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[PATCH v8 0/8] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 27 Jun '22

27 Jun '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v7: - Optimize the code logic, such as the return value and position of a function. - Use devm_pm_runtime_enable() instead of pm_runtime_enable(). - Use pm_runtime_resume_and_get() instead of pm_runtime_get_sync(). Changes since v6: - The Makefile and Kconfig updates are moved into the driver patches so that can keep independence of each patch. Changes since v5: - fix build error about function snd_soc_card_jack_new that modified by: Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com - some have been accepted, details are in the link below: Link: https://lore.kernel.org/all/165459931885.399031.2621579592368573898.b4-ty@k… Changes since v4: - [v5 07/20] - remove unsusd controls - [v5 09/20] - correct indent error - [v5 10/20] - [v5 13/20] - [v5 14/20] - fix the return value if the value is different from the previous value in mixer controls - [v5 17/20] - [v5 19/20] - correct the compatible name with '_' instead of '-' - [v5 18/20] - [v5 20/20] - correct the yaml after 'pip3 install dtschema --upgrade' Changes since v3: - [v4 09/18] - remove DEBUG_COEFF debugging code - [v4 10/18] - simplify the logic of the code - [v4 13/18] - split out the MT6366 bits into mt8186-mt6366-commom.c - fix build error of "error: 'struct dev_pm_info' has no member named 'runtime_error'" - fix bug of adda dai driver - add route for pcm interface channel 2. - [v4 15/18] - [v4 17/18] - commonize the configuration of the codec - move MT6366 common bits into mt8186-mt6366-common.c Changes since v2: - add a new compatible string "mediatek,mt6366-sound" - modify the log level for simplicity - use dev_err_probe(...) instead of dev_err(...) in dev probe() - optimized the logic of some code - use BIT() and GENMASK() macros to descript the registers Thanks for AngeloGioacchino's careful reviews. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (8): dt-bindings: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../devicetree/bindings/sound/mt6358.txt | 4 +- .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 235 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 255 ++ sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3011 +++++++++++++++++ .../mediatek/mt8186/mt8186-mt6366-common.c | 57 + .../mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 ++++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 ++++++ 14 files changed, 5950 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c -- 2.18.0

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attach/detach by mixer class implementation of alsa-lib mixer API
by Takashi Sakamoto 27 Jun '22

27 Jun '22

Hi Jaroslav, Recent years I'm bothered about unexpected abort of pulseaudio and pipewire processes when testing user-defined control element set. They aborts at element removal event. ``` pulseaudio: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed. wireplumber: mixer.c:149: hctl_elem_event_handler: Assertion `bag_empty(bag)' failed. ``` Would I ask your opinion about the design of alsa-lib mixer API? As long as I investigate, these programs seem to have careless coding as alsa-lib mixer API application. Both of them attaches an instance of snd_mixer_elem_t to an instance of snd_hctl_elem_t by calling snd_mixer_elem_attach() when detecting element addition, but never detach it even if detecting element removal. In the case, the link list (=bag) of mixer API internal never becomes empty. It has link entries as much as the number of registered mixer classes which attaches snd_mixer_elem_t. Then it hits the assertion. I think the design of alsa-lib mixer API demands mixer class implementation to detach at element removal which attached at element addition. But I have less conviction about it since enough unfamiliar. I'm glad if receiving your opinion about it. I wrote test program for the issued behaviour: - https://gist.github.com/takaswie/8378fe3bc04652d83428694cd7573bc0 For test, please use sample script in alsa-gobject project: - https://github.com/alsa-project/alsa-gobject/blob/master/samples/ctl The patches for pulseaudio/pipewire are prepared: - https://gitlab.freedesktop.org/takaswie/pulseaudio/-/commit/topic/fix-wrong… - https://gitlab.freedesktop.org/takaswie/pipewire/-/commits/topic/fix-wrong-… Regards Takashi Sakamoto

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mixer: add documentation about postcondition of removal event processing
by GitHub pull_request - opened 27 Jun '22

27 Jun '22

alsa-project/alsa-lib pull request #244 was opened from takaswie: PulseAudio and PipeWire seems to appear including careless code to process events of mixer element in implementation of mixer class. * https://lore.kernel.org/alsa-devel/YrbxZ2b+3rIdi7Ut@workstation/ They register own implementation of mixer class. At addition event, they attach own mixer element into hcontrol element. However at removal event, they never detach the mixer element from hcontrol element. They hit assertion in mixer API internal due to unsatisfied postcondition. This commit adds documentation about postcondition of removal event processing so that developer for implementation of mixer class easily realize the way to satisfy the postcondition. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-lib/pull/244 Patch URL : https://github.com/alsa-project/alsa-lib/pull/244.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH v3 0/2] [PATCH v4 0/2] Add pinctrl support adsp bypass platforms
by Srinivasa Rao Mandadapu 26 Jun '22

26 Jun '22

This patch set is to make clock voting optinal for adsp bypass sc7280 platforms. Changes Since V3: -- Revert compatible type change in dt bindings. -- Update boolean flag description in dt bindings. -- Drop redundant clock optional flag initialization. -- Remove redundant if check. Changes Since V2: -- Remove redundant lpi pincontrol variant data structure and compatible entry. -- Add adsp bypass mode boolean param check. -- Remove compatible name in dt bindings. -- Update dt binding bypass mode boolean param. Changes Since V1: -- Update commit message. Srinivasa Rao Mandadapu (2): dt-bindings: pinctrl: qcom: sc7280: Add boolean param for ADSP bypass platforms pinctrl: qcom: sc7280: Add clock optional check for ADSP bypass targets .../devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 5 +++++ drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 2 ++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 1 - 3 files changed, 7 insertions(+), 1 deletion(-) -- 2.7.4

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Questions about driver implementation (1 I2S controller to 2 codecs)
by Paul Cercueil 25 Jun '22

25 Jun '22

Hi, I'm trying to add support for a new board that uses the Ingenic JZ4770 SoC. This SoC has a I2S/AC97 controller, and has an internal codec. The controller has a register switch to select either the internal codec, or an external codec; both cannot be enabled at the same time. On this board, the external speakers / audio line are wired to the internal codec, while a HDMI chip (ITE66121) is wired as the external codec. I'm having a hard time trying to figure out how the codec selection switch should be exported. Should it be a regular widget, and the user is responsible for selecting the right codec? Should it be a DAPM, and plugging the HDMI cable auto-enables the switch? Is this configuration (one controller to two codecs) already somehow supported by ALSA? Right now we're using a "simple-audio-card" in the device tree, and trying to add HDMI sound support. I am not even sure if we should be using one sound card with the two codecs, or one sound card per codec, sharing the same I2S controller? Any thoughts? Thanks, -Paul

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Quirks for MacroSilicon MS2100/MS2106
by John Veness 25 Jun '22

25 Jun '22

Hello, This is my first ever kernel work, so I hope I am doing everything correctly. I have a USB analogue AV capture dongle with a MacroSilicon MS2100E chip inside. PID 0x534d, VID 0x0021 (534d:0021). Apparently the MS2106 uses the same PID/VID, so hopefully acts the same, but I don't have one to check. For audio, the MS2100E has the same problems as the MS2109 USB HDMI capture dongle, namely claiming 96kHz mono sound when it's actually 48kHz stereo, and with left/right channel problems. Luckily, the fix is the exact same as that for the MS2109, as first implemented by Marcan (Hector Martin) in July 2020. Below is a patch which is basically a copy and paste of his, with a different VID. I have tested this patch in 5.19.0 RC2. I haven't recruited any other testers. Even with this patch, there is a remaining problem, which is not present in the MS2109. The sound sample values range from 0x0000 to 0x7fff, with silence around 0x4000, i.e. 15-bit-ish audio. This actually sounds OK to the ear (although half as loud as it should be), but looks odd when looking at the waveform, and makes volume meters always think the sound is very loud. To convert to s16le, I can bitshift one bit left, and subtract 32768. I'm told that this isn't something that can or should be done in the kernel, but should be in userspace. Any more advice on how to fix this remaining quirk would be very welcome. Nonetheless, as I say, with the following kernel patch the captured audio certainly sounds right, so is a big improvement and makes these dongles usable: diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 4f56e1784932..853da162fd18 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3802,6 +3802,54 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* + * MacroSilicon MS2100/MS2106 based AV capture cards + * + * These claim 96kHz 1ch in the descriptors, but are actually 48kHz 2ch. + * They also need QUIRK_FLAG_ALIGN_TRANSFER, which makes one wonder if + * they pretend to be 96kHz mono as a workaround for stereo being broken + * by that... + * + * They also have an issue with initial stream alignment that causes the + * channels to be swapped and out of phase, which is dealt with in quirks.c. + */ +{ + USB_AUDIO_DEVICE(0x534d, 0x0021), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MacroSilicon", + .product_name = "MS210x", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 2, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + }, + { + .ifnum = -1 + } + } + } +}, + /* * MacroSilicon MS2109 based HDMI capture cards * diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index e8468f9b007d..a72874bc0936 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1478,6 +1478,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x041e, 0x3f19): /* E-Mu 0204 USB */ set_format_emu_quirk(subs, fmt); break; + case USB_ID(0x534d, 0x0021): /* MacroSilicon MS2100/MS2106 */ case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; @@ -1904,6 +1905,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x413c, 0xa506, /* Dell AE515 sound bar */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x534d, 0x0021, /* MacroSilicon MS2100/MS2106 */ + QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x534d, 0x2109, /* MacroSilicon MS2109 */ QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x1224, 0x2a25, /* Jieli Technology USB PHY 2.0 */ Thanks for reading and considering this patch. John Veness

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[PATCH 00/11] ctl: elem-value: rewrite getter methods
by GitHub pull_request - opened 25 Jun '22

25 Jun '22

alsa-project/alsa-gobject pull request #79 was opened from takaswie: This patchset rewrites getter methods of ALSACtl.ElemValue so that they just return pointer of internal storage instead of copying to user buffer. ``` Takashi Sakamoto (11): timer: fix entry in exported symbol map ctl: elem-value: clear value field at setter method ctl: elem-value: rewrite getter for boolean values ctl: elem-value: rewrite getter for 32 bit signed integer values ctl: elem-value: rewrite getter for enumeration index values ctl: elem-value: rewrite getter for 8 bit unsigned integer values ctl: elem-value: rewrite getter for IEC 60958 channel status ctl: elem-value: rewrite getter for IEC 60958 user data ctl: elem-value: rewrite getter for 64 bit signed integer values ctl: elem-value: update class description update README with compatibility note README.rst | 124 ++++++++++++++++++++++++++++++-- samples/ctl | 11 ++- src/ctl/alsactl.map | 15 ++-- src/ctl/elem-value.c | 152 +++++++++++++++++++++------------------- src/ctl/elem-value.h | 14 ++-- src/timer/alsatimer.map | 3 +- 6 files changed, 218 insertions(+), 101 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-gobject/pull/79 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/79.patch Repository URL: https://github.com/alsa-project/alsa-gobject

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sof-soundwire: broken settings for RT711 and RT711-SDCA
by GitHub issues - opened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #179 was opened from plbossart: [plb@fedora ~]$ alsaucm -c sof-soundwire set _verb HiFi ALSA lib main.c:826:(execute_sequence) unable to execute cset 'name='PGA2.0 2 Master Capture Switch' 0' ALSA lib main.c:2573:(set_verb_user) error: failed to initialize new use case: HiFi There's not such thing as a switch here: amixer -Dhw:0 sget 'PGA2.0 2 Master',0 Simple mixer control 'PGA2.0 2 Master',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [30.00dB] Front Right: Capture 80 [100%] [30.00dB] In addition, it makes no sense to use an SOF control to set the volume/switch on the headset codec side. The volume control should be done in the codec, not in the SOF firmware. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/179 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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UCM for Behringer UMC204HD broken
by GitHub issues - reopened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #177 was reopened from gregory112: This is related: [https://github.com/alsa-project/alsa-ucm-conf/pull/128](https://github.com/…. I came from this issue I filed in Pipewire: [https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/2471](https://git…. So previously a while back, Behringer UMC204HD was detected as a single stereo input channel. This sound card has two inputs. However, just recently, the channel was separated into two mono channels, which was perfect. However, with the most recent update (I don't know if it's pipewire, pipewire-alsa, alsa-card-profiles, or alsa-ucm), it reverted back into the stereo single input (left/right) mode. What can I do to bring back the previous correct settings (apart from the bugs I filed in Pipewire)? I did not modify any ALSA UCM files. Previously, a while back: it is detected as UMC204HD 192k (stereo input). Then: UMC204HD 192k Line 1 (mono) and UMC204HD 192k Line 2 (mono) (this is how it's supposed to be, I think it's from the pull request above). Now: Back to UMC204HD 192k (stereo input). Might also be helpful but I am using pipewire. This is the output of `alsaucm listcards`: ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 ALSA lib parser.c:2957:(uc_mgr_scan_master_configs) Unable to open '-hw:2': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` There is an error for hw:2, which I assume is the Behringer. Here is the result of `alsaucm -chw:2 dump text`. ``` ALSA lib ucm_subs.c:799:(uc_mgr_get_substituted_value) variable '${var:DirectCardName}' is not defined in this context! ALSA lib main.c:1541:(snd_use_case_mgr_open) error: failed to import hw:2 use case configuration -22 alsaucm: error failed to open sound card hw:2: Invalid argument ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/177 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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UCM for Behringer UMC204HD not working
by GitHub issues - opened 24 Jun '22

24 Jun '22

alsa-project/alsa-ucm-conf issue #178 was opened from maru-sama: Similar to #177 I have problem with the ucm setting for the UMC204HD. The problem is that apparently no mixer is being created for the device, when trying to open the card with alsamixer it just fails with cannot load mixer controls. the mixer has been created though if I do not missread the output from alsa-info.sh (attached) [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8977069/… I am using alsa-libs 2.0.7 and the latest ucm from master on ubuntu 22.04 to make sure that I have all the minimum requirements. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/178 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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