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waiting for a connection
by folkert 12 Aug '21

12 Aug '21

Hi, When I create an ALSA MIDI port, is it possible (how?) to wait for someone to connect to it/map it to an other port? This would allow me to only allocate extra resources when needed (in my case: connect to ipmidi-peer (in unicast mode)). regards

2 1

[PATCH v4 00/27] ALSA: hda/cirrus: Split generic cirrus HDA codecs and CS8490 bridge into separate modules.
by Vitaly Rodionov 12 Aug '21

12 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 | 1273 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.h | 369 ++++++++ 6 files changed, 2214 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 28

[PATCH] ALSA: oxfw: fix functioal regression for silence in Apogee Duet FireWire
by Takashi Sakamoto 12 Aug '21

12 Aug '21

OXFW 971 has no function to use the value in syt field of received isochronous packet for playback timing generation. In kernel prepatch for v5.14, ALSA OXFW driver got change to send NO_INFO value in the field instead of actual timing value. The change brings Apogee Duet FireWire to generate no playback sound, while output meter moves. As long as I investigate, _any_ value in the syt field takes the device to generate sound. It's reasonable to think that the device just ignores data blocks in packet with NO_INFO value in its syt field for audio data processing. This commit adds a new flag for the quirk to fix regression. Fixes: 029ffc429440 ("ALSA: oxfw: perform sequence replay for media clock recovery") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-stream.c | 9 ++++++++- sound/firewire/oxfw/oxfw.c | 6 ++++-- sound/firewire/oxfw/oxfw.h | 5 +++++ 3 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 0ef242fdd3bc..fff18b5d4e05 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -153,7 +153,7 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) struct cmp_connection *conn; enum cmp_direction c_dir; enum amdtp_stream_direction s_dir; - unsigned int flags = CIP_UNAWARE_SYT; + unsigned int flags = 0; int err; if (!(oxfw->quirks & SND_OXFW_QUIRK_BLOCKING_TRANSMISSION)) @@ -161,6 +161,13 @@ static int init_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream) else flags |= CIP_BLOCKING; + // OXFW 970/971 has no function to generate playback timing according to the sequence + // of value in syt field, thus the packet should include NO_INFO value in the field. + // However, some models just ignore data blocks in packet with NO_INFO for audio data + // processing. + if (!(oxfw->quirks & SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET)) + flags |= CIP_UNAWARE_SYT; + if (stream == &oxfw->tx_stream) { conn = &oxfw->out_conn; c_dir = CMP_OUTPUT; diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 84971d78d152..cb5b5e3a481b 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -159,8 +159,10 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id return snd_oxfw_scs1x_add(oxfw); } - if (entry->vendor_id == OUI_APOGEE && entry->model_id == MODEL_DUET_FW) - oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + if (entry->vendor_id == OUI_APOGEE && entry->model_id == MODEL_DUET_FW) { + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION | + SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET; + } /* * TASCAM FireOne has physical control and requires a pair of additional diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index ee47abcb0c90..c13034f6c2ca 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -42,6 +42,11 @@ enum snd_oxfw_quirk { SND_OXFW_QUIRK_BLOCKING_TRANSMISSION = 0x04, // Stanton SCS1.d and SCS1.m support unique transaction. SND_OXFW_QUIRK_SCS_TRANSACTION = 0x08, + // Apogee Duet FireWire ignores data blocks in packet with NO_INFO for audio data + // processing, while output level meter moves. Any value in syt field of packet takes + // the device to process audio data even if the value is invalid in a point of + // IEC 61883-1/6. + SND_OXFW_QUIRK_IGNORE_NO_INFO_PACKET = 0x10, }; /* This is an arbitrary number for convinience. */ -- 2.30.2

2 1

[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

[PATCH v2] ASoC: Intel: Fix platform ID matching
by Curtis Malainey 11 Aug '21

11 Aug '21

Sparse warnings triggered truncating the IDs of some platform device tables. Unfortunately some of the IDs in the match tables were missed which breaks audio. The KBL change has been verified to fix audio, the CML change was not tested as it was found through grepping the broken changes and found to match the same situation in anticipation that it should also be fixed. Fixes: 94efd726b947 ("ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters") Fixes: 24e46fb811e9 ("ASoC: Intel: bxt_da7219_max98357a: shrink platform_id below 20 characters") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Tested-by: Matt Davis <mattedavis(a)google.com> --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 42ef51c3fb4f..b591c6fd13fd 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -75,7 +75,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "DLGS7219", - .drv_name = "cml_da7219_max98357a", + .drv_name = "cml_da7219_mx98357a", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98390_spk_codecs, .sof_fw_filename = "sof-cml.ri", diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index ba5ff468c265..741bf2f9e081 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -87,7 +87,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", + .drv_name = "kbl_da7219_mx98357a", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98357_codecs, -- 2.32.0.605.g8dce9f2422-goog

5 4

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

11 Aug '21

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) I also want to share my hw-probe output, located here: https://linux-hardware.org/?probe=13e2d05856 It appears there are approximately 50 other users utilizing the same audio driver. From my searches across a few different websites, it seems that my particular audio chip should be very well supported and simply "work". I am now thinking this is a bug, as I can't see a reason why it should be failing in this manner. Thank you for any guidance you can provide.

2 1

Fix alsa-utils building error on NetBSD
by GitHub pull_request - opened 10 Aug '21

10 Aug '21

alsa-project/alsa-lib pull request #169 was opened from triaxx: The following errors occur at least on NetBSD 9.2 amd64 when building alsa-utils: include/alsa/output.h:75:66: error: unknown type name 'va_list'; did you mean '__va_list'? include/alsa/error.h:80:25: error: unknown type name 'va_list'; did you mean '__va_list'? Request URL : https://github.com/alsa-project/alsa-lib/pull/169 Patch URL : https://github.com/alsa-project/alsa-lib/pull/169.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: dt-bindings: rt1015p: fix syntax error in dts-binding document
by Jack Yu 10 Aug '21

10 Aug '21

Fix syntax error in dts-binding document. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 8fcb5f79a1b5..f31d3c4d0192 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -16,8 +16,8 @@ description: | properties: compatible: oneOf: - const: realtek,rt1015p - const: realtek,rt1019p + - const: realtek,rt1015p + - const: realtek,rt1019p sdb-gpios: description: -- 2.31.1

3 2

[PATCH 1/2] ALSA: hda: Release controller display power during shutdown/reboot
by Imre Deak 10 Aug '21

10 Aug '21

Make sure the HDA driver's display power reference is released during shutdown/reboot. During the shutdown/reboot sequence the pci device core calls the pm_runtime_resume handler for all devices before calling the driver's shutdown callback and so the HDA driver's runtime resume callback will acquire a display power reference (on HSW/BDW). This triggers a power reference held WARN on HSW/BDW in the i915 driver's subsequent shutdown handler, which expects all display power references to be released by that time. Since the HDA controller is stopped in the shutdown handler in any case, let's follow here the same sequence as the one during runtime suspend. This will also reset the HDA link and drop the display power reference, getting rid of the above WARN. Tested on HSW. v2: - Fix the build for CONFIG_PM=n (Takashi) - s/__azx_runtime_suspend/azx_shutdown_chip/ Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/3618 References: https://lore.kernel.org/lkml/cea1f9a-52e0-b83-593d-52997fe1aaf6@er-systems.… Reported-and-tested-by: Thomas Voegtle <tv(a)lio96.de> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Imre Deak <imre.deak(a)intel.com> --- sound/pci/hda/hda_intel.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7f8f11536a3dc..f5ab0b682adcc 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -883,6 +883,14 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) return azx_get_pos_posbuf(chip, azx_dev); } +static void azx_shutdown_chip(struct azx *chip) +{ + azx_stop_chip(chip); + azx_enter_link_reset(chip); + azx_clear_irq_pending(chip); + display_power(chip, false); +} + #ifdef CONFIG_PM static DEFINE_MUTEX(card_list_lock); static LIST_HEAD(card_list); @@ -942,14 +950,6 @@ static bool azx_is_pm_ready(struct snd_card *card) return true; } -static void __azx_runtime_suspend(struct azx *chip) -{ - azx_stop_chip(chip); - azx_enter_link_reset(chip); - azx_clear_irq_pending(chip); - display_power(chip, false); -} - static void __azx_runtime_resume(struct azx *chip) { struct hda_intel *hda = container_of(chip, struct hda_intel, chip); @@ -1028,7 +1028,7 @@ static int azx_suspend(struct device *dev) chip = card->private_data; bus = azx_bus(chip); - __azx_runtime_suspend(chip); + azx_shutdown_chip(chip); if (bus->irq >= 0) { free_irq(bus->irq, chip); bus->irq = -1; @@ -1107,7 +1107,7 @@ static int azx_runtime_suspend(struct device *dev) /* enable controller wake up event */ azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | STATESTS_INT_MASK); - __azx_runtime_suspend(chip); + azx_shutdown_chip(chip); trace_azx_runtime_suspend(chip); return 0; } @@ -2383,7 +2383,7 @@ static void azx_shutdown(struct pci_dev *pci) return; chip = card->private_data; if (chip && chip->running) - azx_stop_chip(chip); + azx_shutdown_chip(chip); } /* PCI IDs */ -- 2.27.0

4 4

[PATCH] ALSA: hda/realtek: fix mute/micmute LEDs for HP ProBook 650 G8 Notebook PC
by Jeremy Szu 10 Aug '21

10 Aug '21

The HP ProBook 650 G8 Notebook PC is using ALC236 codec which is using 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Jeremy Szu <jeremy.szu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4043a2362f27..a065260d0d20 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8431,6 +8431,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f4, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.25.1

2 1

[PATCH RFC 0/5] ALSA: Support for non-coherent and non-contiguous page allocation
by Takashi Iwai 10 Aug '21

10 Aug '21

Hi, this is an experimental patch set to add the support for PCM buffers with non-coherent and non-contiguous pages, typically useful for non-x86 architectures. The first patch improves the SG-buffer handling, then add a new PCM info flag that disables the control and status mmap, and implements two new buffer types, SNDRV_DMA_TYPE_NONCONTIG and SNDRV_DMA_TYPE_NONCOHERENT. The former is the SG-buffer and the latter is the continuous page allocation, corresponding to SNDRV_DMA_TYPE_DEV_SG and SNDRV_DMA_TYPE_DEV on x86. Unlike other page types, those are directional (that need the DMA direction at allocation time) and require the explicit sync of buffers around the data transfer (flushing, invalidating). The sync is implemented inside ALSA PCM core and automatically applied at updating the applptr and hwsync via SYNC_PTR ioctl, which should be issued during mmap operation, so it should work transparently as long as applications are running with alsa-lib. For tinyALSA, we might need to revisit the implementation. This is currently an RFC -- more exactly, CALL FOR TESTERS, as I have no Arm machine with the sound device for now. The needed change for an existing driver is simple: * Replace the buffer type from SNDRV_DMA_TYPE_DEV to SNDRV_DMA_TYPE_NONCOHERENT (or from SNDRV_DMA_TYPE_DEV_SG to SNDRV_DMA_TYPE_NONCONTIG), which is set up typically in snd_pcm_set_managed_buffer*() call. * Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag to the PCM runtime hardware.info field It'd be greatly appreciated if anyone can test and try the changes, and help debugging. Again, the code is faily untested. The latest code is found in topic/memalloc-noncontig branch of my sound git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git The patch set is based on for-next branch, i.e. changes for 5.15 are included. thanks, Takashi === Takashi Iwai (5): ALSA: memalloc: Count continuous pages in vmalloc buffer handler ALSA: pcm: Add SNDRV_PCM_INFO_EXPLICIT_SYNC flag ALSA: memalloc: Assign ops field to snd_dma_buffer ALSA: memalloc: Support for non-contiguous page allocation ALSA: memalloc: Support for non-coherent page allocation include/sound/memalloc.h | 48 +++++++- include/uapi/sound/asound.h | 1 + sound/core/memalloc.c | 223 ++++++++++++++++++++++++++++-------- sound/core/memalloc_local.h | 1 + sound/core/pcm_lib.c | 9 ++ sound/core/pcm_memory.c | 13 ++- sound/core/pcm_native.c | 22 ++++ 7 files changed, 263 insertions(+), 54 deletions(-) -- 2.26.2

1 5

[PATCH] ASoC: Intel: Fix platform ID matching
by Curtis Malainey 10 Aug '21

10 Aug '21

Sparse warnings triggered truncating the IDs of some platform device tables. Unfortunately some of the IDs in the match tables were missed which breaks audio. The KBL change has been verified to fix audio, the CML change was not tested as it was found through grepping the broken changes and found to match the same situation in anticipation that it should also be fixed. Fixes: 94efd726b947 ("ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters") Fixes: 24e46fb811e9 ("ASoC: Intel: bxt_da7219_max98357a: shrink platform_id below 20 characters") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Tested-by: Matt Davis <mattedavis(a)chromium.org> --- sound/soc/intel/common/soc-acpi-intel-cml-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cml-match.c b/sound/soc/intel/common/soc-acpi-intel-cml-match.c index 42ef51c3fb4f..b591c6fd13fd 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cml-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cml-match.c @@ -75,7 +75,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_machines[] = { }, { .id = "DLGS7219", - .drv_name = "cml_da7219_max98357a", + .drv_name = "cml_da7219_mx98357a", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &max98390_spk_codecs, .sof_fw_filename = "sof-cml.ri", diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index ba5ff468c265..741bf2f9e081 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -87,7 +87,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", + .drv_name = "kbl_da7219_mx98357a", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98357_codecs, -- 2.32.0.605.g8dce9f2422-goog

1 0

[PATCH 1/2] ASoC: rt1015p: add new acpi id and comapatible id
by Jack Yu 09 Aug '21

09 Aug '21

Add new acpi id and compatible id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015p.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 40f2063aefbe..415cfb3b2f0d 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -127,6 +127,7 @@ static int rt1015p_platform_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id rt1015p_device_id[] = { { .compatible = "realtek,rt1015p" }, + { .compatible = "realtek,rt1019p" }, {} }; MODULE_DEVICE_TABLE(of, rt1015p_device_id); @@ -135,6 +136,7 @@ MODULE_DEVICE_TABLE(of, rt1015p_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt1015p_acpi_match[] = { { "RTL1015", 0}, + { "RTL1019", 0}, { }, }; MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); -- 2.31.1

2 1

[PATCH v2] ASoC: max98390: Add support change dsm param name
by Steve Lee 09 Aug '21

09 Aug '21

In case of using different type of speaker, support using different dsm parameter bin file for each amp connected. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 26 ++++++++++++++++++++------ sound/soc/codecs/max98390.h | 1 + 2 files changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 94773ccee9d5..b392567c2b3e 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -765,17 +765,26 @@ static int max98390_dsm_init(struct snd_soc_component *component) vendor = dmi_get_system_info(DMI_SYS_VENDOR); product = dmi_get_system_info(DMI_PRODUCT_NAME); - if (vendor && product) { - snprintf(filename, sizeof(filename), "dsm_param_%s_%s.bin", - vendor, product); + if (!strcmp(max98390->dsm_param_name, "default")) { + if (vendor && product) { + snprintf(filename, sizeof(filename), + "dsm_param_%s_%s.bin", vendor, product); + } else { + sprintf(filename, "dsm_param.bin"); + } } else { - sprintf(filename, "dsm_param.bin"); + snprintf(filename, sizeof(filename), "%s", + max98390->dsm_param_name); } ret = request_firmware(&fw, filename, component->dev); if (ret) { ret = request_firmware(&fw, "dsm_param.bin", component->dev); - if (ret) - goto err; + if (ret) { + ret = request_firmware(&fw, "dsmparam.bin", + component->dev); + if (ret) + goto err; + } } dev_dbg(component->dev, @@ -1047,6 +1056,11 @@ static int max98390_i2c_probe(struct i2c_client *i2c, __func__, max98390->ref_rdc_value, max98390->ambient_temp_value); + ret = device_property_read_string(&i2c->dev, "maxim,dsm_param_name", + &max98390->dsm_param_name); + if (ret) + max98390->dsm_param_name = "default"; + /* voltage/current slot configuration */ max98390_slot_config(i2c, max98390); diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index e31516717d3b..c250740f73a2 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -662,5 +662,6 @@ struct max98390_priv { unsigned int i_l_slot; unsigned int ref_rdc_value; unsigned int ambient_temp_value; + const char *dsm_param_name; }; #endif -- 2.17.1

2 1

[PATCH 2/2] ASoC: dt-bindings: rt1015p: add new compatible id
by Jack Yu 09 Aug '21

09 Aug '21

Add new compatible ID for rt1015p in dt-bindings document. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 644b68edf3e1..8fcb5f79a1b5 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -15,7 +15,9 @@ description: | properties: compatible: - const: realtek,rt1015p + oneOf: + const: realtek,rt1015p + const: realtek,rt1019p sdb-gpios: description: -- 2.31.1

3 2

[PATCH] ASoC: rt1015p: add new acpi id and comapatible id
by Jack Yu 09 Aug '21

09 Aug '21

Add new acpi id and compatible id for rt1015p. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml | 4 +++- sound/soc/codecs/rt1015p.c | 2 ++ 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml index 644b68edf3e1..b898dfa4eb83 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt1015p.yaml @@ -15,7 +15,9 @@ description: | properties: compatible: - const: realtek,rt1015p + oneOf: + - const: realtek,rt1015p + - const: realtek,rt1019p sdb-gpios: description: diff --git a/sound/soc/codecs/rt1015p.c b/sound/soc/codecs/rt1015p.c index 39f4c4ed2ab8..085a7b3c7bb4 100644 --- a/sound/soc/codecs/rt1015p.c +++ b/sound/soc/codecs/rt1015p.c @@ -126,6 +126,7 @@ static int rt1015p_platform_probe(struct platform_device *pdev) #ifdef CONFIG_OF static const struct of_device_id rt1015p_device_id[] = { { .compatible = "realtek,rt1015p" }, + { .compatible = "realtek,rt1019p" }, {} }; MODULE_DEVICE_TABLE(of, rt1015p_device_id); @@ -134,6 +135,7 @@ MODULE_DEVICE_TABLE(of, rt1015p_device_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt1015p_acpi_match[] = { { "RTL1015", 0}, + { "RTL1019", 0}, { }, }; MODULE_DEVICE_TABLE(acpi, rt1015p_acpi_match); -- 2.29.0

2 1

[PATCH] soundwire: intel: trap TRIGGER_SUSPEND in .trigger callback
by Bard Liao 09 Aug '21

09 Aug '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> This patch provides both a simplification of the suspend flows and a better balanced operation during suspend/resume transition. The exiting code relies on a convoluted way of dealing with suspend signals. Since there is no .suspend DAI callback, we used the component .suspend and marked all the component DAI dmas as 'suspended'. The information was used in the .prepare stage to differentiate resume operations from xrun handling, and only reinitialize SHIM registers and DMA in the former case. While this solution has been working reliably for about 2 years, there is a much better solution consisting in trapping the TRIGGER_SUSPEND in the .trigger DAI ops. The DMA is still marked in the same way for the .prepare op to run, but in addition the callbacks sent to DSP firmware are now balanced. Normal operation: hw_params -> intel_params_stream hw_free -> intel_free_stream suspend -> intel_free_stream prepare -> intel_params_stream This balanced operation was not required with existing SOF firmware relying on static pipelines instantiated at every boot. With the on-going transition to dynamic pipelines, it's however a requirement to keep the use count for the DAI widget balanced across all transitions. Co-developed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 53 +++++++++++++++++++++------------------ 1 file changed, 29 insertions(+), 24 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index fb9c23e13206..a0178779a5ba 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1005,29 +1005,6 @@ static void intel_shutdown(struct snd_pcm_substream *substream, pm_runtime_put_autosuspend(cdns->dev); } -static int intel_component_dais_suspend(struct snd_soc_component *component) -{ - struct sdw_cdns_dma_data *dma; - struct snd_soc_dai *dai; - - for_each_component_dais(component, dai) { - /* - * we don't have a .suspend dai_ops, and we don't have access - * to the substream, so let's mark both capture and playback - * DMA contexts as suspended - */ - dma = dai->playback_dma_data; - if (dma) - dma->suspended = true; - - dma = dai->capture_dma_data; - if (dma) - dma->suspended = true; - } - - return 0; -} - static int intel_pcm_set_sdw_stream(struct snd_soc_dai *dai, void *stream, int direction) { @@ -1056,11 +1033,39 @@ static void *intel_get_sdw_stream(struct snd_soc_dai *dai, return dma->stream; } +static int intel_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) +{ + struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); + struct sdw_intel *sdw = cdns_to_intel(cdns); + struct sdw_cdns_dma_data *dma; + + /* + * The .prepare callback is used to deal with xruns and resume operations. In the case + * of xruns, the DMAs and SHIM registers cannot be touched, but for resume operations the + * DMAs and SHIM registers need to be initialized. + * the .trigger callback is used to track the suspend case only. + */ + if (cmd != SNDRV_PCM_TRIGGER_SUSPEND) + return 0; + + dma = snd_soc_dai_get_dma_data(dai, substream); + if (!dma) { + dev_err(dai->dev, "failed to get dma data in %s\n", + __func__); + return -EIO; + } + + dma->suspended = true; + + return intel_free_stream(sdw, substream, dai, sdw->instance); +} + static const struct snd_soc_dai_ops intel_pcm_dai_ops = { .startup = intel_startup, .hw_params = intel_hw_params, .prepare = intel_prepare, .hw_free = intel_hw_free, + .trigger = intel_trigger, .shutdown = intel_shutdown, .set_sdw_stream = intel_pcm_set_sdw_stream, .get_sdw_stream = intel_get_sdw_stream, @@ -1071,6 +1076,7 @@ static const struct snd_soc_dai_ops intel_pdm_dai_ops = { .hw_params = intel_hw_params, .prepare = intel_prepare, .hw_free = intel_hw_free, + .trigger = intel_trigger, .shutdown = intel_shutdown, .set_sdw_stream = intel_pdm_set_sdw_stream, .get_sdw_stream = intel_get_sdw_stream, @@ -1078,7 +1084,6 @@ static const struct snd_soc_dai_ops intel_pdm_dai_ops = { static const struct snd_soc_component_driver dai_component = { .name = "soundwire", - .suspend = intel_component_dais_suspend }; static int intel_create_dai(struct sdw_cdns *cdns, -- 2.17.1

6 15

[PATCH 0/2] ASoC: qdsp6: cppcheck warnings
by Srinivas Kandagatla 09 Aug '21

09 Aug '21

cppcheck throws below two warnings in qdsp6 code due to unnecessary initialization of variables q6asm.c:1631: (style) Variable 'port' is reassigned a value before the old one has been used. q6adm.c:475]: (style) Variable 'matrix_map' is reassigned a value before the old one has been used. These series fixes it by removing those unnecessary initializations Srinivas Kandagatla (2): ASoC: qdsp6: q6asm: fix cppcheck warnings for unnecessary initialization ASoC: qdsp6: q6adm: fix cppcheck warnings for unnecessary initialization sound/soc/qcom/qdsp6/q6adm.c | 2 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.21.0

2 3

[PATCH v2] ALSA: msnd: Use proper mmap method
by Takashi Iwai 09 Aug '21

09 Aug '21

The old ISA MSND driver basically maps the iomem as the DMA buffer. For this type of buffer, we have already the standard mmap helper, snd_pcm_lib_mmap_iomem(). Correct the PCM info, set the DMA address and use the standard helper for handling the mmap on all architectures properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/msnd/msnd.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 4fbc22a5bc5a..c3fd1eb301bb 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -425,7 +425,7 @@ static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, } static const struct snd_pcm_hardware snd_msnd_playback = { - .info = SNDRV_PCM_INFO_MMAP | + .info = SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH, @@ -444,7 +444,7 @@ static const struct snd_pcm_hardware snd_msnd_playback = { }; static const struct snd_pcm_hardware snd_msnd_capture = { - .info = SNDRV_PCM_INFO_MMAP | + .info = SNDRV_PCM_INFO_MMAP_IOMEM | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH, @@ -473,6 +473,7 @@ static int snd_msnd_playback_open(struct snd_pcm_substream *substream) snd_msnd_enable_irq(chip); runtime->dma_area = (__force void *)chip->mappedbase; + runtime->dma_addr = chip->base; runtime->dma_bytes = 0x3000; chip->playback_substream = substream; @@ -566,6 +567,7 @@ static const struct snd_pcm_ops snd_msnd_playback_ops = { .prepare = snd_msnd_playback_prepare, .trigger = snd_msnd_playback_trigger, .pointer = snd_msnd_playback_pointer, + .mmap = snd_pcm_lib_mmap_iomem, }; static int snd_msnd_capture_open(struct snd_pcm_substream *substream) @@ -576,6 +578,7 @@ static int snd_msnd_capture_open(struct snd_pcm_substream *substream) set_bit(F_AUDIO_READ_INUSE, &chip->flags); snd_msnd_enable_irq(chip); runtime->dma_area = (__force void *)chip->mappedbase + 0x3000; + runtime->dma_addr = chip->base + 0x3000; runtime->dma_bytes = 0x3000; memset(runtime->dma_area, 0, runtime->dma_bytes); chip->capture_substream = substream; @@ -662,6 +665,7 @@ static const struct snd_pcm_ops snd_msnd_capture_ops = { .prepare = snd_msnd_capture_prepare, .trigger = snd_msnd_capture_trigger, .pointer = snd_msnd_capture_pointer, + .mmap = snd_pcm_lib_mmap_iomem, }; -- 2.26.2

1 0

[PATCH] ALSA: msnd: Use proper mmap method
by Takashi Iwai 09 Aug '21

09 Aug '21

The old ISA MSND driver basically maps the iomem as the DMA buffer. For this type of buffer, we have already the standard mmap helper, snd_pcm_lib_mmap_iomem(). Set the DMA address and use the standard helper for allowing the mmap on all architectures properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/isa/msnd/msnd.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c index 4fbc22a5bc5a..e23eaf449c50 100644 --- a/sound/isa/msnd/msnd.c +++ b/sound/isa/msnd/msnd.c @@ -473,6 +473,7 @@ static int snd_msnd_playback_open(struct snd_pcm_substream *substream) snd_msnd_enable_irq(chip); runtime->dma_area = (__force void *)chip->mappedbase; + runtime->dma_addr = chip->base; runtime->dma_bytes = 0x3000; chip->playback_substream = substream; @@ -566,6 +567,7 @@ static const struct snd_pcm_ops snd_msnd_playback_ops = { .prepare = snd_msnd_playback_prepare, .trigger = snd_msnd_playback_trigger, .pointer = snd_msnd_playback_pointer, + .mmap = snd_pcm_lib_mmap_iomem, }; static int snd_msnd_capture_open(struct snd_pcm_substream *substream) @@ -576,6 +578,7 @@ static int snd_msnd_capture_open(struct snd_pcm_substream *substream) set_bit(F_AUDIO_READ_INUSE, &chip->flags); snd_msnd_enable_irq(chip); runtime->dma_area = (__force void *)chip->mappedbase + 0x3000; + runtime->dma_addr = chip->base + 0x3000; runtime->dma_bytes = 0x3000; memset(runtime->dma_area, 0, runtime->dma_bytes); chip->capture_substream = substream; @@ -662,6 +665,7 @@ static const struct snd_pcm_ops snd_msnd_capture_ops = { .prepare = snd_msnd_capture_prepare, .trigger = snd_msnd_capture_trigger, .pointer = snd_msnd_capture_pointer, + .mmap = snd_pcm_lib_mmap_iomem, }; -- 2.26.2

1 1

[PATCH] ASoC: max98390: Add support change dsm param name
by Steve Lee 09 Aug '21

09 Aug '21

In case of using different type of speaker, support using different dsm parameter bin file for each amp connected. Signed-off-by: Steve Lee <steves.lee(a)maximintegrated.com> --- sound/soc/codecs/max98390.c | 20 +++++++++++++++----- sound/soc/codecs/max98390.h | 1 + 2 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 94773ccee9d5..82d2dd52c17d 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -765,15 +765,20 @@ static int max98390_dsm_init(struct snd_soc_component *component) vendor = dmi_get_system_info(DMI_SYS_VENDOR); product = dmi_get_system_info(DMI_PRODUCT_NAME); - if (vendor && product) { - snprintf(filename, sizeof(filename), "dsm_param_%s_%s.bin", - vendor, product); + if (!strcmp(max98390->dsm_param_name, "default")) { + if (vendor && product) { + snprintf(filename, sizeof(filename), + "dsm_param_%s_%s.bin", vendor, product); + } else { + sprintf(filename, "dsmparam.bin"); + } } else { - sprintf(filename, "dsm_param.bin"); + snprintf(filename, sizeof(filename), "%s", + max98390->dsm_param_name); } ret = request_firmware(&fw, filename, component->dev); if (ret) { - ret = request_firmware(&fw, "dsm_param.bin", component->dev); + ret = request_firmware(&fw, "dsmparam.bin", component->dev); if (ret) goto err; } @@ -1047,6 +1052,11 @@ static int max98390_i2c_probe(struct i2c_client *i2c, __func__, max98390->ref_rdc_value, max98390->ambient_temp_value); + ret = device_property_read_string(&i2c->dev, "maxim,dsm_param_name", + &max98390->dsm_param_name); + if (ret) + max98390->dsm_param_name = "default"; + /* voltage/current slot configuration */ max98390_slot_config(i2c, max98390); diff --git a/sound/soc/codecs/max98390.h b/sound/soc/codecs/max98390.h index e31516717d3b..c250740f73a2 100644 --- a/sound/soc/codecs/max98390.h +++ b/sound/soc/codecs/max98390.h @@ -662,5 +662,6 @@ struct max98390_priv { unsigned int i_l_slot; unsigned int ref_rdc_value; unsigned int ambient_temp_value; + const char *dsm_param_name; }; #endif -- 2.17.1

2 4

[PATCH v2] ASoC: qcom: apq8016_sbc: Add SEC_MI2S support
by Vincent Knecht 09 Aug '21

09 Aug '21

This patch adds external codec support on secondary mi2s. It is used for headphones on some devices, eg. alcatel-idol347. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- v1->v2: thanks Srinivas for the review, and Stephan for guidance - Add _SEC suffix to defines to highlight usage for secondary mi2s - Clear TLMM_WS_OUT_SEL and TLMM_WS_EN_SEL fields before setting value --- sound/soc/qcom/apq8016_sbc.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 08a05f0ecad7..ba2a98268ee4 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -30,6 +30,13 @@ struct apq8016_sbc_data { #define MIC_CTRL_QUA_WS_SLAVE_SEL_10 BIT(17) #define MIC_CTRL_TLMM_SCLK_EN BIT(1) #define SPKR_CTL_PRI_WS_SLAVE_SEL_11 (BIT(17) | BIT(16)) +#define SPKR_CTL_TLMM_MCLK_EN BIT(1) +#define SPKR_CTL_TLMM_SCLK_EN BIT(2) +#define SPKR_CTL_TLMM_DATA1_EN BIT(3) +#define SPKR_CTL_TLMM_WS_OUT_SEL_MASK GENMASK(7, 6) +#define SPKR_CTL_TLMM_WS_OUT_SEL_SEC BIT(6) +#define SPKR_CTL_TLMM_WS_EN_SEL_MASK GENMASK(19, 18) +#define SPKR_CTL_TLMM_WS_EN_SEL_SEC BIT(18) #define DEFAULT_MCLK_RATE 9600000 static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) @@ -40,6 +47,7 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); int i, rval; + u32 value; switch (cpu_dai->id) { case MI2S_PRIMARY: @@ -53,6 +61,15 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) MIC_CTRL_TLMM_SCLK_EN, pdata->mic_iomux); break; + case MI2S_SECONDARY: + /* Clear TLMM_WS_OUT_SEL and TLMM_WS_EN_SEL fields */ + value = readl(pdata->spkr_iomux) & + ~(SPKR_CTL_TLMM_WS_OUT_SEL_MASK | SPKR_CTL_TLMM_WS_EN_SEL_MASK); + /* Configure the Sec MI2S to TLMM */ + writel(value | SPKR_CTL_TLMM_MCLK_EN | SPKR_CTL_TLMM_SCLK_EN | + SPKR_CTL_TLMM_DATA1_EN | SPKR_CTL_TLMM_WS_OUT_SEL_SEC | + SPKR_CTL_TLMM_WS_EN_SEL_SEC, pdata->spkr_iomux); + break; case MI2S_TERTIARY: writel(readl(pdata->mic_iomux) | MIC_CTRL_TER_WS_SLAVE_SEL | MIC_CTRL_TLMM_SCLK_EN, -- 2.31.1

3 2

Re: [PATCH] ASoC: rt1015p: add new acpi id and comapatible id
by Mark Brown 09 Aug '21

09 Aug '21

On Mon, Aug 09, 2021 at 08:52:49AM +0000, Jack Yu wrote: > Hi Mark, > > Could you help to review below patch? > https://patchwork.kernel.org/project/alsa-devel/patch/20210317062309.14051-… > I'm not sure if you miss this patch or not. Please don't send content free pings. Sending content free pings adds to the mail volume (if they are seen at all) which is often the problem and since they can't be reviewed directly if something has gone wrong you'll have to resend the patches anyway, so sending again is generally a better approach though there are some other maintainers who like them - if in doubt look at how patches for the subsystem are normally handled.

1 0

[PATCH v3 0/9] ASoC: mediatek: Add support for MT8195 SoC
by Trevor Wu 09 Aug '21

09 Aug '21

This series of patches adds support for Mediatek AFE of MT8195 SoC. Patches are based on broonie tree "for-next" branch. Changes since v2: - added audio clock gate control - added 'mediatek' prefix to private dts properties - added consumed clocks to dt-bindins and adopted suggestions from Rob - refined clock usage and remove unused clock and control code - fixed typos Changes since v1: - fixed some problems related to dt-bindings - added some missing properties to dt-bindings - added depency declaration on dt-bindings - fixed some warnings found by kernel test robot Trevor Wu (9): ASoC: mediatek: mt8195: update mediatek common driver ASoC: mediatek: mt8195: support audsys clock control ASoC: mediatek: mt8195: support etdm in platform driver ASoC: mediatek: mt8195: support adda in platform driver ASoC: mediatek: mt8195: support pcm in platform driver ASoC: mediatek: mt8195: add platform driver dt-bindings: mediatek: mt8195: add audio afe document ASoC: mediatek: mt8195: add machine driver with mt6359, rt1019 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1019-rt5682 document .../bindings/sound/mt8195-afe-pcm.yaml | 184 + .../sound/mt8195-mt6359-rt1019-rt5682.yaml | 39 + sound/soc/mediatek/Kconfig | 23 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 22 +- sound/soc/mediatek/common/mtk-base-afe.h | 10 +- sound/soc/mediatek/mt8195/Makefile | 15 + sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 499 +++ sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 113 + sound/soc/mediatek/mt8195/mt8195-afe-common.h | 158 + sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3283 +++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 217 ++ sound/soc/mediatek/mt8195/mt8195-audsys-clk.h | 15 + .../soc/mediatek/mt8195/mt8195-audsys-clkid.h | 93 + sound/soc/mediatek/mt8195/mt8195-dai-adda.c | 830 +++++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 2652 +++++++++++++ sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 389 ++ .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 977 +++++ sound/soc/mediatek/mt8195/mt8195-reg.h | 2796 ++++++++++++++ 19 files changed, 12311 insertions(+), 5 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1019-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/Makefile create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-adda.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-etdm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h -- 2.18.0

2 10

Re: [PATCH] ASoC: rt1015p: add new acpi id and comapatible id
by Tzung-Bi Shih 09 Aug '21

09 Aug '21

On Mon, Aug 09, 2021 at 08:52:49AM +0000, Jack Yu wrote: > Could you help to review below patch? > https://patchwork.kernel.org/project/alsa-devel/patch/20210317062309.14051-… The patch has been few months ago. You should just resend the patch.

1 0

[PATCH 0/3] ALSA: pcm: More mmap coverages
by Takashi Iwai 09 Aug '21

09 Aug '21

Hi, the recent regression fix enlightened the potentially missing pieces in a couple of drivers for mmap on non-x86 architectures. This patch series tries to cover those. Takashi === Takashi Iwai (3): ALSA: pcm: Check mmap capability of runtime dma buffer at first ALSA: pci: rme: Set up buffer type properly ALSA: pci: cs46xx: Fix set up buffer type properly sound/core/pcm_native.c | 9 +++++++-- sound/pci/cs46xx/cs46xx_lib.c | 30 ++++++++---------------------- sound/pci/rme9652/hdsp.c | 6 ++---- sound/pci/rme9652/rme9652.c | 6 ++---- 4 files changed, 19 insertions(+), 32 deletions(-) -- 2.26.2

1 3

[PATCH] ALSA: harmony: Drop superfluous address setup
by Takashi Iwai 09 Aug '21

09 Aug '21

Harmony driver never uses CONTINUOUS buffer type any longer, hence the code setting the address is superfluous. Moreover, the recent change in the ALSA core already covered the missing address field; it's already set up. So let's drop the superfluous line, which includes the full hw_params callback. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/parisc/harmony.c | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 1440db8b4177..ec4fda324760 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -563,22 +563,9 @@ snd_harmony_capture_close(struct snd_pcm_substream *ss) return 0; } -static int -snd_harmony_hw_params(struct snd_pcm_substream *ss, - struct snd_pcm_hw_params *hw) -{ - struct snd_harmony *h = snd_pcm_substream_chip(ss); - - if (h->dma.type == SNDRV_DMA_TYPE_CONTINUOUS) - ss->runtime->dma_addr = __pa(ss->runtime->dma_area); - - return 0; -} - static const struct snd_pcm_ops snd_harmony_playback_ops = { .open = snd_harmony_playback_open, .close = snd_harmony_playback_close, - .hw_params = snd_harmony_hw_params, .prepare = snd_harmony_playback_prepare, .trigger = snd_harmony_playback_trigger, .pointer = snd_harmony_playback_pointer, @@ -587,7 +574,6 @@ static const struct snd_pcm_ops snd_harmony_playback_ops = { static const struct snd_pcm_ops snd_harmony_capture_ops = { .open = snd_harmony_capture_open, .close = snd_harmony_capture_close, - .hw_params = snd_harmony_hw_params, .prepare = snd_harmony_capture_prepare, .trigger = snd_harmony_capture_trigger, .pointer = snd_harmony_capture_pointer, -- 2.26.2

1 0

Re: Kernel 5.13.6 breaks mmap with snd-hdsp module
by Greg KH 09 Aug '21

09 Aug '21

On Sat, Aug 07, 2021 at 12:49:07AM -0700, Jeff Woods wrote: > Specifically, commit c4824ae7db418aee6f50f308a20b832e58e997fd triggers the problem. Reverting this change restores functionality. > > The device is an RME Multiface II, using the snd-hdsp driver. > > Expected behavior: Device plays sound normally > > Exhibited behavior: When a program attempts to open the device, the following ALSA lib error happens: > > ALSA lib pcm_direct.c:1169:(snd1_pcm_direct_initialize_slave) slave plugin does not support mmap interleaved or mmap noninterleaved access > > This change hasn't affected my other computers with less esoteric hardware, so probably the problem lies with the snd-hdsp driver, but the device is unusable without reverting that commit. > > I am available to test any patches for this issue. Have you notified the developers involved in this change about this issue? Adding them now... thanks, greg k-h

3 6

[PATCH v2] parisc: Make struct parisc_driver::remove() return void
by Uwe Kleine-König 09 Aug '21

09 Aug '21

The caller of this function (parisc_driver_remove() in arch/parisc/kernel/drivers.c) ignores the return value, so better don't return any value at all to not wake wrong expectations in driver authors. The only function that could return a non-zero value before was ipmi_parisc_remove() which returns the return value of ipmi_si_remove_by_dev(). Make this function return void, too, as for all other callers the value is ignored, too. Also fold in a small checkpatch fix for: WARNING: Unnecessary space before function pointer arguments + void (*remove) (struct parisc_device *dev); Acked-by: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> (for drivers/input) Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- changes since v1 sent with Message-Id: 20210806093938.1950990-1-u.kleine-koenig(a)pengutronix.de: - Fix a compiler error noticed by the kernel test robot - Add Ack for Dmitry arch/parisc/include/asm/parisc-device.h | 4 ++-- drivers/char/ipmi/ipmi_si.h | 2 +- drivers/char/ipmi/ipmi_si_intf.c | 6 +----- drivers/char/ipmi/ipmi_si_parisc.c | 4 ++-- drivers/char/ipmi/ipmi_si_platform.c | 4 +++- drivers/input/keyboard/hilkbd.c | 4 +--- drivers/input/serio/gscps2.c | 3 +-- drivers/net/ethernet/i825xx/lasi_82596.c | 3 +-- drivers/parport/parport_gsc.c | 3 +-- drivers/scsi/lasi700.c | 4 +--- drivers/scsi/zalon.c | 4 +--- drivers/tty/serial/mux.c | 3 +-- sound/parisc/harmony.c | 3 +-- 13 files changed, 17 insertions(+), 30 deletions(-) diff --git a/arch/parisc/include/asm/parisc-device.h b/arch/parisc/include/asm/parisc-device.h index d02d144c6012..4de3b391d812 100644 --- a/arch/parisc/include/asm/parisc-device.h +++ b/arch/parisc/include/asm/parisc-device.h @@ -34,8 +34,8 @@ struct parisc_driver { struct parisc_driver *next; char *name; const struct parisc_device_id *id_table; - int (*probe) (struct parisc_device *dev); /* New device discovered */ - int (*remove) (struct parisc_device *dev); + int (*probe)(struct parisc_device *dev); /* New device discovered */ + void (*remove)(struct parisc_device *dev); struct device_driver drv; }; diff --git a/drivers/char/ipmi/ipmi_si.h b/drivers/char/ipmi/ipmi_si.h index 0a4c69539f24..a7ead2a4c753 100644 --- a/drivers/char/ipmi/ipmi_si.h +++ b/drivers/char/ipmi/ipmi_si.h @@ -73,7 +73,7 @@ irqreturn_t ipmi_si_irq_handler(int irq, void *data); void ipmi_irq_start_cleanup(struct si_sm_io *io); int ipmi_std_irq_setup(struct si_sm_io *io); void ipmi_irq_finish_setup(struct si_sm_io *io); -int ipmi_si_remove_by_dev(struct device *dev); +void ipmi_si_remove_by_dev(struct device *dev); struct device *ipmi_si_remove_by_data(int addr_space, enum si_type si_type, unsigned long addr); void ipmi_hardcode_init(void); diff --git a/drivers/char/ipmi/ipmi_si_intf.c b/drivers/char/ipmi/ipmi_si_intf.c index 62929a3e397e..bb466981dc1b 100644 --- a/drivers/char/ipmi/ipmi_si_intf.c +++ b/drivers/char/ipmi/ipmi_si_intf.c @@ -2228,22 +2228,18 @@ static void cleanup_one_si(struct smi_info *smi_info) kfree(smi_info); } -int ipmi_si_remove_by_dev(struct device *dev) +void ipmi_si_remove_by_dev(struct device *dev) { struct smi_info *e; - int rv = -ENOENT; mutex_lock(&smi_infos_lock); list_for_each_entry(e, &smi_infos, link) { if (e->io.dev == dev) { cleanup_one_si(e); - rv = 0; break; } } mutex_unlock(&smi_infos_lock); - - return rv; } struct device *ipmi_si_remove_by_data(int addr_space, enum si_type si_type, diff --git a/drivers/char/ipmi/ipmi_si_parisc.c b/drivers/char/ipmi/ipmi_si_parisc.c index 11c9160275df..2be2967f6b5f 100644 --- a/drivers/char/ipmi/ipmi_si_parisc.c +++ b/drivers/char/ipmi/ipmi_si_parisc.c @@ -29,9 +29,9 @@ static int __init ipmi_parisc_probe(struct parisc_device *dev) return ipmi_si_add_smi(&io); } -static int __exit ipmi_parisc_remove(struct parisc_device *dev) +static void __exit ipmi_parisc_remove(struct parisc_device *dev) { - return ipmi_si_remove_by_dev(&dev->dev); + ipmi_si_remove_by_dev(&dev->dev); } static const struct parisc_device_id ipmi_parisc_tbl[] __initconst = { diff --git a/drivers/char/ipmi/ipmi_si_platform.c b/drivers/char/ipmi/ipmi_si_platform.c index 380a6a542890..505cc978c97a 100644 --- a/drivers/char/ipmi/ipmi_si_platform.c +++ b/drivers/char/ipmi/ipmi_si_platform.c @@ -411,7 +411,9 @@ static int ipmi_probe(struct platform_device *pdev) static int ipmi_remove(struct platform_device *pdev) { - return ipmi_si_remove_by_dev(&pdev->dev); + ipmi_si_remove_by_dev(&pdev->dev); + + return 0; } static int pdev_match_name(struct device *dev, const void *data) diff --git a/drivers/input/keyboard/hilkbd.c b/drivers/input/keyboard/hilkbd.c index 62ccfebf2f60..c1a4d5055de6 100644 --- a/drivers/input/keyboard/hilkbd.c +++ b/drivers/input/keyboard/hilkbd.c @@ -316,11 +316,9 @@ static int __init hil_probe_chip(struct parisc_device *dev) return hil_keyb_init(); } -static int __exit hil_remove_chip(struct parisc_device *dev) +static void __exit hil_remove_chip(struct parisc_device *dev) { hil_keyb_exit(); - - return 0; } static const struct parisc_device_id hil_tbl[] __initconst = { diff --git a/drivers/input/serio/gscps2.c b/drivers/input/serio/gscps2.c index 2f9775de3c5b..a9065c6ab550 100644 --- a/drivers/input/serio/gscps2.c +++ b/drivers/input/serio/gscps2.c @@ -411,7 +411,7 @@ static int __init gscps2_probe(struct parisc_device *dev) * @return: success/error report */ -static int __exit gscps2_remove(struct parisc_device *dev) +static void __exit gscps2_remove(struct parisc_device *dev) { struct gscps2port *ps2port = dev_get_drvdata(&dev->dev); @@ -425,7 +425,6 @@ static int __exit gscps2_remove(struct parisc_device *dev) #endif dev_set_drvdata(&dev->dev, NULL); kfree(ps2port); - return 0; } diff --git a/drivers/net/ethernet/i825xx/lasi_82596.c b/drivers/net/ethernet/i825xx/lasi_82596.c index 96c6f4f36904..48e001881c75 100644 --- a/drivers/net/ethernet/i825xx/lasi_82596.c +++ b/drivers/net/ethernet/i825xx/lasi_82596.c @@ -196,7 +196,7 @@ lan_init_chip(struct parisc_device *dev) return retval; } -static int __exit lan_remove_chip(struct parisc_device *pdev) +static void __exit lan_remove_chip(struct parisc_device *pdev) { struct net_device *dev = parisc_get_drvdata(pdev); struct i596_private *lp = netdev_priv(dev); @@ -205,7 +205,6 @@ static int __exit lan_remove_chip(struct parisc_device *pdev) dma_free_noncoherent(&pdev->dev, sizeof(struct i596_private), lp->dma, lp->dma_addr, DMA_BIDIRECTIONAL); free_netdev (dev); - return 0; } static const struct parisc_device_id lan_tbl[] __initconst = { diff --git a/drivers/parport/parport_gsc.c b/drivers/parport/parport_gsc.c index 1e43b3f399a8..4332692ca4b8 100644 --- a/drivers/parport/parport_gsc.c +++ b/drivers/parport/parport_gsc.c @@ -378,7 +378,7 @@ static int __init parport_init_chip(struct parisc_device *dev) return 0; } -static int __exit parport_remove_chip(struct parisc_device *dev) +static void __exit parport_remove_chip(struct parisc_device *dev) { struct parport *p = dev_get_drvdata(&dev->dev); if (p) { @@ -397,7 +397,6 @@ static int __exit parport_remove_chip(struct parisc_device *dev) parport_put_port(p); kfree (ops); /* hope no-one cached it */ } - return 0; } static const struct parisc_device_id parport_tbl[] __initconst = { diff --git a/drivers/scsi/lasi700.c b/drivers/scsi/lasi700.c index 6d14a7a94d0b..86fe19e0468d 100644 --- a/drivers/scsi/lasi700.c +++ b/drivers/scsi/lasi700.c @@ -134,7 +134,7 @@ lasi700_probe(struct parisc_device *dev) return -ENODEV; } -static int __exit +static void __exit lasi700_driver_remove(struct parisc_device *dev) { struct Scsi_Host *host = dev_get_drvdata(&dev->dev); @@ -146,8 +146,6 @@ lasi700_driver_remove(struct parisc_device *dev) free_irq(host->irq, host); iounmap(hostdata->base); kfree(hostdata); - - return 0; } static struct parisc_driver lasi700_driver __refdata = { diff --git a/drivers/scsi/zalon.c b/drivers/scsi/zalon.c index 7eac76cccc4c..f1e5cf8a17d9 100644 --- a/drivers/scsi/zalon.c +++ b/drivers/scsi/zalon.c @@ -168,15 +168,13 @@ static const struct parisc_device_id zalon_tbl[] __initconst = { MODULE_DEVICE_TABLE(parisc, zalon_tbl); -static int __exit zalon_remove(struct parisc_device *dev) +static void __exit zalon_remove(struct parisc_device *dev) { struct Scsi_Host *host = dev_get_drvdata(&dev->dev); scsi_remove_host(host); ncr53c8xx_release(host); free_irq(dev->irq, host); - - return 0; } static struct parisc_driver zalon_driver __refdata = { diff --git a/drivers/tty/serial/mux.c b/drivers/tty/serial/mux.c index be640d9863cd..643dfbcc43f9 100644 --- a/drivers/tty/serial/mux.c +++ b/drivers/tty/serial/mux.c @@ -496,7 +496,7 @@ static int __init mux_probe(struct parisc_device *dev) return 0; } -static int __exit mux_remove(struct parisc_device *dev) +static void __exit mux_remove(struct parisc_device *dev) { int i, j; int port_count = (long)dev_get_drvdata(&dev->dev); @@ -518,7 +518,6 @@ static int __exit mux_remove(struct parisc_device *dev) } release_mem_region(dev->hpa.start + MUX_OFFSET, port_count * MUX_LINE_OFFSET); - return 0; } /* Hack. This idea was taken from the 8250_gsc.c on how to properly order diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 1440db8b4177..2e3e5aa47682 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -968,11 +968,10 @@ snd_harmony_probe(struct parisc_device *padev) return err; } -static int __exit +static void __exit snd_harmony_remove(struct parisc_device *padev) { snd_card_free(parisc_get_drvdata(padev)); - return 0; } static struct parisc_driver snd_harmony_driver __refdata = { base-commit: c500bee1c5b2f1d59b1081ac879d73268ab0ff17 -- 2.30.2

3 2

undefined reference to `_snd_module_control_empty'
by GitHub issues - edited 08 Aug '21

08 Aug '21

alsa-project/alsa-lib issue #168 was edited from madebr: When linking `aserver` of libalsa 1.2.5.1, the following linking error appears: ``` /usr/bin/ld: ../src/.libs/libasound.a(control_symbols.o):(.data.rel+0x8): undefined reference to `_snd_module_control_empty' collect2: error: ld returned 1 exit status ``` libalsa is built as a static library. Full log: [libalsa.log](https://github.com/alsa-project/alsa-lib/files/6951192/libalsa… (this is using the conan build recipe at https://github.com/conan-io/conan-center-index/tree/master/recipes/libalsa/…) Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/168 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

undefined reference to `_snd_module_control_empty'
by GitHub issues - opened 08 Aug '21

08 Aug '21

alsa-project/alsa-lib issue #168 was opened from madebr: When linking `aserver` of libllsa 1.2.5.1, the following linking error appears: ``` /usr/bin/ld: ../src/.libs/libasound.a(control_symbols.o):(.data.rel+0x8): undefined reference to `_snd_module_control_empty' collect2: error: ld returned 1 exit status ``` libalsa is built as a static library. Full log: [libalsa.log](https://github.com/alsa-project/alsa-lib/files/6951192/libalsa… (this is using the conan build recipe at https://github.com/conan-io/conan-center-index/tree/master/recipes/libalsa/…) Thanks Issue URL : https://github.com/alsa-project/alsa-lib/issues/168 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: memalloc: Fix mmap of SG-buffer with WC pages
by Takashi Iwai 08 Aug '21

08 Aug '21

The code refactoring to move the WC page handling into the common memalloc layer caused a breakage for HD-audio HDMI. I overlooked that the driver is using the SG-buffer, which isn't covered by the patch. This patch adds the mmap workaround for WC pages to SG-buffer handler. A caveat is that it falls back to the default handler by returning an error after setting the pgprot, so it won't work in all cases but merely for PCM (which is currently the only use case). Fixes: 623c10108338 ("ALSA: memalloc: Fix pgprot for WC mmap on x86") Reported-and-tested-by: Andy Lavr <andy.lavr(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/sgbuf.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 9666a95f5a5c..8352a5cdb19f 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -183,10 +183,19 @@ static unsigned int snd_dma_sg_get_chunk_size(struct snd_dma_buffer *dmab, return size; } +static int snd_dma_sg_mmap(struct snd_dma_buffer *dmab, + struct vm_area_struct *area) +{ + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); + return -ENOENT; /* continue with the default mmap handler */ +} + const struct snd_malloc_ops snd_dma_sg_ops = { .alloc = snd_dma_sg_alloc, .free = snd_dma_sg_free, .get_addr = snd_dma_sg_get_addr, .get_page = snd_dma_sg_get_page, .get_chunk_size = snd_dma_sg_get_chunk_size, + .mmap = snd_dma_sg_mmap, }; -- 2.26.2

1 0

usb-audio: New mixer control quirk for Digidesign Mbox
by Damien Zammit 08 Aug '21

08 Aug '21

Hi all, I just figured out how to toggle the input source between analog/spdif on the digidesign mbox. The problem is, when I toggle the spdif source, the hardware auto-updates to spdif clock sync as well, so I need to update the other mixer control to reflect the new hardware state or force an update to revert the clock source, but I need to know the current state of the other mixer control. How do I access a second mixer control from another one to synchronise the mixer states? Also, how would I access the current sample rate from inside a mixer control? Damien

1 0

[PATCH] ALSA: cs46xx: Fix possible mmap breakage
by Takashi Iwai 07 Aug '21

07 Aug '21

CS46xx driver switches the buffer depending on the number of periods, and in some cases it switches to the own buffer without updating the buffer type properly. This may cause a problem due to the recent introduced more strict mmap buffer check. This patch addresses the potential breakage by replacing the buffer setup with the proper macro. It also simplifies the source code, too. Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/cs46xx/cs46xx_lib.c | 30 ++++++++---------------------- 1 file changed, 8 insertions(+), 22 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1e1eb17f8e07..d43927dcd61e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -1121,9 +1121,7 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, if (params_periods(hw_params) == CS46XX_FRAGS) { if (runtime->dma_area != cpcm->hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = cpcm->hw_buf.area; - runtime->dma_addr = cpcm->hw_buf.addr; - runtime->dma_bytes = cpcm->hw_buf.bytes; + snd_pcm_set_runtime_buffer(substream, &cpcm->hw_buf); #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -1143,11 +1141,8 @@ static int snd_cs46xx_playback_hw_params(struct snd_pcm_substream *substream, #endif } else { - if (runtime->dma_area == cpcm->hw_buf.area) { - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; - } + if (runtime->dma_area == cpcm->hw_buf.area) + snd_pcm_set_runtime_buffer(substream, NULL); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) { #ifdef CONFIG_SND_CS46XX_NEW_DSP @@ -1196,9 +1191,7 @@ static int snd_cs46xx_playback_hw_free(struct snd_pcm_substream *substream) if (runtime->dma_area != cpcm->hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; + snd_pcm_set_runtime_buffer(substream, NULL); return 0; } @@ -1287,16 +1280,11 @@ static int snd_cs46xx_capture_hw_params(struct snd_pcm_substream *substream, if (runtime->periods == CS46XX_FRAGS) { if (runtime->dma_area != chip->capt.hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = chip->capt.hw_buf.area; - runtime->dma_addr = chip->capt.hw_buf.addr; - runtime->dma_bytes = chip->capt.hw_buf.bytes; + snd_pcm_set_runtime_buffer(substream, &chip->capt.hw_buf); substream->ops = &snd_cs46xx_capture_ops; } else { - if (runtime->dma_area == chip->capt.hw_buf.area) { - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; - } + if (runtime->dma_area == chip->capt.hw_buf.area) + snd_pcm_set_runtime_buffer(substream, NULL); err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); if (err < 0) return err; @@ -1313,9 +1301,7 @@ static int snd_cs46xx_capture_hw_free(struct snd_pcm_substream *substream) if (runtime->dma_area != chip->capt.hw_buf.area) snd_pcm_lib_free_pages(substream); - runtime->dma_area = NULL; - runtime->dma_addr = 0; - runtime->dma_bytes = 0; + snd_pcm_set_runtime_buffer(substream, NULL); return 0; } -- 2.26.2

1 0

[PATCH] parisc: Make struct parisc_driver::remove() return void
by Uwe Kleine-König 07 Aug '21

07 Aug '21

The caller of this function (parisc_driver_remove() in arch/parisc/kernel/drivers.c) ignores the return value, so better don't return any value at all to not wake wrong expectations in driver authors. The only function that could return a non-zero value before was ipmi_parisc_remove() which returns the return value of ipmi_si_remove_by_dev(). Make this function return void, too, as for all other callers the value is ignored, too. Also fold in a small checkpatch fix for: WARNING: Unnecessary space before function pointer arguments + void (*remove) (struct parisc_device *dev); Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- arch/parisc/include/asm/parisc-device.h | 4 ++-- drivers/char/ipmi/ipmi_si_intf.c | 6 +----- drivers/char/ipmi/ipmi_si_parisc.c | 4 ++-- drivers/char/ipmi/ipmi_si_platform.c | 4 +++- drivers/input/keyboard/hilkbd.c | 4 +--- drivers/input/serio/gscps2.c | 3 +-- drivers/net/ethernet/i825xx/lasi_82596.c | 3 +-- drivers/parport/parport_gsc.c | 3 +-- drivers/scsi/lasi700.c | 4 +--- drivers/scsi/zalon.c | 4 +--- drivers/tty/serial/mux.c | 3 +-- sound/parisc/harmony.c | 3 +-- 12 files changed, 16 insertions(+), 29 deletions(-) diff --git a/arch/parisc/include/asm/parisc-device.h b/arch/parisc/include/asm/parisc-device.h index d02d144c6012..4de3b391d812 100644 --- a/arch/parisc/include/asm/parisc-device.h +++ b/arch/parisc/include/asm/parisc-device.h @@ -34,8 +34,8 @@ struct parisc_driver { struct parisc_driver *next; char *name; const struct parisc_device_id *id_table; - int (*probe) (struct parisc_device *dev); /* New device discovered */ - int (*remove) (struct parisc_device *dev); + int (*probe)(struct parisc_device *dev); /* New device discovered */ + void (*remove)(struct parisc_device *dev); struct device_driver drv; }; diff --git a/drivers/char/ipmi/ipmi_si_intf.c b/drivers/char/ipmi/ipmi_si_intf.c index 62929a3e397e..bb466981dc1b 100644 --- a/drivers/char/ipmi/ipmi_si_intf.c +++ b/drivers/char/ipmi/ipmi_si_intf.c @@ -2228,22 +2228,18 @@ static void cleanup_one_si(struct smi_info *smi_info) kfree(smi_info); } -int ipmi_si_remove_by_dev(struct device *dev) +void ipmi_si_remove_by_dev(struct device *dev) { struct smi_info *e; - int rv = -ENOENT; mutex_lock(&smi_infos_lock); list_for_each_entry(e, &smi_infos, link) { if (e->io.dev == dev) { cleanup_one_si(e); - rv = 0; break; } } mutex_unlock(&smi_infos_lock); - - return rv; } struct device *ipmi_si_remove_by_data(int addr_space, enum si_type si_type, diff --git a/drivers/char/ipmi/ipmi_si_parisc.c b/drivers/char/ipmi/ipmi_si_parisc.c index 11c9160275df..2be2967f6b5f 100644 --- a/drivers/char/ipmi/ipmi_si_parisc.c +++ b/drivers/char/ipmi/ipmi_si_parisc.c @@ -29,9 +29,9 @@ static int __init ipmi_parisc_probe(struct parisc_device *dev) return ipmi_si_add_smi(&io); } -static int __exit ipmi_parisc_remove(struct parisc_device *dev) +static void __exit ipmi_parisc_remove(struct parisc_device *dev) { - return ipmi_si_remove_by_dev(&dev->dev); + ipmi_si_remove_by_dev(&dev->dev); } static const struct parisc_device_id ipmi_parisc_tbl[] __initconst = { diff --git a/drivers/char/ipmi/ipmi_si_platform.c b/drivers/char/ipmi/ipmi_si_platform.c index 380a6a542890..505cc978c97a 100644 --- a/drivers/char/ipmi/ipmi_si_platform.c +++ b/drivers/char/ipmi/ipmi_si_platform.c @@ -411,7 +411,9 @@ static int ipmi_probe(struct platform_device *pdev) static int ipmi_remove(struct platform_device *pdev) { - return ipmi_si_remove_by_dev(&pdev->dev); + ipmi_si_remove_by_dev(&pdev->dev); + + return 0; } static int pdev_match_name(struct device *dev, const void *data) diff --git a/drivers/input/keyboard/hilkbd.c b/drivers/input/keyboard/hilkbd.c index 62ccfebf2f60..c1a4d5055de6 100644 --- a/drivers/input/keyboard/hilkbd.c +++ b/drivers/input/keyboard/hilkbd.c @@ -316,11 +316,9 @@ static int __init hil_probe_chip(struct parisc_device *dev) return hil_keyb_init(); } -static int __exit hil_remove_chip(struct parisc_device *dev) +static void __exit hil_remove_chip(struct parisc_device *dev) { hil_keyb_exit(); - - return 0; } static const struct parisc_device_id hil_tbl[] __initconst = { diff --git a/drivers/input/serio/gscps2.c b/drivers/input/serio/gscps2.c index 2f9775de3c5b..a9065c6ab550 100644 --- a/drivers/input/serio/gscps2.c +++ b/drivers/input/serio/gscps2.c @@ -411,7 +411,7 @@ static int __init gscps2_probe(struct parisc_device *dev) * @return: success/error report */ -static int __exit gscps2_remove(struct parisc_device *dev) +static void __exit gscps2_remove(struct parisc_device *dev) { struct gscps2port *ps2port = dev_get_drvdata(&dev->dev); @@ -425,7 +425,6 @@ static int __exit gscps2_remove(struct parisc_device *dev) #endif dev_set_drvdata(&dev->dev, NULL); kfree(ps2port); - return 0; } diff --git a/drivers/net/ethernet/i825xx/lasi_82596.c b/drivers/net/ethernet/i825xx/lasi_82596.c index 96c6f4f36904..48e001881c75 100644 --- a/drivers/net/ethernet/i825xx/lasi_82596.c +++ b/drivers/net/ethernet/i825xx/lasi_82596.c @@ -196,7 +196,7 @@ lan_init_chip(struct parisc_device *dev) return retval; } -static int __exit lan_remove_chip(struct parisc_device *pdev) +static void __exit lan_remove_chip(struct parisc_device *pdev) { struct net_device *dev = parisc_get_drvdata(pdev); struct i596_private *lp = netdev_priv(dev); @@ -205,7 +205,6 @@ static int __exit lan_remove_chip(struct parisc_device *pdev) dma_free_noncoherent(&pdev->dev, sizeof(struct i596_private), lp->dma, lp->dma_addr, DMA_BIDIRECTIONAL); free_netdev (dev); - return 0; } static const struct parisc_device_id lan_tbl[] __initconst = { diff --git a/drivers/parport/parport_gsc.c b/drivers/parport/parport_gsc.c index 1e43b3f399a8..4332692ca4b8 100644 --- a/drivers/parport/parport_gsc.c +++ b/drivers/parport/parport_gsc.c @@ -378,7 +378,7 @@ static int __init parport_init_chip(struct parisc_device *dev) return 0; } -static int __exit parport_remove_chip(struct parisc_device *dev) +static void __exit parport_remove_chip(struct parisc_device *dev) { struct parport *p = dev_get_drvdata(&dev->dev); if (p) { @@ -397,7 +397,6 @@ static int __exit parport_remove_chip(struct parisc_device *dev) parport_put_port(p); kfree (ops); /* hope no-one cached it */ } - return 0; } static const struct parisc_device_id parport_tbl[] __initconst = { diff --git a/drivers/scsi/lasi700.c b/drivers/scsi/lasi700.c index 6d14a7a94d0b..86fe19e0468d 100644 --- a/drivers/scsi/lasi700.c +++ b/drivers/scsi/lasi700.c @@ -134,7 +134,7 @@ lasi700_probe(struct parisc_device *dev) return -ENODEV; } -static int __exit +static void __exit lasi700_driver_remove(struct parisc_device *dev) { struct Scsi_Host *host = dev_get_drvdata(&dev->dev); @@ -146,8 +146,6 @@ lasi700_driver_remove(struct parisc_device *dev) free_irq(host->irq, host); iounmap(hostdata->base); kfree(hostdata); - - return 0; } static struct parisc_driver lasi700_driver __refdata = { diff --git a/drivers/scsi/zalon.c b/drivers/scsi/zalon.c index 7eac76cccc4c..f1e5cf8a17d9 100644 --- a/drivers/scsi/zalon.c +++ b/drivers/scsi/zalon.c @@ -168,15 +168,13 @@ static const struct parisc_device_id zalon_tbl[] __initconst = { MODULE_DEVICE_TABLE(parisc, zalon_tbl); -static int __exit zalon_remove(struct parisc_device *dev) +static void __exit zalon_remove(struct parisc_device *dev) { struct Scsi_Host *host = dev_get_drvdata(&dev->dev); scsi_remove_host(host); ncr53c8xx_release(host); free_irq(dev->irq, host); - - return 0; } static struct parisc_driver zalon_driver __refdata = { diff --git a/drivers/tty/serial/mux.c b/drivers/tty/serial/mux.c index be640d9863cd..643dfbcc43f9 100644 --- a/drivers/tty/serial/mux.c +++ b/drivers/tty/serial/mux.c @@ -496,7 +496,7 @@ static int __init mux_probe(struct parisc_device *dev) return 0; } -static int __exit mux_remove(struct parisc_device *dev) +static void __exit mux_remove(struct parisc_device *dev) { int i, j; int port_count = (long)dev_get_drvdata(&dev->dev); @@ -518,7 +518,6 @@ static int __exit mux_remove(struct parisc_device *dev) } release_mem_region(dev->hpa.start + MUX_OFFSET, port_count * MUX_LINE_OFFSET); - return 0; } /* Hack. This idea was taken from the 8250_gsc.c on how to properly order diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c index 1440db8b4177..2e3e5aa47682 100644 --- a/sound/parisc/harmony.c +++ b/sound/parisc/harmony.c @@ -968,11 +968,10 @@ snd_harmony_probe(struct parisc_device *padev) return err; } -static int __exit +static void __exit snd_harmony_remove(struct parisc_device *padev) { snd_card_free(parisc_get_drvdata(padev)); - return 0; } static struct parisc_driver snd_harmony_driver __refdata = { -- 2.30.2

3 3

Cyclic SDMA transfers to SAI haulting on imx6
by Andriy Fudimov 07 Aug '21

07 Aug '21

Hi, Im trying to set up a cyclic dma transfer to/from the sai on an imx6, but im noticing weird behavior. First of all if my buffer length is small the DMA transfer will just hault as soon as it reaches the end of the buffer. If I make my buffer length larger it wont hault however it seems that only one of the rx or tx transfers can be on at once. Once my rx callback starts getting called my tx callback stops being called. Any tips would be appreciated. #define HARDWARE_BUFFER_BYTES 64*2 //* 2 #define HARDWARE_BUFFER_PERIOD_BYTES 32*2 static void dma_rx_complete(void *data) { rxstacnt++; printk("rxstacnt: %u\n",txstacnt); } static void dma_tx_complete(void *data) { txstacnt++; printk("txstacnt: %u\n",txstacnt); } static int imx_tx_pingdma_init(struct imx_dma *master, struct device *dev) { int err; dma_addr_t pcm_hardware_reg_base; const __be32 *addr; addr = of_get_address(dev->of_node, 0, NULL, NULL); if (!addr) { dev_err(dev, "could not get pcm hardware register address\n"); } pcm_hardware_reg_base = be32_to_cpup(addr); master->dma_tx = dma_request_slave_channel(dev, "tx"); if (!master->dma_tx) { dev_err(dev, "no tx-dma configuration found - not using dma mode\n"); tdm_dma_release(master); return -1; } master->dma_rx = dma_request_slave_channel(dev, "rx"); if (!master->dma_rx) { dev_err(dev, "no rx-dma configuration found - not using dma mode\n"); tdm_dma_release(master); return -1; } /* configure DMAs */ master->tx_ping_buffer = dma_alloc_coherent(master->dma_tx->device->dev, HARDWARE_BUFFER_BYTES, &master->tx_ping_buffer_dma, GFP_KERNEL); if (!master->tx_ping_buffer) { dev_err(dev, "could not allocate dma address\n"); tdm_dma_release(master); return -1; } memset(master->tx_ping_buffer, 0, HARDWARE_BUFFER_BYTES); master->rx_ping_buffer = dma_alloc_coherent(master->dma_rx->device->dev, HARDWARE_BUFFER_BYTES, &master->rx_ping_buffer_dma, GFP_KERNEL); if (!master->rx_ping_buffer) { dev_err(dev, "could not allocate rx dma address\n"); tdm_dma_release(master); return -1; } memset(master->rx_ping_buffer, 0, HARDWARE_BUFFER_BYTES); master->dma_tx_slave_config.direction = DMA_MEM_TO_DEV; master->dma_tx_slave_config.dst_addr = 0x202c000 + 0x20; master->dma_tx_slave_config.dst_maxburst = 1; master->dma_tx_slave_config.dst_addr_width = 4; master->dma_tx_slave_config.src_addr = master->tx_ping_buffer_dma; master->dma_tx_slave_config.src_maxburst = 1; master->dma_tx_slave_config.src_addr_width = 4; err = dmaengine_slave_config(master->dma_tx, &master->dma_tx_slave_config); if (err < 0) { printk("could not setup slave_tx_config\n"); tdm_dma_release(master); return err; } master->dma_rx_slave_config.direction = DMA_DEV_TO_MEM; master->dma_rx_slave_config.dst_addr = master->rx_ping_buffer_dma; master->dma_rx_slave_config.dst_maxburst = 1; master->dma_rx_slave_config.dst_addr_width = 4; master->dma_rx_slave_config.src_addr = 0x202c000 + 0xa0; master->dma_rx_slave_config.src_maxburst = 1; master->dma_rx_slave_config.src_addr_width = 4; err = dmaengine_slave_config(master->dma_rx, &master->dma_rx_slave_config); if (err < 0) { printk("could not setup slave_rx_config\n"); tdm_dma_release(master); return err; } master->dma_tx_desc = dmaengine_prep_dma_cyclic(master->dma_tx, master->tx_ping_buffer_dma, HARDWARE_BUFFER_BYTES, HARDWARE_BUFFER_PERIOD_BYTES, DMA_MEM_TO_DEV, DMA_CTRL_ACK | DMA_PREP_INTERRUPT); if (!master->dma_tx_desc) { printk("could not setup dma_tx_desc\n"); tdm_dma_release(master); return -1; } master->dma_rx_desc = dmaengine_prep_dma_cyclic(master->dma_rx, master->rx_ping_buffer_dma, HARDWARE_BUFFER_BYTES, HARDWARE_BUFFER_PERIOD_BYTES, DMA_DEV_TO_MEM, DMA_CTRL_ACK | DMA_PREP_INTERRUPT); if (!master->dma_rx_desc) { printk("could not setup dma_rx_desc\n"); tdm_dma_release(master); return -1; } master->dma_tx_desc->callback = dma_tx_complete; master->dma_tx_desc->callback_param = master; master->dma_rx_desc->callback = dma_rx_complete; master->dma_rx_desc->callback_param = master; printk("DMA configured by success!\n"); return 0; } static void start_dma(struct imx_dma *master) { master->dma_tx_cookie = dmaengine_submit(master->dma_tx_desc); dma_async_issue_pending(master->dma_tx); master->dma_rx_cookie = dmaengine_submit(master->dma_rx_desc); dma_async_issue_pending(master->dma_rx); printk("DMA submited! dma_tx_cookie:%d\n", master->dma_tx_cookie); printk("DMA submited! dma_rx_cookie:%d\n", master->dma_rx_cookie); txcnt = 0; }

1 0

UCM SectionModifier doesn't like 'Comment'
by GitHub issues - edited 07 Aug '21

07 Aug '21

alsa-project/alsa-lib issue #167 was edited from plbossart: Adding the following patch generates a crash with alsaucm ```` alsaucm -c sof-soundwire set _verb HiFi set _enadev Headphones alsaucm: conf.c:4355: snd_config_iterator_first: Assertion `config->type == SND_CONFIG_TYPE_COMPOUND' failed. Aborted (core dumped) ```` ````diff diff --git a/ucm2/sof-soundwire/HiFi.conf b/ucm2/sof-soundwire/HiFi.conf index e780ea4..fbadb65 100644 --- a/ucm2/sof-soundwire/HiFi.conf +++ b/ucm2/sof-soundwire/HiFi.conf @@ -29,3 +29,12 @@ If.hsdev { } <sof-soundwire/Hdmi.conf> + +SectionModifier."PlayMusic" { + # this works fine +} + +SectionModifier."PlayRadio" { + # this doesn't + Comment "Play Radio" +} ```` I am not sure if anyone ever used the SectionModifier, if it's a bug in alsa-lib or alsaucm. SectionModifiers are part of the current proposal in PipeWire according to the documentation for "stream endpoints". https://docs.pipewire.org/page_objects_design.html "The available UCM modifiers for each UCM device will be added as streams, plus one "default" stream for accessing the device with no modifiers." Issue URL : https://github.com/alsa-project/alsa-lib/issues/167 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

UCM SectionModifier doesn't like 'Comment'
by GitHub issues - opened 07 Aug '21

07 Aug '21

alsa-project/alsa-lib issue #167 was opened from plbossart: Adding the following patch generates a crash with alsaucm ```` alsaucm -c sof-soundwire set _verb HiFi set _enadev Headphones alsaucm: conf.c:4355: snd_config_iterator_first: Assertion `config->type == SND_CONFIG_TYPE_COMPOUND' failed. Aborted (core dumped) ```` ````diff diff --git a/ucm2/sof-soundwire/HiFi.conf b/ucm2/sof-soundwire/HiFi.conf index e780ea4..fbadb65 100644 --- a/ucm2/sof-soundwire/HiFi.conf +++ b/ucm2/sof-soundwire/HiFi.conf @@ -29,3 +29,12 @@ If.hsdev { } <sof-soundwire/Hdmi.conf> + +SectionModifier."PlayMusic" { + # this works fine +} + +SectionModifier."PlayRadio" { + # this doesn't + Comment "Play Radio" +} I am not sure if anyone ever used the SectionModifier, if it's a bug in alsa-lib or alsaucm. SectionModifiers are part of the current proposal in PipeWire according to the documentation for "stream endpoints". https://docs.pipewire.org/page_objects_design.html "The available UCM modifiers for each UCM device will be added as streams, plus one "default" stream for accessing the device with no modifiers." Issue URL : https://github.com/alsa-project/alsa-lib/issues/167 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/4] soundwire: intel: exit clock-stop mode before system suspend
by Bard Liao 06 Aug '21

06 Aug '21

Intel validation reported an issue where the HW_RST self-clearing bit is not cleared in hardware, which as a ripple effect creates issues with the clock stop mode. This happens is a specific sequence where the Intel manager is pm_runtime suspended with the clock-stop mode enabled. During the system suspend, we currently do nothing, which can lead to potential issues on system resume and the following pm_runtime suspend, depending on the hardware state. This patch suggests a full resume if the clock-stop mode is used. This may require extra time but will make the suspend/resume flows completely symmetric. This also removes a race condition where we could not access SHIM registers if the parent was suspended as well. Resuming the link also resumes the parent by construction. BugLink: https://github.com/thesofproject/linux/issues/2606 Pierre-Louis Bossart (4): soundwire: intel: fix potential race condition during power down soundwire: intel: skip suspend/resume/wake when link was not started soundwire: intel: exit clock stop mode on system suspend soundwire: intel: simplify pm_runtime handling in suspend/resume drivers/soundwire/intel.c | 126 ++++++++++++++++++++++++++++---------- drivers/soundwire/intel.h | 1 + 2 files changed, 94 insertions(+), 33 deletions(-) -- 2.17.1

3 13

[GIT PULL] ASoC fixes for v5.14-rc4
by Mark Brown 06 Aug '21

06 Aug '21

The following changes since commit 82d28b67f780910f816fe1cfb0f676fc38c4cbb3: ASoC: ti: j721e-evm: Check for not initialized parent_clk_id (2021-07-19 12:58:42 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.14-rc4 for you to fetch changes up to e5ada3f6787a4d6234adc6f2f3ae35c6d5b71ba0: ASoC: cs42l42: Fix mono playback (2021-08-05 20:17:16 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.14 Quite a lot of fixes here, the biggest set being for the cs42l42 driver which is reasonably old but has seen a sudden uptick in activity. There's also some fixes for correctly referencing PCM buffer addresses and the removal of some driver-local bodges that had been done for the lack of prefix handling in DAPM which were broken by the core handling that as expected. ---------------------------------------------------------------- Arnd Bergmann (1): ASoC: codecs: wcd938x: fix wcd module dependency Brent Lu (1): ASoC: Intel: sof_da7219_mx98360a: fail to initialize soundcard Derek Fang (1): ASoC: rt5682: Adjust headset volume button threshold Guennadi Liakhovetski (1): ASoC: SOF: Intel: hda-ipc: fix reply size checking Lucas Tanure (1): ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs Marek Vasut (1): ASoC: tlv320aic32x4: Fix TAS2505/TAS2521 processing block selection Mario Limonciello (1): ASoC: amd: renoir: Run hibernation callbacks Mark Brown (3): ASoC: tlv320aic31xx: Fix jack detection after suspend ASoC: component: Remove misplaced prefix handling in pin control functions ASoC: nau8824: Fix open coded prefix handling Peter Ujfalusi (1): ASoC: topology: Select SND_DYNAMIC_MINORS Pierre-Louis Bossart (2): ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and SoundWire ASoC: SOF: Intel: Kconfig: fix SoundWire dependencies Richard Fitzgerald (10): MAINTAINERS: Add sound devicetree bindings for Wolfson Micro devices ASoC: cs42l42: Correct definition of ADC Volume control ASoC: cs42l42: Don't allow SND_SOC_DAIFMT_LEFT_J ASoC: cs42l42: Fix bclk calculation for mono ASoC: cs42l42: Fix inversion of ADC Notch Switch control ASoC: cs42l42: Remove duplicate control for WNF filter frequency ASoC: cs42l42: PLL must be running when changing MCLK_SRC_SEL ASoC: cs42l42: Fix LRCLK frame start edge ASoC: cs42l42: Constrain sample rate to prevent illegal SCLK ASoC: cs42l42: Fix mono playback Takashi Iwai (5): ASoC: intel: atom: Fix reference to PCM buffer address ASoC: xilinx: Fix reference to PCM buffer address ASoC: uniphier: Fix reference to PCM buffer address ASoC: kirkwood: Fix reference to PCM buffer address ASoC: amd: Fix reference to PCM buffer address Vijendar Mukunda (1): ASoC: amd: enable stop_dma_first flag for cz_dai_7219_98357 dai link MAINTAINERS | 3 +- sound/soc/Kconfig | 1 + sound/soc/amd/acp-da7219-max98357a.c | 5 ++ sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/amd/renoir/rn-pci-acp3x.c | 2 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/Makefile | 5 +- sound/soc/codecs/cs42l42.c | 104 ++++++++++++++++++--------- sound/soc/codecs/cs42l42.h | 3 + sound/soc/codecs/nau8824.c | 42 ++--------- sound/soc/codecs/rt5682.c | 1 + sound/soc/codecs/tlv320aic31xx.c | 10 +++ sound/soc/codecs/tlv320aic32x4.c | 33 +++++++-- sound/soc/codecs/wm_adsp.c | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 3 +- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 26 ++++--- sound/soc/soc-component.c | 63 +++++++--------- sound/soc/sof/intel/Kconfig | 4 +- sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda.c | 12 ++++ sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 4 +- 25 files changed, 200 insertions(+), 137 deletions(-)

2 1

[PATCH 1/8] ASoC: cs42l42: PLL must be running when changing MCLK_SRC_SEL
by Richard Fitzgerald 06 Aug '21

06 Aug '21

Both SCLK and PLL clocks must be running to drive the glitch-free mux behind MCLK_SRC_SEL and complete the switchover. This patch moves the writing of MCLK_SRC_SEL to when the PLL is started and stopped, so that it only transitions while the PLL is running. The unconditional write MCLK_SRC_SEL=0 in cs42l42_mute_stream() is safe because if the PLL is not running MCLK_SRC_SEL is already 0. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 43fc357199f9 ("ASoC: cs42l42: Set clock source for both ways of stream") --- sound/soc/codecs/cs42l42.c | 25 ++++++++++++++++++------- sound/soc/codecs/cs42l42.h | 1 + 2 files changed, 19 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index a00dc3c65549..c96549fe6ab2 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -619,6 +619,8 @@ static int cs42l42_pll_config(struct snd_soc_component *component) for (i = 0; i < ARRAY_SIZE(pll_ratio_table); i++) { if (pll_ratio_table[i].sclk == clk) { + cs42l42->pll_config = i; + /* Configure the internal sample rate */ snd_soc_component_update_bits(component, CS42L42_MCLK_CTL, CS42L42_INTERNAL_FS_MASK, @@ -627,14 +629,9 @@ static int cs42l42_pll_config(struct snd_soc_component *component) (pll_ratio_table[i].mclk_int != 24000000)) << CS42L42_INTERNAL_FS_SHIFT); - /* Set the MCLK src (PLL or SCLK) and the divide - * ratio - */ + snd_soc_component_update_bits(component, CS42L42_MCLK_SRC_SEL, - CS42L42_MCLK_SRC_SEL_MASK | CS42L42_MCLKDIV_MASK, - (pll_ratio_table[i].mclk_src_sel - << CS42L42_MCLK_SRC_SEL_SHIFT) | (pll_ratio_table[i].mclk_div << CS42L42_MCLKDIV_SHIFT)); /* Set up the LRCLK */ @@ -892,13 +889,21 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) */ regmap_multi_reg_write(cs42l42->regmap, cs42l42_to_osc_seq, ARRAY_SIZE(cs42l42_to_osc_seq)); + + /* Must disconnect PLL before stopping it */ + snd_soc_component_update_bits(component, + CS42L42_MCLK_SRC_SEL, + CS42L42_MCLK_SRC_SEL_MASK, + 0); + usleep_range(100, 200); + snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 0); } } else { if (!cs42l42->stream_use) { /* SCLK must be running before codec unmute */ - if ((cs42l42->bclk < 11289600) && (cs42l42->sclk < 11289600)) { + if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); @@ -919,6 +924,12 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) CS42L42_PLL_LOCK_TIMEOUT_US); if (ret < 0) dev_warn(component->dev, "PLL failed to lock: %d\n", ret); + + /* PLL must be running to drive glitchless switch logic */ + snd_soc_component_update_bits(component, + CS42L42_MCLK_SRC_SEL, + CS42L42_MCLK_SRC_SEL_MASK, + CS42L42_MCLK_SRC_SEL_MASK); } /* Mark SCLK as present, turn off internal oscillator */ diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 206b3c81d3e0..b92c17be7f58 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -775,6 +775,7 @@ struct cs42l42_private { struct gpio_desc *reset_gpio; struct completion pdn_done; struct snd_soc_jack *jack; + int pll_config; int bclk; u32 sclk; u32 srate; -- 2.11.0

2 10

[PATCH] ASoC: simple-card-utils: Avoid over-allocating DLCs
by Samuel Holland 06 Aug '21

06 Aug '21

The allocation of the DAI link components (DLCs) passed the wrong pointer to sizeof. Since simple_dai_props is much larger than snd_soc_dai_link_component, there was no out of bounds access, only wasted memory. Signed-off-by: Samuel Holland <samuel(a)sholland.org> --- sound/soc/generic/simple-card-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 677f7da93b4b..10c63b73900c 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -640,8 +640,8 @@ int asoc_simple_init_priv(struct asoc_simple_priv *priv, cnf_num += li->num[i].codecs; } - dais = devm_kcalloc(dev, dai_num, sizeof(*dais), GFP_KERNEL); - dlcs = devm_kcalloc(dev, dlc_num, sizeof(*dai_props), GFP_KERNEL); + dais = devm_kcalloc(dev, dai_num, sizeof(*dais), GFP_KERNEL); + dlcs = devm_kcalloc(dev, dlc_num, sizeof(*dlcs), GFP_KERNEL); if (!dais || !dlcs) return -ENOMEM; -- 2.31.1

3 3

[PATCH] ASoC: codecs: wcd938x: add Multi Button Headset Control support
by Srinivas Kandagatla 06 Aug '21

06 Aug '21

WCD938x has Multi Button Headset Control hardware to support Headset insertion, type detection, 8 headset buttons detection, Over Current detection and Impedence measurements. This patch adds support for this using wcd-mbhc apis. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/wcd938x.c | 841 ++++++++++++++++++++++++++++++++++++- 2 files changed, 819 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6d4c65d6fadd..57437eef1125 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1563,6 +1563,7 @@ config SND_SOC_WCD938X config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" select SND_SOC_WCD938X + select SND_SOC_WCD_MBHC depends on SOUNDWIRE select REGMAP_SOUNDWIRE help diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index e5795cd964f7..90e072a78ebf 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -21,6 +21,7 @@ #include <linux/regulator/consumer.h> #include "wcd-clsh-v2.h" +#include "wcd-mbhc-v2.h" #include "wcd938x.h" #define WCD938X_MAX_MICBIAS (4) @@ -173,6 +174,11 @@ struct wcd938x_priv { struct device *rxdev; struct device_node *rxnode, *txnode; struct regmap *regmap; + struct mutex micb_lock; + /* mbhc module */ + struct wcd_mbhc *wcd_mbhc; + struct wcd_mbhc_config mbhc_cfg; + struct wcd_mbhc_intr intr_ids; struct wcd_clsh_ctrl *clsh_info; struct irq_domain *virq; struct regmap_irq_chip *wcd_regmap_irq_chip; @@ -201,24 +207,70 @@ struct wcd938x_priv { bool bcs_dis; }; -enum { - MIC_BIAS_1 = 1, - MIC_BIAS_2, - MIC_BIAS_3, - MIC_BIAS_4 -}; - -enum { - MICB_PULLUP_ENABLE, - MICB_PULLUP_DISABLE, - MICB_ENABLE, - MICB_DISABLE, -}; - static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(ear_pa_gain, 600, -1800); static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(line_gain, 600, -3000); static const SNDRV_CTL_TLVD_DECLARE_DB_MINMAX(analog_gain, 0, 3000); +struct wcd938x_mbhc_zdet_param { + u16 ldo_ctl; + u16 noff; + u16 nshift; + u16 btn5; + u16 btn6; + u16 btn7; +}; + +static struct wcd_mbhc_field wcd_mbhc_fields[WCD_MBHC_REG_FUNC_MAX] = { + WCD_MBHC_FIELD(WCD_MBHC_L_DET_EN, WCD938X_ANA_MBHC_MECH, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_GND_DET_EN, WCD938X_ANA_MBHC_MECH, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_MECH_DETECTION_TYPE, WCD938X_ANA_MBHC_MECH, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_MIC_CLAMP_CTL, WCD938X_MBHC_NEW_PLUG_DETECT_CTL, 0x30), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_DETECTION_TYPE, WCD938X_ANA_MBHC_ELECT, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_HS_L_DET_PULL_UP_CTRL, WCD938X_MBHC_NEW_INT_MECH_DET_CURRENT, 0x1F), + WCD_MBHC_FIELD(WCD_MBHC_HS_L_DET_PULL_UP_COMP_CTRL, WCD938X_ANA_MBHC_MECH, 0x04), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_PLUG_TYPE, WCD938X_ANA_MBHC_MECH, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_GND_PLUG_TYPE, WCD938X_ANA_MBHC_MECH, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_SW_HPH_LP_100K_TO_GND, WCD938X_ANA_MBHC_MECH, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_SCHMT_ISRC, WCD938X_ANA_MBHC_ELECT, 0x06), + WCD_MBHC_FIELD(WCD_MBHC_FSM_EN, WCD938X_ANA_MBHC_ELECT, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_INSREM_DBNC, WCD938X_MBHC_NEW_PLUG_DETECT_CTL, 0x0F), + WCD_MBHC_FIELD(WCD_MBHC_BTN_DBNC, WCD938X_MBHC_NEW_CTL_1, 0x03), + WCD_MBHC_FIELD(WCD_MBHC_HS_VREF, WCD938X_MBHC_NEW_CTL_2, 0x03), + WCD_MBHC_FIELD(WCD_MBHC_HS_COMP_RESULT, WCD938X_ANA_MBHC_RESULT_3, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_IN2P_CLAMP_STATE, WCD938X_ANA_MBHC_RESULT_3, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_MIC_SCHMT_RESULT, WCD938X_ANA_MBHC_RESULT_3, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_SCHMT_RESULT, WCD938X_ANA_MBHC_RESULT_3, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_SCHMT_RESULT, WCD938X_ANA_MBHC_RESULT_3, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_OCP_FSM_EN, WCD938X_HPH_OCP_CTL, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_BTN_RESULT, WCD938X_ANA_MBHC_RESULT_3, 0x07), + WCD_MBHC_FIELD(WCD_MBHC_BTN_ISRC_CTL, WCD938X_ANA_MBHC_ELECT, 0x70), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_RESULT, WCD938X_ANA_MBHC_RESULT_3, 0xFF), + WCD_MBHC_FIELD(WCD_MBHC_MICB_CTRL, WCD938X_ANA_MICB2, 0xC0), + WCD_MBHC_FIELD(WCD_MBHC_HPH_CNP_WG_TIME, WCD938X_HPH_CNP_WG_TIME, 0xFF), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_PA_EN, WCD938X_ANA_HPH, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_PA_EN, WCD938X_ANA_HPH, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_HPH_PA_EN, WCD938X_ANA_HPH, 0xC0), + WCD_MBHC_FIELD(WCD_MBHC_SWCH_LEVEL_REMOVE, WCD938X_ANA_MBHC_RESULT_3, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_ANC_DET_EN, WCD938X_MBHC_CTL_BCS, 0x02), + WCD_MBHC_FIELD(WCD_MBHC_FSM_STATUS, WCD938X_MBHC_NEW_FSM_STATUS, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_MUX_CTL, WCD938X_MBHC_NEW_CTL_2, 0x70), + WCD_MBHC_FIELD(WCD_MBHC_MOISTURE_STATUS, WCD938X_MBHC_NEW_FSM_STATUS, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_GND, WCD938X_HPH_PA_CTL2, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_GND, WCD938X_HPH_PA_CTL2, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_OCP_DET_EN, WCD938X_HPH_L_TEST, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_OCP_DET_EN, WCD938X_HPH_R_TEST, 0x01), + WCD_MBHC_FIELD(WCD_MBHC_HPHL_OCP_STATUS, WCD938X_DIGITAL_INTR_STATUS_0, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_HPHR_OCP_STATUS, WCD938X_DIGITAL_INTR_STATUS_0, 0x20), + WCD_MBHC_FIELD(WCD_MBHC_ADC_EN, WCD938X_MBHC_NEW_CTL_1, 0x08), + WCD_MBHC_FIELD(WCD_MBHC_ADC_COMPLETE, WCD938X_MBHC_NEW_FSM_STATUS, 0x40), + WCD_MBHC_FIELD(WCD_MBHC_ADC_TIMEOUT, WCD938X_MBHC_NEW_FSM_STATUS, 0x80), + WCD_MBHC_FIELD(WCD_MBHC_ADC_RESULT, WCD938X_MBHC_NEW_ADC_RESULT, 0xFF), + WCD_MBHC_FIELD(WCD_MBHC_MICB2_VOUT, WCD938X_ANA_MICB2, 0x3F), + WCD_MBHC_FIELD(WCD_MBHC_ADC_MODE, WCD938X_MBHC_NEW_CTL_1, 0x10), + WCD_MBHC_FIELD(WCD_MBHC_DETECTION_DONE, WCD938X_MBHC_NEW_CTL_1, 0x04), + WCD_MBHC_FIELD(WCD_MBHC_ELECT_ISRC_EN, WCD938X_ANA_MBHC_ZDET, 0x02), +}; + static const struct reg_default wcd938x_defaults[] = { {WCD938X_ANA_PAGE_REGISTER, 0x00}, {WCD938X_ANA_BIAS, 0x00}, @@ -1725,6 +1777,8 @@ static int wcd938x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, usleep_range(7000, 7100); snd_soc_component_write_field(component, WCD938X_ANA_HPH, WCD938X_HPHR_EN_MASK, 0); + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_PRE_HPHR_PA_OFF); set_bit(HPH_PA_DELAY, &wcd938x->status_mask); break; case SND_SOC_DAPM_POST_PMD: @@ -1740,6 +1794,8 @@ static int wcd938x_codec_enable_hphr_pa(struct snd_soc_dapm_widget *w, usleep_range(7000, 7100); clear_bit(HPH_PA_DELAY, &wcd938x->status_mask); } + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_POST_HPHR_PA_OFF); snd_soc_component_write_field(component, WCD938X_ANA_HPH, WCD938X_HPHR_REF_EN_MASK, 0); snd_soc_component_write_field(component, WCD938X_DIGITAL_PDM_WD_CTL1, @@ -1827,6 +1883,7 @@ static int wcd938x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, usleep_range(7000, 7100); snd_soc_component_write_field(component, WCD938X_ANA_HPH, WCD938X_HPHL_EN_MASK, 0); + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, WCD_EVENT_PRE_HPHL_PA_OFF); set_bit(HPH_PA_DELAY, &wcd938x->status_mask); break; case SND_SOC_DAPM_POST_PMD: @@ -1842,6 +1899,8 @@ static int wcd938x_codec_enable_hphl_pa(struct snd_soc_dapm_widget *w, usleep_range(7000, 7100); clear_bit(HPH_PA_DELAY, &wcd938x->status_mask); } + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_POST_HPHL_PA_OFF); snd_soc_component_write_field(component, WCD938X_ANA_HPH, WCD938X_HPHL_REF_EN_MASK, 0); snd_soc_component_write_field(component, WCD938X_DIGITAL_PDM_WD_CTL0, @@ -2352,7 +2411,14 @@ static int wcd938x_micbias_control(struct snd_soc_component *component, snd_soc_component_write_field(component, micb_reg, WCD938X_MICB_EN_MASK, WCD938X_MICB_ENABLE); + if (micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_POST_MICBIAS_2_ON); } + if (micb_num == MIC_BIAS_2 && is_dapm) + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_POST_DAPM_MICBIAS_2_ON); + break; case MICB_DISABLE: @@ -2366,10 +2432,19 @@ static int wcd938x_micbias_control(struct snd_soc_component *component, WCD938X_MICB_PULL_UP); else if ((wcd938x->micb_ref[micb_index] == 0) && (wcd938x->pullup_ref[micb_index] == 0)) { + if (micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_PRE_MICBIAS_2_OFF); snd_soc_component_write_field(component, micb_reg, WCD938X_MICB_EN_MASK, 0); + if (micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_POST_MICBIAS_2_OFF); } + if (is_dapm && micb_num == MIC_BIAS_2) + wcd_mbhc_event_notify(wcd938x->wcd_mbhc, + WCD_EVENT_POST_DAPM_MICBIAS_2_OFF); break; } @@ -2842,6 +2917,704 @@ static int wcd938x_set_swr_port(struct snd_kcontrol *kcontrol, } +/* MBHC related */ +static void wcd938x_mbhc_clk_setup(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_1, + WCD938X_MBHC_CTL_RCO_EN_MASK, enable); +} + +static void wcd938x_mbhc_mbhc_bias_control(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, WCD938X_ANA_MBHC_ELECT, + WCD938X_ANA_MBHC_BIAS_EN, enable); +} + +static void wcd938x_mbhc_program_btn_thr(struct snd_soc_component *component, + int *btn_low, int *btn_high, + int num_btn, bool is_micbias) +{ + int i, vth; + + if (num_btn > WCD_MBHC_DEF_BUTTONS) { + dev_err(component->dev, "%s: invalid number of buttons: %d\n", + __func__, num_btn); + return; + } + + for (i = 0; i < num_btn; i++) { + vth = ((btn_high[i] * 2) / 25) & 0x3F; + snd_soc_component_write_field(component, WCD938X_ANA_MBHC_BTN0 + i, + WCD938X_MBHC_BTN_VTH_MASK, vth); + dev_dbg(component->dev, "%s: btn_high[%d]: %d, vth: %d\n", + __func__, i, btn_high[i], vth); + } +} + +static bool wcd938x_mbhc_micb_en_status(struct snd_soc_component *component, int micb_num) +{ + u8 val; + + if (micb_num == MIC_BIAS_2) { + val = snd_soc_component_read_field(component, + WCD938X_ANA_MICB2, + WCD938X_ANA_MICB2_ENABLE_MASK); + if (val == WCD938X_MICB_ENABLE) + return true; + } + return false; +} + +static void wcd938x_mbhc_hph_l_pull_up_control(struct snd_soc_component *component, + int pull_up_cur) +{ + /* Default pull up current to 2uA */ + if (pull_up_cur > HS_PULLUP_I_OFF || pull_up_cur < HS_PULLUP_I_3P0_UA) + pull_up_cur = HS_PULLUP_I_2P0_UA; + + snd_soc_component_write_field(component, + WCD938X_MBHC_NEW_INT_MECH_DET_CURRENT, + WCD938X_HSDET_PULLUP_C_MASK, pull_up_cur); +} + +static int wcd938x_mbhc_request_micbias(struct snd_soc_component *component, + int micb_num, int req) +{ + return wcd938x_micbias_control(component, micb_num, req, false); +} + +static void wcd938x_mbhc_micb_ramp_control(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_write_field(component, WCD938X_ANA_MICB2_RAMP, + WCD938X_RAMP_SHIFT_CTRL_MASK, 0x0C); + snd_soc_component_write_field(component, WCD938X_ANA_MICB2_RAMP, + WCD938X_RAMP_EN_MASK, 1); + } else { + snd_soc_component_write_field(component, WCD938X_ANA_MICB2_RAMP, + WCD938X_RAMP_EN_MASK, 0); + snd_soc_component_write_field(component, WCD938X_ANA_MICB2_RAMP, + WCD938X_RAMP_SHIFT_CTRL_MASK, 0); + } +} + +static int wcd938x_get_micb_vout_ctl_val(u32 micb_mv) +{ + /* min micbias voltage is 1V and maximum is 2.85V */ + if (micb_mv < 1000 || micb_mv > 2850) + return -EINVAL; + + return (micb_mv - 1000) / 50; +} + +static int wcd938x_mbhc_micb_adjust_voltage(struct snd_soc_component *component, + int req_volt, int micb_num) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + int cur_vout_ctl, req_vout_ctl, micb_reg, micb_en, ret = 0; + + switch (micb_num) { + case MIC_BIAS_1: + micb_reg = WCD938X_ANA_MICB1; + break; + case MIC_BIAS_2: + micb_reg = WCD938X_ANA_MICB2; + break; + case MIC_BIAS_3: + micb_reg = WCD938X_ANA_MICB3; + break; + case MIC_BIAS_4: + micb_reg = WCD938X_ANA_MICB4; + break; + default: + return -EINVAL; + } + mutex_lock(&wcd938x->micb_lock); + /* + * If requested micbias voltage is same as current micbias + * voltage, then just return. Otherwise, adjust voltage as + * per requested value. If micbias is already enabled, then + * to avoid slow micbias ramp-up or down enable pull-up + * momentarily, change the micbias value and then re-enable + * micbias. + */ + micb_en = snd_soc_component_read_field(component, micb_reg, + WCD938X_MICB_EN_MASK); + cur_vout_ctl = snd_soc_component_read_field(component, micb_reg, + WCD938X_MICB_VOUT_MASK); + + req_vout_ctl = wcd938x_get_micb_vout_ctl_val(req_volt); + if (req_vout_ctl < 0) { + ret = -EINVAL; + goto exit; + } + + if (cur_vout_ctl == req_vout_ctl) { + ret = 0; + goto exit; + } + + if (micb_en == WCD938X_MICB_ENABLE) + snd_soc_component_write_field(component, micb_reg, + WCD938X_MICB_EN_MASK, + WCD938X_MICB_PULL_UP); + + snd_soc_component_write_field(component, micb_reg, + WCD938X_MICB_VOUT_MASK, + req_vout_ctl); + + if (micb_en == WCD938X_MICB_ENABLE) { + snd_soc_component_write_field(component, micb_reg, + WCD938X_MICB_EN_MASK, + WCD938X_MICB_ENABLE); + /* + * Add 2ms delay as per HW requirement after enabling + * micbias + */ + usleep_range(2000, 2100); + } +exit: + mutex_unlock(&wcd938x->micb_lock); + return ret; +} + +static int wcd938x_mbhc_micb_ctrl_threshold_mic(struct snd_soc_component *component, + int micb_num, bool req_en) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + int rc, micb_mv; + + if (micb_num != MIC_BIAS_2) + return -EINVAL; + /* + * If device tree micbias level is already above the minimum + * voltage needed to detect threshold microphone, then do + * not change the micbias, just return. + */ + if (wcd938x->micb2_mv >= WCD_MBHC_THR_HS_MICB_MV) + return 0; + + micb_mv = req_en ? WCD_MBHC_THR_HS_MICB_MV : wcd938x->micb2_mv; + + rc = wcd938x_mbhc_micb_adjust_voltage(component, micb_mv, MIC_BIAS_2); + + return rc; +} + +static inline void wcd938x_mbhc_get_result_params(struct wcd938x_priv *wcd938x, + s16 *d1_a, u16 noff, + int32_t *zdet) +{ + int i; + int val, val1; + s16 c1; + s32 x1, d1; + int32_t denom; + int minCode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 + }; + + regmap_update_bits(wcd938x->regmap, WCD938X_ANA_MBHC_ZDET, 0x20, 0x20); + for (i = 0; i < WCD938X_ZDET_NUM_MEASUREMENTS; i++) { + regmap_read(wcd938x->regmap, WCD938X_ANA_MBHC_RESULT_2, &val); + if (val & 0x80) + break; + } + val = val << 0x8; + regmap_read(wcd938x->regmap, WCD938X_ANA_MBHC_RESULT_1, &val1); + val |= val1; + regmap_update_bits(wcd938x->regmap, WCD938X_ANA_MBHC_ZDET, 0x20, 0x00); + x1 = WCD938X_MBHC_GET_X1(val); + c1 = WCD938X_MBHC_GET_C1(val); + /* If ramp is not complete, give additional 5ms */ + if ((c1 < 2) && x1) + usleep_range(5000, 5050); + + if (!c1 || !x1) { + pr_err("%s: Impedance detect ramp error, c1=%d, x1=0x%x\n", + __func__, c1, x1); + goto ramp_down; + } + d1 = d1_a[c1]; + denom = (x1 * d1) - (1 << (14 - noff)); + if (denom > 0) + *zdet = (WCD938X_MBHC_ZDET_CONST * 1000) / denom; + else if (x1 < minCode_param[noff]) + *zdet = WCD938X_ZDET_FLOATING_IMPEDANCE; + + pr_err("%s: d1=%d, c1=%d, x1=0x%x, z_val=%d(milliOhm)\n", + __func__, d1, c1, x1, *zdet); +ramp_down: + i = 0; + while (x1) { + regmap_read(wcd938x->regmap, + WCD938X_ANA_MBHC_RESULT_1, &val); + regmap_read(wcd938x->regmap, + WCD938X_ANA_MBHC_RESULT_2, &val1); + val = val << 0x08; + val |= val1; + x1 = WCD938X_MBHC_GET_X1(val); + i++; + if (i == WCD938X_ZDET_NUM_MEASUREMENTS) + break; + } +} + +static void wcd938x_mbhc_zdet_ramp(struct snd_soc_component *component, + struct wcd938x_mbhc_zdet_param *zdet_param, + int32_t *zl, int32_t *zr, s16 *d1_a) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + int32_t zdet = 0; + + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_ZDET_ANA_CTL, + WCD938X_ZDET_MAXV_CTL_MASK, zdet_param->ldo_ctl); + snd_soc_component_update_bits(component, WCD938X_ANA_MBHC_BTN5, + WCD938X_VTH_MASK, zdet_param->btn5); + snd_soc_component_update_bits(component, WCD938X_ANA_MBHC_BTN6, + WCD938X_VTH_MASK, zdet_param->btn6); + snd_soc_component_update_bits(component, WCD938X_ANA_MBHC_BTN7, + WCD938X_VTH_MASK, zdet_param->btn7); + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_ZDET_ANA_CTL, + WCD938X_ZDET_RANGE_CTL_MASK, zdet_param->noff); + snd_soc_component_update_bits(component, WCD938X_MBHC_NEW_ZDET_RAMP_CTL, + 0x0F, zdet_param->nshift); + + if (!zl) + goto z_right; + /* Start impedance measurement for HPH_L */ + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_ZDET, 0x80, 0x80); + dev_dbg(component->dev, "%s: ramp for HPH_L, noff = %d\n", + __func__, zdet_param->noff); + wcd938x_mbhc_get_result_params(wcd938x, d1_a, zdet_param->noff, &zdet); + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_ZDET, 0x80, 0x00); + + *zl = zdet; + +z_right: + if (!zr) + return; + /* Start impedance measurement for HPH_R */ + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_ZDET, 0x40, 0x40); + dev_dbg(component->dev, "%s: ramp for HPH_R, noff = %d\n", + __func__, zdet_param->noff); + wcd938x_mbhc_get_result_params(wcd938x, d1_a, zdet_param->noff, &zdet); + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_ZDET, 0x40, 0x00); + + *zr = zdet; +} + +static inline void wcd938x_wcd_mbhc_qfuse_cal(struct snd_soc_component *component, + int32_t *z_val, int flag_l_r) +{ + s16 q1; + int q1_cal; + + if (*z_val < (WCD938X_ZDET_VAL_400/1000)) + q1 = snd_soc_component_read(component, + WCD938X_DIGITAL_EFUSE_REG_23 + (2 * flag_l_r)); + else + q1 = snd_soc_component_read(component, + WCD938X_DIGITAL_EFUSE_REG_24 + (2 * flag_l_r)); + if (q1 & 0x80) + q1_cal = (10000 - ((q1 & 0x7F) * 25)); + else + q1_cal = (10000 + (q1 * 25)); + if (q1_cal > 0) + *z_val = ((*z_val) * 10000) / q1_cal; +} + +static void wcd938x_wcd_mbhc_calc_impedance(struct snd_soc_component *component, + uint32_t *zl, uint32_t *zr) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + s16 reg0, reg1, reg2, reg3, reg4; + int32_t z1L, z1R, z1Ls; + int zMono, z_diff1, z_diff2; + bool is_fsm_disable = false; + struct wcd938x_mbhc_zdet_param zdet_param[] = { + {4, 0, 4, 0x08, 0x14, 0x18}, /* < 32ohm */ + {2, 0, 3, 0x18, 0x7C, 0x90}, /* 32ohm < Z < 400ohm */ + {1, 4, 5, 0x18, 0x7C, 0x90}, /* 400ohm < Z < 1200ohm */ + {1, 6, 7, 0x18, 0x7C, 0x90}, /* >1200ohm */ + }; + struct wcd938x_mbhc_zdet_param *zdet_param_ptr = NULL; + s16 d1_a[][4] = { + {0, 30, 90, 30}, + {0, 30, 30, 5}, + {0, 30, 30, 5}, + {0, 30, 30, 5}, + }; + s16 *d1 = NULL; + + reg0 = snd_soc_component_read(component, WCD938X_ANA_MBHC_BTN5); + reg1 = snd_soc_component_read(component, WCD938X_ANA_MBHC_BTN6); + reg2 = snd_soc_component_read(component, WCD938X_ANA_MBHC_BTN7); + reg3 = snd_soc_component_read(component, WCD938X_MBHC_CTL_CLK); + reg4 = snd_soc_component_read(component, WCD938X_MBHC_NEW_ZDET_ANA_CTL); + + if (snd_soc_component_read(component, WCD938X_ANA_MBHC_ELECT) & 0x80) { + is_fsm_disable = true; + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_ELECT, 0x80, 0x00); + } + + /* For NO-jack, disable L_DET_EN before Z-det measurements */ + if (wcd938x->mbhc_cfg.hphl_swh) + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_MECH, 0x80, 0x00); + + /* Turn off 100k pull down on HPHL */ + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_MECH, 0x01, 0x00); + + /* Disable surge protection before impedance detection. + * This is done to give correct value for high impedance. + */ + regmap_update_bits(wcd938x->regmap, + WCD938X_HPH_SURGE_HPHLR_SURGE_EN, 0xC0, 0x00); + /* 1ms delay needed after disable surge protection */ + usleep_range(1000, 1010); + + /* First get impedance on Left */ + d1 = d1_a[1]; + zdet_param_ptr = &zdet_param[1]; + wcd938x_mbhc_zdet_ramp(component, zdet_param_ptr, &z1L, NULL, d1); + + if (!WCD938X_MBHC_IS_SECOND_RAMP_REQUIRED(z1L)) + goto left_ch_impedance; + + /* Second ramp for left ch */ + if (z1L < WCD938X_ZDET_VAL_32) { + zdet_param_ptr = &zdet_param[0]; + d1 = d1_a[0]; + } else if ((z1L > WCD938X_ZDET_VAL_400) && + (z1L <= WCD938X_ZDET_VAL_1200)) { + zdet_param_ptr = &zdet_param[2]; + d1 = d1_a[2]; + } else if (z1L > WCD938X_ZDET_VAL_1200) { + zdet_param_ptr = &zdet_param[3]; + d1 = d1_a[3]; + } + wcd938x_mbhc_zdet_ramp(component, zdet_param_ptr, &z1L, NULL, d1); + +left_ch_impedance: + if ((z1L == WCD938X_ZDET_FLOATING_IMPEDANCE) || + (z1L > WCD938X_ZDET_VAL_100K)) { + *zl = WCD938X_ZDET_FLOATING_IMPEDANCE; + zdet_param_ptr = &zdet_param[1]; + d1 = d1_a[1]; + } else { + *zl = z1L/1000; + wcd938x_wcd_mbhc_qfuse_cal(component, zl, 0); + } + dev_dbg(component->dev, "%s: impedance on HPH_L = %d(ohms)\n", + __func__, *zl); + + /* Start of right impedance ramp and calculation */ + wcd938x_mbhc_zdet_ramp(component, zdet_param_ptr, NULL, &z1R, d1); + if (WCD938X_MBHC_IS_SECOND_RAMP_REQUIRED(z1R)) { + if (((z1R > WCD938X_ZDET_VAL_1200) && + (zdet_param_ptr->noff == 0x6)) || + ((*zl) != WCD938X_ZDET_FLOATING_IMPEDANCE)) + goto right_ch_impedance; + /* Second ramp for right ch */ + if (z1R < WCD938X_ZDET_VAL_32) { + zdet_param_ptr = &zdet_param[0]; + d1 = d1_a[0]; + } else if ((z1R > WCD938X_ZDET_VAL_400) && + (z1R <= WCD938X_ZDET_VAL_1200)) { + zdet_param_ptr = &zdet_param[2]; + d1 = d1_a[2]; + } else if (z1R > WCD938X_ZDET_VAL_1200) { + zdet_param_ptr = &zdet_param[3]; + d1 = d1_a[3]; + } + wcd938x_mbhc_zdet_ramp(component, zdet_param_ptr, NULL, &z1R, d1); + } +right_ch_impedance: + if ((z1R == WCD938X_ZDET_FLOATING_IMPEDANCE) || + (z1R > WCD938X_ZDET_VAL_100K)) { + *zr = WCD938X_ZDET_FLOATING_IMPEDANCE; + } else { + *zr = z1R/1000; + wcd938x_wcd_mbhc_qfuse_cal(component, zr, 1); + } + dev_dbg(component->dev, "%s: impedance on HPH_R = %d(ohms)\n", + __func__, *zr); + + /* Mono/stereo detection */ + if ((*zl == WCD938X_ZDET_FLOATING_IMPEDANCE) && + (*zr == WCD938X_ZDET_FLOATING_IMPEDANCE)) { + dev_dbg(component->dev, + "%s: plug type is invalid or extension cable\n", + __func__); + goto zdet_complete; + } + if ((*zl == WCD938X_ZDET_FLOATING_IMPEDANCE) || + (*zr == WCD938X_ZDET_FLOATING_IMPEDANCE) || + ((*zl < WCD_MONO_HS_MIN_THR) && (*zr > WCD_MONO_HS_MIN_THR)) || + ((*zl > WCD_MONO_HS_MIN_THR) && (*zr < WCD_MONO_HS_MIN_THR))) { + dev_dbg(component->dev, + "%s: Mono plug type with one ch floating or shorted to GND\n", + __func__); + wcd_mbhc_set_hph_type(wcd938x->wcd_mbhc, WCD_MBHC_HPH_MONO); + goto zdet_complete; + } + snd_soc_component_write_field(component, WCD938X_HPH_R_ATEST, + WCD938X_HPHPA_GND_OVR_MASK, 1); + snd_soc_component_write_field(component, WCD938X_HPH_PA_CTL2, + WCD938X_HPHPA_GND_R_MASK, 1); + if (*zl < (WCD938X_ZDET_VAL_32/1000)) + wcd938x_mbhc_zdet_ramp(component, &zdet_param[0], &z1Ls, NULL, d1); + else + wcd938x_mbhc_zdet_ramp(component, &zdet_param[1], &z1Ls, NULL, d1); + snd_soc_component_write_field(component, WCD938X_HPH_PA_CTL2, + WCD938X_HPHPA_GND_R_MASK, 0); + snd_soc_component_write_field(component, WCD938X_HPH_R_ATEST, + WCD938X_HPHPA_GND_OVR_MASK, 0); + z1Ls /= 1000; + wcd938x_wcd_mbhc_qfuse_cal(component, &z1Ls, 0); + /* Parallel of left Z and 9 ohm pull down resistor */ + zMono = ((*zl) * 9) / ((*zl) + 9); + z_diff1 = (z1Ls > zMono) ? (z1Ls - zMono) : (zMono - z1Ls); + z_diff2 = ((*zl) > z1Ls) ? ((*zl) - z1Ls) : (z1Ls - (*zl)); + if ((z_diff1 * (*zl + z1Ls)) > (z_diff2 * (z1Ls + zMono))) { + dev_dbg(component->dev, "%s: stereo plug type detected\n", + __func__); + wcd_mbhc_set_hph_type(wcd938x->wcd_mbhc, WCD_MBHC_HPH_STEREO); + } else { + dev_dbg(component->dev, "%s: MONO plug type detected\n", + __func__); + wcd_mbhc_set_hph_type(wcd938x->wcd_mbhc, WCD_MBHC_HPH_MONO); + } + + /* Enable surge protection again after impedance detection */ + regmap_update_bits(wcd938x->regmap, + WCD938X_HPH_SURGE_HPHLR_SURGE_EN, 0xC0, 0xC0); +zdet_complete: + snd_soc_component_write(component, WCD938X_ANA_MBHC_BTN5, reg0); + snd_soc_component_write(component, WCD938X_ANA_MBHC_BTN6, reg1); + snd_soc_component_write(component, WCD938X_ANA_MBHC_BTN7, reg2); + /* Turn on 100k pull down on HPHL */ + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_MECH, 0x01, 0x01); + + /* For NO-jack, re-enable L_DET_EN after Z-det measurements */ + if (wcd938x->mbhc_cfg.hphl_swh) + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_MECH, 0x80, 0x80); + + snd_soc_component_write(component, WCD938X_MBHC_NEW_ZDET_ANA_CTL, reg4); + snd_soc_component_write(component, WCD938X_MBHC_CTL_CLK, reg3); + if (is_fsm_disable) + regmap_update_bits(wcd938x->regmap, + WCD938X_ANA_MBHC_ELECT, 0x80, 0x80); +} + +static void wcd938x_mbhc_gnd_det_ctrl(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_write_field(component, WCD938X_ANA_MBHC_MECH, + WCD938X_MBHC_HSG_PULLUP_COMP_EN, 1); + snd_soc_component_write_field(component, WCD938X_ANA_MBHC_MECH, + WCD938X_MBHC_GND_DET_EN_MASK, 1); + } else { + snd_soc_component_write_field(component, WCD938X_ANA_MBHC_MECH, + WCD938X_MBHC_GND_DET_EN_MASK, 0); + snd_soc_component_write_field(component, WCD938X_ANA_MBHC_MECH, + WCD938X_MBHC_HSG_PULLUP_COMP_EN, 0); + } +} + +static void wcd938x_mbhc_hph_pull_down_ctrl(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, WCD938X_HPH_PA_CTL2, + WCD938X_HPHPA_GND_R_MASK, enable); + snd_soc_component_write_field(component, WCD938X_HPH_PA_CTL2, + WCD938X_HPHPA_GND_L_MASK, enable); +} + +static void wcd938x_mbhc_moisture_config(struct snd_soc_component *component) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + + if (wcd938x->mbhc_cfg.moist_rref == R_OFF) { + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, R_OFF); + return; + } + + /* Do not enable moisture detection if jack type is NC */ + if (!wcd938x->mbhc_cfg.hphl_swh) { + dev_dbg(component->dev, "%s: disable moisture detection for NC\n", + __func__); + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, R_OFF); + return; + } + + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, wcd938x->mbhc_cfg.moist_rref); +} + +static void wcd938x_mbhc_moisture_detect_en(struct snd_soc_component *component, bool enable) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + + if (enable) + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, wcd938x->mbhc_cfg.moist_rref); + else + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, R_OFF); +} + +static bool wcd938x_mbhc_get_moisture_status(struct snd_soc_component *component) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + bool ret = false; + + if (wcd938x->mbhc_cfg.moist_rref == R_OFF) { + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, R_OFF); + goto done; + } + + /* Do not enable moisture detection if jack type is NC */ + if (!wcd938x->mbhc_cfg.hphl_swh) { + dev_dbg(component->dev, "%s: disable moisture detection for NC\n", + __func__); + snd_soc_component_write_field(component, WCD938X_MBHC_NEW_CTL_2, + WCD938X_M_RTH_CTL_MASK, R_OFF); + goto done; + } + + /* + * If moisture_en is already enabled, then skip to plug type + * detection. + */ + if (snd_soc_component_read_field(component, WCD938X_MBHC_NEW_CTL_2, WCD938X_M_RTH_CTL_MASK)) + goto done; + + wcd938x_mbhc_moisture_detect_en(component, true); + /* Read moisture comparator status */ + ret = ((snd_soc_component_read(component, WCD938X_MBHC_NEW_FSM_STATUS) + & 0x20) ? 0 : 1); + +done: + return ret; + +} + +static void wcd938x_mbhc_moisture_polling_ctrl(struct snd_soc_component *component, + bool enable) +{ + snd_soc_component_write_field(component, + WCD938X_MBHC_NEW_INT_MOISTURE_DET_POLLING_CTRL, + WCD938X_MOISTURE_EN_POLLING_MASK, enable); +} + +static const struct wcd_mbhc_cb mbhc_cb = { + .clk_setup = wcd938x_mbhc_clk_setup, + .mbhc_bias = wcd938x_mbhc_mbhc_bias_control, + .set_btn_thr = wcd938x_mbhc_program_btn_thr, + .micbias_enable_status = wcd938x_mbhc_micb_en_status, + .hph_pull_up_control_v2 = wcd938x_mbhc_hph_l_pull_up_control, + .mbhc_micbias_control = wcd938x_mbhc_request_micbias, + .mbhc_micb_ramp_control = wcd938x_mbhc_micb_ramp_control, + .mbhc_micb_ctrl_thr_mic = wcd938x_mbhc_micb_ctrl_threshold_mic, + .compute_impedance = wcd938x_wcd_mbhc_calc_impedance, + .mbhc_gnd_det_ctrl = wcd938x_mbhc_gnd_det_ctrl, + .hph_pull_down_ctrl = wcd938x_mbhc_hph_pull_down_ctrl, + .mbhc_moisture_config = wcd938x_mbhc_moisture_config, + .mbhc_get_moisture_status = wcd938x_mbhc_get_moisture_status, + .mbhc_moisture_polling_ctrl = wcd938x_mbhc_moisture_polling_ctrl, + .mbhc_moisture_detect_en = wcd938x_mbhc_moisture_detect_en, +}; + +static int wcd938x_get_hph_type(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = wcd_mbhc_get_hph_type(wcd938x->wcd_mbhc); + + return 0; +} + +static int wcd938x_hph_impedance_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + uint32_t zl, zr; + bool hphr; + struct soc_mixer_control *mc; + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + + mc = (struct soc_mixer_control *)(kcontrol->private_value); + hphr = mc->shift; + wcd_mbhc_get_impedance(wcd938x->wcd_mbhc, &zl, &zr); + dev_dbg(component->dev, "%s: zl=%u(ohms), zr=%u(ohms)\n", __func__, zl, zr); + ucontrol->value.integer.value[0] = hphr ? zr : zl; + + return 0; +} + +static const struct snd_kcontrol_new hph_type_detect_controls[] = { + SOC_SINGLE_EXT("HPH Type", 0, 0, UINT_MAX, 0, + wcd938x_get_hph_type, NULL), +}; + +static const struct snd_kcontrol_new impedance_detect_controls[] = { + SOC_SINGLE_EXT("HPHL Impedance", 0, 0, UINT_MAX, 0, + wcd938x_hph_impedance_get, NULL), + SOC_SINGLE_EXT("HPHR Impedance", 0, 1, UINT_MAX, 0, + wcd938x_hph_impedance_get, NULL), +}; + +static int wcd938x_mbhc_init(struct snd_soc_component *component) +{ + struct wcd938x_priv *wcd938x = snd_soc_component_get_drvdata(component); + struct wcd_mbhc_intr *intr_ids = &wcd938x->intr_ids; + + intr_ids->mbhc_sw_intr = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_MBHC_SW_DET); + intr_ids->mbhc_btn_press_intr = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_MBHC_BUTTON_PRESS_DET); + intr_ids->mbhc_btn_release_intr = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_MBHC_BUTTON_RELEASE_DET); + intr_ids->mbhc_hs_ins_intr = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_MBHC_ELECT_INS_REM_LEG_DET); + intr_ids->mbhc_hs_rem_intr = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_MBHC_ELECT_INS_REM_DET); + intr_ids->hph_left_ocp = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_HPHL_OCP_INT); + intr_ids->hph_right_ocp = regmap_irq_get_virq(wcd938x->irq_chip, + WCD938X_IRQ_HPHR_OCP_INT); + + wcd938x->wcd_mbhc = wcd_mbhc_init(component, &mbhc_cb, intr_ids, wcd_mbhc_fields, true); + + snd_soc_add_component_controls(component, impedance_detect_controls, + ARRAY_SIZE(impedance_detect_controls)); + snd_soc_add_component_controls(component, hph_type_detect_controls, + ARRAY_SIZE(hph_type_detect_controls)); + + return 0; +} +/* END MBHC */ + static const struct snd_kcontrol_new wcd938x_snd_controls[] = { SOC_SINGLE_EXT("HPHL_COMP Switch", WCD938X_COMP_L, 0, 1, 0, wcd938x_get_compander, wcd938x_set_compander), @@ -3222,15 +3995,6 @@ static const struct snd_soc_dapm_route wcd938x_audio_map[] = { {"EAR", NULL, "EAR PGA"}, }; -static int wcd938x_get_micb_vout_ctl_val(u32 micb_mv) -{ - /* min micbias voltage is 1V and maximum is 2.85V */ - if (micb_mv < 1000 || micb_mv > 2850) - return -EINVAL; - - return (micb_mv - 1000) / 50; -} - static int wcd938x_set_micbias_data(struct wcd938x_priv *wcd938x) { int vout_ctl_1, vout_ctl_2, vout_ctl_3, vout_ctl_4; @@ -3369,10 +4133,27 @@ static int wcd938x_soc_codec_probe(struct snd_soc_component *component) default: break; } + + ret = wcd938x_mbhc_init(component); + if (ret) + dev_err(component->dev, "mbhc initialization failed\n"); err: return ret; } +static int wcd938x_codec_set_jack(struct snd_soc_component *comp, + struct snd_soc_jack *jack, void *data) +{ + struct wcd938x_priv *wcd = dev_get_drvdata(comp->dev); + + if (!jack) + return wcd_mbhc_start(wcd->wcd_mbhc, &wcd->mbhc_cfg, jack); + + wcd_mbhc_stop(wcd->wcd_mbhc); + + return 0; +} + static const struct snd_soc_component_driver soc_codec_dev_wcd938x = { .name = "wcd938x_codec", .probe = wcd938x_soc_codec_probe, @@ -3382,6 +4163,7 @@ static const struct snd_soc_component_driver soc_codec_dev_wcd938x = { .num_dapm_widgets = ARRAY_SIZE(wcd938x_dapm_widgets), .dapm_routes = wcd938x_audio_map, .num_dapm_routes = ARRAY_SIZE(wcd938x_audio_map), + .set_jack = wcd938x_codec_set_jack, }; static void wcd938x_dt_parse_micbias_info(struct device *dev, struct wcd938x_priv *wcd) @@ -3417,6 +4199,7 @@ static void wcd938x_dt_parse_micbias_info(struct device *dev, struct wcd938x_pri static int wcd938x_populate_dt_data(struct wcd938x_priv *wcd938x, struct device *dev) { + struct wcd_mbhc_config *cfg = &wcd938x->mbhc_cfg; int ret; wcd938x->reset_gpio = of_get_named_gpio(dev->of_node, "reset-gpios", 0); @@ -3445,6 +4228,17 @@ static int wcd938x_populate_dt_data(struct wcd938x_priv *wcd938x, struct device wcd938x_dt_parse_micbias_info(dev, wcd938x); + cfg->mbhc_micbias = MIC_BIAS_2; + cfg->anc_micbias = MIC_BIAS_2; + cfg->v_hs_max = WCD_MBHC_HS_V_MAX; + cfg->num_btn = WCD938X_MBHC_MAX_BUTTONS; + cfg->micb_mv = wcd938x->micb2_mv; + cfg->linein_th = 5000; + cfg->hs_thr = 1700; + cfg->hph_thr = 50; + + wcd_dt_parse_mbhc_data(dev, cfg); + return 0; } @@ -3676,6 +4470,7 @@ static int wcd938x_probe(struct platform_device *pdev) return -ENOMEM; dev_set_drvdata(dev, wcd938x); + mutex_init(&wcd938x->micb_lock); ret = wcd938x_populate_dt_data(wcd938x, dev); if (ret) { -- 2.21.0

2 1

[PATCH 0/3] ASoC: codecs: cppcheck warnings
by Pierre-Louis Bossart 06 Aug '21

06 Aug '21

Two minor corrections for return values, and one more important one for max98090 where duplicate reads don't seem necessary. Pierre-Louis Bossart (3): ASoC: max98090: remove duplicate status reads and useless assignmment ASoC: mt6359-accdet.c: remove useless assignments ASoC: wcd938x: simplify return value sound/soc/codecs/max98090.c | 4 +--- sound/soc/codecs/mt6359-accdet.c | 8 ++++---- sound/soc/codecs/wcd938x.c | 2 +- 3 files changed, 6 insertions(+), 8 deletions(-) -- 2.25.1

2 4

[PATCH 00/12] ASoC: soc-dapm: cleanup cppcheck warning
by Kuninori Morimoto 06 Aug '21

06 Aug '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-dapm. Kuninori Morimoto (12): ASoC: soc-dapm: cleanup cppcheck warning at dapm_wcache_lookup() ASoC: soc-dapm: cleanup cppcheck warning at dapm_connect_mux() ASoC: soc-dapm: cleanup cppcheck warning at dapm_set_mixer_path_status() ASoC: soc-dapm: cleanup cppcheck warning at dapm_new_pga() ASoC: soc-dapm: cleanup cppcheck warning at dapm_new_dai_link() ASoC: soc-dapm: cleanup cppcheck warning at dapm_seq_check_event() ASoC: soc-dapm: cleanup cppcheck warning at dapm_seq_run() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_del_route() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_add_routes() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_weak_routes() ASoC: soc-dapm: cleanup cppcheck warning at snd_soc_dapm_new_controls() ASoC: soc-dapm: cleanup cppcheck warning at soc_dapm_dai_stream_event() sound/soc/soc-dapm.c | 65 ++++++++++++++++++++++---------------------- 1 file changed, 33 insertions(+), 32 deletions(-) -- 2.25.1

2 13

[PATCH v1] ASoC: rt5640: Silence warning message about missing interrupt
by Dmitry Osipenko 06 Aug '21

06 Aug '21

Interrupt is optional for the RT5640 codec. Nexus 7 doesn't use interrupt, this results in a noisy warning message that looks like a error condition. Make interrupt explicitly optional to silence the message, use modern IRQF_NO_AUTOEN flag and correct the reg[q]uest typo in the message. Signed-off-by: Dmitry Osipenko <digetx(a)gmail.com> --- sound/soc/codecs/rt5640.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 9523f4b5c800..0f5087a7644b 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2836,15 +2836,18 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, if (ret) return ret; - ret = devm_request_irq(&i2c->dev, rt5640->irq, rt5640_irq, - IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING - | IRQF_ONESHOT, "rt5640", rt5640); - if (ret == 0) { - /* Gets re-enabled by rt5640_set_jack() */ - disable_irq(rt5640->irq); + if (rt5640->irq) { + /* enabled by rt5640_set_jack() */ + ret = devm_request_irq(&i2c->dev, rt5640->irq, rt5640_irq, + IRQF_TRIGGER_RISING | IRQF_NO_AUTOEN | + IRQF_TRIGGER_FALLING | IRQF_ONESHOT, + "rt5640", rt5640); + if (ret) { + dev_err(&i2c->dev, "Failed to request IRQ %d: %d\n", + rt5640->irq, ret); + return ret; + } } else { - dev_warn(&i2c->dev, "Failed to reguest IRQ %d: %d\n", - rt5640->irq, ret); rt5640->irq = -ENXIO; } -- 2.32.0

2 1

[PATCH] ALSA: usb-audio: fix incorrect clock source setting
by chihhao.chen＠mediatek.com 06 Aug '21

06 Aug '21

From: "chihhao.chen" <chihhao.chen(a)mediatek.com> The following scenario describes an echo test for Samsung USBC Headset (AKG) with VID/PID (0x04e8/0xa051). We first start a capture stream(USB IN transfer) in 96Khz/24bit/1ch mode. In clock find source function, we get value 0x2 for clock selector and 0x1 for clock source. Kernel-4.14 behavior Since clock source is valid so clock selector was not set again. We pass through this function and start a playback stream(USB OUT transfer) in 48Khz/32bit/2ch mode. This time we get value 0x1 for clock selector and 0x1 for clock source. Finally clock id with this setting is 0x9. Kernel-5.10 behavior Clock selector was always set one more time even it is valid. When we start a playback stream, we will get 0x2 for clock selector and 0x1 for clock source. In this case clock id becomes 0xA. This is an incorrect clock source setting and results in severe noises. We see wrong data rate in USB IN transfer. (From 288 bytes/ms becomes 144 bytes/ms) It should keep in 288 bytes/ms. This earphone works fine on older kernel version load because this is a newly-added behavior. Signed-off-by: chihhao.chen <chihhao.chen(a)mediatek.com> --- sound/usb/clock.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 52de522..14456f6 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -324,6 +324,12 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, sources[ret - 1], visited, validate); if (ret > 0) { + /* + * For Samsung USBC Headset (AKG), setting clock selector again + * will result in incorrect default clock setting problems + */ + if (chip->usb_id == USB_ID(0x04e8, 0xa051)) + return ret; err = uac_clock_selector_set_val(chip, entity_id, cur); if (err < 0) return err; -- 1.7.9.5

4 15

[PATCH v3 0/3] Add RZ/G2L Sound support
by Biju Das 05 Aug '21

05 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. v2->v3: * Fixed the dependency on KCONFIG * Merged the binding patch with dma feature added * Updated dt binding example with encoded #dma-cells value. * Improved Error handling in probe function * Removed the passing legacy channel configuration parameters from dmaengine_slave_config function * started using dma_request_chan instead of deprecated dma_request_slave_channel * Removed SoC dtsi and config patches from this series. Will send it later. v1->v2: * Rebased to latest rc kernel Biju Das (3): ASoC: dt-bindings: Document RZ/G2L bindings sound: soc: sh: Add RZ/G2L SSIF-2 driver sound: sh: rz-ssi: Add SSI DMAC support .../bindings/sound/renesas,rz-ssi.yaml | 100 ++ sound/soc/sh/Kconfig | 9 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1062 +++++++++++++++++ 4 files changed, 1175 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

3 8

[PATCH] ALSA: pci/korg1212: completely remove 'set but not used' warnings
by Pierre-Louis Bossart 05 Aug '21

05 Aug '21

Commit 6cdc01ebdfb0 ("ALSA: pci/korg1212: remove 'set but not used' warnings") missed one __maybe_unused, add to enable make W=1 sound/ compilation with x86_64 allmodconfig Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/pci/korg1212/korg1212.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 49ed2bfaf11f..5c9e240ff6a9 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2094,7 +2094,7 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci) { int err, rc; unsigned int i; - unsigned iomem_size; + __maybe_unused unsigned iomem_size; __maybe_unused unsigned ioport_size; __maybe_unused unsigned iomem2_size; struct snd_korg1212 *korg1212 = card->private_data; -- 2.25.1

2 1

[PATCH] ALSA: es1688: Avoid devres management for es1688 object creation
by Takashi Iwai 05 Aug '21

05 Aug '21

The recent refactoring of es1688 object creation with the use of devres caused a problem with the non-PnP probe of GUS driver, as it tries to probe multiple times with different parameters That is, this object needs the immediate resource release and the devres doesn't fit for it. This patch reverts partially the commit for restoring the classic resource management for es1688 object. Fixes: 1bb11c1c7f6e ("ALSA: es1688: Allocate resources with device-managed APIs") Reported-by: kernel test robot <oliver.sang(a)intel.com> Link: https://lore.kernel.org/r/20210805032513.GA30485@xsang-OptiPlex-9020 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/es1688.h | 1 + sound/isa/es1688/es1688_lib.c | 29 ++++++++++++++++++++--------- 2 files changed, 21 insertions(+), 9 deletions(-) diff --git a/include/sound/es1688.h b/include/sound/es1688.h index da577bbf9cc7..099569c31fbb 100644 --- a/include/sound/es1688.h +++ b/include/sound/es1688.h @@ -18,6 +18,7 @@ struct snd_es1688 { unsigned long port; /* port of ESS chip */ + struct resource *res_port; unsigned long mpu_port; /* MPU-401 port of ESS chip */ int irq; /* IRQ number of ESS chip */ int mpu_irq; /* MPU IRQ */ diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 9cd66b236cef..8554cb2263c1 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -580,6 +580,13 @@ static int snd_es1688_free(struct snd_es1688 *chip) { if (chip->hardware != ES1688_HW_UNDEF) snd_es1688_init(chip, 0); + release_and_free_resource(chip->res_port); + if (chip->irq >= 0) + free_irq(chip->irq, (void *) chip); + if (chip->dma8 >= 0) { + disable_dma(chip->dma8); + free_dma(chip->dma8); + } return 0; } @@ -617,25 +624,26 @@ int snd_es1688_create(struct snd_card *card, chip->dma8 = -1; chip->hardware = ES1688_HW_UNDEF; - if (!devm_request_region(card->dev, port + 4, 12, "ES1688")) { + chip->res_port = request_region(port + 4, 12, "ES1688"); + if (chip->res_port == NULL) { snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); - return -EBUSY; + err = -EBUSY; + goto exit; } - err = devm_request_irq(card->dev, irq, snd_es1688_interrupt, 0, - "ES1688", (void *) chip); + err = request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip); if (err < 0) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); - return err; + goto exit; } chip->irq = irq; card->sync_irq = chip->irq; - err = snd_devm_request_dma(card->dev, dma8, "ES1688"); + err = request_dma(dma8, "ES1688"); if (err < 0) { snd_printk(KERN_ERR "es1688: can't grab DMA8 %d\n", dma8); - return err; + goto exit; } chip->dma8 = dma8; @@ -651,14 +659,17 @@ int snd_es1688_create(struct snd_card *card, err = snd_es1688_probe(chip); if (err < 0) - return err; + goto exit; err = snd_es1688_init(chip, 1); if (err < 0) - return err; + goto exit; /* Register device */ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); +exit: + if (err) + snd_es1688_free(chip); return err; } -- 2.26.2

1 0

[PATCH] ASoC: qcom: apq8016_sbc: Add SEC_MI2S support
by Vincent Knecht 04 Aug '21

04 Aug '21

This patch adds external codec support on secondary mi2s. It is used for headphones on some devices, eg. alcatel-idol347. Signed-off-by: Vincent Knecht <vincent.knecht(a)mailoo.org> --- sound/soc/qcom/apq8016_sbc.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index 08a05f0ecad7..53460272eb1e 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -30,6 +30,11 @@ struct apq8016_sbc_data { #define MIC_CTRL_QUA_WS_SLAVE_SEL_10 BIT(17) #define MIC_CTRL_TLMM_SCLK_EN BIT(1) #define SPKR_CTL_PRI_WS_SLAVE_SEL_11 (BIT(17) | BIT(16)) +#define SPKR_CTL_TLMM_MCLK_EN BIT(1) +#define SPKR_CTL_TLMM_SCLK_EN BIT(2) +#define SPKR_CTL_TLMM_DATA1_EN BIT(3) +#define SPKR_CTL_TLMM_WS_OUT_SEL BIT(6) +#define SPKR_CTL_TLMM_WS_EN_SEL BIT(18) #define DEFAULT_MCLK_RATE 9600000 static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) @@ -53,6 +58,13 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) MIC_CTRL_TLMM_SCLK_EN, pdata->mic_iomux); break; + case MI2S_SECONDARY: + /* Configure the Sec MI2S to TLMM */ + writel(readl(pdata->spkr_iomux) | SPKR_CTL_TLMM_MCLK_EN | + SPKR_CTL_TLMM_SCLK_EN | SPKR_CTL_TLMM_DATA1_EN | + SPKR_CTL_TLMM_WS_OUT_SEL | SPKR_CTL_TLMM_WS_EN_SEL, + pdata->spkr_iomux); + break; case MI2S_TERTIARY: writel(readl(pdata->mic_iomux) | MIC_CTRL_TER_WS_SLAVE_SEL | MIC_CTRL_TLMM_SCLK_EN, -- 2.31.1

2 1

[PATCH] ASoC: amd: vangogh: Drop superfluous mmap callback
by Takashi Iwai 04 Aug '21

04 Aug '21

The mmap callback of vangogh driver just calls the default mmap handler, and it's superfluous, as the PCM core would call it if not set. Let's drop the superfluous mmap callback. Fixes: 361414dc1f07 ("ASoC: amd: add vangogh i2s dma driver pm ops") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 9f8eb72ac834..f10de38976cb 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -315,13 +315,6 @@ static int acp5x_dma_new(struct snd_soc_component *component, return 0; } -static int acp5x_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp5x_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -364,7 +357,6 @@ static const struct snd_soc_component_driver acp5x_i2s_component = { .close = acp5x_dma_close, .hw_params = acp5x_dma_hw_params, .pointer = acp5x_dma_pointer, - .mmap = acp5x_dma_mmap, .pcm_construct = acp5x_dma_new, }; -- 2.26.2

2 1

[PATCH 00/15] ALSA: Improved WC memory handling
by Takashi Iwai 04 Aug '21

04 Aug '21

Hi, this is a patchset to enhance the ALSA memory allocation helper to support the allocation of WC pages more generically as well as supporting the fixed size buffer allocation. It allows us to reduce lots of redundant code in various ASoC drivers that still haven't used the standard preallocation. Takashi === Takashi Iwai (15): ALSA: memalloc: Minor refactoring ALSA: memalloc: Correctly name as WC ALSA: pcm: Allow exact buffer preallocation ALSA: memalloc: Support WC allocation on all architectures ALSA: pxa2xx: Use managed PCM buffer allocation ASoC: bcm: Use managed PCM buffer allocation ASoC: fsl: imx-pcm-fiq: Use managed buffer allocation ASoC: fsl: imx-pcm-rpmsg: Use managed buffer allocation ASoC: tegra: Use managed buffer allocation ASoC: fsl_asrc_dma: Use managed buffer allocation ASoC: fsl_dma: Use managed buffer allocation ASoC: mpc5200: Use managed buffer allocation ASoC: qcom: lpass: Use managed buffer allocation ASoC: qcom: qdsp6: Use managed buffer allocation ASoC: sprd: Use managed buffer allocation include/sound/memalloc.h | 6 +- include/sound/pcm.h | 47 ++++++++++++-- include/sound/pxa2xx-lib.h | 13 +--- sound/arm/pxa2xx-ac97.c | 18 ++---- sound/arm/pxa2xx-pcm-lib.c | 90 ++------------------------ sound/core/memalloc.c | 94 +++++++++++++++++----------- sound/core/memalloc_local.h | 2 +- sound/core/pcm_memory.c | 67 +++++++++++++++----- sound/core/sgbuf.c | 21 ++++--- sound/pci/hda/hda_controller.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 82 ++---------------------- sound/soc/fsl/fsl_asrc_dma.c | 47 +------------- sound/soc/fsl/fsl_dma.c | 50 +-------------- sound/soc/fsl/imx-pcm-fiq.c | 75 +--------------------- sound/soc/fsl/imx-pcm-rpmsg.c | 89 +------------------------- sound/soc/fsl/mpc5200_dma.c | 63 +------------------ sound/soc/pxa/pxa-ssp.c | 3 - sound/soc/pxa/pxa2xx-ac97.c | 3 - sound/soc/pxa/pxa2xx-i2s.c | 3 - sound/soc/pxa/pxa2xx-pcm.c | 3 - sound/soc/qcom/lpass-platform.c | 60 +----------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 63 +------------------ sound/soc/sprd/sprd-pcm-dma.c | 67 +------------------- sound/soc/tegra/tegra210_admaif.c | 6 -- sound/soc/tegra/tegra_pcm.c | 93 +-------------------------- sound/soc/tegra/tegra_pcm.h | 7 --- sound/x86/intel_hdmi_audio.c | 2 +- 29 files changed, 209 insertions(+), 871 deletions(-) === Cc: Banajit Goswami <bgoswami(a)codeaurora.org> Cc: Daniel Mack <daniel(a)zonque.org> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Haojian Zhuang <haojian.zhuang(a)gmail.com> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Robert Jarzmik <robert.jarzmik(a)free.fr> Cc: Shengjiu Wang <shengjiu.wang(a)gmail.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> -- 2.26.2

3 18

[PATCH v2] ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too
by Takashi Iwai 04 Aug '21

04 Aug '21

In the recent fix commit eda80d7c9c4d ("ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS"), we replaced the pfn argument of the remap_page_pfn() call from the uninitialized dmab->addr. It was the right fix, but it'd be more generic if we actually initialize dmab->area for SNDRV_DMA_TYPE_CONTINOUS, too. e.g. the field is used in the common snd_sgbuf_get_addr(), too. This patch adds the initialization of addr field and does revert of the previous change to refer to it again in the mmap call. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Fix a wrong buffer pointer to convert sound/core/memalloc.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 084b2600a423..1cea8cb9668f 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -242,8 +242,11 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) { gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); + void *p = alloc_pages_exact(size, gfp); - return alloc_pages_exact(size, gfp); + if (p) + dmab->addr = page_to_phys(virt_to_page(p)); + return p; } static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) @@ -255,7 +258,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - page_to_pfn(virt_to_page(dmab->area)), + dmab->addr >> PAGE_SHIFT, area->vm_end - area->vm_start, area->vm_page_prot); } -- 2.26.2

1 0

[PATCH] ALSA: memalloc: Store snd_dma_buffer.addr for continuous pages, too
by Takashi Iwai 04 Aug '21

04 Aug '21

In the recent fix commit eda80d7c9c4d ("ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS"), we replaced the pfn argument of the remap_page_pfn() call from the uninitialized dmab->addr. It was the right fix, but it'd be more generic if we actually initialize dmab->area for SNDRV_DMA_TYPE_CONTINOUS, too. e.g. the field is used in the common snd_sgbuf_get_addr(), too. This patch adds the initialization of addr field and does revert of the previous change to refer to it again in the mmap call. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 084b2600a423..21f57f1e81a4 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -242,8 +242,11 @@ EXPORT_SYMBOL(snd_sgbuf_get_chunk_size); static void *snd_dma_continuous_alloc(struct snd_dma_buffer *dmab, size_t size) { gfp_t gfp = snd_mem_get_gfp_flags(dmab, GFP_KERNEL); + void *p = alloc_pages_exact(size, gfp); - return alloc_pages_exact(size, gfp); + if (p) + dmab->addr = page_to_phys(virt_to_page(dmab->area)); + return p; } static void snd_dma_continuous_free(struct snd_dma_buffer *dmab) @@ -255,7 +258,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - page_to_pfn(virt_to_page(dmab->area)), + dmab->addr >> PAGE_SHIFT, area->vm_end - area->vm_start, area->vm_page_prot); } -- 2.26.2

1 1

[PATCH] ALSA: memalloc: Fix pgprot for WC mmap on x86
by Takashi Iwai 04 Aug '21

04 Aug '21

We have a special handling of WC pages on x86, and it's currently specific to HD-audio. The last forgotten piece was the pgprot setup for the mmap with WC pages. This patch moves the pgprot setup for WC pages from HD-audio-specific mmap callback to the common helper code. It allows us to remove the superfluous mmap callback in HD-audio and its prepare_mmap redirection. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 4 ++++ sound/pci/hda/hda_controller.c | 11 ----------- sound/pci/hda/hda_controller.h | 2 -- sound/pci/hda/hda_intel.c | 12 ------------ 4 files changed, 4 insertions(+), 25 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 58bab2cfdb87..084b2600a423 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -407,6 +407,10 @@ static void snd_dma_dev_free(struct snd_dma_buffer *dmab) static int snd_dma_dev_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { +#ifdef CONFIG_X86 + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC) + area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); +#endif return dma_mmap_coherent(dmab->dev.dev, area, dmab->area, dmab->addr, dmab->bytes); } diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index ea85c714d1cf..7cd452831fd3 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -669,16 +669,6 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) return err; } -static int azx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - if (chip->ops->pcm_mmap_prepare) - chip->ops->pcm_mmap_prepare(substream, area); - return snd_pcm_lib_default_mmap(substream, area); -} - static const struct snd_pcm_ops azx_pcm_ops = { .open = azx_pcm_open, .close = azx_pcm_close, @@ -688,7 +678,6 @@ static const struct snd_pcm_ops azx_pcm_ops = { .trigger = azx_pcm_trigger, .pointer = azx_pcm_pointer, .get_time_info = azx_get_time_info, - .mmap = azx_pcm_mmap, }; static void azx_pcm_free(struct snd_pcm *pcm) diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e4b290de81de..3062f87380b1 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -74,8 +74,6 @@ struct azx; struct hda_controller_ops { /* Disable msi if supported, PCI only */ int (*disable_msi_reset_irq)(struct azx *); - void (*pcm_mmap_prepare)(struct snd_pcm_substream *substream, - struct vm_area_struct *area); /* Check if current position is acceptable */ int (*position_check)(struct azx *chip, struct azx_dev *azx_dev); /* enable/disable the link power */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9771996e2ef5..3aa432d814a2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2037,17 +2037,6 @@ static int disable_msi_reset_irq(struct azx *chip) return 0; } -static void pcm_mmap_prepare(struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ -#ifdef CONFIG_X86 - struct azx_pcm *apcm = snd_pcm_substream_chip(substream); - struct azx *chip = apcm->chip; - if (chip->uc_buffer) - area->vm_page_prot = pgprot_writecombine(area->vm_page_prot); -#endif -} - /* Denylist for skipping the whole probe: * some HD-audio PCI entries are exposed without any codecs, and such devices * should be ignored from the beginning. @@ -2061,7 +2050,6 @@ static const struct pci_device_id driver_denylist[] = { static const struct hda_controller_ops pci_hda_ops = { .disable_msi_reset_irq = disable_msi_reset_irq, - .pcm_mmap_prepare = pcm_mmap_prepare, .position_check = azx_position_check, }; -- 2.26.2

1 0

[PATCH] ASoC: mediatek: mt6359: convert to use module_platform_driver
by Nobuhiro Iwamatsu 04 Aug '21

04 Aug '21

Simplify the code by using module_platform_driver macro for mt6359-accdet. Signed-off-by: Nobuhiro Iwamatsu <nobuhiro1.iwamatsu(a)toshiba.co.jp> --- sound/soc/codecs/mt6359-accdet.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index 4222aed013f106..b2354a613b1587 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -1057,22 +1057,7 @@ static struct platform_driver mt6359_accdet_driver = { .probe = mt6359_accdet_probe, }; -static int __init mt6359_accdet_driver_init(void) -{ - int ret = 0; - - ret = platform_driver_register(&mt6359_accdet_driver); - if (ret) - return -ENODEV; - return 0; -} - -static void __exit mt6359_accdet_driver_exit(void) -{ - platform_driver_unregister(&mt6359_accdet_driver); -} -module_init(mt6359_accdet_driver_init); -module_exit(mt6359_accdet_driver_exit); +module_platform_driver(mt6359_accdet_driver) /* Module information */ MODULE_DESCRIPTION("MT6359 ALSA SoC codec jack driver"); -- 2.30.0

2 1

[PATCH 0/4] ASoC: soc-ops: cleanup cppcheck warning
by Kuninori Morimoto 04 Aug '21

04 Aug '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-ops. Kuninori Morimoto (4): ASoC: soc-ops: cleanup cppcheck warning at snd_soc_put_volsw_sx() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_limit_volume() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_get_xr_sx() ASoC: soc-ops: cleanup cppcheck warning at snd_soc_put_xr_sx() sound/soc/soc-ops.c | 21 ++++++++++----------- 1 file changed, 10 insertions(+), 11 deletions(-) -- 2.25.1

2 5

[PATCH] ASoC: codecs: ad193x: add support for 96kHz and 192kHz playback rates
by Codrin Ciubotariu 04 Aug '21

04 Aug '21

ad193x devices support 96KHz and 192KHz sampling rates, when PLL/MCLK is referenced to 48kHz. Tested on ad1934. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/ad193x.c | 30 ++++++++++++++++++++++++++++-- sound/soc/codecs/ad193x.h | 4 ++++ 2 files changed, 32 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f37ab7eda615..278a55af158b 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -316,6 +316,13 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, int word_len = 0, master_rate = 0; struct snd_soc_component *component = dai->component; struct ad193x_priv *ad193x = snd_soc_component_get_drvdata(component); + bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u8 dacc0; + + dev_dbg(dai->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + __func__, params_rate(params), params_format(params), + params_width(params), params_channels(params)); + /* bit size */ switch (params_width(params)) { @@ -346,6 +353,25 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream, break; } + if (is_playback) { + switch (params_rate(params)) { + case 48000: + dacc0 = AD193X_DAC_SR_48; + break; + case 96000: + dacc0 = AD193X_DAC_SR_96; + break; + case 192000: + dacc0 = AD193X_DAC_SR_192; + break; + default: + dev_err(dai->dev, "invalid sampling rate: %d\n", params_rate(params)); + return -EINVAL; + } + + regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL0, AD193X_DAC_SR_MASK, dacc0); + } + regmap_update_bits(ad193x->regmap, AD193X_PLL_CLK_CTRL0, AD193X_PLL_INPUT_MASK, master_rate); @@ -385,7 +411,7 @@ static struct snd_soc_dai_driver ad193x_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, @@ -407,7 +433,7 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 8, - .rates = SNDRV_PCM_RATE_48000, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, }, diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 377854712c20..61f4648861d5 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -37,6 +37,10 @@ int ad193x_probe(struct device *dev, struct regmap *regmap, #define AD193X_PLL_CLK_SRC_MCLK (1 << 1) #define AD193X_DAC_CTRL0 0x02 #define AD193X_DAC_POWERDOWN 0x01 +#define AD193X_DAC_SR_MASK 0x06 +#define AD193X_DAC_SR_48 (0 << 1) +#define AD193X_DAC_SR_96 (1 << 1) +#define AD193X_DAC_SR_192 (2 << 1) #define AD193X_DAC_SERFMT_MASK 0xC0 #define AD193X_DAC_SERFMT_STEREO (0 << 6) #define AD193X_DAC_SERFMT_TDM (1 << 6) -- 2.30.2

2 1

[PATCH 1/2] ASoC: cs42l42: Fix inversion of ADC Notch Switch control
by Richard Fitzgerald 04 Aug '21

04 Aug '21

The underlying register field has inverted sense (0 = enabled) so the control definition must be marked as inverted. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") --- sound/soc/codecs/cs42l42.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 08ca05bfbeb3..2db87e3a4dc3 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -437,7 +437,7 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf05_freq_enum, CS42L42_ADC_WNF_HPF_CTL, static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, - CS42L42_ADC_NOTCH_DIS_SHIFT, true, false), + CS42L42_ADC_NOTCH_DIS_SHIFT, true, true), SOC_SINGLE("ADC Weak Force Switch", CS42L42_ADC_CTL, CS42L42_ADC_FORCE_WEAK_VCM_SHIFT, true, false), SOC_SINGLE("ADC Invert Switch", CS42L42_ADC_CTL, -- 2.11.0

2 2

[PATCH] ASoC: SOF: Intel: hda-ipc: fix reply size checking
by Pierre-Louis Bossart 04 Aug '21

04 Aug '21

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Checking that two values don't have common bits makes no sense, strict equality is meant. Fixes: f3b433e4699f ("ASoC: SOF: Implement Probe IPC API") Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-ipc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index c91aa951df22..acfeca42604c 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -107,8 +107,8 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) } else { /* reply correct size ? */ if (reply.hdr.size != msg->reply_size && - /* getter payload is never known upfront */ - !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { + /* getter payload is never known upfront */ + ((reply.hdr.cmd & SOF_GLB_TYPE_MASK) != SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; -- 2.25.1

2 1

[PATCH] ASoC: SOF: Intel: Kconfig: fix SoundWire dependencies
by Pierre-Louis Bossart 04 Aug '21

04 Aug '21

The previous Kconfig cleanup added simplifications but also introduced a new one by moving a boolean to a tristate. This leads to randconfig problems. This patch moves the select operations in the SOUNDWIRE_LINK_BASELINE option. The INTEL_SOUNDWIRE config remains a tristate for backwards compatibility with older configurations but is essentially an on/off switch. Fixes: cf5807f5f814f ('ASoC: SOF: Intel: SoundWire: simplify Kconfig') Reported-by: Arnd Bergmann <arnd(a)arndb.de> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> --- sound/soc/sof/intel/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 4bce89b5ea40..4447f515e8b1 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -278,6 +278,8 @@ config SND_SOC_SOF_HDA config SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE tristate + select SOUNDWIRE_INTEL if SND_SOC_SOF_INTEL_SOUNDWIRE + select SND_INTEL_SOUNDWIRE_ACPI if SND_SOC_SOF_INTEL_SOUNDWIRE config SND_SOC_SOF_INTEL_SOUNDWIRE tristate "SOF support for SoundWire" @@ -285,8 +287,6 @@ config SND_SOC_SOF_INTEL_SOUNDWIRE depends on SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE depends on ACPI && SOUNDWIRE depends on !(SOUNDWIRE=m && SND_SOC_SOF_INTEL_SOUNDWIRE_LINK_BASELINE=y) - select SOUNDWIRE_INTEL - select SND_INTEL_SOUNDWIRE_ACPI help This adds support for SoundWire with Sound Open Firmware for Intel(R) platforms. -- 2.25.1

3 2

[PATCH] ASoC: rt5514: make array div static const, makes object smaller
by Colin King 04 Aug '21

04 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array div on the stack but instead it static const. Makes the object code smaller by 48 bytes. Before: text data bss dec hex filename 53894 16368 128 70390 112f6 ./sound/soc/codecs/rt5514.o After: text data bss dec hex filename 53750 16464 128 70342 112c6 ./sound/soc/codecs/rt5514.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5514.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 7081142a355e..4b1ad5054e8d 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -494,7 +494,7 @@ static const struct snd_kcontrol_new rt5514_sto2_dmic_mux = */ static int rt5514_calc_dmic_clk(struct snd_soc_component *component, int rate) { - int div[] = {2, 3, 4, 8, 12, 16, 24, 32}; + static const int div[] = {2, 3, 4, 8, 12, 16, 24, 32}; int i; if (rate < 1000000 * div[0]) { -- 2.31.1

2 1

[PATCH 0/4] ASoC: SOF/Intel: machine driver updates
by Pierre-Louis Bossart 04 Aug '21

04 Aug '21

Two quirks added for Dell XPS 9710 and Intel reference boards, and two minor cleanups. Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: add quirk for Dell XPS 9710 ASoC: Intel: update sof_pcm512x quirks ASoC: Intel: sof_sdw_max98373: remove useless inits jairaj arava (1): ASoC: SOF: Intel: Use DMI string to search for adl_mx98373_rt5682 variant sound/soc/intel/boards/sof_pcm512x.c | 13 +++++++++++-- sound/soc/intel/boards/sof_rt5682.c | 14 ++++++++++++++ sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ sound/soc/intel/boards/sof_sdw_max98373.c | 4 ++-- sound/soc/sof/sof-pci-dev.c | 9 +++++++++ 5 files changed, 48 insertions(+), 4 deletions(-) -- 2.25.1

2 5

[PATCH][V2] ALSA: usb-audio: make array static const, makes object smaller
by Colin King 03 Aug '21

03 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate array names_to_check on the stack but instead it static. Makes the object code smaller by 56 bytes. Also clean up checkpatch warning by adding extra const for names_to_check and pointer s. Before: text data bss dec hex filename 103512 34380 0 137892 21aa4 ./sound/usb/mixer.o After: text data bss dec hex filename 103264 34572 0 137836 21a6c ./sound/usb/mixer.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- V2: Clean up checkpatch warning on const-ness for names_to_check. Add extra const to clean up build warning on pointer s. --- sound/usb/mixer.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index db7cb6b6dc4e..43bc59575a6e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1572,9 +1572,9 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, struct snd_card *card) { - const char *names_to_check[] = { + static const char * const names_to_check[] = { "Headset", "headset", "Headphone", "headphone", NULL}; - const char **s; + const char * const *s; bool found = false; if (strcmp("Speaker", kctl->id.name)) -- 2.31.1

2 1

[PATCH] ASoC: Intel: sof_rt5682: Add max98390 echo reference support
by Mark Hsieh 03 Aug '21

03 Aug '21

Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_maxim_common.c | 126 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 15 +++ sound/soc/intel/boards/sof_rt5682.c | 43 +++++- .../intel/common/soc-acpi-intel-adl-match.c | 13 ++ 4 files changed, 196 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e9c52f8b6428..179f1d12e1b8 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -11,6 +11,7 @@ #include "sof_maxim_common.h" #define MAX_98373_PIN_NAME 16 +#define MAX_98390_PIN_NAME 16 const struct snd_soc_dapm_route max_98373_dapm_routes[] = { /* speaker */ @@ -133,6 +134,131 @@ void max_98373_set_codec_conf(struct snd_soc_card *card) } EXPORT_SYMBOL_NS(max_98373_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); +/* + * Maxim MAX98390 + */ + +const struct snd_soc_dapm_route max_98390_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "Left BE_OUT" }, + { "Right Spk", NULL, "Right BE_OUT" }, +}; +EXPORT_SYMBOL_NS(max_98390_dapm_routes, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static struct snd_soc_codec_conf max_98390_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { /* For Right */ + .name = MAX_98373_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { /* For Left */ + .name = MAX_98373_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +static int max_98390_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + int j; + + for_each_rtd_codec_dais(rtd, j, codec_dai) { + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x03, 3, 8, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x0C, 3, 8, 32); + } + } + return 0; +} + +int max_98390_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; + int j; + int ret = 0; + + /* set spk pin by playback only */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + return 0; + + cpu_dai = asoc_rtd_to_cpu(rtd, 0); + for_each_rtd_codec_dais(rtd, j, codec_dai) { + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(cpu_dai->component); + char pin_name[MAX_98390_PIN_NAME]; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (!ret) + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (!ret) + snd_soc_dapm_sync(dapm); + break; + default: + break; + } + } + + return ret; +} +EXPORT_SYMBOL_NS(max_98390_trigger, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, + .trigger = max_98390_trigger, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) + dev_err(rtd->dev, "Speaker map addition failed: %d\n", ret); + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card) +{ + card->codec_conf = max_98390_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_codec_conf); +} +EXPORT_SYMBOL_NS(max_98390_set_codec_conf, SND_SOC_INTEL_SOF_MAXIM_COMMON); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 2674f1e373ef..e1ad25dfb915 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,21 @@ int max_98373_spk_codec_init(struct snd_soc_pcm_runtime *rtd); void max_98373_set_codec_conf(struct snd_soc_card *card); int max_98373_trigger(struct snd_pcm_substream *substream, int cmd); +/* + * Maxim MAX98390 + */ +#define MAX_98390_CODEC_DAI "max98390-aif1" +#define MAX_98390_DEV0_NAME "i2c-MX98390:00" +#define MAX_98390_DEV1_NAME "i2c-MX98390:01" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_ops max_98390_ops; +extern const struct snd_soc_dapm_route max_98390_dapm_routes[]; + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); +void max_98390_set_codec_conf(struct snd_soc_card *card); +int max_98390_trigger(struct snd_pcm_substream *substream, int cmd); + /* * Maxim MAX98357A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..ea4b8056db62 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -49,6 +49,7 @@ #define SOF_RT1015P_SPEAKER_AMP_PRESENT BIT(16) #define SOF_MAX98373_SPEAKER_AMP_PRESENT BIT(17) #define SOF_MAX98360A_SPEAKER_AMP_PRESENT BIT(18) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(23) /* BT audio offload: reserve 3 bits for future */ #define SOF_BT_OFFLOAD_SSP_SHIFT 19 @@ -162,6 +163,19 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, {} }; @@ -445,6 +459,14 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) { + /* Disable Left and Right Spk pin after boot */ + snd_soc_dapm_disable_pin(dapm, "Left Spk"); + snd_soc_dapm_disable_pin(dapm, "Right Spk"); + err = snd_soc_dapm_sync(dapm); + if (err < 0) + return err; + } return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -781,6 +803,13 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, } else if (sof_rt5682_quirk & SOF_RT1011_SPEAKER_AMP_PRESENT) { sof_rt1011_dai_link(&links[id]); + } else if (sof_rt5682_quirk & + SOF_MAX98390_SPEAKER_AMP_PRESENT) { + links[id].codecs = max_98390_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_components); + links[id].init = max_98390_spk_codec_init; + links[id].ops = &max_98390_ops; + links[id].dpcm_capture = 1; } else { max_98357a_dai_link(&links[id]); } @@ -917,7 +946,8 @@ static int sof_audio_probe(struct platform_device *pdev) sof_rt1011_codec_conf(&sof_audio_card_rt5682); else if (sof_rt5682_quirk & SOF_RT1015P_SPEAKER_AMP_PRESENT) sof_rt1015p_codec_conf(&sof_audio_card_rt5682); - + else if (sof_rt5682_quirk & SOF_MAX98390_SPEAKER_AMP_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682); if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1043,6 +1073,17 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98390_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..2db152998e4a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98390_amp = { + .num_codecs = 1, + .codecs = {"MX98390"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98390_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98390_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98390-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.17.1

3 2

[PATCH] ASoC: Intel: sof_rt5682: Add support for max98360a speaker amp
by Malik_Hsu 03 Aug '21

03 Aug '21

Signed-off-by: Malik_Hsu <malik_hsu(a)wistron.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 13 +++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 39217223d50c..8d727dd37f12 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1043,6 +1043,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "adl_max98360a_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_NUM_HDMIDEV(4)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-adl-match.c b/sound/soc/intel/common/soc-acpi-intel-adl-match.c index a0f6a69c7038..6855c9e65236 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -280,6 +280,11 @@ static const struct snd_soc_acpi_codecs adl_max98357a_amp = { .codecs = {"MX98357A"} }; +static const struct snd_soc_acpi_codecs adl_max98360a_amp = { + .num_codecs = 1, + .codecs = {"MX98360A"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { { .id = "10EC5682", @@ -297,6 +302,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { .sof_fw_filename = "sof-adl.ri", .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", }, + { + .id = "10EC5682", + .drv_name = "adl_max98360a_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98357a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

2 1

[PATCH 1/2] Revert "ASoC: rt5682: Adjust headset volume button threshold"
by derek.fang＠realtek.com 03 Aug '21

03 Aug '21

From: Derek Fang <derek.fang(a)realtek.com> This reverts commit 6d20bf7c020f417fdef1810a22da17c126603472. Since it may risk affecting other headset Vol- button. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 7dc01ae6bb66..fcf442b8a732 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,7 +44,6 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, - {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.17.1

2 2

[PATCH] ALSA: hda/realtek: add mic quirk for Acer SF314-42
by Alexander Monakov 03 Aug '21

03 Aug '21

The Acer Swift SF314-42 laptop is using Realtek ALC255 codec. Add a quirk so microphone in a headset connected via the right-hand side jack is usable. Signed-off-by: Alexander Monakov <amonakov(a)ispras.ru> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Hui Wang <hui.wang(a)canonical.com> Cc: Kailang Yang <kailang(a)realtek.com> Cc: Jeremy Szu <jeremy.szu(a)canonical.com> Cc: Jian-Hong Pan <jhp(a)endlessos.org> Cc: Chris Chiu <chris.chiu(a)canonical.com> Cc: PeiSen Hou <pshou(a)realtek.com> Cc: alsa-devel(a)alsa-project.org --- Hello, Acer Swift SF314-42 laptop is using AMD Ryzen SoC audio with Realtek ALC255 codec. The are three ALSA cards: HDMI audio, AMD ACP card, and AMD HDA card. There is an internal stereo microphone connected to the ACP card, and a mini-jack connected to the Realtek codec. There are no LEDs. I'd like to be able to use a mic+headphones headset. Unfortunately, out of the box ALSA does not correctly detect the microphone input of the Realtek codec. The suggested patch fixes that, though I cannot be sure it's 100% correct. With the patch, the experience is unfortunately still not ideal. I have noticed a couple of issues: 1) at high enough gain, recording the microphone is picking up what is being played via the headphones; maybe it's supposed to be like that, but it surprised me; 2) there is a very noticeable "pop" when plugging the headset in/out, accompanied by pcieport 0000:00:08.1: PME: Spurious native interrupt! pcieport 0000:00:08.1: PME: Spurious native interrupt! in dmesg. I'd appreciate info and any help about this issue. Thanks. Alexander sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ab5113cccffa..87ad8469dbc5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8191,6 +8191,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x142b, "Acer Swift SF314-42", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), base-commit: 62fb9874f5da54fdb243003b386128037319b219 -- 2.31.1

3 7

[PATCH] ALSA: seq: Fix racy deletion of subscriber
by Takashi Iwai 03 Aug '21

03 Aug '21

It turned out that the current implementation of the port subscription is racy. The subscription contains two linked lists, and we have to add to or delete from both lists. Since both connection and disconnection procedures perform the same order for those two lists (i.e. src list, then dest list), when a deletion happens during a connection procedure, the src list may be deleted before the dest list addition completes, and this may lead to a use-after-free or an Oops, even though the access to both lists are protected via mutex. The simple workaround for this race is to change the access order for the disconnection, namely, dest list, then src list. This assures that the connection has been established when disconnecting, and also the concurrent deletion can be avoided. Reported-and-tested-by: folkert <folkert(a)vanheusden.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/20210801182754.GP890690@belle.intranet.vanheusden… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_ports.c | 39 ++++++++++++++++++++++++++------------ 1 file changed, 27 insertions(+), 12 deletions(-) diff --git a/sound/core/seq/seq_ports.c b/sound/core/seq/seq_ports.c index b9c2ce2b8d5a..84d78630463e 100644 --- a/sound/core/seq/seq_ports.c +++ b/sound/core/seq/seq_ports.c @@ -514,10 +514,11 @@ static int check_and_subscribe_port(struct snd_seq_client *client, return err; } -static void delete_and_unsubscribe_port(struct snd_seq_client *client, - struct snd_seq_client_port *port, - struct snd_seq_subscribers *subs, - bool is_src, bool ack) +/* called with grp->list_mutex held */ +static void __delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) { struct snd_seq_port_subs_info *grp; struct list_head *list; @@ -525,7 +526,6 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, grp = is_src ? &port->c_src : &port->c_dest; list = is_src ? &subs->src_list : &subs->dest_list; - down_write(&grp->list_mutex); write_lock_irq(&grp->list_lock); empty = list_empty(list); if (!empty) @@ -535,6 +535,18 @@ static void delete_and_unsubscribe_port(struct snd_seq_client *client, if (!empty) unsubscribe_port(client, port, grp, &subs->info, ack); +} + +static void delete_and_unsubscribe_port(struct snd_seq_client *client, + struct snd_seq_client_port *port, + struct snd_seq_subscribers *subs, + bool is_src, bool ack) +{ + struct snd_seq_port_subs_info *grp; + + grp = is_src ? &port->c_src : &port->c_dest; + down_write(&grp->list_mutex); + __delete_and_unsubscribe_port(client, port, subs, is_src, ack); up_write(&grp->list_mutex); } @@ -590,27 +602,30 @@ int snd_seq_port_disconnect(struct snd_seq_client *connector, struct snd_seq_client_port *dest_port, struct snd_seq_port_subscribe *info) { - struct snd_seq_port_subs_info *src = &src_port->c_src; + struct snd_seq_port_subs_info *dest = &dest_port->c_dest; struct snd_seq_subscribers *subs; int err = -ENOENT; - down_write(&src->list_mutex); + /* always start from deleting the dest port for avoiding concurrent + * deletions + */ + down_write(&dest->list_mutex); /* look for the connection */ - list_for_each_entry(subs, &src->list_head, src_list) { + list_for_each_entry(subs, &dest->list_head, dest_list) { if (match_subs_info(info, &subs->info)) { - atomic_dec(&subs->ref_count); /* mark as not ready */ + __delete_and_unsubscribe_port(dest_client, dest_port, + subs, false, + connector->number != dest_client->number); err = 0; break; } } - up_write(&src->list_mutex); + up_write(&dest->list_mutex); if (err < 0) return err; delete_and_unsubscribe_port(src_client, src_port, subs, true, connector->number != src_client->number); - delete_and_unsubscribe_port(dest_client, dest_port, subs, false, - connector->number != dest_client->number); kfree(subs); return 0; } -- 2.26.2

1 0

aconnect occasionally causes kernel oopses
by folkert 03 Aug '21

03 Aug '21

[zo aug 1 20:24:51 2021] BUG: unable to handle page fault for address: 00003c366982f750 [zo aug 1 20:24:51 2021] #PF: supervisor read access in kernel mode [zo aug 1 20:24:51 2021] #PF: error_code(0x0000) - not-present page [zo aug 1 20:24:51 2021] PGD 0 P4D 0 [zo aug 1 20:24:51 2021] Oops: 0000 [#1] SMP NOPTI [zo aug 1 20:24:51 2021] CPU: 2 PID: 59575 Comm: aconnect Not tainted 5.11.0-25-generic #27-Ubuntu [zo aug 1 20:24:51 2021] Hardware name: HP HP Laptop 17-by2xxx/865A, BIOS F.58 06/16/2020 [zo aug 1 20:24:51 2021] RIP: 0010:check_and_subscribe_port+0x151/0x210 [snd_seq] [zo aug 1 20:24:51 2021] Code: 49 39 c5 0f 84 39 ff ff ff 41 0f b6 3e eb 0c 48 8b 00 49 39 c5 0f 84 27 ff ff ff 48 8d 70 b0 48 8d 48 a0 45 84 c0 48 0f 45 ce <40> 3a 39 75 e0 0f b6 51 01 41 38 56 01 75 d6 0f b6 51 02 41 38 56 [zo aug 1 20:24:51 2021] RSP: 0018:ffffa57dc9e03cf8 EFLAGS: 00010246 [zo aug 1 20:24:51 2021] RAX: 00003c366982f7b0 RBX: 0000000000000000 RCX: 00003c366982f750 [zo aug 1 20:24:51 2021] RDX: 0000000000000089 RSI: 00003c366982f760 RDI: 0000000000000080 [zo aug 1 20:24:51 2021] RBP: ffffa57dc9e03d38 R08: 0000000000000000 R09: ffff8945226312c0 [zo aug 1 20:24:51 2021] R10: ffff89458b7dda00 R11: ffff89460f08d270 R12: ffff894522631200 [zo aug 1 20:24:51 2021] R13: ffff8945226312c0 R14: ffff894506dd2080 R15: ffff8945226312d8 [zo aug 1 20:24:51 2021] FS: 00007f7fd9633480(0000) GS:ffff894756500000(0000) knlGS:0000000000000000 [zo aug 1 20:24:51 2021] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [zo aug 1 20:24:51 2021] CR2: 00003c366982f750 CR3: 00000001d8318006 CR4: 00000000003706e0 [zo aug 1 20:24:51 2021] Call Trace: [zo aug 1 20:24:51 2021] snd_seq_port_connect+0x113/0x180 [snd_seq] [zo aug 1 20:24:51 2021] snd_seq_ioctl_subscribe_port+0xb9/0x180 [snd_seq] [zo aug 1 20:24:51 2021] ? snd_seq_port_get_subscription+0xbb/0xd0 [snd_seq] [zo aug 1 20:24:51 2021] ? __check_object_size.part.0+0x3a/0x150 [zo aug 1 20:24:51 2021] snd_seq_ioctl+0xe8/0x1b0 [snd_seq] [zo aug 1 20:24:51 2021] __x64_sys_ioctl+0x91/0xc0 [zo aug 1 20:24:51 2021] do_syscall_64+0x38/0x90 [zo aug 1 20:24:51 2021] entry_SYSCALL_64_after_hwframe+0x44/0xa9 [zo aug 1 20:24:51 2021] RIP: 0033:0x7f7fd98b8ecb [zo aug 1 20:24:51 2021] Code: ff ff ff 85 c0 79 8b 49 c7 c4 ff ff ff ff 5b 5d 4c 89 e0 41 5c c3 66 0f 1f 84 00 00 00 00 00 f3 0f 1e fa b8 10 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 8b 0d 6d 1f 0d 00 f7 d8 64 89 01 48 [zo aug 1 20:24:51 2021] RSP: 002b:00007ffdc3e3f038 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 [zo aug 1 20:24:51 2021] RAX: ffffffffffffffda RBX: 00007f7fd998b354 RCX: 00007f7fd98b8ecb [zo aug 1 20:24:51 2021] RDX: 00007ffdc3e3f050 RSI: 0000000040505330 RDI: 0000000000000003 [zo aug 1 20:24:51 2021] RBP: 00007ffdc3e3f120 R08: 00000000ffffffff R09: 0000000000000000 [zo aug 1 20:24:51 2021] R10: 00007f7fd993dac0 R11: 0000000000000246 R12: 00007ffdc3e3f050 [zo aug 1 20:24:51 2021] R13: 00007ffdc3e3f0de R14: 0000000000000000 R15: 00007ffdc3e3f0dc [zo aug 1 20:24:51 2021] Modules linked in: snd_hrtimer snd_usb_audio snd_usbmidi_lib snd_seq_dummy ch341 usbserial ccm xt_CHECKSUM xt_MASQUERADE xt_conntrack ipt_REJECT nf_reject_ipv4 xt_tcpudp nft_compat nft_chain_nat nf_nat nf_conntrack nf_defrag_ipv6 nf_defrag_ipv4 nft_counter nf_tables libcrc32c nfnetlink bridge stp llc rfcomm cmac algif_hash algif_skcipher af_alg bnep binfmt_misc nls_iso8859_1 snd_hda_codec_hdmi snd_sof_pci snd_sof_intel_hda_common snd_hda_codec_realtek snd_soc_hdac_hda snd_hda_codec_generic snd_sof_intel_hda snd_sof_intel_byt snd_sof_intel_ipc snd_sof snd_sof_xtensa_dsp snd_hda_ext_core snd_soc_acpi_intel_match snd_soc_acpi ledtrig_audio snd_hda_intel snd_intel_dspcfg soundwire_intel soundwire_generic_allocation soundwire_cadence snd_hda_codec snd_hda_core snd_hwdep soundwire_bus snd_soc_core snd_compress ac97_bus snd_pcm_dmaengine snd_pcm mei_hdcp rtw88_8723de rtw88_8723d snd_seq_midi x86_pkg_temp_thermal intel_powerclamp snd_seq_midi_event coretemp intel_rapl_msr rtw88_pci [zo aug 1 20:24:51 2021] kvm_intel snd_rawmidi rtw88_core kvm uvcvideo videobuf2_vmalloc rapl intel_cstate mac80211 videobuf2_memops snd_seq joydev btusb btrtl videobuf2_v4l2 snd_seq_device snd_timer btbcm input_leds btintel videobuf2_common videodev hp_wmi efi_pstore mc sparse_keymap serio_raw snd wmi_bmof intel_wmi_thunderbolt cfg80211 mei_me bluetooth soundcore ee1004 libarc4 processor_thermal_device processor_thermal_rfim processor_thermal_mbox mei ecdh_generic processor_thermal_rapl intel_rapl_common ecc intel_pch_thermal int340x_thermal_zone intel_soc_dts_iosf acpi_pad hp_wireless int3400_thermal acpi_tad acpi_thermal_rel mac_hid sch_fq_codel msr parport_pc ppdev lp parport sunrpc ip_tables x_tables autofs4 dm_crypt i915 crct10dif_pclmul crc32_pclmul i2c_algo_bit ghash_clmulni_intel aesni_intel drm_kms_helper crypto_simd syscopyarea cryptd sysfillrect glue_helper sysimgblt fb_sys_fops cec psmouse nvme rc_core intel_lpss_pci intel_lpss nvme_core i2c_i801 r8169 idma64 i2c_smbus ahci realtek [zo aug 1 20:24:51 2021] libahci drm virt_dma xhci_pci xhci_pci_renesas wmi pinctrl_cannonlake video [zo aug 1 20:24:51 2021] CR2: 00003c366982f750 [zo aug 1 20:24:51 2021] ---[ end trace 000787e3cd09f39d ]--- [zo aug 1 20:24:52 2021] RIP: 0010:check_and_subscribe_port+0x151/0x210 [snd_seq] [zo aug 1 20:24:52 2021] Code: 49 39 c5 0f 84 39 ff ff ff 41 0f b6 3e eb 0c 48 8b 00 49 39 c5 0f 84 27 ff ff ff 48 8d 70 b0 48 8d 48 a0 45 84 c0 48 0f 45 ce <40> 3a 39 75 e0 0f b6 51 01 41 38 56 01 75 d6 0f b6 51 02 41 38 56 [zo aug 1 20:24:52 2021] RSP: 0018:ffffa57dc9e03cf8 EFLAGS: 00010246 [zo aug 1 20:24:52 2021] RAX: 00003c366982f7b0 RBX: 0000000000000000 RCX: 00003c366982f750 [zo aug 1 20:24:52 2021] RDX: 0000000000000089 RSI: 00003c366982f760 RDI: 0000000000000080 [zo aug 1 20:24:52 2021] RBP: ffffa57dc9e03d38 R08: 0000000000000000 R09: ffff8945226312c0 [zo aug 1 20:24:52 2021] R10: ffff89458b7dda00 R11: ffff89460f08d270 R12: ffff894522631200 [zo aug 1 20:24:52 2021] R13: ffff8945226312c0 R14: ffff894506dd2080 R15: ffff8945226312d8 [zo aug 1 20:24:52 2021] FS: 00007f7fd9633480(0000) GS:ffff894756500000(0000) knlGS:0000000000000000 [zo aug 1 20:24:52 2021] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [zo aug 1 20:24:52 2021] CR2: 00003c366982f750 CR3: 00000001d8318006 CR4: 00000000003706e0

2 16

Re: [PATCH v2 5/8] ASoC: mediatek: mt8195: add platform driver
by Chen-Yu Tsai 03 Aug '21

03 Aug '21

are all internal Hi, On Tue, Jun 29, 2021 at 9:49 AM Trevor Wu <trevor.wu(a)mediatek.com> wrote: > > This patch adds mt8195 platform and affiliated driver. > > Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> > --- > sound/soc/mediatek/Kconfig | 9 + > sound/soc/mediatek/Makefile | 1 + > sound/soc/mediatek/mt8195/Makefile | 11 + > sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 899 +++++ > sound/soc/mediatek/mt8195/mt8195-afe-clk.h | 201 + > sound/soc/mediatek/mt8195/mt8195-afe-common.h | 200 + > sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 3264 +++++++++++++++++ > sound/soc/mediatek/mt8195/mt8195-reg.h | 2793 ++++++++++++++ > 8 files changed, 7378 insertions(+) > create mode 100644 sound/soc/mediatek/mt8195/Makefile > create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.c > create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-clk.h > create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-common.h > create mode 100644 sound/soc/mediatek/mt8195/mt8195-afe-pcm.c > create mode 100644 sound/soc/mediatek/mt8195/mt8195-reg.h > > diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig > index 74dae4332d17..3389f382be06 100644 > --- a/sound/soc/mediatek/Kconfig > +++ b/sound/soc/mediatek/Kconfig > @@ -184,3 +184,12 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682 > with the MT6359 RT1015 RT5682 audio codec. > Select Y if you have such device. > If unsure select "N". > + > +config SND_SOC_MT8195 > + tristate "ASoC support for Mediatek MT8195 chip" > + select SND_SOC_MEDIATEK > + help > + This adds ASoC platform driver support for Mediatek MT8195 chip > + that can be used with other codecs. > + Select Y if you have such device. > + If unsure select "N". > diff --git a/sound/soc/mediatek/Makefile b/sound/soc/mediatek/Makefile > index f6cb6b8508e3..34778ca12106 100644 > --- a/sound/soc/mediatek/Makefile > +++ b/sound/soc/mediatek/Makefile > @@ -5,3 +5,4 @@ obj-$(CONFIG_SND_SOC_MT6797) += mt6797/ > obj-$(CONFIG_SND_SOC_MT8173) += mt8173/ > obj-$(CONFIG_SND_SOC_MT8183) += mt8183/ > obj-$(CONFIG_SND_SOC_MT8192) += mt8192/ > +obj-$(CONFIG_SND_SOC_MT8195) += mt8195/ > diff --git a/sound/soc/mediatek/mt8195/Makefile b/sound/soc/mediatek/mt8195/Makefile > new file mode 100644 > index 000000000000..b2c9fd88f39e > --- /dev/null > +++ b/sound/soc/mediatek/mt8195/Makefile > @@ -0,0 +1,11 @@ > +# SPDX-License-Identifier: GPL-2.0 > + > +# platform driver > +snd-soc-mt8195-afe-objs := \ > + mt8195-afe-clk.o \ > + mt8195-afe-pcm.o \ > + mt8195-dai-adda.o \ > + mt8195-dai-etdm.o \ > + mt8195-dai-pcm.o > + > +obj-$(CONFIG_SND_SOC_MT8195) += snd-soc-mt8195-afe.o > diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-clk.c b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c > new file mode 100644 > index 000000000000..57aa799b4f41 > --- /dev/null > +++ b/sound/soc/mediatek/mt8195/mt8195-afe-clk.c > @@ -0,0 +1,899 @@ > +// SPDX-License-Identifier: GPL-2.0 > +/* > + * mt8195-afe-clk.c -- Mediatek 8195 afe clock ctrl > + * > + * Copyright (c) 2021 MediaTek Inc. > + * Author: Bicycle Tsai <bicycle.tsai(a)mediatek.com> > + * Trevor Wu <trevor.wu(a)mediatek.com> > + */ > + > +#include <linux/clk.h> > + > +#include "mt8195-afe-common.h" > +#include "mt8195-afe-clk.h" > +#include "mt8195-reg.h" > + > +static const char *aud_clks[MT8195_CLK_NUM] = { Most of these clocks are not described in the device tree binding. If the driver needs to reference them, they should be described. We should not be hard-coding clock names across different drivers. The more important question is, why does the driver need to reference all of them? Maybe we should take a step back and draw out a clock tree diagram for the hardware? > + /* xtal */ > + [MT8195_CLK_XTAL_26M] = "clk26m", > + /* pll */ > + [MT8195_CLK_APMIXED_APLL1] = "apll1", > + [MT8195_CLK_APMIXED_APLL2] = "apll2", > + [MT8195_CLK_APMIXED_APLL3] = "apll3", > + [MT8195_CLK_APMIXED_APLL4] = "apll4", > + [MT8195_CLK_APMIXED_APLL5] = "apll5", > + [MT8195_CLK_APMIXED_HDMIRX_APLL] = "hdmirx_apll", > + /* divider */ > + [MT8195_CLK_TOP_APLL1] = "apll1_ck", > + [MT8195_CLK_TOP_APLL1_D4] = "apll1_d4", > + [MT8195_CLK_TOP_APLL2] = "apll2_ck", > + [MT8195_CLK_TOP_APLL2_D4] = "apll2_d4", > + [MT8195_CLK_TOP_APLL3] = "apll3_ck", > + [MT8195_CLK_TOP_APLL3_D4] = "apll3_d4", > + [MT8195_CLK_TOP_APLL4] = "apll4_ck", > + [MT8195_CLK_TOP_APLL4_D4] = "apll4_d4", > + [MT8195_CLK_TOP_APLL5] = "apll5_ck", > + [MT8195_CLK_TOP_APLL5_D4] = "apll5_d4", > + [MT8195_CLK_TOP_APLL12_DIV0] = "apll12_div0", > + [MT8195_CLK_TOP_APLL12_DIV1] = "apll12_div1", > + [MT8195_CLK_TOP_APLL12_DIV2] = "apll12_div2", > + [MT8195_CLK_TOP_APLL12_DIV3] = "apll12_div3", > + [MT8195_CLK_TOP_APLL12_DIV4] = "apll12_div4", > + [MT8195_CLK_TOP_APLL12_DIV9] = "apll12_div9", > + [MT8195_CLK_TOP_HDMIRX_APLL] = "hdmirx_apll_ck", > + [MT8195_CLK_TOP_MAINPLL_D4_D4] = "mainpll_d4_d4", > + [MT8195_CLK_TOP_MAINPLL_D5_D2] = "mainpll_d5_d2", > + [MT8195_CLK_TOP_MAINPLL_D7_D2] = "mainpll_d7_d2", > + [MT8195_CLK_TOP_UNIVPLL_D4] = "univpll_d4", > + /* mux */ > + [MT8195_CLK_TOP_APLL1_SEL] = "apll1_sel", > + [MT8195_CLK_TOP_APLL2_SEL] = "apll2_sel", > + [MT8195_CLK_TOP_APLL3_SEL] = "apll3_sel", > + [MT8195_CLK_TOP_APLL4_SEL] = "apll4_sel", > + [MT8195_CLK_TOP_APLL5_SEL] = "apll5_sel", > + [MT8195_CLK_TOP_A1SYS_HP_SEL] = "a1sys_hp_sel", > + [MT8195_CLK_TOP_A2SYS_SEL] = "a2sys_sel", > + [MT8195_CLK_TOP_A3SYS_SEL] = "a3sys_sel", > + [MT8195_CLK_TOP_A4SYS_SEL] = "a4sys_sel", > + [MT8195_CLK_TOP_ASM_H_SEL] = "asm_h_sel", > + [MT8195_CLK_TOP_ASM_M_SEL] = "asm_m_sel", > + [MT8195_CLK_TOP_ASM_L_SEL] = "asm_l_sel", > + [MT8195_CLK_TOP_AUD_IEC_SEL] = "aud_iec_sel", > + [MT8195_CLK_TOP_AUD_INTBUS_SEL] = "aud_intbus_sel", > + [MT8195_CLK_TOP_AUDIO_H_SEL] = "audio_h_sel", > + [MT8195_CLK_TOP_AUDIO_LOCAL_BUS_SEL] = "audio_local_bus_sel", > + [MT8195_CLK_TOP_DPTX_M_SEL] = "dptx_m_sel", > + [MT8195_CLK_TOP_INTDIR_SEL] = "intdir_sel", > + [MT8195_CLK_TOP_I2SO1_M_SEL] = "i2so1_m_sel", > + [MT8195_CLK_TOP_I2SO2_M_SEL] = "i2so2_m_sel", > + [MT8195_CLK_TOP_I2SI1_M_SEL] = "i2si1_m_sel", > + [MT8195_CLK_TOP_I2SI2_M_SEL] = "i2si2_m_sel", > + /* clock gate */ > + [MT8195_CLK_TOP_MPHONE_SLAVE_B] = "mphone_slave_b", > + [MT8195_CLK_TOP_CFG_26M_AUD] = "cfg_26m_aud", > + [MT8195_CLK_INFRA_AO_AUDIO] = "infra_ao_audio", > + [MT8195_CLK_INFRA_AO_AUDIO_26M_B] = "infra_ao_audio_26m_b", > + [MT8195_CLK_SCP_ADSP_AUDIODSP] = "scp_adsp_audiodsp", > + [MT8195_CLK_AUD_AFE] = "aud_afe", > + [MT8195_CLK_AUD_LRCK_CNT] = "aud_lrck_cnt", > + [MT8195_CLK_AUD_SPDIFIN_TUNER_APLL] = "aud_spdifin_tuner_apll", > + [MT8195_CLK_AUD_SPDIFIN_TUNER_DBG] = "aud_spdifin_tuner_dbg", > + [MT8195_CLK_AUD_UL_TML] = "aud_ul_tml", > + [MT8195_CLK_AUD_APLL1_TUNER] = "aud_apll1_tuner", > + [MT8195_CLK_AUD_APLL2_TUNER] = "aud_apll2_tuner", > + [MT8195_CLK_AUD_TOP0_SPDF] = "aud_top0_spdf", > + [MT8195_CLK_AUD_APLL] = "aud_apll", > + [MT8195_CLK_AUD_APLL2] = "aud_apll2", > + [MT8195_CLK_AUD_DAC] = "aud_dac", > + [MT8195_CLK_AUD_DAC_PREDIS] = "aud_dac_predis", > + [MT8195_CLK_AUD_TML] = "aud_tml", > + [MT8195_CLK_AUD_ADC] = "aud_adc", > + [MT8195_CLK_AUD_DAC_HIRES] = "aud_dac_hires", > + [MT8195_CLK_AUD_A1SYS_HP] = "aud_a1sys_hp", > + [MT8195_CLK_AUD_AFE_DMIC1] = "aud_afe_dmic1", > + [MT8195_CLK_AUD_AFE_DMIC2] = "aud_afe_dmic2", > + [MT8195_CLK_AUD_AFE_DMIC3] = "aud_afe_dmic3", > + [MT8195_CLK_AUD_AFE_DMIC4] = "aud_afe_dmic4", > + [MT8195_CLK_AUD_AFE_26M_DMIC_TM] = "aud_afe_26m_dmic_tm", > + [MT8195_CLK_AUD_UL_TML_HIRES] = "aud_ul_tml_hires", > + [MT8195_CLK_AUD_ADC_HIRES] = "aud_adc_hires", > + [MT8195_CLK_AUD_ADDA6_ADC] = "aud_adda6_adc", > + [MT8195_CLK_AUD_ADDA6_ADC_HIRES] = "aud_adda6_adc_hires", > + [MT8195_CLK_AUD_LINEIN_TUNER] = "aud_linein_tuner", > + [MT8195_CLK_AUD_EARC_TUNER] = "aud_earc_tuner", > + [MT8195_CLK_AUD_I2SIN] = "aud_i2sin", > + [MT8195_CLK_AUD_TDM_IN] = "aud_tdm_in", > + [MT8195_CLK_AUD_I2S_OUT] = "aud_i2s_out", > + [MT8195_CLK_AUD_TDM_OUT] = "aud_tdm_out", > + [MT8195_CLK_AUD_HDMI_OUT] = "aud_hdmi_out", > + [MT8195_CLK_AUD_ASRC11] = "aud_asrc11", > + [MT8195_CLK_AUD_ASRC12] = "aud_asrc12", > + [MT8195_CLK_AUD_MULTI_IN] = "aud_multi_in", > + [MT8195_CLK_AUD_INTDIR] = "aud_intdir", > + [MT8195_CLK_AUD_A1SYS] = "aud_a1sys", > + [MT8195_CLK_AUD_A2SYS] = "aud_a2sys", > + [MT8195_CLK_AUD_PCMIF] = "aud_pcmif", > + [MT8195_CLK_AUD_A3SYS] = "aud_a3sys", > + [MT8195_CLK_AUD_A4SYS] = "aud_a4sys", > + [MT8195_CLK_AUD_MEMIF_UL1] = "aud_memif_ul1", > + [MT8195_CLK_AUD_MEMIF_UL2] = "aud_memif_ul2", > + [MT8195_CLK_AUD_MEMIF_UL3] = "aud_memif_ul3", > + [MT8195_CLK_AUD_MEMIF_UL4] = "aud_memif_ul4", > + [MT8195_CLK_AUD_MEMIF_UL5] = "aud_memif_ul5", > + [MT8195_CLK_AUD_MEMIF_UL6] = "aud_memif_ul6", > + [MT8195_CLK_AUD_MEMIF_UL8] = "aud_memif_ul8", > + [MT8195_CLK_AUD_MEMIF_UL9] = "aud_memif_ul9", > + [MT8195_CLK_AUD_MEMIF_UL10] = "aud_memif_ul10", > + [MT8195_CLK_AUD_MEMIF_DL2] = "aud_memif_dl2", > + [MT8195_CLK_AUD_MEMIF_DL3] = "aud_memif_dl3", > + [MT8195_CLK_AUD_MEMIF_DL6] = "aud_memif_dl6", > + [MT8195_CLK_AUD_MEMIF_DL7] = "aud_memif_dl7", > + [MT8195_CLK_AUD_MEMIF_DL8] = "aud_memif_dl8", > + [MT8195_CLK_AUD_MEMIF_DL10] = "aud_memif_dl10", > + [MT8195_CLK_AUD_MEMIF_DL11] = "aud_memif_dl11", > + [MT8195_CLK_AUD_GASRC0] = "aud_gasrc0", > + [MT8195_CLK_AUD_GASRC1] = "aud_gasrc1", > + [MT8195_CLK_AUD_GASRC2] = "aud_gasrc2", > + [MT8195_CLK_AUD_GASRC3] = "aud_gasrc3", > + [MT8195_CLK_AUD_GASRC4] = "aud_gasrc4", > + [MT8195_CLK_AUD_GASRC5] = "aud_gasrc5", > + [MT8195_CLK_AUD_GASRC6] = "aud_gasrc6", > + [MT8195_CLK_AUD_GASRC7] = "aud_gasrc7", > + [MT8195_CLK_AUD_GASRC8] = "aud_gasrc8", > + [MT8195_CLK_AUD_GASRC9] = "aud_gasrc9", > + [MT8195_CLK_AUD_GASRC10] = "aud_gasrc10", > + [MT8195_CLK_AUD_GASRC11] = "aud_gasrc11", > + [MT8195_CLK_AUD_GASRC12] = "aud_gasrc12", > + [MT8195_CLK_AUD_GASRC13] = "aud_gasrc13", > + [MT8195_CLK_AUD_GASRC14] = "aud_gasrc14", > + [MT8195_CLK_AUD_GASRC15] = "aud_gasrc15", > + [MT8195_CLK_AUD_GASRC16] = "aud_gasrc16", > + [MT8195_CLK_AUD_GASRC17] = "aud_gasrc17", > + [MT8195_CLK_AUD_GASRC18] = "aud_gasrc18", > + [MT8195_CLK_AUD_GASRC19] = "aud_gasrc19", The MT8195_CLK_AUD_* clocks are all internal to the audio subsystem: the bits that control these clock gates are in the same address space as the audio parts. Would it be possible to model them as internal ASoC SUPPLY widgets? The external ones could be modeled using ASoC CLK_SUPPLY widgets, and the dependencies could be modeled with ASoC routes. The ASoC core could then handle power sequencing, which the driver currently does manually. IMO this is better than having two drivers handling two aspects of the same piece of hardware, while the two aspects are intertwined. Regards ChenYu

2 9

[PATCH] sound: usb: get lock before calling usb_enable_autosuspend()
by Salah Triki 03 Aug '21

03 Aug '21

Based on the documentation of usb_enable_autosuspend(), the caller must hold udev's device lock. Signed-off-by: Salah Triki <salah.triki(a)gmail.com> --- sound/usb/card.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f6a62416c05..9290ba0acd5f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -907,8 +907,13 @@ static void usb_audio_disconnect(struct usb_interface *intf) } } - if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) - usb_enable_autosuspend(interface_to_usbdev(intf)); + if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) { + struct usb_device *udev = interface_to_usbdev(intf); + + usb_lock_device(udev); + usb_enable_autosuspend(udev); + usb_unlock_device(udev); + } chip->num_interfaces--; if (chip->num_interfaces <= 0) { -- 2.25.1

2 1

dmix optimization
by Giuliano Zannetti - ART S.p.A. 03 Aug '21

03 Aug '21

Hi, I'm trying to optimize the dmix because I'm working with a big number of channels (up to 16) and in this case the dmix has not a negligible impact on performance. I'm working with ALSA 1.1.9. I gave my first look to the generic_mix_areas_16_native function (https://github.com/alsa-project/alsa-lib/blob/v1.1.9/src/pcm/pcm_dmix_gener…). I would ask you if I can avoid to check, for each loop iteration, if the current dst sample is not 0. for (;;) { sample = *src; if (! *dst) { *sum = sample; *dst = *src; } else { sample += *sum; *sum = sample; if (sample > 0x7fff) sample = 0x7fff; else if (sample < -0x8000) sample = -0x8000; *dst = sample; } if (!--size) return; src = (signed short *) ((char *)src + src_step); dst = (signed short *) ((char *)dst + dst_step); sum = (signed int *) ((char *)sum + sum_step); } Could it be possible check for the first sample of the period only, as reported in the code below? My assumption is that if dst[0] is 0 also dst[1] ... dst[period-1] will be 0, and I don't need to check every time. This is already an optimization, but it could be also a starting point for other optimization based on my HW. But, first of all, I would ask to you if my assumption is right. if (! *dst) { for (;;) { sample = *src; *sum = sample; *dst = *src; if (!--size) return; src = (signed short *) ((char *)src + src_step); dst = (signed short *) ((char *)dst + dst_step); sum = (signed int *) ((char *)sum + sum_step); } } else { for (;;) { sample = *src; sample += *sum; *sum = sample; if (sample > 0x7fff) sample = 0x7fff; else if (sample < -0x8000) sample = -0x8000; *dst = sample; if (!--size) return; src = (signed short *) ((char *)src + src_step); dst = (signed short *) ((char *)dst + dst_step); sum = (signed int *) ((char *)sum + sum_step); } } Thank you! Best Regards, Giuliano

3 3

[PATCH v3 0/4] ASoC: codecs: wcd934x: add Headset and button detection support
by Srinivas Kandagatla 03 Aug '21

03 Aug '21

This patchset adds support to MBHC(Multi Button Headset Control) block found in Qualcomm WCD codecs. MBHC support headset type detection, both Mechanical and electrical insert/removal detection along with 8 buttons detection, Over current interrupts on HPHL/R, Impedance Measurements on HPHL/R. Eventhough MBHC block supports things like OverCurrent detection, Currently its reported as a kernel debug message. Should this be reported as an uevent to userspace? like the way USB reports? Any suggestions? First patch adds a common mbhc driver and the second one wcd934x specific driver changes along with sdm845 soundcard related changes. Common wcd-mbhc-v2 driver should be reusable across multiple codecs like WCD9335, WCD934x, WCD937x and WCD938x. Most of the work is derived from downstream Qualcomm kernels. Credits to various Qualcomm authors from Patrick Lai's team who have contributed to this code. Changes since v2: - switched to EXPORT_SYMBOL_GPL from EXPORT_SYMBOL - converted one of the if else to switch case. Srinivas Kandagatla (4): ASoC: dt-bindings: wcd934x: add bindings for Headset Button detection ASoC: codecs: wcd: add multi button Headset detection support ASoC: codecs: wcd934x: add mbhc support ASoC: qcom: sdm845: add jack support for WCD934x .../bindings/sound/qcom,wcd934x.yaml | 30 + include/linux/mfd/wcd934x/registers.h | 57 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd-mbhc-v2.c | 1475 +++++++++++++++++ sound/soc/codecs/wcd-mbhc-v2.h | 340 ++++ sound/soc/codecs/wcd934x.c | 884 +++++++++- sound/soc/qcom/sdm845.c | 8 + 8 files changed, 2785 insertions(+), 15 deletions(-) create mode 100644 sound/soc/codecs/wcd-mbhc-v2.c create mode 100644 sound/soc/codecs/wcd-mbhc-v2.h -- 2.21.0

3 6

[PATCH v2 0/6] soundwire/ASoC: abstract platform-dependent bases
by Bard Liao 02 Aug '21

02 Aug '21

shim base and alh base are platform-dependent. This series suggests to use variables for those bases. It allows us to use different bases for new platforms. v2: - Update the commit message of "soundwire: move intel sdw register definitions to sdw_intel.h" Bard Liao (6): soundwire: move intel sdw register definitions to sdw_intel.h ASoC: SOF: intel: add sdw_shim/alh_base to sof_intel_dsp_desc ASoC: SOF: intel: hda: remove HDA_DSP_REG_SNDW_WAKE_STS definition ASoC: SOF: intel: move sof_intel_dsp_desc() forward ASoC: SOF: intel: add snd_sof_dsp_check_sdw_irq ops soundwire: intel: introduce shim and alh base drivers/soundwire/intel.c | 74 ------------------------ drivers/soundwire/intel_init.c | 14 ++--- include/linux/soundwire/sdw_intel.h | 87 +++++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 6 ++ sound/soc/sof/intel/hda.c | 39 ++++++++----- sound/soc/sof/intel/hda.h | 8 ++- sound/soc/sof/intel/icl.c | 3 + sound/soc/sof/intel/shim.h | 3 + sound/soc/sof/intel/tgl.c | 12 ++++ 9 files changed, 149 insertions(+), 97 deletions(-) -- 2.17.1

4 12

[PATCH] ASoC: tlv320aic32x4: make array clocks static, makes object smaller
by Colin King 02 Aug '21

02 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the array clocks on the stack but instead it static. Makes the object code smaller by 316 bytes. Before: text data bss dec hex filename 63668 28264 0 91932 1671c ./sound/soc/codecs/tlv320aic32x4.o After: text data bss dec hex filename 62616 29000 0 91616 165e0 ./sound/soc/codecs/tlv320aic32x4.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tlv320aic32x4.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 254a016cb1f3..d31ffb6d5b4b 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -714,7 +714,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned long adc_clock_rate, dac_clock_rate; int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "pll" }, { .id = "nadc" }, { .id = "madc" }, @@ -878,7 +878,7 @@ static int aic32x4_set_bias_level(struct snd_soc_component *component, { int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "madc" }, { .id = "mdac" }, { .id = "bdiv" }, @@ -994,7 +994,7 @@ static int aic32x4_component_probe(struct snd_soc_component *component) u32 tmp_reg; int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "codec_clkin" }, { .id = "pll" }, { .id = "bdiv" }, @@ -1144,7 +1144,7 @@ static int aic32x4_tas2505_component_probe(struct snd_soc_component *component) u32 tmp_reg; int ret; - struct clk_bulk_data clocks[] = { + static struct clk_bulk_data clocks[] = { { .id = "codec_clkin" }, { .id = "pll" }, { .id = "bdiv" }, -- 2.31.1

3 2

[PATCH v2 0/6] ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 audio routing
by Hans de Goede 02 Aug '21

02 Aug '21

Changes in v2: - Only set lineout_string if BYT_RT5640_LINEOUT is set, since BYT_RT5640_LINEOUT_AS_HP2 only works if the lineout is enabled in the first place Original cover-letter: The HP Elitepad 1000 G2 has 2 headset jacks: 1. on the dock which uses the output of the codecs built-in HP-amp + the standard IN2 input which is always used with the headset-jack. 2. on the tablet itself, this uses the line-out of the codec, combined with an external HP-amp + IN1 for the headset-mic. This series adds support for this, resolving: https://bugzilla.kernel.org/show_bug.cgi?id=213415 Note this series does not add jack-detect support. I plan to add that with a follow-up series when I can make some time to implement that. Regards, Hans Hans de Goede (6): ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output ASoC: Intel: bytcr_rt5640: Add line-out support ASoC: Intel: bytcr_rt5640: Add a byt_rt5640_get_codec_dai() helper ASoC: Intel: bytcr_rt5640: Add support for a second headphones output ASoC: Intel: bytcr_rt5640: Add support for a second headset mic input ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 quirk sound/soc/intel/boards/bytcr_rt5640.c | 118 ++++++++++++++++++++++---- 1 file changed, 102 insertions(+), 16 deletions(-) -- 2.31.1

3 8

Linux daemon +ALSA+pulseaudio problem
by Valery Savchuk 02 Aug '21

02 Aug '21

Hello! I am developing RadioAmateur software AlterFRN-client http://alterfrn.ucoz.ru/ which is using ALSA on Linux. The problem is: the AlterFRN-client daemon (background mode) can not initialize ALSA if pusleaudio is used/running also. In foreground mode all things are OK. AlsaInfo is attached also.

1 0

[PATCH] ASoC: rsnd: make some arrays static const, makes object smaller
by Colin King 02 Aug '21

02 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate arrays on the stack but instead them static const. Makes the object code smaller by 242 bytes. Before: text data bss dec hex filename 23827 8764 0 32591 7f4f ./sound/soc/sh/rcar/ssi.o After: text data bss dec hex filename 23361 8988 0 32349 7e5d ./sound/soc/sh/rcar/ssi.o gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sh/rcar/ssi.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 27f34ca6059d..87e606f688d3 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -165,7 +165,7 @@ static void rsnd_ssi_status_check(struct rsnd_mod *mod, static u32 rsnd_ssi_multi_secondaries(struct rsnd_dai_stream *io) { - enum rsnd_mod_type types[] = { + static const enum rsnd_mod_type types[] = { RSND_MOD_SSIM1, RSND_MOD_SSIM2, RSND_MOD_SSIM3, @@ -228,7 +228,7 @@ unsigned int rsnd_ssi_clk_query(struct rsnd_dai *rdai, int param1, int param2, int *idx) { struct rsnd_priv *priv = rsnd_rdai_to_priv(rdai); - int ssi_clk_mul_table[] = { + static const int ssi_clk_mul_table[] = { 1, 2, 4, 8, 16, 6, 12, }; int j, ret; @@ -1079,7 +1079,7 @@ static void rsnd_ssi_connect(struct rsnd_mod *mod, struct rsnd_dai_stream *io) { struct rsnd_dai *rdai = rsnd_io_to_rdai(io); - enum rsnd_mod_type types[] = { + static const enum rsnd_mod_type types[] = { RSND_MOD_SSI, RSND_MOD_SSIM1, RSND_MOD_SSIM2, -- 2.31.1

3 2

[PATCH v2] ASoC: amd: Fix reference to PCM buffer address
by Takashi Iwai 02 Aug '21

02 Aug '21

PCM buffers might be allocated dynamically when the buffer preallocation failed or a larger buffer is requested, and it's not guaranteed that substream->dma_buffer points to the actually used buffer. The driver needs to refer to substream->runtime->dma_addr instead for the buffer address. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: Fix build error of renoir driver sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 143155a840ac..cc1ce6f22caa 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -969,7 +969,7 @@ static int acp_dma_hw_params(struct snd_soc_component *component, acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = runtime->dma_addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 8148b0d22e88..597d7c4b2a6b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -286,7 +286,7 @@ static int acp3x_dma_hw_params(struct snd_soc_component *component, pr_err("pinfo failed\n"); } size = params_buffer_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp3x_dma(rtd, substream->stream); return 0; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index bd20622b0933..0391c28dd078 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -242,7 +242,7 @@ static int acp_pdm_dma_hw_params(struct snd_soc_component *component, return -EINVAL; size = params_buffer_bytes(params); period_bytes = params_period_bytes(params); - rtd->dma_addr = substream->dma_buffer.addr; + rtd->dma_addr = substream->runtime->dma_addr; rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); config_acp_dma(rtd, substream->stream); init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes, -- 2.26.2

2 1

[PATCH 0/5] ASoC: soc-topology: cleanup cppcheck warning
by Kuninori Morimoto 02 Aug '21

02 Aug '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-topology. Kuninori Morimoto (5): ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_process_headers() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_dai_elems_load() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_dapm_widget_elems_load() ASoC: soc-topology: cleanup cppcheck warning at soc_tplg_kcontrol_elems_load() ASoC: soc-topology: cleanup cppcheck warning at snd_soc_find_dai_link() sound/soc/soc-topology.c | 23 +++++++++++------------ 1 file changed, 11 insertions(+), 12 deletions(-) -- 2.25.1

2 6

[PATCH 0/6] ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 audio routing
by Hans de Goede 02 Aug '21

02 Aug '21

The HP Elitepad 1000 G2 has 2 headset jacks: 1. on the dock which uses the output of the codecs built-in HP-amp + the standard IN2 input which is always used with the headset-jack. 2. on the tablet itself, this uses the line-out of the codec, combined with an external HP-amp + IN1 for the headset-mic. This series adds support for this, resolving: https://bugzilla.kernel.org/show_bug.cgi?id=213415 Note this series does not add jack-detect support. I plan to add that with a follow-up series when I can make some time to implement that. Regards, Hans Hans de Goede (6): ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output ASoC: Intel: bytcr_rt5640: Add line-out support ASoC: Intel: bytcr_rt5640: Add a byt_rt5640_get_codec_dai() helper ASoC: Intel: bytcr_rt5640: Add support for a second headphones output ASoC: Intel: bytcr_rt5640: Add support for a second headset mic input ASoC: Intel: bytcr_rt5640: Fix HP ElitePad 1000 G2 quirk sound/soc/intel/boards/bytcr_rt5640.c | 116 ++++++++++++++++++++++---- 1 file changed, 100 insertions(+), 16 deletions(-) -- 2.31.1

2 10

[PATCH v2 0/8] Add RZ/G2L Sound support
by Biju Das 02 Aug '21

02 Aug '21

This patch series aims to add ASoC support on RZ/G2L SoC's. It is based on the work done by Chris Brandt for RZ/A ASoC driver. Biju Das (8): ASoC: dt-bindings: Document RZ/G2L bindings sound: soc: sh: Add RZ/G2L SSIF-2 driver arm64: dts: renesas: r9a07g044: Add external audio clock nodes arm64: dts: renesas: r9a07g044: Add SSI support arm64: defconfig: Enable ASoC sound support for RZ/G2L SoC ASoC: dt-bindings: sound: renesas,rz-ssi: Document DMA support sound: sh: rz-ssi: Add SSI DMAC support arm64: dts: renesas: r9a07g044: Add SSI DMA support .../bindings/sound/renesas,rz-ssi.yaml | 100 ++ arch/arm64/boot/dts/renesas/r9a07g044.dtsi | 105 ++ arch/arm64/configs/defconfig | 1 + sound/soc/sh/Kconfig | 10 + sound/soc/sh/Makefile | 4 + sound/soc/sh/rz-ssi.c | 1078 +++++++++++++++++ 6 files changed, 1298 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml create mode 100644 sound/soc/sh/rz-ssi.c -- 2.17.1

6 15

[PATCH] ASoC: intel: skylake: Drop superfluous mmap callback
by Takashi Iwai 02 Aug '21

02 Aug '21

skl_platform_soc_mmap() just calls the standard mmap helper, hence it's superfluous. Let's drop it. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/intel/skylake/skl-pcm.c | 8 -------- 1 file changed, 8 deletions(-) diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index b1ca64d2f7ea..c604200de80e 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1214,13 +1214,6 @@ static snd_pcm_uframes_t skl_platform_soc_pointer( return bytes_to_frames(substream->runtime, pos); } -static int skl_platform_soc_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *area) -{ - return snd_pcm_lib_default_mmap(substream, area); -} - static u64 skl_adjust_codec_delay(struct snd_pcm_substream *substream, u64 nsec) { @@ -1460,7 +1453,6 @@ static const struct snd_soc_component_driver skl_component = { .trigger = skl_platform_soc_trigger, .pointer = skl_platform_soc_pointer, .get_time_info = skl_platform_soc_get_time_info, - .mmap = skl_platform_soc_mmap, .pcm_construct = skl_platform_soc_new, .module_get_upon_open = 1, /* increment refcount when a pcm is opened */ }; -- 2.26.2

3 5

[PATCH 0/5] ASoC: Fix reference to PCM buffer address
by Takashi Iwai 02 Aug '21

02 Aug '21

Hi, while updating the memory allocation API for WC pages, I casually stumbled on quite a few drivers dealing with the PCM buffer address wrongly via substream->dma_buffer.addr. This patch series corrects those usages. Takashi === Takashi Iwai (5): ASoC: amd: Fix reference to PCM buffer address ASoC: intel: atom: Fix reference to PCM buffer address ASoC: xilinx: Fix reference to PCM buffer address ASoC: uniphier: Fix reference to PCM buffer address ASoC: kirkwood: Fix reference to PCM buffer address sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 3 +-- sound/soc/kirkwood/kirkwood-dma.c | 26 ++++++++++++++------ sound/soc/uniphier/aio-dma.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 4 +-- 7 files changed, 25 insertions(+), 16 deletions(-) -- 2.26.2

3 9

[PATCH 00/10] soundwire/ASoC: add mockup codec support
by Bard Liao 02 Aug '21

02 Aug '21

Adding mockup SoundWire codec is useful to debug driver/topology changes without having any actual device connected. Bard Liao (2): soundwire: stream: don't abort bank switch on Command_Ignored/-ENODATA soundwire: stream: don't program mockup device ports Pierre-Louis Bossart (8): ASoC: codecs: add SoundWire mockup device support ASoC: soc-acpi: cnl: add table for SoundWire mockup devices ASoC: soc-acpi: tgl: add table for SoundWire mockup devices ASoC: Intel: boards: sof_sdw: add SoundWire mockup codecs for tests soundwire: add flag to ignore all command/control for mockup devices soundwire: bus: squelch error returned by mockup devices soundwire: cadence: add debugfs interface for PDI loopbacks soundwire: cadence: override PDI configurations to create loopback drivers/soundwire/bus.c | 10 +- drivers/soundwire/cadence_master.c | 174 ++++++++-- drivers/soundwire/cadence_master.h | 3 + drivers/soundwire/stream.c | 5 +- include/linux/soundwire/sdw.h | 3 + sound/soc/codecs/Kconfig | 18 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sdw-mockup.c | 312 ++++++++++++++++++ sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_sdw.c | 41 +++ sound/soc/intel/common/Makefile | 3 +- .../intel/common/soc-acpi-intel-cnl-match.c | 15 + .../common/soc-acpi-intel-sdw-mockup-match.c | 166 ++++++++++ .../common/soc-acpi-intel-sdw-mockup-match.h | 17 + .../intel/common/soc-acpi-intel-tgl-match.c | 23 ++ 15 files changed, 754 insertions(+), 39 deletions(-) create mode 100644 sound/soc/codecs/sdw-mockup.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-sdw-mockup-match.h -- 2.17.1

4 15

[PATCH] soundwire: cadence: add paranoid check on self-clearing bits
by Bard Liao 02 Aug '21

02 Aug '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The Cadence IP exposes a small number of self-clearing bits in the MCP_CONTROL and MCP_CONFIG_UPDATE registers. We currently do not check that those bits are indeed cleared, e.g. during resume operations. That could lead to resuming peripheral devices too early. In addition, if we happen to read these registers, update one of the fields and write the register back, we may be writing stale data that might have been cleared in hardware. These sort of race conditions could lead to e.g. doing a hw_reset twice or stopping a clock that just restarted. There is no clear way of avoiding these potential race conditions other than making sure that these registers fields are cleared before any read-modify-write sequence. If we detect this sort of errors, we only log them since there is no clear recovery possible. The only way out is likely to restart the IP with a suspend/resume cycle. Note that the checks are performed before updating the registers, as well as after the Intel 'sync go' sequence in multi-link mode. That should cover both the start and end of suspend/resume hardware configurations. The Multi-Master mode gates the configuration updates until the 'sync go' signal is asserted, so we only check on init and after the end of the 'sync go' sequence. The duration of the usleep_range() was defined by the GSYNC frequency used in multi-master mode. With a 4kHz frequency, any configuration change might be deferred by up to 250us. Extending the range to 1000-1500us should guarantee that the configuration change is completed without any significant impact on the overall resume time. Suggested-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 47 ++++++++++++++++++++++++++++++ drivers/soundwire/cadence_master.h | 4 +++ drivers/soundwire/intel.c | 14 +++++++++ 3 files changed, 65 insertions(+) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index c7372519452b..0b7f037e6cd0 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -979,6 +979,49 @@ static void cdns_update_slave_status_work(struct work_struct *work) } +/* paranoia check to make sure self-cleared bits are indeed cleared */ +void sdw_cdns_check_self_clearing_bits(struct sdw_cdns *cdns, const char *string, + bool initial_delay, int reset_iterations) +{ + u32 mcp_control; + u32 mcp_config_update; + int i; + + if (initial_delay) + usleep_range(1000, 1500); + + mcp_control = cdns_readl(cdns, CDNS_MCP_CONTROL); + + /* the following bits should be cleared immediately */ + if (mcp_control & CDNS_MCP_CONTROL_CMD_RST) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_CMD_RST is not cleared\n", string); + if (mcp_control & CDNS_MCP_CONTROL_SOFT_RST) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_SOFT_RST is not cleared\n", string); + if (mcp_control & CDNS_MCP_CONTROL_SW_RST) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_SW_RST is not cleared\n", string); + if (mcp_control & CDNS_MCP_CONTROL_CLK_STOP_CLR) + dev_err(cdns->dev, "%s failed: MCP_CONTROL_CLK_STOP_CLR is not cleared\n", string); + mcp_config_update = cdns_readl(cdns, CDNS_MCP_CONFIG_UPDATE); + if (mcp_config_update & CDNS_MCP_CONFIG_UPDATE_BIT) + dev_err(cdns->dev, "%s failed: MCP_CONFIG_UPDATE_BIT is not cleared\n", string); + + i = 0; + while (mcp_control & CDNS_MCP_CONTROL_HW_RST) { + if (i == reset_iterations) { + dev_err(cdns->dev, "%s failed: MCP_CONTROL_HW_RST is not cleared\n", string); + break; + } + + dev_dbg(cdns->dev, "%s: MCP_CONTROL_HW_RST is not cleared at iteration %d\n", string, i); + i++; + + usleep_range(1000, 1500); + mcp_control = cdns_readl(cdns, CDNS_MCP_CONTROL); + } + +} +EXPORT_SYMBOL(sdw_cdns_check_self_clearing_bits); + /* * init routines */ @@ -1256,6 +1299,8 @@ int sdw_cdns_init(struct sdw_cdns *cdns) cdns_init_clock_ctrl(cdns); + sdw_cdns_check_self_clearing_bits(cdns, __func__, false, 0); + /* reset msg_count to default value of FIFOLEVEL */ cdns->msg_count = cdns_readl(cdns, CDNS_MCP_FIFOLEVEL); @@ -1500,6 +1545,8 @@ int sdw_cdns_clock_stop(struct sdw_cdns *cdns, bool block_wake) struct sdw_slave *slave; int ret; + sdw_cdns_check_self_clearing_bits(cdns, __func__, false, 0); + /* Check suspend status */ if (sdw_cdns_is_clock_stop(cdns)) { dev_dbg(cdns->dev, "Clock is already stopped\n"); diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index 6c039d456ba8..e587aede63bf 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -187,4 +187,8 @@ int cdns_bus_conf(struct sdw_bus *bus, struct sdw_bus_params *params); int cdns_set_sdw_stream(struct snd_soc_dai *dai, void *stream, bool pcm, int direction); + +void sdw_cdns_check_self_clearing_bits(struct sdw_cdns *cdns, const char *string, + bool initial_delay, int reset_iterations); + #endif /* __SDW_CADENCE_H */ diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 15668d6fecd6..fb9c23e13206 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -23,6 +23,7 @@ #include "intel.h" #define INTEL_MASTER_SUSPEND_DELAY_MS 3000 +#define INTEL_MASTER_RESET_ITERATIONS 10 /* * debug/config flags for the Intel SoundWire Master. @@ -1393,6 +1394,8 @@ int intel_link_startup(struct auxiliary_device *auxdev) goto err_interrupt; } } + sdw_cdns_check_self_clearing_bits(cdns, __func__, + true, INTEL_MASTER_RESET_ITERATIONS); /* Register DAIs */ ret = intel_register_dai(sdw); @@ -1709,6 +1712,8 @@ static int __maybe_unused intel_resume(struct device *dev) return ret; } } + sdw_cdns_check_self_clearing_bits(cdns, __func__, + true, INTEL_MASTER_RESET_ITERATIONS); /* * after system resume, the pm_runtime suspend() may kick in @@ -1793,6 +1798,9 @@ static int __maybe_unused intel_resume_runtime(struct device *dev) return ret; } } + sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime TEARDOWN", + true, INTEL_MASTER_RESET_ITERATIONS); + } else if (clock_stop_quirks & SDW_INTEL_CLK_STOP_BUS_RESET) { ret = intel_init(sdw); if (ret) { @@ -1866,6 +1874,9 @@ static int __maybe_unused intel_resume_runtime(struct device *dev) } } } + sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime BUS_RESET", + true, INTEL_MASTER_RESET_ITERATIONS); + } else if (!clock_stop_quirks) { clock_stop0 = sdw_cdns_is_clock_stop(&sdw->cdns); @@ -1889,6 +1900,9 @@ static int __maybe_unused intel_resume_runtime(struct device *dev) dev_err(dev, "unable to resume master during resume\n"); return ret; } + + sdw_cdns_check_self_clearing_bits(cdns, "intel_resume_runtime no_quirks", + true, INTEL_MASTER_RESET_ITERATIONS); } else { dev_err(dev, "%s clock_stop_quirks %x unsupported\n", __func__, clock_stop_quirks); -- 2.17.1

2 1

[sound:topic/asoc-pcm-addr-fixes 20/24] sound/soc/amd/renoir/acp3x-pdm-dma.c:245:18: error: 'runtime' undeclared
by kernel test robot 01 Aug '21

01 Aug '21

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git topic/asoc-pcm-addr-fixes head: 3ed85d6b39dbdab30d06d6c686739e12c0793cde commit: c4bb3c6064991e5668115533a840609b5dedfb8e [20/24] ASoC: amd: Fix reference to PCM buffer address config: x86_64-randconfig-a006-20210728 (attached as .config) compiler: gcc-10 (Ubuntu 10.3.0-1ubuntu1~20.04) 10.3.0 reproduce (this is a W=1 build): # https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/?id=… git remote add sound https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git git fetch --no-tags sound topic/asoc-pcm-addr-fixes git checkout c4bb3c6064991e5668115533a840609b5dedfb8e # save the attached .config to linux build tree mkdir build_dir make W=1 O=build_dir ARCH=x86_64 SHELL=/bin/bash sound/soc/amd/renoir/ If you fix the issue, kindly add following tag as appropriate Reported-by: kernel test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/amd/renoir/acp3x-pdm-dma.c: In function 'acp_pdm_dma_hw_params': >> sound/soc/amd/renoir/acp3x-pdm-dma.c:245:18: error: 'runtime' undeclared (first use in this function) 245 | rtd->dma_addr = runtime->dma_addr; | ^~~~~~~ sound/soc/amd/renoir/acp3x-pdm-dma.c:245:18: note: each undeclared identifier is reported only once for each function it appears in vim +/runtime +245 sound/soc/amd/renoir/acp3x-pdm-dma.c 232 233 static int acp_pdm_dma_hw_params(struct snd_soc_component *component, 234 struct snd_pcm_substream *substream, 235 struct snd_pcm_hw_params *params) 236 { 237 struct pdm_stream_instance *rtd; 238 size_t size, period_bytes; 239 240 rtd = substream->runtime->private_data; 241 if (!rtd) 242 return -EINVAL; 243 size = params_buffer_bytes(params); 244 period_bytes = params_period_bytes(params); > 245 rtd->dma_addr = runtime->dma_addr; 246 rtd->num_pages = (PAGE_ALIGN(size) >> PAGE_SHIFT); 247 config_acp_dma(rtd, substream->stream); 248 init_pdm_ring_buffer(MEM_WINDOW_START, size, period_bytes, 249 rtd->acp_base); 250 return 0; 251 } 252 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

[PATCH] ALSA: memalloc: Fix regression with SNDRV_DMA_TYPE_CONTINUOUS
by Takashi Iwai 01 Aug '21

01 Aug '21

The recent code refactoring made the mmap of continuous pages to be done via the own helper snd_dma_continuous_mmap() with remap_pfn_range(). There I overlooked that dmab->addr isn't set for the allocation with SNDRV_DMA_TYPE_CONTINUOUS. This resulted always in an error at mmap with this buffer type on the system such as Intel SST Baytrail driver. This patch fixes the regression by passing the correct address. Fixes: 30b7ba6972d5 ("ALSA: core: Add continuous and vmalloc mmap ops") Reported-by: Hans de Goede <hdegoede(a)redhat.com> Link: https://lore.kernel.org/r/8d6674da-7d7b-803e-acc9-7de6cb1223fa@redhat.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 83b79edfa52d..439a358ecfe9 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -215,7 +215,7 @@ static int snd_dma_continuous_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { return remap_pfn_range(area, area->vm_start, - dmab->addr >> PAGE_SHIFT, + page_to_pfn(virt_to_page(dmab->area)), area->vm_end - area->vm_start, area->vm_page_prot); } -- 2.26.2

1 0

[5.14 regression] "ALSA: core: Add continuous and vmalloc mmap ops" breaks Intel SST audio
by Hans de Goede 01 Aug '21

01 Aug '21

Hi All;, As discussed in the "5.14 regression, Intel SST (BYT) audio no longer works" thread, audio on X86 devices using the Intel SST driver no longer works with 5.14. After poking at this for a while I've found the culprit: 30b7ba6972d5 ("ALSA: core: Add continuous and vmalloc mmap ops") If I revert that single commit then everything works fine again with 5.14. This is with a Fedora 34 userspace using pipewire as audiodaemon Regards, Hans

2 3

[PATCH] ALSA: usb-audio: make array static const, makes object smaller
by Colin King 01 Aug '21

01 Aug '21

From: Colin Ian King <colin.king(a)canonical.com> Don't populate array names_to_check on the stack but instead it static. Makes the object code smaller by 56 bytes. Before: text data bss dec hex filename 103512 34380 0 137892 21aa4 ./sound/usb/mixer.o After: text data bss dec hex filename 103264 34572 0 137836 21a6c ./sound/usb/mixer.o gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f4cdaf1ba44a..aec2499284a5 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1572,8 +1572,9 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str) static void check_no_speaker_on_headset(struct snd_kcontrol *kctl, struct snd_card *card) { - const char *names_to_check[] = { - "Headset", "headset", "Headphone", "headphone", NULL}; + static const char *names_to_check[] = { + "Headset", "headset", "Headphone", "headphone", NULL + }; const char **s; bool found = false; -- 2.31.1

2 1

Incorrect device setting for M-Audio Audiophile USB sound card
by GitHub issues - opened 31 Jul '21

31 Jul '21

alsa-project/alsa-lib issue #166 was opened from michele-perrone: ### M-Audio Audiophile USB - fix needed There is a problem with the ALSA configuration of the "M-Audio Audiophile USB" sound card. From the linux kernel documentation (https://www.kernel.org/doc/html/latest/sound/cards/audiophile-usb.html) you can see that this card has four audio interfaces: * Analog 2-channel Input (Ai) * Analog 2-channel Output (Ao) * Digital 2-channel Input (Di) * Digital 2-channel Output (Do) These interfaces are distributed over three devices of the same card: * hw:USBtm,0 is Ao in playback and Di in capture * hw:USBtm,1 is Do in playback and Ai in capture * hw:USBtm,2 is Do in AC3/DTS passthrough mode Now, this is where ALSA makes a bit of a mess. If you look at the output of `aplay - L` (see attachment aplay_out.txt), you can see that the analog audio output is associated to CARD=USBtm,DEV=0 (which is correct), **but** you can also see that two digital outputs are assigned, one to CARD=USBtm,DEV=0 (**wrong!**) and another one to CARD=USBtm,DEV=1 (correct). Speaking now of `arecord - L` (see attachment arecord_out.txt), there is only one input assigned to CARD=USBtm,DEV=0, and it is marked as "Front output / input" instead of SPDIF digital input. And most importantly, the analog input configuration, which should correspond to CARD=USBtm,DEV=1, is completely absent. The result of this is that audio applications relying on ALSA don't know where to pick the correct inputs and outputs. If want both Ao and Ai to work in Jack, I have to select hw:USBtm,1 for the input and hw:USBtm,0 for the output. Alternatively, I have to create an `.asoundrc` file with the `asym` module (see attachment) and then run Jack with the name of my virtual device. However, this is just a workaround. I'm not familiar with the alsa source but I know my way around C code. If you tell me where to look, I may be able to help fixing this. Cheers! Michele Attachments: [aplay_out.txt](https://github.com/alsa-project/alsa-lib/files/6911115/aplay… [arecord_out.txt](https://github.com/alsa-project/alsa-lib/files/6911116/are… [asoundrc.txt](https://github.com/alsa-project/alsa-lib/files/6911119/asound… Issue URL : https://github.com/alsa-project/alsa-lib/issues/166 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: pcxhr: use __func__ to get funcion's name in an output message
by Jason Wang 31 Jul '21

31 Jul '21

Prefer using '"%s...", __func__' to get current function's name in an output message. Signed-off-by: Jason Wang <wangborong(a)cdjrlc.com> --- sound/pci/pcxhr/pcxhr.c | 47 ++++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 24 deletions(-) diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 96b9371c201a..242bd7e04b3e 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -366,7 +366,7 @@ static int pcxhr_sub_set_clock(struct pcxhr_mgr *mgr, mgr->codec_speed = speed; /* save new codec speed */ } - dev_dbg(&mgr->pci->dev, "pcxhr_sub_set_clock to %dHz (realfreq=%d)\n", + dev_dbg(&mgr->pci->dev, "%s to %dHz (realfreq=%d)\n", __func__, rate, realfreq); return 0; } @@ -499,7 +499,7 @@ static int pcxhr_set_stream_state(struct snd_pcxhr *chip, else { if (stream->status != PCXHR_STREAM_STATUS_SCHEDULE_STOP) { dev_err(chip->card->dev, - "pcxhr_set_stream_state CANNOT be stopped\n"); + "%s CANNOT be stopped\n", __func__); return -EINVAL; } start = 0; @@ -524,7 +524,7 @@ static int pcxhr_set_stream_state(struct snd_pcxhr *chip, err = pcxhr_send_msg(chip->mgr, &rmh); if (err) dev_err(chip->card->dev, - "ERROR pcxhr_set_stream_state err=%x;\n", err); + "ERROR %s err=%x;\n", __func__, err); stream->status = start ? PCXHR_STREAM_STATUS_STARTED : PCXHR_STREAM_STATUS_STOPPED; return err; @@ -570,7 +570,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) break; default: dev_err(chip->card->dev, - "error pcxhr_set_format() : unknown format\n"); + "error %s() : unknown format\n", __func__); return -EINVAL; } @@ -615,7 +615,7 @@ static int pcxhr_set_format(struct pcxhr_stream *stream) err = pcxhr_send_msg(chip->mgr, &rmh); if (err) dev_err(chip->card->dev, - "ERROR pcxhr_set_format err=%x;\n", err); + "ERROR %s err=%x;\n", __func__, err); return err; } @@ -630,7 +630,7 @@ static int pcxhr_update_r_buffer(struct pcxhr_stream *stream) stream_num = is_capture ? 0 : subs->number; dev_dbg(chip->card->dev, - "pcxhr_update_r_buffer(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", + "%s(pcm%c%d) : addr(%p) bytes(%zx) subs(%d)\n", __func__, is_capture ? 'c' : 'p', chip->chip_idx, (void *)(long)subs->runtime->dma_addr, subs->runtime->dma_bytes, subs->number); @@ -721,21 +721,20 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) } if (capture_mask == 0 && playback_mask == 0) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : no pipes\n"); + dev_err(&mgr->pci->dev, "%s : no pipes\n", __func__); return; } - dev_dbg(&mgr->pci->dev, "pcxhr_start_linked_stream : " - "playback_mask=%x capture_mask=%x\n", - playback_mask, capture_mask); + dev_dbg(&mgr->pci->dev, "%s : playback_mask=%x capture_mask=%x\n", + __func__, playback_mask, capture_mask); /* synchronous stop of all the pipes concerned */ err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 0); if (err) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : " + dev_err(&mgr->pci->dev, "%s : " "error stop pipes (P%x C%x)\n", - playback_mask, capture_mask); + __func__, playback_mask, capture_mask); return; } @@ -778,9 +777,9 @@ static void pcxhr_start_linked_stream(struct pcxhr_mgr *mgr) err = pcxhr_set_pipe_state(mgr, playback_mask, capture_mask, 1); if (err) { mutex_unlock(&mgr->setup_mutex); - dev_err(&mgr->pci->dev, "pcxhr_start_linked_stream : " + dev_err(&mgr->pci->dev, "%s : " "error start pipes (P%x C%x)\n", - playback_mask, capture_mask); + __func__, playback_mask, capture_mask); return; } @@ -889,7 +888,7 @@ static int pcxhr_hardware_timer(struct pcxhr_mgr *mgr, int start) } err = pcxhr_send_msg(mgr, &rmh); if (err < 0) - dev_err(&mgr->pci->dev, "error pcxhr_hardware_timer err(%x)\n", + dev_err(&mgr->pci->dev, "error %s err(%x)\n", __func__, err); return err; } @@ -904,7 +903,7 @@ static int pcxhr_prepare(struct snd_pcm_substream *subs) int err = 0; dev_dbg(chip->card->dev, - "pcxhr_prepare : period_size(%lx) periods(%x) buffer_size(%lx)\n", + "%s : period_size(%lx) periods(%x) buffer_size(%lx)\n", __func__, subs->runtime->period_size, subs->runtime->periods, subs->runtime->buffer_size); @@ -997,12 +996,12 @@ static int pcxhr_open(struct snd_pcm_substream *subs) runtime->hw = pcxhr_caps; if( subs->stream == SNDRV_PCM_STREAM_PLAYBACK ) { - dev_dbg(chip->card->dev, "pcxhr_open playback chip%d subs%d\n", - chip->chip_idx, subs->number); + dev_dbg(chip->card->dev, "%s playback chip%d subs%d\n", + __func__, chip->chip_idx, subs->number); stream = &chip->playback_stream[subs->number]; } else { - dev_dbg(chip->card->dev, "pcxhr_open capture chip%d subs%d\n", - chip->chip_idx, subs->number); + dev_dbg(chip->card->dev, "%s capture chip%d subs%d\n", + __func__, chip->chip_idx, subs->number); if (mgr->mono_capture) runtime->hw.channels_max = 1; else @@ -1011,8 +1010,8 @@ static int pcxhr_open(struct snd_pcm_substream *subs) } if (stream->status != PCXHR_STREAM_STATUS_FREE){ /* streams in use */ - dev_err(chip->card->dev, "pcxhr_open chip%d subs%d in use\n", - chip->chip_idx, subs->number); + dev_err(chip->card->dev, "%s chip%d subs%d in use\n", + __func__, chip->chip_idx, subs->number); mutex_unlock(&mgr->setup_mutex); return -EBUSY; } @@ -1077,7 +1076,7 @@ static int pcxhr_close(struct snd_pcm_substream *subs) mutex_lock(&mgr->setup_mutex); - dev_dbg(chip->card->dev, "pcxhr_close chip%d subs%d\n", + dev_dbg(chip->card->dev, "%s chip%d subs%d\n", __func__, chip->chip_idx, subs->number); /* sample rate released */ @@ -1572,7 +1571,7 @@ static int pcxhr_probe(struct pci_dev *pci, /* init setup mutex*/ mutex_init(&mgr->setup_mutex); - mgr->prmh = kmalloc(sizeof(*mgr->prmh) + + mgr->prmh = kmalloc(sizeof(*mgr->prmh) + sizeof(u32) * (PCXHR_SIZE_MAX_LONG_STATUS - PCXHR_SIZE_MAX_STATUS), GFP_KERNEL); -- 2.32.0

2 1

[PATCH] ALSA: core: Fix double calls of snd_card_free() via devres
by Takashi Iwai 31 Jul '21

31 Jul '21

At the transition to the devres-managed card release, we've put the check of double-free at trigger_card_release(). But this wasn't enough, as the code path calls snd_card_free() again, and it would lead to the doubly snd_card_free() calls. Actually the v1 patch was correct to handle this, but I forgot that corner case and moved the check to the more obvious place as I thought it's clearer. But, as usual, devils live in details. This patch corrects the check of the double-free to the right place, with a bit more comments. Fixes: e8ad415b7a55 ("ALSA: core: Add managed card creation") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/init.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index e985185ebc91..ac335f5906c6 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -605,6 +605,15 @@ int snd_card_free(struct snd_card *card) DECLARE_COMPLETION_ONSTACK(released); int ret; + /* The call of snd_card_free() is allowed from various code paths; + * a manual call from the driver and the call via devres_free, and + * we need to avoid double-free. Moreover, the release via devres + * may call snd_card_free() twice due to its nature, we need to have + * the check here at the beginning. + */ + if (card->releasing) + return 0; + card->release_completion = &released; ret = snd_card_free_when_closed(card); if (ret) @@ -813,10 +822,7 @@ EXPORT_SYMBOL_GPL(snd_card_add_dev_attr); static void trigger_card_free(void *data) { - struct snd_card *card = data; - - if (!card->releasing) - snd_card_free(data); + snd_card_free(data); } /** -- 2.26.2

1 0

[PATCH] ASoC: amd: Drop superfluous mmap callbacks
by Takashi Iwai 30 Jul '21

30 Jul '21

All ASoC AMD drivers just call the standard mmap handler, hence those are superfluous. Let's drop them. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/amd/acp-pcm-dma.c | 8 -------- sound/soc/amd/raven/acp3x-pcm-dma.c | 8 -------- sound/soc/amd/renoir/acp3x-pdm-dma.c | 8 -------- 3 files changed, 24 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 143155a840ac..8f202423e6ce 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1035,13 +1035,6 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_soc_component *component, return bytes_to_frames(runtime, pos); } -static int acp_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp_dma_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -1205,7 +1198,6 @@ static const struct snd_soc_component_driver acp_asoc_platform = { .hw_params = acp_dma_hw_params, .trigger = acp_dma_trigger, .pointer = acp_dma_pointer, - .mmap = acp_dma_mmap, .prepare = acp_dma_prepare, .pcm_construct = acp_dma_new, }; diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 8148b0d22e88..5f24546eeed0 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -320,13 +320,6 @@ static int acp3x_dma_new(struct snd_soc_component *component, return 0; } -static int acp3x_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp3x_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -370,7 +363,6 @@ static const struct snd_soc_component_driver acp3x_i2s_component = { .close = acp3x_dma_close, .hw_params = acp3x_dma_hw_params, .pointer = acp3x_dma_pointer, - .mmap = acp3x_dma_mmap, .pcm_construct = acp3x_dma_new, }; diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index bd20622b0933..9988a50a81b0 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -291,13 +291,6 @@ static int acp_pdm_dma_new(struct snd_soc_component *component, return 0; } -static int acp_pdm_dma_mmap(struct snd_soc_component *component, - struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - return snd_pcm_lib_default_mmap(substream, vma); -} - static int acp_pdm_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -375,7 +368,6 @@ static const struct snd_soc_component_driver acp_pdm_component = { .close = acp_pdm_dma_close, .hw_params = acp_pdm_dma_hw_params, .pointer = acp_pdm_dma_pointer, - .mmap = acp_pdm_dma_mmap, .pcm_construct = acp_pdm_dma_new, }; -- 2.26.2

2 1

[PATCH] ASoC: nau8824: Fix open coded prefix handling
by Mark Brown 30 Jul '21

30 Jul '21

As with the component layer code the nau8824 driver had been doing some open coded pin manipulation which will have been broken now the core is fixed to handle this properly, remove the open coding to avoid the issue. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/nau8824.c | 42 ++++++-------------------------------- 1 file changed, 6 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 15bd8335f667..906cdd2fb54a 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -828,36 +828,6 @@ static void nau8824_int_status_clear_all(struct regmap *regmap) } } -static void nau8824_dapm_disable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_disable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_disable_pin(dapm, pin); - } -} - -static void nau8824_dapm_enable_pin(struct nau8824 *nau8824, const char *pin) -{ - struct snd_soc_dapm_context *dapm = nau8824->dapm; - const char *prefix = dapm->component->name_prefix; - char prefixed_pin[80]; - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - snd_soc_dapm_force_enable_pin(dapm, prefixed_pin); - } else { - snd_soc_dapm_force_enable_pin(dapm, pin); - } -} - static void nau8824_eject_jack(struct nau8824 *nau8824) { struct snd_soc_dapm_context *dapm = nau8824->dapm; @@ -866,8 +836,8 @@ static void nau8824_eject_jack(struct nau8824 *nau8824) /* Clear all interruption status */ nau8824_int_status_clear_all(regmap); - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); /* Enable the insertion interruption, disable the ejection @@ -897,8 +867,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - nau8824_dapm_enable_pin(nau8824, "MICBIAS"); - nau8824_dapm_enable_pin(nau8824, "SAR"); + snd_soc_dapm_enable_pin(dapm. "MICBIAS"); + snd_soc_dapm_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); @@ -909,8 +879,8 @@ static void nau8824_jdet_work(struct work_struct *work) if (adc_value < HEADSET_SARADC_THD) { event |= SND_JACK_HEADPHONE; - nau8824_dapm_disable_pin(nau8824, "SAR"); - nau8824_dapm_disable_pin(nau8824, "MICBIAS"); + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); snd_soc_dapm_sync(dapm); } else { event |= SND_JACK_HEADSET; -- 2.20.1

1 1

[PATCH] ALSA: pcm - fix mmap capability check for the snd-dummy driver
by Jaroslav Kysela 30 Jul '21

30 Jul '21

The snd-dummy driver (fake_buffer configuration) uses the ops->page callback for the mmap operations. Allow mmap for this case, too. Cc: <stable(a)vger.kernel.org> Fixes: c4824ae7db41 ("ALSA: pcm: Fix mmap capability check") Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 6a2971a7e6a1..09c0e2a6489c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,7 +246,7 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap) + if (substream->ops->mmap || substream->ops->page) return true; switch (substream->dma_buffer.dev.type) { -- 2.31.1

2 1

[PATCH v4 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 30 Jul '21

30 Jul '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v4 changes: Fixed spacing for bracketed declarations Fixed comment style in headers Moved private struct from include/sound to local cs35l41.h Reordered probe operations: 1) hardware errata 2) INT pin config 3) INT mask config 4) IRQ registration Moved set_pdata from component_probe to probe Set initial state of AMP_EN bit and corresponding DAPM widget to off Rename invert-pcm DT prop to invert-class-d Change return code handling in IRQ Use constants for regulator num supplies Error message for no SPI dev in spi_otp_setup Restore 0 initializers in IRQ per Charles's comments David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 398 ++++ include/sound/cs35l41.h | 60 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 116 ++ sound/soc/codecs/cs35l41-spi.c | 145 ++ sound/soc/codecs/cs35l41-tables.c | 612 ++++++ sound/soc/codecs/cs35l41.c | 1775 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 777 ++++++++ 9 files changed, 3899 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

3 8

[PATCH v1] ASoC: Intel: kbl_da7219_max98357a: fix drv_name
by Lukasz Majczak 30 Jul '21

30 Jul '21

platform_id for kbl_da7219_max98357a was shrunk for kbl_da7219_mx98357a, but the drv_name was changed for kbl_da7219_max98373. Tested on a Pixelbook (Atlas). Fixes: 94efd726b947 ("ASoC: Intel: kbl_da7219_max98357a: shrink platform_id below 20 characters") Cc: <stable(a)vger.kernel.org> # 5.4+ Reported-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index ba5ff468c265..8cab91a00b1a 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -87,7 +87,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_max98357a", + .drv_name = "kbl_da7219_mx98357a", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98357_codecs, @@ -113,7 +113,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { }, { .id = "DLGS7219", - .drv_name = "kbl_da7219_mx98373", + .drv_name = "kbl_da7219_max98373", .fw_filename = "intel/dsp_fw_kbl.bin", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &kbl_7219_98373_codecs, -- 2.32.0.554.ge1b32706d8-goog

3 4

5.14 regression, Intel SST (BYT) audio no longer works
by Hans de Goede 30 Jul '21

30 Jul '21

Hi All, I just noticed and I won't have time to dig any deeper for approx. the coming 10 days. Still I wanted to report this in case anyone has any ideas. Or maybe you can reproduce and look for a fix ? After building + installing 5.14-rc1 on a HP Elitepad 1000 G2 (Bay Trail) with a RT5642 codec, I noticed that sound over the speakers and over the docks line-out jack no longer works. Downgrading to 5.13 (without any other changes) fixes this, so this seems to be a regression with 5.14. I've no been able to make time to test this on other BYT/CHT hardware, but I suspect that other models will be affected too. Regards, Hans

2 3

[PATCH v2 00/16] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 30 Jul '21

30 Jul '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 | | q6afe | | | | 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. Sample WIP ASoC graphs are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ Thanks, srini Changes since RFC: - moved GPR support into APR driver to avoid code duplication. - Fixed various dt-bindings-check warnings and errors. - Moved include/dt-bindings headers to dt-bindings patch. - added PGA widget support - Fixed few ordering issues while testing with ASoC Topology 2.0 - Removed duplicate defines in various headers Srinivas Kandagatla (16): soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: make code more reuseable soc: qcom: apr: Add GPR support ASoC: qcom: dt-bindings: add bindings Audio Processing manager ASoC: qcom: audioreach: add basic pkt alloc support ASoC: qcom: audioreach: add q6apm support ASoC: qcom: audioreach: add module configuration command helpers ASoC: qcom: audioreach: add topology support ASoC: qcom: audioreach: add q6apm-dai support ASoC: qcom: audioreach: add bedai support ASoC: qcom: dt-bindings: add bindings for Proxy Resource Manager ASoC: qcom: audioreach: add q6prm support ASoC: qcom: dt-bindings: add audioreach soundcard compatibles ASoC: qcom: audioreach: add volume ctrl module support ASoC: qcom: audioreach: topology add dapm pga support ASoC: qcom: sm8250: Add audioreach support .../bindings/soc/qcom/qcom,gpr.yaml | 83 ++ .../devicetree/bindings/sound/qcom,q6apm.yaml | 87 ++ .../devicetree/bindings/sound/qcom,q6prm.yaml | 43 + .../bindings/sound/qcom,sm8250.yaml | 43 + drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 270 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6apm.h | 8 + include/linux/soc/qcom/apr.h | 69 +- include/uapi/sound/snd_ar_tokens.h | 198 +++ sound/soc/qcom/Kconfig | 19 + sound/soc/qcom/Makefile | 1 + sound/soc/qcom/audioreach/Makefile | 12 + sound/soc/qcom/audioreach/audioreach.c | 1162 +++++++++++++++++ sound/soc/qcom/audioreach/audioreach.h | 676 ++++++++++ sound/soc/qcom/audioreach/q6apm-bedai.c | 323 +++++ sound/soc/qcom/audioreach/q6apm-dai.c | 494 +++++++ sound/soc/qcom/audioreach/q6apm.c | 963 ++++++++++++++ sound/soc/qcom/audioreach/q6apm.h | 174 +++ sound/soc/qcom/audioreach/q6prm.c | 414 ++++++ sound/soc/qcom/audioreach/topology.c | 1075 +++++++++++++++ sound/soc/qcom/sm8250.c | 144 +- 22 files changed, 6236 insertions(+), 48 deletions(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,gpr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6prm.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6apm.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/audioreach/Makefile create mode 100644 sound/soc/qcom/audioreach/audioreach.c create mode 100644 sound/soc/qcom/audioreach/audioreach.h create mode 100644 sound/soc/qcom/audioreach/q6apm-bedai.c create mode 100644 sound/soc/qcom/audioreach/q6apm-dai.c create mode 100644 sound/soc/qcom/audioreach/q6apm.c create mode 100644 sound/soc/qcom/audioreach/q6apm.h create mode 100644 sound/soc/qcom/audioreach/q6prm.c create mode 100644 sound/soc/qcom/audioreach/topology.c -- 2.21.0

5 39

[PATCH] ALSA: pcm: Fix mmap capability check
by Takashi Iwai 30 Jul '21

30 Jul '21

The hw_support_mmap() doesn't cover all memory allocation types and might use a wrong device pointer for checking the capability. Check the all memory allocation types more completely. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c88c4316c417..6919d2943b9d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -246,12 +246,18 @@ static bool hw_support_mmap(struct snd_pcm_substream *substream) if (!(substream->runtime->hw.info & SNDRV_PCM_INFO_MMAP)) return false; - if (substream->ops->mmap || - (substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV && - substream->dma_buffer.dev.type != SNDRV_DMA_TYPE_DEV_UC)) + if (substream->ops->mmap) return true; - return dma_can_mmap(substream->dma_buffer.dev.dev); + switch (substream->dma_buffer.dev.type) { + case SNDRV_DMA_TYPE_UNKNOWN: + return false; + case SNDRV_DMA_TYPE_CONTINUOUS: + case SNDRV_DMA_TYPE_VMALLOC: + return true; + default: + return dma_can_mmap(substream->dma_buffer.dev.dev); + } } static int constrain_mask_params(struct snd_pcm_substream *substream, -- 2.26.2

2 2

[PATCH 1/3] ASoC: cs42l42: Correct definition of ADC Volume control
by Richard Fitzgerald 29 Jul '21

29 Jul '21

The ADC volume is a signed 8-bit number with range -97 to +12, with -97 being mute. Use a SOC_SINGLE_S8_TLV() to define this and fix the DECLARE_TLV_DB_SCALE() to have the correct start and mute flag. Fixes: 2c394ca79604 ("ASoC: Add support for CS42L42 codec") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index eff013f295be..2fd20511f246 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -405,7 +405,7 @@ static const struct regmap_config cs42l42_regmap = { .use_single_write = true, }; -static DECLARE_TLV_DB_SCALE(adc_tlv, -9600, 100, false); +static DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 100, true); static DECLARE_TLV_DB_SCALE(mixer_tlv, -6300, 100, true); static const char * const cs42l42_hpf_freq_text[] = { @@ -444,8 +444,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_ADC_INV_SHIFT, true, false), SOC_SINGLE("ADC Boost Switch", CS42L42_ADC_CTL, CS42L42_ADC_DIG_BOOST_SHIFT, true, false), - SOC_SINGLE_SX_TLV("ADC Volume", CS42L42_ADC_VOLUME, - CS42L42_ADC_VOL_SHIFT, 0xA0, 0x6C, adc_tlv), + SOC_SINGLE_S8_TLV("ADC Volume", CS42L42_ADC_VOLUME, -97, 12, adc_tlv), SOC_SINGLE("ADC WNF Switch", CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_EN_SHIFT, true, false), SOC_SINGLE("ADC HPF Switch", CS42L42_ADC_WNF_HPF_CTL, -- 2.11.0

2 4

[PATCH] ALSA: usb-audio: Avoid unnecessary or invalid connector selection at resume
by Takashi Iwai 29 Jul '21

29 Jul '21

The recent fix for the resume on Lenovo machines seems causing a regression on others. It's because the change always triggers the connector selection no matter which widget node type is. This patch addresses the regression by setting the resume callback selectively only for the connector widget. Fixes: 44609fc01f28 ("ALSA: usb-audio: Check connector value on resume") Cc: <stable(a)vger.kernel.org> BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213897 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer.c | 35 ++++++++++++++++++++--------------- 1 file changed, 20 insertions(+), 15 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index f4cdaf1ba44a..9b713b4a5ec4 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1816,6 +1816,15 @@ static void get_connector_control_name(struct usb_mixer_interface *mixer, strlcat(name, " - Output Jack", name_size); } +/* get connector value to "wake up" the USB audio */ +static int connector_mixer_resume(struct usb_mixer_elem_list *list) +{ + struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); + + get_connector_value(cval, NULL, NULL); + return 0; +} + /* Build a mixer control for a UAC connector control (jack-detect) */ static void build_connector_control(struct usb_mixer_interface *mixer, const struct usbmix_name_map *imap, @@ -1833,6 +1842,10 @@ static void build_connector_control(struct usb_mixer_interface *mixer, if (!cval) return; snd_usb_mixer_elem_init_std(&cval->head, mixer, term->id); + + /* set up a specific resume callback */ + cval->head.resume = connector_mixer_resume; + /* * UAC2: The first byte from reading the UAC2_TE_CONNECTOR control returns the * number of channels connected. @@ -3642,23 +3655,15 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_resume(struct usb_mixer_elem_list *list) -{ - struct usb_mixer_elem_info *cval = mixer_elem_list_to_info(list); - - /* get connector value to "wake up" the USB audio */ - if (cval->val_type == USB_MIXER_BOOLEAN && cval->channels == 1) - get_connector_value(cval, NULL, NULL); - - return 0; -} - static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) { - int err = default_mixer_resume(list); + int err; - if (err < 0) - return err; + if (list->resume) { + err = list->resume(list); + if (err < 0) + return err; + } return restore_mixer_value(list); } @@ -3697,7 +3702,7 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list, list->id = unitid; list->dump = snd_usb_mixer_dump_cval; #ifdef CONFIG_PM - list->resume = default_mixer_resume; + list->resume = NULL; list->reset_resume = default_mixer_reset_resume; #endif } -- 2.26.2

1 0

[PATCH] ASoC: samsung: Constify static snd_soc_ops
by Rikard Falkeborn 29 Jul '21

29 Jul '21

These are only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make them const to allow the compiler to put them in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/samsung/aries_wm8994.c | 2 +- sound/soc/samsung/arndale.c | 4 ++-- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/midas_wm1811.c | 2 +- sound/soc/samsung/neo1973_wm8753.c | 4 ++-- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/smartq_wm8987.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 6 +++--- sound/soc/samsung/tobermory.c | 2 +- 13 files changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/soc/samsung/aries_wm8994.c b/sound/soc/samsung/aries_wm8994.c index 0ac5956ba270..313ab650f8d9 100644 --- a/sound/soc/samsung/aries_wm8994.c +++ b/sound/soc/samsung/aries_wm8994.c @@ -310,7 +310,7 @@ static int aries_hw_free(struct snd_pcm_substream *substream) /* * Main DAI operations */ -static struct snd_soc_ops aries_ops = { +static const struct snd_soc_ops aries_ops = { .hw_params = aries_hw_params, .hw_free = aries_hw_free, }; diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index 28587375813a..606ac5e33a8e 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -48,7 +48,7 @@ static int arndale_rt5631_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops arndale_rt5631_ops = { +static const struct snd_soc_ops arndale_rt5631_ops = { .hw_params = arndale_rt5631_hw_params, }; @@ -80,7 +80,7 @@ static int arndale_wm1811_hw_params(struct snd_pcm_substream *substream, rclk + 1, SND_SOC_CLOCK_IN); } -static struct snd_soc_ops arndale_wm1811_ops = { +static const struct snd_soc_ops arndale_wm1811_ops = { .hw_params = arndale_wm1811_hw_params, }; diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c index 8aa78ff640f5..c994e67d1eaf 100644 --- a/sound/soc/samsung/h1940_uda1380.c +++ b/sound/soc/samsung/h1940_uda1380.c @@ -112,7 +112,7 @@ static int h1940_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops h1940_ops = { +static const struct snd_soc_ops h1940_ops = { .startup = h1940_startup, .hw_params = h1940_hw_params, }; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index a1ff1400857e..390f2dd735ad 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -130,7 +130,7 @@ static int littlemill_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops littlemill_ops = { +static const struct snd_soc_ops littlemill_ops = { .hw_params = littlemill_hw_params, }; diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 1f9a553edf19..a2019535a0b1 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -129,7 +129,7 @@ static int midas_aif1_hw_params(struct snd_pcm_substream *substream, return midas_start_fll1(rtd, pll_out); } -static struct snd_soc_ops midas_aif1_ops = { +static const struct snd_soc_ops midas_aif1_ops = { .hw_params = midas_aif1_hw_params, }; diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index 9266070e0181..c98b68567a89 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -106,7 +106,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) /* * Neo1973 WM8753 HiFi DAI opserations. */ -static struct snd_soc_ops neo1973_hifi_ops = { +static const struct snd_soc_ops neo1973_hifi_ops = { .hw_params = neo1973_hifi_hw_params, .hw_free = neo1973_hifi_hw_free, }; @@ -158,7 +158,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0); } -static struct snd_soc_ops neo1973_voice_ops = { +static const struct snd_soc_ops neo1973_voice_ops = { .hw_params = neo1973_voice_hw_params, .hw_free = neo1973_voice_hw_free, }; diff --git a/sound/soc/samsung/rx1950_uda1380.c b/sound/soc/samsung/rx1950_uda1380.c index 400a7f77c711..6ea1c8cc9167 100644 --- a/sound/soc/samsung/rx1950_uda1380.c +++ b/sound/soc/samsung/rx1950_uda1380.c @@ -62,7 +62,7 @@ static struct snd_soc_jack_gpio hp_jack_gpios[] = { }, }; -static struct snd_soc_ops rx1950_ops = { +static const struct snd_soc_ops rx1950_ops = { .startup = rx1950_startup, .hw_params = rx1950_hw_params, }; diff --git a/sound/soc/samsung/smartq_wm8987.c b/sound/soc/samsung/smartq_wm8987.c index c95629becbc3..cee39ad16667 100644 --- a/sound/soc/samsung/smartq_wm8987.c +++ b/sound/soc/samsung/smartq_wm8987.c @@ -70,7 +70,7 @@ static int smartq_hifi_hw_params(struct snd_pcm_substream *substream, /* * SmartQ WM8987 HiFi DAI operations. */ -static struct snd_soc_ops smartq_hifi_ops = { +static const struct snd_soc_ops smartq_hifi_ops = { .hw_params = smartq_hifi_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8580.c b/sound/soc/samsung/smdk_wm8580.c index ed753a2f202e..78703d095a6f 100644 --- a/sound/soc/samsung/smdk_wm8580.c +++ b/sound/soc/samsung/smdk_wm8580.c @@ -86,7 +86,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8580 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 39a7a449f554..7661b637946d 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -73,7 +73,7 @@ static int smdk_hw_params(struct snd_pcm_substream *substream, /* * SMDK WM8994 DAI operations. */ -static struct snd_soc_ops smdk_ops = { +static const struct snd_soc_ops smdk_ops = { .hw_params = smdk_hw_params, }; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index a01640576f71..029448f5bedb 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -85,7 +85,7 @@ static int smdk_wm8994_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops smdk_wm8994_pcm_ops = { +static const struct snd_soc_ops smdk_wm8994_pcm_ops = { .hw_params = smdk_wm8994_pcm_hw_params, }; diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 84c2c63d5a87..a2c77e6defec 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -126,7 +126,7 @@ static int tm2_aif1_hw_params(struct snd_pcm_substream *substream, return tm2_start_sysclk(rtd->card); } -static struct snd_soc_ops tm2_aif1_ops = { +static const struct snd_soc_ops tm2_aif1_ops = { .hw_params = tm2_aif1_hw_params, }; @@ -200,7 +200,7 @@ static int tm2_aif2_hw_free(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops tm2_aif2_ops = { +static const struct snd_soc_ops tm2_aif2_ops = { .hw_params = tm2_aif2_hw_params, .hw_free = tm2_aif2_hw_free, }; @@ -254,7 +254,7 @@ static int tm2_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tm2_hdmi_ops = { +static const struct snd_soc_ops tm2_hdmi_ops = { .hw_params = tm2_hdmi_hw_params, }; diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index c962d2c2a7f7..15223d860cb7 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -105,7 +105,7 @@ static int tobermory_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tobermory_ops = { +static const struct snd_soc_ops tobermory_ops = { .hw_params = tobermory_hw_params, }; -- 2.32.0

3 2

[PATCH 1/2] cx20442: tty_ldisc_ops::write_wakeup is optional
by Jiri Slaby 29 Jul '21

29 Jul '21

TTY layer does nothing if tty_ldisc_ops::write_wakeup is NULL, so there is no need to implement an empty one in cx20442. Drop it. Signed-off-by: Jiri Slaby <jslaby(a)suse.cz> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/cx20442.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index ec8d6e74b467..65c6b92d7b5f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -279,11 +279,6 @@ static void v253_receive(struct tty_struct *tty, const unsigned char *cp, } } -/* Line discipline .write_wakeup() */ -static void v253_wakeup(struct tty_struct *tty) -{ -} - struct tty_ldisc_ops v253_ops = { .name = "cx20442", .owner = THIS_MODULE, @@ -291,7 +286,6 @@ struct tty_ldisc_ops v253_ops = { .close = v253_close, .hangup = v253_hangup, .receive_buf = v253_receive, - .write_wakeup = v253_wakeup, }; EXPORT_SYMBOL_GPL(v253_ops); -- 2.32.0

2 3

[PATCH 0/7] ASoC: soc-core: cleanup cppcheck warning
by Kuninori Morimoto 29 Jul '21

29 Jul '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-core. Kuninori Morimoto (7): ASoC: soc-core: cleanup cppcheck warning at snd_soc_set_dmi_name() ASoC: soc-core: cleanup cppcheck warning at snd_soc_get_dai_name() ASoC: soc-core: cleanup cppcheck warning at snd_soc_daifmt_parse_format() ASoC: soc-core: cleanup cppcheck warning at snd_soc_unregister_component() ASoC: soc-core: cleanup cppcheck warning at snd_soc_add_controls() ASoC: soc-core: cleanup cppcheck warning at snd_soc_of_parse_audio_simple_widgets() ASoC: soc-core: cleanup cppcheck warning at snd_soc_of_parse_audio_routing() sound/soc/soc-core.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) -- 2.25.1

2 8

[PATCH v2 00/27] ALSA: hda/cirrus: Split generic cirrus HDA codecs and CS8490 bridge into separate modules.
by Vitaly Rodionov 29 Jul '21

29 Jul '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 | 1245 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.h | 369 ++++++++ 6 files changed, 2186 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 00/15] ALSA: usb-audio: Introduce and convert to quirk_flags
by Takashi Iwai 29 Jul '21

29 Jul '21

As more and more device-specific workarounds came up and gathered in various places, it becomes harder to manage. Now it's time to clean up and collect workarounds more consistently and make them more easily applicable. This patch set introduces a new bitmap flags, quirk_flags, to contain various device-specific quirks. Those are separate one from the quirks in quirks-table.h; the quirks-table.h entries are for more intrusive stuff that needs the descriptor override, while the new quirk_flags is for easier ones that are tied with the vendor:product IDs. Takashi === Takashi Iwai (15): ALSA: usb-audio: Introduce quirk_flags field ALSA: usb-audio: Move media-controller API quirk into quirk_flags ALSA: usb-audio: Move txfr_quirk handling to quirk_flags ALSA: usb-audio: Move tx_length quirk handling to quirk_flags ALSA: usb-audio: Move playback_first flag into quirk_flags ALSA: usb-audio: Move clock setup quirk into quirk_flags ALSA: usb-audio: Move ITF-USB DSD quirk handling into quirk_flags ALSA: usb-audio: Move control message delay quirk into quirk_flags ALSA: usb-audio: Move interface setup delay into quirk_flags ALSA: usb-audio: Move rate validation quirk into quirk_flags ALSA: usb-audio: Move autosuspend quirk into quirk_flags ALSA: usb-audio: Move ignore_ctl_error check into quirk_flags ALSA: usb-audio: Move generic DSD raw detection into quirk_flags ALSA: usb-audio: Add quirk_flags module option ALSA: doc: Add the description of quirk_flags option for snd-usb-audio Documentation/sound/alsa-configuration.rst | 26 +- sound/usb/card.c | 42 ++- sound/usb/clock.c | 15 +- sound/usb/endpoint.c | 14 +- sound/usb/format.c | 6 +- sound/usb/implicit.c | 6 +- sound/usb/mixer.c | 6 +- sound/usb/mixer.h | 3 +- sound/usb/mixer_maps.c | 22 -- sound/usb/quirks-table.h | 70 +--- sound/usb/quirks.c | 359 +++++++++++---------- sound/usb/quirks.h | 5 +- sound/usb/stream.c | 4 +- sound/usb/usbaudio.h | 65 +++- 14 files changed, 341 insertions(+), 302 deletions(-) -- 2.26.2

1 15

[PATCH] ASoC: dapm: Revert "use component prefix when checking widget names"
by Richard Fitzgerald 29 Jul '21

29 Jul '21

This reverts commit ae4fc532244b ("ASoC: dapm: use component prefix when checking widget names") That commit breaks all users of the snd_soc_component_*_pin() functions because it results in the prefix being added twice. It also breaks code that correctly uses the snd_soc_dapm_*_pin() functions. Use the snd_soc_component_*_pin() functions if you want the component prefix to be prepended automatically. Use the raw snd_soc_dapm_*_pin() functions if the caller has the full name that should be matched exactly. Fixes: commit ae4fc532244b ("ASoC: dapm: use component prefix when checking widget names") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 91bf939d5233..1369a3fea911 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2528,20 +2528,9 @@ static struct snd_soc_dapm_widget *dapm_find_widget( { struct snd_soc_dapm_widget *w; struct snd_soc_dapm_widget *fallback = NULL; - char prefixed_pin[80]; - const char *pin_name; - const char *prefix = soc_dapm_prefix(dapm); - - if (prefix) { - snprintf(prefixed_pin, sizeof(prefixed_pin), "%s %s", - prefix, pin); - pin_name = prefixed_pin; - } else { - pin_name = pin; - } for_each_card_widgets(dapm->card, w) { - if (!strcmp(w->name, pin_name)) { + if (!strcmp(w->name, pin)) { if (w->dapm == dapm) return w; else -- 2.11.0

2 7

[PATCH] ASoC: wm_adsp: Let soc_cleanup_component_debugfs remove debugfs
by Lucas Tanure 29 Jul '21

29 Jul '21

soc_cleanup_component_debugfs will debugfs_remove_recursive the component->debugfs_root, so adsp doesn't need to also remove the same entry. By doing that adsp also creates a race with core component, which causes a NULL pointer dereference Signed-off-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b395df1eb72d..bbe27ab3b1fc 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -747,7 +747,6 @@ static void wm_adsp2_init_debugfs(struct wm_adsp *dsp, static void wm_adsp2_cleanup_debugfs(struct wm_adsp *dsp) { wm_adsp_debugfs_clear(dsp); - debugfs_remove_recursive(dsp->debugfs_root); } #else static inline void wm_adsp2_init_debugfs(struct wm_adsp *dsp, -- 2.32.0

2 1

[PATCH] ASoC: component: Remove misplaced prefix handling in pin control functions
by Mark Brown 29 Jul '21

29 Jul '21

When the component level pin control functions were added they for some no longer obvious reason handled adding prefixing of widget names. This meant that when the lack of prefix handling in the DAPM level pin operations was fixed by ae4fc532244b3bb4d (ASoC: dapm: use component prefix when checking widget names) the one device using the component level API ended up with the prefix being applied twice, causing all lookups to fail. Fix this by removing the redundant prefixing from the component code, which has the nice side effect of also making that code much simpler. Reported-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-component.c | 63 +++++++++++++++++---------------------- 1 file changed, 27 insertions(+), 36 deletions(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index 3a5e84e16a87..c8dfd0de30e4 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -148,86 +148,75 @@ int snd_soc_component_set_bias_level(struct snd_soc_component *component, return soc_component_ret(component, ret); } -static int soc_component_pin(struct snd_soc_component *component, - const char *pin, - int (*pin_func)(struct snd_soc_dapm_context *dapm, - const char *pin)) -{ - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); - char *full_name; - int ret; - - if (!component->name_prefix) { - ret = pin_func(dapm, pin); - goto end; - } - - full_name = kasprintf(GFP_KERNEL, "%s %s", component->name_prefix, pin); - if (!full_name) { - ret = -ENOMEM; - goto end; - } - - ret = pin_func(dapm, full_name); - kfree(full_name); -end: - return soc_component_ret(component, ret); -} - int snd_soc_component_enable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_enable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_enable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin); int snd_soc_component_enable_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_enable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_enable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_enable_pin_unlocked); int snd_soc_component_disable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_disable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_disable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin); int snd_soc_component_disable_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_disable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_disable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_disable_pin_unlocked); int snd_soc_component_nc_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_nc_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_nc_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin); int snd_soc_component_nc_pin_unlocked(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_nc_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_nc_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_nc_pin_unlocked); int snd_soc_component_get_pin_status(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_get_pin_status); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_get_pin_status(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_get_pin_status); int snd_soc_component_force_enable_pin(struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_force_enable_pin(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin); @@ -235,7 +224,9 @@ int snd_soc_component_force_enable_pin_unlocked( struct snd_soc_component *component, const char *pin) { - return soc_component_pin(component, pin, snd_soc_dapm_force_enable_pin_unlocked); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + return snd_soc_dapm_force_enable_pin_unlocked(dapm, pin); } EXPORT_SYMBOL_GPL(snd_soc_component_force_enable_pin_unlocked); -- 2.20.1

2 2

[PATCH] ASoC: Intel: Fix spelling contraction "cant" -> "can't"
by Colin King 29 Jul '21

29 Jul '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/intel/atom/sst/sst_ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_ipc.c b/sound/soc/intel/atom/sst/sst_ipc.c index a8a9aa0057d3..4e8382097e61 100644 --- a/sound/soc/intel/atom/sst/sst_ipc.c +++ b/sound/soc/intel/atom/sst/sst_ipc.c @@ -128,7 +128,7 @@ int sst_post_message_mrfld(struct intel_sst_drv *sst_drv_ctx, while (header.p.header_high.part.busy) { if (loop_count > 25) { dev_err(sst_drv_ctx->dev, - "sst: Busy wait failed, cant send this msg\n"); + "sst: Busy wait failed, can't send this msg\n"); retval = -EBUSY; goto out; } -- 2.31.1

3 2

[PATCH] MAINTAINERS: Add sound devicetree bindings for Wolfson Micro devices
by Richard Fitzgerald 29 Jul '21

29 Jul '21

Include all wm* sound bindings in the section for Wolfson Micro drivers. This section already includes the actual driver source files. Also update the existing entry to match all wlf,* sound bindings. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 19135a9d778e..d8282bb8a7e1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -20009,7 +20009,8 @@ F: Documentation/devicetree/bindings/extcon/wlf,arizona.yaml F: Documentation/devicetree/bindings/mfd/wlf,arizona.yaml F: Documentation/devicetree/bindings/mfd/wm831x.txt F: Documentation/devicetree/bindings/regulator/wlf,arizona.yaml -F: Documentation/devicetree/bindings/sound/wlf,arizona.yaml +F: Documentation/devicetree/bindings/sound/wlf,*.yaml +F: Documentation/devicetree/bindings/sound/wm* F: Documentation/hwmon/wm83??.rst F: arch/arm/mach-s3c/mach-crag6410* F: drivers/clk/clk-wm83*.c -- 2.11.0

2 1

[PATCH] ASoC: bcm: cygnus-pcm: Fix unused assignment about 'rc'
by Tang Bin 29 Jul '21

29 Jul '21

Delete unused initialized value of 'rc', because it will be assigned by the function devm_request_irq(). Signed-off-by: Zhang Shengju <zhangshengju(a)cmss.chinamobile.com> Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/bcm/cygnus-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c index 7ad07239f..039365f76 100644 --- a/sound/soc/bcm/cygnus-pcm.c +++ b/sound/soc/bcm/cygnus-pcm.c @@ -830,7 +830,7 @@ static struct snd_soc_component_driver cygnus_soc_platform = { int cygnus_soc_platform_register(struct device *dev, struct cygnus_audio *cygaud) { - int rc = 0; + int rc; dev_dbg(dev, "%s Enter\n", __func__); -- 2.20.1.windows.1

2 1

[PATCH 0/7] ALSA/ASoC/SOF/SoundWire: fix Kconfig issues
by Pierre-Louis Bossart 28 Jul '21

28 Jul '21

In January, Intel kbuild bot and Arnd Bergmann reported multiple issues with randconfig. This patchset builds on Arnd's suggestions to a) expose ACPI and PCI devices in separate modules, while sof-acpi-dev and sof-pci-dev become helpers. This will result in minor changes required for developers/testers, i.e. modprobe snd-sof-pci will no longer result in a probe. The SOF CI was already updated to deal with this module dependency change and introduction of new modules. b) Fix SOF/SoundWire/DSP_config dependencies by moving the code required to detect SoundWire presence in ACPI tables to sound/hda. Integration note: This patchset touches directories maintained by Vinod, Takashi and Mark in separate trees, and will impact additional changes to use the auxiliary bus in drivers/soundwire/. I can think of two options, both of which are fine: 1. Mark merges the patches with Vinod and Takashi Acked-by tags, then Mark provides an immutable tag to Vinod. 2. Vinod merges the patches with Mark and Takashi Acked-by tags, then Vinod provides an immutable tag to Mark Acknowledgements: Thanks to Arnd for suggesting fixes and testing these patches with more randconfigs. Thanks to Vinod Koul for his feedback on the move of this common helper to sound/hda/ Arnd Bergmann (1): ASoC: SOF: ACPI: avoid reverse module dependency Pierre-Louis Bossart (6): ASoC: soc-acpi: allow for partial match in parent name ASoC: SOF: pci: split PCI into different drivers ASoC: SOF: pci: move DSP_CONFIG use to platform-specific drivers ASoC: SOF: Intel: SoundWire: simplify Kconfig ALSA: hda: move Intel SoundWire ACPI scan to dedicated module ALSA: hda: intel-sdw-acpi: add missing include files drivers/soundwire/intel.h | 2 - drivers/soundwire/intel_init.c | 158 ------------- include/linux/soundwire/sdw_intel.h | 2 + include/sound/soc-acpi.h | 2 +- sound/hda/Kconfig | 4 + sound/hda/Makefile | 3 + sound/hda/intel-dsp-config.c | 2 +- sound/hda/intel-sdw-acpi.c | 179 +++++++++++++++ sound/soc/sof/Kconfig | 15 +- sound/soc/sof/Makefile | 4 +- sound/soc/sof/intel/Kconfig | 254 +++++++-------------- sound/soc/sof/intel/Makefile | 20 +- sound/soc/sof/intel/bdw.c | 67 +++++- sound/soc/sof/intel/byt.c | 106 ++++++++- sound/soc/sof/intel/hda.c | 18 ++ sound/soc/sof/intel/hda.h | 3 + sound/soc/sof/intel/pci-apl.c | 81 +++++++ sound/soc/sof/intel/pci-cnl.c | 104 +++++++++ sound/soc/sof/intel/pci-icl.c | 84 +++++++ sound/soc/sof/intel/pci-tgl.c | 121 ++++++++++ sound/soc/sof/intel/pci-tng.c | 70 ++++++ sound/soc/sof/intel/shim.h | 6 - sound/soc/sof/sof-acpi-dev.c | 130 +---------- sound/soc/sof/sof-acpi-dev.h | 16 ++ sound/soc/sof/sof-pci-dev.c | 340 +--------------------------- sound/soc/sof/sof-pci-dev.h | 17 ++ 26 files changed, 998 insertions(+), 810 deletions(-) create mode 100644 sound/hda/intel-sdw-acpi.c create mode 100644 sound/soc/sof/intel/pci-apl.c create mode 100644 sound/soc/sof/intel/pci-cnl.c create mode 100644 sound/soc/sof/intel/pci-icl.c create mode 100644 sound/soc/sof/intel/pci-tgl.c create mode 100644 sound/soc/sof/intel/pci-tng.c create mode 100644 sound/soc/sof/sof-acpi-dev.h create mode 100644 sound/soc/sof/sof-pci-dev.h -- 2.25.1

5 20

PCM prepare callback
by Muni Sekhar 28 Jul '21

28 Jul '21

Hi all, Is there any document which explains the sequence of snd_pcm_ops callbacks called during playback and capture? What driver is supposed to return from the PCM prepare callback if the sound card is not ready for playback\capture? -- Thanks, Sekhar

1 0

Copy hw out to loop problem when recording started before playing
by Anton Nikolaev 28 Jul '21

28 Jul '21

Hello everyone. Please help. Do not understand what is wrong Chains is : Plug asym1 - Out= Plug with multi (Copy hardware dmix output to dmix loopout) IN= Dsnoop with hardware dev Plug asym2 - Out= plug dmix loop IN= plug dsnoop loop Work good if playing to plug asym1 started before recording from asym2. Not working play to asym1 if recording from asym2 started before playing to asym1. /Etc/asound.conf is: pcm.hwcard0 { type hw; card 0; device 0; } pcm.!default { type plug; slave.pcm { type asym; playback.pcm "devhwcopy"; capture.pcm "devhwin"; } } pcm.devhwcopy { type plug; route_policy "duplicate"; slave.pcm { type multi; slaves { a { channels 2; pcm "devhwout"; } b { channels 2; pcm "devloopout"; } } bindings { 0 { slave a channel 0 } 1 { slave a channel 1 } 2 { slave b channel 0 } 3 { slave b channel 1 } } } ttable [ [ 1 0 1 0 ] [ 0 1 0 1 ] ] } pcm.devhwout { type dmix; ipc_key 1024001; ipc_perm 0666; hw_ptr_alignment roundup; slave { pcm "hwcard0"; channels 2; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devhwin { type dsnoop; ipc_key 1024002; ipc_perm 0666; slave { pcm "hwcard0"; channels 2; rate 44100; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devloopout { type dmix; ipc_key 1024011; ipc_perm 0666; slave { pcm "hw:LOOPCARD,0,0"; channels 2; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devloopin { type dsnoop; ipc_key 1024012; ipc_perm 0666; slave { pcm "hw:LOOPCARD,1,0"; channels 2; rate 44100; period_size 1024; buffer_size 8192; } bindings { 0 0 1 1 } } pcm.devloop { type plug; slave.pcm { type asym; playback.pcm "devloopout"; capture.pcm "devloopin"; } } (English is not native for me, sorry)

1 0

[PATCH 0/5] ASoC: soc-pcm: cleanup cppcheck warning
by Kuninori Morimoto 28 Jul '21

28 Jul '21

Hi Mark Now I'm posting audio-graph-card2 patch-set, but it seems it needs longer discussion. Thus I want to post more easy patch first, and reduce my local patches. These are cppcheck warning cleanup patches for soc-pcm. Kuninori Morimoto (5): ASoC: soc-pcm: cleanup cppcheck warning at soc_pcm_apply_msb() ASoC: soc-pcm: cleanup cppcheck warning at soc_pcm_components_close() ASoC: soc-pcm: cleanup cppcheck warning at soc_get_playback_capture() ASoC: soc-pcm: cleanup cppcheck warning at dpcm_be_is_active() ASoC: soc-pcm: cleanup cppcheck warning at dpcm_runtime_setup_be_chan() sound/soc/soc-pcm.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) -- 2.25.1 Thank you for your help !! Best regards --- Kuninori Morimoto

3 8

Pulseaudio dropping GPU audio
by Liam Hupfer 27 Jul '21

27 Jul '21

Hello, I’ve set up a Linux VM where I’m passing through an RX 580. I’m using the GPU audio and my monitor’s audio jack. For some reason the Linux VM keeps losing track of the audio output. It always shows up under `lspci' under ID `06:00.0' using the `snd_hda_intel' driver, but `pulseaudio' (I also tried `pipewire', same issue) keep losing it. Most times I can restart `pulseaudio' and it will find the device again; occasionally, it takes multiple restarts. I checked the Arch wiki and I found something[¹] on audio over HDMI (I’m using DisplayPort, if that matters). I checked the device and it did report `Enable+'. Even so, I added the kernel parameter `snd_hda_intel.enable_msi=1', but nothing changed. Here are some `journalctl --user -u pulseaudio' logs: ┌──── │ 20:37:31 host pulseaudio[2373]: ALSA woke us up to write new data to the device, but there was actually nothing to write. │ 20:37:31 host pulseaudio[2373]: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. │ 20:37:31 host pulseaudio[2373]: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. └──── Anyone have any suggestions for troubleshooting? `pulseaudio'’s logs suggested it might have to do with ALSA, so I asked for help here. Thank you! —Liam [¹] <https://wiki.archlinux.org/title/PCI_passthrough_via_OVMF#Slowed_down_audio…>

2 1

alsaloop: Support "Capture Pitch 1000000" rate shift
by GitHub pull_request - opened 27 Jul '21

27 Jul '21

alsa-project/alsa-utils pull request #105 was opened from t123yh: In Linux 5.14, a new feedback mechanism for USB Audio Gadget is introduced (commit e89bb428), along with a new control element "Capture Pitch 1000000". This patch adds support for this feature. Request URL : https://github.com/alsa-project/alsa-utils/pull/105 Patch URL : https://github.com/alsa-project/alsa-utils/pull/105.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and SoundWire
by Pierre-Louis Bossart 27 Jul '21

27 Jul '21

On some platforms with an external HDaudio codec, the DSDT reports the presence of SoundWire devices. Pin-mux restrictions and board reworks usually prevent coexistence between the two types of links, let's prevent unnecessary operations from starting. In the case of a single iDISP codec being detected, we still start the links even if no SoundWire machine configuration was detected, so that we can double-check what the hardware is and add the missing configuration if applicable. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> --- sound/soc/sof/intel/hda.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index e1e368ff2b12..891e6e1b9121 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -187,12 +187,16 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) int hda_sdw_startup(struct snd_sof_dev *sdev) { struct sof_intel_hda_dev *hdev; + struct snd_sof_pdata *pdata = sdev->pdata; hdev = sdev->pdata->hw_pdata; if (!hdev->sdw) return 0; + if (pdata->machine && !pdata->machine->mach_params.link_mask) + return 0; + return sdw_intel_startup(hdev->sdw); } @@ -1002,6 +1006,14 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) hda_mach->mach_params.dmic_num = dmic_num; pdata->machine = hda_mach; pdata->tplg_filename = tplg_filename; + + if (codec_num == 2) { + /* + * Prevent SoundWire links from starting when an external + * HDaudio codec is used + */ + hda_mach->mach_params.link_mask = 0; + } } } -- 2.25.1

2 1

[PATCH] ASoC: topology: Select SND_DYNAMIC_MINORS
by Pierre-Louis Bossart 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> The indexes of the devices are described within the topology file, it is a possibility that the topology encodes invalid indexes when DYNAMIC_MINORS is not enabled in kernel: #define SNDRV_MINOR_COMPRESS 2 /* 2 - 3 */ #define SNDRV_MINOR_HWDEP 4 /* 4 - 7 */ #define SNDRV_MINOR_RAWMIDI 8 /* 8 - 15 */ #define SNDRV_MINOR_PCM_PLAYBACK 16 /* 16 - 23 */ #define SNDRV_MINOR_PCM_CAPTURE 24 /* 24 - 31 */ If the topology assigns an index greater than 7 for PLAYBACK/CAPTURE PCM then there will be minor number collision. As an example: card0 creates a capture PCM with index 10 -> minor = 34 card1 creates compress device with index 0 -> minor = 34 Card1 will fail to instantiate because the minor for the compress stream is already taken. To avoid seemingly mysterious issues with card creation, select the DYNAMIC_MINORS when the topology is enabled. The other option would be to try to do out of bound index checks in case of DYNAMIC_MINOR is not enabled and do not even attempt to create the device with failing the topology load. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8a13462e1a63..5dcf77af07af 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_COMPRESS config SND_SOC_TOPOLOGY bool + select SND_DYNAMIC_MINORS config SND_SOC_TOPOLOGY_KUNIT_TEST tristate "KUnit tests for SoC topology" -- 2.25.1

2 1

[PATCH] ASoC: amd: fix an IS_ERR() vs NULL bug in probe
by Dan Carpenter 27 Jul '21

27 Jul '21

The devm_ioremap() function returns NULL on error, it doesn't return error pointers. Fixes: e550339ee652 ("ASoC: amd: add vangogh i2s controller driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/vangogh/acp5x-i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-i2s.c b/sound/soc/amd/vangogh/acp5x-i2s.c index 86b37c93c3d2..2705e57eb713 100644 --- a/sound/soc/amd/vangogh/acp5x-i2s.c +++ b/sound/soc/amd/vangogh/acp5x-i2s.c @@ -399,8 +399,8 @@ static int acp5x_dai_probe(struct platform_device *pdev) } adata->acp5x_base = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (IS_ERR(adata->acp5x_base)) - return PTR_ERR(adata->acp5x_base); + if (!adata->acp5x_base) + return -ENOMEM; adata->master_mode = I2S_MASTER_MODE_ENABLE; dev_set_drvdata(&pdev->dev, adata); -- 2.20.1

2 1

[PATCH] ASoC: rt5682: enable SAR ADC power saving mode during suspend
by shumingf＠realtek.com 27 Jul '21

27 Jul '21

From: Shuming Fan <shumingf(a)realtek.com> The SAR ADC power saving mode could reduce power consumption on MICVDD rail. Therefore, this patch saves power consumption during suspend state if the headset was connected. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt5682.c | 44 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 44 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 51ecaa2abcd1..7dc01ae6bb66 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2916,10 +2916,46 @@ static void rt5682_remove(struct snd_soc_component *component) static int rt5682_suspend(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + unsigned int val; if (rt5682->is_sdw) return 0; + cancel_delayed_work_sync(&rt5682->jack_detect_work); + cancel_delayed_work_sync(&rt5682->jd_check_work); + if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, + RT5682_CTRL_MB1_REG | RT5682_CTRL_MB2_REG); + val = snd_soc_component_read(component, + RT5682_CBJ_CTRL_2) & RT5682_JACK_TYPE_MASK; + + switch (val) { + case 0x1: + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_SEL_MB1_MASK | RT5682_SAR_SEL_MB2_MASK, + RT5682_SAR_SEL_MB1_NOSEL | RT5682_SAR_SEL_MB2_SEL); + break; + case 0x2: + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_SEL_MB1_MASK | RT5682_SAR_SEL_MB2_MASK, + RT5682_SAR_SEL_MB1_SEL | RT5682_SAR_SEL_MB2_NOSEL); + break; + default: + break; + } + + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, 0); + + /* enter SAR ADC power saving mode */ + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, 0); + snd_soc_component_update_bits(component, RT5682_SAR_IL_CMD_1, + RT5682_SAR_BUTT_DET_MASK | RT5682_SAR_BUTDET_MODE_MASK | RT5682_SAR_BUTDET_RST_MASK, + RT5682_SAR_BUTT_DET_EN | RT5682_SAR_BUTDET_POW_SAV | RT5682_SAR_BUTDET_RST_NORMAL); + } + regcache_cache_only(rt5682->regmap, true); regcache_mark_dirty(rt5682->regmap); return 0; @@ -2935,6 +2971,14 @@ static int rt5682_resume(struct snd_soc_component *component) regcache_cache_only(rt5682->regmap, false); regcache_sync(rt5682->regmap); + if (rt5682->hs_jack && rt5682->jack_type == SND_JACK_HEADSET) { + snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, + RT5682_MB1_PATH_MASK | RT5682_MB2_PATH_MASK, + RT5682_CTRL_MB1_FSM | RT5682_CTRL_MB2_FSM); + snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, + RT5682_PWR_CBJ, RT5682_PWR_CBJ); + } + mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(250)); -- 2.31.1

2 1

[PATCH AUTOSEL 4.19 2/6] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 0b587585b38b..6071de5fca57 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -147,8 +147,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

1 0

[PATCH AUTOSEL 5.4 3/9] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index cb024955c978..73c5f6c8ed69 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 12/17] ASoC: ti: j721e-evm: Check for not initialized parent_clk_id
by Sasha Levin 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit 82d28b67f780910f816fe1cfb0f676fc38c4cbb3 ] During probe the parent_clk_id is set to -1 which should not be used to array index within hsdiv_rates[]. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210717122820.1467-3-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/j721e-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 017c4ad11ca6..265bbc5a2f96 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI", -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 11/17] ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup
by Sasha Levin 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit 78d2a05ef22e7b5863b01e073dd6a06b3979bb00 ] In case of an error within j721e_audio_startup() the domain->active must be decremented to avoid unbalanced counter. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210717122820.1467-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/j721e-evm.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index a7c0484d44ec..017c4ad11ca6 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 08/17] ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend
by Sasha Levin 27 Jul '21

27 Jul '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 6a503e1c455316fd0bfd8188c0a62cce7c5525ca ] While using the DMIC recording, the garbled data will be captured by the DMIC. It is caused by the critical power of PLL closed in the jack detect function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210716085853.20170-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index d9878173ff89..2e41b8c169e5 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -971,10 +971,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, -- 2.30.2

1 0

[PATCH AUTOSEL 5.10 04/17] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 14/21] ASoC: ti: j721e-evm: Check for not initialized parent_clk_id
by Sasha Levin 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit 82d28b67f780910f816fe1cfb0f676fc38c4cbb3 ] During probe the parent_clk_id is set to -1 which should not be used to array index within hsdiv_rates[]. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210717122820.1467-3-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/j721e-evm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index 017c4ad11ca6..265bbc5a2f96 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -197,7 +197,7 @@ static int j721e_configure_refclk(struct j721e_priv *priv, return ret; } - if (priv->hsdiv_rates[domain->parent_clk_id] != scki) { + if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) { dev_dbg(priv->dev, "%s configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n", audio_domain == J721E_AUDIO_DOMAIN_CPB ? "CPB" : "IVI", -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 13/21] ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup
by Sasha Levin 27 Jul '21

27 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit 78d2a05ef22e7b5863b01e073dd6a06b3979bb00 ] In case of an error within j721e_audio_startup() the domain->active must be decremented to avoid unbalanced counter. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210717122820.1467-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/j721e-evm.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c index a7c0484d44ec..017c4ad11ca6 100644 --- a/sound/soc/ti/j721e-evm.c +++ b/sound/soc/ti/j721e-evm.c @@ -278,23 +278,29 @@ static int j721e_audio_startup(struct snd_pcm_substream *substream) j721e_rule_rate, &priv->rate_range, SNDRV_PCM_HW_PARAM_RATE, -1); - mutex_unlock(&priv->mutex); if (ret) - return ret; + goto out; /* Reset TDM slots to 32 */ ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; for_each_rtd_codec_dais(rtd, i, codec_dai) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32); if (ret && ret != -ENOTSUPP) - return ret; + goto out; } - return 0; + if (ret == -ENOTSUPP) + ret = 0; +out: + if (ret) + domain->active--; + mutex_unlock(&priv->mutex); + + return ret; } static int j721e_audio_hw_params(struct snd_pcm_substream *substream, -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 10/21] ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend
by Sasha Levin 27 Jul '21

27 Jul '21

From: Oder Chiou <oder_chiou(a)realtek.com> [ Upstream commit 6a503e1c455316fd0bfd8188c0a62cce7c5525ca ] While using the DMIC recording, the garbled data will be captured by the DMIC. It is caused by the critical power of PLL closed in the jack detect function. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Link: https://lore.kernel.org/r/20210716085853.20170-1-oder_chiou@realtek.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/rt5682.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index e4c91571abae..abcd6f483788 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -973,10 +973,14 @@ int rt5682_headset_detect(struct snd_soc_component *component, int jack_insert) rt5682_enable_push_button_irq(component, false); snd_soc_component_update_bits(component, RT5682_CBJ_CTRL_1, RT5682_TRIG_JD_MASK, RT5682_TRIG_JD_LOW); - if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS")) + if (!snd_soc_dapm_get_pin_status(dapm, "MICBIAS") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_MB, 0); - if (!snd_soc_dapm_get_pin_status(dapm, "Vref2")) + if (!snd_soc_dapm_get_pin_status(dapm, "Vref2") && + !snd_soc_dapm_get_pin_status(dapm, "PLL1") && + !snd_soc_dapm_get_pin_status(dapm, "PLL2B")) snd_soc_component_update_bits(component, RT5682_PWR_ANLG_1, RT5682_PWR_VREF2, 0); snd_soc_component_update_bits(component, RT5682_PWR_ANLG_3, -- 2.30.2

1 0

[PATCH AUTOSEL 5.13 04/21] ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits
by Sasha Levin 27 Jul '21

27 Jul '21

From: Kyle Russell <bkylerussell(a)gmail.com> [ Upstream commit 9cf76a72af6ab81030dea6481b1d7bdd814fbdaf ] These are backwards from Table 7-71 of the TLV320AIC3100 spec [1]. This was broken in 12eb4d66ba2e when BCLK_MASTER and WCLK_MASTER were converted from 0x08 and 0x04 to BIT(2) and BIT(3), respectively. -#define AIC31XX_BCLK_MASTER 0x08 -#define AIC31XX_WCLK_MASTER 0x04 +#define AIC31XX_BCLK_MASTER BIT(2) +#define AIC31XX_WCLK_MASTER BIT(3) Probably just a typo since the defines were not listed in bit order. [1] https://www.ti.com/lit/gpn/tlv320aic3100 Signed-off-by: Kyle Russell <bkylerussell(a)gmail.com> Link: https://lore.kernel.org/r/20210622010941.241386-1-bkylerussell@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 81952984613d..2513922a0292 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -151,8 +151,8 @@ struct aic31xx_pdata { #define AIC31XX_WORD_LEN_24BITS 0x02 #define AIC31XX_WORD_LEN_32BITS 0x03 #define AIC31XX_IFACE1_MASTER_MASK GENMASK(3, 2) -#define AIC31XX_BCLK_MASTER BIT(2) -#define AIC31XX_WCLK_MASTER BIT(3) +#define AIC31XX_BCLK_MASTER BIT(3) +#define AIC31XX_WCLK_MASTER BIT(2) /* AIC31XX_DATA_OFFSET */ #define AIC31XX_DATA_OFFSET_MASK GENMASK(7, 0) -- 2.30.2

1 0

[PATCH 01/54] ASoC: dt-bindings: Add WM8978 Binding
by Maxime Ripard 27 Jul '21

27 Jul '21

Even though we had the wm8978 driver for some time and a number of boards using it already, we never had a binding for it. Let's add it based on what the driver expects and the boards are providing. Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../devicetree/bindings/sound/wlf,wm8978.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8978.yaml diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml new file mode 100644 index 000000000000..6761380261e3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8978.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8978 Codec Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + const: wlf,wm8978 + + reg: + maxItems: 1 + + spi-max-frequency: + maximum: 526000 + +required: + - '#sound-dai-cells' + - compatible + - reg + +additionalProperties: false + +examples: + - | + spi { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + spi-max-frequency = <500000>; + }; + }; + + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@0 { + #sound-dai-cells = <0>; + compatible = "wlf,wm8978"; + reg = <0>; + }; + }; + +... -- 2.31.1

4 4

Pulseaudio craches on start with SegFault error
by GitHub issues - opened 27 Jul '21

27 Jul '21

alsa-project/alsa-plugins issue #31 was opened from jmtrivial: Dear alsa-project developers, I am using a debian (testing) on my computer, together with deb-multimedia.org repository. As some other users with external USB audio devices, I recently discovered a segfault: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=991337 I discussed withe Christian Marillat (maintainer of the deb-multimedia repository) and we tried a couple of alsa versions, but the bug is still present if I'm using a recent libasound2-plugins such as this one: https://www.deb-multimedia.org/tests/libasound2-plugins_1.2.2+git4219f4b-dm… It does not crash with libasound2-plugins 1.2.2-2 from debian testing. You will found attached the log file from pulseaudio -vvv. [pulseaudio.log](https://github.com/alsa-project/alsa-plugins/files/6884875/… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/31 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: seq: Fix comments of wrong client number for MIDI Passthrough
by Takashi Iwai 27 Jul '21

27 Jul '21

MIDI Passthrough sequencer client is assigned always to the fixed number 14, while it's wrongly documented in the comments as if 62, which was an old number that was used during development. Fix all those numbers. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_dummy.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/seq/seq_dummy.c b/sound/core/seq/seq_dummy.c index ac760b1e3d12..8c18d8c4177e 100644 --- a/sound/core/seq/seq_dummy.c +++ b/sound/core/seq/seq_dummy.c @@ -20,15 +20,15 @@ are redirected to output port immediately. The routing can be done via aconnect program in alsa-utils. - Each client has a static client number 62 (= SNDRV_SEQ_CLIENT_DUMMY). + Each client has a static client number 14 (= SNDRV_SEQ_CLIENT_DUMMY). If you want to auto-load this module, you may add the following alias in your /etc/conf.modules file. - alias snd-seq-client-62 snd-seq-dummy + alias snd-seq-client-14 snd-seq-dummy - The module is loaded on demand for client 62, or /proc/asound/seq/ + The module is loaded on demand for client 14, or /proc/asound/seq/ is accessed. If you don't need this module to be loaded, alias - snd-seq-client-62 as "off". This will help modprobe. + snd-seq-client-14 as "off". This will help modprobe. The number of ports to be created can be specified via the module parameter "ports". For example, to create four ports, add the -- 2.26.2

1 0

[PATCH] ALSA: usb-audio: Add registration quirk for JBL Quantum 600
by Alexander Tsoy 27 Jul '21

27 Jul '21

Apparently JBL Quantum 600 has multiple hardware revisions. Apply registration quirk to another device id as well. Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e7accd87e063..326d1b0ea5e6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1899,6 +1899,7 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203c, 2), /* JBL Quantum 600 */ REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; -- 2.31.1

3 3

MIDI Through Port Number in ALSA Sequencer
by rv71＠onet.eu 27 Jul '21

27 Jul '21

Hello, ALSA Sequencer always creates MIDI Through Port with the id 14:0. However, I can't find where it is defined that this special port will have this particular id/number. I also don't see any definition in seq.h regarding this ( unlike the timer and announce special port, for example). Can you explain? Thanks Robert

2 1

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

27 Jul '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 | 11 + 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 | 1234 +++++++++++++++++++++++++++ sound/pci/hda/patch_cs8409.h | 369 ++++++++ 6 files changed, 2176 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

3 30

[PATCH] ALSA: hda/realtek: Fix headset mic for Acer SWIFT SF314-56 (ALC256)
by Nikos Liolios 27 Jul '21

27 Jul '21

The issue on Acer SWIFT SF314-56 is that headset microphone doesn't work. The following quirk fixed headset microphone issue. The fixup was found by trial and error. Note that the fixup of SF314-54/55 (ALC256_FIXUP_ACER_HEADSET_MIC) was not successful on my SF314-56. Signed-off-by: Nikos Liolios <liolios.nk(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index caaf0e8aa..14e1ab7c7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8274,6 +8274,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x129c, "Acer SWIFT SF314-55", ALC256_FIXUP_ACER_HEADSET_MIC), + SND_PCI_QUIRK(0x1025, 0x1300, "Acer SWIFT SF314-56", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1308, "Acer Aspire Z24-890", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x132a, "Acer TravelMate B114-21", ALC233_FIXUP_ACER_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1330, "Acer TravelMate X514-51T", ALC255_FIXUP_ACER_HEADSET_MIC), -- 2.31.1

2 1

Writing a driver for multiple Codecs with one mcAsp
by Kabaz Shlomo 27 Jul '21

27 Jul '21

Hello , I am trying to add to the DM8148 board functionality that support multiple Codecs for One McAsp . The development env is DVRRDK4 of TI . Does any one know how can I do it ? OR what are the concept of adding such support ? Regards , Shlomo Kabaz . The information in this e-mail transmission contains proprietary and business sensitive information. Unauthorized interception of this e-mail may constitute a violation of law. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. You are also asked to contact the sender by reply email and immediately destroy all copies of the original message.

1 0

[PATCH] ASoC: rt1015: Remove unnecessary flush work on rt1015 driver
by Jack Yu 27 Jul '21

27 Jul '21

Remove workqueue of flush work in rt1015 driver since we don't need it after internal discussion, and there is no impact on performance without this workqueue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 39 +-------------------------------------- sound/soc/codecs/rt1015.h | 2 -- 2 files changed, 1 insertion(+), 40 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 9238f12999aa..b3e69bf245dd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -547,31 +547,6 @@ static int rt1015_bypass_boost_put(struct snd_kcontrol *kcontrol, return 0; } -static void rt1015_flush_work(struct work_struct *work) -{ - struct rt1015_priv *rt1015 = container_of(work, struct rt1015_priv, - flush_work.work); - struct snd_soc_component *component = rt1015->component; - unsigned int val, i; - - for (i = 0; i < 200; ++i) { - usleep_range(1000, 1500); - dev_dbg(component->dev, "Flush DAC (retry:%u)\n", i); - regmap_read(rt1015->regmap, RT1015_CLK_DET, &val); - if (val & 0x800) - break; - } - - regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x0597); - regmap_write(rt1015->regmap, RT1015_SYS_RST1, 0x05f7); - regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x0028); - - if (val & 0x800) - dev_dbg(component->dev, "Flush DAC completed.\n"); - else - dev_warn(component->dev, "Fail to flush DAC data.\n"); -} - static const struct snd_kcontrol_new rt1015_snd_controls[] = { SOC_SINGLE_TLV("DAC Playback Volume", RT1015_DAC1, RT1015_DAC_VOL_SFT, 127, 0, dac_vol_tlv), @@ -630,10 +605,6 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, } break; - case SND_SOC_DAPM_POST_PMU: - regmap_write(rt1015->regmap, RT1015_MAN_I2C, 0x00a8); - break; - case SND_SOC_DAPM_POST_PMD: if (rt1015->bypass_boost == RT1015_Enable_Boost) { snd_soc_component_write(component, @@ -653,8 +624,6 @@ static int r1015_dac_event(struct snd_soc_dapm_widget *w, RT1015_SYS_RST2, 0x0b9a); } rt1015->dac_is_used = 0; - - cancel_delayed_work_sync(&rt1015->flush_work); break; default: @@ -687,8 +656,6 @@ static int rt1015_amp_drv_event(struct snd_soc_dapm_widget *w, } break; case SND_SOC_DAPM_POST_PMU: - if (rt1015->hw_config == RT1015_HW_28) - schedule_delayed_work(&rt1015->flush_work, msecs_to_jiffies(10)); msleep(rt1015->pdata.power_up_delay_ms); break; default: @@ -702,7 +669,7 @@ static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { NULL, 0), SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, - r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUT_DRV_E("Amp Drv", SND_SOC_NOPM, 0, 0, NULL, 0, rt1015_amp_drv_event, SND_SOC_DAPM_PRE_PMU | @@ -1028,7 +995,6 @@ static int rt1015_probe(struct snd_soc_component *component) snd_soc_component_get_drvdata(component); rt1015->component = component; - INIT_DELAYED_WORK(&rt1015->flush_work, rt1015_flush_work); return 0; } @@ -1037,7 +1003,6 @@ static void rt1015_remove(struct snd_soc_component *component) { struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); - cancel_delayed_work_sync(&rt1015->flush_work); regmap_write(rt1015->regmap, RT1015_RESET, 0); } @@ -1180,8 +1145,6 @@ static int rt1015_i2c_probe(struct i2c_client *i2c, return ret; } - rt1015->hw_config = (i2c->addr == 0x29) ? RT1015_HW_29 : RT1015_HW_28; - ret = regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); if (ret) { dev_err(&i2c->dev, diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h index 14344532048e..c9f636af7fd1 100644 --- a/sound/soc/codecs/rt1015.h +++ b/sound/soc/codecs/rt1015.h @@ -444,8 +444,6 @@ struct rt1015_priv { int bypass_boost; int dac_is_used; int cali_done; - int hw_config; - struct delayed_work flush_work; }; #endif /* __RT1015_H__ */ -- 2.31.1

2 1

[PATCH] ASoC: Intel: sof_da7219_mx98360a: fail to initialize soundcard
by Brent Lu 27 Jul '21

27 Jul '21

The default codec for speaker amp's DAI Link is max98373 and will be overwritten in probe function if the board id is sof_da7219_mx98360a. However, the probe function does not do it because the board id is changed in earlier commit. Fixes: 1cc04d195dc2 ("ASoC: Intel: sof_da7219_max98373: shrink platform_id below 20 characters") Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_da7219_max98373.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_da7219_max98373.c b/sound/soc/intel/boards/sof_da7219_max98373.c index 896251d742fe..b7b3b0bf994a 100644 --- a/sound/soc/intel/boards/sof_da7219_max98373.c +++ b/sound/soc/intel/boards/sof_da7219_max98373.c @@ -404,7 +404,7 @@ static int audio_probe(struct platform_device *pdev) return -ENOMEM; /* By default dais[0] is configured for max98373 */ - if (!strcmp(pdev->name, "sof_da7219_max98360a")) { + if (!strcmp(pdev->name, "sof_da7219_mx98360a")) { dais[0] = (struct snd_soc_dai_link) { .name = "SSP1-Codec", .id = 0, -- 2.17.1

3 2

[PATCH] ASoC: tlv320aic31xx: Fix jack detection after suspend
by Mark Brown 26 Jul '21

26 Jul '21

The tlv320aic31xx driver relies on regcache_sync() to restore the register contents after going to _BIAS_OFF, for example during system suspend. This does not work for the jack detection configuration since that is configured via the same register that status is read back from so the register is volatile and not cached. This can also cause issues during init if the jack detection ends up getting set up before the CODEC is initially brought out of _BIAS_OFF, we will reset the CODEC and resync the cache as part of that process. Fix this by explicitly reapplying the jack detection configuration after resyncing the register cache during power on. This issue was found by an engineer working off-list on a product kernel, I just wrote up the upstream fix. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index b504d63385b3..e3da8f41fd1a 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1256,6 +1256,13 @@ static int aic31xx_power_on(struct snd_soc_component *component) return ret; } + /* + * The jack detection configuration is in the same register + * that is used to report jack detect status so is volatile + * and not covered by the cache sync, restore it separately. + */ + aic31xx_set_jack(component, aic31xx->jack, NULL); + return 0; } -- 2.20.1

2 3

Writing a driver for multiple Codecs with one mcAsp
by Kabaz Shlomo 26 Jul '21

26 Jul '21

Hello , I am trying to add to the DM8148 board functionality that support multiple Codecs for One McAsp . The development env is DVRRDK4 of TI . Does any one know how can I do it ? OR what are the concept of adding such support ? Regards , Shlomo Kabaz . The information in this e-mail transmission contains proprietary and business sensitive information. Unauthorized interception of this e-mail may constitute a violation of law. If you are not the intended recipient, you are hereby notified that any review, dissemination, distribution or duplication of this communication is strictly prohibited. You are also asked to contact the sender by reply email and immediately destroy all copies of the original message.

1 0

Intel HDA Driver Issue With PCM RW Interleaved
by Ryan McClue 26 Jul '21

26 Jul '21

Hi, I'm trying to play a simple PCM buffer single threaded without buffer underrun. Program source:https://gist.github.com/takaswie/d8a9553c230971d974a3dfc759b2ff67 (struct snd_pcm_sw_params).start_thresholdis altered so as to allow specifying the start of audio frame transmission and prevent underrun issues. In my application, onioctl(fd, SNDRV_PCM_IOCTL_STATUS, &status), I receive:File descriptor in bad state Output of$(dmesg)is:[ 69.013084] snd_hda_intel 0000:00:1f.3: Too big adjustment 128 From a previous response on thealsa-develmailing list: > The driver has some small number of periods to be > put at the beginning of the buffer for correcting the reported > position. Judging from the message, you seem to have set a quite high > rate and a very shot number, and that's possibly too tight to operate > properly. However, I'm not sure what to change in this code to address this issue and get it to work. I tried alteringMY_PLAYBACK_DURATIONandparams.start_thresholdbut get the same error. The program dumps the current hardware parameters: Current hardware parameters: Changed parameters: access format subformat sample-bits frame-bits channels rate period-time period-size period-bytes periods buffer-time buffer-size buffer-bytes tick-time Runtime parameters: access: readwrite-interleaved format: s16-le subformat: std sample-bits: [16, 16], integer, frame-bits: [32, 32], integer, channels: [2, 2], integer, rate: [48000, 48000], integer, period-time: (666, 667), period-size: [32, 32], integer, periods: [32, 32], integer, buffer-time: (21333, 21334), buffer-size: [1024, 1024], integer, buffer-bytes: [4096, 4096], integer, tick-time: [0, 0], integer, info: mmap mmap-valid interleaved block-transfer pause sync-start no-period-wakeup has-wall-clock has-link-atime has-link-synchronized-atime most-significant-bits: 16 rate_num: 48000 rate_den: 1 Thanks

1 0

[PATCH] ALSA: usb-audio: Fix superfluous autosuspend recovery
by Takashi Iwai 26 Jul '21

26 Jul '21

The change to restore the autosuspend from the disabled state uses a wrong check: namely, it should have been the exact comparison of the quirk_type instead of the logical and (&&). Otherwise it matches wrongly with the other quirk types. Although re-enabling the autosuspend for the already enabled device shouldn't matter much, it's better to fix the unbalanced call. Fixes: 9799110825db ("ALSA: usb-audio: Disable USB autosuspend properly in setup_disable_autosuspend()") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 2f6a62416c05..a1f8c3a026f5 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -907,7 +907,7 @@ static void usb_audio_disconnect(struct usb_interface *intf) } } - if (chip->quirk_type & QUIRK_SETUP_DISABLE_AUTOSUSPEND) + if (chip->quirk_type == QUIRK_SETUP_DISABLE_AUTOSUSPEND) usb_enable_autosuspend(interface_to_usbdev(intf)); chip->num_interfaces--; -- 2.26.2

2 2

sof-soundwire: rt1316 - DAC L and R is replaced by DAC
by GitHub pull_request - opened 26 Jul '21

26 Jul '21

alsa-project/alsa-ucm-conf pull request #109 was opened from jason77-wang: In the latest codec driver rt1316-sdw.c, the DAC L and DAC R are replaced by DAC, so we change the ucm to add the condition checking, if 'DAC L' exists, will set those mixers in the old way, otherwise, set the mixers with new name. The codec rt1316 is used on a Dell TGL soundwire audio machine. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/109 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/109.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/3] Get rid of some undesirable characters
by Mauro Carvalho Chehab 25 Jul '21

25 Jul '21

Hi Jon, While rebasing my docs tree, I noticed that there are three remaining patches from my past series that replace some UTF-8 chars by ASCII ones that aren't applied yet. Not sure what happened here. Anyway, those are the missing ones. Mauro Carvalho Chehab (3): docs: sound: kernel-api: writing-an-alsa-driver.rst: replace some characters docs: firmware-guide: acpi: dsd: graph.rst: replace some characters docs: virt: kvm: api.rst: replace some characters .../firmware-guide/acpi/dsd/graph.rst | 2 +- .../kernel-api/writing-an-alsa-driver.rst | 2 +- Documentation/virt/kvm/api.rst | 28 +++++++++---------- 3 files changed, 16 insertions(+), 16 deletions(-) -- 2.31.1

2 2

[syzbot] possible deadlock in snd_timer_interrupt
by syzbot 25 Jul '21

25 Jul '21

Hello, syzbot found the following issue on: HEAD commit: f0fddcec6b62 Merge tag 'for-5.14-rc2-tag' of git://git.ker.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=13cbb70e300000 kernel config: https://syzkaller.appspot.com/x/.config?x=5dc0e3202ae2f574 dashboard link: https://syzkaller.appspot.com/bug?extid=17c16b868e6af41acdd4 compiler: Debian clang version 11.0.1-2, GNU ld (GNU Binutils for Debian) 2.35.1 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=17667ecc300000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=179e0582300000 The issue was bisected to: commit e918188611f073063415f40fae568fa4d86d9044 Author: Boqun Feng <boqun.feng(a)gmail.com> Date: Fri Aug 7 07:42:20 2020 +0000 locking: More accurate annotations for read_lock() bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=109610b2300000 final oops: https://syzkaller.appspot.com/x/report.txt?x=129610b2300000 console output: https://syzkaller.appspot.com/x/log.txt?x=149610b2300000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+17c16b868e6af41acdd4(a)syzkaller.appspotmail.com Fixes: e918188611f0 ("locking: More accurate annotations for read_lock()") ======================================================== WARNING: possible irq lock inversion dependency detected 5.14.0-rc2-syzkaller #0 Not tainted -------------------------------------------------------- syz-executor065/8431 just changed the state of lock: ffff888027e94148 (&timer->lock){..-.}-{2:2}, at: snd_timer_interrupt+0x210/0x1410 sound/core/timer.c:857 but this lock took another, SOFTIRQ-READ-unsafe lock in the past: (&f->f_owner.lock){.+.+}-{2:2} and interrupts could create inverse lock ordering between them. other info that might help us debug this: Chain exists of: &timer->lock --> &new->fa_lock --> &f->f_owner.lock Possible interrupt unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&f->f_owner.lock); local_irq_disable(); lock(&timer->lock); lock(&new->fa_lock); <Interrupt> lock(&timer->lock); *** DEADLOCK *** 4 locks held by syz-executor065/8431: #0: ffff888029390460 (sb_writers#5){.+.+}-{0:0}, at: mnt_want_write+0x3b/0x80 fs/namespace.c:375 #1: ffff888039f60e80 (&type->i_mutex_dir_key#4/1){+.+.}-{3:3}, at: inode_lock_nested include/linux/fs.h:809 [inline] #1: ffff888039f60e80 (&type->i_mutex_dir_key#4/1){+.+.}-{3:3}, at: filename_create+0x19f/0x700 fs/namei.c:3602 #2: ffffffff8c7177c0 (rcu_read_lock){....}-{1:2}, at: rcu_lock_acquire+0x5/0x30 include/linux/rcupdate.h:266 #3: ffffc90000dc0be0 ((&priv->tlist)){+.-.}-{0:0}, at: lockdep_copy_map include/linux/lockdep.h:45 [inline] #3: ffffc90000dc0be0 ((&priv->tlist)){+.-.}-{0:0}, at: call_timer_fn+0xbd/0x210 kernel/time/timer.c:1407 the shortest dependencies between 2nd lock and 1st lock: -> (&f->f_owner.lock){.+.+}-{2:2} { HARDIRQ-ON-R at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_read_lock include/linux/rwlock_api_smp.h:149 [inline] _raw_read_lock+0x32/0x40 kernel/locking/spinlock.c:223 f_getown_ex fs/fcntl.c:211 [inline] do_fcntl+0x150/0x1510 fs/fcntl.c:395 __do_sys_fcntl fs/fcntl.c:471 [inline] __se_sys_fcntl+0xd8/0x1b0 fs/fcntl.c:456 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae SOFTIRQ-ON-R at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_read_lock include/linux/rwlock_api_smp.h:149 [inline] _raw_read_lock+0x32/0x40 kernel/locking/spinlock.c:223 f_getown_ex fs/fcntl.c:211 [inline] do_fcntl+0x150/0x1510 fs/fcntl.c:395 __do_sys_fcntl fs/fcntl.c:471 [inline] __se_sys_fcntl+0xd8/0x1b0 fs/fcntl.c:456 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL READ USE at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:159 [inline] _raw_read_lock_irqsave+0xbb/0x100 kernel/locking/spinlock.c:231 send_sigio+0x2f/0x300 fs/fcntl.c:795 kill_fasync_rcu fs/fcntl.c:1019 [inline] kill_fasync+0x1e3/0x430 fs/fcntl.c:1033 snd_timer_user_ccallback+0x3ee/0x710 sound/core/timer.c:1387 snd_timer_notify1+0x19e/0x340 sound/core/timer.c:516 snd_timer_start1+0x53d/0x640 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:697 [inline] snd_timer_user_start sound/core/timer.c:1985 [inline] __snd_timer_user_ioctl+0xe18/0x5ed0 sound/core/timer.c:2108 snd_timer_user_ioctl+0x5d/0x80 sound/core/timer.c:2129 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:1069 [inline] __se_sys_ioctl+0xfb/0x170 fs/ioctl.c:1055 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff907d04a0>] __alloc_file.__key+0x0/0x10 ... acquired at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:159 [inline] _raw_read_lock_irqsave+0xbb/0x100 kernel/locking/spinlock.c:231 send_sigio+0x2f/0x300 fs/fcntl.c:795 kill_fasync_rcu fs/fcntl.c:1019 [inline] kill_fasync+0x1e3/0x430 fs/fcntl.c:1033 snd_timer_user_ccallback+0x3ee/0x710 sound/core/timer.c:1387 snd_timer_notify1+0x19e/0x340 sound/core/timer.c:516 snd_timer_start1+0x53d/0x640 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:697 [inline] snd_timer_user_start sound/core/timer.c:1985 [inline] __snd_timer_user_ioctl+0xe18/0x5ed0 sound/core/timer.c:2108 snd_timer_user_ioctl+0x5d/0x80 sound/core/timer.c:2129 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:1069 [inline] __se_sys_ioctl+0xfb/0x170 fs/ioctl.c:1055 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&new->fa_lock){....}-{2:2} { INITIAL USE at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_write_lock_irq include/linux/rwlock_api_smp.h:196 [inline] _raw_write_lock_irq+0xae/0xf0 kernel/locking/spinlock.c:311 fasync_remove_entry+0xff/0x1d0 fs/fcntl.c:890 __fput+0x65b/0x7b0 fs/file_table.c:277 task_work_run+0x146/0x1c0 kernel/task_work.c:164 exit_task_work include/linux/task_work.h:32 [inline] do_exit+0x72b/0x2510 kernel/exit.c:825 do_group_exit+0x168/0x2d0 kernel/exit.c:922 get_signal+0x16b0/0x2080 kernel/signal.c:2808 arch_do_signal_or_restart+0x8e/0x6d0 arch/x86/kernel/signal.c:865 handle_signal_work kernel/entry/common.c:148 [inline] exit_to_user_mode_loop kernel/entry/common.c:172 [inline] exit_to_user_mode_prepare+0x191/0x220 kernel/entry/common.c:209 __syscall_exit_to_user_mode_work kernel/entry/common.c:291 [inline] syscall_exit_to_user_mode+0x26/0x60 kernel/entry/common.c:302 do_syscall_64+0x4c/0xb0 arch/x86/entry/common.c:86 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL READ USE at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_read_lock include/linux/rwlock_api_smp.h:149 [inline] _raw_read_lock+0x32/0x40 kernel/locking/spinlock.c:223 kill_fasync_rcu fs/fcntl.c:1012 [inline] kill_fasync+0x13b/0x430 fs/fcntl.c:1033 snd_timer_user_ccallback+0x3ee/0x710 sound/core/timer.c:1387 snd_timer_notify1+0x19e/0x340 sound/core/timer.c:516 snd_timer_start1+0x53d/0x640 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:697 [inline] snd_timer_user_start sound/core/timer.c:1985 [inline] __snd_timer_user_ioctl+0xe18/0x5ed0 sound/core/timer.c:2108 snd_timer_user_ioctl+0x5d/0x80 sound/core/timer.c:2129 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:1069 [inline] __se_sys_ioctl+0xfb/0x170 fs/ioctl.c:1055 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff907d1360>] fasync_insert_entry.__key+0x0/0x40 ... acquired at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_read_lock include/linux/rwlock_api_smp.h:149 [inline] _raw_read_lock+0x32/0x40 kernel/locking/spinlock.c:223 kill_fasync_rcu fs/fcntl.c:1012 [inline] kill_fasync+0x13b/0x430 fs/fcntl.c:1033 snd_timer_user_ccallback+0x3ee/0x710 sound/core/timer.c:1387 snd_timer_notify1+0x19e/0x340 sound/core/timer.c:516 snd_timer_start1+0x53d/0x640 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:697 [inline] snd_timer_user_start sound/core/timer.c:1985 [inline] __snd_timer_user_ioctl+0xe18/0x5ed0 sound/core/timer.c:2108 snd_timer_user_ioctl+0x5d/0x80 sound/core/timer.c:2129 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:1069 [inline] __se_sys_ioctl+0xfb/0x170 fs/ioctl.c:1055 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&timer->lock){..-.}-{2:2} { IN-SOFTIRQ-W at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0xb3/0x100 kernel/locking/spinlock.c:159 snd_timer_interrupt+0x210/0x1410 sound/core/timer.c:857 call_timer_fn+0xf6/0x210 kernel/time/timer.c:1417 expire_timers kernel/time/timer.c:1462 [inline] __run_timers+0x6ff/0x910 kernel/time/timer.c:1731 run_timer_softirq+0x63/0xf0 kernel/time/timer.c:1744 __do_softirq+0x372/0x783 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x21b/0x260 kernel/softirq.c:636 irq_exit_rcu+0x5/0x20 kernel/softirq.c:648 sysvec_apic_timer_interrupt+0x91/0xb0 arch/x86/kernel/apic/apic.c:1100 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 __preempt_count_add arch/x86/include/asm/preempt.h:80 [inline] rcu_is_watching+0xd/0xb0 kernel/rcu/tree.c:1159 rcu_read_lock include/linux/rcupdate.h:688 [inline] get_cached_acl+0x103/0x540 fs/posix_acl.c:44 get_acl+0x20/0x2e0 fs/posix_acl.c:108 posix_acl_create+0x119/0x560 fs/posix_acl.c:616 ext4_init_acl+0xa6/0x3d0 fs/ext4/acl.c:282 __ext4_new_inode+0x44e7/0x5810 fs/ext4/ialloc.c:1316 ext4_mkdir+0x3fa/0xc10 fs/ext4/namei.c:2921 vfs_mkdir+0x451/0x630 fs/namei.c:3823 do_mkdirat+0x267/0x3e0 fs/namei.c:3848 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL USE at: lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0xb3/0x100 kernel/locking/spinlock.c:159 snd_timer_resolution sound/core/timer.c:489 [inline] snd_timer_user_params sound/core/timer.c:1852 [inline] __snd_timer_user_ioctl+0x1df7/0x5ed0 sound/core/timer.c:2101 snd_timer_user_ioctl+0x5d/0x80 sound/core/timer.c:2129 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:1069 [inline] __se_sys_ioctl+0xfb/0x170 fs/ioctl.c:1055 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff90b60220>] snd_timer_new.__key+0x0/0x40 ... acquired at: mark_usage kernel/locking/lockdep.c:4488 [inline] __lock_acquire+0xca1/0x6100 kernel/locking/lockdep.c:4969 lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0xb3/0x100 kernel/locking/spinlock.c:159 snd_timer_interrupt+0x210/0x1410 sound/core/timer.c:857 call_timer_fn+0xf6/0x210 kernel/time/timer.c:1417 expire_timers kernel/time/timer.c:1462 [inline] __run_timers+0x6ff/0x910 kernel/time/timer.c:1731 run_timer_softirq+0x63/0xf0 kernel/time/timer.c:1744 __do_softirq+0x372/0x783 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x21b/0x260 kernel/softirq.c:636 irq_exit_rcu+0x5/0x20 kernel/softirq.c:648 sysvec_apic_timer_interrupt+0x91/0xb0 arch/x86/kernel/apic/apic.c:1100 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 __preempt_count_add arch/x86/include/asm/preempt.h:80 [inline] rcu_is_watching+0xd/0xb0 kernel/rcu/tree.c:1159 rcu_read_lock include/linux/rcupdate.h:688 [inline] get_cached_acl+0x103/0x540 fs/posix_acl.c:44 get_acl+0x20/0x2e0 fs/posix_acl.c:108 posix_acl_create+0x119/0x560 fs/posix_acl.c:616 ext4_init_acl+0xa6/0x3d0 fs/ext4/acl.c:282 __ext4_new_inode+0x44e7/0x5810 fs/ext4/ialloc.c:1316 ext4_mkdir+0x3fa/0xc10 fs/ext4/namei.c:2921 vfs_mkdir+0x451/0x630 fs/namei.c:3823 do_mkdirat+0x267/0x3e0 fs/namei.c:3848 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae stack backtrace: CPU: 1 PID: 8431 Comm: syz-executor065 Not tainted 5.14.0-rc2-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0x1ae/0x29f lib/dump_stack.c:105 print_irq_inversion_bug+0xa58/0xda0 kernel/locking/lockdep.c:4000 check_usage_forwards kernel/locking/lockdep.c:2585 [inline] mark_lock_irq kernel/locking/lockdep.c:4163 [inline] mark_lock+0x1064/0x1eb0 kernel/locking/lockdep.c:4593 mark_usage kernel/locking/lockdep.c:4488 [inline] __lock_acquire+0xca1/0x6100 kernel/locking/lockdep.c:4969 lock_acquire+0x182/0x4a0 kernel/locking/lockdep.c:5625 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0xb3/0x100 kernel/locking/spinlock.c:159 snd_timer_interrupt+0x210/0x1410 sound/core/timer.c:857 call_timer_fn+0xf6/0x210 kernel/time/timer.c:1417 expire_timers kernel/time/timer.c:1462 [inline] __run_timers+0x6ff/0x910 kernel/time/timer.c:1731 run_timer_softirq+0x63/0xf0 kernel/time/timer.c:1744 __do_softirq+0x372/0x783 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x21b/0x260 kernel/softirq.c:636 irq_exit_rcu+0x5/0x20 kernel/softirq.c:648 sysvec_apic_timer_interrupt+0x91/0xb0 arch/x86/kernel/apic/apic.c:1100 </IRQ> asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 RIP: 0010:rcu_dynticks_curr_cpu_in_eqs kernel/rcu/tree.c:339 [inline] RIP: 0010:rcu_is_watching+0xd/0xb0 kernel/rcu/tree.c:1160 Code: 41 5e 41 5f 5d c3 e8 92 a8 6d 08 41 f7 c4 00 02 00 00 75 b6 eb b5 e8 62 a8 6d 08 66 90 41 57 41 56 53 65 83 05 4b e3 96 7e 01 <e8> 1e bb 6d 08 89 c3 83 f8 08 73 74 49 bf 00 00 00 00 00 fc ff df RSP: 0018:ffffc9000164f820 EFLAGS: 00000282 RAX: 12a9aafecb760900 RBX: 1ffff920002c9f0c RCX: ffffffff816213c5 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 0000000000000000 RBP: ffffc9000164f8e8 R08: dffffc0000000000 R09: fffffbfff1f5e1b1 R10: fffffbfff1f5e1b1 R11: 0000000000000000 R12: 0000000000004000 R13: dffffc0000000000 R14: 1ffff920002c9f10 R15: ffff888039f60d48 rcu_read_lock include/linux/rcupdate.h:688 [inline] get_cached_acl+0x103/0x540 fs/posix_acl.c:44 get_acl+0x20/0x2e0 fs/posix_acl.c:108 posix_acl_create+0x119/0x560 fs/posix_acl.c:616 ext4_init_acl+0xa6/0x3d0 fs/ext4/acl.c:282 __ext4_new_inode+0x44e7/0x5810 fs/ext4/ialloc.c:1316 ext4_mkdir+0x3fa/0xc10 fs/ext4/namei.c:2921 vfs_mkdir+0x451/0x630 fs/namei.c:3823 do_mkdirat+0x267/0x3e0 fs/namei.c:3848 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x3d/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x445d87 Code: 73 01 c3 48 c7 c1 bc ff ff ff f7 d8 64 89 01 48 83 c8 ff c3 66 2e 0f 1f 84 00 00 00 00 00 0f 1f 44 00 00 b8 53 00 00 00 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 bc ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007ffc332de6a8 EFLAGS: 00000202 ORIG_RAX: 0000000000000053 RAX: ffffffffffffffda RBX: 000000000000e3bb RCX: 0000000000445d87 RDX: 0000000000000000 RSI: 00000000000001ff RDI: 00007ffc332de740 RBP: 0000000000002ebc R08: 0000000000000000 R09: 0000000000000005 R10: 00007ffc332de445 R11: 0000000000000202 R12: 00007ffc332de6e4 R13: 00007ffc332de740 R14: 00000000000000bd R15: 431bde82d7b634db --- This report is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this issue. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. For information about bisection process see: https://goo.gl/tpsmEJ#bisection syzbot can test patches for this issue, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

Trying to debug in issue in my code
by samiam moorecad.com 24 Jul '21

24 Jul '21

Hi, I am trying to debug an issue in my code, and want to gain a bit more visibility on the issue. My crash is: Thread 1 "connectwave" received signal SIGSEGV, Segmentation fault. 0x00007ffff7dbfae2 in alsa_lib_resampler_process_int () from /lib/x86_64-linux-gnu/alsa-lib/libasound_module_rate_speexrate.so (gdb) bt #0 0x00007ffff7dbfae2 in alsa_lib_resampler_process_int () at /lib/x86_64-linux-gnu/alsa-lib/libasound_module_rate_speexrate.so #1 0x00007ffff7dbfd79 in alsa_lib_resampler_process_interleaved_int () at /lib/x86_64-linux-gnu/alsa-lib/libasound_module_rate_speexrate.so #2 0x00007ffff7dbe22d in () at /lib/x86_64-linux-gnu/alsa-lib/libasound_module_rate_speexrate.so #3 0x00007ffff7d2ac52 in do_convert (dst_areas=0x5555558b4820, dst_offset=0, dst_frames=45812269, src_areas=<optimized out>, src_offset=<optimized out>, src_frames=1024, channels=2, rate=0x5555558b4650) at pcm_rate.c:526 #4 0x00007ffff7d2ba30 in snd_pcm_rate_read_areas1 (slave_offset=<optimized out>, slave_areas=<optimized out>, offset=0, areas=0x5555558b4820, pcm=0x5555558af8e0) at pcm_rate.c:557 #5 snd_pcm_rate_grab_next_period (hw_offset=0, pcm=0x5555558af8e0) at pcm_rate.c:826 #6 snd_pcm_rate_avail_update_capture (slave_size=1056, pcm=0x5555558af8e0) at pcm_rate.c:971 #7 snd_pcm_rate_avail_update (pcm=0x5555558af8e0) at pcm_rate.c:996 #8 0x00007ffff7d101e2 in __snd_pcm_avail_update (pcm=0x5555558af8e0, pcm=0x5555558af8e0) at pcm_local.h:436 #9 snd1_pcm_read_areas (pcm=pcm@entry=0x5555558af8e0, areas=areas@entry=0x7fffffffd5d0, offset=offset@entry=0, size=size@entry=512, func=func@entry=0x7ffff7d1c110 <snd_pcm_mmap_read_areas>) at pcm.c:7437 #10 0x00007ffff7d1c560 in snd_pcm_mmap_readi (pcm=0x5555558af8e0, buffer=<optimized out>, size=512) at pcm_mmap.c:203 #11 0x00007ffff7de5bea in rdalsapcmin (p=21, buff=0x7fffffffd6a0 "\250\327TVUU", len=512) at linux/sound.c:1735 #12 0x00007ffff7dec5a3 in pa_rdwave (p=21, buff=0x7fffffffd6a0 "\250\327TVUU", len=512) at linux/sound.c:5462 #13 0x00005555555555a4 in main (argc=3, argv=0x7fffffffdfb8) at sound_programs/connectwave.c:106 I gather that the target routine of the crash, alsa_lib_resampler_process_int(), is in the alsa-plugins module, so I downloaded alsa-plugins-1.2.5 just so I could gain some insight into what the code is doing when it crashes. I should state quickly here that I believe the crash is my fault, but I am just trying to figure out why it is happening. I was unable to find the routine alsa_lib_resampler_process_int() in that package. Also a ./configure followed by make does not work on it (goes to the idea of substituting my own compiled .so for debug purposes): /home/samiam/projects/alsa/alsa-plugins-1.2.5/usb_stream/pcm_usb_stream.c:480: undefined reference to `snd_config_get_card' Although obviously I didn't put a lot of effort into making it work. DId the routine alsa_lib_resampler_process_int() get eliminated in later editions of the plugins? or am I looking in the wrong place? Thanks in advance, Scott Franco San Jose, CA

2 1

[PATCH 0/4] ALSA: scarlett2: note gen 3 support + fix four issues
by Geoffrey D. Bennett 24 Jul '21

24 Jul '21

Hi Takashi, In the sound-5.14-rc1 merge commit log the highlights included: "Scarlett2 mixer code fixes and enhancements". I think that the new support for Gen 3 devices is significant enough to be worth mentioning. Can you add to the next update a note along the lines of "Support for Focusrite Scarlett Solo/2i2/4i4/8i6/18i8/18i20 Gen 3 audio interface proprietary mixer controls"? This set of patches is relative to v5.14-rc2 and fixes four issues: 1. The Mute/Dim/MSD Mode controls are missing the direction/function parts of the syntax of standard control names as per Documentation/sound/designs/control-names.rst - This could be considered a breaking-stable change if someone is relying on the Mute/Dim control names not changing. I think it's unlikely to be a problem as this driver is still considered experimental and not enabled by default, but if never changing control names is important, then you can drop this patch. 2. The Direct Monitor control on the 2i2 interface is an Enum, not a Switch. - This changes a control name, but that control was only introduced in v5.14-rc1, so not a problem to change it now. 3. Fixes the mute status not being correctly read when the mute button is pressed. - Not applicable for stable; those controls were introduced in v5.14-rc1. 4. Sends the correct notification on line out and speaker switching changes. - Partly applicable to stable, but will need a separate patch. Thanks, Geoffrey. Geoffrey D. Bennett (4): ALSA: scarlett2: Fix Mute/Dim/MSD Mode control names ALSA: scarlett2: Fix Direct Monitor control name for 2i2 ALSA: scarlett2: Correct channel mute status after mute button pressed ALSA: scarlett2: Fix line out/speaker switching notifications sound/usb/mixer_scarlett_gen2.c | 34 +++++++++++++++++++++++---------- 1 file changed, 24 insertions(+), 10 deletions(-) -- 2.31.1

3 6

[PATCH 04/54] ASoC: dt-bindings: Convert Simple Amplifier binding to a schema
by Maxime Ripard 24 Jul '21

24 Jul '21

Simple audio amplifiers are supported by Linux with a matching device tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that driver over to a YAML schema. Cc: alsa-devel(a)alsa-project.org Cc: Jerome Brunet <jbrunet(a)baylibre.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../bindings/sound/simple-amplifier.txt | 17 ------- .../sound/simple-audio-amplifier.yaml | 45 +++++++++++++++++++ 2 files changed, 45 insertions(+), 17 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-amplifier.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-amplifier.txt b/Documentation/devicetree/bindings/sound/simple-amplifier.txt deleted file mode 100644 index b1b097cc9b68..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-amplifier.txt +++ /dev/null @@ -1,17 +0,0 @@ -Simple Amplifier Audio Driver - -Required properties: -- compatible : "dioo,dio2125" or "simple-audio-amplifier" - -Optional properties: -- enable-gpios : the gpio connected to the enable pin of the simple amplifier -- VCC-supply : power supply for the device, as covered - in Documentation/devicetree/bindings/regulator/regulator.txt - -Example: - -amp: analog-amplifier { - compatible = "simple-audio-amplifier"; - VCC-supply = <&regulator>; - enable-gpios = <&gpio GPIOH_3 0>; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml new file mode 100644 index 000000000000..26379377a7ac --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-audio-amplifier.yaml @@ -0,0 +1,45 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-audio-amplifier.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Amplifier Device Tree Bindings + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +properties: + compatible: + enum: + - dioo,dio2125 + - simple-audio-amplifier + + enable-gpios: + maxItems: 1 + + VCC-supply: + description: > + power supply for the device + + sound-name-prefix: + $ref: /schemas/types.yaml#/definitions/string + description: > + See ./name-prefix.txt + +required: + - compatible + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/meson8-gpio.h> + + analog-amplifier { + compatible = "simple-audio-amplifier"; + VCC-supply = <&regulator>; + enable-gpios = <&gpio GPIOH_3 0>; + }; + +... -- 2.31.1

2 1

[PATCH 03/54] ASoC: dt-bindings: Convert SPDIF Transmitter binding to a schema
by Maxime Ripard 24 Jul '21

24 Jul '21

The SPDIF Transmitter binding is used by Linux with a matching Device Tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that driver over to a YAML schema. Cc: alsa-devel(a)alsa-project.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../bindings/sound/linux,spdif-dit.yaml | 32 +++++++++++++++++++ .../bindings/sound/spdif-transmitter.txt | 10 ------ 2 files changed, 32 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml delete mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt diff --git a/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml new file mode 100644 index 000000000000..c6b070e1d014 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,spdif-dit.yaml @@ -0,0 +1,32 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,spdif-dit.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Dummy SPDIF Transmitter Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + compatible: + const: linux,spdif-dit + + "#sound-dai-cells": + const: 0 + +required: + - "#sound-dai-cells" + - compatible + +additionalProperties: false + +examples: + - | + spdif-out { + #sound-dai-cells = <0>; + compatible = "linux,spdif-dit"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt b/Documentation/devicetree/bindings/sound/spdif-transmitter.txt deleted file mode 100644 index 55a85841dd85..000000000000 --- a/Documentation/devicetree/bindings/sound/spdif-transmitter.txt +++ /dev/null @@ -1,10 +0,0 @@ -Device-Tree bindings for dummy spdif transmitter - -Required properties: - - compatible: should be "linux,spdif-dit". - -Example node: - - codec: spdif-transmitter { - compatible = "linux,spdif-dit"; - }; -- 2.31.1

2 1

[PATCH] dt-bindings: Remove "status" from schema examples
by Rob Herring 24 Jul '21

24 Jul '21

There's no reason to have "status" properties in examples. "okay" is the default, and "disabled" turns off some schema checks ('required' specifically). Enabling qca,ar71xx causes a warning, so let's fix the node names: Documentation/devicetree/bindings/net/qca,ar71xx.example.dt.yaml: phy@3: '#phy-cells' is a required property From schema: schemas/phy/phy-provider.yaml Cc: Maxime Ripard <mripard(a)kernel.org> Cc: Chen-Yu Tsai <wens(a)csie.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Sam Ravnborg <sam(a)ravnborg.org> Cc: Rui Miguel Silva <rmfrfs(a)gmail.com> Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: Mauro Carvalho Chehab <mchehab(a)kernel.org> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Robert Marko <robert.marko(a)sartura.hr> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Alessandro Zummo <a.zummo(a)towertech.it> Cc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Cc: Ramesh Shanmugasundaram <rashanmu(a)gmail.com> Cc: "G. Jaya Kumaran" <vineetha.g.jaya.kumaran(a)intel.com> Cc: Linus Walleij <linus.walleij(a)linaro.org> Cc: Oleksij Rempel <o.rempel(a)pengutronix.de> Cc: ChiYuan Huang <cy_huang(a)richtek.com> Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: Dilip Kota <eswara.kota(a)linux.intel.com> Cc: Karol Gugala <kgugala(a)antmicro.com> Cc: Mateusz Holenko <mholenko(a)antmicro.com> Cc: Olivier Moysan <olivier.moysan(a)st.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: dri-devel(a)lists.freedesktop.org Cc: linux-media(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../display/allwinner,sun8i-a83t-dw-hdmi.yaml | 2 -- .../display/panel/boe,tv101wum-nl6.yaml | 1 - .../bindings/media/nxp,imx7-mipi-csi2.yaml | 2 -- .../bindings/media/renesas,drif.yaml | 1 - .../bindings/net/intel,dwmac-plat.yaml | 2 -- .../bindings/net/intel,ixp4xx-ethernet.yaml | 2 -- .../bindings/net/nfc/samsung,s3fwrn5.yaml | 3 --- .../devicetree/bindings/net/qca,ar71xx.yaml | 25 ++++--------------- .../regulator/richtek,rt6245-regulator.yaml | 1 - .../regulator/vqmmc-ipq4019-regulator.yaml | 1 - .../reset/hisilicon,hi3660-reset.yaml | 1 - .../bindings/reset/intel,rcu-gw.yaml | 1 - .../bindings/rtc/microcrystal,rv3032.yaml | 1 - .../soc/litex/litex,soc-controller.yaml | 1 - .../bindings/sound/st,stm32-sai.yaml | 2 -- .../bindings/sound/ti,j721e-cpb-audio.yaml | 2 -- .../sound/ti,j721e-cpb-ivi-audio.yaml | 2 -- 17 files changed, 5 insertions(+), 45 deletions(-) diff --git a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml index 5d42d36608d9..4951b5ef5c6a 100644 --- a/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/allwinner,sun8i-a83t-dw-hdmi.yaml @@ -174,7 +174,6 @@ examples: phy-names = "phy"; pinctrl-names = "default"; pinctrl-0 = <&hdmi_pins>; - status = "disabled"; ports { #address-cells = <1>; @@ -233,7 +232,6 @@ examples: phy-names = "phy"; pinctrl-names = "default"; pinctrl-0 = <&hdmi_pins>; - status = "disabled"; ports { #address-cells = <1>; diff --git a/Documentation/devicetree/bindings/display/panel/boe,tv101wum-nl6.yaml b/Documentation/devicetree/bindings/display/panel/boe,tv101wum-nl6.yaml index 38bc1d1b511e..b87a2e28c866 100644 --- a/Documentation/devicetree/bindings/display/panel/boe,tv101wum-nl6.yaml +++ b/Documentation/devicetree/bindings/display/panel/boe,tv101wum-nl6.yaml @@ -70,7 +70,6 @@ examples: avee-supply = <&ppvarp_lcd>; pp1800-supply = <&pp1800_lcd>; backlight = <&backlight_lcd0>; - status = "okay"; port { panel_in: endpoint { remote-endpoint = <&dsi_out>; diff --git a/Documentation/devicetree/bindings/media/nxp,imx7-mipi-csi2.yaml b/Documentation/devicetree/bindings/media/nxp,imx7-mipi-csi2.yaml index 7c09eec78ce5..877183cf4278 100644 --- a/Documentation/devicetree/bindings/media/nxp,imx7-mipi-csi2.yaml +++ b/Documentation/devicetree/bindings/media/nxp,imx7-mipi-csi2.yaml @@ -200,8 +200,6 @@ examples: clock-names = "pclk", "wrap", "phy", "axi"; power-domains = <&mipi_pd>; - status = "disabled"; - ports { #address-cells = <1>; #size-cells = <0>; diff --git a/Documentation/devicetree/bindings/media/renesas,drif.yaml b/Documentation/devicetree/bindings/media/renesas,drif.yaml index 2867d11fe156..9403b235e976 100644 --- a/Documentation/devicetree/bindings/media/renesas,drif.yaml +++ b/Documentation/devicetree/bindings/media/renesas,drif.yaml @@ -242,7 +242,6 @@ examples: power-domains = <&sysc R8A7795_PD_ALWAYS_ON>; resets = <&cpg 513>; renesas,bonding = <&drif11>; - status = "disabled"; }; drif11: rif@e6f70000 { diff --git a/Documentation/devicetree/bindings/net/intel,dwmac-plat.yaml b/Documentation/devicetree/bindings/net/intel,dwmac-plat.yaml index c1948ce00081..79fa04f5e40d 100644 --- a/Documentation/devicetree/bindings/net/intel,dwmac-plat.yaml +++ b/Documentation/devicetree/bindings/net/intel,dwmac-plat.yaml @@ -116,8 +116,6 @@ examples: snps,mtl-rx-config = <&mtl_rx_setup>; snps,mtl-tx-config = <&mtl_tx_setup>; snps,tso; - status = "okay"; - mdio0 { #address-cells = <1>; #size-cells = <0>; diff --git a/Documentation/devicetree/bindings/net/intel,ixp4xx-ethernet.yaml b/Documentation/devicetree/bindings/net/intel,ixp4xx-ethernet.yaml index f2e91d1bf7d7..378ed2d3b003 100644 --- a/Documentation/devicetree/bindings/net/intel,ixp4xx-ethernet.yaml +++ b/Documentation/devicetree/bindings/net/intel,ixp4xx-ethernet.yaml @@ -71,7 +71,6 @@ examples: ethernet@c8009000 { compatible = "intel,ixp4xx-ethernet"; reg = <0xc8009000 0x1000>; - status = "disabled"; queue-rx = <&qmgr 4>; queue-txready = <&qmgr 21>; intel,npe-handle = <&npe 1>; @@ -82,7 +81,6 @@ examples: ethernet@c800c000 { compatible = "intel,ixp4xx-ethernet"; reg = <0xc800c000 0x1000>; - status = "disabled"; queue-rx = <&qmgr 3>; queue-txready = <&qmgr 20>; intel,npe-handle = <&npe 2>; diff --git a/Documentation/devicetree/bindings/net/nfc/samsung,s3fwrn5.yaml b/Documentation/devicetree/bindings/net/nfc/samsung,s3fwrn5.yaml index 081742c2b726..64995cbb0f97 100644 --- a/Documentation/devicetree/bindings/net/nfc/samsung,s3fwrn5.yaml +++ b/Documentation/devicetree/bindings/net/nfc/samsung,s3fwrn5.yaml @@ -90,14 +90,11 @@ examples: # UART example on Raspberry Pi - | uart0 { - status = "okay"; - nfc { compatible = "samsung,s3fwrn82"; en-gpios = <&gpio 20 GPIO_ACTIVE_HIGH>; wake-gpios = <&gpio 16 GPIO_ACTIVE_HIGH>; - status = "okay"; }; }; diff --git a/Documentation/devicetree/bindings/net/qca,ar71xx.yaml b/Documentation/devicetree/bindings/net/qca,ar71xx.yaml index f0db22645d73..cf4d35edaa1b 100644 --- a/Documentation/devicetree/bindings/net/qca,ar71xx.yaml +++ b/Documentation/devicetree/bindings/net/qca,ar71xx.yaml @@ -101,8 +101,6 @@ examples: phy-mode = "gmii"; - status = "disabled"; - fixed-link { speed = <1000>; full-duplex; @@ -148,32 +146,24 @@ examples: reg = <0x1>; phy-handle = <&phy_port0>; phy-mode = "internal"; - - status = "disabled"; }; switch_port2: port@2 { reg = <0x2>; phy-handle = <&phy_port1>; phy-mode = "internal"; - - status = "disabled"; }; switch_port3: port@3 { reg = <0x3>; phy-handle = <&phy_port2>; phy-mode = "internal"; - - status = "disabled"; }; switch_port4: port@4 { reg = <0x4>; phy-handle = <&phy_port3>; phy-mode = "internal"; - - status = "disabled"; }; }; @@ -183,34 +173,29 @@ examples: interrupt-parent = <&switch10>; - phy_port0: phy@0 { + phy_port0: ethernet-phy@0 { reg = <0x0>; interrupts = <0>; - status = "disabled"; }; - phy_port1: phy@1 { + phy_port1: ethernet-phy@1 { reg = <0x1>; interrupts = <0>; - status = "disabled"; }; - phy_port2: phy@2 { + phy_port2: ethernet-phy@2 { reg = <0x2>; interrupts = <0>; - status = "disabled"; }; - phy_port3: phy@3 { + phy_port3: ethernet-phy@3 { reg = <0x3>; interrupts = <0>; - status = "disabled"; }; - phy_port4: phy@4 { + phy_port4: ethernet-phy@4 { reg = <0x4>; interrupts = <0>; - status = "disabled"; }; }; }; diff --git a/Documentation/devicetree/bindings/regulator/richtek,rt6245-regulator.yaml b/Documentation/devicetree/bindings/regulator/richtek,rt6245-regulator.yaml index 796ceac87445..e983d0e70c9b 100644 --- a/Documentation/devicetree/bindings/regulator/richtek,rt6245-regulator.yaml +++ b/Documentation/devicetree/bindings/regulator/richtek,rt6245-regulator.yaml @@ -77,7 +77,6 @@ examples: rt6245@34 { compatible = "richtek,rt6245"; - status = "okay"; reg = <0x34>; enable-gpios = <&gpio26 2 0>; diff --git a/Documentation/devicetree/bindings/regulator/vqmmc-ipq4019-regulator.yaml b/Documentation/devicetree/bindings/regulator/vqmmc-ipq4019-regulator.yaml index 6f45582c914e..dd7a2f92634c 100644 --- a/Documentation/devicetree/bindings/regulator/vqmmc-ipq4019-regulator.yaml +++ b/Documentation/devicetree/bindings/regulator/vqmmc-ipq4019-regulator.yaml @@ -39,6 +39,5 @@ examples: regulator-min-microvolt = <1500000>; regulator-max-microvolt = <3000000>; regulator-always-on; - status = "disabled"; }; ... diff --git a/Documentation/devicetree/bindings/reset/hisilicon,hi3660-reset.yaml b/Documentation/devicetree/bindings/reset/hisilicon,hi3660-reset.yaml index 9bf40952e5b7..b0c41ab1a746 100644 --- a/Documentation/devicetree/bindings/reset/hisilicon,hi3660-reset.yaml +++ b/Documentation/devicetree/bindings/reset/hisilicon,hi3660-reset.yaml @@ -72,6 +72,5 @@ examples: resets = <&iomcu_rst 0x20 3>; pinctrl-names = "default"; pinctrl-0 = <&i2c0_pmx_func &i2c0_cfg_func>; - status = "disabled"; }; ... diff --git a/Documentation/devicetree/bindings/reset/intel,rcu-gw.yaml b/Documentation/devicetree/bindings/reset/intel,rcu-gw.yaml index 6b2d56cc3f38..13bf6bb3f097 100644 --- a/Documentation/devicetree/bindings/reset/intel,rcu-gw.yaml +++ b/Documentation/devicetree/bindings/reset/intel,rcu-gw.yaml @@ -57,7 +57,6 @@ examples: }; pwm: pwm@e0d00000 { - status = "disabled"; compatible = "intel,lgm-pwm"; reg = <0xe0d00000 0x30>; clocks = <&cgu0 1>; diff --git a/Documentation/devicetree/bindings/rtc/microcrystal,rv3032.yaml b/Documentation/devicetree/bindings/rtc/microcrystal,rv3032.yaml index a2c55303810d..9593840a4a2b 100644 --- a/Documentation/devicetree/bindings/rtc/microcrystal,rv3032.yaml +++ b/Documentation/devicetree/bindings/rtc/microcrystal,rv3032.yaml @@ -53,7 +53,6 @@ examples: rtc@51 { compatible = "microcrystal,rv3032"; reg = <0x51>; - status = "okay"; pinctrl-0 = <&rtc_nint_pins>; interrupts-extended = <&gpio1 16 IRQ_TYPE_LEVEL_HIGH>; trickle-resistor-ohms = <7000>; diff --git a/Documentation/devicetree/bindings/soc/litex/litex,soc-controller.yaml b/Documentation/devicetree/bindings/soc/litex/litex,soc-controller.yaml index c8b57c7fd08c..ecae9fa8561b 100644 --- a/Documentation/devicetree/bindings/soc/litex/litex,soc-controller.yaml +++ b/Documentation/devicetree/bindings/soc/litex/litex,soc-controller.yaml @@ -35,7 +35,6 @@ examples: soc_ctrl0: soc-controller@f0000000 { compatible = "litex,soc-controller"; reg = <0xf0000000 0xc>; - status = "okay"; }; ... diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index 06e83461705c..f97132400bb6 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -180,7 +180,6 @@ examples: pinctrl-names = "default", "sleep"; pinctrl-0 = <&sai2a_pins_a>, <&sai2b_pins_b>; pinctrl-1 = <&sai2a_sleep_pins_a>, <&sai2b_sleep_pins_b>; - status = "okay"; sai2a: audio-controller@4400b004 { #sound-dai-cells = <0>; @@ -190,7 +189,6 @@ examples: dma-names = "tx"; clocks = <&rcc SAI2_K>; clock-names = "sai_ck"; - status = "okay"; }; }; diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml index ec06789b21df..6806f53a4aed 100644 --- a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-audio.yaml @@ -127,8 +127,6 @@ examples: compatible = "ti,j721e-cpb-audio"; model = "j721e-cpb"; - status = "okay"; - ti,cpb-mcasp = <&mcasp10>; ti,cpb-codec = <&pcm3168a_1>; diff --git a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml index ee9f960de36b..859d369c71e2 100644 --- a/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml +++ b/Documentation/devicetree/bindings/sound/ti,j721e-cpb-ivi-audio.yaml @@ -119,8 +119,6 @@ examples: compatible = "ti,j721e-cpb-ivi-audio"; model = "j721e-cpb-ivi"; - status = "okay"; - ti,cpb-mcasp = <&mcasp10>; ti,cpb-codec = <&pcm3168a_1>; -- 2.27.0

9 10

kbl-r5514-5663-max Headphone does not work
by GitHub issues - opened 23 Jul '21

23 Jul '21

alsa-project/alsa-ucm-conf issue #108 was opened from jmontleon: On a Pixelbook I have copied these firmware files in place: - /opt/google/dsm/dsmparam.bin - /lib/firmware/9d71-GOOGLE-EVEMAX-0-tplg.bin - /lib/firmware/dsp_lib_dsm_core_spt_release.bin - /lib/firmware/intel/dsp_fw_C75061F3-F2B2-4DCC-8F9F-82ABB4131E66.bin With that alone the speakers started working after a reboot. I have been trying to modify the ucm HiFi.conf to ucm2 on Fedora. If I comment out all of `SectionDevice."Headphones"` the speakers and internal microphone work. If I leave it I see a bunch of error messages in dmesg. Pulseaudio also produces some error messages and then just loads a dummy device. dmesg: ``` [ 7866.107942] Kbl Audio Headset Playback: ASoC: no backend DAIs enabled for Kbl Audio Headset Playback [ 7866.107950] Kbl Audio Headset Playback: ASoC: dpcm_fe_dai_prepare() failed (-22) ``` pulseaudio: ``` D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 21845 ms I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) D: [pulseaudio] alsa-util.c: Set neither period nor buffer size. I: [pulseaudio] (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_PREPARE failed (-22) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Invalid argument D: [pulseaudio] alsa-util.c: Trying plug:_ucm0006.hw:kblr55145663max,2 with SND_PCM_NO_AUTO_FORMAT ... I: [pulseaudio] (alsa-lib)conf.c: Unknown parameter 1 I: [pulseaudio] (alsa-lib)conf.c: Parse arguments error: No such file or directory I: [pulseaudio] (alsa-lib)pcm.c: Unknown PCM plug:_ucm0006.hw:kblr55145663max,2 I: [pulseaudio] alsa-util.c: Error opening PCM device plug:_ucm0006.hw:kblr55145663max,2: No such file or directory D: [pulseaudio] alsa-mixer.c: Profile set 0x555cededf730, auto_profiles=no, probed=yes, n_mappings=0, n_profiles=0, n_decibel_fixes=0 E: [pulseaudio] module-alsa-card.c: Failed to find a working profile. E: [pulseaudio] module.c: Failed to load module "module-alsa-card" (argument: "device_id="0" name="platform-kbl_r5514_5663_max" card_name="alsa_card.platform-kbl_r5514_5663_max" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=yes use_ucm=yes avoid_resampling=no card_properties="module-udev-detect.discovered=1""): initialization failed. I: [pulseaudio] module-udev-detect.c: Card /devices/platform/kbl_r5514_5663_max/sound/card0 (alsa_card.platform-kbl_r5514_5663_max) failed to load module. ``` >From reading I gather there's probably some routing incorrectly or incompletely set up that is the cause, but I'm at a bit of a loss on where to go from here. I saw a suggestion somewhere to set this module option, but as far as I can tell it makes no difference: ``` cat /etc/modprobe.d/dsp.conf options snd_intel_dspcfg dsp_driver=2 ``` Not really an alsa / ucm problem as far as I am aware but in the interest of giving as much info as possible I originally switched to pulseaudio because I had an easier time finding meaningful logs. I switched back to pipewire later to realize the mic is just producing awful noise so I went back to pulseaudio for the time being. Might need to file a pipewire bug later. The HiFi.conf and alsa-info are attached. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/6870590/… [Hifi.conf.txt](https://github.com/alsa-project/alsa-ucm-conf/files/6870593/… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/108 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 1/2] ASoC: amd: Don't show messages about deferred probing by default
by Mario Limonciello 23 Jul '21

23 Jul '21

Nearly every boot with a Lenovo P14s is showing acp_pdm_mach acp_pdm_mach.0: snd_soc_register_card(acp) failed: -517 This isn't useful to a user, especially as probing will run again. Use the dev_err_probe helper to hide the deferrerd probing messages. CC: markpearson(a)lenovo.com Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/renoir/acp3x-rn.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) v1->v2: * Adjust return codes v2->v3: * Use deferred probing helper * Rebase on asoc-next/for-5.15 diff --git a/sound/soc/amd/renoir/acp3x-rn.c b/sound/soc/amd/renoir/acp3x-rn.c index 306134b89a82..5d979a7b77fb 100644 --- a/sound/soc/amd/renoir/acp3x-rn.c +++ b/sound/soc/amd/renoir/acp3x-rn.c @@ -54,10 +54,9 @@ static int acp_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, - "snd_soc_register_card(%s) failed: %d\n", - acp_card.name, ret); - return ret; + return dev_err_probe(&pdev->dev, ret, + "snd_soc_register_card(%s) failed\n", + card->name); } return 0; } -- 2.25.1

3 5

tlv320aic3x audio driver -- possible bug
by Frederick Gotham 23 Jul '21

23 Jul '21

In the C source file "tlv320aic3x" in the function "aic3x_set_power", when the argument for power is 0, the hardware registers are saved to the Linux computer as follows: regcache_cache_only(aic3x->regmap, true); And then later when "aic3x_set_power" is called with power set to 1, the previous values are then restored back to their hardware registers: regcache_cache_only(aic3x->regmap, false); regcache_sync(aic3x->regmap); When the device is powered back on, the previous setting for microphone bias (MICBIAS) is lost. If it was previously set to 2 volts or 2.5 volts, then when it's powered back on it becomes Off. I'm not very familiar with audio codec chips and how they work, but I think maybe the order of function calls in the "else" branch might be incorrect. Right now the code looks like this: /* * Do soft reset to this codec instance in order to clear * possible VDD leakage currents in case the supply regulators * remain on */ snd_soc_component_write(component, AIC3X_RESET, SOFT_RESET); regcache_mark_dirty(aic3x->regmap); aic3x->power = 0; /* HW writes are needless when bias is off */ regcache_cache_only(aic3x->regmap, true); ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies); However I think that the call to "regcache_cache_only" should come before the soft reset takeS place. The setting for MICBIAS is lost at this line: snd_soc_component_write(component, AIC3X_RESET, SOFT_RESET); And so I think that the call to "regcache_cache_only" should be moved up above "snd_soc_component_write", as follows: /* * Do soft reset to this codec instance in order to clear * possible VDD leakage currents in case the supply regulators * remain on */ regcache_cache_only(aic3x->regmap, true); snd_soc_component_write(component, AIC3X_RESET, SOFT_RESET); regcache_mark_dirty(aic3x->regmap); aic3x->power = 0; ret = regulator_bulk_disable(ARRAY_SIZE(aic3x->supplies), aic3x->supplies);

2 1

[PATCH 0/6] soundwire/ASoC: abstract platform-dependent bases
by Bard Liao 23 Jul '21

23 Jul '21

shim base and alh base are platform-dependent. This series suggests to use variables for those bases. It allows us to use different bases for new platforms. Bard Liao (6): soundwire: move intel sdw register definitions to sdw_intel.h ASoC: SOF: intel: add sdw_shim/alh_base to sof_intel_dsp_desc ASoC: SOF: intel: hda: remove HDA_DSP_REG_SNDW_WAKE_STS definition ASoC: SOF: intel: move sof_intel_dsp_desc() forward ASoC: SOF: intel: add snd_sof_dsp_check_sdw_irq ops soundwire: intel: introduce shim and alh base drivers/soundwire/intel.c | 74 ------------------------ drivers/soundwire/intel_init.c | 14 ++--- include/linux/soundwire/sdw_intel.h | 87 +++++++++++++++++++++++++++++ sound/soc/sof/intel/cnl.c | 6 ++ sound/soc/sof/intel/hda.c | 39 ++++++++----- sound/soc/sof/intel/hda.h | 8 ++- sound/soc/sof/intel/icl.c | 3 + sound/soc/sof/intel/shim.h | 3 + sound/soc/sof/intel/tgl.c | 12 ++++ 9 files changed, 149 insertions(+), 97 deletions(-) -- 2.17.1

3 8

[PATCH 0/2] ASoC/soundwire: add quirks for Intel 'Bishop County' NUC M15
by Pierre-Louis Bossart 23 Jul '21

23 Jul '21

The 'Bishop County' NUC Laptops designed by Intel rely on SoundWire peripherals. Quirks are required on the soundwire/ side to work-around ACPI issues (bad _ADR) and sound/soc/ side for jack detection. The two patches are independent and can be merged in the two subsystem independently if desired. Both patches are however required in any backport. Thanks to Gilles van Eeden for reporting this issue and testing these patches. Pierre-Louis Bossart (2): soundwire: dmi-quirks: add quirk for Intel 'Bishop County' NUC M15 ASOC: Intel: sof_sdw: add quirk for Intel 'Bishop County' NUC M15 drivers/soundwire/dmi-quirks.c | 16 ++++++++++++---- sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 23 insertions(+), 4 deletions(-) -- 2.25.1

3 8

sof-hda-dsp: Set Capture Switch on in the BootSequence
by GitHub pull_request - opened 23 Jul '21

23 Jul '21

alsa-project/alsa-ucm-conf pull request #107 was opened from jason77-wang: We found an issue that the PA source of Mic2/Headset and Mic2 is muted by default after newly install an OS, the root cause is the 'Capture Switch' is set to off in the kernel. Without ucm, the /usr/share/alsa/init/default will set the 'Capture Switch' to on, similarly we set it to on in the BootSequence of ucm. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/107 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/107.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ASRock (A320M-HDV R4.0) No analog audio driver detected -
by GitHub issues - opened 22 Jul '21

22 Jul '21

alsa-project/alsa-lib issue #165 was opened from PedePanoOriginal: Hello guys! I just got a new PC and I was so happy about it. It's a Ryzen 5 - 3600, GeForce GT 1030 GPU, the motherboard is a ASRock A320M-HDV R4.0. The first thing I did when I pluged it in was installing Pop!_OS 20.04 LTS. Everything worked fine except my audio analog ports. They are simply not detected by OS, the only output audio device detected is "GP108 High Definition Audio Controller Digital Stereo". I've done a lot of googling, tried everything I could find, but, no success. I noticed that when I turn on the PC a message is displayed, something like that: "snd_hda_intel 0000:23:00.4: no codecs found". This message disappear if I turn off onboard audio stuff at the bios. I would love some help if possible. I'm a linux noob, so I'm sorry to not be able to provide better information now but, I can do it later with some help. Issue URL : https://github.com/alsa-project/alsa-lib/issues/165 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: codecs: wcd938x: fix wcd module dependency
by Arnd Bergmann 22 Jul '21

22 Jul '21

From: Arnd Bergmann <arnd(a)arndb.de> With SND_SOC_ALL_CODECS=y and SND_SOC_WCD938X_SDW=m, there is a link error from a reverse dependency, since the built-in codec driver calls into the modular soundwire back-end: x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_free': wcd938x.c:(.text+0x2c0): undefined reference to `wcd938x_sdw_free' x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_hw_params': wcd938x.c:(.text+0x2f6): undefined reference to `wcd938x_sdw_hw_params' x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_codec_set_sdw_stream': wcd938x.c:(.text+0x332): undefined reference to `wcd938x_sdw_set_sdw_stream' x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_tx_swr_ctrl': wcd938x.c:(.text+0x23de): undefined reference to `wcd938x_swr_get_current_bank' x86_64-linux-ld: sound/soc/codecs/wcd938x.o: in function `wcd938x_bind': wcd938x.c:(.text+0x2579): undefined reference to `wcd938x_sdw_device_get' x86_64-linux-ld: wcd938x.c:(.text+0x25a1): undefined reference to `wcd938x_sdw_device_get' x86_64-linux-ld: wcd938x.c:(.text+0x262a): undefined reference to `__devm_regmap_init_sdw' Work around this using two small hacks: An added Kconfig dependency prevents the main driver from being built-in when soundwire support itself is a loadable module to allow calling devm_regmap_init_sdw(), and a Makefile trick links the wcd938x-sdw backend as built-in if needed to solve the dependency between the two modules. Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- I saw this on -rc2, please ignore if this is already fixed --- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/Makefile | 5 ++++- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ebae3f09435..459deba31485 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1558,6 +1558,7 @@ config SND_SOC_WCD934X config SND_SOC_WCD938X tristate + depends on SOUNDWIRE || !SOUNDWIRE config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index de8b83dd2c76..7bb38c370842 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -583,7 +583,10 @@ obj-$(CONFIG_SND_SOC_WCD_MBHC) += snd-soc-wcd-mbhc.o obj-$(CONFIG_SND_SOC_WCD9335) += snd-soc-wcd9335.o obj-$(CONFIG_SND_SOC_WCD934X) += snd-soc-wcd934x.o obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x.o -obj-$(CONFIG_SND_SOC_WCD938X_SDW) += snd-soc-wcd938x-sdw.o +ifdef CONFIG_SND_SOC_WCD938X_SDW +# avoid link failure by forcing sdw code built-in when needed +obj-$(CONFIG_SND_SOC_WCD938X) += snd-soc-wcd938x-sdw.o +endif obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o -- 2.29.2

2 1

[PATCH] ASoC: tlv320aic32x4: Fix TAS2505/TAS2521 processing block selection
by Marek Vasut 22 Jul '21

22 Jul '21

The TAS2505/TAS2521 does support only three processing block options, unlike TLV320AIC32x4 which supports 25. This is documented in TI slau472 2.5.1.2 Processing Blocks and Page 0 / Register 60: DAC Instruction Set - 0x00 / 0x3C. Limit the Processing Blocks maximum value to 3 on TAS2505/TAS2521 and select processing block PRB_P1 always, because for the configuration of teh codec implemented in this driver, this is the best quality option. Fixes: b4525b6196cd7 ("ASoC: tlv320aic32x4: add support for TAS2505") Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Claudius Heine <ch(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic32x4.c | 33 +++++++++++++++++++++++++------- 1 file changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c51267ce67cf..4d80879dd144 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -682,11 +682,20 @@ static int aic32x4_set_dosr(struct snd_soc_component *component, u16 dosr) static int aic32x4_set_processing_blocks(struct snd_soc_component *component, u8 r_block, u8 p_block) { - if (r_block > 18 || p_block > 25) - return -EINVAL; + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); + + if (aic32x4->type == AIC32X4_TYPE_TAS2505) { + if (r_block || p_block > 3) + return -EINVAL; - snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); - snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + } else { /* AIC32x4 */ + if (r_block > 18 || p_block > 25) + return -EINVAL; + + snd_soc_component_write(component, AIC32X4_ADCSPB, r_block); + snd_soc_component_write(component, AIC32X4_DACSPB, p_block); + } return 0; } @@ -695,6 +704,7 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, unsigned int sample_rate, unsigned int channels, unsigned int bit_depth) { + struct aic32x4_priv *aic32x4 = snd_soc_component_get_drvdata(component); u8 aosr; u16 dosr; u8 adc_resource_class, dac_resource_class; @@ -721,19 +731,28 @@ static int aic32x4_setup_clocks(struct snd_soc_component *component, adc_resource_class = 6; dac_resource_class = 8; dosr_increment = 8; - aic32x4_set_processing_blocks(component, 1, 1); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 1, 1); } else if (sample_rate <= 96000) { aosr = 64; adc_resource_class = 6; dac_resource_class = 8; dosr_increment = 4; - aic32x4_set_processing_blocks(component, 1, 9); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 1, 9); } else if (sample_rate == 192000) { aosr = 32; adc_resource_class = 3; dac_resource_class = 4; dosr_increment = 2; - aic32x4_set_processing_blocks(component, 13, 19); + if (aic32x4->type == AIC32X4_TYPE_TAS2505) + aic32x4_set_processing_blocks(component, 0, 1); + else + aic32x4_set_processing_blocks(component, 13, 19); } else { dev_err(component->dev, "Sampling rate not supported\n"); return -EINVAL; -- 2.30.2

2 1

[PATCH] ASoC: rt5682: Adjust headset volume button threshold
by derek.fang＠realtek.com 22 Jul '21

22 Jul '21

From: Derek Fang <derek.fang(a)realtek.com> Adjust the threshold of headset button volume+ to fix the wrong button detection issue with some brand headsets. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index abcd6f483788..51ecaa2abcd1 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -44,6 +44,7 @@ static const struct reg_sequence patch_list[] = { {RT5682_I2C_CTRL, 0x000f}, {RT5682_PLL2_INTERNAL, 0x8266}, {RT5682_SAR_IL_CMD_3, 0x8365}, + {RT5682_SAR_IL_CMD_6, 0x0180}, }; void rt5682_apply_patch_list(struct rt5682_priv *rt5682, struct device *dev) -- 2.17.1

2 1

[PATCH] ASoC: amd: enable stop_dma_first flag for cz_dai_7219_98357 dai link
by Vijendar Mukunda 22 Jul '21

22 Jul '21

DMA driver stop sequence should be invoked first before invoking I2S controller driver stop sequence for Stoneyridge platform. Enable stop_dma_first flag for cz_dai_7219_98357 dai link structure. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 9449fb40a956..3c60c5f96dcb 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -525,6 +525,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_da7219_play_ops, SND_SOC_DAILINK_REG(designware1, dlgs, platform), }, @@ -534,6 +535,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_da7219_cap_ops, SND_SOC_DAILINK_REG(designware2, dlgs, platform), }, @@ -543,6 +545,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_playback = 1, + .stop_dma_first = 1, .ops = &cz_max_play_ops, SND_SOC_DAILINK_REG(designware3, mx, platform), }, @@ -553,6 +556,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_dmic0_cap_ops, SND_SOC_DAILINK_REG(designware3, adau, platform), }, @@ -563,6 +567,7 @@ static struct snd_soc_dai_link cz_dai_7219_98357[] = { .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, + .stop_dma_first = 1, .ops = &cz_dmic1_cap_ops, SND_SOC_DAILINK_REG(designware2, adau, platform), }, -- 2.17.1

2 1

[PATCH v2] ASoC: amd: Don't show messages about deferred probing by default
by Mario Limonciello 22 Jul '21

22 Jul '21

Nearly every boot with a Lenovo P14s is showing acp_pdm_mach acp_pdm_mach.0: snd_soc_register_card(acp) failed: -517 This isn't useful to a user, especially as probing will run again. Downgrade the message to debug, and only show errors. CC: markpearson(a)lenovo.com Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/renoir/acp3x-rn.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) Changes from v1->v2: * Simplify down to one return line. diff --git a/sound/soc/amd/renoir/acp3x-rn.c b/sound/soc/amd/renoir/acp3x-rn.c index 306134b89a82..ed73da496744 100644 --- a/sound/soc/amd/renoir/acp3x-rn.c +++ b/sound/soc/amd/renoir/acp3x-rn.c @@ -54,12 +54,16 @@ static int acp_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, - "snd_soc_register_card(%s) failed: %d\n", - acp_card.name, ret); - return ret; + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card(%s) failed: %d\n", + card->name, ret); + else + dev_dbg(&pdev->dev, + "snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); } - return 0; + return ret; } static struct platform_driver acp_mach_driver = { -- 2.25.1

3 5

[PATCH v5 00/12] Add Vangogh ACP ASoC driver
by Vijendar Mukunda 22 Jul '21

22 Jul '21

This adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Vangogh APU. Changes since v4: - Fixed indentaion issue - Fixed error handling - reverted hex notation for macros - Fixed typo error in Kconfig file Changes since v3: - Fixed typo errors - Fixed platform devices unregister logic - Updated acp resources count with macro value - updated ACP5x_I2S_MODE macro value to avoid unregistering platform devices for other ACP audio configurations in driver remove callback - Fixed error handling - Fixed indentation issues in irq handler - Add __maybe_unused for PM functions Changes since v2: - remove extra stuff from Make file Changes since v1: - add comment for acp_readl() & acp_writel() API's in common header file. - add amd: tag in commit message - modify condition check in acp5x_power_on() - removed acp_power_gating flag related code - drop acp5x_audio_remove() API - use single flag for handling irq status - fixed indetation issue and used dev_err instead of pr_err - drop acp5x_dai_remove() API - use SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK - use PM related Macros in ACP PCI driver - use PM related Macros in ACP DMA driver - remove extra line in Make file Vijendar Mukunda (12): ASoC: amd: add Vangogh ACP5x IP register header ASoC: amd: add Vangogh ACP PCI driver ASoc: amd: add acp5x init/de-init functions ASoC: amd: create acp5x platform devices ASoC: amd: add ACP5x PCM platform driver ASoC: amd: irq handler changes for ACP5x PCM dma driver ASoC: amd: add ACP5x pcm dma driver ops ASoC: amd: add vangogh i2s controller driver ASoC: amd: add vangogh i2s dai driver ops ASoC: amd: add vangogh pci driver pm ops ASoC: amd: add vangogh i2s dma driver pm ops ASoC: amd: enable vangogh acp5x driver build sound/soc/amd/Kconfig | 9 + sound/soc/amd/Makefile | 1 + sound/soc/amd/vangogh/Makefile | 9 + sound/soc/amd/vangogh/acp5x-i2s.c | 427 ++++++++++++++++ sound/soc/amd/vangogh/acp5x-pcm-dma.c | 525 ++++++++++++++++++++ sound/soc/amd/vangogh/acp5x.h | 193 +++++++ sound/soc/amd/vangogh/pci-acp5x.c | 326 ++++++++++++ sound/soc/amd/vangogh/vg_chip_offset_byte.h | 337 +++++++++++++ 8 files changed, 1827 insertions(+) create mode 100644 sound/soc/amd/vangogh/Makefile create mode 100644 sound/soc/amd/vangogh/acp5x-i2s.c create mode 100644 sound/soc/amd/vangogh/acp5x-pcm-dma.c create mode 100644 sound/soc/amd/vangogh/acp5x.h create mode 100644 sound/soc/amd/vangogh/pci-acp5x.c create mode 100644 sound/soc/amd/vangogh/vg_chip_offset_byte.h -- 2.17.1

2 13

[PATCH v3 0/2] Cirrus Logic CS35L41 Amplifier
by David Rhodes 22 Jul '21

22 Jul '21

ASoC driver and devicetree documentation for a new Cirrus Logic amplifier CS35L41 v3 changes: Removed redundant info from license header Alphabetized includes Changed comment on ACPI ID to reflect that vendor portion is a PnP ID Removed some unnecessary initializations Changed some uses of (x == NULL) to (!x) David Rhodes (2): ASoC: cs35l41: CS35L41 Boosted Smart Amplifier ASoC: cs35l41: Add bindings for CS35L41 .../devicetree/bindings/sound/cs35l41.yaml | 395 ++++ include/sound/cs35l41.h | 79 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs35l41-i2c.c | 120 ++ sound/soc/codecs/cs35l41-spi.c | 147 ++ sound/soc/codecs/cs35l41-tables.c | 612 ++++++ sound/soc/codecs/cs35l41.c | 1765 +++++++++++++++++ sound/soc/codecs/cs35l41.h | 750 +++++++ 9 files changed, 3884 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs35l41.yaml create mode 100644 include/sound/cs35l41.h create mode 100644 sound/soc/codecs/cs35l41-i2c.c create mode 100644 sound/soc/codecs/cs35l41-spi.c create mode 100644 sound/soc/codecs/cs35l41-tables.c create mode 100644 sound/soc/codecs/cs35l41.c create mode 100644 sound/soc/codecs/cs35l41.h -- 2.25.1

3 11

[PATCH] soundwire: bus: update Slave status in sdw_clear_slave_status
by Bard Liao 22 Jul '21

22 Jul '21

Call to sdw_update_slave_status() needs to be added to sdw_clear_slave_ status() to ensure Slaves are informed of the new status via update_ status() callback. This will enable codec drivers to reset their internal states and make sure the register settings are properly restored on pm_runtime or system resume BugLink: https://github.com/thesofproject/linux/issues/2908 BugLink: https://github.com/thesofproject/linux/issues/2637 Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> --- drivers/soundwire/bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index baa236cb5484..1b115734a8f6 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1860,6 +1860,7 @@ void sdw_clear_slave_status(struct sdw_bus *bus, u32 request) if (slave->status != SDW_SLAVE_UNATTACHED) { sdw_modify_slave_status(slave, SDW_SLAVE_UNATTACHED); slave->first_interrupt_done = false; + sdw_update_slave_status(slave, SDW_SLAVE_UNATTACHED); } /* keep track of request, used in pm_runtime resume */ -- 2.17.1

2 1

[PATCH] soundwire: cadence: Remove ret variable from sdw_cdns_irq()
by Bard Liao 22 Jul '21

22 Jul '21

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> The ret is not used in the interrupt handler, it is just returned without any condition or change. We can return the IRQ_HANDLED directly. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/cadence_master.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 25950422b085..9a9b6110e763 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -822,7 +822,6 @@ irqreturn_t sdw_cdns_irq(int irq, void *dev_id) { struct sdw_cdns *cdns = dev_id; u32 int_status; - int ret = IRQ_HANDLED; /* Check if the link is up */ if (!cdns->link_up) @@ -900,7 +899,7 @@ irqreturn_t sdw_cdns_irq(int irq, void *dev_id) } cdns_writel(cdns, CDNS_MCP_INTSTAT, int_status); - return ret; + return IRQ_HANDLED; } EXPORT_SYMBOL(sdw_cdns_irq); -- 2.17.1

2 1

[PATCH] soundwire: bus: filter out more -EDATA errors on clock stop
by Bard Liao 22 Jul '21

22 Jul '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> We've added quite a few filters to avoid throwing errors if a Device does not respond to commands during the clock stop sequences, but we missed one. This will lead to an isolated message [ 6115.294412] soundwire sdw-master-1: SDW_SCP_STAT bread failed:-61 The callers already filter this error code, so there's no point in keeping it at the lower level. Since this is a recoverable error, make this dev_err() conditional and only log cases with Command Failed. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 3e6d4addac2f..278a4fbdb88d 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -896,7 +896,8 @@ static int sdw_bus_wait_for_clk_prep_deprep(struct sdw_bus *bus, u16 dev_num) do { val = sdw_bread_no_pm(bus, dev_num, SDW_SCP_STAT); if (val < 0) { - dev_err(bus->dev, "SDW_SCP_STAT bread failed:%d\n", val); + if (val != -ENODATA) + dev_err(bus->dev, "SDW_SCP_STAT bread failed:%d\n", val); return val; } val &= SDW_SCP_STAT_CLK_STP_NF; -- 2.17.1

2 1

[PATCH] soundwire: dmi-quirks: add ull suffix for SoundWire _ADR values
by Bard Liao 22 Jul '21

22 Jul '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Sparse throws the following type of warnings: drivers/soundwire/dmi-quirks.c:25:17: error: constant 0x000010025D070100 is so big it is long Let's add the 'ull' suffix to make this go away and find real issues. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/dmi-quirks.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 5db0a2443a1d..1ac16687e315 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -22,12 +22,12 @@ struct adr_remap { */ static const struct adr_remap hp_spectre_360[] = { { - 0x000010025D070100, - 0x000020025D071100 + 0x000010025D070100ull, + 0x000020025D071100ull }, { - 0x000110025d070100, - 0x000120025D130800 + 0x000110025d070100ull, + 0x000120025D130800ull }, {} }; @@ -39,18 +39,18 @@ static const struct adr_remap hp_spectre_360[] = { static const struct adr_remap dell_sku_0A3E[] = { /* rt715 on link0 */ { - 0x00020025d071100, - 0x00021025d071500 + 0x00020025d071100ull, + 0x00021025d071500ull }, /* rt711 on link1 */ { - 0x000120025d130800, - 0x000120025d071100, + 0x000120025d130800ull, + 0x000120025d071100ull, }, /* rt1308 on link2 */ { - 0x000220025d071500, - 0x000220025d130800 + 0x000220025d071500ull, + 0x000220025d130800ull }, {} }; -- 2.17.1

2 1

[PATCH v2] ASoC: amd: renoir: Run hibernation callbacks
by Mario Limonciello 22 Jul '21

22 Jul '21

The registers need to be re-initialized after hibernation or microphone may be non-functional. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=213793 Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 2 ++ 1 file changed, 2 insertions(+) v1-> v2: * Resend as a separate patch from threaded series diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 19438da5dfa5..7b8040e812a1 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -382,6 +382,8 @@ static const struct dev_pm_ops rn_acp_pm = { .runtime_resume = snd_rn_acp_resume, .suspend = snd_rn_acp_suspend, .resume = snd_rn_acp_resume, + .restore = snd_rn_acp_resume, + .poweroff = snd_rn_acp_suspend, }; static void snd_rn_acp_remove(struct pci_dev *pci) -- 2.25.1

1 0

[PATCH] soundwire: stream: add s_rt into slave_rt_list before sdw_config_stream
by Dongliang Mu 22 Jul '21

22 Jul '21

The commit 48f17f96a817 ("soundwire: stream: fix memory leak in stream config error path") fixes the memory leak by implicitly freeing the s_rt object. However, this fixing style is not very good. The better fix is to move list_add_tail before sdw_config_stream and revert the previous commit. Signed-off-by: Dongliang Mu <mudongliangabcd(a)gmail.com> --- drivers/soundwire/stream.c | 14 +++----------- 1 file changed, 3 insertions(+), 11 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 1a18308f4ef4..66a4ce4f923f 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1373,19 +1373,11 @@ int sdw_stream_add_slave(struct sdw_slave *slave, goto stream_error; } + list_add_tail(&s_rt->m_rt_node, &m_rt->slave_rt_list); + ret = sdw_config_stream(&slave->dev, stream, stream_config, true); - if (ret) { - /* - * sdw_release_master_stream will release s_rt in slave_rt_list in - * stream_error case, but s_rt is only added to slave_rt_list - * when sdw_config_stream is successful, so free s_rt explicitly - * when sdw_config_stream is failed. - */ - kfree(s_rt); + if (ret) goto stream_error; - } - - list_add_tail(&s_rt->m_rt_node, &m_rt->slave_rt_list); ret = sdw_slave_port_config(slave, s_rt, port_config, num_ports); if (ret) -- 2.25.1

2 3

[PATCH v4 0/5] bus: Make remove callback return void
by Uwe Kleine-König 22 Jul '21

22 Jul '21

Hello, this is v4 of the final patch set for my effort to make struct bus_type::remove return void. The first four patches contain cleanups that make some of these callbacks (more obviously) always return 0. They are acked by the respective maintainers. Bjorn Helgaas explicitly asked to include the pci patch (#1) into this series, so Greg taking this is fine. I assume the s390 people are fine with Greg taking patches #2 to #4, too, they didn't explicitly said so though. The last patch actually changes the prototype and so touches quite some drivers and has the potential to conflict with future developments, so I consider it beneficial to put these patches into next soon. I expect that it will be Greg who takes the complete series, he already confirmed via irc (for v2) to look into this series. The only change compared to v3 is in the fourth patch where I modified a few more drivers to fix build failures. Some of them were found by build bots (thanks!), some of them I found myself using a regular expression search. The newly modified files are: arch/sparc/kernel/vio.c drivers/nubus/bus.c drivers/sh/superhyway/superhyway.c drivers/vlynq/vlynq.c drivers/zorro/zorro-driver.c sound/ac97/bus.c Best regards Uwe Uwe Kleine-König (5): PCI: endpoint: Make struct pci_epf_driver::remove return void s390/cio: Make struct css_driver::remove return void s390/ccwgroup: Drop if with an always false condition s390/scm: Make struct scm_driver::remove return void bus: Make remove callback return void arch/arm/common/locomo.c | 3 +-- arch/arm/common/sa1111.c | 4 +--- arch/arm/mach-rpc/ecard.c | 4 +--- arch/mips/sgi-ip22/ip22-gio.c | 3 +-- arch/parisc/kernel/drivers.c | 5 ++--- arch/powerpc/platforms/ps3/system-bus.c | 3 +-- arch/powerpc/platforms/pseries/ibmebus.c | 3 +-- arch/powerpc/platforms/pseries/vio.c | 3 +-- arch/s390/include/asm/eadm.h | 2 +- arch/sparc/kernel/vio.c | 4 +--- drivers/acpi/bus.c | 3 +-- drivers/amba/bus.c | 4 +--- drivers/base/auxiliary.c | 4 +--- drivers/base/isa.c | 4 +--- drivers/base/platform.c | 4 +--- drivers/bcma/main.c | 6 ++---- drivers/bus/sunxi-rsb.c | 4 +--- drivers/cxl/core.c | 3 +-- drivers/dax/bus.c | 4 +--- drivers/dma/idxd/sysfs.c | 4 +--- drivers/firewire/core-device.c | 4 +--- drivers/firmware/arm_scmi/bus.c | 4 +--- drivers/firmware/google/coreboot_table.c | 4 +--- drivers/fpga/dfl.c | 4 +--- drivers/hid/hid-core.c | 4 +--- drivers/hid/intel-ish-hid/ishtp/bus.c | 4 +--- drivers/hv/vmbus_drv.c | 5 +---- drivers/hwtracing/intel_th/core.c | 4 +--- drivers/i2c/i2c-core-base.c | 5 +---- drivers/i3c/master.c | 4 +--- drivers/input/gameport/gameport.c | 3 +-- drivers/input/serio/serio.c | 3 +-- drivers/ipack/ipack.c | 4 +--- drivers/macintosh/macio_asic.c | 4 +--- drivers/mcb/mcb-core.c | 4 +--- drivers/media/pci/bt8xx/bttv-gpio.c | 3 +-- drivers/memstick/core/memstick.c | 3 +-- drivers/mfd/mcp-core.c | 3 +-- drivers/misc/mei/bus.c | 4 +--- drivers/misc/tifm_core.c | 3 +-- drivers/mmc/core/bus.c | 4 +--- drivers/mmc/core/sdio_bus.c | 4 +--- drivers/net/netdevsim/bus.c | 3 +-- drivers/ntb/core.c | 4 +--- drivers/ntb/ntb_transport.c | 4 +--- drivers/nubus/bus.c | 6 ++---- drivers/nvdimm/bus.c | 3 +-- drivers/pci/endpoint/pci-epf-core.c | 7 ++----- drivers/pci/pci-driver.c | 3 +-- drivers/pcmcia/ds.c | 4 +--- drivers/platform/surface/aggregator/bus.c | 4 +--- drivers/platform/x86/wmi.c | 4 +--- drivers/pnp/driver.c | 3 +-- drivers/rapidio/rio-driver.c | 4 +--- drivers/rpmsg/rpmsg_core.c | 7 ++----- drivers/s390/block/scm_drv.c | 4 +--- drivers/s390/cio/ccwgroup.c | 6 +----- drivers/s390/cio/chsc_sch.c | 3 +-- drivers/s390/cio/css.c | 7 +++---- drivers/s390/cio/css.h | 2 +- drivers/s390/cio/device.c | 9 +++------ drivers/s390/cio/eadm_sch.c | 4 +--- drivers/s390/cio/scm.c | 5 +++-- drivers/s390/cio/vfio_ccw_drv.c | 3 +-- drivers/s390/crypto/ap_bus.c | 4 +--- drivers/scsi/scsi_debug.c | 3 +-- drivers/sh/superhyway/superhyway.c | 8 ++------ drivers/siox/siox-core.c | 4 +--- drivers/slimbus/core.c | 4 +--- drivers/soc/qcom/apr.c | 4 +--- drivers/spi/spi.c | 4 +--- drivers/spmi/spmi.c | 3 +-- drivers/ssb/main.c | 4 +--- drivers/staging/fieldbus/anybuss/host.c | 4 +--- drivers/staging/greybus/gbphy.c | 4 +--- drivers/target/loopback/tcm_loop.c | 5 ++--- drivers/thunderbolt/domain.c | 4 +--- drivers/tty/serdev/core.c | 4 +--- drivers/usb/common/ulpi.c | 4 +--- drivers/usb/serial/bus.c | 4 +--- drivers/usb/typec/bus.c | 4 +--- drivers/vdpa/vdpa.c | 4 +--- drivers/vfio/mdev/mdev_driver.c | 4 +--- drivers/virtio/virtio.c | 3 +-- drivers/vlynq/vlynq.c | 4 +--- drivers/vme/vme.c | 4 +--- drivers/xen/xenbus/xenbus.h | 2 +- drivers/xen/xenbus/xenbus_probe.c | 4 +--- drivers/zorro/zorro-driver.c | 3 +-- include/linux/device/bus.h | 2 +- include/linux/pci-epf.h | 2 +- sound/ac97/bus.c | 6 ++---- sound/aoa/soundbus/core.c | 4 +--- 93 files changed, 107 insertions(+), 263 deletions(-) base-commit: e73f0f0ee7541171d89f2e2491130c7771ba58d3 -- 2.30.2

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Any way of making this Intel SoC sound cluster work?
by Valentin Villenave 22 Jul '21

22 Jul '21

Greetings, I recently found myself with a rather atypical laptop from a niche Chinese manufacturer (relying on an old J3455 SoC); the `Intel Smart Sound Technology` output does work under Windows with the appropriate drivers, but under GNU/Linux (no matter which distro or release I test: Debian, Ubuntu, ClearLinux,…) no audio output is reportedly available other than HDMI. Lspci does list an “Audio Cluster” [10ec:10fc] and supposedly handles it with snd_hda_intel, but that’s as far as I got. This looks to me like a missing firmware issue, but I’ve spent hours trawling the Internets to no avail. I also tried fiddling with the modules, by either blacklisting snd_hda_intel or loading snd_soc_skl and virtually every other combination I could think of. If there is an arcane option to be passed to snd_hda_intel, I have yet to find which one. Attached are the alsa-info output, the verbose lspci output and (just in case) the output of `wmic path win32_pnpentity` on Windows. If some of you fine alsa-devel experts could give me any pointers, I’d greatly appreciate it! Cheers, -- Valentin (from France).

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[PATCH 02/54] ASoC: dt-bindings: Convert Bluetooth SCO Link binding to a schema
by Maxime Ripard 22 Jul '21

22 Jul '21

Bluetooth SCO Link are supported by Linux with a matching device tree binding. Now that we have the DT validation in place, let's convert the device tree bindings for that driver over to a YAML schema. The value expected for #sound-dai-cells wasn't documented though, and the users were inconsistent. The example didn't list it, and across the 4 users we have in tree: - 1 had a cells value of 1, but using only 0 as argument - 1 had a cells value of 0, - 2 didn't have this property at all, behaving as if it was 0, It seems like the consensus seems to be that it should be 0, so let's enforce it. Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Samuel Holland <samuel(a)sholland.org> Signed-off-by: Maxime Ripard <maxime(a)cerno.tech> --- .../devicetree/bindings/sound/bt-sco.txt | 13 ------- .../bindings/sound/linux,bt-sco.yaml | 34 +++++++++++++++++++ 2 files changed, 34 insertions(+), 13 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/bt-sco.txt create mode 100644 Documentation/devicetree/bindings/sound/linux,bt-sco.yaml diff --git a/Documentation/devicetree/bindings/sound/bt-sco.txt b/Documentation/devicetree/bindings/sound/bt-sco.txt deleted file mode 100644 index 641edf75e184..000000000000 --- a/Documentation/devicetree/bindings/sound/bt-sco.txt +++ /dev/null @@ -1,13 +0,0 @@ -Bluetooth-SCO audio CODEC - -This device support generic Bluetooth SCO link. - -Required properties: - - - compatible : "delta,dfbmcs320" or "linux,bt-sco" - -Example: - -codec: bt_sco { - compatible = "delta,dfbmcs320"; -}; diff --git a/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml new file mode 100644 index 000000000000..334b508205cd --- /dev/null +++ b/Documentation/devicetree/bindings/sound/linux,bt-sco.yaml @@ -0,0 +1,34 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/linux,bt-sco.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Bluetooth SCO Audio Codec Device Tree Bindings + +maintainers: + - Mark Brown <broonie(a)kernel.org> + +properties: + '#sound-dai-cells': + const: 0 + + compatible: + enum: + - delta,dfbmcs320 + - linux,bt-sco + +required: + - '#sound-dai-cells' + - compatible + +additionalProperties: false + +examples: + - | + codec { + #sound-dai-cells = <0>; + compatible = "linux,bt-sco"; + }; + +... -- 2.31.1

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[PATCH] ALSA: usb-audio: Add registration quirk for JBL Quantum headsets
by Alexander Tsoy 22 Jul '21

22 Jul '21

These devices has two interfaces, but only the second interface contains the capture endpoint, thus quirk is required to delay the registration until the second interface appears. Tested-by: Jakub Fišer <jakub(a)ufiseru.cz> Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/quirks.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 8b8bee3c3dd6..e7accd87e063 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1897,6 +1897,9 @@ static const struct registration_quirk registration_quirks[] = { REG_QUIRK_ENTRY(0x0951, 0x16d8, 2), /* Kingston HyperX AMP */ REG_QUIRK_ENTRY(0x0951, 0x16ed, 2), /* Kingston HyperX Cloud Alpha S */ REG_QUIRK_ENTRY(0x0951, 0x16ea, 2), /* Kingston HyperX Cloud Flight S */ + REG_QUIRK_ENTRY(0x0ecb, 0x1f46, 2), /* JBL Quantum 600 */ + REG_QUIRK_ENTRY(0x0ecb, 0x2039, 2), /* JBL Quantum 400 */ + REG_QUIRK_ENTRY(0x0ecb, 0x203e, 2), /* JBL Quantum 800 */ { 0 } /* terminator */ }; -- 2.31.1

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mic not working on JBL Quantum series USB headsets
by Jakub Fišer 22 Jul '21

22 Jul '21

Hello Alsa people, hope this is the right place to post. My Quantum 800's microphone doesn't work and I found other reports suggesting this issue affects the whole Quantum product line. It's a wireless headset with USB dongle. I *think* the issue is with Alsa because the Capture device is "detected" but not populated under /sys: arecord list of devices: ``` % arecord -l **** List of CAPTURE Hardware Devices **** (...) card 1: Q800 [Quantum 800], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` amixer knobs: ``` % amixer -c1 Simple mixer control 'PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PCM',1 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Headset',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 64 Mono: Capture 64 [100%] [0.00dB] [on] ``` arecord trying to use the capture device: ``` % arecord -Dhw:1,0 /tmp/out.wav arecord: main:831: audio open error: No such file or directory ``` /sys pcm "endpoints" (missing capture endpoint): ``` % ls -1d /sys/class/sound/pcm* /sys/class/sound/pcmC1D0p /sys/class/sound/pcmC1D1p ``` nothing suspicious in dmesg: ``` usb 3-1.3: new full-speed USB device number 12 using xhci_hcd usb 3-1.3: New USB device found, idVendor=0ecb, idProduct=203e, bcdDevice= 1.00 usb 3-1.3: New USB device strings: Mfr=4, Product=5, SerialNumber=0 usb 3-1.3: Product: Quantum 800 usb 3-1.3: Manufacturer: JBL usb 3-1.3: Found post-registration device assignment: 0ecb203e:02 input: JBL Quantum 800 Consumer Control as /devices/pci0000:00/0000:00:1d.4/0000:05:00.0/0000:06:01.0/0000:08:00.0/0000:09:02.0/0000:0a:00.0/us 3/3-1/3-1.3/3-1.3:1.5/0003:0ECB:203E.0024/input/input51 input: JBL Quantum 800 as /devices/pci0000:00/0000:00:1d.4/0000:05:00.0/0000:06:01.0/0000:08:00.0/0000:09:02.0/0000:0a:00.0/us 3/3-1/3-1.3/3-1.3:1.5/0003:0ECB:203E.0024/input/input52 hid-generic 0003:0ECB:203E.0024: input,hiddev96,hidraw0: USB HID v1.11 Device [JBL Quantum 800] on usb-0000:0a:00.0-1.3/input5 ``` Nothing at all in either Pulse or Pipewire logs. alsa-info: http://alsa-project.org/db/?f=d9e7a91d97b2f64a0b2d00a05dfc9f9dace3a6f6 Linux 5.12.14-lqx2 Other reports: https://bbs.archlinux.org/viewtopic.php?pid=1983686 https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/issues/998 The headset also works for me on Windows 10 out of the box (and according to the Internet, it works on Mac, too). Note that I updated the headset and dongle to the latest firmware (v3.1.5.0) with no effect. The 800 model also has Bluetooth with both A2DP and HSP profile. Not sure if this is relevant but for clarity: - with Pulse A2DP profile works only as output device (SBC and AAC), other profiles, such as HSP, are listed but "unavailable" - with Pipewire only HSP profiles are listed and available (mSBC, CVSD and "plain"), microphone *works* I'd like to help with debugging, just tell me how (or if) I can. thanks, -jakub.

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Re: mic not working on JBL Quantum series USB headsets
by Alexander Tsoy 22 Jul '21

22 Jul '21

В Ср, 21/07/2021 в 22:34 +0000, Jakub Fišer пишет: > July 21, 2021 10:17 PM, "Alexander Tsoy" <alexander(a)tsoy.me> wrote: > > > В Вт, 20/07/2021 в 18:00 +0000, Jakub Fišer пишет: > > > > > My Quantum 800's microphone doesn't work and I found other reports > > > suggesting > > > this issue affects the whole Quantum product line. > > > > This device consists of two interfaces. The first interface contains > > a > > single playback endpoint and the second interface contains playback > > and > > capture endpoints. So JBL Quantum probably needs a registration quirk > > similar to Kingston HyperX devices. Please try the patch below: > > The patch works! Thanks. Two new profiles appeared in Pulseaudio (via > Pipewire) with "Mono input" and the profile "Pro Audio" gained the > mono input as well. Both analog and digital profiles work for playback > but only the "Pro Audio" profile exposes the second output as well. > This > profile is only available when using Pipewire (not in "plain" Pulse) > but > I guess this is not an Alsa-related issue. > > The microphone works in all three profiles as expected, everything > is there out-of-the-box with the patched kernel. What's the process to > get this patch to mainline? Should I be involved in it somehow? Thanks for testing. I'll submit this patch. > > > Just FYI, the headset also claims to have support for DTS Surround and > a 7.1 ultimate-something, the latter being explicitly a proprietary > thing. > > I guess DTS support is proprietary, too, but I couldn't find anything > about > it except that "it is there". In Windows both modes are only available > via > the proprietary application. According to lsusb output [1] this device works in UAC 1.0 mode and doesn't expose any of these capabilities. Presumably Windows driver is switching the device into some proprietary vendor specific mode or it is processing surround sound entirely in software. [1] https://github.com/linuxhw/LsUSB/blob/master/Desktop/Gigabyte%20Technology/…

2 1

out of memory issue
by Alex Roberts 21 Jul '21

21 Jul '21

Hello, I am developing a dummy codec to interface with an 8-channel, 24-bit ADC. I've got it working on an NXP imx8m through the fsl_sai driver on kernel 5.4.85. I can capture all 8 channels at varying sample rates using arecord, and I've verified correct data capture via opening the resulting .wav file in Audacity. The problem I am having is that occasionally, upon starting arecord - after a fresh power cycle - I get an out of memory error. Other times I get an out of memory after a non-deterministic period of capture. Starting capture again also reports out of memory, but if I wait several minutes and start capture it will start recording again. A power cycle usually helps, but as stated earlier, not 100% of the time. I'm trying to track down where the oom error is coming from, but haven't had much luck. My colleague tried running arecord with valgrind to check for memory leaks and nothing of note was observed. My suspicion is there's something going on with allocated memory for DMA, like fragmentation starts to happen and it can't get a contiguous region for operation. Reserving a large pool - either via device tree or boot cmd arguments in the bootoader - did not seem to help. Another thought is that it's a boundary/alignment issue due to the 24-bit data, and the error is the result of trying to allocate a chunk of memory for DMA that doesn't align. I'm very new to ALSA dev with some exposure to kernel dev in general, so please correct me if I'm wrong or completely mis-understanding something. Any suggestions on where I should / how I can debug this memory error? Thanks, Alex.

1 0

[GIT PULL] ASoC fixes for v5.14-rc2
by Mark Brown 21 Jul '21

21 Jul '21

The following changes since commit 6a7f5bd6185e1c86256d5e52c3bb7a4d390d6e19: ASoC: qcom: lpass-cpu: mark IRQ_CLEAR register as volatile and readable (2021-06-24 19:29:53 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.14-rc2 for you to fetch changes up to 82d28b67f780910f816fe1cfb0f676fc38c4cbb3: ASoC: ti: j721e-evm: Check for not initialized parent_clk_id (2021-07-19 12:58:42 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.14 A collection of fixes for ASoC that have come in since the merge window, all driver specific. There is a new core feature added for reversing the order of operations when shutting down, this is needed to fix a bug with the AMD Stonyridge platform, and we also tweak the Kconfig to make the SSM2518 driver user selectable so it can be used with generic cards but that requires no actual code changes. ---------------------------------------------------------------- Charles Keepax (2): ASoC: wm_adsp: Correct wm_coeff_tlv_get handling ASoC: wm_adsp: Add CCM_CORE_RESET to Halo start core Kyle Russell (1): ASoC: tlv320aic31xx: fix reversed bclk/wclk master bits Lucas Stach (1): ASoC: codecs: allow SSM2518 to be selected by the user Marek Vasut (1): ASoC: tlv320aic32x4: Fix TAS2505 volume controls Mark Brown (3): Merge series "ASoC: Intel: machine driver corrections" from Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>: ASoC: tlv320aic31xx: Make regmap cache only on probe() Merge series "arm64: tegra: Enable audio IOMMU support on Tegra194" from Thierry Reding <thierry.reding(a)gmail.com> Thierry Reding <treding(a)nvidia.com>: Maxim Schwalm (1): ASoC: rt5631: Fix regcache sync errors on resume Oder Chiou (1): ASoC: rt5682: Fix the issue of garbled recording after powerd_dbus_suspend Peter Robinson (1): ASoC: remove zte zx dangling kconfig Peter Ujfalusi (2): ASoC: ti: j721e-evm: Fix unbalanced domain activity tracking during startup ASoC: ti: j721e-evm: Check for not initialized parent_clk_id Rander Wang (1): ASoC: Intel: boards: fix xrun issue on platform with max98373 Rob Herring (1): ASoC: dt-bindings: renesas: rsnd: Fix incorrect 'port' regex schema Sathya Prakash M R (1): ASoC: SOF: Intel: Update ADL descriptor to use ACPI power states Srinivas Kandagatla (2): ASoC: codecs: wcd938x: make sdw dependency explicit in Kconfig ASoC: codecs: wcd938x: setup irq during component bind Thierry Reding (1): ASoC: tegra: Use ADMAIF component for DMA allocations Vijendar Mukunda (2): ASoC: soc-pcm: add a flag to reverse the stop sequence ASoC: amd: reverse stop sequence for stoneyridge platform .../devicetree/bindings/sound/renesas,rsnd.yaml | 2 +- include/sound/soc.h | 6 ++ sound/soc/amd/acp-da7219-max98357a.c | 5 ++ sound/soc/codecs/Kconfig | 8 +-- sound/soc/codecs/rt5631.c | 2 + sound/soc/codecs/rt5682.c | 8 ++- sound/soc/codecs/tlv320aic31xx.c | 2 + sound/soc/codecs/tlv320aic31xx.h | 4 +- sound/soc/codecs/tlv320aic32x4.c | 27 ++++---- sound/soc/codecs/wcd938x.c | 18 ++--- sound/soc/codecs/wm_adsp.c | 6 +- sound/soc/intel/boards/sof_sdw_max98373.c | 81 ++++++++++++++-------- sound/soc/soc-pcm.c | 22 ++++-- sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/tegra/tegra_pcm.c | 30 ++++---- sound/soc/ti/j721e-evm.c | 18 +++-- 16 files changed, 152 insertions(+), 88 deletions(-)

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[PATCH 0/3] Read firmware, tplg and machine driver name from dts node
by Daniel Baluta 21 Jul '21

21 Jul '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries adds support for reading the firmware name, topology name and machine driver name from dsp dts node. Intel side uses ACPI to read this info. We should use DT for i.MX. First patches should go via sof-dev tree. Also last one could go via the same tree after we get an Ack from Rob. Daniel Baluta (3): ASoC: SOF: Parse fw/tplg filename from DT ASoC: SOF: Introduce machine driver name dt-bindings: dsp: fsl: Document newly introduced fsl,properties .../devicetree/bindings/dsp/fsl,dsp.yaml | 20 +++++++++++ include/sound/sof.h | 1 + sound/soc/sof/pcm.c | 5 ++- sound/soc/sof/sof-audio.c | 2 +- sound/soc/sof/sof-of-dev.c | 34 +++++++++++++++++++ 5 files changed, 60 insertions(+), 2 deletions(-) -- 2.27.0

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[PATCH] ASoC: amd: Don't show messages about deferred probing by default
by Mario Limonciello 21 Jul '21

21 Jul '21

Nearly every boot with a Lenovo P14s is showing acp_pdm_mach acp_pdm_mach.0: snd_soc_register_card(acp) failed: -517 This isn't useful to a user, especially as probing will run again. Downgrade the message to debug, and only show errors. CC: markpearson(a)lenovo.com Signed-off-by: Mario Limonciello <mario.limonciello(a)amd.com> --- sound/soc/amd/renoir/acp3x-rn.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/renoir/acp3x-rn.c b/sound/soc/amd/renoir/acp3x-rn.c index 306134b89a82..e785994de614 100644 --- a/sound/soc/amd/renoir/acp3x-rn.c +++ b/sound/soc/amd/renoir/acp3x-rn.c @@ -54,9 +54,14 @@ static int acp_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, machine); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, - "snd_soc_register_card(%s) failed: %d\n", - acp_card.name, ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card(%s) failed: %d\n", + card->name, ret); + else + dev_dbg(&pdev->dev, + "snd_soc_register_card(%s) probe deferred: %d\n", + card->name, ret); return ret; } return 0; -- 2.25.1

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[PATCH] ALSA: korg1212: Fix crash in snd_korg1212_create()
by Dan Carpenter 21 Jul '21

21 Jul '21

The "dsp_code" is not initialized until the next line so we can't dereference it. Fixes: b5cde369b618 ("ALSA: korg1212: Allocate resources with device-managed APIs") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/pci/korg1212/korg1212.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 7872abbd4587..3b2df5d06839 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -2256,17 +2256,17 @@ static int snd_korg1212_create(struct snd_card *card, struct pci_dev *pci) korg1212->AdatTimeCodePhy = korg1212->sharedBufferPhy + offsetof(struct KorgSharedBuffer, AdatTimeCode); - korg1212->dma_dsp = snd_devm_alloc_pages(&pci->dev, SNDRV_DMA_TYPE_DEV, - dsp_code->size); - if (!korg1212->dma_dsp) - return -ENOMEM; - err = request_firmware(&dsp_code, "korg/k1212.dsp", &pci->dev); if (err < 0) { snd_printk(KERN_ERR "firmware not available\n"); return err; } + korg1212->dma_dsp = snd_devm_alloc_pages(&pci->dev, SNDRV_DMA_TYPE_DEV, + dsp_code->size); + if (!korg1212->dma_dsp) + return -ENOMEM; + K1212_DEBUG_PRINTK("K1212_DEBUG: DSP Code area = 0x%p (0x%08x) %d bytes [%s]\n", korg1212->dma_dsp->area, korg1212->dma_dsp->addr, dsp_code->size, stateName[korg1212->cardState]); -- 2.20.1

2 1

[bug report] ALSA: als300: Allocate resources with device-managed APIs
by Dan Carpenter 21 Jul '21

21 Jul '21

Hello Takashi Iwai, The patch 21a9314cf93b: "ALSA: als300: Allocate resources with device-managed APIs" from Jul 15, 2021, leads to the following static checker warning: sound/pci/als300.c:713 snd_als300_probe() error: potentially dereferencing uninitialized 'chip'. sound/pci/als300.c 686 static int snd_als300_probe(struct pci_dev *pci, 687 const struct pci_device_id *pci_id) 688 { 689 static int dev; 690 struct snd_card *card; 691 struct snd_als300 *chip; 692 int err, chip_type; 693 694 if (dev >= SNDRV_CARDS) 695 return -ENODEV; 696 if (!enable[dev]) { 697 dev++; 698 return -ENOENT; 699 } 700 701 err = snd_devm_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, 702 sizeof(*chip), &card); 703 if (err < 0) 704 return err; 705 706 chip_type = pci_id->driver_data; 707 708 err = snd_als300_create(card, pci, chip_type); 709 if (err < 0) 710 return err; 711 712 strcpy(card->driver, "ALS300"); --> 713 if (chip->chip_type == DEVICE_ALS300_PLUS) ^^^^^^^^^^^^^^^ chip isn't initialized now. 714 /* don't know much about ALS300+ yet 715 * print revision number for now */ 716 sprintf(card->shortname, "ALS300+ (Rev. %d)", chip->revision); 717 else 718 sprintf(card->shortname, "ALS300 (Rev. %c)", 'A' + 719 chip->revision - 1); 720 sprintf(card->longname, "%s at 0x%lx irq %i", 721 card->shortname, chip->port, chip->irq); 722 723 err = snd_card_register(card); 724 if (err < 0) 725 return err; 726 727 pci_set_drvdata(pci, card); 728 dev++; 729 return 0; 730 } regards, dan carpenter

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[PATCH v4 00/12] Add Vangogh ACP ASoC driver
by Vijendar Mukunda 21 Jul '21

21 Jul '21

This adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Vangogh APU. Changes since v3: - Fixed typo errors - Fixed platform devices unregister logic - Updated acp resources count with macro value - updated ACP5x_I2S_MODE macro value to avoid unregistering platform devices for other ACP audio configurations in driver remove callback - Fixed error handling - Fixed indentation issues in irq handler - Add __maybe_unused for PM functions Changes since v2: - remove extra stuff from Make file Changes since v1: - add comment for acp_readl() & acp_writel() API's in common header file. - add amd: tag in commit message - modify condition check in acp5x_power_on() - removed acp_power_gating flag related code - drop acp5x_audio_remove() API - use single flag for handling irq status - fixed indetation issue and used dev_err instead of pr_err - drop acp5x_dai_remove() API - use SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK - use PM related Macros in ACP PCI driver - use PM related Macros in ACP DMA driver - remove extra line in Make file Vijendar Mukunda (12): ASoC: amd: add Vangogh ACP5x IP register header ASoC: amd: add Vangogh ACP PCI driver ASoc: amd: add acp5x init/de-init functions ASoC: amd: create acp5x platform devices ASoC: amd: add ACP5x PCM platform driver ASoC: amd: irq handler changes for ACP5x PCM dma driver ASoC: amd: add ACP5x pcm dma driver ops ASoC: amd: add vangogh i2s controller driver ASoC: amd: add vangogh i2s dai driver ops ASoC: amd: add vangogh pci driver pm ops ASoC: amd: add vangogh i2s dma driver pm ops ASoC: amd: enable vangogh acp5x driver build sound/soc/amd/Kconfig | 9 + sound/soc/amd/Makefile | 1 + sound/soc/amd/vangogh/Makefile | 9 + sound/soc/amd/vangogh/acp5x-i2s.c | 427 ++++++++++++++++ sound/soc/amd/vangogh/acp5x-pcm-dma.c | 524 ++++++++++++++++++++ sound/soc/amd/vangogh/acp5x.h | 193 +++++++ sound/soc/amd/vangogh/pci-acp5x.c | 326 ++++++++++++ sound/soc/amd/vangogh/vg_chip_offset_byte.h | 337 +++++++++++++ 8 files changed, 1826 insertions(+) create mode 100644 sound/soc/amd/vangogh/Makefile create mode 100644 sound/soc/amd/vangogh/acp5x-i2s.c create mode 100644 sound/soc/amd/vangogh/acp5x-pcm-dma.c create mode 100644 sound/soc/amd/vangogh/acp5x.h create mode 100644 sound/soc/amd/vangogh/pci-acp5x.c create mode 100644 sound/soc/amd/vangogh/vg_chip_offset_byte.h -- 2.17.1

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