn't possible to switch between the low and high frequency ranges on-the-fly - Prevent the PLL configuration being written while the PLL is running - Correctly set the MCLK for the sample rate converters ASP bugfixes: - Ensure the TX (capture) ASP is properly configured when recording mono audio. Other bugfixes: - Prevent NULL pointer deref if there is an interrupt before soc component probe Code cleanup: - Remove the unused runtime suspend/resume functions. - Remove some code made redundant by an earlier patch Improvement: - Add ALSA control for HP path attenuation. Previously this was always set to -6dB with no way for the user to configure it. - Add ALSA control to disable the soft volume ramp at the start of audio. Richard Fitzgerald (12): ASoC: cs42l42: Use PLL for SCLK > 12.188MHz ASoC: cs42l42: Don't claim to support 192k ASoC: cs42l42: Always configure both ASP TX channels ASoC: cs42l42: Don't reconfigure the PLL while it is running ASoC: cs42l42: Set correct SRC MCLK ASoC: cs42l42: Mark OSC_SWITCH_STATUS register volatile ASoC: cs42l42: Allow time for HP/ADC to power-up after enable ASoC: cs42l42: Prevent NULL pointer deref in interrupt handler ASoC: cs42l42: Remove runtime_suspend/runtime_resume callbacks ASoC: cs42l42: Remove redundant pll_divout member ASoC: cs42l42: Add HP Volume Scale control ASoC: cs42l42: Add control for audio slow-start switch sound/soc/codecs/cs42l42.c | 281 ++++++++++++++++++++++++--------------------- sound/soc/codecs/cs42l42.h | 9 +- 2 files changed, 156 insertions(+), 134 deletions(-) -- 2.11.0

2 18

USB-Audio: Device or resource busy
by Muni Sekhar 11 Aug '21

11 Aug '21

Hi All, $ cat /proc/asound/cards 0 [USB ]: USB-Audio - Plantronics .Audio 628 USB Plantronics Plantronics .Audio 628 USB at usb-0000:00:14.0-2, full speed I am using a Plantronics USB Audio headset. $ arecord --device hw:0,0 --channels 2 --format S16_LE --rate 44100Hz x.wav arecord: main:722: audio open error: Device or resource busy 'arecord' command always fails the first time after system boot in my system. But subsequent execution of the 'arecord' command runs fine. I've attached the strace log for the "audio open error: Device or resource busy" failure. Is there any fix available for this issue? -- Thanks, Sekhar

1 0

Unable to clone repo from windows or yocto
by GitHub issues - edited 11 Aug '21

11 Aug '21

alsa-project/tinycompress issue #11 was edited from boppana113: Attempted to clone the github repo using the following command: - git clone git://git.alsa-project.org/tinycompress.git I got the following error: ``` $ git clone git://git.alsa-project.org/tinycompress.git Cloning into 'tinycompress'... fatal: unable to connect to git.alsa-project.org: git.alsa-project.org[0: 207.180.221.201]: errno=Unknown error ``` The git clone command appears to timeout Issue URL : https://github.com/alsa-project/tinycompress/issues/11 Repository URL: https://github.com/alsa-project/tinycompress

1 0

Unable to clone repo from windows or yocto
by GitHub issues - opened 11 Aug '21

11 Aug '21

alsa-project/tinycompress issue #11 was opened from boppana113: Attempted to clone the github repo using the following command: - git clone git://git.alsa-project.org/tinycompress.git I got the following error: ``` $ git clone git://git.alsa-project.org/tinycompress.git Cloning into 'tinycompress'... fatal: unable to connect to git.alsa-project.org: git.alsa-project.org[0: 207.180.221.201]: errno=Unknown error ``` Issue URL : https://github.com/alsa-project/tinycompress/issues/11 Repository URL: https://github.com/alsa-project/tinycompress

1 0

Re: CORB reset timeout, Intel 3400 onboard chip
by tom 11 Aug '21

11 Aug '21

On 21/08/10 09:21PM, tom wrote: > On 21/08/10 09:16PM, tom wrote: > > On 21/08/10 06:03PM, Pierre-Louis Bossart wrote: > > > > > > > > > On 8/10/21 5:35 PM, tom wrote: > > > > Hello all, > > > > I am experiencing an issue with my onboard sound card, where I have no sound > > > > output. Please see a copy of my alsa-info.sh for my current > > > > /etc/modprobe.d/*.conf config, located here: > > > > > > > > http://ix.io/3vAV > > > > > > > > I attempted troubleshooting using some config options that I saw listed in > > > > HDAudio.txt. Specifically, I tried model options for "generic". I also tried a > > > > few options for probe_mask, including "1" and "0x100". > > > > > > > > > > > > It seems that no matter which options I put in, I always get a > > > > CORB timeout from desg: > > > > > > > > [ 4.052955] snd_hda_intel 0000:00:1b.0: CORB reset timeout#1, CORBRP = 0 > > > > [ 4.053038] snd_hda_intel 0000:00:1b.0: no codecs initialized > > > > > > > > When I remove my probe_mask options, it first switches to polling mode, then > > > > single_cmd mode, and finally fails to initialize any codecs. I looked at the > > > > hda_intel.c file and saw a section for my onboard sound card (ref line 2515): > > > > > > > > 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High > > > > Definition Audio (rev 05) > > > > > > In the alsa-info.sh log, I can see this > > > [ 3.950319] snd_hda_intel 0000:00:1b.0: codec_mask forced to 0x0 > > > > > > I am not sure how this might happen, this looks like a bad configuration > > > to me? > > > > > > /* check forced option */ > > > if (chip->codec_probe_mask != -1 && > > > (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) { > > > azx_bus(chip)->codec_mask = chip->codec_probe_mask & 0xff; > > > dev_info(chip->card->dev, "codec_mask forced to 0x%x\n", > > > (int)azx_bus(chip)->codec_mask); > > > } > > Hello Pierre, My /etc/modprobe.d/alsa-base.conf file looks as follows: options snd_hda_intel probe_mask=0x100 And this was based on a suggestion from "HDAudio.txt" If I comment this line out, I will instead get the following from the command "sudo dmesg | grep -i snd": [ 4.110018] snd_hda_intel 0000:00:1b.0: CORB reset timeout#1, CORBRP = 0 [ 5.114462] snd_hda_intel 0000:00:1b.0: azx_get_response timeout, switching to polling mode: last cmd=0x100f0000 [ 6.124488] snd_hda_intel 0000:00:1b.0: No response from codec, disabling MSI: last cmd=0x100f0000 [ 7.131158] snd_hda_intel 0000:00:1b.0: Codec #1 probe error; disabling it... [ 7.236557] snd_hda_intel 0000:00:1b.0: CORB reset timeout#1, CORBRP = 0 [ 8.244423] snd_hda_intel 0000:00:1b.0: azx_get_respons timeout, switching to single_cmd mode: last cmd=0x100f0000 [ 8.244955] snd_hda_intel 0000:00:1b.0: no codecs initialized I hope this helps!

1 0