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

None
by srinivas.kandagatla＠linaro.org 11 Oct '21

11 Oct '21

On 08/10/2021 16:30, Mark Brown wrote: > On Fri, Oct 08, 2021 at 07:37:12PM +0530, Srinivasa Rao Mandadapu wrote: >> On 9/28/2021 2:01 PM, Srinivas Kandagatla wrote: > >>> In that case you should probably consider using regmap_register_patch() >>> for corrections to the default registers in sm8250 case. > >> As regmap_register_patch() not working in this use case, updating defaults >> in probe for sc7280 specific differences. > > In what way is regmap_register_patch() not working? SM8250 and SC7280 has different register defaults values for few (atleast 2 that we are aware of) registers. Initially we thought regmap_register_patch() would help in this case, but later learned that it was designed for completely different usecase and this will not update the cache values with the new values from reg_sequence passed to regmap_register_patch(). What we want was to be able to reflect that default values in the regcaches, so that subsequent writes are not ignored based on incorrect cache values. There are two ways to fix this, one is to come up with a full default register list specific to SC7280 or reuse SM8250 defaults with some modifications, other option is to update default registers based on compatible string before regmap creation. I suggested Srinivasa to use the second option as 99% of the defaults match to SM8250. --srini >

1 0

[PATCH] ALSA: hda/realtek: Fix for quirk to enable speaker output on the Lenovo 13s Gen2.
by Cameron Berkenpas 11 Oct '21

11 Oct '21

The previous patch's HDA verb initialization for the Lenovo 13s sequence was slightly off. This updated verb sequence has been tested and confirmed working. Fixes: ad7cc2d41b7a ("ALSA: hda/realtek: Quirks to enable speaker output for Lenovo Legion 7i 15IMHG05, Yoga 7i 14ITL5/15ITL5, and 13s Gen2 laptops.") BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Cc: <stable(a)vger.kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..84d3e062a335 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8344,7 +8344,7 @@ static const struct hda_fixup alc269_fixups[] = { .v.verbs = (const struct hda_verb[]) { { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, -- 2.30.2

2 1

None
by srivasam＠codeaurora.org 11 Oct '21

11 Oct '21

On 10/8/2021 9:00 PM, Mark Brown wrote: > On Fri, Oct 08, 2021 at 07:37:12PM +0530, Srinivasa Rao Mandadapu wrote: >> On 9/28/2021 2:01 PM, Srinivas Kandagatla wrote: >>> In that case you should probably consider using regmap_register_patch() >>> for corrections to the default registers in sm8250 case. >> As regmap_register_patch() not working in this use case, updating defaults >> in probe for sc7280 specific differences. > In what way is regmap_register_patch() not working? Even after using this API, reg cache is not updating with the latest default values. The regmap_register_patch() implementation is bypassing Cache and writing values to HW. The requirement for updating previous defaults in cache is not solving here. -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 0

None
by srivasam＠codeaurora.org 11 Oct '21

11 Oct '21

On 10/8/2021 9:00 PM, Mark Brown wrote: > On Fri, Oct 08, 2021 at 07:37:12PM +0530, Srinivasa Rao Mandadapu wrote: >> On 9/28/2021 2:01 PM, Srinivas Kandagatla wrote: >>> In that case you should probably consider using regmap_register_patch() >>> for corrections to the default registers in sm8250 case. >> As regmap_register_patch() not working in this use case, updating defaults >> in probe for sc7280 specific differences. > In what way is regmap_register_patch() not working? Even after using this API, reg cache is not updating with the latest default values. The regmap_register_patch() implementation is bypassing Cache and writing values to HW. The requirement for updating previous defaults in cache is not solving here. -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

1 0

[PATCH 1/3] ASoc: amd: create platform device for VG machine driver
by Vijendar Mukunda 11 Oct '21

11 Oct '21

Create platform device for Vangogh machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/vangogh/acp5x.h | 2 +- sound/soc/amd/vangogh/pci-acp5x.c | 3 +++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index a808635f9740..fe5e1fa98974 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -23,7 +23,7 @@ #define ACP_ERR_INTR_MASK 0x20000000 #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF -#define ACP5x_DEVS 3 +#define ACP5x_DEVS 4 #define ACP5x_REG_START 0x1240000 #define ACP5x_REG_END 0x1250200 #define ACP5x_I2STDM_REG_START 0x1242400 diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index a57b762d9f2e..137e64c47882 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -213,6 +213,9 @@ static int snd_acp5x_probe(struct pci_dev *pci, pdevinfo[2].num_res = 1; pdevinfo[2].res = &adata->res[2]; + pdevinfo[3].name = "acp5x_nu8821_cs35l41_mach"; + pdevinfo[3].id = 0; + pdevinfo[3].parent = &pci->dev; for (i = 0; i < ACP5x_DEVS; i++) { adata->pdev[i] = platform_device_register_full(&pdevinfo[i]); -- 2.25.1

3 6

[PATCH] ALSA: hda/realtek: Fix for quirk to enable speaker output on the Lenovo 13s Gen2.
by Cameron Berkenpas 10 Oct '21

10 Oct '21

The previous patch's HDA verb initialization for the Lenovo 13s sequence was slightly off. This updated verb sequence has been tested and confirmed working. --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..84d3e062a335 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8344,7 +8344,7 @@ static const struct hda_fixup alc269_fixups[] = { .v.verbs = (const struct hda_verb[]) { { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, - { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x2 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, -- 2.30.2

2 1

[PATCH v4 0/4] ASoC: Intel: bytcht_es8316: few cleanups
by Andy Shevchenko 10 Oct '21

10 Oct '21

The small set of cleanups against bytcht_es8316 board file. In v4: - fixed Pierre's email (Pierre) - added Hans to the Cc list In v3: - actually added a Pierre's tag (Mark) In v2: - added tag (Pierre) - added commit message to the patch 2 (Joe) Andy Shevchenko (4): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation sound/soc/intel/boards/bytcht_es8316.c | 37 +++++++++----------------- 1 file changed, 12 insertions(+), 25 deletions(-) -- 2.33.0

2 5

Tenacity throws error when running "build/bin/Debug/tenacity" to run it
by GitHub issues - edited 10 Oct '21

10 Oct '21

alsa-project/alsa-lib issue #184 was edited from User10987654321: context: https://github.com/tenacityteam/tenacity/issues/649 `cat /proc/asound/version` output: Advanced Linux Sound Architecture Driver Version k5.10.0-kali9-amd64 backtrace, which is only produced in a popup once I comment out the lines "pcm.rear #cards.pcm.rear pcm.center_lfe #cards.pcm.center_lfe pcm.side #cards.pcm.side", which produce the following backtrace in a popup labeled "../src/AudioIOBase.cpp(501): assert "false" failed in getRecordDevIndex():": ASSERT INFO: ../src/AudioIOBase.cpp(501): assert "false" failed in getRecordDevIndex(). BACKTRACE: [1] AudioIOBase::getRecordDevIndex(wxString const&) [2] AudioIO::Init() [3] AudacityApp::InitPart2() [4] AudacityApp::OnInit() [5] wxAppConsoleBase::CallOnInit() [6] wxEntry(int&, wchar_t**) [7] wxEntry(int&, char**) [8] main [9] __libc_start_main [10] _sta When going through the steps in https://github.com/tenacityteam/tenacity/blob/master/BUILDING.md, I eventually get to a point where I run the command in the issue title which brings up the tenacity logo on screen and then provides the following errors in the terminal: ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp ALSA lib pcm_a52.c:835:(_snd_pcm_a52_open) a52 is only for playback zsh: segmentation fault ./tenacity/build/bin/Debug/tenacity How can I mitigate this to use tenacity successfully? Issue URL : https://github.com/alsa-project/alsa-lib/issues/184 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Tenacity throws error when running "build/bin/Debug/tenacity" to run it
by GitHub issues - edited 10 Oct '21

10 Oct '21

alsa-project/alsa-lib issue #184 was edited from User10987654321: context: https://github.com/tenacityteam/tenacity/issues/649 `cat /proc/asound/version` output: Advanced Linux Sound Architecture Driver Version k5.10.0-kali9-amd64 When going through the steps in https://github.com/tenacityteam/tenacity/blob/master/BUILDING.md, I eventually get to a point where I run the command in the issue title which brings up the tenacity logo on screen and then provides the following errors in the terminal: ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp ALSA lib pcm_a52.c:835:(_snd_pcm_a52_open) a52 is only for playback zsh: segmentation fault ./tenacity/build/bin/Debug/tenacity How can I mitigate this to use tenacity successfully? Issue URL : https://github.com/alsa-project/alsa-lib/issues/184 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Tenacity throws error when running "build/bin/Debug/tenacity" to run it
by GitHub issues - opened 09 Oct '21

09 Oct '21

alsa-project/alsa-lib issue #184 was opened from User10987654321: context: https://github.com/tenacityteam/tenacity/issues/649 When going through the steps in https://github.com/tenacityteam/tenacity/blob/master/BUILDING.md, I eventually get to a point where I run the command in the issue title which brings up the tenacity logo on screen and then provides the following errors in the terminal: ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe ALSA lib pcm.c:2660:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map ALSA lib pcm_route.c:877:(find_matching_chmap) Found no matching channel map connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed connect(2) call to /dev/shm/jack-1000/default/jack_0 failed (err=No such file or directory) attempt to connect to server failed ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp ALSA lib pcm_oss.c:397:(_snd_pcm_oss_open) Cannot open device /dev/dsp ALSA lib pcm_a52.c:835:(_snd_pcm_a52_open) a52 is only for playback zsh: segmentation fault ./tenacity/build/bin/Debug/tenacity How can I mitigate this to use tenacity successfully? Issue URL : https://github.com/alsa-project/alsa-lib/issues/184 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/8] ASoC: Add common modules support for ACP hw block
by Ajit Kumar Pandey 08 Oct '21

08 Oct '21

We need to support I2S audio on various different platforms SOC in near future. At present we are creating new folder under soc/amd/ for each platforms support with separate I2S and DMA device modules across each SOC. Actually AMD's audio coprocessor block commonly known as ACP hw IP block consist of I2S controllers registers configuration and registers specs are common across all SOC. This patch series add common folder named acp in soc/amd/ that exposes common driver modules to configure I2S and DMA registers across different platforms. Any platforms specific configurations and dai's will be exposed in separate platform modules. We will use SOF based PCI driver or common PCI driver to register and probe platform device modules. We also add common and more generic machine driver support within this patch chain. This machine driver can be used across different machines on AMD SOC with just some minor tweaks related to codecs connections. Ajit Kumar Pandey (8): ASoC: amd: Add common framework to support I2S on ACP SOC ASoC: amd: acp: Add I2S support on Renoir platform ASoC: amd: acp: Add callback for machine driver on ACP ASoC: amd: acp: Add generic machine driver support for ACP cards ASoC: amd: acp: Add legacy sound card support for Guybrush board ASoC: amd: acp: Add SOF sound card support on Guybrush board ASoC: amd: acp: Add support for Maxim amplifier codec ASoC: amd: acp: Add support for RT5682-VS codec sound/soc/amd/Kconfig | 2 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Kconfig | 49 +++ sound/soc/amd/acp/Makefile | 26 ++ sound/soc/amd/acp/acp-i2s.c | 353 +++++++++++++++++ sound/soc/amd/acp/acp-legacy-mach.c | 104 +++++ sound/soc/amd/acp/acp-mach-common.c | 557 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-mach.h | 57 +++ sound/soc/amd/acp/acp-platform.c | 325 ++++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 144 +++++++ sound/soc/amd/acp/acp-sof-mach.c | 103 +++++ sound/soc/amd/acp/amd.h | 146 +++++++ sound/soc/amd/acp/chip_offset_byte.h | 76 ++++ 13 files changed, 1943 insertions(+) create mode 100644 sound/soc/amd/acp/Kconfig create mode 100644 sound/soc/amd/acp/Makefile create mode 100644 sound/soc/amd/acp/acp-i2s.c create mode 100644 sound/soc/amd/acp/acp-legacy-mach.c create mode 100644 sound/soc/amd/acp/acp-mach-common.c create mode 100644 sound/soc/amd/acp/acp-mach.h create mode 100644 sound/soc/amd/acp/acp-platform.c create mode 100644 sound/soc/amd/acp/acp-renoir.c create mode 100644 sound/soc/amd/acp/acp-sof-mach.c create mode 100644 sound/soc/amd/acp/amd.h create mode 100644 sound/soc/amd/acp/chip_offset_byte.h -- 2.25.1

3 12

[PATCH v2 0/5] Update Lpass digital codec macro drivers
by Srinivasa Rao Mandadapu 08 Oct '21

08 Oct '21

This patch set is to add support for lpass sc7280 based targets. Upadate compatible name and change of bulk clock voting to optional clock voting in digital codecs va, rx, tx macro drivers. Chnages Since V1: -- Removed individual clock voting and used bulk clock optional. -- Removed volatile changes and fixed default values. -- Typo errors. Srinivasa Rao Mandadapu (5): ASoC: qcom: Add compatible names in va,wsa,rx,tx codec drivers for sc7280 ASoC: qcom: dt-bindings: Add compatible names for lpass sc7280 digital codecs ASoC: codecs: tx-macro: Enable tx top soundwire mic clock ASoC: codecs: tx-macro: Update tx default values ASoC: codecs: Change bulk clock voting to optional voting in digital codecs .../devicetree/bindings/sound/qcom,lpass-rx-macro.yaml | 4 +++- .../devicetree/bindings/sound/qcom,lpass-tx-macro.yaml | 4 +++- .../devicetree/bindings/sound/qcom,lpass-va-macro.yaml | 4 +++- .../devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml | 4 +++- sound/soc/codecs/lpass-rx-macro.c | 3 ++- sound/soc/codecs/lpass-tx-macro.c | 11 +++++++---- sound/soc/codecs/lpass-va-macro.c | 3 ++- sound/soc/codecs/lpass-wsa-macro.c | 1 + 8 files changed, 24 insertions(+), 10 deletions(-) -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

4 16

alsabat returns 0 when more than 1 signal is found and all are near DC
by GitHub issues - opened 08 Oct '21

08 Oct '21

alsa-project/alsa-utils issue #114 was opened from hollerdawg: alsabat should return -EDCONLY (-1002) when only dc signals can be found and there's more than 1 signal but instead returns 0. Issue URL : https://github.com/alsa-project/alsa-utils/issues/114 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH 1/2][next] ASoC: rockchip: i2s-tdm: Remove call to rockchip_i2s_ch_to_io
by Colin King 08 Oct '21

08 Oct '21

From: Colin Ian King <colin.king(a)canonical.com> The call to rockchip_i2s_ch_to_io is only useful for its return value which is not being used. The function call also has no side effects, the call is effectively useless and can be removed. Addresses-Coverity: ("Useless call") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index b08b15071993..cc5a2f9d3948 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -848,8 +848,6 @@ static int rockchip_i2s_io_multiplex(struct snd_pcm_substream *substream, to_ch_num(val), usable_chs); return -EINVAL; } - - rockchip_i2s_ch_to_io(val, false); } val <<= i2s_tdm->soc_data->grf_shift; -- 2.32.0

1 1

[PATCH] ASoC: SOF: prepare code to allocate IPC messages in fw_ready
by Peter Ujfalusi 08 Oct '21

08 Oct '21

From: Rander Wang <rander.wang(a)intel.com> The fixed maximum size of IPC message does not allow for large transfers, e.g. for filter data. Currently such messages will be divided into smaller pieces and sent to firmware in multiple chunks. For future IPC, this strategy is not suitable. The maximum IPC message size is limited by host box size which can be known when firmware is ready, so the fw_ready callback can allocate IPC messages with platform-specific sizes instead of the current fixed-size. To be compatible with released firmware, current platforms will still use SOF_IPC_MSG_MAX_SIZE. For future platforms, there will be a new fw_ready function and the platform-specific allocation will take place there. Signed-off-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/ipc.c | 27 ++++++++++++++++----------- sound/soc/sof/loader.c | 3 ++- sound/soc/sof/sof-priv.h | 1 + 3 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index b2e556b64994..e6c53c6c470e 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -919,6 +919,22 @@ int snd_sof_ipc_valid(struct snd_sof_dev *sdev) } EXPORT_SYMBOL(snd_sof_ipc_valid); +int sof_ipc_init_msg_memory(struct snd_sof_dev *sdev) +{ + struct snd_sof_ipc_msg *msg; + + msg = &sdev->ipc->msg; + msg->msg_data = devm_kzalloc(sdev->dev, SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!msg->msg_data) + return -ENOMEM; + + msg->reply_data = devm_kzalloc(sdev->dev, SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!msg->reply_data) + return -ENOMEM; + + return 0; +} + struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) { struct snd_sof_ipc *ipc; @@ -935,17 +951,6 @@ struct snd_sof_ipc *snd_sof_ipc_init(struct snd_sof_dev *sdev) /* indicate that we aren't sending a message ATM */ msg->ipc_complete = true; - /* pre-allocate message data */ - msg->msg_data = devm_kzalloc(sdev->dev, SOF_IPC_MSG_MAX_SIZE, - GFP_KERNEL); - if (!msg->msg_data) - return NULL; - - msg->reply_data = devm_kzalloc(sdev->dev, SOF_IPC_MSG_MAX_SIZE, - GFP_KERNEL); - if (!msg->reply_data) - return NULL; - init_waitqueue_head(&msg->waitq); return ipc; diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 54aa7764d2b3..c04646647637 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -13,6 +13,7 @@ #include <linux/firmware.h> #include <sound/sof.h> #include <sound/sof/ext_manifest.h> +#include "sof-priv.h" #include "ops.h" static int get_ext_windows(struct snd_sof_dev *sdev, @@ -517,7 +518,7 @@ int sof_fw_ready(struct snd_sof_dev *sdev, u32 msg_id) sof_get_windows(sdev); - return 0; + return sof_ipc_init_msg_memory(sdev); } EXPORT_SYMBOL(sof_fw_ready); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index efaec2989a27..ba341b1bda0c 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -526,6 +526,7 @@ int sof_ipc_tx_message(struct snd_sof_ipc *ipc, u32 header, int sof_ipc_tx_message_no_pm(struct snd_sof_ipc *ipc, u32 header, void *msg_data, size_t msg_bytes, void *reply_data, size_t reply_bytes); +int sof_ipc_init_msg_memory(struct snd_sof_dev *sdev); /* * Trace/debug -- 2.33.0

1 0

[PATCH] ASoC: mediatek: mt8195: update audsys clock parent name
by Trevor Wu 08 Oct '21

08 Oct '21

Because clock names are modified in mediatek CCF driver, sync the updated clock names to audsys driver. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- The clock patch is accepted in clk-next branch. clk: mediatek: Add MT8195 topckgen clock support https://patchwork.kernel.org/project/linux-mediatek/patch/20210914021633.26… --- sound/soc/mediatek/mt8195/mt8195-audsys-clk.c | 152 +++++++++--------- 1 file changed, 76 insertions(+), 76 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c b/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c index 740aa6ddda0e..e0670e0dbd5b 100644 --- a/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c +++ b/sound/soc/mediatek/mt8195/mt8195-audsys-clk.c @@ -59,93 +59,93 @@ struct afe_gate { static const struct afe_gate aud_clks[CLK_AUD_NR_CLK] = { /* AUD0 */ - GATE_AUD0(CLK_AUD_AFE, "aud_afe", "a1sys_hp_sel", 2), - GATE_AUD0(CLK_AUD_LRCK_CNT, "aud_lrck_cnt", "a1sys_hp_sel", 4), - GATE_AUD0(CLK_AUD_SPDIFIN_TUNER_APLL, "aud_spdifin_tuner_apll", "apll4_sel", 10), - GATE_AUD0(CLK_AUD_SPDIFIN_TUNER_DBG, "aud_spdifin_tuner_dbg", "apll4_sel", 11), - GATE_AUD0(CLK_AUD_UL_TML, "aud_ul_tml", "a1sys_hp_sel", 18), - GATE_AUD0(CLK_AUD_APLL1_TUNER, "aud_apll1_tuner", "apll1_sel", 19), - GATE_AUD0(CLK_AUD_APLL2_TUNER, "aud_apll2_tuner", "apll2_sel", 20), - GATE_AUD0(CLK_AUD_TOP0_SPDF, "aud_top0_spdf", "aud_iec_sel", 21), - GATE_AUD0(CLK_AUD_APLL, "aud_apll", "apll1_sel", 23), - GATE_AUD0(CLK_AUD_APLL2, "aud_apll2", "apll2_sel", 24), - GATE_AUD0(CLK_AUD_DAC, "aud_dac", "a1sys_hp_sel", 25), - GATE_AUD0(CLK_AUD_DAC_PREDIS, "aud_dac_predis", "a1sys_hp_sel", 26), - GATE_AUD0(CLK_AUD_TML, "aud_tml", "a1sys_hp_sel", 27), - GATE_AUD0(CLK_AUD_ADC, "aud_adc", "a1sys_hp_sel", 28), - GATE_AUD0(CLK_AUD_DAC_HIRES, "aud_dac_hires", "audio_h_sel", 31), + GATE_AUD0(CLK_AUD_AFE, "aud_afe", "top_a1sys_hp", 2), + GATE_AUD0(CLK_AUD_LRCK_CNT, "aud_lrck_cnt", "top_a1sys_hp", 4), + GATE_AUD0(CLK_AUD_SPDIFIN_TUNER_APLL, "aud_spdifin_tuner_apll", "top_apll4", 10), + GATE_AUD0(CLK_AUD_SPDIFIN_TUNER_DBG, "aud_spdifin_tuner_dbg", "top_apll4", 11), + GATE_AUD0(CLK_AUD_UL_TML, "aud_ul_tml", "top_a1sys_hp", 18), + GATE_AUD0(CLK_AUD_APLL1_TUNER, "aud_apll1_tuner", "top_apll1", 19), + GATE_AUD0(CLK_AUD_APLL2_TUNER, "aud_apll2_tuner", "top_apll2", 20), + GATE_AUD0(CLK_AUD_TOP0_SPDF, "aud_top0_spdf", "top_aud_iec_clk", 21), + GATE_AUD0(CLK_AUD_APLL, "aud_apll", "top_apll1", 23), + GATE_AUD0(CLK_AUD_APLL2, "aud_apll2", "top_apll2", 24), + GATE_AUD0(CLK_AUD_DAC, "aud_dac", "top_a1sys_hp", 25), + GATE_AUD0(CLK_AUD_DAC_PREDIS, "aud_dac_predis", "top_a1sys_hp", 26), + GATE_AUD0(CLK_AUD_TML, "aud_tml", "top_a1sys_hp", 27), + GATE_AUD0(CLK_AUD_ADC, "aud_adc", "top_a1sys_hp", 28), + GATE_AUD0(CLK_AUD_DAC_HIRES, "aud_dac_hires", "top_audio_h", 31), /* AUD1 */ - GATE_AUD1(CLK_AUD_A1SYS_HP, "aud_a1sys_hp", "a1sys_hp_sel", 2), - GATE_AUD1(CLK_AUD_AFE_DMIC1, "aud_afe_dmic1", "a1sys_hp_sel", 10), - GATE_AUD1(CLK_AUD_AFE_DMIC2, "aud_afe_dmic2", "a1sys_hp_sel", 11), - GATE_AUD1(CLK_AUD_AFE_DMIC3, "aud_afe_dmic3", "a1sys_hp_sel", 12), - GATE_AUD1(CLK_AUD_AFE_DMIC4, "aud_afe_dmic4", "a1sys_hp_sel", 13), - GATE_AUD1(CLK_AUD_AFE_26M_DMIC_TM, "aud_afe_26m_dmic_tm", "a1sys_hp_sel", 14), - GATE_AUD1(CLK_AUD_UL_TML_HIRES, "aud_ul_tml_hires", "audio_h_sel", 16), - GATE_AUD1(CLK_AUD_ADC_HIRES, "aud_adc_hires", "audio_h_sel", 17), - GATE_AUD1(CLK_AUD_ADDA6_ADC, "aud_adda6_adc", "a1sys_hp_sel", 18), - GATE_AUD1(CLK_AUD_ADDA6_ADC_HIRES, "aud_adda6_adc_hires", "audio_h_sel", 19), + GATE_AUD1(CLK_AUD_A1SYS_HP, "aud_a1sys_hp", "top_a1sys_hp", 2), + GATE_AUD1(CLK_AUD_AFE_DMIC1, "aud_afe_dmic1", "top_a1sys_hp", 10), + GATE_AUD1(CLK_AUD_AFE_DMIC2, "aud_afe_dmic2", "top_a1sys_hp", 11), + GATE_AUD1(CLK_AUD_AFE_DMIC3, "aud_afe_dmic3", "top_a1sys_hp", 12), + GATE_AUD1(CLK_AUD_AFE_DMIC4, "aud_afe_dmic4", "top_a1sys_hp", 13), + GATE_AUD1(CLK_AUD_AFE_26M_DMIC_TM, "aud_afe_26m_dmic_tm", "top_a1sys_hp", 14), + GATE_AUD1(CLK_AUD_UL_TML_HIRES, "aud_ul_tml_hires", "top_audio_h", 16), + GATE_AUD1(CLK_AUD_ADC_HIRES, "aud_adc_hires", "top_audio_h", 17), + GATE_AUD1(CLK_AUD_ADDA6_ADC, "aud_adda6_adc", "top_a1sys_hp", 18), + GATE_AUD1(CLK_AUD_ADDA6_ADC_HIRES, "aud_adda6_adc_hires", "top_audio_h", 19), /* AUD3 */ - GATE_AUD3(CLK_AUD_LINEIN_TUNER, "aud_linein_tuner", "apll5_sel", 5), - GATE_AUD3(CLK_AUD_EARC_TUNER, "aud_earc_tuner", "apll3_sel", 7), + GATE_AUD3(CLK_AUD_LINEIN_TUNER, "aud_linein_tuner", "top_apll5", 5), + GATE_AUD3(CLK_AUD_EARC_TUNER, "aud_earc_tuner", "top_apll3", 7), /* AUD4 */ - GATE_AUD4(CLK_AUD_I2SIN, "aud_i2sin", "a1sys_hp_sel", 0), - GATE_AUD4(CLK_AUD_TDM_IN, "aud_tdm_in", "a1sys_hp_sel", 1), - GATE_AUD4(CLK_AUD_I2S_OUT, "aud_i2s_out", "a1sys_hp_sel", 6), - GATE_AUD4(CLK_AUD_TDM_OUT, "aud_tdm_out", "a1sys_hp_sel", 7), - GATE_AUD4(CLK_AUD_HDMI_OUT, "aud_hdmi_out", "a1sys_hp_sel", 8), - GATE_AUD4(CLK_AUD_ASRC11, "aud_asrc11", "a1sys_hp_sel", 16), - GATE_AUD4(CLK_AUD_ASRC12, "aud_asrc12", "a1sys_hp_sel", 17), + GATE_AUD4(CLK_AUD_I2SIN, "aud_i2sin", "top_a1sys_hp", 0), + GATE_AUD4(CLK_AUD_TDM_IN, "aud_tdm_in", "top_a1sys_hp", 1), + GATE_AUD4(CLK_AUD_I2S_OUT, "aud_i2s_out", "top_a1sys_hp", 6), + GATE_AUD4(CLK_AUD_TDM_OUT, "aud_tdm_out", "top_a1sys_hp", 7), + GATE_AUD4(CLK_AUD_HDMI_OUT, "aud_hdmi_out", "top_a1sys_hp", 8), + GATE_AUD4(CLK_AUD_ASRC11, "aud_asrc11", "top_a1sys_hp", 16), + GATE_AUD4(CLK_AUD_ASRC12, "aud_asrc12", "top_a1sys_hp", 17), GATE_AUD4(CLK_AUD_MULTI_IN, "aud_multi_in", "mphone_slave_b", 19), - GATE_AUD4(CLK_AUD_INTDIR, "aud_intdir", "intdir_sel", 20), - GATE_AUD4(CLK_AUD_A1SYS, "aud_a1sys", "a1sys_hp_sel", 21), - GATE_AUD4(CLK_AUD_A2SYS, "aud_a2sys", "a2sys_sel", 22), - GATE_AUD4(CLK_AUD_PCMIF, "aud_pcmif", "a1sys_hp_sel", 24), - GATE_AUD4(CLK_AUD_A3SYS, "aud_a3sys", "a3sys_sel", 30), - GATE_AUD4(CLK_AUD_A4SYS, "aud_a4sys", "a4sys_sel", 31), + GATE_AUD4(CLK_AUD_INTDIR, "aud_intdir", "top_intdir", 20), + GATE_AUD4(CLK_AUD_A1SYS, "aud_a1sys", "top_a1sys_hp", 21), + GATE_AUD4(CLK_AUD_A2SYS, "aud_a2sys", "top_a2sys_hf", 22), + GATE_AUD4(CLK_AUD_PCMIF, "aud_pcmif", "top_a1sys_hp", 24), + GATE_AUD4(CLK_AUD_A3SYS, "aud_a3sys", "top_a3sys_hf", 30), + GATE_AUD4(CLK_AUD_A4SYS, "aud_a4sys", "top_a4sys_hf", 31), /* AUD5 */ - GATE_AUD5(CLK_AUD_MEMIF_UL1, "aud_memif_ul1", "a1sys_hp_sel", 0), - GATE_AUD5(CLK_AUD_MEMIF_UL2, "aud_memif_ul2", "a1sys_hp_sel", 1), - GATE_AUD5(CLK_AUD_MEMIF_UL3, "aud_memif_ul3", "a1sys_hp_sel", 2), - GATE_AUD5(CLK_AUD_MEMIF_UL4, "aud_memif_ul4", "a1sys_hp_sel", 3), - GATE_AUD5(CLK_AUD_MEMIF_UL5, "aud_memif_ul5", "a1sys_hp_sel", 4), - GATE_AUD5(CLK_AUD_MEMIF_UL6, "aud_memif_ul6", "a1sys_hp_sel", 5), - GATE_AUD5(CLK_AUD_MEMIF_UL8, "aud_memif_ul8", "a1sys_hp_sel", 7), - GATE_AUD5(CLK_AUD_MEMIF_UL9, "aud_memif_ul9", "a1sys_hp_sel", 8), - GATE_AUD5(CLK_AUD_MEMIF_UL10, "aud_memif_ul10", "a1sys_hp_sel", 9), - GATE_AUD5(CLK_AUD_MEMIF_DL2, "aud_memif_dl2", "a1sys_hp_sel", 18), - GATE_AUD5(CLK_AUD_MEMIF_DL3, "aud_memif_dl3", "a1sys_hp_sel", 19), - GATE_AUD5(CLK_AUD_MEMIF_DL6, "aud_memif_dl6", "a1sys_hp_sel", 22), - GATE_AUD5(CLK_AUD_MEMIF_DL7, "aud_memif_dl7", "a1sys_hp_sel", 23), - GATE_AUD5(CLK_AUD_MEMIF_DL8, "aud_memif_dl8", "a1sys_hp_sel", 24), - GATE_AUD5(CLK_AUD_MEMIF_DL10, "aud_memif_dl10", "a1sys_hp_sel", 26), - GATE_AUD5(CLK_AUD_MEMIF_DL11, "aud_memif_dl11", "a1sys_hp_sel", 27), + GATE_AUD5(CLK_AUD_MEMIF_UL1, "aud_memif_ul1", "top_a1sys_hp", 0), + GATE_AUD5(CLK_AUD_MEMIF_UL2, "aud_memif_ul2", "top_a1sys_hp", 1), + GATE_AUD5(CLK_AUD_MEMIF_UL3, "aud_memif_ul3", "top_a1sys_hp", 2), + GATE_AUD5(CLK_AUD_MEMIF_UL4, "aud_memif_ul4", "top_a1sys_hp", 3), + GATE_AUD5(CLK_AUD_MEMIF_UL5, "aud_memif_ul5", "top_a1sys_hp", 4), + GATE_AUD5(CLK_AUD_MEMIF_UL6, "aud_memif_ul6", "top_a1sys_hp", 5), + GATE_AUD5(CLK_AUD_MEMIF_UL8, "aud_memif_ul8", "top_a1sys_hp", 7), + GATE_AUD5(CLK_AUD_MEMIF_UL9, "aud_memif_ul9", "top_a1sys_hp", 8), + GATE_AUD5(CLK_AUD_MEMIF_UL10, "aud_memif_ul10", "top_a1sys_hp", 9), + GATE_AUD5(CLK_AUD_MEMIF_DL2, "aud_memif_dl2", "top_a1sys_hp", 18), + GATE_AUD5(CLK_AUD_MEMIF_DL3, "aud_memif_dl3", "top_a1sys_hp", 19), + GATE_AUD5(CLK_AUD_MEMIF_DL6, "aud_memif_dl6", "top_a1sys_hp", 22), + GATE_AUD5(CLK_AUD_MEMIF_DL7, "aud_memif_dl7", "top_a1sys_hp", 23), + GATE_AUD5(CLK_AUD_MEMIF_DL8, "aud_memif_dl8", "top_a1sys_hp", 24), + GATE_AUD5(CLK_AUD_MEMIF_DL10, "aud_memif_dl10", "top_a1sys_hp", 26), + GATE_AUD5(CLK_AUD_MEMIF_DL11, "aud_memif_dl11", "top_a1sys_hp", 27), /* AUD6 */ - GATE_AUD6(CLK_AUD_GASRC0, "aud_gasrc0", "asm_h_sel", 0), - GATE_AUD6(CLK_AUD_GASRC1, "aud_gasrc1", "asm_h_sel", 1), - GATE_AUD6(CLK_AUD_GASRC2, "aud_gasrc2", "asm_h_sel", 2), - GATE_AUD6(CLK_AUD_GASRC3, "aud_gasrc3", "asm_h_sel", 3), - GATE_AUD6(CLK_AUD_GASRC4, "aud_gasrc4", "asm_h_sel", 4), - GATE_AUD6(CLK_AUD_GASRC5, "aud_gasrc5", "asm_h_sel", 5), - GATE_AUD6(CLK_AUD_GASRC6, "aud_gasrc6", "asm_h_sel", 6), - GATE_AUD6(CLK_AUD_GASRC7, "aud_gasrc7", "asm_h_sel", 7), - GATE_AUD6(CLK_AUD_GASRC8, "aud_gasrc8", "asm_h_sel", 8), - GATE_AUD6(CLK_AUD_GASRC9, "aud_gasrc9", "asm_h_sel", 9), - GATE_AUD6(CLK_AUD_GASRC10, "aud_gasrc10", "asm_h_sel", 10), - GATE_AUD6(CLK_AUD_GASRC11, "aud_gasrc11", "asm_h_sel", 11), - GATE_AUD6(CLK_AUD_GASRC12, "aud_gasrc12", "asm_h_sel", 12), - GATE_AUD6(CLK_AUD_GASRC13, "aud_gasrc13", "asm_h_sel", 13), - GATE_AUD6(CLK_AUD_GASRC14, "aud_gasrc14", "asm_h_sel", 14), - GATE_AUD6(CLK_AUD_GASRC15, "aud_gasrc15", "asm_h_sel", 15), - GATE_AUD6(CLK_AUD_GASRC16, "aud_gasrc16", "asm_h_sel", 16), - GATE_AUD6(CLK_AUD_GASRC17, "aud_gasrc17", "asm_h_sel", 17), - GATE_AUD6(CLK_AUD_GASRC18, "aud_gasrc18", "asm_h_sel", 18), - GATE_AUD6(CLK_AUD_GASRC19, "aud_gasrc19", "asm_h_sel", 19), + GATE_AUD6(CLK_AUD_GASRC0, "aud_gasrc0", "top_asm_h", 0), + GATE_AUD6(CLK_AUD_GASRC1, "aud_gasrc1", "top_asm_h", 1), + GATE_AUD6(CLK_AUD_GASRC2, "aud_gasrc2", "top_asm_h", 2), + GATE_AUD6(CLK_AUD_GASRC3, "aud_gasrc3", "top_asm_h", 3), + GATE_AUD6(CLK_AUD_GASRC4, "aud_gasrc4", "top_asm_h", 4), + GATE_AUD6(CLK_AUD_GASRC5, "aud_gasrc5", "top_asm_h", 5), + GATE_AUD6(CLK_AUD_GASRC6, "aud_gasrc6", "top_asm_h", 6), + GATE_AUD6(CLK_AUD_GASRC7, "aud_gasrc7", "top_asm_h", 7), + GATE_AUD6(CLK_AUD_GASRC8, "aud_gasrc8", "top_asm_h", 8), + GATE_AUD6(CLK_AUD_GASRC9, "aud_gasrc9", "top_asm_h", 9), + GATE_AUD6(CLK_AUD_GASRC10, "aud_gasrc10", "top_asm_h", 10), + GATE_AUD6(CLK_AUD_GASRC11, "aud_gasrc11", "top_asm_h", 11), + GATE_AUD6(CLK_AUD_GASRC12, "aud_gasrc12", "top_asm_h", 12), + GATE_AUD6(CLK_AUD_GASRC13, "aud_gasrc13", "top_asm_h", 13), + GATE_AUD6(CLK_AUD_GASRC14, "aud_gasrc14", "top_asm_h", 14), + GATE_AUD6(CLK_AUD_GASRC15, "aud_gasrc15", "top_asm_h", 15), + GATE_AUD6(CLK_AUD_GASRC16, "aud_gasrc16", "top_asm_h", 16), + GATE_AUD6(CLK_AUD_GASRC17, "aud_gasrc17", "top_asm_h", 17), + GATE_AUD6(CLK_AUD_GASRC18, "aud_gasrc18", "top_asm_h", 18), + GATE_AUD6(CLK_AUD_GASRC19, "aud_gasrc19", "top_asm_h", 19), }; int mt8195_audsys_clk_register(struct mtk_base_afe *afe) -- 2.18.0

1 0

[PATCH] ASoC: qcom: soundwire: Enable soundwire bus clock for version 1.6
by Srinivasa Rao Mandadapu 08 Oct '21

08 Oct '21

Add support for soundwire 1.6 version to gate RX/TX bus clock. Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- drivers/soundwire/qcom.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 0ef79d6..599b3ed 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -127,6 +127,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + char __iomem *swrm_hctl_reg; struct completion broadcast; struct completion enumeration; struct work_struct slave_work; @@ -610,6 +611,12 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + if (ctrl->swrm_hctl_reg) { + val = ioread32(ctrl->swrm_hctl_reg); + val &= 0xFFFFFFFD; + iowrite32(val, ctrl->swrm_hctl_reg); + } + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1200,7 +1207,7 @@ static int qcom_swrm_probe(struct platform_device *pdev) struct qcom_swrm_ctrl *ctrl; const struct qcom_swrm_data *data; int ret; - u32 val; + int val, swrm_hctl_reg = 0; ctrl = devm_kzalloc(dev, sizeof(*ctrl), GFP_KERNEL); if (!ctrl) @@ -1251,6 +1258,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + if (!of_property_read_u32(dev->of_node, "qcom,swrm-hctl-reg", &swrm_hctl_reg)) + ctrl->swrm_hctl_reg = devm_ioremap(&pdev->dev, swrm_hctl_reg, 0x4); + ret = qcom_swrm_get_port_config(ctrl); if (ret) goto err_clk; -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 6

[PATCH v2] ASoC: dt-bindings: lpass: add binding headers for digital codecs
by Srinivasa Rao Mandadapu 08 Oct '21

08 Oct '21

Add header defining for lpass internal digital codecs rx,tx and va dai node id's. Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes since v1: -- Add missing dai node ID's include/dt-bindings/sound/qcom,lpass.h | 31 +++++++++++++++++++++++++++++++ 1 file changed, 31 insertions(+) diff --git a/include/dt-bindings/sound/qcom,lpass.h b/include/dt-bindings/sound/qcom,lpass.h index 7b0b80b..a9404c3 100644 --- a/include/dt-bindings/sound/qcom,lpass.h +++ b/include/dt-bindings/sound/qcom,lpass.h @@ -10,6 +10,37 @@ #define LPASS_DP_RX 5 +#define LPASS_CDC_DMA_RX0 6 +#define LPASS_CDC_DMA_RX1 7 +#define LPASS_CDC_DMA_RX2 8 +#define LPASS_CDC_DMA_RX3 9 +#define LPASS_CDC_DMA_RX4 10 +#define LPASS_CDC_DMA_RX5 11 +#define LPASS_CDC_DMA_RX6 12 +#define LPASS_CDC_DMA_RX7 13 +#define LPASS_CDC_DMA_RX8 14 +#define LPASS_CDC_DMA_RX9 15 + +#define LPASS_CDC_DMA_TX0 16 +#define LPASS_CDC_DMA_TX1 17 +#define LPASS_CDC_DMA_TX2 18 +#define LPASS_CDC_DMA_TX3 19 +#define LPASS_CDC_DMA_TX4 20 +#define LPASS_CDC_DMA_TX5 21 +#define LPASS_CDC_DMA_TX6 22 +#define LPASS_CDC_DMA_TX7 23 +#define LPASS_CDC_DMA_TX8 24 + +#define LPASS_CDC_DMA_VA_TX0 25 +#define LPASS_CDC_DMA_VA_TX1 26 +#define LPASS_CDC_DMA_VA_TX2 27 +#define LPASS_CDC_DMA_VA_TX3 28 +#define LPASS_CDC_DMA_VA_TX4 29 +#define LPASS_CDC_DMA_VA_TX5 30 +#define LPASS_CDC_DMA_VA_TX6 31 +#define LPASS_CDC_DMA_VA_TX7 32 +#define LPASS_CDC_DMA_VA_TX8 33 + #define LPASS_MCLK0 0 #endif /* __DT_QCOM_LPASS_H */ -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 3

[RESEND PATCH v4] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 08 Oct '21

08 Oct '21

The document was merged as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [1] revealed that the commit 1afc60e00de3 breaks dt_binding_check due to dt-bindings/clock/mt8192-clk.h doesn't exist. As a temporary fix, commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") reverted commit 1afc60e00de3. dt-bindings/clock/mt8192-clk.h is in mainline per commit f35f1a23e0e1 ("clk: mediatek: Add dt-bindings of MT8192 clocks"). Re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Changes from v3 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-September/189705…): - Fix the indent errors. Changes from v2 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189058.ht…): - Simplify the commit message. Changes from v1 (https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189048.ht…): - Add more context to the commit message. .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..5e9fe067f440 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.882.g93a45727a2-goog

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Re: [PATCH 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by ChiYuan Huang 08 Oct '21

08 Oct '21

Hi, Mark: ChiYuan Huang <u0084500(a)gmail.com> 於 2021年10月5日週二下午8:39寫道： > > Mark Brown <broonie(a)kernel.org> 於 2021年10月5日週二下午8:29寫道： > > > > On Tue, Oct 05, 2021 at 08:25:43PM +0800, ChiYuan Huang wrote: > > > Mark Brown <broonie(a)kernel.org> 於 2021年10月5日週二下午7:49寫道： > > > > On Tue, Oct 05, 2021 at 09:16:54AM +0800, cy_huang wrote: > > > > > > > + richtek,use-dvdd-1p8v: > > > > > + description: Indicate DVDD 1P8V is used, default for 3P3V or 5V design > > > > > + type: boolean > > > > > > I would expect this to be done through the regulator bindings, they > > > > would allow the driver to query the supply voltage. > > > > > It's more like as the I/O pad voltage. > > > Must be the same as I2C and I2S signal high level. > > > It depends on the application SOC design. > > > From my understanding, not all application SOC I/O voltage uses > > > regulator interface. > > > > It doesn't really matter what the SoC is doing here, you can always add > > regulator support to your device - you'd be requesting the supplies to > > your device, if the SoC doesn't request the supplies that go to it that > > doesn't really make a difference to what your driver does. > > > > Please don't take things off-list unless there is a really strong reason > > to do so. Sending things to the list ensures that everyone gets a > > chance to read and comment on things. > After contacting our HW RD, to support DVDD 1.8V not just SW config, also HW connections. To get only DVDD supply voltage is not enough to meet the HW design. The property seems indeed and need to be used by user's HW connection. Can this property to be kept? > Sorry, my fault. > I just noticed the mail not reply all. Loop all again.

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[PATCH] ASoC: wcd938x: Fix jack detection issue
by Srinivasa Rao Mandadapu 08 Oct '21

08 Oct '21

This patch is to fix audio 3.5mm jack detection failure on wcd938x codec based target. Fixes: bcee7ed09b8e (ASoC: codecs: wcd938x: add Multi Button Headset Control support) Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/codecs/wcd938x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index f0daf8d..52de7d1 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -4144,10 +4144,10 @@ static int wcd938x_codec_set_jack(struct snd_soc_component *comp, { struct wcd938x_priv *wcd = dev_get_drvdata(comp->dev); - if (!jack) + if (jack) return wcd_mbhc_start(wcd->wcd_mbhc, &wcd->mbhc_cfg, jack); - - wcd_mbhc_stop(wcd->wcd_mbhc); + else + wcd_mbhc_stop(wcd->wcd_mbhc); return 0; } -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH 0/4] Introduce new SOF helpers
by Daniel Baluta 08 Oct '21

08 Oct '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries adds new helpers in order to reduce code duplication and prepare for compress audio support with SOF. Bud Liviu-Alexandru (1): ASoC: SOF: Make Intel IPC stream ops generic Daniel Baluta (3): ASoC: SOF: Introduce snd_sof_mailbox_read / snd_sof_mailbox_write callbacks ASoC: SOF: imx: Use newly introduced generic IPC stream ops ASoC: SOF: Introduce fragment elapsed notification API sound/soc/sof/Kconfig | 4 ++ sound/soc/sof/Makefile | 3 +- sound/soc/sof/compress.c | 51 +++++++++++++++++++ sound/soc/sof/imx/Kconfig | 2 + sound/soc/sof/imx/imx8.c | 39 +++++++------- sound/soc/sof/imx/imx8m.c | 26 ++++------ sound/soc/sof/intel/Makefile | 3 -- sound/soc/sof/intel/apl.c | 4 ++ sound/soc/sof/intel/bdw.c | 12 +++-- sound/soc/sof/intel/byt.c | 24 ++++++--- sound/soc/sof/intel/cnl.c | 4 ++ sound/soc/sof/intel/icl.c | 4 ++ sound/soc/sof/intel/pci-tng.c | 12 +++-- sound/soc/sof/intel/tgl.c | 4 ++ sound/soc/sof/ipc.c | 6 ++- sound/soc/sof/ops.h | 15 ++++++ sound/soc/sof/pcm.c | 7 ++- sound/soc/sof/sof-audio.h | 11 +++- sound/soc/sof/sof-priv.h | 28 ++++++---- .../sof/{intel/intel-ipc.c => stream-ipc.c} | 50 +++++++++--------- sound/soc/sof/topology.c | 6 ++- 21 files changed, 218 insertions(+), 97 deletions(-) create mode 100644 sound/soc/sof/compress.c rename sound/soc/sof/{intel/intel-ipc.c => stream-ipc.c} (58%) -- 2.27.0

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[PATCH] ASoC: rt5682s: Fix hp pop produced immediately after resuming
by derek.fang＠realtek.com 08 Oct '21

08 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> When the system plays a sound immediately after resuming from S3, it could hear a little pop from headphones. It is due to the HP was unmuted before the completion of jack re-detection finished in parallel. This patch adds a lock to make sure the HP unmute after jack detect handler, and adds a few depop changes. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 27 ++++++++++++++++++++------- sound/soc/codecs/rt5682s.h | 6 ++++++ 2 files changed, 26 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 639a50fa95e2..9dd1796f3339 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -790,6 +790,7 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ snd_soc_component_update_bits(component, RT5682S_HP_CHARGE_PUMP_2, RT5682S_OSW_L_MASK | RT5682S_OSW_R_MASK, RT5682S_OSW_L_EN | RT5682S_OSW_R_EN); + usleep_range(35000, 40000); } else { rt5682s_sar_power_mode(component, SAR_PWR_OFF, 1); rt5682s_disable_push_button_irq(component); @@ -829,6 +830,7 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) while (!rt5682s->component->card->instantiated) usleep_range(10000, 15000); + mutex_lock(&rt5682s->jdet_mutex); mutex_lock(&rt5682s->calibrate_mutex); val = snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) @@ -896,6 +898,7 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) cancel_delayed_work_sync(&rt5682s->jd_check_work); mutex_unlock(&rt5682s->calibrate_mutex); + mutex_unlock(&rt5682s->jdet_mutex); } static void rt5682s_jd_check_handler(struct work_struct *work) @@ -1318,34 +1321,41 @@ static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_component_update_bits(component, RT5682S_DEPOP_1, RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN, RT5682S_OUT_HP_L_EN | RT5682S_OUT_HP_R_EN); + usleep_range(15000, 20000); snd_soc_component_update_bits(component, RT5682S_DEPOP_1, RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN); - break; - - case SND_SOC_DAPM_POST_PMU: - usleep_range(30000, 35000); snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_11, 0x6666); snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_12, 0xa82a); + + mutex_lock(&rt5682s->jdet_mutex); + snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | RT5682S_HPO_SEL_IP_EN_SW, RT5682S_HPO_L_PATH_EN | RT5682S_HPO_R_PATH_EN | RT5682S_HPO_IP_EN_GATING); - snd_soc_component_write(component, RT5682S_HP_AMP_DET_CTL_1, 0x3050); + usleep_range(5000, 10000); + snd_soc_component_update_bits(component, RT5682S_HP_AMP_DET_CTL_1, + RT5682S_CP_SW_SIZE_MASK, RT5682S_CP_SW_SIZE_L | RT5682S_CP_SW_SIZE_S); + + mutex_unlock(&rt5682s->jdet_mutex); break; case SND_SOC_DAPM_POST_PMD: snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | RT5682S_HPO_SEL_IP_EN_SW, 0); + snd_soc_component_update_bits(component, RT5682S_HP_AMP_DET_CTL_1, + RT5682S_CP_SW_SIZE_MASK, RT5682S_CP_SW_SIZE_M); snd_soc_component_update_bits(component, RT5682S_DEPOP_1, RT5682S_LDO_PUMP_EN | RT5682S_PUMP_EN | RT5682S_CAPLESS_L_EN | RT5682S_CAPLESS_R_EN, 0); @@ -1734,7 +1744,7 @@ static const struct snd_soc_dapm_widget rt5682s_dapm_widgets[] = { /* HPO */ SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5682s_hp_amp_event, - SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), /* CLK DET */ SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5682S_CLK_DET, @@ -3061,6 +3071,7 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c, mutex_init(&rt5682s->calibrate_mutex); mutex_init(&rt5682s->sar_mutex); + mutex_init(&rt5682s->jdet_mutex); rt5682s_calibrate(rt5682s); regmap_update_bits(rt5682s->regmap, RT5682S_MICBIAS_2, @@ -3072,6 +3083,8 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c, RT5682S_HP_SIG_SRC_MASK, RT5682S_HP_SIG_SRC_1BIT_CTL); regmap_update_bits(rt5682s->regmap, RT5682S_HP_CHARGE_PUMP_2, RT5682S_PM_HP_MASK, RT5682S_PM_HP_HV); + regmap_update_bits(rt5682s->regmap, RT5682S_HP_AMP_DET_CTL_1, + RT5682S_CP_SW_SIZE_MASK, RT5682S_CP_SW_SIZE_M); /* DMIC data pin */ switch (rt5682s->pdata.dmic1_data_pin) { diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 59ba4ea3b062..1bf2ef7ce578 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1366,6 +1366,11 @@ #define RT5682S_SAR_SOUR_BTN (0x3f) #define RT5682S_SAR_SOUR_TYPE (0x0) +/* Headphone Amp Detection Control 1 (0x3b00) */ +#define RT5682S_CP_SW_SIZE_MASK (0x7 << 4) +#define RT5682S_CP_SW_SIZE_L (0x4 << 4) +#define RT5682S_CP_SW_SIZE_M (0x2 << 4) +#define RT5682S_CP_SW_SIZE_S (0x1 << 4) #define RT5682S_STEREO_RATES SNDRV_PCM_RATE_8000_192000 #define RT5682S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ @@ -1441,6 +1446,7 @@ struct rt5682s_priv { struct delayed_work jd_check_work; struct mutex calibrate_mutex; struct mutex sar_mutex; + struct mutex jdet_mutex; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682S_DAI_NUM_CLKS]; -- 2.17.1

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[PATCH v5 0/4] Rockchip I2S/TDM controller
by Nicolas Frattaroli 08 Oct '21

08 Oct '21

Hello, this is version 5 of the I2S/TDM driver patchset. A big thanks to everyone who has provided their valuable feedback so far. Changes in v5: driver: - change comment style of the first comment to C++ style - make refcount non-atomic, as it's only ever used inside a spinlock - use newer SND_SOC_DAIFMT_CB* defines - change ternary statements to if/else conditions - make _clk_compensation_put return 1 if clock changed - implement set_bclk_ratio callback - always set half frame sync mode in TDM mode - automatically enable mclk-calibrate mode when the clocks for it are specified in the device tree bindings: - add Reviewed-by: Rob Herring - drop rockchip,frame-width property (done by set_bclk_ratio) - drop rockchip,fsync-half-frame property - drop rockchip,mclk-calibrate property dts: - drop empty codec block from Quartz64 device tree Changes in v4: driver: - factor TDE/RDE enable/disable into their own inlined functions - add an RDE disable in a location where it looks like it was forgotten (rxctrl else), judging by corresponding TDE code - remove parentheses around CLK_PPM_MIN/MAX values - wording + titlecasing in the clock compensation control - use if statement in precious_reg instead - refactor rockchip_i2s_io_multiplex to have the switch statements in a function call to make the function less unwieldy - get rid of IS_ERR checks around clk enable/disable calls where already checked before by the probe - reworded some error message strings - fix potential deadlock in txrxctrl found by Sugar Zhang using spin_lock_irqsave - fix potential deadlock in trcm_mode found by Sugar Zhang using spin_lock_irqsave - use devm_platform_get_and_ioremap_resource in probe - only set DMA things if controller has capture/playback ability. Did not move this into init_dai because I'd then need to pass in the res and probe it earlier in the function, and it's also used elsewhere in the probe function - use _get_optional_exclusive for reset controls, as some controllers only have capture or playback capability bindings: - remove status = "okay" since that's the default - change the path configs to be an enum - rename "foo" to "bus" - make resets optional as controller may lack either playback or capture capability, and therefore also doesn't have a reset. At least one reset is still required, because a controller with no playback and no capture is not very useful Changes in v3: driver: - alphabetically sort includes - check pm_runtime_get_sync return value, act on it - remove unnecessary initialisers in set_fmt - use udelay(15) in retry code: 10 retries * 15 = 150, so at worst we wait the full i2s register access delay - fix some weird returns to return directly - use __maybe_unused instead of #ifdef CONFIG_PM_SLEEP, also put __maybe_unused on the runtime callbacks - use (foo) instead of foo in header macros for precedence reasons - when using mclk-calibrate, also turn off/on those clocks during suspend and resume operations - remove mclk_tx and mclk_rx reenablement code in remove - move hclk enablement further down the probe, and disable it on probe failure - make reset controls mandatory, since the bindings state this too - use _exclusive for getting the reset controls - change reset assert/deassert delays to both be 10 usec (thank you Sugar Zhang!) - properly prepare and enable all mclks in probe, especially before calling clk_get_rate on them - if registering PCM fails, also use the cleanup error path instead of returning directly - bring back playback and capture only but in the way Sugar Zhang suggested it: set those modes depending on dma-names - rework clock enablement in general. Probe now always enables these, instead of relying on the pm resume thing - add myself to MAINTAINERS for this driver dt bindings: - fix a description still mentioning clk-trcm in the schema - document rockchip,io-multiplex, a property that describes the hardware as having multiplexed I2S GPIOs so direction needs to be changed dynamically - document rockchip,mclk-calibrate, which allows specifying different clocks for the two sample rate bases and switch between them as needed - dma-names now doesn't have a set order and items can be absent to indicate that the controller doesn't support this mode - add myself to MAINTAINERS for these bindings Changes in v2: - remove ad-hoc writeq and needless (and broken) optimisation in reset assert/deassert. This wouldn't have worked on Big Endian, and would've been pointless on any other platform, as the overhead for saving one write was comparatively big - fix various checkpatch issues - get rid of leftover clk-trcm in schema - set status = "okay" in example in schema instead of "disabled" - change dma-names so rx is first, adjust device trees as necessary - properly reference uint32-array for rx-route and tx-route instead of uint32 - replace trcm-sync with two boolean properties, adjust DT changes accordingly and also get rid of the header file - get rid of rockchip,no-dmaengine. This was only needed for some downstream driver and shouldn't be in the DT - get rid of rockchip,capture-only/playback-only. Rationale being that I have no way to test whether they're needed, and unconditionally setting channels_min to 0 breaks everything - change hclk description in "clocks" Nicolas Frattaroli (4): ASoC: rockchip: add support for i2s-tdm controller ASoC: dt-bindings: rockchip: add i2s-tdm bindings arm64: dts: rockchip: add i2s1 on rk356x arm64: dts: rockchip: add analog audio on Quartz64 .../bindings/sound/rockchip,i2s-tdm.yaml | 198 ++ MAINTAINERS | 7 + .../boot/dts/rockchip/rk3566-quartz64-a.dts | 31 +- arch/arm64/boot/dts/rockchip/rk356x.dtsi | 26 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_i2s_tdm.c | 1848 +++++++++++++++++ sound/soc/rockchip/rockchip_i2s_tdm.h | 398 ++++ 8 files changed, 2520 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,i2s-tdm.yaml create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.c create mode 100644 sound/soc/rockchip/rockchip_i2s_tdm.h -- 2.33.0

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[PATCH 00/19] ASoC: SOF: Improvements for debugging
by Peter Ujfalusi 08 Oct '21

08 Oct '21

Hi, The aim of this series is to clean up, make it easier to interpret and less 'chatty' prints aimed for debugging errors. For example currently the DSP/IPC dump is printed every time we have an IPC timeout and it is posible to lost the first and more indicative dump to find the rootcause. Regards, Peter --- Peter Ujfalusi (18): ASoC: SOF: debug: Swap the dsp_dump and ipc_dump sequence for fw_exception ASoC: SOF: ipc and dsp dump: Add markers for better visibility ASoC: SOF: Print the dbg_dump and ipc_dump once to reduce kernel log noise ASoC: SOF: loader: Print the DSP dump if boot fails ASoC: SOF: intel: atom: No need to do a DSP dump in atom_run() ASoC: SOF: debug/ops: Move the IPC and DSP dump functions out from the header ASoC: SOF: debug: Add SOF_DBG_DUMP_OPTIONAL flag for DSP dumping ASoC: SOF: intel: hda-loader: Use snd_sof_dsp_dbg_dump() for DSP dump ASoC: SOF: Drop SOF_DBG_DUMP_FORCE_ERR_LEVEL and sof_dev_dbg_or_err ASoC: SOF: debug: Print out the fw_state along with the DSP dump ASoC: SOF: ipc: Re-enable dumps after successful IPC tx ASoC: SOF: ops: Force DSP panic dumps to be printed ASoC: SOF: Introduce macro to set the firmware state ASoC: SOF: intel: hda: Drop 'error' prefix from error dump functions ASoC: SOF: core: Clean up snd_sof_get_status() prints ASoC: SOF: loader: Drop SOF_DBG_DUMP_REGS flag when firmware start fails ASoC: SOF: Intel: hda-loader: Drop SOF_DBG_DUMP_REGS flag from dbg_dump calls ASoC: SOF: Intel: hda: Dump registers and stack when SOF_DBG_DUMP_REGS is set Pierre-Louis Bossart (1): ASoC: SOF: core: debug: force all processing on primary core sound/soc/sof/core.c | 24 ++++++------- sound/soc/sof/debug.c | 61 ++++++++++++++++++++++++++++++-- sound/soc/sof/intel/atom.c | 5 +-- sound/soc/sof/intel/hda-loader.c | 11 +++--- sound/soc/sof/intel/hda.c | 16 +++------ sound/soc/sof/ipc.c | 10 ++++-- sound/soc/sof/loader.c | 11 ++++-- sound/soc/sof/ops.c | 3 ++ sound/soc/sof/ops.h | 12 +------ sound/soc/sof/pm.c | 6 ++-- sound/soc/sof/sof-priv.h | 31 ++++++++++------ sound/soc/sof/topology.c | 6 ++++ 12 files changed, 131 insertions(+), 65 deletions(-) -- 2.33.0

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[PATCH] ASoC: SOF: pipelines: Harmonize all functions to use struct snd_sof_dev
by Peter Ujfalusi 08 Oct '21

08 Oct '21

First thing the pipelines function which have "struct device *dev" as parameter do is: struct snd_sof_dev *sdev = dev_get_drvdata(dev); and in all cases the passed dev is actually coming from sdev->dev. Skip this steps and pass directly the sdev to all pipelines related functions as few of them already does this. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/pm.c | 4 ++-- sound/soc/sof/sof-audio.c | 12 +++++------- sound/soc/sof/sof-audio.h | 8 ++++---- sound/soc/sof/topology.c | 16 +++++++--------- 4 files changed, 18 insertions(+), 22 deletions(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index 77a3496d3dbd..e65f4f4d6df9 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -157,7 +157,7 @@ static int sof_resume(struct device *dev, bool runtime_resume) } /* restore pipelines */ - ret = sof_set_up_pipelines(sdev->dev, false); + ret = sof_set_up_pipelines(sdev, false); if (ret < 0) { dev_err(sdev->dev, "error: failed to restore pipeline after resume %d\n", @@ -208,7 +208,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) if (target_state == SOF_DSP_PM_D0) goto suspend; - sof_tear_down_pipelines(dev, false); + sof_tear_down_pipelines(sdev, false); /* release trace */ snd_sof_release_trace(sdev); diff --git a/sound/soc/sof/sof-audio.c b/sound/soc/sof/sof-audio.c index 262cb3ad4674..7cbe757c1fe2 100644 --- a/sound/soc/sof/sof-audio.c +++ b/sound/soc/sof/sof-audio.c @@ -203,7 +203,7 @@ int sof_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) break; case snd_soc_dapm_scheduler: pipeline = swidget->private; - ret = sof_load_pipeline_ipc(sdev->dev, pipeline, &r); + ret = sof_load_pipeline_ipc(sdev, pipeline, &r); break; default: hdr = swidget->private; @@ -428,7 +428,7 @@ int sof_widget_list_setup(struct snd_sof_dev *sdev, struct snd_sof_pcm *spcm, in if (pipe_widget->complete) continue; - pipe_widget->complete = snd_sof_complete_pipeline(sdev->dev, pipe_widget); + pipe_widget->complete = snd_sof_complete_pipeline(sdev, pipe_widget); if (pipe_widget->complete < 0) { ret = pipe_widget->complete; goto widget_free; @@ -593,9 +593,8 @@ const struct sof_ipc_pipe_new *snd_sof_pipeline_find(struct snd_sof_dev *sdev, return NULL; } -int sof_set_up_pipelines(struct device *dev, bool verify) +int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct snd_sof_widget *swidget; struct snd_sof_route *sroute; int ret; @@ -654,7 +653,7 @@ int sof_set_up_pipelines(struct device *dev, bool verify) continue; swidget->complete = - snd_sof_complete_pipeline(dev, swidget); + snd_sof_complete_pipeline(sdev, swidget); break; default: break; @@ -668,9 +667,8 @@ int sof_set_up_pipelines(struct device *dev, bool verify) * This function doesn't free widgets during suspend. It only resets the set up status for all * routes and use_count for all widgets. */ -int sof_tear_down_pipelines(struct device *dev, bool verify) +int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct snd_sof_widget *swidget; struct snd_sof_route *sroute; int ret; diff --git a/sound/soc/sof/sof-audio.h b/sound/soc/sof/sof-audio.h index 149b3dbcddd1..fe2ffe02fdfb 100644 --- a/sound/soc/sof/sof-audio.h +++ b/sound/soc/sof/sof-audio.h @@ -181,10 +181,10 @@ void snd_sof_control_notify(struct snd_sof_dev *sdev, * be freed by snd_soc_unregister_component, */ int snd_sof_load_topology(struct snd_soc_component *scomp, const char *file); -int snd_sof_complete_pipeline(struct device *dev, +int snd_sof_complete_pipeline(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget); -int sof_load_pipeline_ipc(struct device *dev, +int sof_load_pipeline_ipc(struct snd_sof_dev *sdev, struct sof_ipc_pipe_new *pipeline, struct sof_ipc_comp_reply *r); int sof_pipeline_core_enable(struct snd_sof_dev *sdev, @@ -246,8 +246,8 @@ int snd_sof_ipc_set_get_comp_data(struct snd_sof_control *scontrol, int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params); /* PM */ -int sof_set_up_pipelines(struct device *dev, bool verify); -int sof_tear_down_pipelines(struct device *dev, bool verify); +int sof_set_up_pipelines(struct snd_sof_dev *sdev, bool verify); +int sof_tear_down_pipelines(struct snd_sof_dev *sdev, bool verify); int sof_set_hw_params_upon_resume(struct device *dev); bool snd_sof_stream_suspend_ignored(struct snd_sof_dev *sdev); bool snd_sof_dsp_only_d0i3_compatible_stream_active(struct snd_sof_dev *sdev); diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 534f004f6162..696c254f101c 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1690,11 +1690,10 @@ static int sof_widget_load_pcm(struct snd_soc_component *scomp, int index, /* * Pipeline Topology */ -int sof_load_pipeline_ipc(struct device *dev, +int sof_load_pipeline_ipc(struct snd_sof_dev *sdev, struct sof_ipc_pipe_new *pipeline, struct sof_ipc_comp_reply *r) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); int ret = sof_core_enable(sdev, pipeline->core); if (ret < 0) @@ -1703,7 +1702,7 @@ int sof_load_pipeline_ipc(struct device *dev, ret = sof_ipc_tx_message(sdev->ipc, pipeline->hdr.cmd, pipeline, sizeof(*pipeline), r, sizeof(*r)); if (ret < 0) - dev_err(dev, "error: load pipeline ipc failure\n"); + dev_err(sdev->dev, "error: load pipeline ipc failure\n"); return ret; } @@ -3373,15 +3372,14 @@ static int sof_route_load(struct snd_soc_component *scomp, int index, return ret; } -int snd_sof_complete_pipeline(struct device *dev, +int snd_sof_complete_pipeline(struct snd_sof_dev *sdev, struct snd_sof_widget *swidget) { - struct snd_sof_dev *sdev = dev_get_drvdata(dev); struct sof_ipc_pipe_ready ready; struct sof_ipc_reply reply; int ret; - dev_dbg(dev, "tplg: complete pipeline %s id %d\n", + dev_dbg(sdev->dev, "tplg: complete pipeline %s id %d\n", swidget->widget->name, swidget->comp_id); memset(&ready, 0, sizeof(ready)); @@ -3460,13 +3458,13 @@ static int sof_complete(struct snd_soc_component *scomp) /* verify topology components loading including dynamic pipelines */ if (sof_core_debug & SOF_DBG_VERIFY_TPLG) { - ret = sof_set_up_pipelines(scomp->dev, true); + ret = sof_set_up_pipelines(sdev, true); if (ret < 0) { dev_err(sdev->dev, "error: topology verification failed %d\n", ret); return ret; } - ret = sof_tear_down_pipelines(scomp->dev, true); + ret = sof_tear_down_pipelines(sdev, true); if (ret < 0) { dev_err(sdev->dev, "error: topology tear down pipelines failed %d\n", ret); return ret; @@ -3474,7 +3472,7 @@ static int sof_complete(struct snd_soc_component *scomp) } /* set up static pipelines */ - return sof_set_up_pipelines(scomp->dev, false); + return sof_set_up_pipelines(sdev, false); } /* manifest - optional to inform component of manifest */ -- 2.33.0

2 1

[PATCH v4 0/2] Add reset-gpios handling for max98927
by Alejandro Tafalla 08 Oct '21

08 Oct '21

The max98927 codec on some devices (i.e. Xiaomi Mi A2 Lite phone) requires hardware-resetting the codec by driving a reset-gpio. This series adds support for it through an optional reset-gpios property. v4: * Correctly assert/deassert the GPIO states * Wait for the i2c port to be ready after reset * Reset device when removed v3: * Fix indentation on the dev_err_probe line v2: * Use dev_err_probe instead of dev_err Alejandro Tafalla (2): ASoC: max98927: Handle reset gpio when probing i2c dt-bindings: sound: max98927: Add reset-gpios optional property .../devicetree/bindings/sound/max9892x.txt | 3 +++ sound/soc/codecs/max98927.c | 25 +++++++++++++++++++ sound/soc/codecs/max98927.h | 1 + 3 files changed, 29 insertions(+) -- 2.33.0

3 7

[PATCH] ASoC: SOF: topology: do not power down primary core during topology removal
by Peter Ujfalusi 08 Oct '21

08 Oct '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> When removing the topology components, do not power down the primary core. Doing so will result in an IPC timeout when the SOF PCI device runtime suspends. Fixes: 0dcdf84289fb ("ASoC: SOF: add a "core" parameter to widget loading functions") Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/topology.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 534f004f6162..1723dd11acd7 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -2514,6 +2514,15 @@ static int sof_widget_unload(struct snd_soc_component *scomp, /* power down the pipeline schedule core */ pipeline = swidget->private; + + /* + * Runtime PM should still function normally if topology loading fails and + * it's components are unloaded. Do not power down the primary core so that the + * CTX_SAVE IPC can succeed during runtime suspend. + */ + if (pipeline->core == SOF_DSP_PRIMARY_CORE) + break; + ret = snd_sof_dsp_core_power_down(sdev, 1 << pipeline->core); if (ret < 0) dev_err(scomp->dev, "error: powering down pipeline schedule core %d\n", -- 2.33.0

2 1

[PATCH] ASoC: DAPM: Fix missing kctl change notifications
by Takashi Iwai 08 Oct '21

08 Oct '21

The put callback of a kcontrol is supposed to return 1 when the value is changed, and this will be notified to user-space. However, some DAPM kcontrols always return 0 (except for errors), hence the user-space misses the update of a control value. This patch corrects the behavior by properly returning 1 when the value gets updated. Reported-and-tested-by: Hans de Goede <hdegoede(a)redhat.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/soc-dapm.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7b67f1e19ae9..59d07648a7e7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2561,6 +2561,7 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, const char *pin, int status) { struct snd_soc_dapm_widget *w = dapm_find_widget(dapm, pin, true); + int ret = 0; dapm_assert_locked(dapm); @@ -2573,13 +2574,14 @@ static int snd_soc_dapm_set_pin(struct snd_soc_dapm_context *dapm, dapm_mark_dirty(w, "pin configuration"); dapm_widget_invalidate_input_paths(w); dapm_widget_invalidate_output_paths(w); + ret = 1; } w->connected = status; if (status == 0) w->force = 0; - return 0; + return ret; } /** @@ -3583,14 +3585,15 @@ int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol, { struct snd_soc_card *card = snd_kcontrol_chip(kcontrol); const char *pin = (const char *)kcontrol->private_value; + int ret; if (ucontrol->value.integer.value[0]) - snd_soc_dapm_enable_pin(&card->dapm, pin); + ret = snd_soc_dapm_enable_pin(&card->dapm, pin); else - snd_soc_dapm_disable_pin(&card->dapm, pin); + ret = snd_soc_dapm_disable_pin(&card->dapm, pin); snd_soc_dapm_sync(&card->dapm); - return 0; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch); @@ -4023,7 +4026,7 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, rtd->params_select = ucontrol->value.enumerated.item[0]; - return 0; + return 1; } static void -- 2.26.2

2 1

[PATCH v1 1/4] ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata()
by Andy Shevchenko 08 Oct '21

08 Oct '21

Access to platform data via dev_get_platdata() getter to make code cleaner. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/boards/bytcht_es8316.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 1bb9b8e7bcc7..4360519fbb0c 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -456,12 +456,12 @@ static const struct dmi_system_id byt_cht_es8316_quirk_table[] = { static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; static const char * const mic_name[] = { "in1", "in2" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct property_entry props[MAX_NO_PROPS] = {}; struct byt_cht_es8316_private *priv; const struct dmi_system_id *dmi_id; - struct device *dev = &pdev->dev; - struct snd_soc_acpi_mach *mach; struct fwnode_handle *fwnode; const char *platform_name; struct acpi_device *adev; @@ -476,7 +476,6 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) if (!priv) return -ENOMEM; - mach = dev->platform_data; /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { if (!strcmp(byt_cht_es8316_dais[i].codecs->name, -- 2.33.0

4 9

[PATCH] ASoC: soc-pcm: restore mixer functionality
by Pierre-Louis Bossart 08 Oct '21

08 Oct '21

Recent changes in soc-pcm completely broke basic support for mixers on Intel systems: the filters on BE states prevent the connection of a second mixer input while the back-end is already active. Rather than reverting the changes, which would be problematic for Tegra systems, this patch suggests an additional filter which will only apply to Tegra systems. This is a temporary solution which will have to be revisited - additional issues have been reported with DPCM. Fixes: 0c25db3f7621 ('ASoC: soc-pcm: Don't reconnect an already active BE') Suggested-by: Sameer Pujar <spujar(a)nvidia.com> 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: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/soc-pcm.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index b5825d8d6caa..0aa0ae9703d1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1395,7 +1395,13 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; - if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && + /* + * Filter for systems with 'component_chaining' enabled. + * This helps to avoid unnecessary re-configuration of an + * already active BE on such systems. + */ + if (fe->card->component_chaining && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_NEW) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_CLOSE)) continue; -- 2.25.1

2 1

[PATCH] ASoC: codec: wcd938x: Add irq config support
by Srinivasa Rao Mandadapu 08 Oct '21

08 Oct '21

This patch fixes compilation error in wcd98x codec driver. Fixes: 045442228868 ("ASoC: codecs: wcd938x: add audio routing and Kconfig") Signed-off-by: Venkata Prasad Potturu <potturu(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 82ee233..216cea0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1583,6 +1583,7 @@ config SND_SOC_WCD938X_SDW tristate "WCD9380/WCD9385 Codec - SDW" select SND_SOC_WCD938X select SND_SOC_WCD_MBHC + select REGMAP_IRQ depends on SOUNDWIRE select REGMAP_SOUNDWIRE help -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 2

[PATCH v2 0/3] Multiple headphone codec driver support
by Brent Lu 07 Oct '21

07 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". V2 Changes: - remove useless 'NULL', 'false' in if-condition - can use 'comp_ids' field alone to enumerate driver - add comma to the end of entry in structure initialization - keep the table of byt/cht/cml/icl untouched Brent Lu (3): ASoC: soc-acpi: add comp_ids field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use comp_ids to enumerate rt5682s include/sound/soc-acpi.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 34 +++--------------- .../intel/common/soc-acpi-intel-adl-match.c | 11 ++++-- .../intel/common/soc-acpi-intel-jsl-match.c | 35 +++++-------------- .../intel/common/soc-acpi-intel-tgl-match.c | 11 ++++-- sound/soc/soc-acpi.c | 24 +++++++++++-- 6 files changed, 52 insertions(+), 65 deletions(-) -- 2.25.1

3 6

[PATCH v2 00/13] Extend AHUB audio support for Tegra210 and later
by Sameer Pujar 07 Oct '21

07 Oct '21

Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Changelog ========= v1 -> v2 -------- * Put comments for soft reset application in the drivers. * Split out mute/volume control logic in put() calls of MVC driver and use separate callbacks for the respective kcontrols. * Update kcontrol put() callback in MVC driver to return 1 whenever there is change. Similar change is done in other drivers too. * Use name-prefix.yaml reference for the driver documentation now. * Add sound-name-prefix pattern for MIXER driver and use prefix accordingly in DT. Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 76 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 76 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 74 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 76 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 73 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 +++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 531 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 600 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 674 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 645 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3549 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13647 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4

4 18

[PATCH 0/3] Multiple headphone codec driver support
by Brent Lu 07 Oct '21

07 Oct '21

Support multiple headphone drivers in same machine driver. In this case, both rt5682 and rt5682s are supported and enumerated by different ACPI HID "10EC5682" and "RTL5682". Brent Lu (3): ASoC: soc-acpi: add alternative id field for machine driver matching ASoC: Intel: sof_rt5682: detect codec variant in probe function ASoC: Intel: sof_rt5682: use id_alt to enumerate rt5682s include/sound/soc-acpi.h | 2 ++ sound/soc/intel/boards/sof_rt5682.c | 34 +++---------------- .../intel/common/soc-acpi-intel-adl-match.c | 8 +++++ .../intel/common/soc-acpi-intel-byt-match.c | 6 ++++ .../intel/common/soc-acpi-intel-cht-match.c | 6 ++++ .../intel/common/soc-acpi-intel-cml-match.c | 8 +++++ .../intel/common/soc-acpi-intel-icl-match.c | 6 ++++ .../intel/common/soc-acpi-intel-jsl-match.c | 32 +++++------------ .../intel/common/soc-acpi-intel-tgl-match.c | 8 +++++ sound/soc/soc-acpi.c | 21 +++++++++++- 10 files changed, 76 insertions(+), 55 deletions(-) -- 2.25.1

7 16

DAPM PIN switches do not update in alsamixer when changed through UCM profile
by Hans de Goede 07 Oct '21

07 Oct '21

Hi All, I notice that DAPM PIN switches, such as e.g. the "Headphone" SOC_DAPM_PIN_SWITCH defined in: sound/soc/intel/boards/cht_bsw_nau8824.c: static const struct snd_kcontrol_new cht_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Headphone"), SOC_DAPM_PIN_SWITCH("Headset Mic"), SOC_DAPM_PIN_SWITCH("Int Mic"), SOC_DAPM_PIN_SWITCH("Ext Spk"), }; Do not get updated to reflect state-changes when the output is switched between e.g. Headphone / "Ext Spk" by pulseaudio/pipewire through the UCM profile mechanism. If I exit alsa-mixer after changing the output and start it again then the control does show the expect value. So it seems that we are failing to send a change event about this somewhere? Regards, Hans

3 5

dlmisc.c: Avoid double unlock in snd_dlobj_cache_get0()
by GitHub pull_request - opened 07 Oct '21

07 Oct '21

alsa-project/alsa-lib pull request #182 was opened from johado: Remove call to snd_dlobj_unlock() in snd_dlobj_cache_get0() All lock/unlock is done by callers of the function. Fixes: https://github.com/alsa-project/alsa-lib/issues/181 Request URL : https://github.com/alsa-project/alsa-lib/pull/182 Patch URL : https://github.com/alsa-project/alsa-lib/pull/182.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Double snd_dlobj_unlock()
by GitHub issues - edited 07 Oct '21

07 Oct '21

alsa-project/alsa-lib issue #181 was edited from johado: In dlmisc.c:snd_dlobj_cache_get0() there is a call to snd_dlobj_unlock() in the __err: path which should not be there. All lock/unlock are done by the callers to this function. ``` --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -359,7 +359,6 @@ snd_dlobj_cache_get0(const char *lib, const char *name, free(c); __err: snd_dlclose(dlobj); - snd_dlobj_unlock(); return NULL; } c->dlobj = dlobj; ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/181 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Double snd_dlobj_unlock()
by GitHub issues - opened 07 Oct '21

07 Oct '21

alsa-project/alsa-lib issue #181 was opened from johado: In dlmisc.c:snd_dlobj_cache_get0() there is a call to snd_dlobj_unlock() in the __err: path which should not be there. All lock/unlock are done by the callers to this function. --- a/src/dlmisc.c +++ b/src/dlmisc.c @@ -359,7 +359,6 @@ snd_dlobj_cache_get0(const char *lib, const char *name, free(c); __err: snd_dlclose(dlobj); - snd_dlobj_unlock(); return NULL; } c->dlobj = dlobj; Issue URL : https://github.com/alsa-project/alsa-lib/issues/181 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: usb-audio: Pass JOINT_DUPLEX info flag for implicit fb streams
by Takashi Iwai 07 Oct '21

07 Oct '21

When a stream is in the implicit feedback mode, it's more or less tied with a capture stream. Passing SNDRV_PCM_INFO_JOINT_DUPLEX may help for user-space to understand the situation. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index f09c7380a923..3bb095a3c9b6 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1080,6 +1080,13 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre if (err < 0) return err; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (fp->implicit_fb) { + runtime->hw.info |= SNDRV_PCM_INFO_JOINT_DUPLEX; + break; + } + } + return 0; } -- 2.31.1

1 0

ALC236 recording issue
by Kailang 07 Oct '21

07 Oct '21

Hi Takashi, Attach patch will solve alc236 recording issue for HP platform. Many Thanks. BR, Kailang

2 4

[PATCH v3 0/4] ASoC: Intel: bytcht_es8316: few cleanups
by Andy Shevchenko 06 Oct '21

06 Oct '21

The small set of cleanups against bytcht_es8316 board file. In v3: - actually added a Pierre's tag (Mark) In v2: - added tag (Pierre) - added commit message to the patch 2 (Joe) Andy Shevchenko (4): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation sound/soc/intel/boards/bytcht_es8316.c | 37 +++++++++----------------- 1 file changed, 12 insertions(+), 25 deletions(-) -- 2.33.0

1 4

[PATCH v1 1/4] ASoC: Intel: bytcr_rt5651: Get platform data via dev_get_platdata()
by Andy Shevchenko 06 Oct '21

06 Oct '21

Access to platform data via dev_get_platdata() getter to make code cleaner. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 00c347b52863..2335cca17cc8 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -894,9 +894,10 @@ struct acpi_chan_package { /* ACPICA seems to require 64 bit integers */ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; static const char * const mic_name[] = { "dmic", "in1", "in2", "in12" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); struct byt_rt5651_private *priv; - struct snd_soc_acpi_mach *mach; const char *platform_name; struct acpi_device *adev; struct device *codec_dev; @@ -912,8 +913,6 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* register the soc card */ byt_rt5651_card.dev = &pdev->dev; - - mach = byt_rt5651_card.dev->platform_data; snd_soc_card_set_drvdata(&byt_rt5651_card, priv); /* fix index of codec dai */ -- 2.33.0

5 11

[PATCH v2 0/4] ASoC: Intel: bytcht_es8316: few cleanups
by Andy Shevchenko 06 Oct '21

06 Oct '21

The small set of cleanups against bytcht_es8316 board file. In v2: - added tag (Pierre) - added commit message to the patch 2 (Joe) Andy Shevchenko (4): ASoC: Intel: bytcht_es8316: Get platform data via dev_get_platdata() ASoC: Intel: bytcht_es8316: Use temporary variable for struct device ASoC: Intel: bytcht_es8316: Switch to use gpiod_get_optional() ASoC: Intel: bytcht_es8316: Utilize dev_err_probe() to avoid log saturation sound/soc/intel/boards/bytcht_es8316.c | 37 +++++++++----------------- 1 file changed, 12 insertions(+), 25 deletions(-) -- 2.33.0

2 5

[PATCH v1 1/4] ASoC: Intel: bytcr_rt5640: Get platform data via dev_get_platdata()
by Andy Shevchenko 06 Oct '21

06 Oct '21

Access to platform data via dev_get_platdata() getter to make code cleaner. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index c28abe74816f..43997048a825 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1495,12 +1495,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; static const char * const map_name[] = { "dmic1", "dmic2", "in1", "in3", "none" }; + struct snd_soc_acpi_mach *mach = dev_get_platdata(dev); __maybe_unused const char *spk_type; const struct dmi_system_id *dmi_id; const char *headset2_string = ""; const char *lineout_string = ""; struct byt_rt5640_private *priv; - struct snd_soc_acpi_mach *mach; const char *platform_name; struct acpi_device *adev; struct device *codec_dev; @@ -1517,7 +1517,6 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* register the soc card */ byt_rt5640_card.dev = &pdev->dev; - mach = byt_rt5640_card.dev->platform_data; snd_soc_card_set_drvdata(&byt_rt5640_card, priv); /* fix index of codec dai */ -- 2.33.0

4 11

[PATCH] ASoC: soc-acpi: add alternative id field for machine driver matching
by Brent Lu 06 Oct '21

06 Oct '21

Current design to support second headphone driver in the same machine driver is to duplicate the entries in snd_soc_acpi_mach array and board configs in machine driver. We can avoid this by adding an id_alt field in snd_soc_acpi_mach structure to specify alternative ACPI HIDs for machine driver enumeration and leave the codec type detection to machine driver if necessary. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- include/sound/soc-acpi.h | 2 ++ sound/soc/soc-acpi.c | 21 ++++++++++++++++++++- 2 files changed, 22 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index 2f3fa385c092..fcf6bae9f9d7 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -129,6 +129,7 @@ struct snd_soc_acpi_link_adr { * all firmware/topology related fields. * * @id: ACPI ID (usually the codec's) used to find a matching machine driver. + * @id_alt: array of ACPI IDs used as an alternative of id field. * @link_mask: describes required board layout, e.g. for SoundWire. * @links: array of link _ADR descriptors, null terminated. * @drv_name: machine driver name @@ -146,6 +147,7 @@ struct snd_soc_acpi_link_adr { /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { const u8 id[ACPI_ID_LEN]; + struct snd_soc_acpi_codecs *id_alt; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; const char *drv_name; diff --git a/sound/soc/soc-acpi.c b/sound/soc/soc-acpi.c index 395229bf5c51..ab67d640c20f 100644 --- a/sound/soc/soc-acpi.c +++ b/sound/soc/soc-acpi.c @@ -8,6 +8,25 @@ #include <linux/module.h> #include <sound/soc-acpi.h> +static bool snd_soc_acpi_id_present(struct snd_soc_acpi_mach *machine) +{ + struct snd_soc_acpi_codecs *id_alt = machine->id_alt; + int i; + + if (acpi_dev_present(machine->id, NULL, -1)) + return true; + + if (id_alt == NULL) + return false; + + for (i = 0; i < id_alt->num_codecs; i++) { + if (acpi_dev_present(id_alt->codecs[i], NULL, -1)) + return true; + } + + return false; +} + struct snd_soc_acpi_mach * snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) { @@ -15,7 +34,7 @@ snd_soc_acpi_find_machine(struct snd_soc_acpi_mach *machines) struct snd_soc_acpi_mach *mach_alt; for (mach = machines; mach->id[0]; mach++) { - if (acpi_dev_present(mach->id, NULL, -1)) { + if (snd_soc_acpi_id_present(mach) != false) { if (mach->machine_quirk) { mach_alt = mach->machine_quirk(mach); if (!mach_alt) -- 2.25.1

3 2

[PATCH] ALSA: hda/realtek: Add quirk for TongFang PHxTxX1
by Werner Sembach 06 Oct '21

06 Oct '21

This applies a SND_PCI_QUIRK(...) to the TongFang PHxTxX1 barebone. This fixes the issue of the internal Microphone not working after booting another OS. When booting a certain another OS this barebone keeps some coeff settings even after a cold shutdown. These coeffs prevent the microphone detection from working in Linux, making the Laptop think that there is always an external microphone plugged-in and therefore preventing the use of the internal one. The relevant indexes and values where gathered by naively diff-ing and reading a working and a non-working coeff dump. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> --- sound/pci/hda/patch_realtek.c | 26 +++++++++++++++++++++++++- 1 file changed, 25 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..55d3c9d85fd2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6449,6 +6449,24 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" +static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + /* + * A certain other OS sets these coeffs to different values. On at least one TongFang + * barebone these settings might survive even a cold reboot. So to restore a clean slate the + * values are explicitly reset to default here. Without this, the external microphone is + * always in a plugged-in state, while the internal microphone is always in an unplugged + * state, breaking the ability to use the internal microphone. + */ + alc_write_coef_idx(codec, 0x24, 0x0000); + alc_write_coef_idx(codec, 0x26, 0x0000); + alc_write_coef_idx(codec, 0x29, 0x3000); + alc_write_coef_idx(codec, 0x37, 0xfe05); + alc_write_coef_idx(codec, 0x45, 0x5089); +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6663,7 +6681,8 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS, ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, - ALC287_FIXUP_13S_GEN2_SPEAKERS + ALC287_FIXUP_13S_GEN2_SPEAKERS, + ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, }; static const struct hda_fixup alc269_fixups[] = { @@ -8361,6 +8380,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, + [ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_tongfang_reset_persistent_settings, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8789,6 +8812,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), + SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.25.1

2 3

[PATCH] ALSA: pcm: Add more disconnection checks at file ops
by Takashi Iwai 06 Oct '21

06 Oct '21

In the case of hot-disconnection of a PCM device, all file operations except for close should be rejected. This patch adds more sanity checks in the file operation code paths. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index ee6d17caf2ce..0c0a825af773 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3234,6 +3234,9 @@ static int snd_pcm_common_ioctl(struct file *file, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; + if (substream->runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + return -EBADFD; + res = snd_power_wait(substream->pcm->card); if (res < 0) return res; @@ -3360,6 +3363,9 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, snd_pcm_uframes_t *frames = arg; snd_pcm_sframes_t result; + if (substream->runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + return -EBADFD; + switch (cmd) { case SNDRV_PCM_IOCTL_FORWARD: { @@ -3402,7 +3408,8 @@ static ssize_t snd_pcm_read(struct file *file, char __user *buf, size_t count, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->status->state == SNDRV_PCM_STATE_OPEN || + runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!frame_aligned(runtime, count)) return -EINVAL; @@ -3426,7 +3433,8 @@ static ssize_t snd_pcm_write(struct file *file, const char __user *buf, if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->status->state == SNDRV_PCM_STATE_OPEN || + runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!frame_aligned(runtime, count)) return -EINVAL; @@ -3452,7 +3460,8 @@ static ssize_t snd_pcm_readv(struct kiocb *iocb, struct iov_iter *to) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->status->state == SNDRV_PCM_STATE_OPEN || + runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!iter_is_iovec(to)) return -EINVAL; @@ -3488,7 +3497,8 @@ static ssize_t snd_pcm_writev(struct kiocb *iocb, struct iov_iter *from) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; - if (runtime->status->state == SNDRV_PCM_STATE_OPEN) + if (runtime->status->state == SNDRV_PCM_STATE_OPEN || + runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; if (!iter_is_iovec(from)) return -EINVAL; @@ -3527,6 +3537,9 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) return ok | EPOLLERR; runtime = substream->runtime; + if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + return ok | EPOLLERR; + poll_wait(file, &runtime->sleep, wait); mask = 0; @@ -3836,6 +3849,8 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; + if (substream->runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + return -EBADFD; offset = area->vm_pgoff << PAGE_SHIFT; switch (offset) { @@ -3872,6 +3887,8 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; + if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) + return -EBADFD; return fasync_helper(fd, file, on, &runtime->fasync); } -- 2.31.1

1 0

[PATCH] ALSA: hda: intel: Allow repeatedly probing on codec configuration errors
by Takashi Iwai 06 Oct '21

06 Oct '21

It seems that a few recent AMD systems show the codec configuration errors at the early boot, while loading the driver at a later stage works magically. Although the root cause of the error isn't clear, it's certainly not bad to allow retrying the codec probe in such a case if that helps. This patch adds the capability for retrying the probe upon codec probe errors on the certain AMD platforms. The probe_work is changed to a delayed work, and at the secondary call, it'll jump to the codec probing. Note that, not only adding the re-probing, this includes the behavior changes in the codec configuration function. Namely, snd_hda_codec_configure() won't unregister the codec at errors any longer. Instead, its caller, azx_codec_configure() unregisters the codecs with the probe failures *if* any codec has been successfully configured. If all codec probe failed, it doesn't unregister but let it re-probed -- which is the most case we're seeing and this patch tries to improve. Even if the driver doesn't re-probe or give up, it will go to the "free-all" error path, hence the leftover codecs shall be disabled / deleted in anyway. BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190801 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_codec.h | 1 + sound/pci/hda/hda_bind.c | 20 +++++++++++--------- sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_controller.c | 24 ++++++++++++++++-------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 29 +++++++++++++++++++++++------ sound/pci/hda/hda_intel.h | 4 +++- 7 files changed, 56 insertions(+), 25 deletions(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 01570dbda503..0e45963bb767 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -224,6 +224,7 @@ struct hda_codec { #endif /* misc flags */ + unsigned int configured:1; /* codec was configured */ unsigned int in_freeing:1; /* being released */ unsigned int registered:1; /* codec was registered */ unsigned int display_power_control:1; /* needs display power */ diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c index 2523b23389e9..1c8bffc3eec6 100644 --- a/sound/pci/hda/hda_bind.c +++ b/sound/pci/hda/hda_bind.c @@ -298,29 +298,31 @@ int snd_hda_codec_configure(struct hda_codec *codec) { int err; + if (codec->configured) + return 0; + if (is_generic_config(codec)) codec->probe_id = HDA_CODEC_ID_GENERIC; else codec->probe_id = 0; - err = snd_hdac_device_register(&codec->core); - if (err < 0) - return err; + if (!device_is_registered(&codec->core.dev)) { + err = snd_hdac_device_register(&codec->core); + if (err < 0) + return err; + } if (!codec->preset) codec_bind_module(codec); if (!codec->preset) { err = codec_bind_generic(codec); if (err < 0) { - codec_err(codec, "Unable to bind the codec\n"); - goto error; + codec_dbg(codec, "Unable to bind the codec\n"); + return err; } } + codec->configured = 1; return 0; - - error: - snd_hdac_device_unregister(&codec->core); - return err; } EXPORT_SYMBOL_GPL(snd_hda_codec_configure); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index a9ebefd60cf6..0c4a337c9fc0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -791,6 +791,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec) snd_array_free(&codec->nids); remove_conn_list(codec); snd_hdac_regmap_exit(&codec->core); + codec->configured = 0; } EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind); diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 7cd452831fd3..930ae4002a81 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include <sound/initval.h> #include "hda_controller.h" +#include "hda_local.h" #define CREATE_TRACE_POINTS #include "hda_controller_trace.h" @@ -1248,17 +1249,24 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs); int azx_codec_configure(struct azx *chip) { struct hda_codec *codec, *next; + int success = 0; - /* use _safe version here since snd_hda_codec_configure() deregisters - * the device upon error and deletes itself from the bus list. - */ - list_for_each_codec_safe(codec, next, &chip->bus) { - snd_hda_codec_configure(codec); + list_for_each_codec(codec, &chip->bus) { + if (!snd_hda_codec_configure(codec)) + success++; } - if (!azx_bus(chip)->num_codecs) - return -ENODEV; - return 0; + if (success) { + /* unregister failed codecs if any codec has been probed */ + list_for_each_codec_safe(codec, next, &chip->bus) { + if (!codec->configured) { + codec_err(codec, "Unable to configure, disabling\n"); + snd_hdac_device_unregister(&codec->core); + } + } + } + + return success ? 0 : -ENODEV; } EXPORT_SYMBOL_GPL(azx_codec_configure); diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 3062f87380b1..f5bf295eb830 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -41,7 +41,7 @@ /* 24 unused */ #define AZX_DCAPS_COUNT_LPIB_DELAY (1 << 25) /* Take LPIB as delay */ #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ -/* 27 unused */ +#define AZX_DCAPS_RETRY_PROBE (1 << 27) /* retry probe if no codec is configured */ #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28) /* CORBRP clears itself after reset */ #define AZX_DCAPS_NO_MSI64 (1 << 29) /* Stick to 32-bit MSIs */ #define AZX_DCAPS_SEPARATE_STREAM_TAG (1 << 30) /* capture and playback use separate stream tag */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 47777439961c..4d22e7adeee8 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -307,7 +307,8 @@ enum { /* quirks for AMD SB */ #define AZX_DCAPS_PRESET_AMD_SB \ (AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\ - AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME) + AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME |\ + AZX_DCAPS_RETRY_PROBE) /* quirks for Nvidia */ #define AZX_DCAPS_PRESET_NVIDIA \ @@ -1723,7 +1724,7 @@ static void azx_check_snoop_available(struct azx *chip) static void azx_probe_work(struct work_struct *work) { - struct hda_intel *hda = container_of(work, struct hda_intel, probe_work); + struct hda_intel *hda = container_of(work, struct hda_intel, probe_work.work); azx_probe_continue(&hda->chip); } @@ -1828,7 +1829,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, } /* continue probing in work context as may trigger request module */ - INIT_WORK(&hda->probe_work, azx_probe_work); + INIT_DELAYED_WORK(&hda->probe_work, azx_probe_work); *rchip = chip; @@ -2142,7 +2143,7 @@ static int azx_probe(struct pci_dev *pci, #endif if (schedule_probe) - schedule_work(&hda->probe_work); + schedule_delayed_work(&hda->probe_work, 0); dev++; if (chip->disabled) @@ -2228,6 +2229,11 @@ static int azx_probe_continue(struct azx *chip) int dev = chip->dev_index; int err; + if (chip->disabled || hda->init_failed) + return -EIO; + if (hda->probe_retry) + goto probe_retry; + to_hda_bus(bus)->bus_probing = 1; hda->probe_continued = 1; @@ -2289,10 +2295,20 @@ static int azx_probe_continue(struct azx *chip) #endif } #endif + + probe_retry: if (bus->codec_mask && !(probe_only[dev] & 1)) { err = azx_codec_configure(chip); - if (err < 0) + if (err) { + if ((chip->driver_caps & AZX_DCAPS_RETRY_PROBE) && + ++hda->probe_retry < 60) { + schedule_delayed_work(&hda->probe_work, + msecs_to_jiffies(1000)); + return 0; /* keep things up */ + } + dev_err(chip->card->dev, "Cannot probe codecs, giving up\n"); goto out_free; + } } err = snd_card_register(chip->card); @@ -2322,6 +2338,7 @@ static int azx_probe_continue(struct azx *chip) display_power(chip, false); complete_all(&hda->probe_wait); to_hda_bus(bus)->bus_probing = 0; + hda->probe_retry = 0; return 0; } @@ -2347,7 +2364,7 @@ static void azx_remove(struct pci_dev *pci) * device during cancel_work_sync() call. */ device_unlock(&pci->dev); - cancel_work_sync(&hda->probe_work); + cancel_delayed_work_sync(&hda->probe_work); device_lock(&pci->dev); snd_card_free(card); diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h index 3fb119f09040..0f39418f9328 100644 --- a/sound/pci/hda/hda_intel.h +++ b/sound/pci/hda/hda_intel.h @@ -14,7 +14,7 @@ struct hda_intel { /* sync probing */ struct completion probe_wait; - struct work_struct probe_work; + struct delayed_work probe_work; /* card list (for power_save trigger) */ struct list_head list; @@ -30,6 +30,8 @@ struct hda_intel { unsigned int freed:1; /* resources already released */ bool need_i915_power:1; /* the hda controller needs i915 power */ + + int probe_retry; /* being probe-retry */ }; #endif -- 2.31.1

1 0

[PATCH MANUALSEL 5.14 9/9] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index ed40d0f7432c..773db4bf0876 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

[PATCH MANUALSEL 4.4] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 3689f5f6be64..3766624c018f 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -144,7 +144,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

[PATCH MANUALSEL 4.9] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 3689f5f6be64..3766624c018f 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -144,7 +144,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

[PATCH MANUALSEL 4.14] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 8f0f05bbc081..23a15d892e50 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -145,7 +145,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

[PATCH MANUALSEL 4.19 2/2] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 8f0f05bbc081..23a15d892e50 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -145,7 +145,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

[PATCH MANUALSEL 5.4 4/4] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index 8f0f05bbc081..23a15d892e50 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -145,7 +145,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

[PATCH MANUALSEL 5.10 7/7] ALSA: pcsp: Make hrtimer forwarding more robust
by Sasha Levin 06 Oct '21

06 Oct '21

From: Thomas Gleixner <tglx(a)linutronix.de> [ Upstream commit f2ff7147c6834f244b8ce636b12e71a3bd044629 ] The hrtimer callback pcsp_do_timer() prepares rearming of the timer with hrtimer_forward(). hrtimer_forward() is intended to provide a mechanism to forward the expiry time of the hrtimer by a multiple of the period argument so that the expiry time greater than the time provided in the 'now' argument. pcsp_do_timer() invokes hrtimer_forward() with the current timer expiry time as 'now' argument. That's providing a periodic timer expiry, but is not really robust when the timer callback is delayed so that the resulting new expiry time is already in the past which causes the callback to be invoked immediately again. If the timer is delayed then the back to back invocation is not really making it better than skipping the missed periods. Sound is distorted in any case. Use hrtimer_forward_now() which ensures that the next expiry is in the future. This prevents hogging the CPU in the timer expiry code and allows later on to remove hrtimer_forward() from the public interfaces. Signed-off-by: Thomas Gleixner <tglx(a)linutronix.de> Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Link: https://lore.kernel.org/r/20210923153339.623208460@linutronix.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/drivers/pcsp/pcsp_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index ed40d0f7432c..773db4bf0876 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle) if (pointer_update) pcsp_pointer_update(chip); - hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns)); + hrtimer_forward_now(handle, ns_to_ktime(ns)); return HRTIMER_RESTART; } -- 2.33.0

1 0

Re: [PATCH 2/2] ASoC: rt9120: Add rt9210 audio amplifier support
by ChiYuan Huang 06 Oct '21

06 Oct '21

Hi, Mark: Mark Brown <broonie(a)kernel.org> 於 2021年10月5日週二下午8:44寫道： > > On Tue, Oct 05, 2021 at 08:36:44PM +0800, ChiYuan Huang wrote: > > Mark Brown <broonie(a)kernel.org> 於 2021年10月5日週二下午7:54寫道： > > > On Tue, Oct 05, 2021 at 09:16:55AM +0800, cy_huang wrote: > > > > > + /* Default config volume to 0dB */ > > > > + snd_soc_component_write(comp, RT9120_REG_MSVOL, 0x180); > > > > + /* Mute by default */ > > > > + snd_soc_component_update_bits(comp, RT9120_REG_VOLRAMP, > > > > + RT9120_MUTE_MASK, RT9120_MUTE_MASK); > > > > As ever you should leave the defaults at whatever the hardware defaults > > > to, the defaults for one machine may not be suitable for another so we > > > shouldn't be trying to do that in software. > > > The default volume will be kept in value 0x7ff (mute). > > I just want to follow the ASoC flow to control mute/unmute mask by AMP on/off. > > If to default set volume to 0dB and mute is improper, user have to use > > mixer control to configure the volume. > > Does mute function also need to be removed also? > > It's totally fine and indeed quite common for devices to be muted by > default - usually systems will have UCM profiles that unmute things by > the time users actually interact with them. > > Please don't take things off-list unless there is a really strong reason > to do so. Sending things to the list ensures that everyone gets a > chance to read and comment on things. After asking the HW member, there's already builtin HW volume ramp function. Mute API is still no need. There's already no pop issue without mute function. So the next change, I'll remove the default volume and mute config, and also mute API. Thx.

1 0

[RFC PATCH v1 0/1] ASoC: meson: aiu: HDMI codec control questions and issues
by Martin Blumenstingl 06 Oct '21

06 Oct '21

Hello Jerome et al., on Amlogic Meson SoCs which use the AIU audio controller we have the so-called "machine gun noise" (MGN) issue, reproducible for example in Kodi. So far nobody was able to identify the cause of this issue. My idea was to at least narrow down the issue so we can categorize it. For that I wanted to get the SPDIF output from AIU to the HDMI controller working. On Amlogic Meson GXBB/GXL/GXM SoCs a DesignWare HDMI TX controller is used. This has an SPDIF input but there's currently not driver for it. On Meson8/8b/8m2 SoCs I am working on a HDMI driver for the TransSwitch HDMI controller which - just like DesignWare HDMI TX - supports SPDIF and I2S inputs. I decided to add SPDIF support to the latter since the code from the vendor driver is much easier. It took me a while to figure out why I would not get any audio output from AIU SPDIF to the HDMI controller - or from there to the sink. The "fix" for this issue is the RFC patch which is part of this series. Any feedback would be great as I am still new to the ASoC subsystem. Another part I am still struggling with is the audio "routing" (due to lack of a better term - please correct me if this is not the right word to use for this case). I have the following description in my .dts: sound { compatible = "amlogic,gx-sound-card"; model = "M8B-ODROID-C1"; assigned-clocks = <&clkc CLKID_MPLL0>, <&clkc CLKID_MPLL1>; assigned-clock-rates = <294912000>, <270950400>; dai-link-0 { sound-dai = <&aiu AIU_CPU CPU_I2S_FIFO>; }; dai-link-1 { sound-dai = <&aiu AIU_CPU CPU_SPDIF_FIFO>; }; dai-link-2 { sound-dai = <&aiu AIU_CPU CPU_I2S_ENCODER>; dai-format = "i2s"; mclk-fs = <256>; codec-0 { sound-dai = <&aiu AIU_HDMI CTRL_I2S>; }; }; dai-link-3 { sound-dai = <&aiu AIU_CPU CPU_SPDIF_ENCODER>; codec-0 { sound-dai = <&aiu AIU_HDMI CTRL_PCM>; }; }; dai-link-4 { sound-dai = <&aiu AIU_HDMI CTRL_OUT>; codec-0 { sound-dai = <&hdmi_tx>; }; }; }; The driver for &hdmi_tx defines: struct hdmi_codec_pdata pdata = { .ops = &txc_48352_hdmi_codec_ops, .i2s = 1, .spdif = 1, .max_i2s_channels = 8, .data = priv, }; In hdmi_codec_ops.hw_params I always get fmt->fmt HDMI_I2S unless I remove all I2S references from the .dts snipped above (only then HDMI_SPDIF is detected). Based on the selection of the "HDMI Source" enum in aiu-codec-ctrl I was expecting the format to update as well. That unfortunately doesn't happen and I don't know how that can be achieved. Best regards, Martin Martin Blumenstingl (1): ASoC: meson: aiu: Fix HDMI codec control selection sound/soc/meson/aiu-codec-ctrl.c | 108 ++++++++++++++++++++++-------- sound/soc/meson/aiu-encoder-i2s.c | 6 -- 2 files changed, 80 insertions(+), 34 deletions(-) -- 2.33.0

5 13

[PATCH] alsaloop: fixed parsing value 'auto' of option 'sync'
by Pavel Hofman 05 Oct '21

05 Oct '21

The value 'auto' was not checked for which resulted in using SYNC_TYPE_NONE instead of SYNC_TYPE_AUTO. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 4192712..f5f2e37 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -528,6 +528,8 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, arg_sync = SYNC_TYPE_PLAYRATESHIFT; else if (optarg[0] == 'r') arg_sync = SYNC_TYPE_SAMPLERATE; + else if (optarg[0] == 'a') + arg_sync = SYNC_TYPE_AUTO; else arg_sync = atoi(optarg); if (arg_sync < 0 || arg_sync > SYNC_TYPE_LAST) -- 2.25.1

1 0

[PATCH] alsaloop: added option prateshift for ctl elem used in PLAYSHIFT sync
by Pavel Hofman 05 Oct '21

05 Oct '21

If snd-aloop device is on playback side, the required sync mode is PLAYSHIFT. That means Loopback ctl elem "PCM Rate Shift 100000" of the corresponding capture side of the Loopback pipe must be controlled (by a reciprocal). ASCII name of the playback rate shift ctl elem is specified with newly added option -x/--prateshift, e.g.: -P hw:Loopback,0 -S playshift \ -x iface=PCM,name='PCM Rate Shift 100000',device=1 Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/alsaloop.1 | 11 ++++-- alsaloop/alsaloop.c | 11 +++++- alsaloop/alsaloop.h | 1 + alsaloop/pcmjob.c | 86 ++++++++++++++++++++++++++++++++++----------- 4 files changed, 86 insertions(+), 23 deletions(-) diff --git a/alsaloop/alsaloop.1 b/alsaloop/alsaloop.1 index 6bacfd5..006494d 100644 --- a/alsaloop/alsaloop.1 +++ b/alsaloop/alsaloop.1 @@ -57,6 +57,11 @@ Use given CTL device for playback. Use given CTL device for capture. +.TP +\fI\-x <ctl_ascii_name>\fP | \fI\-\-prateshift=<ctl_ascii_name>\fP + +Specify ctl ascii name for playshift sync mode (see the Examples section). + .TP \fI\-l <latency>\fP | \fI\-\-latency=<frames>\fP @@ -195,10 +200,12 @@ Verbose xrun profiling. Set process wake timeout. .SH EXAMPLES - -.TP +.nf \fBalsaloop \-C hw:0,0 \-P hw:1,0 \-t 50000\fR +\fBalsaloop \-C hw:soundcard,0 \-P hw:Loopback,0 \-t 50000 \-S playshift \\ +\-x iface=PCM,name='PCM Rate Shift 100000',device=1\fR +.ne .SH BUGS None known. .SH AUTHOR diff --git a/alsaloop/alsaloop.c b/alsaloop/alsaloop.c index 06ffadf..4192712 100644 --- a/alsaloop/alsaloop.c +++ b/alsaloop/alsaloop.c @@ -175,6 +175,7 @@ void help(void) "-C,--cdevice capture device\n" "-X,--pctl playback ctl device\n" "-Y,--cctl capture ctl device\n" +"-x,--prateshift playback 'PCM Rate Shift 100000' ascii ctl name\n" "-l,--latency requested latency in frames\n" "-t,--tlatency requested latency in usec (1/1000000sec)\n" "-f,--format sample format\n" @@ -362,6 +363,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, {"cdevice", 1, NULL, 'C'}, {"pctl", 1, NULL, 'X'}, {"cctl", 1, NULL, 'Y'}, + {"prateshift", 1, NULL, 'x'}, {"latency", 1, NULL, 'l'}, {"tlatency", 1, NULL, 't'}, {"format", 1, NULL, 'f'}, @@ -391,6 +393,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, char *arg_cdevice = NULL; char *arg_pctl = NULL; char *arg_cctl = NULL; + char *arg_prateshift = NULL; unsigned int arg_latency_req = 0; unsigned int arg_latency_reqtime = 10000; snd_pcm_format_t arg_format = SND_PCM_FORMAT_S16_LE; @@ -420,7 +423,7 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, while (1) { int c; if ((c = getopt_long(argc, argv, - "hdg:P:C:X:Y:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", + "hdg:P:C:X:Y:x:l:t:F:f:c:r:s:benvA:S:a:m:T:O:w:UW:z", long_option, NULL)) < 0) break; switch (c) { @@ -446,6 +449,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, case 'Y': arg_cctl = strdup(optarg); break; + case 'x': + arg_prateshift = strdup(optarg); + break; case 'l': err = atoi(optarg); arg_latency_req = err >= 4 ? err : 4; @@ -627,6 +633,9 @@ static int parse_config(int argc, char *argv[], snd_output_t *output, logit(LOG_CRIT, "Unable to add ossmixer controls.\n"); exit(EXIT_FAILURE); } + if (arg_prateshift) + play->prateshift_name = arg_prateshift; + #ifdef USE_SAMPLERATE loop->src_enable = arg_samplerate > 0; if (loop->src_enable) diff --git a/alsaloop/alsaloop.h b/alsaloop/alsaloop.h index 1dbcefe..7a98ef3 100644 --- a/alsaloop/alsaloop.h +++ b/alsaloop/alsaloop.h @@ -127,6 +127,7 @@ struct loopback_handle { snd_ctl_elem_value_t *ctl_format; snd_ctl_elem_value_t *ctl_rate; snd_ctl_elem_value_t *ctl_channels; + char *prateshift_name; /* ascii name for the playback rate shift ctl elem */ }; struct loopback { diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 845ab82..4b56d03 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1101,7 +1101,8 @@ void update_pitch(struct loopback *loop) #endif } else if (loop->sync == SYNC_TYPE_PLAYRATESHIFT) { - set_rate_shift(loop->play, pitch); + // pitch is capture-based, playback side requires reciprocal + set_rate_shift(loop->play, 1 / pitch); #ifdef USE_SAMPLERATE if (loop->use_samplerate) { loop->src_data.src_ratio = @@ -1172,44 +1173,89 @@ static int get_channels(struct loopback_handle *lhandle) return snd_ctl_elem_value_get_integer(lhandle->ctl_channels, 0); } -static void openctl_elem(struct loopback_handle *lhandle, - int device, int subdevice, - const char *name, - snd_ctl_elem_value_t **elem) +static int openctl_elem_id(struct loopback_handle *lhandle, snd_ctl_elem_id_t *id, + snd_ctl_elem_value_t **elem) { int err; if (snd_ctl_elem_value_malloc(elem) < 0) { *elem = NULL; + return -ENOMEM; + } + snd_ctl_elem_value_set_id(*elem, id); + snd_ctl_elem_value_set_interface(*elem, SND_CTL_ELEM_IFACE_PCM); + err = snd_ctl_elem_read(lhandle->ctl, *elem); + if (err < 0) { + snd_ctl_elem_id_free(id); + snd_ctl_elem_value_free(*elem); + *elem = NULL; + return err; } else { - snd_ctl_elem_value_set_interface(*elem, - SND_CTL_ELEM_IFACE_PCM); - snd_ctl_elem_value_set_device(*elem, device); - snd_ctl_elem_value_set_subdevice(*elem, subdevice); - snd_ctl_elem_value_set_name(*elem, name); - err = snd_ctl_elem_read(lhandle->ctl, *elem); - if (err < 0) { - snd_ctl_elem_value_free(*elem); - *elem = NULL; - } else { - if (verbose) - snd_output_printf(lhandle->loopback->output, - "Opened PCM element %s of %s, device %d, subdevice %d\n", - name, snd_ctl_name(lhandle->ctl), device, subdevice); - } + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + snd_ctl_elem_id_get_name(id), snd_ctl_name(lhandle->ctl), + snd_ctl_elem_id_get_device(id), + snd_ctl_elem_id_get_subdevice(id)); + return 0; } } +static int openctl_elem(struct loopback_handle *lhandle, + int device, int subdevice, + const char *name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + if (snd_ctl_elem_id_malloc(&id)) + return -ENOMEM; + snd_ctl_elem_id_set_device(id, device); + snd_ctl_elem_id_set_subdevice(id, subdevice); + snd_ctl_elem_id_set_name(id, name); + return openctl_elem_id(lhandle, id, elem); +} + +static int openctl_elem_ascii(struct loopback_handle *lhandle, char *ascii_name, + snd_ctl_elem_value_t **elem) +{ + snd_ctl_elem_id_t *id; + + if (snd_ctl_elem_value_malloc(elem) < 0) + return -ENOMEM; + if (snd_ctl_elem_id_malloc(&id)) { + snd_ctl_elem_value_free(*elem); + return -ENOMEM; + } + if (snd_ctl_ascii_elem_id_parse(id, ascii_name)) { + snd_ctl_elem_id_free(id); + snd_ctl_elem_value_free(*elem); + fprintf(stderr, "Wrong control identifier: %s\n", ascii_name); + return -EINVAL; + } + return openctl_elem_id(lhandle, id, elem); +} + static int openctl(struct loopback_handle *lhandle, int device, int subdevice) { int err; lhandle->ctl_rate_shift = NULL; if (lhandle->loopback->play == lhandle) { + // play only + if (lhandle->prateshift_name) { + err = openctl_elem_ascii(lhandle, lhandle->prateshift_name, + &lhandle->ctl_rate_shift); + if (err < 0) { + logit(LOG_CRIT, "Unable to open playback PCM Rate Shift elem '%s'.\n", + lhandle->prateshift_name); + exit(EXIT_FAILURE); + } + } if (lhandle->loopback->controls) goto __events; return 0; } + // capture only openctl_elem(lhandle, device, subdevice, "PCM Notify", &lhandle->ctl_notify); openctl_elem(lhandle, device, subdevice, "PCM Rate Shift 100000", -- 2.25.1

1 0

[PATCH v3 00/16] ASoC: Add Rich Graph Card support
by Kuninori Morimoto 05 Oct '21

05 Oct '21

Hi Mark We already have Audio-Graph-Card which is Of-Graph base general sound card driver. Basically it supports basic CPU-Codec connection, and is also supporting DPCM connection. Because it was forcibly expanded to DPCM, DT parsing is very limited and very difficult to add new features on it, for example Multi-CPU/Codec support, Codec2Codec support, etc. This patch adds more flexible new Rich-Graph-Card driver for it. Audio-Graph-Card and Rich-Graph-Card are similar, but don't have full compatibility. The reason why I need Rich-Graph-Card instead of updating Audio-Graph-Card is that it is very difficult to keep compatibility. Rich-Graph-Card supports Normal/DPCM/Codec2Codec Connection wich Single/Multi DAIs. And it is possible to Customizing. This patch-set adds Rich-Graph-Card driver and customized driver sample, and DT settings sample which can be used for testing. To enable testing/debuging, this patch-set also adds Test-Component driver. We already have Dummy Component and/or Dummy DAI on soc-utils, but 1) we can't use it from DT, 2) it do nothing. Added new Test-Component can be used from DT, and it can indicate called function name. We can use it to trace callback order, understanding ALSA SoC behavior, etc, etc... Sample DT settings of Rich Graph Card is using Test-Component as CPU/Codec DAI. You can easily try to use/test it if you added below line to your DT file. Your .config needs to have below CONFIGs to use/test it. It will probe sample Sound Card which has Normal/DPCM/Multi/Codec2Codec connections. #include "../../../../../sound/soc/generic/rich-graph-card-sample.dtsi" CONFIG_SND_RICH_GRAPH_CARD CONFIG_SND_RICH_CUSTOM_CARD_SAMPLE CONFIG_SND_TEST_COMPONENT Because Audio Graph Card2 is still under experimental stage, it will indicate such warning when probing, and the DT might be updated/exchanged. It can use Codec2Codec, but it will start automatically when probed, and can't stop it so far. It should be updated. Link: https://lore.kernel.org/r/87k0xszlep.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871r8u4s6q.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87a6mhwyqn.wl-kuninori.morimoto.gx@renesas.com v1 -> v2 - don't use "port" base for_each loop v2 -> v3 - Rename audio-graph-card2 to rich-graph-card - Rename DSP to DPCM not to confuse - Normal/DPCM/Codec2Codec can use Single/Multi DAIs. - use dpcm/multi/codec2codec node instead of using extra compatible - Sample DTSI patch is separated to Single/Multi. Kuninori Morimoto (16): ASoC: test-component: add Test Component YAML bindings ASoC: test-component: add Test Component for Sound debug/test ASoC: simple-card-utils: add asoc_graph_is_ports0() ASoC: simple-card-utils: add codec2codec support ASoC: add Rich Graph Card driver ASoC: rich-graph-card: add Multi CPU/Codec support ASoC: rich-graph-card: add DPCM support ASoC: rich-graph-card: add Codec2Codec support ASoC: add Rich Graph Card Yaml Document ASoC: add Rich Graph Card Custom Sample ASoC: rich-graph-card-sample.dtsi: add Sample DT for Normal (Single) ASoC: rich-graph-card-sample.dtsi: add Sample DT for Normal (Nulti) ASoC: rich-graph-card-sample.dtsi: add DPCM sample (Single) ASoC: rich-graph-card-sample.dtsi: add DPCM sample (Multi) ASoC: rich-graph-card-sample.dtsi: add Codec2Codec sample (Single) ASoC: rich-graph-card-sample.dtsi: add Codec2Codec sample (Multi) .../bindings/sound/rich-graph-card.yaml | 57 + .../bindings/sound/test-component.yaml | 33 + include/sound/graph_card.h | 21 + include/sound/simple_card_utils.h | 4 + sound/soc/generic/Kconfig | 20 + sound/soc/generic/Makefile | 6 + sound/soc/generic/rich-custom-card-sample.c | 174 +++ sound/soc/generic/rich-graph-card-sample.dtsi | 225 +++ sound/soc/generic/rich-graph-card.c | 1277 +++++++++++++++++ sound/soc/generic/simple-card-utils.c | 46 +- sound/soc/generic/test-component.c | 659 +++++++++ 11 files changed, 2521 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/rich-graph-card.yaml create mode 100644 Documentation/devicetree/bindings/sound/test-component.yaml create mode 100644 sound/soc/generic/rich-custom-card-sample.c create mode 100644 sound/soc/generic/rich-graph-card-sample.dtsi create mode 100644 sound/soc/generic/rich-graph-card.c create mode 100644 sound/soc/generic/test-component.c -- 2.25.1

3 28

[PATCH 0/3] ASoC: SOF: topology: minor updates
by Pierre-Louis Bossart 05 Oct '21

05 Oct '21

Two patches to visualize settings and override dynamic pipelines + one correction for errors on connections. Pierre-Louis Bossart (2): ASoC: SOF: topology: show clks_control value in dynamic debug ASoC: SOF: topology: allow for dynamic pipelines override for debug Ranjani Sridharan (1): ASoC: SOF: topology: return error if sof_connect_dai_widget() fails sound/soc/sof/sof-priv.h | 6 ++++++ sound/soc/sof/topology.c | 24 +++++++++++++++--------- 2 files changed, 21 insertions(+), 9 deletions(-) -- 2.25.1

2 4

[PATCH 0/5] ASoC: SOF: Intel: add flags to turn on SSP clocks early
by Pierre-Louis Bossart 05 Oct '21

05 Oct '21

With the chip shortage, some GeminiLake Intel-based designs were respun and now rely on codecs that need the SSP bit clock turned on in the hw_params stage, not the trigger stage. This patchset mirrors the flags added in the SOF DAI_CONFIG IPC, and sets the flags when this capability is indicated as necessary in the topology files where the SSP configuration is stored. We initially considered a more generic solution with an on-demand SSP clock activation using the common clock framework. This would be a more elegant solution indeed, but it would have required more intrusive changes that would conflict with the SOF multi-client support (in-development), and more backport hassles on product branches. The on-demand activation of clocks is still a desired feature that will be enabled at a later point. Bard Liao (1): ASoC: SOF: dai-intel: add SOF_DAI_INTEL_SSP_CLKCTRL_MCLK/BCLK_ES bits Pierre-Louis Bossart (4): ASoC: SOF: dai: mirror group_id definition added in firmware ASoC: SOF: dai: include new flags for DAI_CONFIG ASoC: SOF: Intel: hda: add new flags for DAI_CONFIG ASoC: SOF: Intel: hda-dai: improve SSP DAI handling for dynamic pipelines include/sound/sof/dai-intel.h | 4 ++ include/sound/sof/dai.h | 10 ++++- sound/soc/sof/intel/hda-dai.c | 82 ++++++++++++++++++++++++++++++++++- sound/soc/sof/intel/hda.c | 6 +++ sound/soc/sof/sof-audio.c | 4 ++ 5 files changed, 103 insertions(+), 3 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: amd: acp-rt5645: Constify static snd_soc_ops
by Rikard Falkeborn 05 Oct '21

05 Oct '21

The struct cz_aif1_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/amd/acp-rt5645.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c index d6ba94677ac2..6d5c547a32de 100644 --- a/sound/soc/amd/acp-rt5645.c +++ b/sound/soc/amd/acp-rt5645.c @@ -91,7 +91,7 @@ static int cz_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static struct snd_soc_ops cz_aif1_ops = { +static const struct snd_soc_ops cz_aif1_ops = { .hw_params = cz_aif1_hw_params, }; -- 2.33.0

2 1

[PATCH] ASoC: SOF: OF: Add fw_path and tplg_path parameters
by Daniel Baluta 05 Oct '21

05 Oct '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This allows specifying an alternate path for SOF firmware or SOF topology. This is particularly useful for i.MX when running Linux vs Android. Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/sof-of-dev.c | 21 ++++++++++++++++++--- 1 file changed, 18 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index f0f819a46456..885430a42226 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -7,12 +7,21 @@ #include <linux/firmware.h> #include <linux/module.h> +#include <linux/moduleparam.h> #include <linux/pm_runtime.h> #include <sound/sof.h> #include "ops.h" #include "imx/imx-ops.h" +static char *fw_path; +module_param(fw_path, charp, 0444); +MODULE_PARM_DESC(fw_path, "alternate path for SOF firmware."); + +static char *tplg_path; +module_param(tplg_path, charp, 0444); +MODULE_PARM_DESC(tplg_path, "alternate path for SOF topology."); + /* platform specific devices */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) static struct sof_dev_desc sof_of_imx8qxp_desc = { @@ -87,9 +96,15 @@ static int sof_of_probe(struct platform_device *pdev) sof_pdata->dev = &pdev->dev; sof_pdata->fw_filename = desc->default_fw_filename; - /* TODO: read alternate fw and tplg filenames from DT */ - sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path; - sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path; + if (fw_path) + sof_pdata->fw_filename_prefix = fw_path; + else + sof_pdata->fw_filename_prefix = sof_pdata->desc->default_fw_path; + + if (tplg_path) + sof_pdata->tplg_filename_prefix = tplg_path; + else + sof_pdata->tplg_filename_prefix = sof_pdata->desc->default_tplg_path; /* set callback to be called on successful device probe to enable runtime_pm */ sof_pdata->sof_probe_complete = sof_of_probe_complete; -- 2.27.0

2 1

[PATCH AUTOSEL 5.14 15/40] ASoC: SOF: loader: release_firmware() on load failure to avoid batching
by Sasha Levin 05 Oct '21

05 Oct '21

From: Marc Herbert <marc.herbert(a)intel.com> [ Upstream commit 8a8e1813ffc35111fc0b6db49968ceb0e1615ced ] Invoke release_firmware() when the firmware fails to boot in sof_probe_continue(). The request_firmware() framework must be informed of failures in sof_probe_continue() otherwise its internal "batching" feature (different from caching) cached the firmware image forever. Attempts to correct the file in /lib/firmware/ were then silently and confusingly ignored until the next reboot. Unloading the drivers did not help because from their disconnected perspective the firmware had failed so there was nothing to release. Also leverage the new snd_sof_fw_unload() function to simplify the snd_sof_device_remove() function. Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/20210916085008.28929-1-peter.ujfalusi@linux.intel… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 4 +--- sound/soc/sof/loader.c | 2 ++ 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e4dd4a86363..59d0d7b2b55c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev) dev_warn(dev, "error: %d failed to prepare DSP for device removal", ret); - snd_sof_fw_unload(sdev); snd_sof_ipc_free(sdev); snd_sof_free_debug(sdev); snd_sof_free_trace(sdev); @@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev) snd_sof_remove(sdev); /* release firmware */ - release_firmware(pdata->fw); - pdata->fw = NULL; + snd_sof_fw_unload(sdev); return 0; } diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..9c3f251a0dd0 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware); void snd_sof_fw_unload(struct snd_sof_dev *sdev) { /* TODO: support module unloading at runtime */ + release_firmware(sdev->pdata->fw); + sdev->pdata->fw = NULL; } EXPORT_SYMBOL(snd_sof_fw_unload); -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 08/40] ALSA: usb-audio: Unify mixer resume and reset_resume procedure
by Sasha Levin 05 Oct '21

05 Oct '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 7b9cf9036609428e845dc300aec13822ba2c4ab3 ] USB-audio driver assumes that the normal resume would preserve the device configuration while reset_resume wouldn't, and tries to restore the mixer elements only at reset_resume callback. However, this seems too naive, and some devices do behave differently, resetting the volume at the normal resume; this resulted in the inconsistent volume that surprised users. This patch changes the mixer resume code to handle both the normal and reset resume in the same way, always restoring the original mixer element values. This allows us to unify the both callbacks as well as dropping the no longer used reset_resume field, which ends up with a good code reduction. A slight behavior change by this patch is that now we assign restore_mixer_value() as the default resume callback, and the function is no longer called at reset-resume when the resume callback is overridden by the quirk function. That is, if needed, the quirk resume function would have to handle similarly as restore_mixer_value() by itself. Reported-by: En-Shuo Hsu <enshuo(a)chromium.org> Cc: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> Link: https://lore.kernel.org/r/CADDZ45UPsbpAAqP6=ZkTT8BE-yLii4Y7xSDnjK550G2DhQsM… Link: https://lore.kernel.org/r/20210910105155.12862-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/usb/card.c | 18 ++++-------------- sound/usb/mixer.c | 26 ++++---------------------- sound/usb/mixer.h | 3 +-- sound/usb/mixer_quirks.c | 2 +- 4 files changed, 10 insertions(+), 39 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 6abfc9d079e7..fa75b7e72ad1 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -1020,7 +1020,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message) return 0; } -static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) +static int usb_audio_resume(struct usb_interface *intf) { struct snd_usb_audio *chip = usb_get_intfdata(intf); struct snd_usb_stream *as; @@ -1046,7 +1046,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) * we just notify and restart the mixers */ list_for_each_entry(mixer, &chip->mixer_list, list) { - err = snd_usb_mixer_resume(mixer, reset_resume); + err = snd_usb_mixer_resume(mixer); if (err < 0) goto err_out; } @@ -1066,20 +1066,10 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume) atomic_dec(&chip->active); /* allow autopm after this point */ return err; } - -static int usb_audio_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, false); -} - -static int usb_audio_reset_resume(struct usb_interface *intf) -{ - return __usb_audio_resume(intf, true); -} #else #define usb_audio_suspend NULL #define usb_audio_resume NULL -#define usb_audio_reset_resume NULL +#define usb_audio_resume NULL #endif /* CONFIG_PM */ static const struct usb_device_id usb_audio_ids [] = { @@ -1101,7 +1091,7 @@ static struct usb_driver usb_audio_driver = { .disconnect = usb_audio_disconnect, .suspend = usb_audio_suspend, .resume = usb_audio_resume, - .reset_resume = usb_audio_reset_resume, + .reset_resume = usb_audio_resume, .id_table = usb_audio_ids, .supports_autosuspend = 1, }; diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 9b713b4a5ec4..fa7cf982d39e 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -3655,33 +3655,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list) return 0; } -static int default_mixer_reset_resume(struct usb_mixer_elem_list *list) -{ - int err; - - if (list->resume) { - err = list->resume(list); - if (err < 0) - return err; - } - return restore_mixer_value(list); -} - -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume) +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer) { struct usb_mixer_elem_list *list; - usb_mixer_elem_resume_func_t f; int id, err; /* restore cached mixer values */ for (id = 0; id < MAX_ID_ELEMS; id++) { for_each_mixer_elem(list, mixer, id) { - if (reset_resume) - f = list->reset_resume; - else - f = list->resume; - if (f) { - err = f(list); + if (list->resume) { + err = list->resume(list); if (err < 0) return err; } @@ -3702,7 +3685,6 @@ 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 = NULL; - list->reset_resume = default_mixer_reset_resume; + list->resume = restore_mixer_value; #endif } diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h index ea41e7a1f7bf..16567912b998 100644 --- a/sound/usb/mixer.h +++ b/sound/usb/mixer.h @@ -70,7 +70,6 @@ struct usb_mixer_elem_list { bool is_std_info; usb_mixer_elem_dump_func_t dump; usb_mixer_elem_resume_func_t resume; - usb_mixer_elem_resume_func_t reset_resume; }; /* iterate over mixer element list of the given unit id */ @@ -122,7 +121,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, #ifdef CONFIG_PM int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer); -int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume); +int snd_usb_mixer_resume(struct usb_mixer_interface *mixer); #endif int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0a3cb8fd7d00..4a4d3361ac04 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer, *listp = list; list->mixer = mixer; list->id = id; - list->reset_resume = resume; + list->resume = resume; kctl = snd_ctl_new1(knew, list); if (!kctl) { kfree(list); -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 05/40] ALSA: oxfw: fix transmission method for Loud models based on OXFW971
by Sasha Levin 05 Oct '21

05 Oct '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 64794d6db49730d22f440aef0cf4da98a56a4ea3 ] Loud Technologies Mackie Onyx 1640i (former model) is identified as the model which uses OXFW971. The analysis of packet dump shows that it transfers events in blocking method of IEC 61883-6, however the default behaviour of ALSA oxfw driver is for non-blocking method. This commit adds code to detect it assuming that all of loud models based on OXFW971 have such quirk. It brings no functional change except for alignment rule of PCM buffer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210913021042.10085-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/oxfw/oxfw.c | 13 ++++++++----- 1 file changed, 8 insertions(+), 5 deletions(-) diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index cb5b5e3a481b..daf731364695 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id model = val; } - /* - * Mackie Onyx Satellite with base station has a quirk to report a wrong - * value in 'dbs' field of CIP header against its format information. - */ - if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + if (vendor == VENDOR_LOUD) { + // Mackie Onyx Satellite with base station has a quirk to report a wrong + // value in 'dbs' field of CIP header against its format information. oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS; + // OXFW971-based models may transfer events by blocking method. + if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD)) + oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION; + } + return 0; } -- 2.33.0

1 0

[PATCH AUTOSEL 5.14 03/40] ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic
by Sasha Levin 05 Oct '21

05 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 58eafe1ff52ee1ce255759fc15729519af180cbb ] The SoundWire BEs make use of 'stream' functions for .prepare and .trigger. These functions will in turn force a Bank Switch, which implies a wait operation. Mark SoundWire BEs as nonatomic for consistency, but keep all other types of BEs as is. The initialization of .nonatomic is done outside of the create_sdw_dailink helper to avoid adding more parameters to deal with a single exception to the rule that BEs are atomic. Suggested-by: Takashi Iwai <tiwai(a)suse.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: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Link: https://lore.kernel.org/r/20210907184436.33152-1-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1a867c73a48e..cb3afc4519cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -860,6 +860,11 @@ static int create_sdw_dailink(struct device *dev, int *be_index, cpus + *cpu_id, cpu_dai_num, codecs, codec_num, NULL, &sdw_ops); + /* + * SoundWire DAILINKs use 'stream' functions and Bank Switch operations + * based on wait_for_completion(), tag them as 'nonatomic'. + */ + dai_links[*be_index].nonatomic = true; ret = set_codec_init_func(link, dai_links + (*be_index)++, playback, group_id); -- 2.33.0

1 0

Re: [PATCH v2 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:41:31AM +0800, cy_huang wrote: > Very appoligized. I forget to migrate this line from my test 5.10 kernel. > Without this line will cause sound card codec link fail. > Please use the patch v2 to review the bindings and source code. Sorry, saw v1 before I saw this but I guess the issues still apply - a quick check seems like it.

1 0

Re: [PATCH 2/2] ASoC: rt9120: Add rt9210 audio amplifier support
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:16:55AM +0800, cy_huang wrote: > +static const char * const sdo_select_text[] = { > + "NONE", "INTF", "FINAL", "RMS Detect" > +}; Why not None and Final? > + if (event == SND_SOC_DAPM_PRE_PMU) > + snd_soc_component_write(comp, RT9120_REG_ERRRPT, 0); > + else if (event == SND_SOC_DAPM_POST_PMU) > + msleep(RT9120_AMPON_WAITMS); > + else if (event == SND_SOC_DAPM_POST_PMD) > + msleep(RT9120_AMPOFF_WAITMS); This should be a switch statement, it'd be clearer. > + /* Default config volume to 0dB */ > + snd_soc_component_write(comp, RT9120_REG_MSVOL, 0x180); > + /* Mute by default */ > + snd_soc_component_update_bits(comp, RT9120_REG_VOLRAMP, > + RT9120_MUTE_MASK, RT9120_MUTE_MASK); As ever you should leave the defaults at whatever the hardware defaults to, the defaults for one machine may not be suitable for another so we shouldn't be trying to do that in software.

1 0

Re: [PATCH 1/2] ASoC: dt-bindings: rt9120: Add initial bindings
by Mark Brown 05 Oct '21

05 Oct '21

On Tue, Oct 05, 2021 at 09:16:54AM +0800, cy_huang wrote: > + richtek,use-dvdd-1p8v: > + description: Indicate DVDD 1P8V is used, default for 3P3V or 5V design > + type: boolean I would expect this to be done through the regulator bindings, they would allow the driver to query the supply voltage.

1 0

[PATCH] ASoC: rt5682: move clk related code to rt5682_i2c_probe
by Jack Yu 05 Oct '21

05 Oct '21

The DAI clock is only used in I2S mode, to make it clear and to fix clock resource release issue, we move CCF clock related code to rt5682_i2c_probe to fix clock register/unregister issue. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt5682-i2c.c | 22 +++++++++++ sound/soc/codecs/rt5682.c | 70 +++++++++++++---------------------- sound/soc/codecs/rt5682.h | 3 ++ 3 files changed, 51 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index b9d5d7a0975b..a30e42932cf7 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -139,6 +139,8 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, rt5682); + rt5682->i2c_dev = &i2c->dev; + rt5682->pdata = i2s_default_platform_data; if (pdata) @@ -276,6 +278,26 @@ static int rt5682_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } +#ifdef CONFIG_COMMON_CLK + /* Check if MCLK provided */ + rt5682->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(rt5682->mclk)) { + if (PTR_ERR(rt5682->mclk) != -ENOENT) { + ret = PTR_ERR(rt5682->mclk); + return ret; + } + rt5682->mclk = NULL; + } + + /* Register CCF DAI clock control */ + ret = rt5682_register_dai_clks(rt5682); + if (ret) + return ret; + + /* Initial setup for CCF */ + rt5682->lrck[RT5682_AIF1] = 48000; +#endif + return devm_snd_soc_register_component(&i2c->dev, &rt5682_soc_component_dev, rt5682_dai, ARRAY_SIZE(rt5682_dai)); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 12113c2dcae2..914fe7debc05 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2510,7 +2510,7 @@ static int rt5682_set_bias_level(struct snd_soc_component *component, static bool rt5682_clk_check(struct rt5682_priv *rt5682) { if (!rt5682->master[RT5682_AIF1]) { - dev_dbg(rt5682->component->dev, "sysclk/dai not set correctly\n"); + dev_dbg(rt5682->i2c_dev, "sysclk/dai not set correctly\n"); return false; } return true; @@ -2521,13 +2521,15 @@ static int rt5682_wclk_prepare(struct clk_hw *hw) struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); + struct snd_soc_component *component; + struct snd_soc_dapm_context *dapm; if (!rt5682_clk_check(rt5682)) return -EINVAL; + component = rt5682->component; + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); @@ -2557,13 +2559,15 @@ static void rt5682_wclk_unprepare(struct clk_hw *hw) struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; - struct snd_soc_dapm_context *dapm = - snd_soc_component_get_dapm(component); + struct snd_soc_component *component; + struct snd_soc_dapm_context *dapm; if (!rt5682_clk_check(rt5682)) return; + component = rt5682->component; + dapm = snd_soc_component_get_dapm(component); + snd_soc_dapm_mutex_lock(dapm); snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); @@ -2587,7 +2591,6 @@ static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; const char * const clk_name = clk_hw_get_name(hw); if (!rt5682_clk_check(rt5682)) @@ -2597,7 +2600,7 @@ static unsigned long rt5682_wclk_recalc_rate(struct clk_hw *hw, */ if (rt5682->lrck[RT5682_AIF1] != CLK_48 && rt5682->lrck[RT5682_AIF1] != CLK_44) { - dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + dev_warn(rt5682->i2c_dev, "%s: clk %s only support %d or %d Hz output\n", __func__, clk_name, CLK_44, CLK_48); return 0; } @@ -2611,7 +2614,6 @@ static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; const char * const clk_name = clk_hw_get_name(hw); if (!rt5682_clk_check(rt5682)) @@ -2621,7 +2623,7 @@ static long rt5682_wclk_round_rate(struct clk_hw *hw, unsigned long rate, * It will force to 48kHz if not both. */ if (rate != CLK_48 && rate != CLK_44) { - dev_warn(component->dev, "%s: clk %s only support %d or %d Hz output\n", + dev_warn(rt5682->i2c_dev, "%s: clk %s only support %d or %d Hz output\n", __func__, clk_name, CLK_44, CLK_48); rate = CLK_48; } @@ -2635,7 +2637,7 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_WCLK_IDX]); - struct snd_soc_component *component = rt5682->component; + struct snd_soc_component *component; struct clk_hw *parent_hw; const char * const clk_name = clk_hw_get_name(hw); int pre_div; @@ -2644,6 +2646,8 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, if (!rt5682_clk_check(rt5682)) return -EINVAL; + component = rt5682->component; + /* * Whether the wclk's parent clk (mclk) exists or not, please ensure * it is fixed or set to 48MHz before setting wclk rate. It's a @@ -2653,12 +2657,12 @@ static int rt5682_wclk_set_rate(struct clk_hw *hw, unsigned long rate, */ parent_hw = clk_hw_get_parent(hw); if (!parent_hw) - dev_warn(component->dev, + dev_warn(rt5682->i2c_dev, "Parent mclk of wclk not acquired in driver. Please ensure mclk was provided as %d Hz.\n", CLK_PLL2_FIN); if (parent_rate != CLK_PLL2_FIN) - dev_warn(component->dev, "clk %s only support %d Hz input\n", + dev_warn(rt5682->i2c_dev, "clk %s only support %d Hz input\n", clk_name, CLK_PLL2_FIN); /* @@ -2690,10 +2694,9 @@ static unsigned long rt5682_bclk_recalc_rate(struct clk_hw *hw, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_BCLK_IDX]); - struct snd_soc_component *component = rt5682->component; unsigned int bclks_per_wclk; - bclks_per_wclk = snd_soc_component_read(component, RT5682_TDM_TCON_CTRL); + regmap_read(rt5682->regmap, RT5682_TDM_TCON_CTRL, &bclks_per_wclk); switch (bclks_per_wclk & RT5682_TDM_BCLK_MS1_MASK) { case RT5682_TDM_BCLK_MS1_256: @@ -2754,20 +2757,22 @@ static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, struct rt5682_priv *rt5682 = container_of(hw, struct rt5682_priv, dai_clks_hw[RT5682_DAI_BCLK_IDX]); - struct snd_soc_component *component = rt5682->component; + struct snd_soc_component *component; struct snd_soc_dai *dai; unsigned long factor; if (!rt5682_clk_check(rt5682)) return -EINVAL; + component = rt5682->component; + factor = rt5682_bclk_get_factor(rate, parent_rate); for_each_component_dais(component, dai) if (dai->id == RT5682_AIF1) break; if (!dai) { - dev_err(component->dev, "dai %d not found in component\n", + dev_err(rt5682->i2c_dev, "dai %d not found in component\n", RT5682_AIF1); return -ENODEV; } @@ -2790,10 +2795,9 @@ static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { }, }; -static int rt5682_register_dai_clks(struct snd_soc_component *component) +int rt5682_register_dai_clks(struct rt5682_priv *rt5682) { - struct device *dev = component->dev; - struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct device *dev = rt5682->i2c_dev; struct rt5682_platform_data *pdata = &rt5682->pdata; struct clk_hw *dai_clk_hw; int i, ret; @@ -2851,6 +2855,7 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) return 0; } +EXPORT_SYMBOL_GPL(rt5682_register_dai_clks); #endif /* CONFIG_COMMON_CLK */ static int rt5682_probe(struct snd_soc_component *component) @@ -2860,9 +2865,6 @@ static int rt5682_probe(struct snd_soc_component *component) unsigned long time; struct snd_soc_dapm_context *dapm = &component->dapm; -#ifdef CONFIG_COMMON_CLK - int ret; -#endif rt5682->component = component; if (rt5682->is_sdw) { @@ -2874,26 +2876,6 @@ static int rt5682_probe(struct snd_soc_component *component) dev_err(&slave->dev, "Initialization not complete, timed out\n"); return -ETIMEDOUT; } - } else { -#ifdef CONFIG_COMMON_CLK - /* Check if MCLK provided */ - rt5682->mclk = devm_clk_get(component->dev, "mclk"); - if (IS_ERR(rt5682->mclk)) { - if (PTR_ERR(rt5682->mclk) != -ENOENT) { - ret = PTR_ERR(rt5682->mclk); - return ret; - } - rt5682->mclk = NULL; - } - - /* Register CCF DAI clock control */ - ret = rt5682_register_dai_clks(component); - if (ret) - return ret; - - /* Initial setup for CCF */ - rt5682->lrck[RT5682_AIF1] = CLK_48; -#endif } snd_soc_dapm_disable_pin(dapm, "MICBIAS"); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b59221048ebf..262512babc50 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1408,6 +1408,7 @@ enum { struct rt5682_priv { struct snd_soc_component *component; + struct device *i2c_dev; struct rt5682_platform_data pdata; struct regmap *regmap; struct regmap *sdw_regmap; @@ -1462,6 +1463,8 @@ void rt5682_calibrate(struct rt5682_priv *rt5682); void rt5682_reset(struct rt5682_priv *rt5682); int rt5682_parse_dt(struct rt5682_priv *rt5682, struct device *dev); +int rt5682_register_dai_clks(struct rt5682_priv *rt5682); + #define RT5682_REG_NUM 318 extern const struct reg_default rt5682_reg[RT5682_REG_NUM]; -- 2.33.0

3 2

User unsubscribed from mailing list
by Manish DUBEY 05 Oct '21

05 Oct '21

Hello Team, Hope you are doing well. We have some users in ST environment being subscribed to your Mailing list. After few days they use to get an email stating that "Your membership in the mailing list has been disabled due to excessive bounces".. We have verified at our end and didn't find any bounce back email getting generated. To investigate the case, could you please share couple of sample bounce back (NDR) email received by you from ST environment. The NDR may help us to investigate the cause of bounce back email getting generated. If possible, we can schedule a call as well to discuss the case. Feel free to share your availability to discuss the case and we can share a Teams invite accordingly. Awaiting your response. [cid:image001.png@01D7B9D6.F6BFBDE0] Manish Dubey | Messaging Support Mobile: +91 9716060208 ST Restricted

2 1

User unsubscribed from mailing list
by Manish DUBEY 05 Oct '21

05 Oct '21

Hello Team, Hope you are doing well. We have some users in ST environment being subscribed to your Mailing list. After few days they use to get an email stating that "Your membership in the mailing list has been disabled due to excessive bounces".. We have verified at our end and didn't find any bounce back email getting generated. To investigate the case, could you please share couple of sample bounce back (NDR) email received by you from ST environment. The NDR may help us to investigate the cause of bounce back email getting generated. If possible, we can schedule a call as well to discuss the case. Feel free to share your availability to discuss the case and we can share a Teams invite accordingly. Awaiting your response. [cid:image001.png@01D7B9D6.F6BFBDE0] Manish Dubey | Messaging Support Mobile: +91 9716060208 [cid:image001.png@01D7B9D6.F6BFBDE0] Manish Dubey | Messaging Support Mobile: +91 9716060208 ST Restricted

1 0

[PATCH] ASoC: qcom: apq8096: Constify static snd_soc_ops
by Rikard Falkeborn 05 Oct '21

05 Oct '21

The struct iapq8096_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/qcom/apq8096.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1a69baefc5ce..c7b7d0864d1a 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -60,7 +60,7 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops apq8096_ops = { +static const struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, }; -- 2.33.0

1 0

MIDI 2.0 support in ALSA
by GitHub issues - opened 05 Oct '21

05 Oct '21

alsa-project/alsa-lib issue #180 was opened from robertwu1: Hi, now that USB MIDI 2.0 is officially out (https://www.midi.org/specifications/midi-2-0-specifications), are there any updates for support in USB MIDI 2.0 in ALSA? https://github.com/alsa-project/alsa-lib/issues/24 was closed in February of 2020 but it's been over a year and a half. There seems to be quite a bit of interest in the issue since it had been closed. Thanks! Issue URL : https://github.com/alsa-project/alsa-lib/issues/180 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[BUG?] Setting (start_threshold > stop_threshold) makes snd_pcm_writei(a_small_buffer) XRUN immediately
by Zhang Boyang 05 Oct '21

05 Oct '21

Hello, I'm using ALSA to develop an audio-streaming application. I try to use start_threshold and stop_threshold in combination with small buffers. However, I think I probably found a bug about this. I'm setting start_threshold=100 and stop_threshold=50. I'm also using a buffer of 44 frames. When I call snd_pcm_writei(the_small_44_frames_buffer), pcm state came to XRUN from PREPARED directly. I think this is a bug because the stream hasn't started. It's hard to say a xrun occurred while stream not started. I'm wondering if this is a ALSA-bug or a misuse of ALSA. A simple bug test program is attached. Thank you very much! Zhang Boyang p.s. I dug into kernel code. After writting hardware buffer, __snd_pcm_lib_xfer() called snd_pcm_update_state(), which set the XRUN state.

2 5

[PATCH 1/2] ASoC: es8316: Use IRQF_NO_AUTOEN when requesting the IRQ
by Hans de Goede 04 Oct '21

04 Oct '21

Use the new IRQF_NO_AUTOEN flag when requesting the IRQ, rather then disabling it immediately after requesting it. This fixes a possible race where the IRQ might trigger between requesting and disabling it; and this also leads to a small code cleanup. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/es8316.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 067757d1d70a..5fb02635c140 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -811,12 +811,9 @@ static int es8316_i2c_probe(struct i2c_client *i2c_client, mutex_init(&es8316->lock); ret = devm_request_threaded_irq(dev, es8316->irq, NULL, es8316_irq, - IRQF_TRIGGER_HIGH | IRQF_ONESHOT, + IRQF_TRIGGER_HIGH | IRQF_ONESHOT | IRQF_NO_AUTOEN, "es8316", es8316); - if (ret == 0) { - /* Gets re-enabled by es8316_set_jack() */ - disable_irq(es8316->irq); - } else { + if (ret) { dev_warn(dev, "Failed to get IRQ %d: %d\n", es8316->irq, ret); es8316->irq = -ENXIO; } -- 2.31.1

2 2

[PATCH] ASoC: dt-bindings: uniphier: Add description of each port number
by Kunihiko Hayashi 04 Oct '21

04 Oct '21

To eliminate FIXME, add descriptions of what each port number is associated with. Cc: Rob Herring <robh(a)kernel.org> Signed-off-by: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> --- .../bindings/sound/socionext,uniphier-aio.yaml | 22 +++++++++++++++++++++- .../bindings/sound/socionext,uniphier-evea.yaml | 6 +++++- 2 files changed, 26 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml index 55ae198..70f62ec 100644 --- a/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-aio.yaml @@ -46,7 +46,27 @@ properties: patternProperties: "^port@[0-9]$": - description: FIXME, Need to define what each port is. + description: | + Port number of DT node is specified by the following DAI channels that + depends on SoC. + ld11-aio,ld20-aio: + 0: hdmi + 1: pcmin2 + 2: line + 3: hpcmout1 + 4: pcmout3 + 5: hiecout1 + 6: epcmout2 + 7: epcmout3 + 8: hieccompout1 + pxs2-aio: + 0: hdmi + 1: line + 2: aux + 3: hiecout1 + 4: iecout1 + 5: hieccompout1 + 6: ieccompout1 $ref: audio-graph-port.yaml# unevaluatedProperties: false diff --git a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml index 48ddfcb..be6acfd 100644 --- a/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml +++ b/Documentation/devicetree/bindings/sound/socionext,uniphier-evea.yaml @@ -40,7 +40,11 @@ properties: patternProperties: "^port@[0-9]$": - description: FIXME, Need to define what each port is. + description: | + Port number of DT node is specified by the following DAI channels. + 0: line1 + 1: hp + 2: line2 $ref: audio-graph-port.yaml# unevaluatedProperties: false -- 2.7.4

2 1

[PATCH 1/4] ASoC: nau8824: Fix NAU8824_JACK_LOGIC define
by Hans de Goede 04 Oct '21

04 Oct '21

The NAU8824_JACK_LOGIC define was wrong, for active high jack-detect to work bit 1 needs to be set, rather then bit 0. The correct bit was found in the Android kernel source dump for a Cyberbook T116 tablet; and this was also tested on that same tablet. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/nau8824.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.h b/sound/soc/codecs/nau8824.h index 1d7bdd8e0523..6e61405f623b 100644 --- a/sound/soc/codecs/nau8824.h +++ b/sound/soc/codecs/nau8824.h @@ -197,7 +197,7 @@ /* JACK_DET_CTRL (0x0D) */ #define NAU8824_JACK_EJECT_DT_SFT 2 #define NAU8824_JACK_EJECT_DT_MASK (0x3 << NAU8824_JACK_EJECT_DT_SFT) -#define NAU8824_JACK_LOGIC 0x1 +#define NAU8824_JACK_LOGIC (0x1 << 1) /* INTERRUPT_SETTING_1 (0x0F) */ -- 2.31.1

3 5

[PATCH 0/8] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 04 Oct '21

04 Oct '21

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. This patch set depends on: -- https://patchwork.kernel.org/project/alsa-devel/list/?series=549577 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=548765 -- https://patchwork.kernel.org/project/alsa-devel/list/?series=543829 Srinivasa Rao Mandadapu (8): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lapss CPU driver for codec dma control ASoC: qcom: Add support for codec dma driver ASoC: dt-bindings: Add SC7280 sound card bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoc: qcom: lpass: Add suspend and resume for sc7280 platform .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 69 ++- sound/soc/qcom/lpass-cdc-dma.c | 263 +++++++++ sound/soc/qcom/lpass-cpu.c | 334 +++++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 103 +++- sound/soc/qcom/lpass-platform.c | 504 +++++++++++++++-- sound/soc/qcom/lpass-sc7280.c | 602 +++++++++++++++++++++ sound/soc/qcom/lpass.h | 156 ++++++ 7 files changed, 1974 insertions(+), 57 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

3 13

[PATCH 1/2] dt-bindings: sound: tlv320adc3xxx: New codec driver
by Ricard Wanderlof 04 Oct '21

04 Oct '21

DT bindings for tlv320adc3xxx driver, currently supporting Texas Instruments TLV320ADC3001 and TLV320ADC3101 audio ADCs. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- .../bindings/sound/ti,tlv320adc3xxx.yaml | 126 ++++++++++++++++++ include/dt-bindings/sound/tlv320adc3xxx.h | 33 +++++ 2 files changed, 159 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml new file mode 100644 index 000000000000..4720eac65177 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml @@ -0,0 +1,126 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320adc3xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADC3001/TLV320ADC3101 Stereo ADC + +maintainers: + - Ricard Wanderlof <ricardw(a)axis.com> + +description: | + Texas Instruments TLV320ADC3001 and TLV320ADC3101 Stereo ADC + https://www.ti.com/product/TLV320ADC3001 + https://www.ti.com/product/TLV320ADC3101 + +properties: + compatible: + enum: + - ti,tlv320adc3001 + - ti,tlv320adc3101 + + reg: + maxItems: 1 + description: I2C address + + "#sound-dai-cells": + const: 0 + + gpios: + maxItems: 1 + description: GPIO pin used for codec reset (RESET pin) + + clocks: + maxItems: 1 + description: Master clock (MCLK) + + ti,pll-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_PLL_DONT_SET - leave mode unchanged + - 1 # ADC3XXX_PLL_ENABLE - use the on-chip PLL + - 2 # ADC3XXX_PLL_BYPASS - do not use the on-chip PLL + - 3 # ADC3XXX_PLL_AUTO - use on-chip PLL if possible + default: 0 + description: | + Set on-chip PLL mode. + ADC3XXX_PLL_DONT_SET is intended for use when setting the clock mod + via the clock id in a machine driver and it is not to be changed + from the previous setting. + ADC3XXX_PLL_AUTO will enable on-chip PLL if possible, depending on the + master clock and sample rate combination. + + ti,dmdin-gpio1: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMDIN/GPIO1 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + + ti,dmclk-gpio2: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # ADC3XXX_GPIO_DISABLED - I/O buffers powered down and not used + - 1 # ADC3XXX_GPIO_INPUT - Various non-GPIO input functions + - 2 # ADC3XXX_GPIO_GPI - General purpose input + - 3 # ADC3XXX_GPIO_GPO - General purpose output + - 4 # ADC3XXX_GPIO_CLKOUT - Clock source set in CLKOUT_MUX reg + - 5 # ADC3XXX_GPIO_INT1 - INT1 output + - 6 # ADC3XXX_GPIO_SECONDARY_BCLK - Codec interface secondary BCLK + - 7 # ADC3XXX_GPIO_SECONDARY_WCLK - Codec interface secondary WCLK + default: 0 + description: | + Configuration for DMCLK/GPIO2 pin. + + When ADC3XXX_GPIO_GPO is configured, this causes corresponding the + ALSA control "GPIOx Output" to appear, as a switch control. + + Note that there is currently no support for reading the GPIO pins as + inputs. + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/sound/tlv320adc3xxx.h> + + i2c { + #address-cells = <1>; + #size-cells = <0>; + tlv320adc3101: tlv320adc3101@18 { + compatible = "ti,tlv320adc3101"; + reg = <0x18>; + gpios = <&gpio_pc 3 GPIO_ACTIVE_LOW>; + clocks = <&audio_mclk>; + ti,pll-mode = <ADC3XXX_PLL_BYPASS>; + ti,dmdin-gpio1 = <ADC3XXX_GPIO_GPO>; + }; + }; + + audio_mclk: clock { + compatible = "fixed-clock"; + #clock-cells = <0>; + clock-frequency = <24576000>; + }; +... diff --git a/include/dt-bindings/sound/tlv320adc3xxx.h b/include/dt-bindings/sound/tlv320adc3xxx.h new file mode 100644 index 000000000000..8d488a6c3186 --- /dev/null +++ b/include/dt-bindings/sound/tlv320adc3xxx.h @@ -0,0 +1,33 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_TLV320ADC3XXX_H +#define __DT_TLV320ADC3XXX_H + +/* PLL modes, derived from clk_id of set_sysclk callback, or set in + * devicetree using the ti,pll-mode property. + * + * The default behavior is to take the first matching entry in the clock + * table, which is intended to be the PLL based one if there is more than one. + * + * Setting the clock source using simple-card (clocks or + * system-clock-frequency property) sets clk_id = 0 = ADC3XXX_CLK_DONT_SET, + * which doesn't change whatever clock setting was previously set up. + */ +#define ADC3XXX_PLL_DONT_SET 0 /* Don't change mode */ +#define ADC3XXX_PLL_ENABLE 1 /* Use PLL for clock generation */ +#define ADC3XXX_PLL_BYPASS 2 /* Don't use PLL for clock generation */ +#define ADC3XXX_PLL_AUTO 3 /* Use first available mode */ + +#define ADC3XXX_GPIO_DISABLED 0 /* I/O buffers powered down */ +#define ADC3XXX_GPIO_INPUT 1 /* Various non-GPIO inputs */ +#define ADC3XXX_GPIO_GPI 2 /* General purpose input */ +#define ADC3XXX_GPIO_GPO 3 /* General purpose output */ +#define ADC3XXX_GPIO_CLKOUT 4 /* Source set in reg. CLKOUT_MUX */ +#define ADC3XXX_GPIO_INT1 5 /* INT1 output */ +#define ADC3XXX_GPIO_INT2 6 /* INT2 output */ +/* value 7 is reserved */ +#define ADC3XXX_GPIO_SECONDARY_BCLK 8 /* Codec interface secondary BCLK */ +#define ADC3XXX_GPIO_SECONDARY_WCLK 9 /* Codec interface secondary WCLK */ +#define ADC3XXX_GPIO_ADC_MOD_CLK 10 /* Clock output for digital mics */ +/* values 11-15 reserved */ + +#endif /* __DT_TLV320ADC3XXX_H */ -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 3

Re: [PATCH] ASoc: fix ASoC driver to support ops to register get_time_info
by Mark Brown 04 Oct '21

04 Oct '21

On Mon, Oct 04, 2021 at 07:38:22PM +0800, Chien-Yu Lin wrote: > Dear Sir, > > Sorry for bothering you. > > Does it have any status updated, please? Please don't send content free pings and please allow a reasonable time for review. People get busy, go on holiday, attend conferences and so on so unless there is some reason for urgency (like critical bug fixes) please allow at least a couple of weeks for review. If there have been review comments then people may be waiting for those to be addressed. 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 RFC] ASoC: sh: rcar: dma: : use proper DMAENGINE API for termination
by Wolfram Sang 04 Oct '21

04 Oct '21

dmaengine_terminate_all() is deprecated in favor of explicitly saying if it should be sync or async. Here, we want dmaengine_terminate_sync() because there is no other synchronization code in the driver to handle an async case. Signed-off-by: Wolfram Sang <wsa+renesas(a)sang-engineering.com> --- Morimoto-san, can you please double check if this works. I don't know this driver very well. Thank you! sound/soc/sh/rcar/dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 95aa26d62e4f..d1b46057df9e 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -101,7 +101,7 @@ static int rsnd_dmaen_stop(struct rsnd_mod *mod, struct rsnd_dmaen *dmaen = rsnd_dma_to_dmaen(dma); if (dmaen->chan) - dmaengine_terminate_all(dmaen->chan); + dmaengine_terminate_sync(dmaen->chan); return 0; } -- 2.30.2

3 3

[PATCH 0/2] Codec driver for TLV320ADC3001 and 3101
by Ricard Wanderlof 04 Oct '21

04 Oct '21

This is a codec driver for the TLV320ADC3001 and TLV320ADC3101 2 channel audio ADC chips from Texas Instruments. Based on an old driver for these chips that I was referred to by Texas Instruments, but which apparently was never upstreamed, I've taken patches from various incarnations, ported them and augmented them with various new features in order to make a working driver. There is currently no support for the on-chip "miniDSP". Although the TLV320ADC3001 and TLV320ADC3101 have similarities with other TI codec chips for which there are existing drivers, upon closer inspection I decided the differences were too great to warrant attempting to modify an existing driver, especially without access to all supported chips and resources for testing them all. There are certainly things that could be improved, among others in the area of PLL register value calculations, but it has been tested on the Axis ARTPEC-8 platform and has been found to work satisfactorily at least with a TLV320ADC3101 operating as an I2S slave at 48 kHz with 32 bit bit depth, 1 or 2 channels, both with the on-chip PLL enabled and disabled. I have not tested this on a TLV320ADC3001 chip, but I've retained the apparent differences from the '3101 (which seem quite small) from the original driver. /Ricard Ricard Wanderlof (2): dt-bindings: sound: tlv320adc3xxx: New codec driver ASoC: codec: tlv320adc3xxx: New codec driver .../bindings/sound/ti,tlv320adc3xxx.yaml | 126 ++ include/dt-bindings/sound/tlv320adc3xxx.h | 33 + sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adc3xxx.c | 1239 +++++++++++++++++ sound/soc/codecs/tlv320adc3xxx.h | 381 +++++ 6 files changed, 1788 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320adc3xxx.yaml create mode 100644 include/dt-bindings/sound/tlv320adc3xxx.h create mode 100644 sound/soc/codecs/tlv320adc3xxx.c create mode 100644 sound/soc/codecs/tlv320adc3xxx.h -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

1 0

[PATCH] alsaloop: Adding openctl_elem debug log
by Pavel Hofman 04 Oct '21

04 Oct '21

When debugging what ctl elem was found and opened (if any) a debug log is helpful. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> --- alsaloop/pcmjob.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/alsaloop/pcmjob.c b/alsaloop/pcmjob.c index 8b72af4..845ab82 100644 --- a/alsaloop/pcmjob.c +++ b/alsaloop/pcmjob.c @@ -1191,6 +1191,11 @@ static void openctl_elem(struct loopback_handle *lhandle, if (err < 0) { snd_ctl_elem_value_free(*elem); *elem = NULL; + } else { + if (verbose) + snd_output_printf(lhandle->loopback->output, + "Opened PCM element %s of %s, device %d, subdevice %d\n", + name, snd_ctl_name(lhandle->ctl), device, subdevice); } } } -- 2.25.1

2 1

[PATCH] ALSA: usb-audio: Enable rate validation for Scarlett devices
by Takashi Iwai 04 Oct '21

04 Oct '21

The Scarlett device series from Focusrite Novation seem requiring the sample rate validations as we've done for MOTU devices; otherwise the driver probes invalid audioformat entries that contain the sample rates that actually don't work, and this may result in an incomplete setup as reported recently. This patch adds the needed quirk flag for enabling the sample rate validation for Focusrite Novation devices. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=214493 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 6ee6d24c847f..889c855addfc 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1900,6 +1900,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY), VENDOR_FLG(0x07fd, /* MOTU */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1235, /* Focusrite Novation */ + QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ -- 2.26.2

1 0

[PATCH 0/2] ALSA: hda/realtek: Fix name and add quirk for new Clevo
by Werner Sembach 04 Oct '21

04 Oct '21

Add quirk for Clevo X170KM-G to fix internal speaker not working and rename previously named "Clevo X170" into the more verbose "Clevo X170SM" to make the devices differentiable. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org>

2 3

[PATCH] ASoC: fsl: add COMPILE_TEST for SND_SOC_FSL_ASRC
by Randy Dunlap 04 Oct '21

04 Oct '21

Geert pointed out that since sound/soc has the soc_dummy_driver for NO_DMA platforms, it is possible (desirable) to have drivers that depend on HAS_DMA to alternately depend on COMPILE_TEST. This means that SND_ATMEL_FSL_ASRC can depend on HAS_DMA || COMPIE_TEST. Fixes: 121a01521b1e ("ASoC: fsl: fix build failure") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Suggested-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> Cc: Shengjiu Wang <shengjiu.wang(a)nxp.com> Cc: Fabio Estevam <festevam(a)gmail.com> --- sound/soc/fsl/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20211001.orig/sound/soc/fsl/Kconfig +++ linux-next-20211001/sound/soc/fsl/Kconfig @@ -5,7 +5,7 @@ comment "Common SoC Audio options for Fr config SND_SOC_FSL_ASRC tristate "Asynchronous Sample Rate Converter (ASRC) module support" - depends on HAS_DMA + depends on HAS_DMA || COMPILE_TEST select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM help

2 1

[PATCH 00/13] Extend AHUB audio support for Tegra210 and later
by Sameer Pujar 04 Oct '21

04 Oct '21

Earlier as part of series [0], support for ADMAIF and I/O modules (such as I2S, DMIC and DSPK) was added. This series aims at exposing some of the AHUB internal modules (listed below), which can be used for audio pre or post processing. * SFC (Sampling Frequency Converter) * MVC (Master Volume Control) * AMX (Audio Multiplexer) * ADX (Audio Demultiplexer) * Mixer These modules can be plugged into audio paths and relevant processing can be done. The MUX routes are extended to allow add or remove above modules in the path via mixer controls. This is similar to how specific ADMAIF channels are connected to relevant I/O module instances at the moment. Some of these modules can alter PCM parameters. Consider example of resampler (44.1 -> 48 kHz) in the path. aplay(44.1 kHz) -> ADMAIF -> SFC -> (48 kHz) I2S -> (48kHz) Codec The modules following SFC should be using converted sample rate and DAIs need to be configured accordingly. The audio-graph driver provides a mechanism to fixup the new parameters which can be specified in DT for a given DAI. Then core uses these new values via fixup callback and then pass it to respective DAIs hw_param() callback. The "convert-rate", described in [1], property can be used when there is rate conversion in the audio path. Similarly "convert-channels" can be used when there is channel conversion in the path. There is no "convert-xxx" property for sample size conversions. It can be added if necessary. [0] https://www.lkml.org/lkml/2020/7/21/1357 [1] Documentation/devicetree/bindings/sound/audio-graph-port.yaml Sameer Pujar (13): ASoC: soc-pcm: Don't reconnect an already active BE ASoC: simple-card-utils: Increase maximum DAI links limit to 512 ASoC: audio-graph: Fixup CPU endpoint hw_params in a BE<->BE link ASoC: dt-bindings: tegra: Few more Tegra210 AHUB modules ASoC: tegra: Add routes for few AHUB modules ASoC: tegra: Add Tegra210 based MVC driver ASoC: tegra: Add Tegra210 based SFC driver ASoC: tegra: Add Tegra210 based AMX driver ASoC: tegra: Add Tegra210 based ADX driver ASoC: tegra: Add Tegra210 based Mixer driver arm64: defconfig: Enable few Tegra210 based AHUB drivers arm64: tegra: Add few AHUB devices for Tegra210 and later arm64: tegra: Extend APE audio support on Jetson platforms .../bindings/sound/nvidia,tegra210-adx.yaml | 74 + .../bindings/sound/nvidia,tegra210-ahub.yaml | 20 + .../bindings/sound/nvidia,tegra210-amx.yaml | 72 + .../bindings/sound/nvidia,tegra210-mixer.yaml | 67 + .../bindings/sound/nvidia,tegra210-mvc.yaml | 79 + .../bindings/sound/nvidia,tegra210-sfc.yaml | 76 + arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 1554 ++++++++- arch/arm64/boot/dts/nvidia/tegra186.dtsi | 120 + arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 1493 ++++++++- .../arm64/boot/dts/nvidia/tegra194-p3509-0000.dtsi | 1520 ++++++++- arch/arm64/boot/dts/nvidia/tegra194.dtsi | 116 + arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 876 +++++ arch/arm64/boot/dts/nvidia/tegra210.dtsi | 77 + arch/arm64/configs/defconfig | 5 + include/sound/simple_card_utils.h | 2 +- sound/soc/generic/audio-graph-card.c | 4 +- sound/soc/soc-pcm.c | 4 + sound/soc/tegra/Kconfig | 48 + sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra210_adx.c | 527 +++ sound/soc/tegra/tegra210_adx.h | 72 + sound/soc/tegra/tegra210_ahub.c | 511 ++- sound/soc/tegra/tegra210_amx.c | 595 ++++ sound/soc/tegra/tegra210_amx.h | 93 + sound/soc/tegra/tegra210_mixer.c | 667 ++++ sound/soc/tegra/tegra210_mixer.h | 100 + sound/soc/tegra/tegra210_mvc.c | 629 ++++ sound/soc/tegra/tegra210_mvc.h | 117 + sound/soc/tegra/tegra210_sfc.c | 3542 ++++++++++++++++++++ sound/soc/tegra/tegra210_sfc.h | 78 + 31 files changed, 13601 insertions(+), 423 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-adx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-amx.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mixer.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mvc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-sfc.yaml create mode 100644 sound/soc/tegra/tegra210_adx.c create mode 100644 sound/soc/tegra/tegra210_adx.h create mode 100644 sound/soc/tegra/tegra210_amx.c create mode 100644 sound/soc/tegra/tegra210_amx.h create mode 100644 sound/soc/tegra/tegra210_mixer.c create mode 100644 sound/soc/tegra/tegra210_mixer.h create mode 100644 sound/soc/tegra/tegra210_mvc.c create mode 100644 sound/soc/tegra/tegra210_mvc.h create mode 100644 sound/soc/tegra/tegra210_sfc.c create mode 100644 sound/soc/tegra/tegra210_sfc.h -- 2.7.4

5 33

mpv segfaults in a52 plugin
by GitHub issues - opened 02 Oct '21

02 Oct '21

alsa-project/alsa-plugins issue #35 was opened from ykurlaev: When using mpv with the option `audio-device=alsa/<a52 device>` it gets a segfault on this line https://github.com/alsa-project/alsa-plugins/blob/da157e978d73f1947bb77637c… because it calls `dump` before `prepare` has been called and `rec->avctx` is NULL. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/35 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ASoC: nau8821: new driver
by Seven Lee 02 Oct '21

02 Oct '21

The driver is for codec NAU88L21 of Nuvoton Technology Corporation. The NAU88L21 is an ultra-low power high performance audio codec that supports both analog and digital audio functions. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8821.c | 1712 +++++++++++++++++ sound/soc/codecs/nau8821.h | 533 +++++ 5 files changed, 2307 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 sound/soc/codecs/nau8821.c create mode 100644 sound/soc/codecs/nau8821.h diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt new file mode 100644 index 000000000000..6c3baf7a5f21 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8821.txt @@ -0,0 +1,55 @@ +Nuvoton NAU88L21 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8821" + + - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. + + +Example: + + headset: nau8821@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + nuvoton,dmic-clk-threshold = 3072000; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index deda5ee02ebb..75506bd8fc0d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -138,6 +138,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_NAU8315 imply SND_SOC_NAU8540 imply SND_SOC_NAU8810 + imply SND_SOC_NAU8821 imply SND_SOC_NAU8822 imply SND_SOC_NAU8824 imply SND_SOC_NAU8825 @@ -1922,6 +1923,10 @@ config SND_SOC_NAU8810 tristate "Nuvoton Technology Corporation NAU88C10 CODEC" depends on I2C +config SND_SOC_NAU8821 + tristate "Nuvoton Technology Corporation NAU88L21 CODEC" + depends on I2C + config SND_SOC_NAU8822 tristate "Nuvoton Technology Corporation NAU88C22 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4cf939d0d3fb..90eedab4ee7a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -143,6 +143,7 @@ snd-soc-mt6660-objs := mt6660.o snd-soc-nau8315-objs := nau8315.o snd-soc-nau8540-objs := nau8540.o snd-soc-nau8810-objs := nau8810.o +snd-soc-nau8821-objs := nau8821.o snd-soc-nau8822-objs := nau8822.o snd-soc-nau8824-objs := nau8824.o snd-soc-nau8825-objs := nau8825.o @@ -473,6 +474,7 @@ obj-$(CONFIG_SND_SOC_MT6660) += snd-soc-mt6660.o obj-$(CONFIG_SND_SOC_NAU8315) += snd-soc-nau8315.o obj-$(CONFIG_SND_SOC_NAU8540) += snd-soc-nau8540.o obj-$(CONFIG_SND_SOC_NAU8810) += snd-soc-nau8810.o +obj-$(CONFIG_SND_SOC_NAU8821) += snd-soc-nau8821.o obj-$(CONFIG_SND_SOC_NAU8822) += snd-soc-nau8822.o obj-$(CONFIG_SND_SOC_NAU8824) += snd-soc-nau8824.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c new file mode 100644 index 000000000000..4ea56d2f9509 --- /dev/null +++ b/sound/soc/codecs/nau8821.c @@ -0,0 +1,1712 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// nau8821.c -- Nuvoton NAU88L21 audio codec driver +// +// Copyright 2021 Nuvoton Technology Corp. +// Author: John Hsu <kchsu0(a)nuvoton.com> +// Co-author: Seven Lee <wtli(a)nuvoton.com> +// + +#include <linux/acpi.h> +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/math64.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "nau8821.h" + +#define NAU_FREF_MAX 13500000 +#define NAU_FVCO_MAX 100000000 +#define NAU_FVCO_MIN 90000000 + +/* the maximum frequency of CLK_ADC and CLK_DAC */ +#define CLK_DA_AD_MAX 6144000 + +static int nau8821_configure_sysclk(struct nau8821 *nau8821, + int clk_id, unsigned int freq); + +struct nau8821_fll { + int mclk_src; + int ratio; + int fll_frac; + int fll_int; + int clk_ref_div; +}; + +struct nau8821_fll_attr { + unsigned int param; + unsigned int val; +}; + +/* scaling for mclk from sysclk_src output */ +static const struct nau8821_fll_attr mclk_src_scaling[] = { + { 1, 0x0 }, + { 2, 0x2 }, + { 4, 0x3 }, + { 8, 0x4 }, + { 16, 0x5 }, + { 32, 0x6 }, + { 3, 0x7 }, + { 6, 0xa }, + { 12, 0xb }, + { 24, 0xc }, + { 48, 0xd }, + { 96, 0xe }, + { 5, 0xf }, +}; + +/* ratio for input clk freq */ +static const struct nau8821_fll_attr fll_ratio[] = { + { 512000, 0x01 }, + { 256000, 0x02 }, + { 128000, 0x04 }, + { 64000, 0x08 }, + { 32000, 0x10 }, + { 8000, 0x20 }, + { 4000, 0x40 }, +}; + +static const struct nau8821_fll_attr fll_pre_scalar[] = { + { 0, 0x0 }, + { 1, 0x1 }, + { 2, 0x2 }, + { 3, 0x3 }, +}; + +/* over sampling rate */ +struct nau8821_osr_attr { + unsigned int osr; + unsigned int clk_src; +}; + +static const struct nau8821_osr_attr osr_dac_sel[] = { + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 256, 0 }, /* OSR 256, SRC 1 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 0, 0 }, + { 32, 3 }, /* OSR 32, SRC 1/8 */ +}; + +static const struct nau8821_osr_attr osr_adc_sel[] = { + { 32, 3 }, /* OSR 32, SRC 1/8 */ + { 64, 2 }, /* OSR 64, SRC 1/4 */ + { 128, 1 }, /* OSR 128, SRC 1/2 */ + { 256, 0 }, /* OSR 256, SRC 1 */ +}; + +struct nau8821_dmic_speed { + unsigned int param; + unsigned int val; +}; + +static const struct nau8821_dmic_speed dmic_speed_sel[] = { + { 0, 0x0 }, /*SPEED 1, SRC 1 */ + { 1, 0x1 }, /*SPEED 2, SRC 1/2 */ + { 2, 0x2 }, /*SPEED 4, SRC 1/4 */ + { 3, 0x3 }, /*SPEED 8, SRC 1/8 */ +}; + +static const struct reg_default nau8821_reg_defaults[] = { + { NAU8821_R01_ENA_CTRL, 0x00ff }, + { NAU8821_R03_CLK_DIVIDER, 0x0050 }, + { NAU8821_R04_FLL1, 0x0 }, + { NAU8821_R05_FLL2, 0x00bc }, + { NAU8821_R06_FLL3, 0x0008 }, + { NAU8821_R07_FLL4, 0x0010 }, + { NAU8821_R08_FLL5, 0x4000 }, + { NAU8821_R09_FLL6, 0x6900 }, + { NAU8821_R0A_FLL7, 0x0031 }, + { NAU8821_R0B_FLL8, 0x26e9 }, + { NAU8821_R0D_JACK_DET_CTRL, 0x0 }, + { NAU8821_R0F_INTERRUPT_MASK, 0x0 }, + { NAU8821_R12_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8821_R13_DMIC_CTRL, 0x0 }, + { NAU8821_R1A_GPIO12_CTRL, 0x0 }, + { NAU8821_R1B_TDM_CTRL, 0x0 }, + { NAU8821_R1C_I2S_PCM_CTRL1, 0x000a }, + { NAU8821_R1D_I2S_PCM_CTRL2, 0x8010 }, + { NAU8821_R1E_LEFT_TIME_SLOT, 0x0 }, + { NAU8821_R1F_RIGHT_TIME_SLOT, 0x0 }, + { NAU8821_R21_BIQ0_COF1, 0x0 }, + { NAU8821_R22_BIQ0_COF2, 0x0 }, + { NAU8821_R23_BIQ0_COF3, 0x0 }, + { NAU8821_R24_BIQ0_COF4, 0x0 }, + { NAU8821_R25_BIQ0_COF5, 0x0 }, + { NAU8821_R26_BIQ0_COF6, 0x0 }, + { NAU8821_R27_BIQ0_COF7, 0x0 }, + { NAU8821_R28_BIQ0_COF8, 0x0 }, + { NAU8821_R29_BIQ0_COF9, 0x0 }, + { NAU8821_R2A_BIQ0_COF10, 0x0 }, + { NAU8821_R2B_ADC_RATE, 0x0002 }, + { NAU8821_R2C_DAC_CTRL1, 0x0082 }, + { NAU8821_R2D_DAC_CTRL2, 0x0 }, + { NAU8821_R2F_DAC_DGAIN_CTRL, 0x0 }, + { NAU8821_R30_ADC_DGAIN_CTRL, 0x0 }, + { NAU8821_R31_MUTE_CTRL, 0x0 }, + { NAU8821_R32_HSVOL_CTRL, 0x0 }, + { NAU8821_R34_DACR_CTRL, 0xcfcf }, + { NAU8821_R35_ADC_DGAIN_CTRL1, 0xcfcf }, + { NAU8821_R36_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8821_R37_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8821_R38_ADC_DRC_SLOPES, 0x25ff }, + { NAU8821_R39_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8821_R3A_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8821_R3B_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8821_R3C_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8821_R3D_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8821_R41_BIQ1_COF1, 0x0 }, + { NAU8821_R42_BIQ1_COF2, 0x0 }, + { NAU8821_R43_BIQ1_COF3, 0x0 }, + { NAU8821_R44_BIQ1_COF4, 0x0 }, + { NAU8821_R45_BIQ1_COF5, 0x0 }, + { NAU8821_R46_BIQ1_COF6, 0x0 }, + { NAU8821_R47_BIQ1_COF7, 0x0 }, + { NAU8821_R48_BIQ1_COF8, 0x0 }, + { NAU8821_R49_BIQ1_COF9, 0x0 }, + { NAU8821_R4A_BIQ1_COF10, 0x0 }, + { NAU8821_R4B_CLASSG_CTRL, 0x0 }, + { NAU8821_R4C_IMM_MODE_CTRL, 0x0 }, + { NAU8821_R4D_IMM_RMS_L, 0x0 }, + { NAU8821_R53_OTPDOUT_1, 0xaad8 }, + { NAU8821_R54_OTPDOUT_2, 0x0002 }, + { NAU8821_R55_MISC_CTRL, 0x0 }, + { NAU8821_R66_BIAS_ADJ, 0x0 }, + { NAU8821_R68_TRIM_SETTINGS, 0x0 }, + { NAU8821_R69_ANALOG_CONTROL_1, 0x0 }, + { NAU8821_R6A_ANALOG_CONTROL_2, 0x0 }, + { NAU8821_R6B_PGA_MUTE, 0x0 }, + { NAU8821_R71_ANALOG_ADC_1, 0x0011 }, + { NAU8821_R72_ANALOG_ADC_2, 0x0020 }, + { NAU8821_R73_RDAC, 0x0008 }, + { NAU8821_R74_MIC_BIAS, 0x0006 }, + { NAU8821_R76_BOOST, 0x0 }, + { NAU8821_R77_FEPGA, 0x0 }, + { NAU8821_R7E_PGA_GAIN, 0x0 }, + { NAU8821_R7F_POWER_UP_CONTROL, 0x0 }, + { NAU8821_R80_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8821_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET ... NAU8821_R01_ENA_CTRL: + case NAU8821_R03_CLK_DIVIDER ... NAU8821_R0B_FLL8: + case NAU8821_R0D_JACK_DET_CTRL: + case NAU8821_R0F_INTERRUPT_MASK ... NAU8821_R13_DMIC_CTRL: + case NAU8821_R1A_GPIO12_CTRL ... NAU8821_R1F_RIGHT_TIME_SLOT: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2D_DAC_CTRL2: + case NAU8821_R2F_DAC_DGAIN_CTRL ... NAU8821_R32_HSVOL_CTRL: + case NAU8821_R34_DACR_CTRL ... NAU8821_R3D_DAC_DRC_ATKDCY: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4F_FUSE_CTRL3: + case NAU8821_R51_FUSE_CTRL1: + case NAU8821_R53_OTPDOUT_1 ... NAU8821_R55_MISC_CTRL: + case NAU8821_R58_I2C_DEVICE_ID ... NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R66_BIAS_ADJ: + case NAU8821_R68_TRIM_SETTINGS ... NAU8821_R6B_PGA_MUTE: + case NAU8821_R71_ANALOG_ADC_1 ... NAU8821_R74_MIC_BIAS: + case NAU8821_R76_BOOST ... NAU8821_R77_FEPGA: + case NAU8821_R7E_PGA_GAIN ... NAU8821_R82_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static bool nau8821_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET ... NAU8821_R01_ENA_CTRL: + case NAU8821_R03_CLK_DIVIDER ... NAU8821_R0B_FLL8: + case NAU8821_R0D_JACK_DET_CTRL: + case NAU8821_R0F_INTERRUPT_MASK: + case NAU8821_R11_INT_CLR_KEY_STATUS ... NAU8821_R13_DMIC_CTRL: + case NAU8821_R1A_GPIO12_CTRL ... NAU8821_R1F_RIGHT_TIME_SLOT: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2D_DAC_CTRL2: + case NAU8821_R2F_DAC_DGAIN_CTRL ... NAU8821_R32_HSVOL_CTRL: + case NAU8821_R34_DACR_CTRL ... NAU8821_R3D_DAC_DRC_ATKDCY: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4C_IMM_MODE_CTRL: + case NAU8821_R4E_FUSE_CTRL2 ... NAU8821_R4F_FUSE_CTRL3: + case NAU8821_R51_FUSE_CTRL1: + case NAU8821_R55_MISC_CTRL: + case NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R66_BIAS_ADJ: + case NAU8821_R68_TRIM_SETTINGS ... NAU8821_R6B_PGA_MUTE: + case NAU8821_R71_ANALOG_ADC_1 ... NAU8821_R74_MIC_BIAS: + case NAU8821_R76_BOOST ... NAU8821_R77_FEPGA: + case NAU8821_R7E_PGA_GAIN ... NAU8821_R80_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8821_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8821_R00_RESET: + case NAU8821_R10_IRQ_STATUS ... NAU8821_R11_INT_CLR_KEY_STATUS: + case NAU8821_R21_BIQ0_COF1 ... NAU8821_R2A_BIQ0_COF10: + case NAU8821_R41_BIQ1_COF1 ... NAU8821_R4A_BIQ1_COF10: + case NAU8821_R4D_IMM_RMS_L: + case NAU8821_R53_OTPDOUT_1 ... NAU8821_R54_OTPDOUT_2: + case NAU8821_R58_I2C_DEVICE_ID ... NAU8821_R5A_SOFTWARE_RST: + case NAU8821_R81_CHARGE_PUMP_INPUT_READ ... NAU8821_R82_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8821_biq_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + if (!component->regmap) + return -EINVAL; + + regmap_raw_read(component->regmap, NAU8821_R21_BIQ0_COF1, + ucontrol->value.bytes.data, params->max); + + return 0; +} + +static int nau8821_biq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + + if (!component->regmap) + return -EINVAL; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + regmap_raw_write(component->regmap, NAU8821_R21_BIQ0_COF1, + data, params->max); + + kfree(data); + + return 0; +} + +static const char * const nau8821_adc_decimation[] = { + "32", "64", "128", "256" }; + +static const struct soc_enum nau8821_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8821_R2B_ADC_RATE, NAU8821_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8821_adc_decimation), nau8821_adc_decimation); + +static const char * const nau8821_dac_oversampl[] = { + "64", "256", "128", "", "32" }; + +static const struct soc_enum nau8821_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8821_R2C_DAC_CTRL1, NAU8821_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8821_dac_oversampl), nau8821_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -6600, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(hp_vol_tlv, -900, 0); +static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -6600, 50, 1); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); +static const DECLARE_TLV_DB_MINMAX_MUTE(crosstalk_vol_tlv, -7000, 2400); + +static const struct snd_kcontrol_new nau8821_controls[] = { + SOC_DOUBLE_TLV("Mic Volume", NAU8821_R35_ADC_DGAIN_CTRL1, + NAU8821_ADCL_CH_VOL_SFT, NAU8821_ADCR_CH_VOL_SFT, + 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Headphone Bypass Volume", NAU8821_R30_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("Headphone Volume", NAU8821_R32_HSVOL_CTRL, + NAU8821_HPL_VOL_SFT, NAU8821_HPR_VOL_SFT, 0x3, 1, hp_vol_tlv), + SOC_DOUBLE_TLV("Digital Playback Volume", NAU8821_R34_DACR_CTRL, + NAU8821_DACL_CH_VOL_SFT, NAU8821_DACR_CH_VOL_SFT, + 0xcf, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Frontend PGA Volume", NAU8821_R7E_PGA_GAIN, + NAU8821_PGA_GAIN_L_SFT, NAU8821_PGA_GAIN_R_SFT, + 37, 0, fepga_gain_tlv), + SOC_DOUBLE_TLV("Headphone Crosstalk Volume", + NAU8821_R2F_DAC_DGAIN_CTRL, + 0, 8, 0xff, 0, crosstalk_vol_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8821_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8821_dac_oversampl_enum), + SND_SOC_BYTES_EXT("BIQ Coefficients", 20, + nau8821_biq_coeff_get, nau8821_biq_coeff_put), + SOC_SINGLE("ADC Phase Switch", NAU8821_R1B_TDM_CTRL, + NAU8821_ADCPHS_SFT, 1, 0), +}; + +static const struct snd_kcontrol_new nau8821_dmic_mode_switch = + SOC_DAPM_SINGLE("Switch", NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_EN_SFT, 1, 0); + +static int dmic_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + int i, speed_selection = -1, clk_adc_src, clk_adc; + unsigned int clk_divider_r03; + + /* The DMIC clock is gotten from adc clock divided by + * CLK_DMIC_SRC (1, 2, 4, 8). The clock has to be equal or + * less than nau8821->dmic_clk_threshold. + */ + regmap_read(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + &clk_divider_r03); + clk_adc_src = (clk_divider_r03 & NAU8821_CLK_ADC_SRC_MASK) + >> NAU8821_CLK_ADC_SRC_SFT; + clk_adc = (nau8821->fs * 256) >> clk_adc_src; + + for (i = 0 ; i < 4 ; i++) + if ((clk_adc >> dmic_speed_sel[i].param) <= + nau8821->dmic_clk_threshold) { + speed_selection = dmic_speed_sel[i].val; + break; + } + if (speed_selection < 0) + return -EINVAL; + + dev_dbg(nau8821->dev, + "clk_adc=%d, dmic_clk_threshold = %d, param=%d, val = %d\n", + clk_adc, nau8821->dmic_clk_threshold, + dmic_speed_sel[i].param, dmic_speed_sel[i].val); + regmap_update_bits(nau8821->regmap, NAU8821_R13_DMIC_CTRL, + NAU8821_DMIC_SRC_MASK, + (speed_selection << NAU8821_DMIC_SRC_SFT)); + + return 0; +} + +static int nau8821_left_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(125); + regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCL, NAU8821_EN_ADCL); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, + NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCL, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_right_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(125); + regmap_update_bits(nau8821->regmap, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCR, NAU8821_EN_ADCR); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, + NAU8821_R01_ENA_CTRL, NAU8821_EN_ADCR, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = + snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(20); + regmap_update_bits(nau8821->regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_JAMNODCLOW, NAU8821_JAMNODCLOW); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_update_bits(nau8821->regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_JAMNODCLOW, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int nau8821_output_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /* Disables the TESTDAC to let DAC signal pass through. */ + regmap_update_bits(nau8821->regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_TESTDAC_EN, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(nau8821->regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_TESTDAC_EN, NAU8821_BIAS_TESTDAC_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget nau8821_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_POWERUP_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC Clock", SND_SOC_NOPM, 0, 0, + dmic_clock_control, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC("ADCL Power", NULL, NAU8821_R72_ANALOG_ADC_2, + NAU8821_POWERUP_ADCL_SFT, 0), + SND_SOC_DAPM_ADC("ADCR Power", NULL, NAU8821_R72_ANALOG_ADC_2, + NAU8821_POWERUP_ADCR_SFT, 0), + SND_SOC_DAPM_PGA_S("Frontend PGA L", 1, NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_PGA_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Frontend PGA R", 1, NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_PGA_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADCL Digital path", 0, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCL_SFT, 0, nau8821_left_adc_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("ADCR Digital path", 0, NAU8821_R01_ENA_CTRL, + NAU8821_EN_ADCR_SFT, 0, nau8821_right_adc_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("DMIC Enable", SND_SOC_NOPM, + 0, 0, &nau8821_dmic_mode_switch), + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_TRISTATE_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIFRX", "Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8821_R73_RDAC, + NAU8821_DACL_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8821_R73_RDAC, + NAU8821_DACR_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACL Clock", 3, NAU8821_R73_RDAC, + NAU8821_DACL_CLK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("ADACR Clock", 3, NAU8821_R73_RDAC, + NAU8821_DACR_CLK_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8821_R01_ENA_CTRL, + NAU8821_EN_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8821_R01_ENA_CTRL, + NAU8821_EN_DACL_SFT, 0), + SND_SOC_DAPM_PGA_S("HP amp L", 0, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_LDAC_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HP amp R", 0, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_RDAC_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Charge Pump", 1, NAU8821_R80_CHARGE_PUMP, + NAU8821_CHANRGE_PUMP_EN_SFT, 0, nau8821_pump_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 1", 4, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_INTEG_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 1", 4, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_INTEG_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 2", 5, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_DRV_INSTG_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 2", 5, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_DRV_INSTG_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 6, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_MAIN_DRV_R_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 6, + NAU8821_R7F_POWER_UP_CONTROL, + NAU8821_PUP_MAIN_DRV_L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACL", 7, + NAU8821_R80_CHARGE_PUMP, NAU8821_POWER_DOWN_DACL_SFT, + 0, nau8821_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_S("Output DACR", 7, + NAU8821_R80_CHARGE_PUMP, NAU8821_POWER_DOWN_DACR_SFT, + 0, nau8821_output_dac_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* HPOL/R are ungrounded by disabling 16 Ohm pull-downs on playback */ + SND_SOC_DAPM_PGA_S("HPOL Pulldown", 8, + NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1L_SFT, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("HPOR Pulldown", 8, + NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R_SFT, 0, NULL, 0), + + /* High current HPOL/R boost driver */ + SND_SOC_DAPM_PGA_S("HP Boost Driver", 9, + NAU8821_R76_BOOST, NAU8821_HP_BOOST_DIS_SFT, 1, NULL, 0), + SND_SOC_DAPM_PGA("Class G", NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_INPUT("DMIC"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8821_dapm_routes[] = { + {"DMIC Enable", "Switch", "DMIC"}, + {"DMIC Enable", NULL, "DMIC Clock"}, + + {"Frontend PGA L", NULL, "MICL"}, + {"Frontend PGA R", NULL, "MICR"}, + {"Frontend PGA L", NULL, "MICBIAS"}, + {"Frontend PGA R", NULL, "MICBIAS"}, + + {"ADCL Power", NULL, "Frontend PGA L"}, + {"ADCR Power", NULL, "Frontend PGA R"}, + + {"ADCL Digital path", NULL, "ADCL Power"}, + {"ADCR Digital path", NULL, "ADCR Power"}, + {"ADCL Digital path", NULL, "DMIC Enable"}, + {"ADCR Digital path", NULL, "DMIC Enable"}, + + {"AIFTX", NULL, "ADCL Digital path"}, + {"AIFTX", NULL, "ADCR Digital path"}, + + {"DDACL", NULL, "AIFRX"}, + {"DDACR", NULL, "AIFRX"}, + + {"HP amp L", NULL, "DDACL"}, + {"HP amp R", NULL, "DDACR"}, + + {"Charge Pump", NULL, "HP amp L"}, + {"Charge Pump", NULL, "HP amp R"}, + + {"ADACL", NULL, "Charge Pump"}, + {"ADACR", NULL, "Charge Pump"}, + {"ADACL Clock", NULL, "ADACL"}, + {"ADACR Clock", NULL, "ADACR"}, + + {"Output Driver L Stage 1", NULL, "ADACL Clock"}, + {"Output Driver R Stage 1", NULL, "ADACR Clock"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + + {"HPOL Pulldown", NULL, "Output DACL"}, + {"HPOR Pulldown", NULL, "Output DACR"}, + {"HP Boost Driver", NULL, "HPOL Pulldown"}, + {"HP Boost Driver", NULL, "HPOR Pulldown"}, + + {"Class G", NULL, "HP Boost Driver"}, + {"HPOL", NULL, "Class G"}, + {"HPOR", NULL, "Class G"}, +}; + +static int nau8821_clock_check(struct nau8821 *nau8821, + int stream, int rate, int osr) +{ + int osrate = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (osr >= ARRAY_SIZE(osr_dac_sel)) + return -EINVAL; + osrate = osr_dac_sel[osr].osr; + } else { + if (osr >= ARRAY_SIZE(osr_adc_sel)) + return -EINVAL; + osrate = osr_adc_sel[osr].osr; + } + + if (!osrate || rate * osrate > CLK_DA_AD_MAX) { + dev_err(nau8821->dev, + "exceed the maximum frequency of CLK_ADC or CLK_DAC"); + return -EINVAL; + } + + return 0; +} + +static int nau8821_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int val_len = 0, osr, ctrl_val, bclk_fs, clk_div; + + nau8821->fs = params_rate(params); + /* CLK_DAC or CLK_ADC = OSR * FS + * DAC or ADC clock frequency is defined as Over Sampling Rate (OSR) + * multiplied by the audio sample rate (Fs). Note that the OSR and Fs + * values must be selected such that the maximum frequency is less + * than 6.144 MHz. + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_read(nau8821->regmap, NAU8821_R2C_DAC_CTRL1, &osr); + osr &= NAU8821_DAC_OVERSAMPLE_MASK; + if (nau8821_clock_check(nau8821, substream->stream, + nau8821->fs, osr)) { + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_DAC_SRC_MASK, + osr_dac_sel[osr].clk_src << NAU8821_CLK_DAC_SRC_SFT); + } else { + regmap_read(nau8821->regmap, NAU8821_R2B_ADC_RATE, &osr); + osr &= NAU8821_ADC_SYNC_DOWN_MASK; + if (nau8821_clock_check(nau8821, substream->stream, + nau8821->fs, osr)) { + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_ADC_SRC_MASK, + osr_adc_sel[osr].clk_src << NAU8821_CLK_ADC_SRC_SFT); + } + + /* make BCLK and LRC divde configuration if the codec as master. */ + regmap_read(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, &ctrl_val); + if (ctrl_val & NAU8821_I2S_MS_MASTER) { + /* get the bclk and fs ratio */ + bclk_fs = snd_soc_params_to_bclk(params) / nau8821->fs; + if (bclk_fs <= 32) + clk_div = 3; + else if (bclk_fs <= 64) + clk_div = 2; + else if (bclk_fs <= 128) + clk_div = 1; + else { + return -EINVAL; + } + regmap_update_bits(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_LRC_DIV_MASK | NAU8821_I2S_BLK_DIV_MASK, + (clk_div << NAU8821_I2S_LRC_DIV_SFT) | clk_div); + } + + switch (params_width(params)) { + case 16: + val_len |= NAU8821_I2S_DL_16; + break; + case 20: + val_len |= NAU8821_I2S_DL_20; + break; + case 24: + val_len |= NAU8821_I2S_DL_24; + break; + case 32: + val_len |= NAU8821_I2S_DL_32; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8821->regmap, NAU8821_R1C_I2S_PCM_CTRL1, + NAU8821_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8821_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + ctrl2_val |= NAU8821_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8821_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8821_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8821_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8821_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8821_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8821_I2S_DF_PCM_AB; + ctrl1_val |= NAU8821_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8821->regmap, NAU8821_R1C_I2S_PCM_CTRL1, + NAU8821_I2S_DL_MASK | NAU8821_I2S_DF_MASK | + NAU8821_I2S_BP_MASK | NAU8821_I2S_PCMB_MASK, ctrl1_val); + regmap_update_bits(nau8821->regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static int nau8821_digital_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + unsigned int val = 0; + + if (mute) + val = NAU8821_DAC_SOFT_MUTE; + + return regmap_update_bits(nau8821->regmap, + NAU8821_R31_MUTE_CTRL, NAU8821_DAC_SOFT_MUTE, val); +} + +static const struct snd_soc_dai_ops nau8821_dai_ops = { + .hw_params = nau8821_hw_params, + .set_fmt = nau8821_set_dai_fmt, + .mute_stream = nau8821_digital_mute, +}; + +#define NAU8821_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8821_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8821_dai = { + .name = NUVOTON_CODEC_DAI, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8821_RATES, + .formats = NAU8821_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8821_RATES, + .formats = NAU8821_FORMATS, + }, + .ops = &nau8821_dai_ops, +}; + + +static bool nau8821_is_jack_inserted(struct regmap *regmap) +{ + bool active_high, is_high; + int status, jkdet; + + regmap_read(regmap, NAU8821_R0D_JACK_DET_CTRL, &jkdet); + active_high = jkdet & NAU8821_JACK_POLARITY; + regmap_read(regmap, NAU8821_R82_GENERAL_STATUS, &status); + is_high = status & NAU8821_GPIO2_IN; + /* return jack connection status according to jack insertion logic + * active high or active low. + */ + return active_high == is_high; +} + +static void nau8821_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8821_R10_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8821_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8821_R11_INT_CLR_KEY_STATUS, clear_irq); + } +} + +static void nau8821_eject_jack(struct nau8821 *nau8821) +{ + struct snd_soc_dapm_context *dapm = nau8821->dapm; + struct regmap *regmap = nau8821->regmap; + struct snd_soc_component *component = snd_soc_dapm_to_component(dapm); + + /* Detach 2kOhm Resistors from MICBIAS to MICGND */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, 0); + /* HPL/HPR short to ground */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R | NAU8821_SPKR_DWN1L, 0); + snd_soc_component_disable_pin(component, "MICBIAS"); + snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8821_int_status_clear_all(regmap); + + /* Enable the insertion interruption, disable the ejection inter- + * ruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_EJECT_DIS | NAU8821_IRQ_INSERT_DIS, + NAU8821_IRQ_EJECT_DIS); + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_EJECT_EN); + + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, NAU8821_JACK_DET_DB_BYPASS); + + /* Close clock for jack type detection at manual mode */ + if (dapm->bias_level < SND_SOC_BIAS_PREPARE) + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + + /* Recover to normal channel input */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_R_SRC_EN, 0); +} + +static void nau8821_jdet_work(struct work_struct *work) +{ + struct nau8821 *nau8821 = + container_of(work, struct nau8821, jdet_work); + struct snd_soc_dapm_context *dapm = nau8821->dapm; + struct snd_soc_component *component = snd_soc_dapm_to_component(dapm); + struct regmap *regmap = nau8821->regmap; + int jack_status_reg, mic_detected, event = 0, event_mask = 0; + + snd_soc_component_force_enable_pin(component, "MICBIAS"); + snd_soc_dapm_sync(dapm); + msleep(20); + + regmap_read(regmap, NAU8821_R58_I2C_DEVICE_ID, &jack_status_reg); + mic_detected = !(jack_status_reg & NAU8821_KEYDET); + if (mic_detected) { + dev_dbg(nau8821->dev, "Headset connected\n"); + event |= SND_JACK_HEADSET; + + /* 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, NAU8821_MICBIAS_JKR2); + /* Latch Right Channel Analog data + * input into the Right Channel Filter + */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_R_SRC_EN, NAU8821_ADC_R_SRC_EN); + } else { + dev_dbg(nau8821->dev, "Headphone connected\n"); + event |= SND_JACK_HEADPHONE; + snd_soc_component_disable_pin(component, "MICBIAS"); + snd_soc_dapm_sync(dapm); + } + event_mask |= SND_JACK_HEADSET; + snd_soc_jack_report(nau8821->jack, event, event_mask); +} + +/* Enable interruptions with internal clock. */ +static void nau8821_setup_inserted_irq(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Enable internal VCO needed for interruptions */ + if (nau8821->dapm->bias_level < SND_SOC_BIAS_PREPARE) + nau8821_configure_sysclk(nau8821, NAU8821_CLK_INTERNAL, 0); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, NAU8821_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8821_R1D_I2S_PCM_CTRL2, + NAU8821_I2S_MS_MASK, NAU8821_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, 0); + + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN, 0); + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_EJECT_DIS, 0); +} + +static irqreturn_t nau8821_interrupt(int irq, void *data) +{ + struct nau8821 *nau8821 = (struct nau8821 *)data; + struct regmap *regmap = nau8821->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + if (regmap_read(regmap, NAU8821_R10_IRQ_STATUS, &active_irq)) { + dev_err(nau8821->dev, "failed to read irq status\n"); + return IRQ_NONE; + } + + dev_dbg(nau8821->dev, "IRQ %d\n", active_irq); + + if ((active_irq & NAU8821_JACK_EJECT_IRQ_MASK) == + NAU8821_JACK_EJECT_DETECTED) { + regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, + NAU8821_MICDET_MASK, NAU8821_MICDET_DIS); + nau8821_eject_jack(nau8821); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8821_JACK_EJECT_IRQ_MASK; + } else if ((active_irq & NAU8821_JACK_INSERT_IRQ_MASK) == + NAU8821_JACK_INSERT_DETECTED) { + regmap_update_bits(regmap, NAU8821_R71_ANALOG_ADC_1, + NAU8821_MICDET_MASK, NAU8821_MICDET_EN); + if (nau8821_is_jack_inserted(regmap)) { + /* detect microphone and jack type */ + cancel_work_sync(&nau8821->jdet_work); + schedule_work(&nau8821->jdet_work); + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_INSERT_DIS, + NAU8821_IRQ_INSERT_DIS); + regmap_update_bits(regmap, + NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_INSERT_EN); + nau8821_setup_inserted_irq(nau8821); + } else { + dev_warn(nau8821->dev, + "Inserted IRQ fired but not connected\n"); + nau8821_eject_jack(nau8821); + } + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8821_R11_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8821->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static const struct regmap_config nau8821_regmap_config = { + .val_bits = NAU8821_REG_DATA_LEN, + .reg_bits = NAU8821_REG_ADDR_LEN, + + .max_register = NAU8821_REG_MAX, + .readable_reg = nau8821_readable_reg, + .writeable_reg = nau8821_writeable_reg, + .volatile_reg = nau8821_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8821_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8821_reg_defaults), +}; + +static int nau8821_component_probe(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + + nau8821->dapm = dapm; + + return 0; +} + +/** + * nau8821_calc_fll_param - Calculate FLL parameters. + * @fll_in: external clock provided to codec. + * @fs: sampling rate. + * @fll_param: Pointer to structure of FLL parameters. + * + * Calculate FLL parameters to configure codec. + * + * Returns 0 for success or negative error code. + */ +static int nau8821_calc_fll_param(unsigned int fll_in, + unsigned int fs, struct nau8821_fll *fll_param) +{ + u64 fvco, fvco_max; + unsigned int fref, i, fvco_sel; + + /* Ensure the reference clock frequency (FREF) is <= 13.5MHz by + * dividing freq_in by 1, 2, 4, or 8 using FLL pre-scalar. + * FREF = freq_in / NAU8821_FLL_REF_DIV_MASK + */ + for (i = 0; i < ARRAY_SIZE(fll_pre_scalar); i++) { + fref = fll_in >> fll_pre_scalar[i].param; + if (fref <= NAU_FREF_MAX) + break; + } + if (i == ARRAY_SIZE(fll_pre_scalar)) + return -EINVAL; + fll_param->clk_ref_div = fll_pre_scalar[i].val; + + /* Choose the FLL ratio based on FREF */ + for (i = 0; i < ARRAY_SIZE(fll_ratio); i++) { + if (fref >= fll_ratio[i].param) + break; + } + if (i == ARRAY_SIZE(fll_ratio)) + return -EINVAL; + fll_param->ratio = fll_ratio[i].val; + + /* Calculate the frequency of DCO (FDCO) given freq_out = 256 * Fs. + * FDCO must be within the 90MHz - 100MHz or the FFL cannot be + * guaranteed across the full range of operation. + * FDCO = freq_out * 2 * mclk_src_scaling + */ + fvco_max = 0; + fvco_sel = ARRAY_SIZE(mclk_src_scaling); + for (i = 0; i < ARRAY_SIZE(mclk_src_scaling); i++) { + fvco = 256ULL * fs * 2 * mclk_src_scaling[i].param; + if (fvco > NAU_FVCO_MIN && fvco < NAU_FVCO_MAX && + fvco_max < fvco) { + fvco_max = fvco; + fvco_sel = i; + } + } + if (ARRAY_SIZE(mclk_src_scaling) == fvco_sel) + return -EINVAL; + fll_param->mclk_src = mclk_src_scaling[fvco_sel].val; + + /* Calculate the FLL 10-bit integer input and the FLL 24-bit fractional + * input based on FDCO, FREF and FLL ratio. + */ + fvco = div_u64(fvco_max << 24, fref * fll_param->ratio); + fll_param->fll_int = (fvco >> 24) & 0x3ff; + fll_param->fll_frac = fvco & 0xffffff; + + return 0; +} + +static void nau8821_fll_apply(struct nau8821 *nau8821, + struct nau8821_fll *fll_param) +{ + struct regmap *regmap = nau8821->regmap; + + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK | NAU8821_CLK_MCLK_SRC_MASK, + NAU8821_CLK_SRC_MCLK | fll_param->mclk_src); + /* Make DSP operate at high speed for better performance. */ + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_FLL_RATIO_MASK | NAU8821_ICTRL_LATCH_MASK, + fll_param->ratio | (0x6 << NAU8821_ICTRL_LATCH_SFT)); + /* FLL 24-bit fractional input */ + regmap_write(regmap, NAU8821_R0A_FLL7, + (fll_param->fll_frac >> 16) & 0xff); + regmap_write(regmap, NAU8821_R0B_FLL8, fll_param->fll_frac & 0xffff); + /* FLL 10-bit integer input */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_INTEGER_MASK, fll_param->fll_int); + /* FLL pre-scaler */ + regmap_update_bits(regmap, NAU8821_R07_FLL4, + NAU8821_HIGHBW_EN | NAU8821_FLL_REF_DIV_MASK, + NAU8821_HIGHBW_EN | + (fll_param->clk_ref_div << NAU8821_FLL_REF_DIV_SFT)); + /* select divided VCO input */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_CLK_SW_MASK, NAU8821_FLL_CLK_SW_REF); + /* Disable free-running mode */ + regmap_update_bits(regmap, + NAU8821_R09_FLL6, NAU8821_DCO_EN, 0); + if (fll_param->fll_frac) { + /* set FLL loop filter enable and cutoff frequency at 500Khz */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_MASK, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_FILTER); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN | NAU8821_CUTOFF500, + NAU8821_SDM_EN | NAU8821_CUTOFF500); + } else { + /* disable FLL loop filter and cutoff frequency */ + regmap_update_bits(regmap, NAU8821_R08_FLL5, + NAU8821_FLL_PDB_DAC_EN | NAU8821_FLL_LOOP_FTR_EN | + NAU8821_FLL_FTR_SW_MASK, NAU8821_FLL_FTR_SW_ACCU); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN | NAU8821_CUTOFF500, 0); + } +} + +/** + * nau8821_set_fll - FLL configuration of nau8821 + * @codec: codec component + * @freq_in: frequency of input clock source + * @freq_out: must be 256*Fs in order to achieve the best performance + * + * The FLL function can select BCLK or MCLK as the input clock source. + * + * Returns 0 if the parameters have been applied successfully + * or negative error code. + */ +static int nau8821_set_fll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, unsigned int freq_out) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct nau8821_fll fll_set_param, *fll_param = &fll_set_param; + int ret, fs; + + fs = freq_out >> 8; + ret = nau8821_calc_fll_param(freq_in, fs, fll_param); + if (ret) { + dev_err(nau8821->dev, + "Unsupported input clock %d to output clock %d\n", + freq_in, freq_out); + return ret; + } + dev_dbg(nau8821->dev, + "mclk_src=%x ratio=%x fll_frac=%x fll_int=%x clk_ref_div=%x\n", + fll_param->mclk_src, fll_param->ratio, fll_param->fll_frac, + fll_param->fll_int, fll_param->clk_ref_div); + + nau8821_fll_apply(nau8821, fll_param); + mdelay(2); + regmap_update_bits(nau8821->regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_VCO); + + return 0; +} + +static void nau8821_configure_mclk_as_sysclk(struct regmap *regmap) +{ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_DCO_EN, 0); + /* Make DSP operate as default setting for power saving. */ + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_ICTRL_LATCH_MASK, 0); +} + +static int nau8821_configure_sysclk(struct nau8821 *nau8821, + int clk_id, unsigned int freq) +{ + struct regmap *regmap = nau8821->regmap; + + switch (clk_id) { + case NAU8821_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + nau8821_configure_mclk_as_sysclk(regmap); + break; + case NAU8821_CLK_MCLK: + nau8821_configure_mclk_as_sysclk(regmap); + /* MCLK not changed by clock tree */ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_MCLK_SRC_MASK, 0); + break; + case NAU8821_CLK_INTERNAL: + if (nau8821_is_jack_inserted(regmap)) { + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_DCO_EN, NAU8821_DCO_EN); + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_SRC_MASK, NAU8821_CLK_SRC_VCO); + /* Decrease the VCO frequency and make DSP operate + * as default setting for power saving. + */ + regmap_update_bits(regmap, NAU8821_R03_CLK_DIVIDER, + NAU8821_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8821_R04_FLL1, + NAU8821_ICTRL_LATCH_MASK | + NAU8821_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8821_R09_FLL6, + NAU8821_SDM_EN, NAU8821_SDM_EN); + } + break; + case NAU8821_CLK_FLL_MCLK: + /* Higher FLL reference input frequency can only set lower + * gain error, such as 0000 for input reference from MCLK + * 12.288Mhz. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_MCLK | 0); + break; + case NAU8821_CLK_FLL_BLK: + /* If FLL reference input is from low frequency source, + * higher error gain can apply such as 0xf which has + * the most sensitive gain error correction threshold, + * Therefore, FLL has the most accurate DCO to + * target frequency. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_BLK | + (0xf << NAU8821_GAIN_ERR_SFT)); + break; + case NAU8821_CLK_FLL_FS: + /* If FLL reference input is from low frequency source, + * higher error gain can apply such as 0xf which has + * the most sensitive gain error correction threshold, + * Therefore, FLL has the most accurate DCO to + * target frequency. + */ + regmap_update_bits(regmap, NAU8821_R06_FLL3, + NAU8821_FLL_CLK_SRC_MASK | NAU8821_GAIN_ERR_MASK, + NAU8821_FLL_CLK_SRC_FS | + (0xf << NAU8821_GAIN_ERR_SFT)); + break; + default: + dev_err(nau8821->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + nau8821->clk_id = clk_id; + dev_dbg(nau8821->dev, "Sysclk is %dHz and clock id is %d\n", freq, + nau8821->clk_id); + + return 0; +} + +static int nau8821_set_sysclk(struct snd_soc_component *component, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + return nau8821_configure_sysclk(nau8821, clk_id, freq); +} + +static int nau8821_resume_setup(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8821_configure_sysclk(nau8821, NAU8821_CLK_DIS, 0); + if (nau8821->irq) { + /* Clear all interruption status */ + nau8821_int_status_clear_all(regmap); + + /* Enable both insertion and ejection interruptions, and then + * bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, 0); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_DET_DB_BYPASS, + NAU8821_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8821_R12_INTERRUPT_DIS_CTRL, + NAU8821_IRQ_INSERT_DIS | NAU8821_IRQ_EJECT_DIS, 0); + } + + return 0; +} + +static int nau8821_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + struct regmap *regmap = nau8821->regmap; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + /* Setup codec configuration after resume */ + if (snd_soc_component_get_bias_level(component) == + SND_SOC_BIAS_OFF) + nau8821_resume_setup(nau8821); + break; + + case SND_SOC_BIAS_OFF: + /* HPL/HPR short to ground */ + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_SPKR_DWN1R | NAU8821_SPKR_DWN1L, 0); + if (nau8821->irq) { + /* Reset the configuration of jack type for detection. + * Detach 2kOhm Resistors from MICBIAS to MICGND1/2. + */ + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_JKR2, 0); + /* Turn off all interruptions before system shutdown. + * Keep theinterruption quiet before resume + * setup completes. + */ + regmap_write(regmap, + NAU8821_R12_INTERRUPT_DIS_CTRL, 0xffff); + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN, + NAU8821_IRQ_EJECT_EN | NAU8821_IRQ_INSERT_EN); + } + break; + default: + break; + } + + return 0; +} + +static int __maybe_unused nau8821_suspend(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + if (nau8821->irq) + disable_irq(nau8821->irq); + snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); + /* Power down codec power; don't support button wakeup */ + snd_soc_component_disable_pin(component, "MICBIAS"); + snd_soc_dapm_sync(nau8821->dapm); + regcache_cache_only(nau8821->regmap, true); + regcache_mark_dirty(nau8821->regmap); + + return 0; +} + +static int __maybe_unused nau8821_resume(struct snd_soc_component *component) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(nau8821->regmap, false); + regcache_sync(nau8821->regmap); + if (nau8821->irq) + enable_irq(nau8821->irq); + + return 0; +} + +static const struct snd_soc_component_driver nau8821_component_driver = { + .probe = nau8821_component_probe, + .set_sysclk = nau8821_set_sysclk, + .set_pll = nau8821_set_fll, + .set_bias_level = nau8821_set_bias_level, + .suspend = nau8821_suspend, + .resume = nau8821_resume, + .controls = nau8821_controls, + .num_controls = ARRAY_SIZE(nau8821_controls), + .dapm_widgets = nau8821_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8821_dapm_widgets), + .dapm_routes = nau8821_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8821_dapm_routes), + .suspend_bias_off = 1, + .non_legacy_dai_naming = 1, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, +}; + +/** + * nau8821_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8821_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct nau8821 *nau8821 = snd_soc_component_get_drvdata(component); + int ret; + + nau8821->jack = jack; + /* Initiate jack detection work queue */ + INIT_WORK(&nau8821->jdet_work, nau8821_jdet_work); + ret = devm_request_threaded_irq(nau8821->dev, nau8821->irq, NULL, + nau8821_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8821", nau8821); + if (ret) { + dev_err(nau8821->dev, "Cannot request irq %d (%d)\n", + nau8821->irq, ret); + return ret; + } + + return ret; +} +EXPORT_SYMBOL_GPL(nau8821_enable_jack_detect); + +static void nau8821_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8821_R00_RESET, 0xffff); + regmap_write(regmap, NAU8821_R00_RESET, 0xffff); +} + +static void nau8821_print_device_properties(struct nau8821 *nau8821) +{ + struct device *dev = nau8821->dev; + + dev_dbg(dev, "jkdet-enable: %d\n", nau8821->jkdet_enable); + dev_dbg(dev, "jkdet-pull-enable: %d\n", nau8821->jkdet_pull_enable); + dev_dbg(dev, "jkdet-pull-up: %d\n", nau8821->jkdet_pull_up); + dev_dbg(dev, "jkdet-polarity: %d\n", nau8821->jkdet_polarity); + dev_dbg(dev, "micbias-voltage: %d\n", nau8821->micbias_voltage); + dev_dbg(dev, "vref-impedance: %d\n", nau8821->vref_impedance); + dev_dbg(dev, "jack-insert-debounce: %d\n", + nau8821->jack_insert_debounce); + dev_dbg(dev, "jack-eject-debounce: %d\n", + nau8821->jack_eject_debounce); + dev_dbg(dev, "dmic-clk-threshold: %d\n", + nau8821->dmic_clk_threshold); +} + +static int nau8821_read_device_properties(struct device *dev, + struct nau8821 *nau8821) +{ + int ret; + + nau8821->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8821->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8821->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + ret = device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8821->jkdet_polarity); + if (ret) + nau8821->jkdet_polarity = 1; + ret = device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8821->micbias_voltage); + if (ret) + nau8821->micbias_voltage = 6; + ret = device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8821->vref_impedance); + if (ret) + nau8821->vref_impedance = 2; + ret = device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8821->jack_insert_debounce); + if (ret) + nau8821->jack_insert_debounce = 7; + ret = device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8821->jack_eject_debounce); + if (ret) + nau8821->jack_eject_debounce = 0; + ret = device_property_read_u32(dev, "nuvoton,dmic-clk-threshold", + &nau8821->dmic_clk_threshold); + if (ret) + nau8821->dmic_clk_threshold = 3072000; + + return 0; +} + +static void nau8821_init_regs(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Enable Bias/Vmid */ + regmap_update_bits(regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_VMID, NAU8821_BIAS_VMID); + regmap_update_bits(regmap, NAU8821_R76_BOOST, + NAU8821_GLOBAL_BIAS_EN, NAU8821_GLOBAL_BIAS_EN); + /* VMID Tieoff setting and enable TESTDAC. + * This sets the analog DAC inputs to a '0' input signal to avoid + * any glitches due to power up transients in both the analog and + * digital DAC circuit. + */ + regmap_update_bits(regmap, NAU8821_R66_BIAS_ADJ, + NAU8821_BIAS_VMID_SEL_MASK | NAU8821_BIAS_TESTDAC_EN, + (nau8821->vref_impedance << NAU8821_BIAS_VMID_SEL_SFT) | + NAU8821_BIAS_TESTDAC_EN); + /* Disable short Frame Sync detection logic */ + regmap_update_bits(regmap, NAU8821_R1E_LEFT_TIME_SLOT, + NAU8821_DIS_FS_SHORT_DET, NAU8821_DIS_FS_SHORT_DET); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8821_R76_BOOST, + NAU8821_PRECHARGE_DIS | NAU8821_HP_BOOST_DIS | + NAU8821_HP_BOOST_G_DIS | NAU8821_SHORT_SHUTDOWN_EN, + NAU8821_PRECHARGE_DIS | NAU8821_HP_BOOST_DIS | + NAU8821_HP_BOOST_G_DIS | NAU8821_SHORT_SHUTDOWN_EN); + /* Class G timer 64ms */ + regmap_update_bits(regmap, NAU8821_R4B_CLASSG_CTRL, + NAU8821_CLASSG_TIMER_MASK, + 0x20 << NAU8821_CLASSG_TIMER_SFT); + /* Class AB bias current to 2x, DAC Capacitor enable MSB/LSB */ + regmap_update_bits(regmap, NAU8821_R6A_ANALOG_CONTROL_2, + NAU8821_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8821_DAC_CAPACITOR_MSB | NAU8821_DAC_CAPACITOR_LSB, + NAU8821_HP_NON_CLASSG_CURRENT_2xADJ | + NAU8821_DAC_CAPACITOR_MSB | NAU8821_DAC_CAPACITOR_LSB); + /* Disable DACR/L power */ + regmap_update_bits(regmap, NAU8821_R80_CHARGE_PUMP, + NAU8821_POWER_DOWN_DACR | NAU8821_POWER_DOWN_DACL, 0); + /* DAC clock delay 2ns, VREF */ + regmap_update_bits(regmap, NAU8821_R73_RDAC, + NAU8821_DAC_CLK_DELAY_MASK | NAU8821_DAC_VREF_MASK, + (0x2 << NAU8821_DAC_CLK_DELAY_SFT) | + (0x3 << NAU8821_DAC_VREF_SFT)); + + regmap_update_bits(regmap, NAU8821_R74_MIC_BIAS, + NAU8821_MICBIAS_VOLTAGE_MASK, nau8821->micbias_voltage); + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. + */ + regmap_update_bits(regmap, NAU8821_R2B_ADC_RATE, + NAU8821_ADC_SYNC_DOWN_MASK, NAU8821_ADC_SYNC_DOWN_64); + regmap_update_bits(regmap, NAU8821_R2C_DAC_CTRL1, + NAU8821_DAC_OVERSAMPLE_MASK, NAU8821_DAC_OVERSAMPLE_64); +} + +static int nau8821_setup_irq(struct nau8821 *nau8821) +{ + struct regmap *regmap = nau8821->regmap; + + /* Jack detection */ + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_OUTPUT_EN, + nau8821->jkdet_enable ? 0 : NAU8821_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_PULL_EN, + nau8821->jkdet_pull_enable ? 0 : NAU8821_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8821_R1A_GPIO12_CTRL, + NAU8821_JKDET_PULL_UP, + nau8821->jkdet_pull_up ? NAU8821_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8821->jkdet_polarity ? 0 : NAU8821_JACK_POLARITY); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_INSERT_DEBOUNCE_MASK, + nau8821->jack_insert_debounce << + NAU8821_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8821_R0D_JACK_DET_CTRL, + NAU8821_JACK_EJECT_DEBOUNCE_MASK, + nau8821->jack_eject_debounce << + NAU8821_JACK_EJECT_DEBOUNCE_SFT); + /* Pull up IRQ pin */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, + NAU8821_IRQ_PIN_PULL_UP | NAU8821_IRQ_PIN_PULL_EN | + NAU8821_IRQ_OUTPUT_EN, NAU8821_IRQ_PIN_PULL_UP | + NAU8821_IRQ_PIN_PULL_EN | NAU8821_IRQ_OUTPUT_EN); + /* Disable interruption before codec initiation done */ + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8821_R0F_INTERRUPT_MASK, 0x3f5, 0x3f5); + + return 0; +} + +static int nau8821_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8821 *nau8821 = dev_get_platdata(&i2c->dev); + int ret, value; + + if (!nau8821) { + nau8821 = devm_kzalloc(dev, sizeof(*nau8821), GFP_KERNEL); + if (!nau8821) + return -ENOMEM; + nau8821_read_device_properties(dev, nau8821); + } + i2c_set_clientdata(i2c, nau8821); + + nau8821->regmap = devm_regmap_init_i2c(i2c, &nau8821_regmap_config); + if (IS_ERR(nau8821->regmap)) + return PTR_ERR(nau8821->regmap); + + nau8821->dev = dev; + nau8821->irq = i2c->irq; + nau8821_print_device_properties(nau8821); + + nau8821_reset_chip(nau8821->regmap); + ret = regmap_read(nau8821->regmap, NAU8821_R58_I2C_DEVICE_ID, &value); + if (ret) { + dev_err(dev, "Failed to read device id (%d)\n", ret); + return ret; + } + nau8821_init_regs(nau8821); + + if (i2c->irq) + nau8821_setup_irq(nau8821); + + ret = devm_snd_soc_register_component(&i2c->dev, + &nau8821_component_driver, &nau8821_dai, 1); + + return ret; +} + +static int nau8821_i2c_remove(struct i2c_client *i2c_client) +{ + struct nau8821 *nau8821 = i2c_get_clientdata(i2c_client); + + devm_free_irq(nau8821->dev, nau8821->irq, nau8821); + + return 0; +} + +static const struct i2c_device_id nau8821_i2c_ids[] = { + { "nau8821", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8821_i2c_ids); + +#ifdef CONFIG_OF +static const struct of_device_id nau8821_of_ids[] = { + { .compatible = "nuvoton,nau8821", }, + {} +}; +MODULE_DEVICE_TABLE(of, nau8821_of_ids); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id nau8821_acpi_match[] = { + { "NVTN2020", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, nau8821_acpi_match); +#endif + +static struct i2c_driver nau8821_driver = { + .driver = { + .name = "nau8821", + .of_match_table = of_match_ptr(nau8821_of_ids), + .acpi_match_table = ACPI_PTR(nau8821_acpi_match), + }, + .probe = nau8821_i2c_probe, + .remove = nau8821_i2c_remove, + .id_table = nau8821_i2c_ids, +}; +module_i2c_driver(nau8821_driver); + +MODULE_DESCRIPTION("ASoC nau8821 driver"); +MODULE_AUTHOR("John Hsu <kchsu0(a)nuvoton.com>"); +MODULE_AUTHOR("Seven Lee <wtli(a)nuvoton.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8821.h b/sound/soc/codecs/nau8821.h new file mode 100644 index 000000000000..a92edfeb9d3a --- /dev/null +++ b/sound/soc/codecs/nau8821.h @@ -0,0 +1,533 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * NAU88L21 ALSA SoC audio driver + * + * Copyright 2021 Nuvoton Technology Corp. + * Author: John Hsu <kchsu0(a)nuvoton.com> + * Co-author: Seven Lee <wtli(a)nuvoton.com> + */ + +#ifndef __NAU8821_H__ +#define __NAU8821_H__ + +#define NAU8821_R00_RESET 0x00 +#define NAU8821_R01_ENA_CTRL 0x01 +#define NAU8821_R03_CLK_DIVIDER 0x03 +#define NAU8821_R04_FLL1 0x04 +#define NAU8821_R05_FLL2 0x05 +#define NAU8821_R06_FLL3 0x06 +#define NAU8821_R07_FLL4 0x07 +#define NAU8821_R08_FLL5 0x08 +#define NAU8821_R09_FLL6 0x09 +#define NAU8821_R0A_FLL7 0x0a +#define NAU8821_R0B_FLL8 0x0b +#define NAU8821_R0D_JACK_DET_CTRL 0x0d +#define NAU8821_R0F_INTERRUPT_MASK 0x0f +#define NAU8821_R10_IRQ_STATUS 0x10 +#define NAU8821_R11_INT_CLR_KEY_STATUS 0x11 +#define NAU8821_R12_INTERRUPT_DIS_CTRL 0x12 +#define NAU8821_R13_DMIC_CTRL 0x13 +#define NAU8821_R1A_GPIO12_CTRL 0x1a +#define NAU8821_R1B_TDM_CTRL 0x1b +#define NAU8821_R1C_I2S_PCM_CTRL1 0x1c +#define NAU8821_R1D_I2S_PCM_CTRL2 0x1d +#define NAU8821_R1E_LEFT_TIME_SLOT 0x1e +#define NAU8821_R1F_RIGHT_TIME_SLOT 0x1f +#define NAU8821_R21_BIQ0_COF1 0x21 +#define NAU8821_R22_BIQ0_COF2 0x22 +#define NAU8821_R23_BIQ0_COF3 0x23 +#define NAU8821_R24_BIQ0_COF4 0x24 +#define NAU8821_R25_BIQ0_COF5 0x25 +#define NAU8821_R26_BIQ0_COF6 0x26 +#define NAU8821_R27_BIQ0_COF7 0x27 +#define NAU8821_R28_BIQ0_COF8 0x28 +#define NAU8821_R29_BIQ0_COF9 0x29 +#define NAU8821_R2A_BIQ0_COF10 0x2a +#define NAU8821_R2B_ADC_RATE 0x2b +#define NAU8821_R2C_DAC_CTRL1 0x2c +#define NAU8821_R2D_DAC_CTRL2 0x2d +#define NAU8821_R2F_DAC_DGAIN_CTRL 0x2f +#define NAU8821_R30_ADC_DGAIN_CTRL 0x30 +#define NAU8821_R31_MUTE_CTRL 0x31 +#define NAU8821_R32_HSVOL_CTRL 0x32 +#define NAU8821_R34_DACR_CTRL 0x34 +#define NAU8821_R35_ADC_DGAIN_CTRL1 0x35 +#define NAU8821_R36_ADC_DRC_KNEE_IP12 0x36 +#define NAU8821_R37_ADC_DRC_KNEE_IP34 0x37 +#define NAU8821_R38_ADC_DRC_SLOPES 0x38 +#define NAU8821_R39_ADC_DRC_ATKDCY 0x39 +#define NAU8821_R3A_DAC_DRC_KNEE_IP12 0x3a +#define NAU8821_R3B_DAC_DRC_KNEE_IP34 0x3b +#define NAU8821_R3C_DAC_DRC_SLOPES 0x3c +#define NAU8821_R3D_DAC_DRC_ATKDCY 0x3d +#define NAU8821_R41_BIQ1_COF1 0x41 +#define NAU8821_R42_BIQ1_COF2 0x42 +#define NAU8821_R43_BIQ1_COF3 0x43 +#define NAU8821_R44_BIQ1_COF4 0x44 +#define NAU8821_R45_BIQ1_COF5 0x45 +#define NAU8821_R46_BIQ1_COF6 0x46 +#define NAU8821_R47_BIQ1_COF7 0x47 +#define NAU8821_R48_BIQ1_COF8 0x48 +#define NAU8821_R49_BIQ1_COF9 0x49 +#define NAU8821_R4A_BIQ1_COF10 0x4a +#define NAU8821_R4B_CLASSG_CTRL 0x4b +#define NAU8821_R4C_IMM_MODE_CTRL 0x4c +#define NAU8821_R4D_IMM_RMS_L 0x4d +#define NAU8821_R4E_FUSE_CTRL2 0x4e +#define NAU8821_R4F_FUSE_CTRL3 0x4f +#define NAU8821_R51_FUSE_CTRL1 0x51 +#define NAU8821_R53_OTPDOUT_1 0x53 +#define NAU8821_R54_OTPDOUT_2 0x54 +#define NAU8821_R55_MISC_CTRL 0x55 +#define NAU8821_R58_I2C_DEVICE_ID 0x58 +#define NAU8821_R59_SARDOUT_RAM_STATUS 0x59 +#define NAU8821_R5A_SOFTWARE_RST 0x5a +#define NAU8821_R66_BIAS_ADJ 0x66 +#define NAU8821_R68_TRIM_SETTINGS 0x68 +#define NAU8821_R69_ANALOG_CONTROL_1 0x69 +#define NAU8821_R6A_ANALOG_CONTROL_2 0x6a +#define NAU8821_R6B_PGA_MUTE 0x6b +#define NAU8821_R71_ANALOG_ADC_1 0x71 +#define NAU8821_R72_ANALOG_ADC_2 0x72 +#define NAU8821_R73_RDAC 0x73 +#define NAU8821_R74_MIC_BIAS 0x74 +#define NAU8821_R76_BOOST 0x76 +#define NAU8821_R77_FEPGA 0x77 +#define NAU8821_R7E_PGA_GAIN 0x7e +#define NAU8821_R7F_POWER_UP_CONTROL 0x7f +#define NAU8821_R80_CHARGE_PUMP 0x80 +#define NAU8821_R81_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8821_R82_GENERAL_STATUS 0x82 +#define NAU8821_REG_MAX NAU8821_R82_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8821_REG_ADDR_LEN 16 +#define NAU8821_REG_DATA_LEN 16 + +/* ENA_CTRL (0x01) */ +#define NAU8821_CLK_DAC_INV_SFT 14 +#define NAU8821_CLK_DAC_INV (0x1 << NAU8821_CLK_DAC_INV) +#define NAU8821_EN_DACR_SFT 11 +#define NAU8821_EN_DACR (0x1 << NAU8821_EN_DACR_SFT) +#define NAU8821_EN_DACL_SFT 10 +#define NAU8821_EN_DACL (0x1 << NAU8821_EN_DACL_SFT) +#define NAU8821_EN_ADCR_SFT 9 +#define NAU8821_EN_ADCR (0x1 << NAU8821_EN_ADCR_SFT) +#define NAU8821_EN_ADCL_SFT 8 +#define NAU8821_EN_ADCL (0x1 << NAU8821_EN_ADCL_SFT) +#define NAU8821_EN_ADC_CLK_SFT 7 +#define NAU8821_EN_ADC_CLK (0x1 << NAU8821_EN_ADC_CLK_SFT) +#define NAU8821_EN_DAC_CLK_SFT 6 +#define NAU8821_EN_DAC_CLK (0x1 << NAU8821_EN_DAC_CLK_SFT) +#define NAU8821_EN_I2S_CLK_SFT 4 +#define NAU8821_EN_I2S_CLK (0x1 << NAU8821_EN_I2S_CLK_SFT) +#define NAU8821_EN_DRC_CLK_SFT 0 +#define NAU8821_EN_DRC_CLK (0x1 << NAU8821_EN_DRC_CLK_SFT) + +/* CLK_DIVIDER (0x03) */ +#define NAU8821_CLK_SRC_SFT 15 +#define NAU8821_CLK_SRC_MASK (0x1 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_SRC_VCO (0x1 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_SRC_MCLK (0x0 << NAU8821_CLK_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_SFT 13 +#define NAU8821_CLK_CODEC_SRC_MASK (0x1 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_VCO (0x1 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_CODEC_SRC_MCLK (0x0 << NAU8821_CLK_CODEC_SRC_SFT) +#define NAU8821_CLK_ADC_SRC_SFT 6 +#define NAU8821_CLK_ADC_SRC_MASK (0x3 << NAU8821_CLK_ADC_SRC_SFT) +#define NAU8821_CLK_DAC_SRC_SFT 4 +#define NAU8821_CLK_DAC_SRC_MASK (0x3 << NAU8821_CLK_DAC_SRC_SFT) +#define NAU8821_CLK_MCLK_SRC_MASK 0xf + +/* FLL1 (0x04) */ +#define NAU8821_ICTRL_LATCH_SFT 10 +#define NAU8821_ICTRL_LATCH_MASK (0x7 << NAU8821_ICTRL_LATCH_SFT) +#define NAU8821_FLL_RATIO_MASK 0x7f + +/* FLL3 (0x06) */ +#define NAU8821_GAIN_ERR_SFT 12 +#define NAU8821_GAIN_ERR_MASK (0xf << NAU8821_GAIN_ERR_SFT) +#define NAU8821_FLL_CLK_SRC_SFT 10 +#define NAU8821_FLL_CLK_SRC_MASK (0x3 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_FS (0x3 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_BLK (0x2 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_CLK_SRC_MCLK (0x0 << NAU8821_FLL_CLK_SRC_SFT) +#define NAU8821_FLL_INTEGER_MASK 0x3ff + +/* FLL4 (0x07) */ +#define NAU8821_HIGHBW_EN_SFT 15 +#define NAU8821_HIGHBW_EN (0x1 << NAU8821_HIGHBW_EN_SFT) +#define NAU8821_FLL_REF_DIV_SFT 10 +#define NAU8821_FLL_REF_DIV_MASK (0x3 << NAU8821_FLL_REF_DIV_SFT) + +/* FLL5 (0x08) */ +#define NAU8821_FLL_PDB_DAC_EN (0x1 << 15) +#define NAU8821_FLL_LOOP_FTR_EN (0x1 << 14) +#define NAU8821_FLL_CLK_SW_SFT 13 +#define NAU8821_FLL_CLK_SW_MASK (0x1 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_CLK_SW_N2 (0x1 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_CLK_SW_REF (0x0 << NAU8821_FLL_CLK_SW_SFT) +#define NAU8821_FLL_FTR_SW_SFT 12 +#define NAU8821_FLL_FTR_SW_MASK (0x1 << NAU8821_FLL_FTR_SW_SFT) +#define NAU8821_FLL_FTR_SW_ACCU (0x1 << NAU8821_FLL_FTR_SW_SFT) +#define NAU8821_FLL_FTR_SW_FILTER (0x0 << NAU8821_FLL_FTR_SW_SFT) + +/* FLL6 (0x09) */ +#define NAU8821_DCO_EN (0x1 << 15) +#define NAU8821_SDM_EN (0x1 << 14) +#define NAU8821_CUTOFF500 (0x1 << 13) + +/* FLL7 (0x0a) */ +#define NAU8821_FLL_FRACH_MASK 0xff + +/* FLL8 (0x0b) */ +#define NAU8821_FLL_FRACL_MASK 0xffff + +/* JACK_DET_CTRL (0x0d) */ +/* 0 - open, 1 - short to GND */ +#define NAU8821_SPKR_DWN1R_SFT 15 +#define NAU8821_SPKR_DWN1R (0x1 << NAU8821_SPKR_DWN1R_SFT) +#define NAU8821_SPKR_DWN1L_SFT 14 +#define NAU8821_SPKR_DWN1L (0x1 << NAU8821_SPKR_DWN1L_SFT) +#define NAU8821_JACK_DET_RESTART (0x1 << 9) +#define NAU8821_JACK_DET_DB_BYPASS (0x1 << 8) +#define NAU8821_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8821_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8821_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8821_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8821_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8821_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8821_JACK_POLARITY (0x1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0x0f) */ +#define NAU8821_IRQ_PIN_PULL_UP (0x1 << 14) +#define NAU8821_IRQ_PIN_PULL_EN (0x1 << 13) +#define NAU8821_IRQ_OUTPUT_EN (0x1 << 11) +#define NAU8821_IRQ_RMS_EN (0x1 << 8) +#define NAU8821_IRQ_KEY_RELEASE_EN (0x1 << 7) +#define NAU8821_IRQ_KEY_PRESS_EN (0x1 << 6) +#define NAU8821_IRQ_MIC_DET_EN (0x1 << 4) +#define NAU8821_IRQ_EJECT_EN (0x1 << 2) +#define NAU8821_IRQ_INSERT_EN 0x1 + +/* IRQ_STATUS (0x10) */ +#define NAU8821_SHORT_CIRCUIT_IRQ (0x1 << 9) +#define NAU8821_IMPEDANCE_MEAS_IRQ (0x1 << 8) +#define NAU8821_KEY_IRQ_SFT 6 +#define NAU8821_KEY_IRQ_MASK (0x3 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_KEY_RELEASE_IRQ (0x2 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_KEY_SHORT_PRESS_IRQ (0x1 << NAU8821_KEY_IRQ_SFT) +#define NAU8821_MIC_DETECT_IRQ (0x1 << 4) +#define NAU8821_JACK_EJECT_IRQ_MASK (0x3 << 2) +#define NAU8821_JACK_EJECT_DETECTED (0x1 << 2) +#define NAU8821_JACK_INSERT_IRQ_MASK 0x3 +#define NAU8821_JACK_INSERT_DETECTED 0x1 + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8821_IRQ_KEY_RELEASE_DIS (0x1 << 7) +#define NAU8821_IRQ_KEY_PRESS_DIS (0x1 << 6) +#define NAU8821_IRQ_MIC_DIS (0x1 << 4) +#define NAU8821_IRQ_EJECT_DIS (0x1 << 2) +#define NAU8821_IRQ_INSERT_DIS 0x1 + +/* DMIC_CTRL (0x13) */ +#define NAU8821_DMIC_DS_SFT 7 +#define NAU8821_DMIC_DS_MASK (0x1 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_DS_HIGH (0x1 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_DS_LOW (0x0 << NAU8821_DMIC_DS_SFT) +#define NAU8821_DMIC_SRC_SFT 1 +#define NAU8821_DMIC_SRC_MASK (0x3 << NAU8821_DMIC_SRC_SFT) +#define NAU8821_CLK_DMIC_SRC (0x2 << NAU8821_DMIC_SRC_SFT) +#define NAU8821_DMIC_EN_SFT 0 + +/* GPIO12_CTRL (0x1a) */ +#define NAU8821_JKDET_PULL_UP (0x1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8821_JKDET_PULL_EN (0x1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8821_JKDET_OUTPUT_EN (0x1 << 8) /* 0 - enable input, 1 - enable output */ + +/* TDM_CTRL (0x1b) */ +#define NAU8821_TDM_EN_SFT 15 +#define NAU8821_TDM_EN (0x1 << NAU8821_TDM_EN_SFT) +#define NAU8821_ADCPHS_SFT 13 +#define NAU8821_DACL_CH_SFT 7 +#define NAU8821_DACL_CH_MASK (0x7 << NAU8821_DACL_CH_SFT) +#define NAU8821_DACR_CH_SFT 4 +#define NAU8821_DACR_CH_MASK (0x7 << NAU8821_DACR_CH_SFT) +#define NAU8821_ADCL_CH_SFT 2 +#define NAU8821_ADCL_CH_MASK (0x3 << NAU8821_ADCL_CH_SFT) +#define NAU8821_ADCR_CH_SFT 0 +#define NAU8821_ADCR_CH_MASK 0x3 + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8821_I2S_BP_SFT 7 +#define NAU8821_I2S_BP_MASK (0x1 << NAU8821_I2S_BP_SFT) +#define NAU8821_I2S_BP_INV (0x1 << NAU8821_I2S_BP_SFT) +#define NAU8821_I2S_PCMB_SFT 6 +#define NAU8821_I2S_PCMB_MASK (0x1 << NAU8821_I2S_PCMB_SFT) +#define NAU8821_I2S_PCMB_EN (0x1 << NAU8821_I2S_PCMB_SFT) +#define NAU8821_I2S_DL_SFT 2 +#define NAU8821_I2S_DL_MASK (0x3 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_32 (0x3 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_24 (0x2 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_20 (0x1 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DL_16 (0x0 << NAU8821_I2S_DL_SFT) +#define NAU8821_I2S_DF_MASK 0x3 +#define NAU8821_I2S_DF_PCM_AB 0x3 +#define NAU8821_I2S_DF_I2S 0x2 +#define NAU8821_I2S_DF_LEFT 0x1 +#define NAU8821_I2S_DF_RIGTH 0x0 + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8821_I2S_TRISTATE_SFT 15 +#define NAU8821_I2S_TRISTATE (0x1 << NAU8821_I2S_TRISTATE_SFT) +#define NAU8821_I2S_LRC_DIV_SFT 12 +#define NAU8821_I2S_LRC_DIV_MASK (0x3 << NAU8821_I2S_LRC_DIV_SFT) +#define NAU8821_I2S_MS_SFT 3 +#define NAU8821_I2S_MS_MASK (0x1 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_MS_MASTER (0x1 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_MS_SLAVE (0x0 << NAU8821_I2S_MS_SFT) +#define NAU8821_I2S_BLK_DIV_MASK 0x7 + +/* LEFT_TIME_SLOT (0x1e) */ +#define NAU8821_TSLOT_L_OFFSET_MASK 0x3ff +#define NAU8821_DIS_FS_SHORT_DET (0x1 << 13) + +/* RIGHT_TIME_SLOT (0x1f) */ +#define NAU8821_TSLOT_R_OFFSET_MASK 0x3ff + +/* BIQ0_COF10 (0x2a) */ +#define NAU8821_BIQ0_ADC_EN_SFT 3 +#define NAU8821_BIQ0_ADC_EN_EN (0x1 << NAU8821_BIQ0_ADC_EN_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8821_ADC_SYNC_DOWN_SFT 0 +#define NAU8821_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8821_ADC_SYNC_DOWN_256 0x3 +#define NAU8821_ADC_SYNC_DOWN_128 0x2 +#define NAU8821_ADC_SYNC_DOWN_64 0x1 +#define NAU8821_ADC_SYNC_DOWN_32 0x0 +#define NAU8821_ADC_L_SRC_SFT 15 +#define NAU8821_ADC_L_SRC_EN (0x1 << NAU8821_ADC_L_SRC_SFT) +#define NAU8821_ADC_R_SRC_SFT 14 +#define NAU8821_ADC_R_SRC_EN (0x1 << NAU8821_ADC_R_SRC_SFT) + +/* DAC_CTRL1 (0x2c) */ +#define NAU8821_DAC_OVERSAMPLE_SFT 0 +#define NAU8821_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8821_DAC_OVERSAMPLE_32 0x4 +#define NAU8821_DAC_OVERSAMPLE_128 0x2 +#define NAU8821_DAC_OVERSAMPLE_256 0x1 +#define NAU8821_DAC_OVERSAMPLE_64 0x0 + +/* DAC_DGAIN_CTRL (0x2f) */ +#define NAU8821_DAC1_TO_DAC0_ST_SFT 8 +#define NAU8821_DAC1_TO_DAC0_ST_MASK (0xff << NAU8821_DAC1_TO_DAC0_ST_SFT) +#define NAU8821_DAC0_TO_DAC1_ST_SFT 0 +#define NAU8821_DAC0_TO_DAC1_ST_MASK 0xff + +/* MUTE_CTRL (0x31) */ +#define NAU8821_DAC_ZC_EN (0x1 << 12) +#define NAU8821_DAC_SOFT_MUTE (0x1 << 9) +#define NAU8821_ADC_ZC_EN (0x1 << 2) +#define NAU8821_ADC_SOFT_MUTE (0x1 << 1) + +/* HSVOL_CTRL (0x32) */ +#define NAU8821_HP_MUTE (0x1 << 15) +#define NAU8821_HP_MUTE_AUTO (0x1 << 14) +#define NAU8821_HPL_MUTE (0x1 << 13) +#define NAU8821_HPR_MUTE (0x1 << 12) +#define NAU8821_HPL_VOL_SFT 4 +#define NAU8821_HPL_VOL_MASK (0x3 << NAU8821_HPL_VOL_SFT) +#define NAU8821_HPR_VOL_SFT 0 +#define NAU8821_HPR_VOL_MASK (0x3 << NAU8821_HPR_VOL_SFT) + +/* DACR_CTRL (0x34) */ +#define NAU8821_DACR_CH_VOL_SFT 8 +#define NAU8821_DACR_CH_VOL_MASK (0xff << NAU8821_DACR_CH_VOL_SFT) +#define NAU8821_DACL_CH_VOL_SFT 0 +#define NAU8821_DACL_CH_VOL_MASK 0xff + +/* ADC_DGAIN_CTRL1 (0x35) */ +#define NAU8821_ADCR_CH_VOL_SFT 8 +#define NAU8821_ADCR_CH_VOL_MASK (0xff << NAU8821_ADCR_CH_VOL_SFT) +#define NAU8821_ADCL_CH_VOL_SFT 0 +#define NAU8821_ADCL_CH_VOL_MASK 0xff + +/* BIQ1_COF10 (0x4a) */ +#define NAU8821_BIQ1_DAC_EN_SFT 3 +#define NAU8821_BIQ1_DAC_EN_EN (0x1 << NAU8821_BIQ1_DAC_EN_SFT) + +/* CLASSG_CTRL (0x4b) */ +#define NAU8821_CLASSG_TIMER_SFT 8 +#define NAU8821_CLASSG_TIMER_MASK (0x3f << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_64MS (0x20 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_32MS (0x10 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_16MS (0x8 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_8MS (0x4 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_2MS (0x2 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_TIMER_1MS (0x1 << NAU8821_CLASSG_TIMER_SFT) +#define NAU8821_CLASSG_RDAC_EN_SFT 2 +#define NAU8821_CLASSG_RDAC_EN (0x1 << NAU8821_CLASSG_RDAC_EN_SFT) +#define NAU8821_CLASSG_LDAC_EN_SFT 1 +#define NAU8821_CLASSG_LDAC_EN (0x1 << NAU8821_CLASSG_LDAC_EN_SFT) +#define NAU8821_CLASSG_EN_SFT 0 +#define NAU8821_CLASSG_EN 0x1 + +/* IMM_MODE_CTRL (0x4c) */ +#define NAU8821_IMM_THD_SFT 8 +#define NAU8821_IMM_THD_MASK (0x3f << NAU8821_IMM_THD_SFT) +#define NAU8821_IMM_GEN_VOL_SFT 6 +#define NAU8821_IMM_GEN_VOL_MASK (0x3 << NAU8821_IMM_GEN_VOL_SFT) +#define NAU8821_IMM_CYC_SFT 4 +#define NAU8821_IMM_CYC_MASK (0x3 << NAU8821_IMM_CYC_SFT) +#define NAU8821_IMM_EN (0x1 << 3) +#define NAU8821_IMM_DAC_SRC_MASK 0x3 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8821_KEYDET (0x1 << 7) +#define NAU8821_MICDET (0x1 << 6) +#define NAU8821_SOFTWARE_ID_MASK 0x3 + +/* BIAS_ADJ (0x66) */ +#define NAU8821_BIAS_HP_IMP (0x1 << 15) +#define NAU8821_BIAS_TESTDAC_SFT 8 +#define NAU8821_BIAS_TESTDAC_EN (0x3 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_TESTDACR_EN (0x2 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_TESTDACL_EN (0x1 << NAU8821_BIAS_TESTDAC_SFT) +#define NAU8821_BIAS_VMID (0x1 << 6) +#define NAU8821_BIAS_VMID_SEL_SFT 4 +#define NAU8821_BIAS_VMID_SEL_MASK (0x3 << NAU8821_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_1 (0x69) */ +#define NAU8821_JD_POL_SFT 2 +#define NAU8821_JD_POL_MASK (0x1 << NAU8821_JD_POL_SFT) +#define NAU8821_JD_POL_INV (0x1 << NAU8821_JD_POL_SFT) +#define NAU8821_JD_OUT_POL_SFT 1 +#define NAU8821_JD_OUT_POL_MASK (0x1 << NAU8821_JD_OUT_POL_SFT) +#define NAU8821_JD_OUT_POL_INV (0x1 << NAU8821_JD_OUT_POL_SFT) +#define NAU8821_JD_EN_SFT 0 +#define NAU8821_JD_EN 0x1 + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8821_HP_NON_CLASSG_CURRENT_2xADJ (0x1 << 12) +#define NAU8821_DAC_CAPACITOR_MSB (0x1 << 1) +#define NAU8821_DAC_CAPACITOR_LSB 0x1 + +/* ANALOG_ADC_1 (0x71) */ +#define NAU8821_MICDET_EN_SFT 0 +#define NAU8821_MICDET_MASK 0x1 +#define NAU8821_MICDET_DIS 0x1 +#define NAU8821_MICDET_EN 0x0 + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8821_ADC_VREFSEL_SFT 8 +#define NAU8821_ADC_VREFSEL_MASK (0x3 << NAU8821_ADC_VREFSEL_SFT) +#define NAU8821_POWERUP_ADCL_SFT 6 +#define NAU8821_POWERUP_ADCL (0x1 << NAU8821_POWERUP_ADCL_SFT) +#define NAU8821_POWERUP_ADCR_SFT 4 +#define NAU8821_POWERUP_ADCR (0x1 << NAU8821_POWERUP_ADCR_SFT) + +/* RDAC (0x73) */ +#define NAU8821_DACR_EN_SFT 13 +#define NAU8821_DACR_EN (0x3 << NAU8821_DACR_EN_SFT) +#define NAU8821_DACL_EN_SFT 12 +#define NAU8821_DACL_EN (0x3 << NAU8821_DACL_EN_SFT) +#define NAU8821_DACR_CLK_EN_SFT 9 +#define NAU8821_DACR_CLK_EN (0x3 << NAU8821_DACR_CLK_EN_SFT) +#define NAU8821_DACL_CLK_EN_SFT 8 +#define NAU8821_DACL_CLK_EN (0x3 << NAU8821_DACL_CLK_EN_SFT) +#define NAU8821_DAC_CLK_DELAY_SFT 4 +#define NAU8821_DAC_CLK_DELAY_MASK (0x7 << NAU8821_DAC_CLK_DELAY_SFT) +#define NAU8821_DAC_VREF_SFT 2 +#define NAU8821_DAC_VREF_MASK (0x3 << NAU8821_DAC_VREF_SFT) + +/* MIC_BIAS (0x74) */ +#define NAU8821_MICBIAS_JKR2 (0x1 << 12) +#define NAU8821_MICBIAS_POWERUP_SFT 8 +#define NAU8821_MICBIAS_VOLTAGE_SFT 0 +#define NAU8821_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8821_PRECHARGE_DIS (0x1 << 13) +#define NAU8821_GLOBAL_BIAS_EN (0x1 << 12) +#define NAU8821_HP_BOOST_DIS_SFT 9 +#define NAU8821_HP_BOOST_DIS (0x1 << NAU8821_HP_BOOST_DIS_SFT) +#define NAU8821_HP_BOOST_G_DIS (0x1 << 8) +#define NAU8821_SHORT_SHUTDOWN_EN (0x1 << 6) + +/* FEPGA (0x77) */ +#define NAU8821_FEPGA_MODEL_SFT 4 +#define NAU8821_FEPGA_MODEL_MASK (0xf << NAU8821_FEPGA_MODEL_SFT) +#define NAU8821_FEPGA_MODER_SFT 0 +#define NAU8821_FEPGA_MODER_MASK 0xf + +/* PGA_GAIN (0x7e) */ +#define NAU8821_PGA_GAIN_L_SFT 8 +#define NAU8821_PGA_GAIN_L_MASK (0x3f << NAU8821_PGA_GAIN_L_SFT) +#define NAU8821_PGA_GAIN_R_SFT 0 +#define NAU8821_PGA_GAIN_R_MASK 0x3f + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8821_PUP_PGA_L_SFT 15 +#define NAU8821_PUP_PGA_L (0x1 << NAU8821_PUP_PGA_L_SFT) +#define NAU8821_PUP_PGA_R_SFT 14 +#define NAU8821_PUP_PGA_R (0x1 << NAU8821_PUP_PGA_R_SFT) +#define NAU8821_PUP_INTEG_R_SFT 5 +#define NAU8821_PUP_INTEG_R (0x1 << NAU8821_PUP_INTEG_R_SFT) +#define NAU8821_PUP_INTEG_L_SFT 4 +#define NAU8821_PUP_INTEG_L (0x1 << NAU8821_PUP_INTEG_L_SFT) +#define NAU8821_PUP_DRV_INSTG_R_SFT 3 +#define NAU8821_PUP_DRV_INSTG_R (0x1 << NAU8821_PUP_DRV_INSTG_R_SFT) +#define NAU8821_PUP_DRV_INSTG_L_SFT 2 +#define NAU8821_PUP_DRV_INSTG_L (0x1 << NAU8821_PUP_DRV_INSTG_L_SFT) +#define NAU8821_PUP_MAIN_DRV_R_SFT 1 +#define NAU8821_PUP_MAIN_DRV_R (0x1 << NAU8821_PUP_MAIN_DRV_R_SFT) +#define NAU8821_PUP_MAIN_DRV_L_SFT 0 +#define NAU8821_PUP_MAIN_DRV_L 0x1 + +/* CHARGE_PUMP (0x80) */ +#define NAU8821_JAMNODCLOW (0x1 << 10) +#define NAU8821_POWER_DOWN_DACR_SFT 9 +#define NAU8821_POWER_DOWN_DACR (0x1 << NAU8821_POWER_DOWN_DACR_SFT) +#define NAU8821_POWER_DOWN_DACL_SFT 8 +#define NAU8821_POWER_DOWN_DACL (0x1 << NAU8821_POWER_DOWN_DACL_SFT) +#define NAU8821_CHANRGE_PUMP_EN_SFT 5 +#define NAU8821_CHANRGE_PUMP_EN (0x1 << NAU8821_CHANRGE_PUMP_EN_SFT) + +/* GENERAL_STATUS (0x82) */ +#define NAU8821_GPIO2_IN_SFT 1 +#define NAU8821_GPIO2_IN (0x1 << NAU8821_GPIO2_IN_SFT) + +#define NUVOTON_CODEC_DAI "nau8821-hifi" + +/* System Clock Source */ +enum { + NAU8821_CLK_DIS, + NAU8821_CLK_MCLK, + NAU8821_CLK_INTERNAL, + NAU8821_CLK_FLL_MCLK, + NAU8821_CLK_FLL_BLK, + NAU8821_CLK_FLL_FS, +}; + +struct nau8821 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + struct snd_soc_jack *jack; + struct work_struct jdet_work; + int irq; + int clk_id; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int jack_insert_debounce; + int jack_eject_debounce; + int fs; + int dmic_clk_threshold; +}; + +int nau8821_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack); + +#endif /* __NAU8821_H__ */ -- 2.25.1

2 1

Scripts to save and reload ALSA settings on shutdown/reboot doesn't work for Baytrail atom
by GitHub issues - opened 02 Oct '21

02 Oct '21

alsa-project/alsa-utils issue #113 was opened from Pic-master: Issue persists across various distributions. "best" workaround I've found is to save a copy of asound.state to user home directory, and run alsactl --file /home/asound.state restore at boot. Janky work around that only works for one user. Chased the issue as far as I could. Running sudo alsactl restore also works, but must be done after each reboot (also had to manually "sudo alsactl store" first) Currently on GalliumOS 3.0 Acer CB-131 N15Q10 Issue URL : https://github.com/alsa-project/alsa-utils/issues/113 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: fsl_spdif: implement bypass mode from in to out
by Shengjiu Wang 02 Oct '21

02 Oct '21

From: Viorel Suman <viorel.suman(a)nxp.com> Implement SPDIF bypass mode. It implies internal SoC routing of SPDIF input signal to SPDIF output signal. The test bed requires two boards: B1 configured in bypass mode, and B2 to feed B1 SPDIF RX port and read B1 SPDIF TX port: B2 TX -> B1 RX, B2 RX <- B1 TX. The test procedure: a) Boot both boards b) B2: start "arecord <spdifcard> -r 48kHz | aplay <local DAC>" c) B2: start "aplay <spdifcard> -r 48kHz <2ch 48kHz audio file>" d) B1: enable bypass mode: amixer -cimxspdif cset numid=8,iface=PCM,name='Bypass Mode' on e) B2: check DAC audio, make sure the same sample rate is used at steps b) and c), in example above the rate is 48kHz. f) B1: try to run "aplay" or "arecord" on imxspdif card while in bypass mode - both must fail until bypass mode is disabled g) B1: disable bypass mode: amixer -cimxspdif cset numid=8,iface=PCM,name='Bypass Mode' off h) B1: check the usual playback and capture on imxspdif card. During this test try to set bypass mode - must not be allowed while playback or capture is running. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 74 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 74 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 928b59069283..d178b479c8bd 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -111,6 +111,7 @@ struct spdif_mixer_control { * @dma_params_tx: DMA parameters for transmit channel * @dma_params_rx: DMA parameters for receive channel * @regcache_srpc: regcache for SRPC + * @bypass: status of bypass input to output */ struct fsl_spdif_priv { const struct fsl_spdif_soc_data *soc; @@ -133,6 +134,7 @@ struct fsl_spdif_priv { struct snd_dmaengine_dai_dma_data dma_params_rx; /* regcache for SRPC */ u32 regcache_srpc; + bool bypass; }; static struct fsl_spdif_soc_data fsl_spdif_vf610 = { @@ -905,6 +907,69 @@ static int fsl_spdif_rx_rcm_put(struct snd_kcontrol *kcontrol, return 0; } +static int fsl_spdif_bypass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *priv = snd_soc_dai_get_drvdata(dai); + + ucontrol->value.integer.value[0] = priv->bypass ? 1 : 0; + + return 0; +} + +static int fsl_spdif_bypass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol); + struct fsl_spdif_priv *priv = snd_soc_dai_get_drvdata(dai); + struct snd_soc_card *card = dai->component->card; + bool set = (ucontrol->value.integer.value[0] != 0); + struct regmap *regmap = priv->regmap; + struct snd_soc_pcm_runtime *rtd; + u32 scr, mask; + int stream; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link); + + if (priv->bypass == set) + return 0; /* nothing to do */ + + if (snd_soc_dai_active(dai)) { + dev_err(dai->dev, "Cannot change BYPASS mode while stream is running.\n"); + return -EBUSY; + } + + pm_runtime_get_sync(dai->dev); + + if (set) { + /* Disable interrupts */ + regmap_update_bits(regmap, REG_SPDIF_SIE, 0xffffff, 0); + + /* Configure BYPASS mode */ + scr = SCR_TXSEL_RX | SCR_RXFIFO_OFF; + mask = SCR_RXFIFO_FSEL_MASK | SCR_RXFIFO_AUTOSYNC_MASK | + SCR_RXFIFO_CTL_MASK | SCR_RXFIFO_OFF_MASK | SCR_TXSEL_MASK; + /* Power up SPDIF module */ + mask |= SCR_LOW_POWER; + } else { + /* Power down SPDIF module, disable TX */ + scr = SCR_LOW_POWER | SCR_TXSEL_OFF; + mask = SCR_LOW_POWER | SCR_TXSEL_MASK; + } + + regmap_update_bits(regmap, REG_SPDIF_SCR, mask, scr); + + /* Disable playback & capture if BYPASS mode is enabled, enable otherwise */ + for_each_pcm_streams(stream) + rtd->pcm->streams[stream].substream_count = (set ? 0 : 1); + + priv->bypass = set; + pm_runtime_put_sync(dai->dev); + + return 0; +} + /* DPLL lock information */ static int fsl_spdif_rxrate_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1075,6 +1140,15 @@ static struct snd_kcontrol_new fsl_spdif_ctrls[] = { .info = fsl_spdif_rxrate_info, .get = fsl_spdif_rxrate_get, }, + /* RX bypass controller */ + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "Bypass Mode", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = fsl_spdif_bypass_get, + .put = fsl_spdif_bypass_put, + }, /* User bit sync mode set/get controller */ { .iface = SNDRV_CTL_ELEM_IFACE_PCM, -- 2.17.1

2 1

[PATCH] ASoC: soc-component: Remove conditional definition of debugfs data members
by Simon Trimmer 02 Oct '21

02 Oct '21

This simplification allows the use of the standard kernel pattern of static inline dummy functions for debugfs code. Most systems will only have a small number of snd_soc_components so the memory impact is minimal. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Suggested-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc-component.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index e09a2d108e8c..3a35d149e92f 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -227,10 +227,8 @@ struct snd_soc_component { struct snd_compr_stream *mark_compr_open; void *mark_pm; -#ifdef CONFIG_DEBUG_FS struct dentry *debugfs_root; const char *debugfs_prefix; -#endif }; #define for_each_component_dais(component, dai)\ -- 2.33.0

2 1

[PATCH] ASoC: Intel: sof_rt5682: Add support for max98360a speaker amp
by Brent Lu 02 Oct '21

02 Oct '21

From: Malik_Hsu <malik_hsu(a)wistron.corp-partner.google.com> Add a board config adl_mx98360a_rt5682 to support alc5682 headset codec and max98360a speaker amplifier. Follow Intel BT offload design by connecting alc5682 to SSP0 and max98360a to SSP1. Signed-off-by: Malik_Hsu <malik_hsu(a)wistron.corp-partner.google.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 11 +++++++++++ sound/soc/intel/common/soc-acpi-intel-adl-match.c | 13 +++++++++++++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ad42d4c7ade5..613662eedd0d 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1076,6 +1076,17 @@ static const struct platform_device_id board_ids[] = { SOF_MAX98360A_SPEAKER_AMP_PRESENT | SOF_RT5682_SSP_AMP(1)), }, + { + .name = "adl_mx98360_rt5682", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + 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 e4ff280eac23..f5b21a95d222 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_mx98360_rt5682", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &adl_max98360a_amp, + .sof_fw_filename = "sof-adl.ri", + .sof_tplg_filename = "sof-adl-max98360a-rt5682.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_adl_machines); -- 2.25.1

4 3

[PATCH v3 00/12] Add support for on demand pipeline setup/destroy
by Peter Ujfalusi 02 Oct '21

02 Oct '21

Hi, The previous, v2 of this series was sent by Daniel Baluta: https://lore.kernel.org/alsa-devel/20210917143659.401102-1-daniel.baluta@os… We have agreed that it might be better that someone from Intel is going to take it from here as we already have the infrastructure up to test and verify the dynamic pipelines support. Changes since v2 (sent by Daniel Baluta): - patch 10: Fix NULL point dereference in hda_dai_update_config() - I have kept Daniel's SoB for the series. Changes since v1: - Signed-off-by tag added by Daniel This series implements initial support for dynamic pipelines to setup/teardown pipeline as needed when a PCM is open/closed. Initially dynamic pipelines are only supported with single core setup which will be expanded with a follow-up series. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 The feature has been merged on 1st of April to sof-dev, all issues found since has been fixed and squashed to this upstream series. Regards, Peter --- Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 174 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1032 insertions(+), 538 deletions(-) -- 2.33.0

2 13

[PATCH 1/2] ASoC: rt5682s: Enable ASRC auto-disable to fix pop during jack plug-in while playback
by derek.fang＠realtek.com 02 Oct '21

02 Oct '21

From: Derek Fang <derek.fang(a)realtek.com> When codec's ASRC was enabled, the DA and AD filters use the reference clk which tracks the I2S clks. And if the I2S clks' timing of open and close are not expected, this patch allows the filters to switch to use sysclk if ASRC's ref clks disappeared and could fix the below possible issues: 1. Avoid DA filter to keep surplus samples. 2. Avoid that AD filter works failed during dapm's power on. For example, if I2S clks were closed before dacdat during playback off due to jack unplug, it causes ref clks disappeared and DA filter remained some samples which will produce pop noise on the next HP playback. ASRC auto-disable could clear the samples during the playback off. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 3 +++ sound/soc/codecs/rt5682s.h | 13 +++++++++++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 122adeb7beb1..5ae54a51ff9e 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -1061,6 +1061,9 @@ int rt5682s_sel_asrc_clk_src(struct snd_soc_component *component, RT5682S_FILTER_CLK_SEL_MASK, clk_src << RT5682S_FILTER_CLK_SEL_SFT); } + snd_soc_component_update_bits(component, RT5682S_PLL_TRACK_11, + RT5682S_ASRCIN_AUTO_CLKOUT_MASK, RT5682S_ASRCIN_AUTO_CLKOUT_EN); + return 0; } EXPORT_SYMBOL_GPL(rt5682s_sel_asrc_clk_src); diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 55f1febb81e9..59ba4ea3b062 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -994,6 +994,19 @@ #define RT5682S_ASRCIN_FTK_M2_MASK (0x7 << 4) #define RT5682S_ASRCIN_FTK_M2_SFT 4 +/* ASRC Control 11 (0x008c) */ +#define RT5682S_ASRCIN_AUTO_CLKOUT_MASK (0x1 << 5) +#define RT5682S_ASRCIN_AUTO_CLKOUT_EN (0x1 << 5) +#define RT5682S_ASRCIN_AUTO_CLKOUT_DIS (0x0 << 5) +#define RT5682S_ASRCIN_AUTO_RST_MASK (0x1 << 4) +#define RT5682S_ASRCIN_AUTO_RST_EN (0x1 << 4) +#define RT5682S_ASRCIN_AUTO_RST_DIS (0x0 << 4) +#define RT5682S_SEL_LRCK_DET_MASK (0x3) +#define RT5682S_SEL_LRCK_DET_DIV8 (0x3) +#define RT5682S_SEL_LRCK_DET_DIV4 (0x2) +#define RT5682S_SEL_LRCK_DET_DIV2 (0x1) +#define RT5682S_SEL_LRCK_DET_DIV1 (0x0) + /* Depop Mode Control 1 (0x008e) */ #define RT5682S_OUT_HP_L_EN (0x1 << 6) #define RT5682S_OUT_HP_R_EN (0x1 << 5) -- 2.17.1

2 2

[PATCH] ASoC: tegra: Constify static snd_soc_ops
by Rikard Falkeborn 02 Oct '21

02 Oct '21

The struct tegra_machine_snd_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/tegra/tegra_asoc_machine.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra_asoc_machine.c b/sound/soc/tegra/tegra_asoc_machine.c index 735909310a26..3cbe6ef1cf9f 100644 --- a/sound/soc/tegra/tegra_asoc_machine.c +++ b/sound/soc/tegra/tegra_asoc_machine.c @@ -313,7 +313,7 @@ static int tegra_machine_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops tegra_machine_snd_ops = { +static const struct snd_soc_ops tegra_machine_snd_ops = { .hw_params = tegra_machine_hw_params, }; -- 2.33.0

2 1

[PATCH] ASoC: qcom: apq8096: Constify static snd_soc_ops
by Rikard Falkeborn 02 Oct '21

02 Oct '21

The struct iapq8096_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/qcom/apq8096.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index 1a69baefc5ce..c7b7d0864d1a 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -60,7 +60,7 @@ static int msm_snd_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops apq8096_ops = { +static const struct snd_soc_ops apq8096_ops = { .hw_params = msm_snd_hw_params, }; -- 2.33.0

2 1

[PATCH] ASoC: mediatek: mt8195: move of_node_put to remove function
by Trevor Wu 02 Oct '21

02 Oct '21

platforms->of_node and codes->of_node are assigned in probe function, and of_node_put is called at the end of probe function, because of_node seems to be not used after probe functon In this patch, of_node_put is moved to platform remove function in case of_node is used at any occasions after probe function in the future. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 69 +++++++++++-------- 1 file changed, 42 insertions(+), 27 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index cb15467e5fc5..95abaadcd842 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -26,6 +26,9 @@ #define RT5682_DEV0_NAME "rt5682.2-001a" struct mt8195_mt6359_rt1019_rt5682_priv { + struct device_node *platform_node; + struct device_node *hdmi_node; + struct device_node *dp_node; struct snd_soc_jack headset_jack; struct snd_soc_jack dp_jack; struct snd_soc_jack hdmi_jack; @@ -994,33 +997,36 @@ static struct snd_soc_card mt8195_mt6359_rt1019_rt5682_soc_card = { static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8195_mt6359_rt1019_rt5682_soc_card; - struct device_node *platform_node; - struct device_node *dp_node; - struct device_node *hdmi_node; struct snd_soc_dai_link *dai_link; - struct mt8195_mt6359_rt1019_rt5682_priv *priv = NULL; + struct mt8195_mt6359_rt1019_rt5682_priv *priv; int ret, i; card->dev = &pdev->dev; - platform_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,platform", 0); - if (!platform_node) { + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->platform_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,platform", 0); + if (!priv->platform_node) { dev_dbg(&pdev->dev, "Property 'platform' missing or invalid\n"); return -EINVAL; } for_each_card_prelinks(card, i, dai_link) { if (!dai_link->platforms->name) - dai_link->platforms->of_node = platform_node; + dai_link->platforms->of_node = priv->platform_node; if (strcmp(dai_link->name, "DPTX_BE") == 0) { - dp_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,dptx-codec", 0); - if (!dp_node) { + priv->dp_node = + of_parse_phandle(pdev->dev.of_node, + "mediatek,dptx-codec", 0); + + if (!priv->dp_node) { dev_dbg(&pdev->dev, "No property 'dptx-codec'\n"); } else { - dai_link->codecs->of_node = dp_node; + dai_link->codecs->of_node = priv->dp_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_dptx_codec_init; @@ -1028,12 +1034,13 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - hdmi_node = of_parse_phandle(pdev->dev.of_node, - "mediatek,hdmi-codec", 0); - if (!hdmi_node) { + priv->hdmi_node = + of_parse_phandle(pdev->dev.of_node, + "mediatek,hdmi-codec", 0); + if (!priv->hdmi_node) { dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n"); } else { - dai_link->codecs->of_node = hdmi_node; + dai_link->codecs->of_node = priv->hdmi_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_hdmi_codec_init; @@ -1041,26 +1048,33 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } } - priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) { - ret = -ENOMEM; - goto out; - } - snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret) { dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); + of_node_put(priv->hdmi_node); + of_node_put(priv->dp_node); + of_node_put(priv->platform_node); + } -out: - of_node_put(hdmi_node); - of_node_put(dp_node); - of_node_put(platform_node); return ret; } +static int mt8195_mt6359_rt1019_rt5682_dev_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct mt8195_mt6359_rt1019_rt5682_priv *priv = + snd_soc_card_get_drvdata(card); + + of_node_put(priv->hdmi_node); + of_node_put(priv->dp_node); + of_node_put(priv->platform_node); + + return 0; +} + #ifdef CONFIG_OF static const struct of_device_id mt8195_mt6359_rt1019_rt5682_dt_match[] = { {.compatible = "mediatek,mt8195_mt6359_rt1019_rt5682",}, @@ -1082,6 +1096,7 @@ static struct platform_driver mt8195_mt6359_rt1019_rt5682_driver = { .pm = &mt8195_mt6359_rt1019_rt5682_pm_ops, }, .probe = mt8195_mt6359_rt1019_rt5682_dev_probe, + .remove = mt8195_mt6359_rt1019_rt5682_dev_remove, }; module_platform_driver(mt8195_mt6359_rt1019_rt5682_driver); -- 2.18.0

2 1

[PATCH] ASoC: SOF: Intel: hda-stream: limit PROCEN workaround
by Peter Ujfalusi 01 Oct '21

01 Oct '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The work-around enabled in hda-stream.c is only required on earlier versions of SOCs/PCH (Skylake, KabyLake, ApolloLake, GeminiLake). Before setting the format on the host DMA, it is required to couple the host and link DMA - which as a consequence shall use the same format. This patch introduces a quirk field in the platform descriptor and makes the work-around conditional. Newer platforms have no limitations on the use of host and link DMA, which can use different formats. 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: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/hda-stream.c | 18 ++++++++++++------ sound/soc/sof/intel/shim.h | 4 ++++ 3 files changed, 17 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/apl.c b/sound/soc/sof/intel/apl.c index e6a1f6532547..16b7d8b1adff 100644 --- a/sound/soc/sof/intel/apl.c +++ b/sound/soc/sof/intel/apl.c @@ -143,5 +143,6 @@ const struct sof_intel_dsp_desc apl_chip_info = { .rom_init_timeout = 150, .ssp_count = APL_SSP_COUNT, .ssp_base_offset = APL_SSP_BASE_OFFSET, + .quirks = SOF_INTEL_PROCEN_FMT_QUIRK, }; EXPORT_SYMBOL_NS(apl_chip_info, SND_SOC_SOF_INTEL_HDA_COMMON); diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..47a465b06da5 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -405,6 +405,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, struct snd_dma_buffer *dmab, struct snd_pcm_hw_params *params) { + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); struct hdac_bus *bus = sof_to_bus(sdev); struct hdac_stream *hstream = &stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); @@ -538,6 +539,7 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, /* * Recommended hardware programming sequence for HDAudio DMA format + * on earlier platforms - this is not needed on newer platforms * * 1. Put DMA into coupled mode by clearing PPCTL.PROCEN bit * for corresponding stream index before the time of writing @@ -547,9 +549,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, * enable decoupled mode */ - /* couple host and link DMA, disable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, 0); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* couple host and link DMA, disable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, 0); + } /* program stream format */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, @@ -557,9 +561,11 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, SOF_HDA_ADSP_REG_CL_SD_FORMAT, 0xffff, hstream->format_val); - /* decouple host and link DMA, enable DSP features */ - snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, - mask, mask); + if (chip->quirks & SOF_INTEL_PROCEN_FMT_QUIRK) { + /* decouple host and link DMA, enable DSP features */ + snd_sof_dsp_update_bits(sdev, HDA_DSP_PP_BAR, SOF_HDA_REG_PP_PPCTL, + mask, mask); + } /* program last valid index */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, diff --git a/sound/soc/sof/intel/shim.h b/sound/soc/sof/intel/shim.h index e9f7d4d7fcce..8e2a0f4c0f66 100644 --- a/sound/soc/sof/intel/shim.h +++ b/sound/soc/sof/intel/shim.h @@ -151,6 +151,9 @@ #define PCI_PMCS 0x84 #define PCI_PMCS_PS_MASK 0x3 +/* Intel quirks */ +#define SOF_INTEL_PROCEN_FMT_QUIRK BIT(0) + /* DSP hardware descriptor */ struct sof_intel_dsp_desc { int cores_num; @@ -166,6 +169,7 @@ struct sof_intel_dsp_desc { int ssp_base_offset; /* base address of the SSPs */ u32 sdw_shim_base; u32 sdw_alh_base; + u32 quirks; bool (*check_sdw_irq)(struct snd_sof_dev *sdev); }; -- 2.33.0

3 4

[PATCH][next] ALSA: usb-audio: Fix sum of uninitialized variable sample_accum
by Colin King 01 Oct '21

01 Oct '21

From: Colin Ian King <colin.king(a)canonical.com> Variable sample_accum is not being intialized and then has ep->sample_rem added to it, leading to a bogus value. One solution is to initialize it to zero at declaration time, but it is probably best to just assign it to ep->sample_rem on first use. Addresses-Coveriry: ("Uninitialized scalar variable") Fixes: f0bd62b64016 ("ALSA: usb-audio: Improve frames size computation") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 42c0d2db8ba8..c6a33732db3f 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -182,7 +182,7 @@ static int next_packet_size(struct snd_usb_endpoint *ep, unsigned int avail) if (ep->fill_max) return ep->maxframesize; - sample_accum += ep->sample_rem; + sample_accum = ep->sample_rem; if (sample_accum >= ep->pps) { sample_accum -= ep->pps; ret = ep->packsize[1]; -- 2.32.0

3 2

[PATCH] ALSA: usb-audio: Fix packet size calculation regression
by Takashi Iwai 01 Oct '21

01 Oct '21

The commit d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") introduced the available frame size check, but the conversion forgot to initialize the temporary variable properly, and it resulted in a bogus calculation. This patch fixes it. Fixes: d215f63d49da ("ALSA: usb-audio: Check available frames for the next packet size") Reported-by: Colin Ian King <colin.king(a)canonical.com> Link: https://lore.kernel.org/r/20211001104417.14291-1-colin.king@canonical.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 42c0d2db8ba8..743b8287cfcd 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -182,7 +182,7 @@ static int next_packet_size(struct snd_usb_endpoint *ep, unsigned int avail) if (ep->fill_max) return ep->maxframesize; - sample_accum += ep->sample_rem; + sample_accum = ep->sample_accum + ep->sample_rem; if (sample_accum >= ep->pps) { sample_accum -= ep->pps; ret = ep->packsize[1]; -- 2.26.2

1 0

ALSA kernel projects - for academic purposes
by Muni Sekhar 01 Oct '21

01 Oct '21

Hi all, What small projects would you suggest to a novice with the ALSA kernel. The aim is to develop a familiarity with the ALSA kernel source code, and also to submit it for academic purposes. -- Thanks, Sekhar

7 15

[PATCH v2 0/2] ASoC: mediatek: Add support for MT8195 sound card with rt1011 and rt5682
by Trevor Wu 01 Oct '21

01 Oct '21

This series of patches adds support for mt8195 board with mt6359, rt1011 and rt5682. Patches are based on broonie tree "for-next" branch. Changes since v1: - refined code based on Pierre-Louis's suggestions - added missing of_node_put - fixed wrong fixup assignment on HDMITX Trevor Wu (2): ASoC: mediatek: mt8195: add machine driver with mt6359, rt1011 and rt5682 dt-bindings: mediatek: mt8195: add mt8195-mt6359-rt1011-rt5682 document .../sound/mt8195-mt6359-rt1011-rt5682.yaml | 47 + sound/soc/mediatek/Kconfig | 15 + sound/soc/mediatek/mt8195/Makefile | 1 + .../mt8195/mt8195-mt6359-rt1011-rt5682.c | 1155 +++++++++++++++++ 4 files changed, 1218 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8195-mt6359-rt1011-rt5682.yaml create mode 100644 sound/soc/mediatek/mt8195/mt8195-mt6359-rt1011-rt5682.c -- 2.18.0

1 2

[PATCH] ALSA: hda - Enable headphone mic on Dell Latitude laptops with ALC3254
by Chris Chiu 01 Oct '21

01 Oct '21

The headphone mic is not working on Dell Latitude laptops with ALC3254. The codec vendor id is 0x10ec0295 and share the same pincfg as defined in ALC295_STANDARD_PINS. So the ALC269_FIXUP_DELL1_MIC_NO_PRESENCE will be applied per alc269_pin_fixup_tbl[] but actually the headphone mic is using NID 0x1b instead of 0x1a. The ALC269_FIXUP_DELL4_MIC_NO_PRESENCE need to be applied instead. Use ALC269_FIXUP_DELL4_MIC_NO_PRESENCE for particular models before a generic fixup comes out. Signed-off-by: Chris Chiu <chris.chiu(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4407f7da57c4..70778aa137cf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8452,6 +8452,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.20.1

2 1

[PATCH] ALSA: usb-audio: disable implicit feedback sync for Behringer UFX1204 and UFX1604
by Geraldo Nascimento 01 Oct '21

01 Oct '21

Hi Takashi, Hopefully you will be able to apply this cleanly to branch master of your sound.git. Thanks! Geraldo Nascimento --- Behringer UFX1204 and UFX1604 have Synchronous endpoints to which current ALSA code applies implicit feedback sync as if they were Asynchronous endpoints. This breaks UAC compliance and is unneeded. The commit 5e35dc0338d85ccebacf3f77eca1e5dea73155e8 and subsequent 1a15718b41df026cffd0e42cfdc38a1384ce19f9 were meant to clear up noise. Unfortunately, noise persisted for those using higher sample rates and this was only solved by commit d2e8f641257d0d3af6e45d6ac2d6f9d56b8ea964 Since there are no more reports of noise, let's get rid of the implicit-fb quirks breaking UAC compliance. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 23767a14d126..70319c822c10 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -54,8 +54,6 @@ static const struct snd_usb_implicit_fb_match playback_implicit_fb_quirks[] = { /* Fixed EP */ /* FIXME: check the availability of generic matching */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0001, 0x81, 1), /* Behringer UFX1604 */ - IMPLICIT_FB_FIXED_DEV(0x1397, 0x0002, 0x81, 1), /* Behringer UFX1204 */ IMPLICIT_FB_FIXED_DEV(0x2466, 0x8010, 0x81, 2), /* Fractal Audio Axe-Fx III */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0001, 0x81, 2), /* Solid State Logic SSL2 */ IMPLICIT_FB_FIXED_DEV(0x31e9, 0x0002, 0x81, 2), /* Solid State Logic SSL2+ */

2 1

[PATCH v2] soundwire: debugfs: use controller id and link_id for debugfs
by Srinivas Kandagatla 01 Oct '21

01 Oct '21

link_id can be zero and if we have multiple controller instances in a system like Qualcomm debugfs will end-up with duplicate namespace resulting in incorrect debugfs entries. Using bus-id and link-id combination should give a unique debugfs directory entry and should fix below warning too. "debugfs: Directory 'master-0' with parent 'soundwire' already present!" Fixes: bf03473d5bcc ("soundwire: add debugfs support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: Added Link ID along with bus id. drivers/soundwire/debugfs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/debugfs.c b/drivers/soundwire/debugfs.c index b6cad0d59b7b..49900cd207bc 100644 --- a/drivers/soundwire/debugfs.c +++ b/drivers/soundwire/debugfs.c @@ -19,7 +19,7 @@ void sdw_bus_debugfs_init(struct sdw_bus *bus) return; /* create the debugfs master-N */ - snprintf(name, sizeof(name), "master-%d", bus->link_id); + snprintf(name, sizeof(name), "master-%d-%d", bus->id, bus->link_id); bus->debugfs = debugfs_create_dir(name, sdw_debugfs_root); } -- 2.21.0

3 2

[PATCH] ASoC: max98373: Mark cache dirty before entering sleep
by Ryan Lee 01 Oct '21

01 Oct '21

Amp lose its register values in case amp power loss or 'ForceReset' over Soundwire SCP_ctrl register(0x0044) or HW_RESET pin control during the audio suspend and resume. Mark cache dirty before audio suspension to restore existing values when audio resume. Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- sound/soc/codecs/max98373-sdw.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98373-sdw.c b/sound/soc/codecs/max98373-sdw.c index dc520effc61c..a7e4a6e880b0 100644 --- a/sound/soc/codecs/max98373-sdw.c +++ b/sound/soc/codecs/max98373-sdw.c @@ -259,6 +259,7 @@ static __maybe_unused int max98373_suspend(struct device *dev) regmap_read(max98373->regmap, max98373->cache[i].reg, &max98373->cache[i].val); regcache_cache_only(max98373->regmap, true); + regcache_mark_dirty(max98373->regmap); return 0; } -- 2.17.1

3 16

[PATCH regression fix] ASoC: nau8824: Fix headphone vs headset, button-press detection no longer working
by Hans de Goede 30 Sep '21

30 Sep '21

Commit 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") replaced the nau8824_dapm_enable_pin() helper with direct calls to snd_soc_dapm_enable_pin(), but the helper was using snd_soc_dapm_force_enable_pin() and not forcing the MICBIAS + SAR supplies on breaks headphone vs headset and button-press detection. Replace the snd_soc_dapm_enable_pin() calls with snd_soc_dapm_force_enable_pin() to fix this. Cc: stable(a)vger.kernel.org Fixes: 1d25684e2251 ("ASoC: nau8824: Fix open coded prefix handling") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/codecs/nau8824.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index cea1d43e0cb8..77f647529354 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -874,8 +874,8 @@ static void nau8824_jdet_work(struct work_struct *work) struct regmap *regmap = nau8824->regmap; int adc_value, event = 0, event_mask = 0; - snd_soc_dapm_enable_pin(dapm, "MICBIAS"); - snd_soc_dapm_enable_pin(dapm, "SAR"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); snd_soc_dapm_sync(dapm); msleep(100); -- 2.31.1

2 1

[PATCH 1/3] ASoC: rt5682s: Remove the volatile SW reset register from reg_default
by derek.fang＠realtek.com 30 Sep '21

30 Sep '21

From: Derek Fang <derek.fang(a)realtek.com> This reg is for SW reset. It shouldn't have default value, so remove. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d3e965b2e707..2b229c074a31 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -72,7 +72,6 @@ static void rt5682s_apply_patch_list(struct rt5682s_priv *rt5682s, } const struct reg_default rt5682s_reg[] = { - {0x0000, 0x0001}, {0x0002, 0x8080}, {0x0003, 0x0001}, {0x0005, 0x0000}, -- 2.17.1

2 3

[PATCH] ASoC: dt-bindings: rt5682s: fix the device-tree schema errors
by derek.fang＠realtek.com 30 Sep '21

30 Sep '21

From: Derek Fang <derek.fang(a)realtek.com> Fix the device-tree schema errors that be reported by using 'make dt_binding_check'. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- .../devicetree/bindings/sound/realtek,rt5682s.yaml | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml index 7a5f1d0fd3e2..2b8b7b51fe55 100644 --- a/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml +++ b/Documentation/devicetree/bindings/sound/realtek,rt5682s.yaml @@ -29,6 +29,8 @@ properties: - 0 # dmic1 data is not used - 1 # using GPIO2 pin as dmic1 data pin - 2 # using GPIO5 pin as dmic1 data pin + description: | + Specify which GPIO pin be used as DMIC1 data pin. realtek,dmic1-clk-pin: $ref: /schemas/types.yaml#/definitions/uint32 @@ -36,12 +38,16 @@ properties: - 0 # dmic1 clk is not used - 1 # using GPIO1 pin as dmic1 clock pin - 2 # using GPIO3 pin as dmic1 clock pin + description: | + Specify which GPIO pin be used as DMIC1 clk pin. realtek,jd-src: $ref: /schemas/types.yaml#/definitions/uint32 enum: - 0 # No JD is used - 1 # using JD1 as JD source + description: | + Specify which JD source be used. realtek,ldo1-en-gpios: description: | @@ -72,9 +78,9 @@ properties: const: 1 clock-output-names: - items: - - description: Name given for DAI word clock output. - - description: Name given for DAI bit clock output. + minItems: 2 + maxItems: 2 + description: Name given for DAI word clock and bit clock outputs. additionalProperties: false -- 2.17.1

2 1

[PATCH] ASoC: fsl_rpmsg: Add rpmsg audio support for i.MX8ULP
by Shengjiu Wang 30 Sep '21

30 Sep '21

On i.MX8ULP the audio interface and codec are controlled by Cortex-M domain, Cortex-M core provides audio service over rpmsg. The rpmsg audio function is almost same as i.MX7ULP platform, so share same configuration. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_rpmsg.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 07abad7fe372..8508bc7f239d 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -174,6 +174,7 @@ static const struct of_device_id fsl_rpmsg_ids[] = { { .compatible = "fsl,imx8mm-rpmsg-audio", .data = &imx8mm_data}, { .compatible = "fsl,imx8mn-rpmsg-audio", .data = &imx8mn_data}, { .compatible = "fsl,imx8mp-rpmsg-audio", .data = &imx8mp_data}, + { .compatible = "fsl,imx8ulp-rpmsg-audio", .data = &imx7ulp_data}, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_rpmsg_ids); -- 2.17.1

2 1

[PATCH] ASoC: SOF: Change SND_SOC_SOF_TOPLEVEL from config to menuconfig
by Peter Ujfalusi 30 Sep '21

30 Sep '21

We have growing number of options under SND_SOC_SOF_TOPLEVEL as SOF adaptation is growing (Intel, NXP, AMD and Mediatek) and new features are added. It will make the menuconfig user experience much cleaner if we move the SOF options under a separate page. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- Hi, screen shots of before and after of menuconfig is attached to the PR for sof-dev: https://github.com/thesofproject/linux/pull/3148 Regards, Peter sound/soc/sof/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index cd659493b5df..94d1a859fedc 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -1,5 +1,5 @@ # SPDX-License-Identifier: GPL-2.0-only -config SND_SOC_SOF_TOPLEVEL +menuconfig SND_SOC_SOF_TOPLEVEL bool "Sound Open Firmware Support" help This adds support for Sound Open Firmware (SOF). SOF is free and -- 2.33.0

2 1

[PATCH] ASoC: wm_adsp: remove a repeated including
by Guo Zhengkui 30 Sep '21

30 Sep '21

Remove a repeated "#include <linux/firmware.h>" in line 32. Signed-off-by: Guo Zhengkui <guozhengkui(a)vivo.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 6c5d55b3b311..b9fb4f547d5e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -29,7 +29,6 @@ #include <sound/jack.h> #include <sound/initval.h> #include <sound/tlv.h> -#include <linux/firmware.h> #include "wm_adsp.h" -- 2.20.1

3 2

[PATCH] ALSA: hda/realtek: Enable 4-speaker output for Dell Precision 5560 laptop
by Takashi Iwai 30 Sep '21

30 Sep '21

From: John Liu <johnliu55tw(a)gmail.com> The Dell Precision 5560 laptop appears to use the 4-speakers-on-ALC289 audio just like its sibling product XPS 9510, so it requires the same quirk to enable woofer output. Tested on my Dell Precision 5560. Signed-off-by: John Liu <johnliu55tw(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- 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 4407f7da57c4..f17bfc6007d3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8452,6 +8452,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), + SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.26.2

1 0

[PATCH v2 0/2] ALSA: hda: Reduce CPU hog with SKL+ position reporting
by Takashi Iwai 30 Sep '21

30 Sep '21

Hi, this is a v2 patch set for reducing the CPU hog with the HD-audio PCM position reporting. The first patch is almost same, while the second patch is added to take back to the position buffer as suggested by Pierre. Takashi v1: https://lore.kernel.org/r/20210910141002.32749-1-tiwai@suse.de === Takashi Iwai (2): ALSA: hda: Reduce udelay() at SKL+ position reporting ALSA: hda: Use position buffer for SKL+ again sound/pci/hda/hda_intel.c | 49 ++++++++++++++++++--------------------- 1 file changed, 23 insertions(+), 26 deletions(-) -- 2.26.2

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[PATCH][next] ALSA: virtio: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 30 Sep '21

30 Sep '21

There is a regular need in the kernel to provide a way to declare having a dynamically sized set of trailing elements in a structure. Kernel code should always use “flexible array members”[1] for these cases. The older style of one-element or zero-length arrays should no longer be used[2]. Also, make use of the struct_size() helper in kzalloc(). [1] https://en.wikipedia.org/wiki/Flexible_array_member [2] https://www.kernel.org/doc/html/v5.10/process/deprecated.html#zero-length-a… Link: https://github.com/KSPP/linux/issues/78 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/virtio/virtio_pcm_msg.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/virtio/virtio_pcm_msg.c b/sound/virtio/virtio_pcm_msg.c index f88c8f29cbd8..aca2dc1989ba 100644 --- a/sound/virtio/virtio_pcm_msg.c +++ b/sound/virtio/virtio_pcm_msg.c @@ -20,7 +20,7 @@ struct virtio_pcm_msg { struct virtio_snd_pcm_xfer xfer; struct virtio_snd_pcm_status status; size_t length; - struct scatterlist sgs[0]; + struct scatterlist sgs[]; }; /** @@ -146,8 +146,7 @@ int virtsnd_pcm_msg_alloc(struct virtio_pcm_substream *vss, int sg_num = virtsnd_pcm_sg_num(data, period_bytes); struct virtio_pcm_msg *msg; - msg = kzalloc(sizeof(*msg) + sizeof(*msg->sgs) * (sg_num + 2), - GFP_KERNEL); + msg = kzalloc(struct_size(msg, sgs, sg_num + 2), GFP_KERNEL); if (!msg) return -ENOMEM; -- 2.27.0

2 1

Behringer UFX1604 and UFX1204 unneeded implicit feedback
by Geraldo Nascimento 30 Sep '21

30 Sep '21

Hello Takashi and everyone, Behringer UFX1204 and UFX1604 have sync endpoints and we set up implicit feedback sync on them. This is against the UAC spec. We've already fixed a major source of noise by explictly setting up the Clock Selector. Therefore in my humble opinion we should now disable implicit feedback sync for those two devices. What do you think? Thank you, Geraldo Nascimento

3 3

[PATCH] Fixes HP Spectre x360 15-eb1xxx speakers
by Davide Baldo 30 Sep '21

30 Sep '21

First kernel contribution, I only tested this patch myself on a 5.14.7. There is probably some way to fix the remaining issues, but I need some guidance on how I could test the different pinouts since hdajackretask hangs the audio card and I can't test out combinations. In the meantime this commit fix the most serius problem: silent speakers. In laptop 'HP Spectre x360 Convertible 15-eb1xxx/8811' both front and rear speakers are silent, this patch fixes that by overriding the pin layout and by initializing the amplifier which needs a GPIO pin to be set to 1 then 0, similar to the existing HP Spectre x360 14 model. This patch also correctly map the mute LED but since there is no microphone on/off switch exposed by the alsa subsystem it never turns on by itself. There are still known audio issues in this laptop: headset microphone doesn't work, the button to mute/unmute microphone is not yet mapped, the LED of the mute/unmute speakers doesn't seems to be exposed via GPIO and never turns on. Signed-off-by: Davide Baldo <davide(a)baldo.me> --- sound/pci/hda/patch_realtek.c | 41 ++++++++++++++++++++++++++++++++++- 1 file changed, 40 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 70516527ebce..90f845976793 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6414,6 +6414,38 @@ static void alc_fixup_no_int_mic(struct hda_codec *codec, } } +/* GPIO1 = amplifier on/off + * GPIO3 = mic mute LED + */ +static void alc285_fixup_hp_spectre_x360_eb1(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + static const struct hda_pintbl pincfgs[] = { + { 0x14, 0x90170110 }, /* front speakers */ + { } + }; + + //enable micmute led + alc_fixup_hp_gpio_led(codec, action, 0x00, 0x04); + spec->micmute_led_polarity = 1; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + /* needed for amp of back speakers */ + spec->gpio_mask |= 0x01; + spec->gpio_dir |= 0x01; + snd_hda_apply_pincfgs(codec, pincfgs); + break; + case HDA_FIXUP_ACT_INIT: + /* need to toggle GPIO to enable the amp of back speakers */ + alc_update_gpio_data(codec, 0x01, true); + msleep(100); + alc_update_gpio_data(codec, 0x01, false); + break; + } +} + static void alc285_fixup_hp_spectre_x360(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -6534,6 +6566,7 @@ enum { ALC269_FIXUP_HP_DOCK_GPIO_MIC1_LED, ALC280_FIXUP_HP_9480M, ALC245_FIXUP_HP_X360_AMP, + ALC285_FIXUP_HP_SPECTRE_X360_EB1, ALC288_FIXUP_DELL_HEADSET_MODE, ALC288_FIXUP_DELL1_MIC_NO_PRESENCE, ALC288_FIXUP_DELL_XPS_13, @@ -6658,7 +6691,7 @@ enum { ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, ALC623_FIXUP_LENOVO_THINKSTATION_P340, ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, - ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST, + ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST }; static const struct hda_fixup alc269_fixups[] = { @@ -8222,6 +8255,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360, }, + [ALC285_FIXUP_HP_SPECTRE_X360_EB1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_hp_spectre_x360_eb1 + }, [ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_ideapad_s740_coef, @@ -8415,6 +8452,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x84da, "HP OMEN dc0019-ur", ALC295_FIXUP_HP_OMEN), SND_PCI_QUIRK(0x103c, 0x84e7, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8519, "HP Spectre x360 15-df0xxx", ALC285_FIXUP_HP_SPECTRE_X360), + SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x861f, "HP Elite Dragonfly G1", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x869d, "HP", ALC236_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x86c7, "HP Envy AiO 32", ALC274_FIXUP_HP_ENVY_GPIO), @@ -8858,6 +8896,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC245_FIXUP_HP_X360_AMP, .name = "alc245-hp-x360-amp"}, {.id = ALC295_FIXUP_HP_OMEN, .name = "alc295-hp-omen"}, {.id = ALC285_FIXUP_HP_SPECTRE_X360, .name = "alc285-hp-spectre-x360"}, + {.id = ALC285_FIXUP_HP_SPECTRE_X360_EB1, .name = "alc285-hp-spectre-x360-eb1"}, {.id = ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP, .name = "alc287-ideapad-bass-spk-amp"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, -- 2.32.0

3 5

[PATCH] ALSA: seq: Fix a potential UAF by wrong private_free call order
by Takashi Iwai 30 Sep '21

30 Sep '21

John Keeping reported and posted a patch for a potential UAF in rawmidi sequencer destruction: the snd_rawmidi_dev_seq_free() may be called after the associated rawmidi object got already freed. After a deeper look, it turned out that the bug is rather the incorrect private_free call order for a snd_seq_device. The snd_seq_device private_free gets called at the release callback of the sequencer device object, while this was rather expected to be executed at the snd_device call chains that runs at the beginning of the whole card-free procedure. It's been broken since the rewrite of sequencer-device binding (although it hasn't surfaced because the sequencer device release happens usually right along with the card device release). This patch corrects the private_free call to be done in the right place, at snd_seq_device_dev_free(). Fixes: 7c37ae5c625a ("ALSA: seq: Rewrite sequencer device binding with standard bus") Reported-and-tested-by: John Keeping <john(a)metanate.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq_device.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index 382275c5b193..7f3fd8eb016f 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -156,6 +156,8 @@ static int snd_seq_device_dev_free(struct snd_device *device) struct snd_seq_device *dev = device->device_data; cancel_autoload_drivers(); + if (dev->private_free) + dev->private_free(dev); put_device(&dev->dev); return 0; } @@ -183,11 +185,7 @@ static int snd_seq_device_dev_disconnect(struct snd_device *device) static void snd_seq_dev_release(struct device *dev) { - struct snd_seq_device *sdev = to_seq_dev(dev); - - if (sdev->private_free) - sdev->private_free(sdev); - kfree(sdev); + kfree(to_seq_dev(dev)); } /* -- 2.26.2

1 0

[PATCH] ALSA: rawmidi: Fix potential UAF from sequencer destruction
by John Keeping 30 Sep '21

30 Sep '21

If the sequencer device outlives the rawmidi device, then snd_rawmidi_dev_seq_free() will run after release_rawmidi_device() has freed the snd_rawmidi structure. This can easily be reproduced with CONFIG_DEBUG_KOBJECT_RELEASE. Keep a reference to the rawmidi device until the sequencer has been destroyed in order to avoid this. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/core/rawmidi.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6f30231bdb88..b015f5f69175 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -1860,6 +1860,7 @@ static void snd_rawmidi_dev_seq_free(struct snd_seq_device *device) struct snd_rawmidi *rmidi = device->private_data; rmidi->seq_dev = NULL; + put_device(&rmidi->dev); } #endif @@ -1936,6 +1937,9 @@ static int snd_rawmidi_dev_register(struct snd_device *device) #if IS_ENABLED(CONFIG_SND_SEQUENCER) if (!rmidi->ops || !rmidi->ops->dev_register) { /* own registration mechanism */ if (snd_seq_device_new(rmidi->card, rmidi->device, SNDRV_SEQ_DEV_ID_MIDISYNTH, 0, &rmidi->seq_dev) >= 0) { + /* Ensure we outlive the sequencer (see snd_rawmidi_dev_seq_free). */ + get_device(&rmidi->dev); + rmidi->seq_dev->private_data = rmidi; rmidi->seq_dev->private_free = snd_rawmidi_dev_seq_free; sprintf(rmidi->seq_dev->name, "MIDI %d-%d", rmidi->card->number, rmidi->device); -- 2.33.0

2 8

[PATCH] ASoC: dpcm: fix race condition to dpcm links in dpcm_be_dai_trigger
by Gyeongtaek Lee 30 Sep '21

30 Sep '21

If routing change and underrun stop is run at the same time, data abort can be occurred by the following sequence. CPU0: Processing underrun CPU1: Processing routing change dpcm_be_dai_trigger(): dpcm_be_disconnect(): for_each_dpcm_be(fe, stream, dpcm) { spin_lock_irqsave(&fe->card->dpcm_lock, flags); list_del(&dpcm->list_be); list_del(&dpcm->list_fe); spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); kfree(dpcm); struct snd_soc_pcm_runtime *be = dpcm->be; <-- Accessing freed memory To prevent this situation, dpcm_lock is needed during accessing the lists for dpcm links. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- sound/soc/soc-pcm.c | 53 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 50 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 48f71bb81a2f..df2cd4c0dabe 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1993,17 +1993,63 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, return ret; } +struct dpcm_be_list { + unsigned int num; + struct snd_soc_pcm_runtime *be[]; +}; + +static int dpcm_create_be_list(struct snd_soc_pcm_runtime *fe, int stream, + struct dpcm_be_list **be_list) +{ + struct snd_soc_dpcm *dpcm; + struct dpcm_be_list *be; + int size = 0; + int ret = 0; + unsigned long flags; + + spin_lock_irqsave(&fe->card->dpcm_lock, flags); + + for_each_dpcm_be(fe, stream, dpcm) + size++; + + be = kzalloc(struct_size(be, be, size), GFP_ATOMIC); + if (!be) { + ret = -ENOMEM; + } else { + unsigned int i = 0; + + for_each_dpcm_be(fe, stream, dpcm) + be->be[i++] = dpcm->be; + + *be_list = be; + } + + spin_unlock_irqrestore(&fe->card->dpcm_lock, flags); + + return ret; +} + +static void dpcm_free_be_list(struct dpcm_be_list *be_list) +{ + kfree(be_list); +} + int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_pcm_runtime *be; - struct snd_soc_dpcm *dpcm; + struct dpcm_be_list *be_list; int ret = 0; + int i; - for_each_dpcm_be(fe, stream, dpcm) { + ret = dpcm_create_be_list(fe, stream, &be_list); + if (ret < 0) + return ret; + + for(i = 0; i < be_list->num; i++) { struct snd_pcm_substream *be_substream; - be = dpcm->be; + be = be_list->be[i]; be_substream = snd_soc_dpcm_get_substream(be, stream); /* is this op for this BE ? */ @@ -2092,6 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, if (ret < 0) dev_err(fe->dev, "ASoC: %s() failed at %s (%d)\n", __func__, be->dai_link->name, ret); + dpcm_free_be_list(be_list); return ret; } EXPORT_SYMBOL_GPL(dpcm_be_dai_trigger); base-commit: 4ac6d90867a4de2e12117e755dbd76e08d88697f -- 2.21.0

2 3

Re: [PATCH] ASoC: dt-bindings: rt5682s: fix the device-tree schema errors
by David Heidelberg 30 Sep '21

30 Sep '21

Our CI seems to be happy again. Thanks Reviewed-by: David Heidelberg <david(a)ixit.cz> Best regards David Heidelberg

1 0

[PATCH] ASoC: wm_adsp: Fix compilation when debugfs is disabled
by Simon Trimmer 29 Sep '21

29 Sep '21

Guard the use of component->debufs_root against CONFIG_DEBUG_FS Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f17c749c24c3..03cfaa82b031 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1028,7 +1028,9 @@ int wm_adsp2_component_probe(struct wm_adsp *dsp, struct snd_soc_component *comp snprintf(preload, ARRAY_SIZE(preload), "%s Preload", dsp->cs_dsp.name); snd_soc_component_disable_pin(component, preload); +#if IS_ENABLED(CONFIG_DEBUG_FS) cs_dsp_init_debugfs(&dsp->cs_dsp, component->debugfs_root); +#endif dsp->component = component; -- 2.33.0

2 3

[PATCH] ASoC: ux500: mop500: Constify static snd_soc_ops
by Rikard Falkeborn 29 Sep '21

29 Sep '21

The struct mop500_ab8500_ops is only assigned to the ops field in the snd_soc_dai_link struct which is a pointer to const struct snd_soc_ops. Make it const to allow the compiler to put it in read-only memory. Signed-off-by: Rikard Falkeborn <rikard.falkeborn(a)gmail.com> --- sound/soc/ux500/mop500_ab8500.c | 2 +- sound/soc/ux500/mop500_ab8500.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 2c39c7a2fd7d..3e654e708f78 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -348,7 +348,7 @@ static int mop500_ab8500_hw_free(struct snd_pcm_substream *substream) return 0; } -struct snd_soc_ops mop500_ab8500_ops[] = { +const struct snd_soc_ops mop500_ab8500_ops[] = { { .hw_params = mop500_ab8500_hw_params, .hw_free = mop500_ab8500_hw_free, diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h index 8138a4e9aaf5..087ef246d87d 100644 --- a/sound/soc/ux500/mop500_ab8500.h +++ b/sound/soc/ux500/mop500_ab8500.h @@ -11,7 +11,7 @@ #ifndef MOP500_AB8500_H #define MOP500_AB8500_H -extern struct snd_soc_ops mop500_ab8500_ops[]; +extern const struct snd_soc_ops mop500_ab8500_ops[]; int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd); void mop500_ab8500_remove(struct snd_soc_card *card); -- 2.33.0

2 1

[PATCH] ASoC: ti: Constify static snd_soc_ops
by Rikard Falkeborn 29 Sep '21

29 Sep '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/ti/davinci-evm.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index b043a0070d20..68d69e32681a 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -73,7 +73,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops evm_ops = { +static const struct snd_soc_ops evm_ops = { .startup = evm_startup, .shutdown = evm_shutdown, .hw_params = evm_hw_params, diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c index 2e3d1eea77c1..da809c7f25a4 100644 --- a/sound/soc/ti/omap-abe-twl6040.c +++ b/sound/soc/ti/omap-abe-twl6040.c @@ -96,7 +96,7 @@ static int omap_abe_dmic_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops omap_abe_dmic_ops = { +static const struct snd_soc_ops omap_abe_dmic_ops = { .hw_params = omap_abe_dmic_hw_params, }; -- 2.33.0

2 1

[PATCH] ASoC: rt5682s Minor modifications to look more sensibly
by derek.fang＠realtek.com 29 Sep '21

29 Sep '21

From: Derek Fang <derek.fang(a)realtek.com> These modifications wouldn't affect the driver's behavior, but the code could look more sensibly. 1. Remove the volatile reg (0x0000) in reg_default. 2. Use dev_dbg instead of pr_debug. 3. Revise the macro RT5682S_PLLB_SRC_MASK to 0x1 because the mux is only two-source. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 3 +-- sound/soc/codecs/rt5682s.h | 2 +- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index d3e965b2e707..122adeb7beb1 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -72,7 +72,6 @@ static void rt5682s_apply_patch_list(struct rt5682s_priv *rt5682s, } const struct reg_default rt5682s_reg[] = { - {0x0000, 0x0001}, {0x0002, 0x8080}, {0x0003, 0x0001}, {0x0005, 0x0000}, @@ -769,7 +768,7 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ count++; } while (val == 0 && count < 50); - pr_debug("%s, val=%d, count=%d\n", __func__, val, count); + dev_dbg(component->dev, "%s, val=%d, count=%d\n", __func__, val, count); switch (val) { case 0x1: diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 7c755e5efb81..55f1febb81e9 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1094,7 +1094,7 @@ #define RT5682S_PLLA_K_BP_SFT 6 /* PLL M/N/K Code Control 7 (0x009e) */ -#define RT5682S_PLLB_SRC_MASK (0x3 << 0) +#define RT5682S_PLLB_SRC_MASK (0x1) #define RT5682S_PLLB_SRC_DFIN (0x1) #define RT5682S_PLLB_SRC_PLLA (0x0) -- 2.17.1

2 1

[PATCH alsa-ucm-conf 0/5] bytcr-rt5640: Fixes and improvements
by Hans de Goede 29 Sep '21

29 Sep '21

Hi Jaroslav, Here is a patch series with 2 bug-fixes for X86 devices with a rt5640 codec (these 2 are a resend) + 3 patches adding support for the second headset-jack (on the dock) found on the HP elitepad 1000 G2. The new kernel-bits for the second headset-jack have already landed, so this is ready for merging. Regards, Hans

2 6

[PATCH alsa-ucm-conf 1/2] bytcr-rt5640: Fix 'Headphone Switch' / 'Headset Mic Switch' no longer getting set
by Hans de Goede 29 Sep '21

29 Sep '21

Commit 446d0a71b47f ("codecs/rt5640: Move out BayTrail-specific pin switches") moved the setting of the 'Headphone Switch' / 'Headset Mic Switch' into "If.hp" resp "If.hsmic" blocks. But instead of making them part of the True {} block inside that If, the statements where added add the top-level of the If {} block where they are no-ops. Move them to inside the True blocks so that these Switches again get properly turned on/off. This fixes the Headphones and Headset-mic no longer working on bytcr-rt5640 devices. Cc: Dmitry Osipenko <digetx(a)gmail.com> Fixes: 446d0a71b47f ("codecs/rt5640: Move out BayTrail-specific pin switches") Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/bytcr-rt5640/HiFi.conf | 36 ++++++++++++++++++++---------------- 1 file changed, 20 insertions(+), 16 deletions(-) diff --git a/ucm2/bytcr-rt5640/HiFi.conf b/ucm2/bytcr-rt5640/HiFi.conf index a5b47df..08127e0 100644 --- a/ucm2/bytcr-rt5640/HiFi.conf +++ b/ucm2/bytcr-rt5640/HiFi.conf @@ -100,16 +100,18 @@ If.mono { If.hp { Condition { Type String Empty "" } - True.Include.hs.File "/codecs/rt5640/HeadPhones.conf" + True { + Include.hs.File "/codecs/rt5640/HeadPhones.conf" - SectionDevice."Headphones" { - EnableSequence [ - cset "name='Headphone Switch' on" - ] + SectionDevice."Headphones" { + EnableSequence [ + cset "name='Headphone Switch' on" + ] - DisableSequence [ - cset "name='Headphone Switch' off" - ] + DisableSequence [ + cset "name='Headphone Switch' off" + ] + } } } @@ -178,15 +180,17 @@ If.in3 { If.hsmic { Condition { Type String Empty "" } - True.Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" + True { + Include.hsmic.File "/codecs/rt5640/HeadsetMic.conf" - SectionDevice."Headset" { - EnableSequence [ - cset "name='Headset Mic Switch' on" - ] + SectionDevice."Headset" { + EnableSequence [ + cset "name='Headset Mic Switch' on" + ] - DisableSequence [ - cset "name='Headset Mic Switch' off" - ] + DisableSequence [ + cset "name='Headset Mic Switch' off" + ] + } } } -- 2.31.1

2 3

[PATCH] ALSA: usb-audio: Fix a missing error check in scarlett gen2 mixer
by Takashi Iwai 29 Sep '21

29 Sep '21

The check of the returned error code is missing in scarlett2_update_monitor_other(). Let's fix it. Fixes: d5bda7e03982 ("ALSA: usb-audio: scarlett2: Add support for the talkback feature") Reported-by: kernel test robot <lkp(a)intel.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/202109131831.9IodEzRx-lkp@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_scarlett_gen2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 3d5848d5481b..53ebabf42472 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -2450,6 +2450,8 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer) err = scarlett2_usb_get_config(mixer, SCARLETT2_CONFIG_TALKBACK_MAP, 1, &bitmap); + if (err < 0) + return err; for (i = 0; i < num_mixes; i++, bitmap >>= 1) private->talkback_map[i] = bitmap & 1; } -- 2.26.2

1 0

[PATCH] ALSA: hda: Reduce udelay() at SKL+ position reporting
by Takashi Iwai 29 Sep '21

29 Sep '21

The position reporting on Intel Skylake and later chips via azx_get_pos_skl() contains a udelay(20) call for the capture streams. A call for this alone doesn't sound too harmful. However, as the pointer PCM ops is one of the hottest path in the PCM operations -- especially for the timer-scheduled operations like PulseAudio -- such a delay hogs CPU usage significantly in the total performance. The function there was taken from the original code in ASoC SST Skylake driver blindly. The udelay() is a workaround for the case where the reported position is behind the period boundary at the timing triggered from interrupts; applications often expect that the full data is available for the whole period when returned (and also that's the definition of the ALSA PCM period). OTOH, HD-audio (legacy) driver has already some workarounds for the delayed position reporting due to its relatively large FIFO, such as the BDL position adjustment and the delayed period-elapsed call in the work. That said, the udelay() is almost superfluous for HD-audio driver unlike SST, and we can drop the udelay(). Though, the current code doesn't guarantee the full period readiness as mentioned in the above, but rather it checks the wallclock and detects the unexpected jump. That's one missing piece, and the drop of udelay() needs a bit more sanity checks for the delayed handling. This patch implements those: the drop of udelay() call in azx_get_pos_skl() and the more proper check of hwptr in azx_position_ok(). The latter change is applied only for the case where the stream is running in the normal mode without no_period_wakeup flag. When no_period_wakeup is set, it essentially ignores the period handling and rather concentrates only on the current position; which implies that we don't need to care about the period boundary at all. Fixes: f87e7f25893d ("ALSA: hda - Improved position reporting on SKL+") Reported-by: Jens Axboe <axboe(a)kernel.dk> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3aa432d814a2..faeeeb923d5e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -637,13 +637,17 @@ static int azx_position_check(struct azx *chip, struct azx_dev *azx_dev) * the update-IRQ timing. The IRQ is issued before actually the * data is processed. So, we need to process it afterwords in a * workqueue. + * + * Returns 1 if OK to proceed, 0 for delay handling, -1 for skipping update */ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) { struct snd_pcm_substream *substream = azx_dev->core.substream; + struct snd_pcm_runtime *runtime = substream->runtime; int stream = substream->stream; u32 wallclk; unsigned int pos; + snd_pcm_uframes_t hwptr, target; wallclk = azx_readl(chip, WALLCLK) - azx_dev->core.start_wallclk; if (wallclk < (azx_dev->core.period_wallclk * 2) / 3) @@ -680,6 +684,24 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) /* NG - it's below the first next period boundary */ return chip->bdl_pos_adj ? 0 : -1; azx_dev->core.start_wallclk += wallclk; + + if (azx_dev->core.no_period_wakeup) + return 1; /* OK, no need to check period boundary */ + + if (runtime->hw_ptr_base != runtime->hw_ptr_interrupt) + return 1; /* OK, already in hwptr updating process */ + + /* check whether the period gets really elapsed */ + pos = bytes_to_frames(runtime, pos); + hwptr = runtime->hw_ptr_base + pos; + if (hwptr < runtime->status->hw_ptr) + hwptr += runtime->buffer_size; + target = runtime->hw_ptr_interrupt + runtime->period_size; + if (hwptr < target) { + /* too early wakeup, process it later */ + return chip->bdl_pos_adj ? 0 : -1; + } + return 1; /* OK, it's fine */ } @@ -874,11 +896,7 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev) if (azx_dev->core.substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return azx_skl_get_dpib_pos(chip, azx_dev); - /* For capture, we need to read posbuf, but it requires a delay - * for the possible boundary overlap; the read of DPIB fetches the - * actual posbuf - */ - udelay(20); + /* read of DPIB fetches the actual posbuf */ azx_skl_get_dpib_pos(chip, azx_dev); return azx_get_pos_posbuf(chip, azx_dev); } -- 2.26.2

3 7

[PATCH] ASoC: SOF: add error handling to snd_sof_ipc_msg_data()
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> If an invalid stream is passed to snd_sof_ipc_msg_data() it won't fill the provided object with data. The caller has to be able to recognise such cases to avoid handling invalid data. Make the function return an error when failing. Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Reviewed-by: Peter Ujfalusi <peter.ujfalusi(a)intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/imx/imx8.c | 7 +++--- sound/soc/sof/imx/imx8m.c | 7 +++--- sound/soc/sof/intel/hda-ipc.c | 15 +++++++----- sound/soc/sof/intel/hda.h | 6 ++--- sound/soc/sof/intel/intel-ipc.c | 14 +++++++---- sound/soc/sof/ipc.c | 41 +++++++++++++++++++++++++-------- sound/soc/sof/ops.h | 8 +++---- sound/soc/sof/sof-priv.h | 12 +++++----- 8 files changed, 71 insertions(+), 39 deletions(-) diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 82ca920a69ab..e6eed53f8644 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -375,11 +375,12 @@ static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type) } } -static void imx8_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +static int imx8_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); + return 0; } static int imx8_ipc_pcm_params(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 76f4ffd9d540..5579cbef4a93 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -238,11 +238,12 @@ static int imx8m_get_bar_index(struct snd_sof_dev *sdev, u32 type) } } -static void imx8m_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +static int imx8m_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); + return 0; } static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index acfeca42604c..11f20a5a62df 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -253,9 +253,9 @@ int hda_dsp_ipc_get_window_offset(struct snd_sof_dev *sdev, u32 id) return SRAM_WINDOW_OFFSET(id); } -void hda_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +int hda_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { if (!substream || !sdev->stream_box.size) { sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); @@ -268,10 +268,13 @@ void hda_ipc_msg_data(struct snd_sof_dev *sdev, hda_stream.hstream); /* The stream might already be closed */ - if (hstream) - sof_mailbox_read(sdev, hda_stream->stream.posn_offset, - p, sz); + if (!hstream) + return -ESTRPIPE; + + sof_mailbox_read(sdev, hda_stream->stream.posn_offset, p, sz); } + + return 0; } int hda_ipc_pcm_params(struct snd_sof_dev *sdev, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 087fa06d5210..cb5a1b17f153 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -563,9 +563,9 @@ int hda_dsp_stream_spib_config(struct snd_sof_dev *sdev, struct hdac_ext_stream *stream, int enable, u32 size); -void hda_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz); +int hda_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz); int hda_ipc_pcm_params(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, const struct sof_ipc_pcm_params_reply *reply); diff --git a/sound/soc/sof/intel/intel-ipc.c b/sound/soc/sof/intel/intel-ipc.c index de66f8a82a07..df37187c8427 100644 --- a/sound/soc/sof/intel/intel-ipc.c +++ b/sound/soc/sof/intel/intel-ipc.c @@ -25,9 +25,9 @@ struct intel_stream { }; /* Mailbox-based Intel IPC implementation */ -void intel_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +int intel_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { if (!substream || !sdev->stream_box.size) { sof_mailbox_read(sdev, sdev->dsp_box.offset, p, sz); @@ -35,9 +35,13 @@ void intel_ipc_msg_data(struct snd_sof_dev *sdev, struct intel_stream *stream = substream->runtime->private_data; /* The stream might already be closed */ - if (stream) - sof_mailbox_read(sdev, stream->posn_offset, p, sz); + if (!stream) + return -ESTRPIPE; + + sof_mailbox_read(sdev, stream->posn_offset, p, sz); } + + return 0; } EXPORT_SYMBOL_NS(intel_ipc_msg_data, SND_SOC_SOF_INTEL_HIFI_EP_IPC); diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 5d41924f37a6..152d36a6253d 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -394,6 +394,7 @@ static void ipc_comp_notification(struct snd_sof_dev *sdev, { u32 msg_type = hdr->cmd & SOF_CMD_TYPE_MASK; struct sof_ipc_ctrl_data *cdata; + int ret; switch (msg_type) { case SOF_IPC_COMP_GET_VALUE: @@ -403,7 +404,12 @@ static void ipc_comp_notification(struct snd_sof_dev *sdev, return; /* read back full message */ - snd_sof_ipc_msg_data(sdev, NULL, cdata, hdr->size); + ret = snd_sof_ipc_msg_data(sdev, NULL, cdata, hdr->size); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to read component event: %d\n", ret); + goto err; + } break; default: dev_err(sdev->dev, "error: unhandled component message %#x\n", msg_type); @@ -412,6 +418,7 @@ static void ipc_comp_notification(struct snd_sof_dev *sdev, snd_sof_control_notify(sdev, cdata); +err: kfree(cdata); } @@ -420,10 +427,14 @@ void snd_sof_ipc_msgs_rx(struct snd_sof_dev *sdev) { struct sof_ipc_cmd_hdr hdr; u32 cmd, type; - int err = 0; + int err; /* read back header */ - snd_sof_ipc_msg_data(sdev, NULL, &hdr, sizeof(hdr)); + err = snd_sof_ipc_msg_data(sdev, NULL, &hdr, sizeof(hdr)); + if (err < 0) { + dev_warn(sdev->dev, "failed to read IPC header: %d\n", err); + return; + } ipc_log_header(sdev->dev, "ipc rx", hdr.cmd); cmd = hdr.cmd & SOF_GLB_TYPE_MASK; @@ -477,12 +488,16 @@ EXPORT_SYMBOL(snd_sof_ipc_msgs_rx); static void ipc_trace_message(struct snd_sof_dev *sdev, u32 msg_type) { struct sof_ipc_dma_trace_posn posn; + int ret; switch (msg_type) { case SOF_IPC_TRACE_DMA_POSITION: /* read back full message */ - snd_sof_ipc_msg_data(sdev, NULL, &posn, sizeof(posn)); - snd_sof_trace_update_pos(sdev, &posn); + ret = snd_sof_ipc_msg_data(sdev, NULL, &posn, sizeof(posn)); + if (ret < 0) + dev_warn(sdev->dev, "failed to read trace position: %d\n", ret); + else + snd_sof_trace_update_pos(sdev, &posn); break; default: dev_err(sdev->dev, "error: unhandled trace message %#x\n", msg_type); @@ -500,7 +515,7 @@ static void ipc_period_elapsed(struct snd_sof_dev *sdev, u32 msg_id) struct snd_sof_pcm_stream *stream; struct sof_ipc_stream_posn posn; struct snd_sof_pcm *spcm; - int direction; + int direction, ret; spcm = snd_sof_find_spcm_comp(scomp, msg_id, &direction); if (!spcm) { @@ -511,7 +526,11 @@ static void ipc_period_elapsed(struct snd_sof_dev *sdev, u32 msg_id) } stream = &spcm->stream[direction]; - snd_sof_ipc_msg_data(sdev, stream->substream, &posn, sizeof(posn)); + ret = snd_sof_ipc_msg_data(sdev, stream->substream, &posn, sizeof(posn)); + if (ret < 0) { + dev_warn(sdev->dev, "failed to read stream position: %d\n", ret); + return; + } dev_vdbg(sdev->dev, "posn : host 0x%llx dai 0x%llx wall 0x%llx\n", posn.host_posn, posn.dai_posn, posn.wallclock); @@ -530,7 +549,7 @@ static void ipc_xrun(struct snd_sof_dev *sdev, u32 msg_id) struct snd_sof_pcm_stream *stream; struct sof_ipc_stream_posn posn; struct snd_sof_pcm *spcm; - int direction; + int direction, ret; spcm = snd_sof_find_spcm_comp(scomp, msg_id, &direction); if (!spcm) { @@ -540,7 +559,11 @@ static void ipc_xrun(struct snd_sof_dev *sdev, u32 msg_id) } stream = &spcm->stream[direction]; - snd_sof_ipc_msg_data(sdev, stream->substream, &posn, sizeof(posn)); + ret = snd_sof_ipc_msg_data(sdev, stream->substream, &posn, sizeof(posn)); + if (ret < 0) { + dev_warn(sdev->dev, "failed to read overrun position: %d\n", ret); + return; + } dev_dbg(sdev->dev, "posn XRUN: host %llx comp %d size %d\n", posn.host_posn, posn.xrun_comp_id, posn.xrun_size); diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index d40ce2581d94..a93aa5b943b2 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -422,11 +422,11 @@ static inline int snd_sof_load_firmware(struct snd_sof_dev *sdev) } /* host DSP message data */ -static inline void snd_sof_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz) +static inline int snd_sof_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz) { - sof_ops(sdev)->ipc_msg_data(sdev, substream, p, sz); + return sof_ops(sdev)->ipc_msg_data(sdev, substream, p, sz); } /* host configure DSP HW parameters */ diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 6b1dbae3344c..a6ded5bd0674 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -210,9 +210,9 @@ struct snd_sof_dsp_ops { #endif /* host read DSP stream data */ - void (*ipc_msg_data)(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz); /* mandatory */ + int (*ipc_msg_data)(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz); /* mandatory */ /* host configure DSP HW parameters */ int (*ipc_pcm_params)(struct snd_sof_dev *sdev, @@ -567,9 +567,9 @@ int sof_block_read(struct snd_sof_dev *sdev, enum snd_sof_fw_blk_type blk_type, int sof_fw_ready(struct snd_sof_dev *sdev, u32 msg_id); -void intel_ipc_msg_data(struct snd_sof_dev *sdev, - struct snd_pcm_substream *substream, - void *p, size_t sz); +int intel_ipc_msg_data(struct snd_sof_dev *sdev, + struct snd_pcm_substream *substream, + void *p, size_t sz); int intel_ipc_pcm_params(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, const struct sof_ipc_pcm_params_reply *reply); -- 2.33.0

2 1

[PATCH] ASoC: SOF: prefix some terse and cryptic dev_dbg() with __func__
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Marc Herbert <marc.herbert(a)intel.com> These helped troubleshoot some DMA issue in SOF. Signed-off-by: Marc Herbert <marc.herbert(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 7 ++++--- sound/soc/sof/trace.c | 5 +++-- 2 files changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..ab051f67e0e8 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -89,13 +89,13 @@ int hda_dsp_stream_setup_bdl(struct snd_sof_dev *sdev, int remain, ioc; period_bytes = stream->period_bytes; - dev_dbg(sdev->dev, "period_bytes:0x%x\n", period_bytes); + dev_dbg(sdev->dev, "%s: period_bytes:0x%x\n", __func__, period_bytes); if (!period_bytes) period_bytes = stream->bufsize; periods = stream->bufsize / period_bytes; - dev_dbg(sdev->dev, "periods:%d\n", periods); + dev_dbg(sdev->dev, "%s: periods:%d\n", __func__, periods); remain = stream->bufsize % period_bytes; if (remain) @@ -244,7 +244,8 @@ int hda_dsp_stream_put(struct snd_sof_dev *sdev, int direction, int stream_tag) HDA_VS_INTEL_EM2_L1SEN, HDA_VS_INTEL_EM2_L1SEN); if (!found) { - dev_dbg(sdev->dev, "stream_tag %d not opened!\n", stream_tag); + dev_dbg(sdev->dev, "%s: stream_tag %d not opened!\n", + __func__, stream_tag); return -ENODEV; } diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c index 58f6ca5cf491..e3afc3dac7d1 100644 --- a/sound/soc/sof/trace.c +++ b/sound/soc/sof/trace.c @@ -417,7 +417,7 @@ int snd_sof_init_trace_ipc(struct snd_sof_dev *sdev) "error: fail in snd_sof_dma_trace_init %d\n", ret); return ret; } - dev_dbg(sdev->dev, "stream_tag: %d\n", params.stream_tag); + dev_dbg(sdev->dev, "%s: stream_tag: %d\n", __func__, params.stream_tag); /* send IPC to the DSP */ ret = sof_ipc_tx_message(sdev->ipc, @@ -480,7 +480,8 @@ int snd_sof_init_trace(struct snd_sof_dev *sdev) goto table_err; sdev->dma_trace_pages = ret; - dev_dbg(sdev->dev, "dma_trace_pages: %d\n", sdev->dma_trace_pages); + dev_dbg(sdev->dev, "%s: dma_trace_pages: %d\n", + __func__, sdev->dma_trace_pages); if (sdev->first_boot) { ret = trace_debugfs_create(sdev); -- 2.33.0

2 1

[PATCH] ASoC: intel: sof_rt5682: update platform device name for Maxim amplifier
by Brent Lu 28 Sep '21

28 Sep '21

To follow 20-character length limitation of platform device name, we have only 7 character space for amplifier. Therefore, the last character of mx98357a and mx98360a is removed to save space. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/sof_rt5682.c | 8 ++++---- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 4 ++-- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 2 +- 4 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3f6f19d9b19e..ad42d4c7ade5 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -959,7 +959,7 @@ static const struct platform_device_id board_ids[] = { .name = "sof_rt5682", }, { - .name = "tgl_mx98357a_rt5682", + .name = "tgl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | @@ -989,7 +989,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "jsl_rt5682_mx98360a", + .name = "jsl_rt5682_mx98360", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | @@ -1039,7 +1039,7 @@ static const struct platform_device_id board_ids[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, { - .name = "adl_mx98357a_rt5682", + .name = "adl_mx98357_rt5682", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_SSP_CODEC(0) | SOF_SPEAKER_AMP_PRESENT | @@ -1067,7 +1067,7 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1)), }, { - .name = "jsl_rt5682s_mx98360a", + .name = "jsl_rt5682s_mx98360", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | SOF_RT5682_MCLK_24MHZ | SOF_RT5682_SSP_CODEC(0) | 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..e4ff280eac23 100644 --- a/sound/soc/intel/common/soc-acpi-intel-adl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-adl-match.c @@ -291,7 +291,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[] = { }, { .id = "10EC5682", - .drv_name = "adl_mx98357a_rt5682", + .drv_name = "adl_mx98357_rt5682", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &adl_max98357a_amp, .sof_fw_filename = "sof-adl.ri", diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index cfd6bf756215..20fd9dcc74af 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -67,7 +67,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { }, { .id = "10EC5682", - .drv_name = "jsl_rt5682_mx98360a", + .drv_name = "jsl_rt5682_mx98360", .sof_fw_filename = "sof-jsl.ri", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, @@ -99,7 +99,7 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { }, { .id = "RTL5682", - .drv_name = "jsl_rt5682s_mx98360a", + .drv_name = "jsl_rt5682s_mx98360", .sof_fw_filename = "sof-jsl.ri", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &mx98360a_spk, diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 785d5f5f8a9c..91ef71c2535d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -333,7 +333,7 @@ static const struct snd_soc_acpi_codecs tgl_rt1011_amp = { struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[] = { { .id = "10EC5682", - .drv_name = "tgl_mx98357a_rt5682", + .drv_name = "tgl_mx98357_rt5682", .machine_quirk = snd_soc_acpi_codec_list, .quirk_data = &tgl_codecs, .sof_fw_filename = "sof-tgl.ri", -- 2.25.1

3 2

[PATCH] ASoC: SOF: ipc: Make the error prints consistent in tx_wait_done()
by Peter Ujfalusi 28 Sep '21

28 Sep '21

If we get an error on reply (msg->reply_error) then we should print the error value out. At the same time extend the print to include the message size as well and do the same in case of a timeout. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 5d41924f37a6..95b1d875c3c0 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -249,15 +249,17 @@ static int tx_wait_done(struct snd_sof_ipc *ipc, struct snd_sof_ipc_msg *msg, msecs_to_jiffies(sdev->ipc_timeout)); if (ret == 0) { - dev_err(sdev->dev, "error: ipc timed out for 0x%x size %d\n", - hdr->cmd, hdr->size); + dev_err(sdev->dev, + "ipc tx timed out for %#x (msg/reply size: %d/%zu)\n", + hdr->cmd, hdr->size, msg->reply_size); snd_sof_handle_fw_exception(ipc->sdev); ret = -ETIMEDOUT; } else { ret = msg->reply_error; if (ret < 0) { - dev_err(sdev->dev, "error: ipc error for 0x%x size %zu\n", - hdr->cmd, msg->reply_size); + dev_err(sdev->dev, + "ipc tx error for %#x (msg/reply size: %d/%zu): %d\n", + hdr->cmd, hdr->size, msg->reply_size, ret); } else { ipc_log_header(sdev->dev, "ipc tx succeeded", hdr->cmd); if (msg->reply_size) -- 2.33.0

2 1

[PATCH] ASoC: mediatek: mt8195: add missing of_node_put in probe
by Trevor Wu 28 Sep '21

28 Sep '21

dp node and hdmi node are retrieved from of_parse_phandle(), so using of_node_put() on them before return. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- .../mt8195/mt8195-mt6359-rt1019-rt5682.c | 25 +++++++++++-------- 1 file changed, 15 insertions(+), 10 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c index a3fa8efc8f81..cb15467e5fc5 100644 --- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c +++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c @@ -995,6 +995,8 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) { struct snd_soc_card *card = &mt8195_mt6359_rt1019_rt5682_soc_card; struct device_node *platform_node; + struct device_node *dp_node; + struct device_node *hdmi_node; struct snd_soc_dai_link *dai_link; struct mt8195_mt6359_rt1019_rt5682_priv *priv = NULL; int ret, i; @@ -1013,12 +1015,12 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) dai_link->platforms->of_node = platform_node; if (strcmp(dai_link->name, "DPTX_BE") == 0) { - dai_link->codecs->of_node = - of_parse_phandle(pdev->dev.of_node, - "mediatek,dptx-codec", 0); - if (!dai_link->codecs->of_node) { + dp_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,dptx-codec", 0); + if (!dp_node) { dev_dbg(&pdev->dev, "No property 'dptx-codec'\n"); } else { + dai_link->codecs->of_node = dp_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_dptx_codec_init; @@ -1026,12 +1028,12 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) } if (strcmp(dai_link->name, "ETDM3_OUT_BE") == 0) { - dai_link->codecs->of_node = - of_parse_phandle(pdev->dev.of_node, - "mediatek,hdmi-codec", 0); - if (!dai_link->codecs->of_node) { + hdmi_node = of_parse_phandle(pdev->dev.of_node, + "mediatek,hdmi-codec", 0); + if (!hdmi_node) { dev_dbg(&pdev->dev, "No property 'hdmi-codec'\n"); } else { + dai_link->codecs->of_node = hdmi_node; dai_link->codecs->name = NULL; dai_link->codecs->dai_name = "i2s-hifi"; dai_link->init = mt8195_hdmi_codec_init; @@ -1041,8 +1043,8 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); if (!priv) { - of_node_put(platform_node); - return -ENOMEM; + ret = -ENOMEM; + goto out; } snd_soc_card_set_drvdata(card, priv); @@ -1052,6 +1054,9 @@ static int mt8195_mt6359_rt1019_rt5682_dev_probe(struct platform_device *pdev) dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", __func__, ret); +out: + of_node_put(hdmi_node); + of_node_put(dp_node); of_node_put(platform_node); return ret; } -- 2.18.0

2 1

[PATCH] ASoC: SOF: pm: fix a stale comment
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> There is no restore_stream flag anymmore. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <bard.liao(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/pm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/pm.c b/sound/soc/sof/pm.c index c83fb6255961..89b4b3d76539 100644 --- a/sound/soc/sof/pm.c +++ b/sound/soc/sof/pm.c @@ -191,7 +191,7 @@ static int sof_suspend(struct device *dev, bool runtime_suspend) if (sdev->fw_state != SOF_FW_BOOT_COMPLETE) goto suspend; - /* set restore_stream for all streams during system suspend */ + /* prepare for streams to be resumed properly upon resume */ if (!runtime_suspend) { ret = sof_set_hw_params_upon_resume(sdev->dev); if (ret < 0) { -- 2.33.0

2 1

[PATCH] ASoC: SOF: Intel: hda-stream: Print stream name on STREAM_SD_OFFSET timeout
by Peter Ujfalusi 28 Sep '21

28 Sep '21

In order to provide more information in case of timeout observed while reading STREAM_SD_OFFSET, print out the stream name or in case there is no audio stream associated (like dma-trace), print "--" Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/sof/intel/hda-stream.c | 85 ++++++++++++++++++++++---------- 1 file changed, 59 insertions(+), 26 deletions(-) diff --git a/sound/soc/sof/intel/hda-stream.c b/sound/soc/sof/intel/hda-stream.c index 63c367478f1c..db028b83c5f2 100644 --- a/sound/soc/sof/intel/hda-stream.c +++ b/sound/soc/sof/intel/hda-stream.c @@ -25,6 +25,33 @@ #define HDA_LTRP_GB_VALUE_US 95 +static inline const char *hda_hstream_direction_str(struct hdac_stream *hstream) +{ + if (hstream->direction == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + +static char *hda_hstream_dbg_get_stream_info_str(struct hdac_stream *hstream) +{ + struct snd_soc_pcm_runtime *rtd; + + if (hstream->substream) + rtd = asoc_substream_to_rtd(hstream->substream); + else if (hstream->cstream) + rtd = hstream->cstream->private_data; + else + /* Non audio DMA user, like dma-trace */ + return kasprintf(GFP_KERNEL, "-- (%s, stream_tag: %u)", + hda_hstream_direction_str(hstream), + hstream->stream_tag); + + return kasprintf(GFP_KERNEL, "dai_link \"%s\" (%s, stream_tag: %u)", + rtd->dai_link->name, hda_hstream_direction_str(hstream), + hstream->stream_tag); +} + /* * set up one of BDL entries for a stream */ @@ -257,7 +284,7 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, struct hdac_stream *hstream = &stream->hstream; int sd_offset = SOF_STREAM_SD_OFFSET(hstream); u32 dma_start = SOF_HDA_SD_CTL_DMA_START; - int ret; + int ret = 0; u32 run; /* cmd must be for audio stream */ @@ -283,14 +310,9 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_STREAM_RUN_TIMEOUT); - if (ret < 0) { - dev_err(sdev->dev, - "error: %s: cmd %d: timeout on STREAM_SD_OFFSET read\n", - __func__, cmd); - return ret; - } + if (ret >= 0) + hstream->running = true; - hstream->running = true; break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: @@ -306,27 +328,32 @@ int hda_dsp_stream_trigger(struct snd_sof_dev *sdev, HDA_DSP_REG_POLL_INTERVAL_US, HDA_DSP_STREAM_RUN_TIMEOUT); - if (ret < 0) { - dev_err(sdev->dev, - "error: %s: cmd %d: timeout on STREAM_SD_OFFSET read\n", - __func__, cmd); - return ret; - } + if (ret >= 0) { + snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, + sd_offset + SOF_HDA_ADSP_REG_CL_SD_STS, + SOF_HDA_CL_DMA_SD_INT_MASK); - snd_sof_dsp_write(sdev, HDA_DSP_HDA_BAR, sd_offset + - SOF_HDA_ADSP_REG_CL_SD_STS, - SOF_HDA_CL_DMA_SD_INT_MASK); - - hstream->running = false; - snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, - 1 << hstream->index, 0x0); + hstream->running = false; + snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, + SOF_HDA_INTCTL, + 1 << hstream->index, 0x0); + } break; default: dev_err(sdev->dev, "error: unknown command: %d\n", cmd); return -EINVAL; } - return 0; + if (ret < 0) { + char *stream_name = hda_hstream_dbg_get_stream_info_str(hstream); + + dev_err(sdev->dev, + "%s: cmd %d on %s: timeout on STREAM_SD_OFFSET read\n", + __func__, cmd, stream_name ? stream_name : "unknown stream"); + kfree(stream_name); + } + + return ret; } /* minimal recommended programming for ICCMAX stream */ @@ -440,9 +467,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, HDA_DSP_STREAM_RUN_TIMEOUT); if (ret < 0) { + char *stream_name = hda_hstream_dbg_get_stream_info_str(hstream); + dev_err(sdev->dev, - "error: %s: timeout on STREAM_SD_OFFSET read1\n", - __func__); + "%s: on %s: timeout on STREAM_SD_OFFSET read1\n", + __func__, stream_name ? stream_name : "unknown stream"); + kfree(stream_name); return ret; } @@ -506,9 +536,12 @@ int hda_dsp_stream_hw_params(struct snd_sof_dev *sdev, HDA_DSP_STREAM_RUN_TIMEOUT); if (ret < 0) { + char *stream_name = hda_hstream_dbg_get_stream_info_str(hstream); + dev_err(sdev->dev, - "error: %s: timeout on STREAM_SD_OFFSET read2\n", - __func__); + "%s: on %s: timeout on STREAM_SD_OFFSET read1\n", + __func__, stream_name ? stream_name : "unknown stream"); + kfree(stream_name); return ret; } -- 2.33.0

2 1

[PATCH] ASoC: SOF: imx: add header file for ops
by Peter Ujfalusi 28 Sep '21

28 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Sparse (make C=2) complains about undeclared variables. Fix by adding a real prototype instead of 'extern' in sof-of-dev.c Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)gmail.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/imx/imx-ops.h | 10 ++++++++++ sound/soc/sof/imx/imx8.c | 1 + sound/soc/sof/imx/imx8m.c | 1 + sound/soc/sof/sof-of-dev.c | 5 +---- 4 files changed, 13 insertions(+), 4 deletions(-) create mode 100644 sound/soc/sof/imx/imx-ops.h diff --git a/sound/soc/sof/imx/imx-ops.h b/sound/soc/sof/imx/imx-ops.h new file mode 100644 index 000000000000..24235ef8c8fa --- /dev/null +++ b/sound/soc/sof/imx/imx-ops.h @@ -0,0 +1,10 @@ +/* SPDX-License-Identifier: (GPL-2.0-only OR BSD-3-Clause) */ + +#ifndef __IMX_OPS_H__ +#define __IMX_OPS_H__ + +extern struct snd_sof_dsp_ops sof_imx8_ops; +extern struct snd_sof_dsp_ops sof_imx8x_ops; +extern struct snd_sof_dsp_ops sof_imx8m_ops; + +#endif diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 82ca920a69ab..cf27ee66b577 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -22,6 +22,7 @@ #include <dt-bindings/firmware/imx/rsrc.h> #include "../ops.h" #include "imx-common.h" +#include "imx-ops.h" /* DSP memories */ #define IRAM_OFFSET 0x10000 diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 76f4ffd9d540..932c0e73a8b0 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -18,6 +18,7 @@ #include "../ops.h" #include "imx-common.h" +#include "imx-ops.h" #define MBOX_OFFSET 0x800000 #define MBOX_SIZE 0x1000 diff --git a/sound/soc/sof/sof-of-dev.c b/sound/soc/sof/sof-of-dev.c index d1a21edfa05d..f0f819a46456 100644 --- a/sound/soc/sof/sof-of-dev.c +++ b/sound/soc/sof/sof-of-dev.c @@ -11,10 +11,7 @@ #include <sound/sof.h> #include "ops.h" - -extern struct snd_sof_dsp_ops sof_imx8_ops; -extern struct snd_sof_dsp_ops sof_imx8x_ops; -extern struct snd_sof_dsp_ops sof_imx8m_ops; +#include "imx/imx-ops.h" /* platform specific devices */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_IMX8) -- 2.33.0

2 1

[PATCH v8 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 28 Sep '21

28 Sep '21

Many thanks for reviewing v7 This version addresses all the comments raised as part of v8 review. 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 simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | 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/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v7: - removed returning errors in dsp command callbacks as it has no effect - removed uncessary err prints - removed default topology file as its impossible to have one for all boards. - removed references of Elite in commit log - return correct error codes returned from functions. Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h 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-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings 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 Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai 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 | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1135 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 823 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1113 ++++++++++++++++ 31 files changed, 7170 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-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-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.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-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-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

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[patch 00/11] hrtimers: Cleanup hrtimer_forward() [ab]use
by Thomas Gleixner 28 Sep '21

28 Sep '21

A recent syzbot report unearthed abuse of hrtimer_forward() which can cause runaway timers hogging the CPU in timer expiry context by rearming the timer in the past over and over. This happens when the caller uses timer->expiry for the 'now' argument of hrtimer_forward(). That works as long as the timer expiry is on time, but can cause a long period of rearm/fire loops which hog the CPU. Expiring late can have various causes, but obviously virtualization is prone to that due to VCPU scheduling. The correct usage of hrtimer_forward() is to hand the current time to the 'now' argument which ensures that the next event on the periodic time line is past now. This is what hrtimer_forward_now() provides. The following series addresses this: 1) Add a debug mechanism to the hrtimer expiry loop 2) Convert all hrtimer_forward() usage outside of kernel/time/ to use hrtimer_forward_now(). 3) Confine hrtimer_forward() to kernel/time/ core code. The mac80211_hwsim patch has already been picked up by the wireless maintainer and all other patches which affect usage outside the core code can be picked up by the relevant subsystems. If a maintainer wants me to pick a particular patch up, please let me know. The last patch which confines hrtimer_forward() will be postponed until all other patches have been merged into Linus tree. The series is also available from git: git://git.kernel.org/pub/scm/linux/kernel/git/tglx/devel.git hrtimer Thanks, tglx --- drivers/gpu/drm/i915/i915_pmu.c | 2 - drivers/net/wireless/mac80211_hwsim.c | 4 +- drivers/net/wwan/iosm/iosm_ipc_imem.c | 4 +- drivers/power/reset/ltc2952-poweroff.c | 4 -- include/linux/hrtimer.h | 26 ----------------- include/linux/posix-timers.h | 3 ++ kernel/signal.c | 14 +-------- kernel/time/hrtimer.c | 48 ++++++++++++++++++++++++++++++++- kernel/time/itimer.c | 13 ++++++++ kernel/time/posix-timers.c | 42 +++++++++++----------------- kernel/time/tick-internal.h | 1 net/can/bcm.c | 2 - sound/drivers/pcsp/pcsp_lib.c | 2 - 13 files changed, 92 insertions(+), 73 deletions(-)

2 2

[PATCH] snd-usb-audio: fix comment reference in __uac_clock_find_source
by Geraldo Nascimento 28 Sep '21

28 Sep '21

snd_usb_find_clock_source and snd_usb_find_clock_selector are helper macros that look at an entity id and validate that this entity id is in fact a clock source or a clock selector. The present comments inside __uac_clock_find_source give the reader the impression we're looking for an entity id. We're looking for an entity id indeed, the clock source, but since __uac_clock_find_source is recursive, we're also looking *at* the entity ids, in the search for the one clock source. Fix the comment so we don't give readers a wrong idea. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 81d5ce07d548..7dd71d342443 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -271,7 +271,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, return -EINVAL; } - /* first, see if the ID we're looking for is a clock source already */ + /* first, see if the ID we're looking at is a clock source already */ source = snd_usb_find_clock_source(chip, entity_id, proto); if (source) { entity_id = GET_VAL(source, proto, bClockID); @@ -297,7 +297,7 @@ static int __uac_clock_find_source(struct snd_usb_audio *chip, goto find_source; } - /* the entity ID we are looking for is a selector. + /* the entity ID we are looking at is a selector. * find out what it currently selects */ ret = uac_clock_selector_get_val(chip, clock_id); if (ret < 0) {

2 1

[RFC PATCH 0/4] ALSA: hda: potential hdac_stream locking issues?
by Pierre-Louis Bossart 28 Sep '21

28 Sep '21

While reviewing the HDAudio DMA handling, I found a number of inconsistencies in how spin_locks are used. It's not clear what the HDaudio bus->reg_lock is supposed to protect. In most cases only the writes to specific boolean status flags are protected, and there are multiple cases of taking the lock after testing a status flag. This patchset suggests a more consistent locking pattern, but it's entirely possible that the bus->reg_lock is only intented to protect register read/write access on the HDaudio bus, and not the status flags, and that this entire piece of code is completely over-engineered. On the Intel side no one knows why this spinlock was used, the reasons are lost to history. I set the 'RFC' status on purpose in the hope that Takashi might recall what this lock is supposed to protect. The diff format makes this patchset difficult to review, it's recommended to apply the patches and look at entire functions with changes to get a better idea of the suggested changes. Pierre-Louis Bossart (4): ALSA: hda: hdac_stream: fix potential locking issue in snd_hdac_stream_assign() ALSA: hda: hdac_stream: reset assigned_key in stream_release() ALSA: hda: hdac_ext_stream: fix potential locking issues ASoC: SOF: Intel: hda-dai: fix potential locking issue include/sound/hdaudio_ext.h | 2 ++ sound/hda/ext/hdac_ext_stream.c | 46 ++++++++++++++++++++------------- sound/hda/hdac_stream.c | 5 ++-- sound/soc/sof/intel/hda-dai.c | 7 ++--- 4 files changed, 37 insertions(+), 23 deletions(-) -- 2.25.1

3 7

[RESEND PATCH v4] ASoC: dt-bindings: mediatek: mt8192: re-add audio afe document
by Tzung-Bi Shih 28 Sep '21

28 Sep '21

The document was merged as commit 1afc60e00de3 ("dt-bindings: mediatek: mt8192: add audio afe document"). However, [1] revealed that the commit 1afc60e00de3 breaks dt_binding_check due to dt-bindings/clock/mt8192-clk.h doesn't exist. As a temporary fix, commit 7d94ca3c8acd ("ASoC: mt8192: revert add audio afe document") reverted commit 1afc60e00de3. dt-bindings/clock/mt8192-clk.h is in mainline per commit f35f1a23e0e1 ("clk: mediatek: Add dt-bindings of MT8192 clocks"). Re-adds the document back. [1]: https://mailman.alsa-project.org/pipermail/alsa-devel/2020-November/176873.… Reviewed-by: Rob Herring <robh(a)kernel.org> Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../bindings/sound/mt8192-afe-pcm.yaml | 100 ++++++++++++++++++ 1 file changed, 100 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml diff --git a/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml new file mode 100644 index 000000000000..5e9fe067f440 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/mt8192-afe-pcm.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/mt8192-afe-pcm.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Mediatek AFE PCM controller for mt8192 + +maintainers: + - Jiaxin Yu <jiaxin.yu(a)mediatek.com> + - Shane Chien <shane.chien(a)mediatek.com> + +properties: + compatible: + const: mediatek,mt8192-audio + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + reset-names: + const: audiosys + + mediatek,apmixedsys: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek apmixedsys controller + + mediatek,infracfg: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek infracfg controller + + mediatek,topckgen: + $ref: "/schemas/types.yaml#/definitions/phandle" + description: The phandle of the mediatek topckgen controller + + power-domains: + maxItems: 1 + + clocks: + items: + - description: AFE clock + - description: ADDA DAC clock + - description: ADDA DAC pre-distortion clock + - description: audio infra sys clock + - description: audio infra 26M clock + + clock-names: + items: + - const: aud_afe_clk + - const: aud_dac_clk + - const: aud_dac_predis_clk + - const: aud_infra_clk + - const: aud_infra_26m_clk + +required: + - compatible + - interrupts + - resets + - reset-names + - mediatek,apmixedsys + - mediatek,infracfg + - mediatek,topckgen + - power-domains + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/mt8192-clk.h> + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + #include <dt-bindings/power/mt8192-power.h> + #include <dt-bindings/reset-controller/mt8192-resets.h> + + afe: mt8192-afe-pcm { + compatible = "mediatek,mt8192-audio"; + interrupts = <GIC_SPI 202 IRQ_TYPE_LEVEL_HIGH>; + resets = <&watchdog MT8192_TOPRGU_AUDIO_SW_RST>; + reset-names = "audiosys"; + mediatek,apmixedsys = <&apmixedsys>; + mediatek,infracfg = <&infracfg>; + mediatek,topckgen = <&topckgen>; + power-domains = <&scpsys MT8192_POWER_DOMAIN_AUDIO>; + clocks = <&audsys CLK_AUD_AFE>, + <&audsys CLK_AUD_DAC>, + <&audsys CLK_AUD_DAC_PREDIS>, + <&infracfg CLK_INFRA_AUDIO>, + <&infracfg CLK_INFRA_AUDIO_26M_B>; + clock-names = "aud_afe_clk", + "aud_dac_clk", + "aud_dac_predis_clk", + "aud_infra_clk", + "aud_infra_26m_clk"; + }; + +... -- 2.33.0.685.g46640cef36-goog

1 1

[PATCH] ASoC: zl38060: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the zl38060 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/zl38060.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/zl38060.c b/sound/soc/codecs/zl38060.c index d21a72314d37..d20ec1571010 100644 --- a/sound/soc/codecs/zl38060.c +++ b/sound/soc/codecs/zl38060.c @@ -250,8 +250,8 @@ static int zl38_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: /* always 32 bits per frame (= 16 bits/channel, 2 channels) */ err = regmap_update_bits(priv->regmap, REG_TDMA_CFG_CLK, CFG_CLK_MASTER | CFG_CLK_PCLK_MASK, -- 2.20.1

1 1

[PATCH] ASoC: bcm: Convert to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of the effort to remove our old APIs based on outdated terminology update the Broadcom drivers to use modern terminology. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/bcm/bcm2835-i2s.c | 56 ++++++++++++++++++------------------- sound/soc/bcm/cygnus-ssp.c | 6 ++-- 2 files changed, 31 insertions(+), 31 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 3d668f449bc1..e3fc4bee8cfd 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -127,14 +127,14 @@ struct bcm2835_i2s_dev { static void bcm2835_i2s_start_clock(struct bcm2835_i2s_dev *dev) { - unsigned int master = dev->fmt & SND_SOC_DAIFMT_MASTER_MASK; + unsigned int provider = dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; if (dev->clk_prepared) return; - switch (master) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBS_CFM: + switch (provider) { + case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_CBC_CFP: clk_prepare_enable(dev->clk); dev->clk_prepared = true; break; @@ -337,8 +337,8 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, unsigned int rx_mask, tx_mask; unsigned int rx_ch1_pos, rx_ch2_pos, tx_ch1_pos, tx_ch2_pos; unsigned int mode, format; - bool bit_clock_master = false; - bool frame_sync_master = false; + bool bit_clock_provider = false; + bool frame_sync_provider = false; bool frame_start_falling_edge = false; uint32_t csreg; int ret = 0; @@ -383,36 +383,36 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, if (data_length > slot_width) return -EINVAL; - /* Check if CPU is bit clock master */ - switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - bit_clock_master = true; + /* Check if CPU is bit clock provider */ + switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_CBC_CFP: + bit_clock_provider = true; break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBM_CFM: - bit_clock_master = false; + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBP_CFP: + bit_clock_provider = false; break; default: return -EINVAL; } - /* Check if CPU is frame sync master */ - switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - frame_sync_master = true; + /* Check if CPU is frame sync provider */ + switch (dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + case SND_SOC_DAIFMT_CBP_CFC: + frame_sync_provider = true; break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFM: - frame_sync_master = false; + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFP: + frame_sync_provider = false; break; default: return -EINVAL; } /* Clock should only be set up here if CPU is clock master */ - if (bit_clock_master && + if (bit_clock_provider && (!dev->clk_prepared || dev->clk_rate != bclk_rate)) { if (dev->clk_prepared) bcm2835_i2s_stop_clock(dev); @@ -501,11 +501,11 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable - * if bcm2835 is the frame clock master. + * if bcm2835 is the frame clock provider. */ - if ((!rx_ch1_pos || !tx_ch1_pos) && !frame_sync_master) + if ((!rx_ch1_pos || !tx_ch1_pos) && !frame_sync_provider) dev_warn(dev->dev, - "Unstable slave config detected, L/R may be swapped"); + "Unstable consumer config detected, L/R may be swapped"); /* * Set format for both streams. @@ -538,11 +538,11 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, mode |= BCM2835_I2S_FSLEN(framesync_length); /* CLKM selects bcm2835 clock slave mode */ - if (!bit_clock_master) + if (!bit_clock_provider) mode |= BCM2835_I2S_CLKM; /* FSM selects bcm2835 frame sync slave mode */ - if (!frame_sync_master) + if (!frame_sync_provider) mode |= BCM2835_I2S_FSM; /* CLKI selects normal clocking mode, sampling on rising edge */ diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index fca5a3f2eec5..9698f4531c90 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -848,12 +848,12 @@ static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssp_newcfg = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: ssp_newcfg |= BIT(I2S_OUT_CFGX_SLAVE_MODE); aio->is_slave = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: ssp_newcfg &= ~BIT(I2S_OUT_CFGX_SLAVE_MODE); aio->is_slave = 0; break; -- 2.20.1

1 1

[PATCH] ASoC: cpcap: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cpcap driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cpcap.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 05bbacd0d174..9a932d088fb5 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1168,15 +1168,15 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * "HiFi Playback" should always be configured as - * SND_SOC_DAIFMT_CBM_CFM - codec clk & frm master + * SND_SOC_DAIFMT_CBP_CFP - codec clk & frm provider * SND_SOC_DAIFMT_I2S - I2S mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val &= ~BIT(CPCAP_BIT_SMB_ST_DAC); break; default: - dev_err(dev, "HiFi dai fmt failed: CPCAP should be master"); + dev_err(dev, "HiFi dai fmt failed: CPCAP should be provider"); return -EINVAL; } -- 2.20.1

1 1

[PATCH] ASoC: dt-bindings: wlf,wm8978: Fix I2C address in example
by Geert Uytterhoeven 27 Sep '21

27 Sep '21

According to the WM8978 datasheet, the address of the device is fixed as 0011010 in 2-wire mode. Fixes: 5bd5699c494f42a3 ("ASoC: dt-bindings: Add WM8978 Binding") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- Documentation/devicetree/bindings/sound/wlf,wm8978.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml index 944e5859c962e634..0b80ac3a7106658c 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -48,10 +48,10 @@ examples: #address-cells = <1>; #size-cells = <0>; - codec@0 { + codec@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8978"; - reg = <0>; + reg = <1a>; }; }; -- 2.25.1

3 2

[PATCH] ASoC: ak4104: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4104 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4104.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 979cfb165eed..dc4747c77a7a 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -81,8 +81,8 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - /* This device can only be slave */ - if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + /* This device can only be consumer */ + if ((format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, -- 2.20.1

1 1

[PATCH 00/16] add driver to support firmware loading on Cirrus Logic DSPs
by Simon Trimmer 27 Sep '21

27 Sep '21

This series of patches gradually separates the ASoC specific wm_adsp code from that required to manage firmware in Cirrus Logic DSPs. The series starts with renaming, progresses to splitting the functionality before finally moving the independent functionality into drivers/firmware so that it can be used by both the existing ASoC wm_adsp and new non-audio parts. Charles Keepax (3): ASoC: wm_adsp: Move check for control existence ASoC: wm_adsp: Switch to using wm_coeff_read_ctrl for compressed buffers ASoC: wm_adsp: Move sys_config_size to wm_adsp Simon Trimmer (13): ASoC: wm_adsp: Remove use of snd_ctl_elem_type_t ASoC: wm_adsp: Cancel ongoing work when removing controls ASoC: wm_adsp: Rename generic DSP support ASoC: wm_adsp: Introduce cs_dsp logging macros ASoC: wm_adsp: Separate some ASoC and generic functions ASoC: wm_adsp: Split DSP power operations into helper functions ASoC: wm_adsp: Separate generic cs_dsp_coeff_ctl handling ASoC: wm_adsp: Move check of dsp->running to better place ASoC: wm_adsp: Pass firmware names as parameters when starting DSP core ASoC: wm_adsp: move firmware loading to client ASoC: wm_adsp: Split out struct cs_dsp from struct wm_adsp ASoC: wm_adsp: Separate wm_adsp specifics in cs_dsp_client_ops firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs MAINTAINERS | 11 + drivers/firmware/Kconfig | 1 + drivers/firmware/Makefile | 1 + drivers/firmware/cirrus/Kconfig | 5 + drivers/firmware/cirrus/Makefile | 3 + drivers/firmware/cirrus/cs_dsp.c | 3109 ++++++++++++++++ include/linux/firmware/cirrus/cs_dsp.h | 242 ++ .../linux/firmware/cirrus}/wmfw.h | 8 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs47l15.c | 22 +- sound/soc/codecs/cs47l24.c | 20 +- sound/soc/codecs/cs47l35.c | 24 +- sound/soc/codecs/cs47l85.c | 32 +- sound/soc/codecs/cs47l90.c | 34 +- sound/soc/codecs/cs47l92.c | 20 +- sound/soc/codecs/madera.c | 18 +- sound/soc/codecs/wm2200.c | 30 +- sound/soc/codecs/wm5102.c | 16 +- sound/soc/codecs/wm5110.c | 24 +- sound/soc/codecs/wm_adsp.c | 3188 ++--------------- sound/soc/codecs/wm_adsp.h | 105 +- 21 files changed, 3794 insertions(+), 3120 deletions(-) create mode 100644 drivers/firmware/cirrus/Kconfig create mode 100644 drivers/firmware/cirrus/Makefile create mode 100644 drivers/firmware/cirrus/cs_dsp.c create mode 100644 include/linux/firmware/cirrus/cs_dsp.h rename {sound/soc/codecs => include/linux/firmware/cirrus}/wmfw.h (91%) -- 2.33.0

2 17

[PATCH] ASoC: dwc-i2s: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the dwc-i2s driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/dwc/dwc-i2s.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 33ce257ae198..5cb58929090d 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -356,25 +356,25 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); int ret = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: if (dev->capability & DW_I2S_SLAVE) ret = 0; else ret = -EINVAL; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: if (dev->capability & DW_I2S_MASTER) ret = 0; else ret = -EINVAL; break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBC_CFP: ret = -EINVAL; break; default: - dev_dbg(dev->dev, "dwc : Invalid master/slave format\n"); + dev_dbg(dev->dev, "dwc : Invalid clock provider format\n"); ret = -EINVAL; break; } -- 2.20.1

1 1

[PATCH] ASoC: wcd9335: Use correct version to initialize Class H
by Yassine Oudjana 27 Sep '21

27 Sep '21

The versioning scheme was changed in an earlier patch, which caused the version being used to initialize WCD9335 to be interpreted as if it was WCD937X, which changed code paths causing broken headphones output. Pass WCD9335 instead of WCD9335_VERSION_2_0 to wcd_clsh_ctrl_alloc to fix it. Fixes: 19c5d1f6a0c3 ("ASoC: codecs: wcd-clsh: add new version support") Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> --- sound/soc/codecs/wcd9335.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index d885ced34f60..bc5d68c53e5a 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -4859,7 +4859,7 @@ static int wcd9335_codec_probe(struct snd_soc_component *component) snd_soc_component_init_regmap(component, wcd->regmap); /* Class-H Init*/ - wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, wcd->version); + wcd->clsh_ctrl = wcd_clsh_ctrl_alloc(component, WCD9335); if (IS_ERR(wcd->clsh_ctrl)) return PTR_ERR(wcd->clsh_ctrl); -- 2.33.0

3 2

[PATCH] ASoC: tegra: Constify static snd_soc_dai_ops structs
by Rikard Falkeborn 27 Sep '21

27 Sep '21

The only usage of these is to assign their address to the ops field in the snd_soc_dai_driver struct, which is a pointer to const. 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/tegra/tegra210_adx.c | 4 ++-- sound/soc/tegra/tegra210_amx.c | 4 ++-- sound/soc/tegra/tegra210_mixer.c | 4 ++-- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 4 ++-- 5 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index 78d660447bb1..d7c7849c2f92 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -206,12 +206,12 @@ static int tegra210_adx_put_byte_map(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_soc_dai_ops tegra210_adx_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_adx_in_dai_ops = { .hw_params = tegra210_adx_in_hw_params, .startup = tegra210_adx_startup, }; -static struct snd_soc_dai_ops tegra210_adx_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_adx_out_dai_ops = { .hw_params = tegra210_adx_out_hw_params, }; diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index 83176e1663ab..af9bddfc3120 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -241,12 +241,12 @@ static int tegra210_amx_put_byte_map(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_soc_dai_ops tegra210_amx_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_amx_out_dai_ops = { .hw_params = tegra210_amx_out_hw_params, .startup = tegra210_amx_startup, }; -static struct snd_soc_dai_ops tegra210_amx_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_amx_in_dai_ops = { .hw_params = tegra210_amx_in_hw_params, }; diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 53fcd8f6605a..55e61776c565 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -283,11 +283,11 @@ static int tegra210_mixer_out_hw_params(struct snd_pcm_substream *substream, dai->id - TEGRA210_MIXER_RX_MAX); } -static struct snd_soc_dai_ops tegra210_mixer_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_mixer_out_dai_ops = { .hw_params = tegra210_mixer_out_hw_params, }; -static struct snd_soc_dai_ops tegra210_mixer_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_mixer_in_dai_ops = { .hw_params = tegra210_mixer_in_hw_params, }; diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index 3646ce9b0fd1..7b9c7006e419 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -387,7 +387,7 @@ static int tegra210_mvc_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_dai_ops tegra210_mvc_dai_ops = { +static const struct snd_soc_dai_ops tegra210_mvc_dai_ops = { .hw_params = tegra210_mvc_hw_params, }; diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index 260dca6f6d25..dc477ee1b82c 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3287,12 +3287,12 @@ static int tegra210_sfc_put_control(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_soc_dai_ops tegra210_sfc_in_dai_ops = { +static const struct snd_soc_dai_ops tegra210_sfc_in_dai_ops = { .hw_params = tegra210_sfc_in_hw_params, .startup = tegra210_sfc_startup, }; -static struct snd_soc_dai_ops tegra210_sfc_out_dai_ops = { +static const struct snd_soc_dai_ops tegra210_sfc_out_dai_ops = { .hw_params = tegra210_sfc_out_hw_params, }; -- 2.33.0

3 2

[PATCH] ASoC: ak4458: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4458 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4458.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 29eb78702bf3..baa9ff5d0ce5 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -464,14 +464,14 @@ static int ak4458_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct ak4458_priv *ak4458 = snd_soc_component_get_drvdata(component); int ret; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: /* Slave Mode */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: /* Consumer Mode */ break; - case SND_SOC_DAIFMT_CBM_CFM: /* Master Mode is not supported */ - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFP: /* Provider Mode is not supported */ + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: default: - dev_err(component->dev, "Master mode unsupported\n"); + dev_err(component->dev, "Clock provider mode unsupported\n"); return -EINVAL; } -- 2.20.1

1 1

[PATCH] ASoC: cs4341: Add SPI device ID table
by Mark Brown 27 Sep '21

27 Sep '21

Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: patches(a)opensource.cirrus.com --- sound/soc/codecs/cs4341.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/cs4341.c b/sound/soc/codecs/cs4341.c index 7d3e54d8eef3..29d05e32d341 100644 --- a/sound/soc/codecs/cs4341.c +++ b/sound/soc/codecs/cs4341.c @@ -305,12 +305,19 @@ static int cs4341_spi_probe(struct spi_device *spi) return cs4341_probe(&spi->dev); } +static const struct spi_device_id cs4341_spi_ids[] = { + { "cs4341a" }, + { } +}; +MODULE_DEVICE_TABLE(spi, cs4341_spi_ids); + static struct spi_driver cs4341_spi_driver = { .driver = { .name = "cs4341-spi", .of_match_table = of_match_ptr(cs4341_dt_ids), }, .probe = cs4341_spi_probe, + .id_table = cs4341_spi_ids, }; #endif -- 2.20.1

2 2

[PATCH v2] ASoC: dt-bindings: wlf,wm8978: Fix I2C address in example
by Geert Uytterhoeven 27 Sep '21

27 Sep '21

According to the WM8978 datasheet, the address of the device is fixed as 0011010 in 2-wire mode. Fixes: 5bd5699c494f42a3 ("ASoC: dt-bindings: Add WM8978 Binding") Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- v2: - Fix silly typo (missing 0x prefix). --- Documentation/devicetree/bindings/sound/wlf,wm8978.yaml | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml index 944e5859c962e634..96cf9fc9c8b0098b 100644 --- a/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml +++ b/Documentation/devicetree/bindings/sound/wlf,wm8978.yaml @@ -48,10 +48,10 @@ examples: #address-cells = <1>; #size-cells = <0>; - codec@0 { + codec@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8978"; - reg = <0>; + reg = <0x1a>; }; }; -- 2.25.1

2 1

[PATCH v2] ASoC: cpcap: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cpcap driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cpcap.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index 05bbacd0d174..598e09024e23 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -1168,15 +1168,15 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * "HiFi Playback" should always be configured as - * SND_SOC_DAIFMT_CBM_CFM - codec clk & frm master + * SND_SOC_DAIFMT_CBP_CFP - codec clk & frm provider * SND_SOC_DAIFMT_I2S - I2S mode */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val &= ~BIT(CPCAP_BIT_SMB_ST_DAC); break; default: - dev_err(dev, "HiFi dai fmt failed: CPCAP should be master"); + dev_err(dev, "HiFi dai fmt failed: CPCAP should be provider"); return -EINVAL; } @@ -1318,15 +1318,15 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, /* * "Voice Playback" and "Voice Capture" should always be - * configured as SND_SOC_DAIFMT_CBM_CFM - codec clk & frm - * master + * configured as SND_SOC_DAIFMT_CBP_CFP - codec clk & frm + * provider */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val &= ~BIT(CPCAP_BIT_SMB_CDC); break; default: - dev_err(component->dev, "Voice dai fmt failed: CPCAP should be the master"); + dev_err(component->dev, "Voice dai fmt failed: CPCAP should be the provider"); val &= ~BIT(CPCAP_BIT_SMB_CDC); break; } -- 2.20.1

1 1

[PATCH] ASoC: pcm179x: Add missing entries SPI to device ID table
by Mark Brown 27 Sep '21

27 Sep '21

Currently autoloading for SPI devices does not use the DT ID table, it uses SPI modalises. Supporting OF modalises is going to be difficult if not impractical, an attempt was made but has been reverted, so ensure that module autoloading works for this driver by adding SPI IDs for parts that only have a compatible listed. Fixes: 96c8395e2166 ("spi: Revert modalias changes") Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm179x-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/pcm179x-spi.c b/sound/soc/codecs/pcm179x-spi.c index 0a542924ec5f..ebf63ea90a1c 100644 --- a/sound/soc/codecs/pcm179x-spi.c +++ b/sound/soc/codecs/pcm179x-spi.c @@ -36,6 +36,7 @@ static const struct of_device_id pcm179x_of_match[] = { MODULE_DEVICE_TABLE(of, pcm179x_of_match); static const struct spi_device_id pcm179x_spi_ids[] = { + { "pcm1792a", 0 }, { "pcm179x", 0 }, { }, }; -- 2.20.1

1 1

[PATCH] ASoC: ad1836: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ad1836 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ad1836.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 08a5651bed9f..29e1689da67f 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -148,9 +148,9 @@ static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - /* ALCLK,ABCLK are both output, AD1836 can only be master */ - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + /* ALCLK,ABCLK are both output, AD1836 can only be provider */ + case SND_SOC_DAIFMT_CBP_CFP: break; default: return -EINVAL; -- 2.20.1

1 1

[PATCH] ASoC: ak4642: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4642 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4642.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index c49c58eeb476..c284dcc5af76 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -392,13 +392,13 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) data = MCKO | PMPLL; /* use MCKO */ bcko = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set clocking for audio interface */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: data |= MS; bcko = BCKO_64; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

1 1

[PATCH 1/6] ASoC: adau1372: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the adau1372 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/adau1372.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/adau1372.c b/sound/soc/codecs/adau1372.c index 6811a8b3866d..1faa4c426365 100644 --- a/sound/soc/codecs/adau1372.c +++ b/sound/soc/codecs/adau1372.c @@ -30,7 +30,7 @@ struct adau1372 { void (*switch_mode)(struct device *dev); bool use_pll; bool enabled; - bool master; + bool clock_provider; struct snd_pcm_hw_constraint_list rate_constraints; unsigned int slot_width; @@ -578,13 +578,13 @@ static int adau1372_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) unsigned int sai0 = 0, sai1 = 0; bool invert_lrclk = false; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - adau1372->master = true; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + adau1372->clock_provider = true; sai1 |= ADAU1372_SAI1_MS; break; - case SND_SOC_DAIFMT_CBS_CFS: - adau1372->master = false; + case SND_SOC_DAIFMT_CBC_CFC: + adau1372->clock_provider = false; break; default: return -EINVAL; @@ -714,7 +714,7 @@ static int adau1372_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, break; case 4: sai0 = ADAU1372_SAI0_SAI_TDM4; - if (adau1372->master) + if (adau1372->clock_provider) adau1372->rate_constraints.mask = ADAU1372_RATE_MASK_TDM4_MASTER; else adau1372->rate_constraints.mask = ADAU1372_RATE_MASK_TDM4; -- 2.20.1

1 6

[PATCH] ASoC: dmaengine: Introduce module option prealloc_buffer_size_kbytes
by Sugar Zhang 27 Sep '21

27 Sep '21

Currently, The fixed 512KB prealloc buffer size is too larger for tiny memory kernel (such as 16MB memory). This patch adds the module option "prealloc_buffer_size_kbytes" to specify prealloc buffer size. It's suitable for cards which use the generic dmaengine pcm driver with no config. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> --- Documentation/sound/alsa-configuration.rst | 9 +++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +++++- 2 files changed, 14 insertions(+), 1 deletion(-) diff --git a/Documentation/sound/alsa-configuration.rst b/Documentation/sound/alsa-configuration.rst index 65f6169..34888d4 100644 --- a/Documentation/sound/alsa-configuration.rst +++ b/Documentation/sound/alsa-configuration.rst @@ -100,6 +100,15 @@ amidi_map MIDI device number maps assigned to the 2st OSS device; Default: 1 +Module snd-soc-core +------------------- + +The soc core module. It is used by all ALSA card drivers. +It takes the following options which have global effects. + +prealloc_buffer_size_kbytes + Specify prealloc buffer size in kbytes (default: 512). + Common parameters for top sound card modules -------------------------------------------- diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 4aa48c7..c54c8ca 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -15,6 +15,10 @@ #include <sound/dmaengine_pcm.h> +static unsigned int prealloc_buffer_size_kbytes = 512; +module_param(prealloc_buffer_size_kbytes, uint, 0444); +MODULE_PARM_DESC(prealloc_buffer_size_kbytes, "Preallocate DMA buffer size (KB)."); + /* * The platforms dmaengine driver does not support reporting the amount of * bytes that are still left to transfer. @@ -237,7 +241,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, prealloc_buffer_size = config->prealloc_buffer_size; max_buffer_size = config->pcm_hardware->buffer_bytes_max; } else { - prealloc_buffer_size = 512 * 1024; + prealloc_buffer_size = prealloc_buffer_size_kbytes * 1024; max_buffer_size = SIZE_MAX; } -- 2.7.4

2 1

[PATCH] ASoC: ep93xx: Convert to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of the effort to remove our old APIs based on outdated terminology update the EP93xx drivers to use modern terminology. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 10 +++++----- sound/soc/cirrus/snappercl15.c | 2 +- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 7b6cdc9c8a23..385290202912 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -60,7 +60,7 @@ static struct snd_soc_dai_link edb93xx_dai = { .name = "CS4271", .stream_name = "CS4271 HiFi", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ops = &edb93xx_ops, SND_SOC_DAILINK_REG(hifi), }; diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 0d26550d0df8..06c315a4d20a 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -245,14 +245,14 @@ static int ep93xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* CPU is master */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + /* CPU is provider */ clk_cfg |= EP93XX_I2S_CLKCFG_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFM: - /* Codec is master */ + case SND_SOC_DAIFMT_CBP_CFP: + /* Codec is provider */ clk_cfg &= ~EP93XX_I2S_CLKCFG_MASTER; break; diff --git a/sound/soc/cirrus/snappercl15.c b/sound/soc/cirrus/snappercl15.c index c4b112921661..a286f5beeaeb 100644 --- a/sound/soc/cirrus/snappercl15.c +++ b/sound/soc/cirrus/snappercl15.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link snappercl15_dai = { .name = "tlv320aic23", .stream_name = "AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, + SND_SOC_DAIFMT_CBC_CFC, .ops = &snappercl15_ops, SND_SOC_DAILINK_REG(aic23), }; -- 2.20.1

1 1

[PATCH] ASoC: ak4671: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4671 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4671.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index eb435235b5a3..e9d1251c4265 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -520,11 +520,11 @@ static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) /* set master/slave audio interface */ mode = snd_soc_component_read(component, AK4671_PLL_MODE_SELECT1); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: mode |= AK4671_M_S; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: mode &= ~(AK4671_M_S); break; default: -- 2.20.1

1 1

[PATCH] ASoC: ak4118: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak4118 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak4118.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/ak4118.c b/sound/soc/codecs/ak4118.c index 5d46ae85566c..e0a6451851e8 100644 --- a/sound/soc/codecs/ak4118.c +++ b/sound/soc/codecs/ak4118.c @@ -151,8 +151,8 @@ static const struct snd_soc_dapm_route ak4118_dapm_routes[] = { }; -static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, - unsigned int format) +static int ak4118_set_dai_fmt_provider(struct ak4118_priv *ak4118, + unsigned int format) { int dif; @@ -173,8 +173,8 @@ static int ak4118_set_dai_fmt_master(struct ak4118_priv *ak4118, return dif; } -static int ak4118_set_dai_fmt_slave(struct ak4118_priv *ak4118, - unsigned int format) +static int ak4118_set_dai_fmt_consumer(struct ak4118_priv *ak4118, + unsigned int format) { int dif; @@ -201,14 +201,12 @@ static int ak4118_set_dai_fmt(struct snd_soc_dai *dai, int dif; int ret = 0; - switch (format & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - /* component is master */ - dif = ak4118_set_dai_fmt_master(ak4118, format); + switch (format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + dif = ak4118_set_dai_fmt_provider(ak4118, format); break; - case SND_SOC_DAIFMT_CBS_CFS: - /*component is slave */ - dif = ak4118_set_dai_fmt_slave(ak4118, format); + case SND_SOC_DAIFMT_CBC_CFC: + dif = ak4118_set_dai_fmt_consumer(ak4118, format); break; default: ret = -ENOTSUPP; -- 2.20.1

1 1

[PATCH] ASoC: codecs: Fix spelling mistake "Unsupport" -> "Unsupported"
by Colin King 27 Sep '21

27 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There are spelling mistakes in dev_err error messages. Fix them. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt1015.c | 2 +- sound/soc/codecs/rt1016.c | 2 +- sound/soc/codecs/rt1019.c | 2 +- sound/soc/codecs/rt1305.c | 2 +- sound/soc/codecs/rt1308.c | 2 +- sound/soc/codecs/rt5514.c | 2 +- sound/soc/codecs/rt5616.c | 2 +- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 2 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5670.c | 2 +- sound/soc/codecs/rt5677.c | 2 +- sound/soc/codecs/rt5682.c | 6 +++--- 18 files changed, 20 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c0c5952cdff7..6a27dfacd81c 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -864,7 +864,7 @@ static int rt1015_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1016.c b/sound/soc/codecs/rt1016.c index 7561d202274c..9845cdddcb4c 100644 --- a/sound/soc/codecs/rt1016.c +++ b/sound/soc/codecs/rt1016.c @@ -490,7 +490,7 @@ static int rt1016_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out * 4, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1019.c b/sound/soc/codecs/rt1019.c index 8c0b00242bb8..80b7ca0e4e1e 100644 --- a/sound/soc/codecs/rt1019.c +++ b/sound/soc/codecs/rt1019.c @@ -359,7 +359,7 @@ static int rt1019_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1305.c b/sound/soc/codecs/rt1305.c index 7a0094578e46..a9c473537a91 100644 --- a/sound/soc/codecs/rt1305.c +++ b/sound/soc/codecs/rt1305.c @@ -841,7 +841,7 @@ static int rt1305_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt1308.c b/sound/soc/codecs/rt1308.c index b4e5546e2e21..c555b77b3c5c 100644 --- a/sound/soc/codecs/rt1308.c +++ b/sound/soc/codecs/rt1308.c @@ -664,7 +664,7 @@ static int rt1308_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 4b1ad5054e8d..577680df7052 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -936,7 +936,7 @@ static int rt5514_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5616.c b/sound/soc/codecs/rt5616.c index fd0d3a08e9dd..8e6414468a87 100644 --- a/sound/soc/codecs/rt5616.c +++ b/sound/soc/codecs/rt5616.c @@ -1133,7 +1133,7 @@ static int rt5616_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index cd1db5caabad..d01fe73ab9c8 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1909,7 +1909,7 @@ static int rt5640_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 9408ee63cb26..197c56047947 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2969,7 +2969,7 @@ static int rt5645_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index fc0c83b73f09..e78ea101bc8d 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1487,7 +1487,7 @@ static int rt5651_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5659.c b/sound/soc/codecs/rt5659.c index 4a50b169fe03..e1503c2eee81 100644 --- a/sound/soc/codecs/rt5659.c +++ b/sound/soc/codecs/rt5659.c @@ -3509,7 +3509,7 @@ static int rt5659_set_component_pll(struct snd_soc_component *component, int pll ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 33ff9156358b..3b50fb29864e 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1046,7 +1046,7 @@ static int rt5660_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index be9fc58ff681..0389b2bb360e 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -2941,7 +2941,7 @@ static int rt5663_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index e59323fd5bf2..33e889802ff8 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4374,7 +4374,7 @@ static int rt5665_set_component_pll(struct snd_soc_component *component, int pll ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index 6ab1a8bc3735..fb09715bf932 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2171,7 +2171,7 @@ static int rt5668_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index ecbaf129a6e3..ce7684752bb0 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2577,7 +2577,7 @@ static int rt5670_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index f655228c8c4b..4a8c267d4fbc 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -4557,7 +4557,7 @@ static int rt5677_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, ret = rt5677_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 4a64cab99c55..12113c2dcae2 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2327,7 +2327,7 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, pll2_fout1 = 3840000; ret = rl6231_pll_calc(freq_in, pll2_fout1, &pll2f_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } @@ -2339,7 +2339,7 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(pll2_fout1, freq_out, &pll2b_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", + dev_err(component->dev, "Unsupported input clock %d\n", pll2_fout1); return ret; } @@ -2390,7 +2390,7 @@ static int rt5682_set_component_pll(struct snd_soc_component *component, ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); if (ret < 0) { - dev_err(component->dev, "Unsupport input clock %d\n", + dev_err(component->dev, "Unsupported input clock %d\n", freq_in); return ret; } -- 2.32.0

2 1

[PATCH] ASoC: cs42l42: Use two thresholds and increased wait time for manual type detection
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Some headsets require very different comparator thresholds for type detection, as well as longer settling times. In order to detect a larger number of headsets, use 2 thresholds to give maximum coverage (1.25V and 1.75V), as well as a longer settling time of 100ms. This will not affect default audotodetect mode and applies to manual mode type detection only. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 84 +++++++++++++++++++++++++++----------- sound/soc/codecs/cs42l42.h | 5 +++ 2 files changed, 66 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index 78f8c89410b6..3f202acd4a8e 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1048,7 +1048,8 @@ static struct snd_soc_dai_driver cs42l42_dai = { static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) { unsigned int hs_det_status; - unsigned int hs_det_comp; + unsigned int hs_det_comp1; + unsigned int hs_det_comp2; unsigned int hs_det_sw; /* Set hs detect to manual, active mode */ @@ -1063,23 +1064,40 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) (0 << CS42L42_HSBIAS_REF_SHIFT) | (0 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + /* Configure HS DET comparator reference levels. */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL1, + CS42L42_HSDET_COMP1_LVL_MASK | + CS42L42_HSDET_COMP2_LVL_MASK, + (CS42L42_HSDET_COMP1_LVL_VAL << CS42L42_HSDET_COMP1_LVL_SHIFT) | + (CS42L42_HSDET_COMP2_LVL_VAL << CS42L42_HSDET_COMP2_LVL_SHIFT)); + /* Open the SW_HSB_HS3 switch and close SW_HSB_HS4 for a Type 1 headset. */ regmap_write(cs42l42->regmap, CS42L42_HS_SWITCH_CTL, CS42L42_HSDET_SW_COMP1); + msleep(100); + regmap_read(cs42l42->regmap, CS42L42_HS_DET_STATUS, &hs_det_status); - hs_det_comp = (hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> + hs_det_comp1 = (hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> CS42L42_HSDET_COMP1_OUT_SHIFT; + hs_det_comp2 = (hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> + CS42L42_HSDET_COMP2_OUT_SHIFT; /* Close the SW_HSB_HS3 switch for a Type 2 headset. */ regmap_write(cs42l42->regmap, CS42L42_HS_SWITCH_CTL, CS42L42_HSDET_SW_COMP2); + msleep(100); + regmap_read(cs42l42->regmap, CS42L42_HS_DET_STATUS, &hs_det_status); - hs_det_comp |= ((hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> + hs_det_comp1 |= ((hs_det_status & CS42L42_HSDET_COMP1_OUT_MASK) >> + CS42L42_HSDET_COMP1_OUT_SHIFT) << 1; + hs_det_comp2 |= ((hs_det_status & CS42L42_HSDET_COMP2_OUT_MASK) >> CS42L42_HSDET_COMP2_OUT_SHIFT) << 1; - switch (hs_det_comp) { + /* Use Comparator 1 with 1.25V Threshold. */ + switch (hs_det_comp1) { case CS42L42_HSDET_COMP_TYPE1: cs42l42->hs_type = CS42L42_PLUG_CTIA; hs_det_sw = CS42L42_HSDET_SW_TYPE1; @@ -1088,14 +1106,26 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = CS42L42_PLUG_OMTP; hs_det_sw = CS42L42_HSDET_SW_TYPE2; break; - case CS42L42_HSDET_COMP_TYPE3: - cs42l42->hs_type = CS42L42_PLUG_HEADPHONE; - hs_det_sw = CS42L42_HSDET_SW_TYPE3; - break; default: - cs42l42->hs_type = CS42L42_PLUG_INVALID; - hs_det_sw = CS42L42_HSDET_SW_TYPE4; - break; + /* Fallback to Comparator 2 with 1.75V Threshold. */ + switch (hs_det_comp2) { + case CS42L42_HSDET_COMP_TYPE1: + cs42l42->hs_type = CS42L42_PLUG_CTIA; + hs_det_sw = CS42L42_HSDET_SW_TYPE1; + break; + case CS42L42_HSDET_COMP_TYPE2: + cs42l42->hs_type = CS42L42_PLUG_OMTP; + hs_det_sw = CS42L42_HSDET_SW_TYPE2; + break; + case CS42L42_HSDET_COMP_TYPE3: + cs42l42->hs_type = CS42L42_PLUG_HEADPHONE; + hs_det_sw = CS42L42_HSDET_SW_TYPE3; + break; + default: + cs42l42->hs_type = CS42L42_PLUG_INVALID; + hs_det_sw = CS42L42_HSDET_SW_TYPE4; + break; + } } /* Set Switches */ @@ -1112,6 +1142,14 @@ static void cs42l42_manual_hs_type_detect(struct cs42l42_private *cs42l42) (0 << CS42L42_HSDET_SET_SHIFT) | (0 << CS42L42_HSBIAS_REF_SHIFT) | (0 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + + /* Configure HS DET comparator reference levels. */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL1, + CS42L42_HSDET_COMP1_LVL_MASK | + CS42L42_HSDET_COMP2_LVL_MASK, + (CS42L42_HSDET_COMP1_LVL_DEFAULT << CS42L42_HSDET_COMP1_LVL_SHIFT) | + (CS42L42_HSDET_COMP2_LVL_DEFAULT << CS42L42_HSDET_COMP2_LVL_SHIFT)); } static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) @@ -1134,6 +1172,18 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = (hs_det_status & CS42L42_HSDET_TYPE_MASK) >> CS42L42_HSDET_TYPE_SHIFT; + /* Set hs detect to automatic, disabled mode */ + regmap_update_bits(cs42l42->regmap, + CS42L42_HSDET_CTL2, + CS42L42_HSDET_CTRL_MASK | + CS42L42_HSDET_SET_MASK | + CS42L42_HSBIAS_REF_MASK | + CS42L42_HSDET_AUTO_TIME_MASK, + (2 << CS42L42_HSDET_CTRL_SHIFT) | + (2 << CS42L42_HSDET_SET_SHIFT) | + (0 << CS42L42_HSBIAS_REF_SHIFT) | + (3 << CS42L42_HSDET_AUTO_TIME_SHIFT)); + /* Run Manual detection if auto detect has not found a headset. * We Re-Run with Manual Detection if the original detection was invalid or headphones, * to ensure that a headset mic is detected in all cases. @@ -1142,18 +1192,6 @@ static void cs42l42_process_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type == CS42L42_PLUG_HEADPHONE) { dev_dbg(cs42l42->component->dev, "Running Manual Detection Fallback\n"); cs42l42_manual_hs_type_detect(cs42l42); - } else { - /* Set hs detect to automatic, disabled mode */ - regmap_update_bits(cs42l42->regmap, - CS42L42_HSDET_CTL2, - CS42L42_HSDET_CTRL_MASK | - CS42L42_HSDET_SET_MASK | - CS42L42_HSBIAS_REF_MASK | - CS42L42_HSDET_AUTO_TIME_MASK, - (2 << CS42L42_HSDET_CTRL_SHIFT) | - (2 << CS42L42_HSDET_SET_SHIFT) | - (0 << CS42L42_HSBIAS_REF_SHIFT) | - (3 << CS42L42_HSDET_AUTO_TIME_SHIFT)); } /* Set up button detection */ diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 2aeabba73e05..0704c902475f 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -188,6 +188,11 @@ #define CS42L42_HSDET_COMP2_LVL_SHIFT 4 #define CS42L42_HSDET_COMP2_LVL_MASK (15 << CS42L42_HSDET_COMP2_LVL_SHIFT) +#define CS42L42_HSDET_COMP1_LVL_VAL 12 /* 1.25V Comparator */ +#define CS42L42_HSDET_COMP2_LVL_VAL 2 /* 1.75V Comparator */ +#define CS42L42_HSDET_COMP1_LVL_DEFAULT 7 /* 1V Comparator */ +#define CS42L42_HSDET_COMP2_LVL_DEFAULT 7 /* 2V Comparator */ + #define CS42L42_HSDET_CTL2 (CS42L42_PAGE_11 + 0x20) #define CS42L42_HSDET_AUTO_TIME_SHIFT 0 #define CS42L42_HSDET_AUTO_TIME_MASK (3 << CS42L42_HSDET_AUTO_TIME_SHIFT) -- 2.25.1

2 1

[PATCH] ASoC: ad193x: Update to modern clocking terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ad193x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ad193x.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 278a55af158b..30b98b4267e1 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -243,22 +243,22 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai, if (fmt & SND_SOC_DAIFMT_DSP_A) dac_fmt ^= AD193X_DAC_LEFT_HIGH; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & frm master */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: adc_fmt |= AD193X_ADC_LCR_MASTER; adc_fmt |= AD193X_ADC_BCLK_MASTER; dac_fmt |= AD193X_DAC_LCR_MASTER; dac_fmt |= AD193X_DAC_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & frm master */ + case SND_SOC_DAIFMT_CBC_CFP: adc_fmt |= AD193X_ADC_LCR_MASTER; dac_fmt |= AD193X_DAC_LCR_MASTER; break; - case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + case SND_SOC_DAIFMT_CBP_CFC: adc_fmt |= AD193X_ADC_BCLK_MASTER; dac_fmt |= AD193X_DAC_BCLK_MASTER; break; - case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & frm slave */ + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

1 1

[PATCH] ASoC: meson: aiu: Fix spelling mistake "Unsupport" -> "Unsupported"
by Colin King 27 Sep '21

27 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err error message. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/meson/aiu-encoder-spdif.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-spdif.c b/sound/soc/meson/aiu-encoder-spdif.c index de850913975f..97da60db2c4d 100644 --- a/sound/soc/meson/aiu-encoder-spdif.c +++ b/sound/soc/meson/aiu-encoder-spdif.c @@ -113,7 +113,7 @@ static int aiu_encoder_spdif_hw_params(struct snd_pcm_substream *substream, val |= AIU_958_MISC_MODE_32BITS; break; default: - dev_err(dai->dev, "Unsupport physical width\n"); + dev_err(dai->dev, "Unsupported physical width\n"); return -EINVAL; } -- 2.32.0

3 2

[PATCH] ASoC: samsung: add missing "fallthrough;"
by Arnd Bergmann 27 Sep '21

27 Sep '21

From: Arnd Bergmann <arnd(a)arndb.de> With gcc, we get a warning in this file: In file included from include/linux/io.h:13, from sound/soc/samsung/s3c-i2s-v2.c:16: sound/soc/samsung/s3c-i2s-v2.c: In function 's3c2412_i2s_trigger': arch/arm/include/asm/io.h:92:22: error: this statement may fall through [-Werror=implicit-fallthrough=] #define __raw_writel __raw_writel ^ arch/arm/include/asm/io.h:299:29: note: in expansion of macro '__raw_writel' #define writel_relaxed(v,c) __raw_writel((__force u32) cpu_to_le32(v),c) ^~~~~~~~~~~~ arch/arm/include/asm/io.h:307:36: note: in expansion of macro 'writel_relaxed' #define writel(v,c) ({ __iowmb(); writel_relaxed(v,c); }) ^~~~~~~~~~~~~~ sound/soc/samsung/s3c-i2s-v2.c:398:3: note: in expansion of macro 'writel' writel(0x0, i2s->regs + S3C2412_IISFIC); ^~~~~~ sound/soc/samsung/s3c-i2s-v2.c:400:2: note: here case SNDRV_PCM_TRIGGER_RESUME: ^~~~ >From all I can tell, this was indeed meant to fall through, so add "fallthrough;" statement to avoid the warning. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/samsung/s3c-i2s-v2.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index e9481187a08c..de66cc422e6e 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -397,6 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, /* clear again, just in case */ writel(0x0, i2s->regs + S3C2412_IISFIC); + fallthrough; + case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!i2s->master) { -- 2.29.2

3 2

[PATCH] ASoC: alc5623: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the alc5623 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/alc5623.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 54f489837162..b10357a6d655 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -641,12 +641,12 @@ static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; u16 iface = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface = ALC5623_DAI_SDP_MASTER_MODE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface = ALC5623_DAI_SDP_SLAVE_MODE; break; default: -- 2.20.1

1 1

[PATCH] ASoC: alc5632: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the alc5632 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/alc5632.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 79813882a955..6d7af3736a91 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -815,12 +815,12 @@ static int alc5632_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; u16 iface = 0; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* set audio interface clocking */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: iface = ALC5632_DAI_SDP_MASTER_MODE; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: iface = ALC5632_DAI_SDP_SLAVE_MODE; break; default: -- 2.20.1

1 1

[PATCH] ASoC: ak5558: Use modern ASoC DAI format terminology
by Mark Brown 27 Sep '21

27 Sep '21

As part of moving to remove the old style defines for the bus clocks update the ak5558 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ak5558.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/ak5558.c b/sound/soc/codecs/ak5558.c index 37d4600b6f2c..c94cfde3e4a8 100644 --- a/sound/soc/codecs/ak5558.c +++ b/sound/soc/codecs/ak5558.c @@ -198,13 +198,13 @@ static int ak5558_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct snd_soc_component *component = dai->component; u8 format; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBP_CFC: default: dev_err(dai->dev, "Clock mode unsupported"); return -EINVAL; -- 2.20.1

1 1

[PATCH] ASoC: cs42l42: Add HP Volume Scale control
by Vitaly Rodionov 27 Sep '21

27 Sep '21

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> cs42l42 has a configurable scaling of the maximum volume. Add an ALSA control for this. Note that the datasheet name is "full scale volume" but this conflicts with ALSA naming convention so the control is named "HP Volume Scale". Before this change the FULL_SCALE_VOLUME was set based on the value in RLA_STAT, but as there isn't any load detection result this always set the scaling to -6dB instead of the default 0dB. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 25 ++++++++++++------------- 1 file changed, 12 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d5e1e5228b5f..78f8c89410b6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -425,6 +425,14 @@ static SOC_ENUM_SINGLE_DECL(cs42l42_wnf3_freq_enum, CS42L42_ADC_WNF_HPF_CTL, CS42L42_ADC_WNF_CF_SHIFT, cs42l42_wnf3_freq_text); +static const char * const cs42l42_full_scale_vol_text[] = { + "0dB", "-6dB" +}; + +static SOC_ENUM_SINGLE_DECL(cs42l42_full_scale_vol_enum, CS42L42_HP_CTL, + CS42L42_HP_FULL_SCALE_VOL_SHIFT, + cs42l42_full_scale_vol_text); + static const struct snd_kcontrol_new cs42l42_snd_controls[] = { /* ADC Volume and Filter Controls */ SOC_SINGLE("ADC Notch Switch", CS42L42_ADC_CTL, @@ -450,6 +458,7 @@ static const struct snd_kcontrol_new cs42l42_snd_controls[] = { CS42L42_DACB_INV_SHIFT, true, false), SOC_SINGLE("DAC HPF Switch", CS42L42_DAC_CTL2, CS42L42_DAC_HPF_EN_SHIFT, true, false), + SOC_ENUM("HP Volume Scale", cs42l42_full_scale_vol_enum), SOC_DOUBLE_R_TLV("Mixer Volume", CS42L42_MIXER_CHA_VOL, CS42L42_MIXER_CHB_VOL, CS42L42_MIXER_CH_VOL_SHIFT, 0x3f, 1, mixer_tlv) @@ -922,7 +931,6 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) struct snd_soc_component *component = dai->component; struct cs42l42_private *cs42l42 = snd_soc_component_get_drvdata(component); unsigned int regval; - u8 fullScaleVol; int ret; if (mute) { @@ -993,20 +1001,11 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) cs42l42->stream_use |= 1 << stream; if (stream == SNDRV_PCM_STREAM_PLAYBACK) { - /* Read the headphone load */ - regval = snd_soc_component_read(component, CS42L42_LOAD_DET_RCSTAT); - if (((regval & CS42L42_RLA_STAT_MASK) >> CS42L42_RLA_STAT_SHIFT) == - CS42L42_RLA_STAT_15_OHM) { - fullScaleVol = CS42L42_HP_FULL_SCALE_VOL_MASK; - } else { - fullScaleVol = 0; - } - - /* Un-mute the headphone, set the full scale volume flag */ + /* Un-mute the headphone */ snd_soc_component_update_bits(component, CS42L42_HP_CTL, CS42L42_HP_ANA_AMUTE_MASK | - CS42L42_HP_ANA_BMUTE_MASK | - CS42L42_HP_FULL_SCALE_VOL_MASK, fullScaleVol); + CS42L42_HP_ANA_BMUTE_MASK, + 0); } } -- 2.25.1

2 1

Broken Pipe with SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Ryan McClue 27 Sep '21

27 Sep '21

Hello again. I'm using alsa-kernel. My configuration: SNDRV_PCM_HW_PARAM_ACCESS = SNDRV_PCM_ACCESS_RW_INTERLEAVED SNDRV_PCM_HW_PARAM_FORMAT = SNDRV_PCM_FORMAT_S16_LE SNDRV_PCM_HW_PARAM_SUBFORMAT = SNDRV_PCM_SUBFORMAT_STD SNDRV_PCM_HW_PARAM_CHANNELS = 2 SNDRV_PCM_HW_PARAM_RATE = 48000 I have a vsynced game loop that is running at 60fps: > int num_base_samples = 48000 * (1 / 60); > int num_samples = num_base_samples * 2; > int16_t buffer[num_samples] = {}; > > while (true) { > int16_t *samples = buffer; > for (int sample_i = 0; sample_i < num_base_samples; sample_i++) { > *samples++ = 0x33; > *samples++ = 0x33; > } > > struct snd_xferi transfer = {}; > transfer.buf = buffer; > transfer.frames = num_base_samples; > ioctl(fd, SNDRV_PCM_IOCTL_WRITEI_FRAMES, &transfer); > > // ioctl(fd, SNDRV_PCM_IOCTL_DRAIN); --> NECESSARY??? > // ioctl(fd, SNDRV_PCM_IOCTL_PREPARE); --> NECESSARY??? > } On the first iteration of SNDRV_PCM_IOCTL_WRITEI_FRAMES I get no error. All subsequent iterations, I get Broken Pipe error. So, to counteract this at the end of each frame I call SNDRV_PCM_IOCTL_DRAIN and SNDRV_PCM_IOCTL_PREPARE. This removes the Broken Pipe error however slows the program down by half and no sound is heard. How best to solve this issue? Thanks again -- Ryan McClue, Sydney

2 3

[PATCH v2 00/12] Add support for on demand pipeline setup/destroy
by Daniel Baluta 27 Sep '21

27 Sep '21

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries implements the new feature to setup/teardown pipeline as needed when a PCM is open/closed. This only only support single cores, with support for multicore arriving with a future patchseries. Review with SOF community at https://github.com/thesofproject/linux/pull/2794 Changes since v1: - added my own Signed-off-by tag. Ranjani Sridharan (12): ASoC: topology: change the complete op in snd_soc_tplg_ops to return int ASoC: SOF: control: Add access field in struct snd_sof_control ASoC: SOF: topology: Add new token for dynamic pipeline ASoC: SOF: sof-audio: add helpers for widgets, kcontrols and dai config set up AsoC: dapm: export a couple of functions ASoC: SOF: Add new fields to snd_sof_route ASoC: SOF: restore kcontrols for widget during set up ASoC: SOF: Don't set up widgets during topology parsing ASoC: SOF: Introduce widget use_count ASoC: SOF: Intel: hda: make sure DAI widget is set up before IPC ASoC: SOF: Add support for dynamic pipelines ASoC: SOF: topology: Add kernel parameter for topology verification include/sound/soc-dpcm.h | 1 + include/sound/soc-topology.h | 2 +- include/uapi/sound/sof/tokens.h | 1 + sound/soc/intel/skylake/skl-topology.c | 6 +- sound/soc/soc-dapm.c | 2 + sound/soc/soc-pcm.c | 4 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/hda-dai.c | 176 +++--- sound/soc/sof/intel/hda.c | 177 ++++-- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/ipc.c | 22 + sound/soc/sof/pcm.c | 58 +- sound/soc/sof/pm.c | 4 +- sound/soc/sof/sof-audio.c | 709 +++++++++++++++++++------ sound/soc/sof/sof-audio.h | 32 +- sound/soc/sof/sof-priv.h | 1 + sound/soc/sof/topology.c | 362 +++++-------- 17 files changed, 1034 insertions(+), 538 deletions(-) -- 2.27.0

4 17

[PATCH v3] ASoC: Intel: boards: add max98390 2/4 speakers support
by Mac Chiang 24 Sep '21

24 Sep '21

support 2 hw boards. 1. SSP2 connects max98390, 2 speakers. 2. SSP1 connects max98390, 2/4 speakers. 2 or 4 speakers playback add echo reference capture add bt offload support add DMI_OEM_STRING for board varients add ALC5682I-VS support Signed-off-by: Mark Hsieh <mark_hsieh(a)wistron.corp-partner.google.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Kieth Tzeng <keith.tzeng(a)quantatw.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- v3: - add acpi_dev_present() for additional tweeter condition, otherwise stereo speakers. - add ALC5682I-VS headphone codec support --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_maxim_common.c | 181 ++++++++++++++++++ sound/soc/intel/boards/sof_maxim_common.h | 16 ++ sound/soc/intel/boards/sof_rt5682.c | 84 ++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 21 ++ sound/soc/sof/sof-pci-dev.c | 9 + 6 files changed, 312 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 3e20c697b569..760d845ac6a1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -466,6 +466,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ (SND_SOC_SOF_BAYTRAIL && (X86_INTEL_LPSS || COMPILE_TEST)) select SND_SOC_MAX98373_I2C + select SND_SOC_MAX98390 select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P diff --git a/sound/soc/intel/boards/sof_maxim_common.c b/sound/soc/intel/boards/sof_maxim_common.c index e66dfe666915..ea5058a5b331 100644 --- a/sound/soc/intel/boards/sof_maxim_common.c +++ b/sound/soc/intel/boards/sof_maxim_common.c @@ -5,6 +5,7 @@ #include <linux/string.h> #include <sound/pcm.h> #include <sound/soc.h> +#include <sound/soc-acpi.h> #include <sound/soc-dai.h> #include <sound/soc-dapm.h> #include <uapi/sound/asound.h> @@ -133,6 +134,186 @@ 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" }, +}; + +static const struct snd_kcontrol_new max_98390_tt_kcontrols[] = { + SOC_DAPM_PIN_SWITCH("TL Spk"), + SOC_DAPM_PIN_SWITCH("TR Spk"), +}; + +static const struct snd_soc_dapm_widget max_98390_tt_dapm_widgets[] = { + SND_SOC_DAPM_SPK("TL Spk", NULL), + SND_SOC_DAPM_SPK("TR Spk", NULL), +}; + +const struct snd_soc_dapm_route max_98390_tt_dapm_routes[] = { + /* Tweeter speaker */ + { "TL Spk", NULL, "Tweeter Left BE_OUT" }, + { "TR Spk", NULL, "Tweeter Right BE_OUT" }, +}; + +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", + }, +}; + +static struct snd_soc_codec_conf max_98390_4spk_codec_conf[] = { + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV0_NAME), + .name_prefix = "Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV1_NAME), + .name_prefix = "Left", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV2_NAME), + .name_prefix = "Tweeter Right", + }, + { + .dlc = COMP_CODEC_CONF(MAX_98390_DEV3_NAME), + .name_prefix = "Tweeter Left", + }, +}; + +struct snd_soc_dai_link_component max_98390_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_components, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_dai_link_component max_98390_4spk_components[] = { + { + .name = MAX_98390_DEV0_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV1_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV2_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, + { + .name = MAX_98390_DEV3_NAME, + .dai_name = MAX_98390_CODEC_DAI, + }, +}; +EXPORT_SYMBOL_NS(max_98390_4spk_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 i; + + for_each_rtd_codec_dais(rtd, i, codec_dai) { + + if (i >= ARRAY_SIZE(max_98390_4spk_components)) { + dev_err(codec_dai->dev, "invalid codec index %d\n", i); + return -ENODEV; + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV0_NAME)) { + /* DEV0 tdm slot configuration Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x01, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV1_NAME)) { + /* DEV1 tdm slot configuration Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x02, 3, 4, 32); + } + + if (!strcmp(codec_dai->component->name, MAX_98390_DEV2_NAME)) { + /* DEVi2 tdm slot configuration Tweeter Right */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x04, 3, 4, 32); + } + if (!strcmp(codec_dai->component->name, MAX_98390_DEV3_NAME)) { + /* DEV3 tdm slot configuration Tweeter Left */ + snd_soc_dai_set_tdm_slot(codec_dai, 0x08, 3, 4, 32); + } + } + return 0; +} + +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + /* add regular speakers dapm route */ + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_dapm_routes, + ARRAY_SIZE(max_98390_dapm_routes)); + if (ret) { + dev_err(rtd->dev, "unable to add Left/Right Speaker dapm, ret %d\n", ret); + return ret; + } + + /* add widgets/controls/dapm for tweeter speakers */ + if (acpi_dev_present("MX98390", "3", -1)) { + ret = snd_soc_dapm_new_controls(&card->dapm, max_98390_tt_dapm_widgets, + ARRAY_SIZE(max_98390_tt_dapm_widgets)); + + if (ret) { + dev_err(rtd->dev, "unable to add tweeter dapm controls, ret %d\n", ret); + /* Don't need to add routes if widget addition failed */ + return ret; + } + + ret = snd_soc_add_card_controls(card, max_98390_tt_kcontrols, + ARRAY_SIZE(max_98390_tt_kcontrols)); + if (ret) { + dev_err(rtd->dev, "unable to add tweeter card controls, ret %d\n", ret); + return ret; + } + + ret = snd_soc_dapm_add_routes(&card->dapm, max_98390_tt_dapm_routes, + ARRAY_SIZE(max_98390_tt_dapm_routes)); + if (ret) + dev_err(rtd->dev, + "unable to add Tweeter Left/Right Speaker dapm, ret %d\n", ret); + } + return ret; +} +EXPORT_SYMBOL_NS(max_98390_spk_codec_init, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +struct snd_soc_ops max_98390_ops = { + .hw_params = max_98390_hw_params, +}; +EXPORT_SYMBOL_NS(max_98390_ops, SND_SOC_INTEL_SOF_MAXIM_COMMON); + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch) +{ + if (ch == ARRAY_SIZE(max_98390_4spk_codec_conf)) { + card->codec_conf = max_98390_4spk_codec_conf; + card->num_configs = ARRAY_SIZE(max_98390_4spk_codec_conf); + } else { + 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/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_maxim_common.h b/sound/soc/intel/boards/sof_maxim_common.h index 3ff5e8fec4de..9dd69f5d8721 100644 --- a/sound/soc/intel/boards/sof_maxim_common.h +++ b/sound/soc/intel/boards/sof_maxim_common.h @@ -24,6 +24,22 @@ 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" +#define MAX_98390_DEV2_NAME "i2c-MX98390:02" +#define MAX_98390_DEV3_NAME "i2c-MX98390:03" + +extern struct snd_soc_dai_link_component max_98390_components[2]; +extern struct snd_soc_dai_link_component max_98390_4spk_components[4]; +extern struct snd_soc_ops max_98390_ops; + +void max_98390_set_codec_conf(struct snd_soc_card *card, int ch); +int max_98390_spk_codec_init(struct snd_soc_pcm_runtime *rtd); + /* * Maxim MAX98357A/MAX98360A */ diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 3f6f19d9b19e..774e4dd9539c 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -59,6 +59,9 @@ (((quirk) << SOF_BT_OFFLOAD_SSP_SHIFT) & SOF_BT_OFFLOAD_SSP_MASK) #define SOF_SSP_BT_OFFLOAD_PRESENT BIT(22) #define SOF_RT5682S_HEADPHONE_CODEC_PRESENT BIT(23) +#define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) +#define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) + /* Default: MCLK on, MCLK 19.2M, SSP0 */ static unsigned long sof_rt5682_quirk = SOF_RT5682_MCLK_EN | @@ -179,6 +182,36 @@ 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)), + }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S_4SPK"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_MAX98390_TWEETER_SPEAKER_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + + }, {} }; @@ -486,6 +519,7 @@ static int sof_card_late_probe(struct snd_soc_card *card) if (err < 0) return err; } + return hdac_hdmi_jack_port_init(component, &card->dapm); } @@ -784,6 +818,20 @@ 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) { + if (sof_rt5682_quirk & + SOF_MAX98390_TWEETER_SPEAKER_PRESENT) { + links[id].codecs = max_98390_4spk_components; + links[id].num_codecs = ARRAY_SIZE(max_98390_4spk_components); + } else { + 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]); } @@ -864,6 +912,10 @@ static int sof_audio_probe(struct platform_device *pdev) if ((sof_rt5682_quirk & SOF_SPEAKER_AMP_PRESENT) && !mach->quirk_data) sof_rt5682_quirk &= ~SOF_SPEAKER_AMP_PRESENT; + /* Detect the headset codec variant to support machines in DMI quirk */ + if (acpi_dev_present("RTL5682", NULL, -1)) + sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; dmic_be_num = 0; @@ -920,6 +972,14 @@ 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) { + if (sof_rt5682_quirk & SOF_MAX98390_TWEETER_SPEAKER_PRESENT) + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_4spk_components)); + else + max_98390_set_codec_conf(&sof_audio_card_rt5682, + ARRAY_SIZE(max_98390_components)); + } if (sof_rt5682_quirk & SOF_SSP_BT_OFFLOAD_PRESENT) sof_audio_card_rt5682.num_links++; @@ -1046,6 +1106,29 @@ 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(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, + { + .name = "adl_mx98390_rt5682s", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_RT5682S_HEADPHONE_CODEC_PRESENT | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98390_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1) | + SOF_RT5682_NUM_HDMIDEV(4) | + SOF_BT_OFFLOAD_SSP(2) | + SOF_SSP_BT_OFFLOAD_PRESENT), + }, { .name = "jsl_rt5682s_rt1015", .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | @@ -1095,6 +1178,7 @@ MODULE_DESCRIPTION("SOF Audio Machine driver"); MODULE_AUTHOR("Bard Liao <bard.liao(a)intel.com>"); MODULE_AUTHOR("Sathya Prakash M R <sathya.prakash.m.r(a)intel.com>"); MODULE_AUTHOR("Brent Lu <brent.lu(a)intel.com>"); +MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); MODULE_IMPORT_NS(SND_SOC_INTEL_SOF_MAXIM_COMMON); 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..a62a78edc855 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 +302,22 @@ 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", + }, + { + .id = "RTL5682", + .drv_name = "adl_mx98390_rt5682s", + .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); diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index bc9e70765678..be90a5f79766 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -59,6 +59,15 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-rt5682-ssp0-max98373-ssp2.tplg", }, + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98390_ALC5682I_I2S"), + }, + .driver_data = "sof-adl-max98390-ssp2-rt5682-ssp0.tplg", + }, + {} }; -- 2.20.1

2 1

include a sample `alsamixer.rc` file?
by GitHub issues - opened 24 Sep '21

24 Sep '21

alsa-project/alsa-utils issue #112 was opened from int-e: I think it would be useful to include an example `alsamixer.rc` file; see https://gist.github.com/int-e/25bd77d4e59531fad7d48bfaf1fc1046 for a candidate. In particular this example replicates the default color theme, making it easier to adjust (say, replace `default` by `black` to get sane colors back in terminals with bright backgrounds). The default keybindings are also replicated but that's probably less useful. Issue URL : https://github.com/alsa-project/alsa-utils/issues/112 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v2] ALSA: rawmidi: introduce SNDRV_RAWMIDI_IOCTL_USER_PVERSION
by Jaroslav Kysela 23 Sep '21

23 Sep '21

The new framing mode causes the user space regression, because the alsa-lib code does not initialize the reserved space in the params structure when the device is opened. This change adds SNDRV_RAWMIDI_IOCTL_USER_PVERSION like we do for the PCM interface for the protocol acknowledgment. Cc: David Henningsson <coding(a)diwic.se> Cc: <stable(a)vger.kernel.org> Fixes: 08fdced60ca0 ("ALSA: rawmidi: Add framing mode") BugLink: https://github.com/alsa-project/alsa-lib/issues/178 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- include/sound/rawmidi.h | 1 + include/uapi/sound/asound.h | 1 + sound/core/rawmidi.c | 9 +++++++++ 3 files changed, 11 insertions(+) diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index 989e1517332d..7a08ed2acd60 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -98,6 +98,7 @@ struct snd_rawmidi_file { struct snd_rawmidi *rmidi; struct snd_rawmidi_substream *input; struct snd_rawmidi_substream *output; + unsigned int user_pversion; /* supported protocol version */ }; struct snd_rawmidi_str { diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 1d84ec9db93b..5859ca0a1439 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -784,6 +784,7 @@ struct snd_rawmidi_status { #define SNDRV_RAWMIDI_IOCTL_PVERSION _IOR('W', 0x00, int) #define SNDRV_RAWMIDI_IOCTL_INFO _IOR('W', 0x01, struct snd_rawmidi_info) +#define SNDRV_RAWMIDI_IOCTL_USER_PVERSION _IOW('W', 0x02, int) #define SNDRV_RAWMIDI_IOCTL_PARAMS _IOWR('W', 0x10, struct snd_rawmidi_params) #define SNDRV_RAWMIDI_IOCTL_STATUS _IOWR('W', 0x20, struct snd_rawmidi_status) #define SNDRV_RAWMIDI_IOCTL_DROP _IOW('W', 0x30, int) diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 6c0a4a67ad2e..6f30231bdb88 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long return -EINVAL; } } + case SNDRV_RAWMIDI_IOCTL_USER_PVERSION: + if (get_user(rfile->user_pversion, (unsigned int __user *)arg)) + return -EFAULT; + return 0; + case SNDRV_RAWMIDI_IOCTL_PARAMS: { struct snd_rawmidi_params params; if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params))) return -EFAULT; + if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) { + params.mode = 0; + memset(params.reserved, 0, sizeof(params.reserved)); + } switch (params.stream) { case SNDRV_RAWMIDI_STREAM_OUTPUT: if (rfile->output == NULL) -- 2.31.1

2 1

[PATCH v7 00/22] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 22 Sep '21

22 Sep '21

Many thanks for reviewing v6 This version addresses all the comments raised as part of v7 review. 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 simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | 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/ and Qualcomm AudioReach DSP headers are available at: https://source.codeaurora.org/quic/la/platform/vendor/opensource/arspf-head… Note: There is one false positive warning in this patchset: audioreach.c:80:45: warning: array of flexible structures Thanks, srini Changes since v6: - fixed last usage of flexible array in apr - removed leftover comments - added support to derive topology firmware name from soundcard to support different variants, as suggested by Pierre. Srinivas Kandagatla (22): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h 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-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings 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 Kconfig and Makefile ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai 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 | 177 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 53 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 77 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 205 +++ drivers/soc/qcom/Kconfig | 2 +- drivers/soc/qcom/apr.c | 287 ++++- include/dt-bindings/soc/qcom,gpr.h | 19 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 208 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1135 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 726 +++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 415 ++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 260 ++++ sound/soc/qcom/qdsp6/q6apm.c | 836 ++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 152 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 202 +++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1133 ++++++++++++++++ 31 files changed, 7203 insertions(+), 1415 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-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-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.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-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-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

2 31

[PATCH 01/16] ASoC: eureka-tlv320: Update to modern clocking terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the eureka-tlv320 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e13271ea84de..8b61582753c8 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &eukrea_tlv320_snd_ops, SND_SOC_DAILINK_REG(hifi), }; -- 2.20.1

2 18

[PATCH] ASoC: cros_ec_codec: Use modern ASoC DAI format terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the cros_ec_codec driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/cros_ec_codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index a201d652aca2..9b92e1a0d1a3 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -283,8 +283,8 @@ static int i2s_rx_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct ec_param_ec_codec_i2s_rx p; enum ec_codec_i2s_rx_daifmt daifmt; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: break; default: return -EINVAL; -- 2.20.1

2 2

[PATCH v2 01/16] ASoC: eureka-tlv320: Update to modern clocking terminology
by Mark Brown 22 Sep '21

22 Sep '21

As part of moving to remove the old style defines for the bus clocks update the eureka-tlv320 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index e13271ea84de..8b61582753c8 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -70,7 +70,7 @@ static struct snd_soc_dai_link eukrea_tlv320_dai = { .name = "tlv320aic23", .stream_name = "TLV320AIC23", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &eukrea_tlv320_snd_ops, SND_SOC_DAILINK_REG(hifi), }; -- 2.20.1

2 17

Redesign the rawmidi midi read with timestamping API
by GitHub pull_request - opened 22 Sep '21

22 Sep '21

alsa-project/alsa-lib pull request #179 was opened from perexg: It's follow-up for https://github.com/alsa-project/alsa-lib/pull/173 and https://github.com/alsa-project/alsa-lib/issues/172 . I am confident, that the direct framing mechanism for the transfer of the midi stream with the timestamping should be hidden and abstracted in the alsa-lib's API. This version uses a new read function (`snd_rawmidi_tread()`) which decodes and merges the data from the kernel frames. The framing mechanism is very specific to the kernel API and the cost of added data copy is negligible in this case (the kernel code basically does similar thing). I kept the `snd_rawmidi_read()` functionality to read raw frames), but the structure for the parsing must be defined on it's own. This mechanism is unsupported. This version seems more abstract and easy to use for apps than the separate decoder function for `snd_rawmidi_read()`. Request URL : https://github.com/alsa-project/alsa-lib/pull/179 Patch URL : https://github.com/alsa-project/alsa-lib/pull/179.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[RFC PATCH] component: do not leave master devres group open after bind
by Kai Vehmanen 22 Sep '21

22 Sep '21

In current code, the devres group for aggregate master is left open after call to component_master_add_*(). This leads to problems when the master does further managed allocations on its own. When any participating driver calls component_del(), this leads to immediate release of resources. This came up when investigating a page fault occurring with i915 DRM driver unbind with 5.15-rc1 kernel. The following sequence occurs: i915_pci_remove() -> intel_display_driver_unregister() -> i915_audio_component_cleanup() -> component_del() -> component.c:take_down_master() -> hdac_component_master_unbind() [via master->ops->unbind()] -> devres_release_group(master->parent, NULL) With older kernels this has not caused issues, but with audio driver moving to use managed interfaces for more of its allocations, this no longer works. Devres log shows following to occur: component_master_add_with_match() [ 126.886032] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000323ccdc5 devm_component_match_release (24 bytes) [ 126.886045] snd_hda_intel 0000:00:1f.3: DEVRES ADD 00000000865cdb29 grp< (0 bytes) [ 126.886049] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 grp< (0 bytes) audio driver completes its PCI probe() [ 126.892238] snd_hda_intel 0000:00:1f.3: DEVRES ADD 000000001b480725 pcim_iomap_release (48 bytes) component_del() called() at DRM/i915 unbind() [ 137.579422] i915 0000:00:02.0: DEVRES REL 00000000ef44c293 grp< (0 bytes) [ 137.579445] snd_hda_intel 0000:00:1f.3: DEVRES REL 00000000865cdb29 grp< (0 bytes) [ 137.579458] snd_hda_intel 0000:00:1f.3: DEVRES REL 000000001b480725 pcim_iomap_release (48 bytes) So the "devres_release_group(master->parent, NULL)" ends up freeing the pcim_iomap allocation. Upon next runtime resume, the audio driver will cause a page fault as the iomap alloc was released without the driver knowing about it. Fix this issue by using the "struct master" pointer as identifier for the devres group, and by closing the devres group after the master->ops->bind() call is done. This allows devres allocations done by the driver acting as master to be isolated from the binding state of the aggregate driver. This modifies the logic originally introduced in commit 9e1ccb4a7700 ("drivers/base: fix devres handling for master device"). BugLink: https://gitlab.freedesktop.org/drm/intel/-/issues/4136 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- drivers/base/component.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) Hi, I'm sending this as RFC as I'm not sure of the implications of not leaving the devres group open might have to other users of the component framework. For audio, the current behaviour seems very problematic. The display codec is usually just one of many audio codecs attached to the controller, and unbind of the display codec (and the aggregate driver created with DRM), should not bring down the whole audio card. However, now all allocations audio driver does after call to component_master_add_with_match(), will be freed when display driver calls component_del(). Closing the devres group at end of component_master_add_*() would seem the cleanest option. Looking for feedback whether this approach is feasible. One alternative would be for the audio driver to close the "last opened" group after its call to component_master_add(), but this seems messy (audio would make assumptions on component.c internals). diff --git a/drivers/base/component.c b/drivers/base/component.c index 5e79299f6c3f..870485cbbb87 100644 --- a/drivers/base/component.c +++ b/drivers/base/component.c @@ -246,7 +246,7 @@ static int try_to_bring_up_master(struct master *master, return 0; } - if (!devres_open_group(master->parent, NULL, GFP_KERNEL)) + if (!devres_open_group(master->parent, master, GFP_KERNEL)) return -ENOMEM; /* Found all components */ @@ -258,6 +258,7 @@ static int try_to_bring_up_master(struct master *master, return ret; } + devres_close_group(master->parent, NULL); master->bound = true; return 1; } @@ -282,7 +283,7 @@ static void take_down_master(struct master *master) { if (master->bound) { master->ops->unbind(master->parent); - devres_release_group(master->parent, NULL); + devres_release_group(master->parent, master); master->bound = false; } } base-commit: 930e99a51fcc8b1254e0a45fbe0cd5a5b8a704a5 -- 2.32.0

3 2

Re: [LAD] Mackie Onyx 1640i: Long run playback results in xrun and hang.
by AreYouLoco? 21 Sep '21

21 Sep '21

I am going to recompile the kernel with options mentioned here: https://www.alsa-project.org/main/index.php/XRUN_Debug And try to debug why the xrun appears. I also checked with buffer size 1024 and exaclty one xrun and then hang repeats. On 9/13/21 12:23 PM, AreYouLoco? wrote: > Hi, > > I am using custom build real-time kernel 5.12.2-rt3-fw. JACK with ALSA > backend on Mackie Onyx 1640i. > Lenovo Thinkpad T420 with FireWire. Debian Bullseye (11.0 - stable) > JACKD version 1.9.17. > I am using 512 samples 3 periods @48kHz. > > The problem is that long run playback only around 1h (but the time > differs), result in > XRUN detected by JACK logs and the whole audio stops. So I have to > restart JACK in order to make it work. > DMESG is silent about that issue. But sometimes I get: > > snd_oxfw fw1.0: Detect discontinuity of CIP: E8 00 > And different hex values there. > > Here is interesting part from the log: > > Sun Sep 12 22:10:12 2021: Jack: JackRequest::Notification > Sun Sep 12 22:10:12 2021: Jack: **** alsa_pcm: xrun of at least 0.027 msecs > Sun Sep 12 22:10:12 2021: Jack: Repreparing capture > Sun Sep 12 22:10:12 2021: Jack: Repreparing playback > Sun Sep 12 22:10:12 2021: Jack: ALSA XRun wait_status = 0 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackClient::ClientNotify ref = 2 name = > dbusapi notify = 3 > Sun Sep 12 22:10:12 2021: Jack: JackClient::kXRunCallback > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:12 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 22:10:28 2021: ERROR: ALSA: poll time out, polled for > 15998655 usecs, Retrying with a recovery, retry cnt = 1 > Sun Sep 12 22:10:44 2021: ERROR: ALSA: poll time out, polled for > 15998654 usecs, Retrying with a recovery, retry cnt = 2 > Sun Sep 12 22:11:00 2021: ERROR: ALSA: poll time out, polled for > 15998659 usecs, Retrying with a recovery, retry cnt = 3 > Sun Sep 12 22:11:16 2021: ERROR: ALSA: poll time out, polled for > 15998649 usecs, Retrying with a recovery, retry cnt = 4 > > Anyone have idea what might be the cause of this XRUN? Or how to debug that? > > Its reproducible on my setup. Another log catched the same issue: > > Sun Sep 12 16:50:44 2021: Jack: **** alsa_pcm: xrun of at least 0.030 msecs > Sun Sep 12 16:50:44 2021: Jack: Repreparing capture > Sun Sep 12 16:50:44 2021: Jack: Repreparing playback > Sun Sep 12 16:50:44 2021: Jack: ALSA XRun wait_status = 0 > Sun Sep 12 16:50:44 2021: Jack: JackRequest::Notification > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackClient::ClientNotify ref = 2 name = > dbusapi notify = 3 > Sun Sep 12 16:50:44 2021: Jack: JackClient::kXRunCallback > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:50:44 2021: Jack: JackEngine::ClientNotify: no callback > for notification = 3 > Sun Sep 12 16:51:00 2021: ERROR: ALSA: poll time out, polled for > 15998366 usecs, Retrying with a recovery, retry cnt = 1 > Sun Sep 12 16:51:16 2021: ERROR: ALSA: poll time out, polled for > 15998353 usecs, Retrying with a recovery, retry cnt = 2 > Sun Sep 12 16:51:32 2021: ERROR: ALSA: poll time out, polled for > 15998352 usecs, Retrying with a recovery, retry cnt = 3 > Sun Sep 12 16:51:48 2021: ERROR: ALSA: poll time out, polled for > 15998367 usecs, Retrying with a recovery, retry cnt = 4 > Sun Sep 12 16:52:04 2021: ERROR: ALSA: poll time out, polled for > 15998353 usecs, Retrying with a recovery, retry cnt = 5 > Sun Sep 12 16:52:20 2021: ERROR: ALSA: poll time out, polled for > 15998371 usecs, Reached max retry cnt = 5, Exiting > Sun Sep 12 16:52:20 2021: ERROR: JackAudioDriver::ProcessAsync: read > error, stopping... > Sun Sep 12 16:52:20 2021: Jack: JackPosixThread::ThreadHandler : exit > > I am willing to provide additional info. > > Thank you in advance, > > AreYouLoco? > _______________________________________________ > Linux-audio-dev mailing list > Linux-audio-dev(a)lists.linuxaudio.org > https://lists.linuxaudio.org/listinfo/linux-audio-dev

1 1

[PATCH] ASoC: Fix warning related to 'sound-name-prefix' binding
by Sameer Pujar 21 Sep '21

21 Sep '21

commit 82d3ec1d89fa ("ASoC: Use schema reference for sound-name-prefix") added name-prefix.yaml schema and the same reference was used in couple of other schemas. But this is causing following warning and the same is fixed in current patch. Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml: audio-codec@34: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml Documentation/devicetree/bindings/sound/nxp,tfa989x.example.dt.yaml: audio-codec@36: 'sound-name-prefix' does not match any of the regexes: 'pinctrl-[0-9]+' From schema: Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml Fixes: 82d3ec1d89fa ("ASoC: Use schema reference for sound-name-prefix") Reported-by: Rob Herring <robh+dt(a)kernel.org> Suggested-by: Rob Herring <robh+dt(a)kernel.org> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml | 2 ++ Documentation/devicetree/bindings/sound/simple-audio-mux.yaml | 2 ++ 2 files changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml index 68e5ad2..7667471 100644 --- a/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml +++ b/Documentation/devicetree/bindings/sound/nxp,tfa989x.yaml @@ -24,6 +24,8 @@ properties: '#sound-dai-cells': const: 0 + sound-name-prefix: true + vddd-supply: description: regulator phandle for the VDDD power supply. diff --git a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml index c597d9d..b5fc35e 100644 --- a/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml +++ b/Documentation/devicetree/bindings/sound/simple-audio-mux.yaml @@ -24,6 +24,8 @@ properties: description: | GPIOs used to select the input line. + sound-name-prefix: true + required: - compatible - mux-gpios -- 2.7.4

2 1

[PATCH] ASoC: pcm512x: Mend accesses to the I2S_1 and I2S_2 registers
by Peter Rosin 21 Sep '21

21 Sep '21

>From 625f858894af2b7e547cc723b97361081438b123 Mon Sep 17 00:00:00 2001 From: Peter Rosin <peda(a)axentia.se> Commit 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") breaks the TSE-850 device, which is using a pcm5142 in I2S and CBM_CFS mode (maybe not relevant). Without this fix, the result is: pcm512x 0-004c: Failed to set data format: -16 And after that, no sound. This fix is not 100% correct. The datasheet of at least the pcm5142 states that four bits (0xcc) in the I2S_1 register are "RSV" ("Reserved. Do not access.") and no hint is given as to what the initial values are supposed to be. So, specifying defaults for these bits is wrong. But perhaps better than a broken driver? Fixes: 25d27c4f68d2 ("ASoC: pcm512x: Add support for more data formats") 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: Kirill Marinushkin <kmarinushkin(a)birdec.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4dc844f3c1fc..60dee41816dc 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -116,6 +116,8 @@ static const struct reg_default pcm512x_reg_defaults[] = { { PCM512x_FS_SPEED_MODE, 0x00 }, { PCM512x_IDAC_1, 0x01 }, { PCM512x_IDAC_2, 0x00 }, + { PCM512x_I2S_1, 0x02 }, + { PCM512x_I2S_2, 0x00 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) -- 2.20.1

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[PATCH 0/7] add internal channels support
by Olivier Moysan 21 Sep '21

21 Sep '21

This patchset adds support of ADC2 internal channels VDDCORE, VREFINT and VBAT on STM32MP15x SoCs. The generic IIO channel bindings is also introduced here to provide this feature. The legacy channel binding is kept for backward compatibility. Olivier Moysan (7): dt-bindings: iio: adc: add generic channel binding dt-bindings: iio: adc: add nvmem support for vrefint internal channel iio: adc stm32-adc: split channel init into several routines iio: adc: stm32-adc: add support of generic channels binding iio: adc: stm32-adc: add support of internal channels iio: adc: stm32-adc: add vrefint calibration support iio: adc: stm32-adc: use generic binding for sample-time .../bindings/iio/adc/st,stm32-adc.yaml | 108 ++++- drivers/iio/adc/stm32-adc-core.h | 8 + drivers/iio/adc/stm32-adc.c | 418 ++++++++++++++++-- 3 files changed, 482 insertions(+), 52 deletions(-) -- 2.17.1

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[PATCH] ALSA: firewire-motu: fix truncated bytes in message tracepoints
by Takashi Sakamoto 21 Sep '21

21 Sep '21

In MOTU protocol v2/v3, first two data chunks across 2nd and 3rd data channels includes message bytes from device. The total size of message is 48 bits per data block. The 'data_block_message' tracepoints event produced by ALSA firewire-motu driver exposes the sequence of messages to userspace in 64 bit storage, however lower 32 bits are actually available since current implementation truncates 16 bits in upper of the message as a result of bit shift operation within 32 bit storage. This commit fixes the bug by perform the bit shift in 64 bit storage. Fixes: c6b0b9e65f09 ("ALSA: firewire-motu: add tracepoints for messages for unique protocol") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index 5388b85fb60e..a18c2c033e83 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer, /* This is just for v2/v3 protocol. */ for (i = 0; i < data_blocks; ++i) { - *frames = (be32_to_cpu(buffer[1]) << 16) | - (be32_to_cpu(buffer[2]) >> 16); + *frames = be32_to_cpu(buffer[1]); + *frames <<= 16; + *frames |= be32_to_cpu(buffer[2]) >> 16; + ++frames; buffer += data_block_quadlets; - frames++; } } -- 2.30.2

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[GIT PULL] ASoC fixes for v5.15-rc2
by Mark Brown 21 Sep '21

21 Sep '21

The following changes since commit fcb958ee8e832e9cdf43408535207e15f14af755: ASoC: rockchip: i2s: Fix concurrency between tx/rx (2021-09-03 16:19:01 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc2 for you to fetch changes up to cfacfefd382af3b42905108b54f02820dca225c4: ASoC: SOF: trace: Omit error print when waking up trace sleepers (2021-09-17 13:16:36 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A crop of mostly device specific fixes that have been applied since the merge window, nothing particularly standout here. ---------------------------------------------------------------- Arnaud Pouliquen (1): MAINTAINERS: fix update references to stm32 audio bindings Marc Herbert (1): ASoC: SOF: loader: release_firmware() on load failure to avoid batching Mark Brown (1): Merge series "ASoC: fsl: register platform component before registering cpu dai" from Shengjiu Wang <shengjiu.wang(a)nxp.com>: Peter Ujfalusi (4): ASoC: SOF: imx: imx8: Bar index is only valid for IRAM and SRAM types ASoC: SOF: imx: imx8m: Bar index is only valid for IRAM and SRAM types ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication ASoC: SOF: trace: Omit error print when waking up trace sleepers Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: tag SoundWire BEs as non-atomic Shengjiu Wang (5): ASoC: fsl_sai: register platform component before registering cpu dai ASoC: fsl_esai: register platform component before registering cpu dai ASoC: fsl_micfil: register platform component before registering cpu dai ASoC: fsl_spdif: register platform component before registering cpu dai ASoC: fsl_xcvr: register platform component before registering cpu dai Trevor Wu (3): ASoC: mediatek: add required config dependency ASoC: mediatek: common: handle NULL case in suspend/resume function ASoC: mediatek: mt8195: remove wrong fixup assignment on HDMITX Yong Zhi (1): ASoC: SOF: Fix DSP oops stack dump output contents MAINTAINERS | 3 ++- sound/soc/fsl/fsl_esai.c | 16 ++++++++++------ sound/soc/fsl/fsl_micfil.c | 15 ++++++++++----- sound/soc/fsl/fsl_sai.c | 14 +++++++++----- sound/soc/fsl/fsl_spdif.c | 14 +++++++++----- sound/soc/fsl/fsl_xcvr.c | 15 ++++++++++----- sound/soc/intel/boards/sof_sdw.c | 5 +++++ sound/soc/mediatek/Kconfig | 3 +++ sound/soc/mediatek/common/mtk-afe-fe-dai.c | 19 +++++++++++-------- .../soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 7 +++---- sound/soc/sof/core.c | 4 +--- sound/soc/sof/imx/imx8.c | 9 ++++++++- sound/soc/sof/imx/imx8m.c | 9 ++++++++- sound/soc/sof/loader.c | 8 +++++--- sound/soc/sof/trace.c | 1 - sound/soc/sof/xtensa/core.c | 4 ++-- 16 files changed, 96 insertions(+), 50 deletions(-)

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Re: [PATCH] ALSA: usx2y: Prefer struct_size over open coded arithmetic
by Takashi Iwai 21 Sep '21

21 Sep '21

On Sun, 19 Sep 2021 15:37:27 +0200, Len Baker wrote: > > As noted in the "Deprecated Interfaces, Language Features, Attributes, > and Conventions" documentation [1], size calculations (especially > multiplication) should not be performed in memory allocator (or similar) > function arguments due to the risk of them overflowing. This could lead > to values wrapping around and a smaller allocation being made than the > caller was expecting. Using those allocations could lead to linear > overflows of heap memory and other misbehaviors. > > In this case this is not actually dynamic size: all the operands > involved in the calculation are constant values. However it is better to > refactor this anyway, just to keep the open-coded math idiom out of > code. > > So, use the struct_size() helper to do the arithmetic instead of the > argument "size + size * count" in the kzalloc() function. > > Also, take the opportunity to refactor the declaration variables to make > it more easy to read. > > [1] https://www.kernel.org/doc/html/latest/process/deprecated.html#open-coded-a… > > Signed-off-by: Len Baker <len.baker(a)gmx.com> Thanks, applied. Takashi

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[PATCH] ASoC: fsl_spdif: Add support for i.MX8ULP
by Shengjiu Wang 21 Sep '21

21 Sep '21

On i.MX8ULP the spdif works with EDMA, so add compatible string and soc specific data for i.MX8ULP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 8ffb1a6048d6..61ba08322f23 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -186,6 +186,16 @@ static struct fsl_spdif_soc_data fsl_spdif_imx8mm = { .tx_formats = FSL_SPDIF_FORMATS_PLAYBACK, }; +static struct fsl_spdif_soc_data fsl_spdif_imx8ulp = { + .imx = true, + .shared_root_clock = true, + .raw_capture_mode = false, + .interrupts = 1, + .tx_burst = 2, /* Applied for EDMA */ + .rx_burst = 2, /* Applied for EDMA */ + .tx_formats = SNDRV_PCM_FMTBIT_S24_LE, /* Applied for EDMA */ +}; + /* Check if clk is a root clock that does not share clock source with others */ static inline bool fsl_spdif_can_set_clk_rate(struct fsl_spdif_priv *spdif, int clk) { @@ -1556,6 +1566,7 @@ static const struct of_device_id fsl_spdif_dt_ids[] = { { .compatible = "fsl,imx6sx-spdif", .data = &fsl_spdif_imx6sx, }, { .compatible = "fsl,imx8qm-spdif", .data = &fsl_spdif_imx8qm, }, { .compatible = "fsl,imx8mm-spdif", .data = &fsl_spdif_imx8mm, }, + { .compatible = "fsl,imx8ulp-spdif", .data = &fsl_spdif_imx8ulp, }, {} }; MODULE_DEVICE_TABLE(of, fsl_spdif_dt_ids); -- 2.17.1

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[PATCH] ASoC: fsl_xcvr: Fix channel swap issue with ARC
by Shengjiu Wang 21 Sep '21

21 Sep '21

With pause and resume test for ARC, there is occasionally channel swap issue. The reason is that currently driver set the DPATH out of reset first, then start the DMA, the first data got from FIFO may not be the Left channel. Moving DPATH out of reset operation after the dma enablement to fix this issue. Fixes: 28564486866f ("ASoC: fsl_xcvr: Add XCVR ASoC CPU DAI driver") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 31c5ee641fe7..6e67033b6cde 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -487,8 +487,9 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, return ret; } - /* clear DPATH RESET */ + /* set DPATH RESET */ m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); + v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -590,10 +591,6 @@ static void fsl_xcvr_shutdown(struct snd_pcm_substream *substream, val |= FSL_XCVR_EXT_CTRL_CMDC_RESET(tx); } - /* set DPATH RESET */ - mask |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - val |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, mask, val); if (ret < 0) { dev_err(dai->dev, "Err setting DPATH RESET: %d\n", ret); @@ -643,6 +640,16 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, dev_err(dai->dev, "Failed to enable DMA: %d\n", ret); return ret; } + + /* clear DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear DPATH RESET: %d\n", ret); + return ret; + } + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: -- 2.17.1

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[PATCH] ASoC: qdsp6: q6afe-dai: Fix spelling mistake "Fronend" -> "Frontend"
by Colin King 21 Sep '21

21 Sep '21

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in the module description. Fix it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index ac8f7324e94b..8b664cbf6fa6 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -1747,5 +1747,5 @@ static struct platform_driver q6afe_dai_platform_driver = { }; module_platform_driver(q6afe_dai_platform_driver); -MODULE_DESCRIPTION("Q6 Audio Fronend dai driver"); +MODULE_DESCRIPTION("Q6 Audio Frontend dai driver"); MODULE_LICENSE("GPL v2"); -- 2.32.0

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[PATCH] ASoC: fsl: Constify static snd_soc_ops
by Rikard Falkeborn 21 Sep '21

21 Sep '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/fsl/imx-audmix.c | 4 ++-- sound/soc/fsl/imx-card.c | 4 ++-- sound/soc/fsl/imx-hdmi.c | 2 +- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/fsl/imx-audmix.c b/sound/soc/fsl/imx-audmix.c index a364e2415de0..0d637929bfef 100644 --- a/sound/soc/fsl/imx-audmix.c +++ b/sound/soc/fsl/imx-audmix.c @@ -132,12 +132,12 @@ static int imx_audmix_be_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops imx_audmix_fe_ops = { +static const struct snd_soc_ops imx_audmix_fe_ops = { .startup = imx_audmix_fe_startup, .hw_params = imx_audmix_fe_hw_params, }; -static struct snd_soc_ops imx_audmix_be_ops = { +static const struct snd_soc_ops imx_audmix_be_ops = { .hw_params = imx_audmix_be_hw_params, }; diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 58fd0639a069..05dff2dc1d19 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -442,12 +442,12 @@ static int imx_aif_startup(struct snd_pcm_substream *substream) return ret; } -static struct snd_soc_ops imx_aif_ops = { +static const struct snd_soc_ops imx_aif_ops = { .hw_params = imx_aif_hw_params, .startup = imx_aif_startup, }; -static struct snd_soc_ops imx_aif_ops_be = { +static const struct snd_soc_ops imx_aif_ops_be = { .hw_params = imx_aif_hw_params, }; diff --git a/sound/soc/fsl/imx-hdmi.c b/sound/soc/fsl/imx-hdmi.c index 34a0dceae621..2b663c39edb2 100644 --- a/sound/soc/fsl/imx-hdmi.c +++ b/sound/soc/fsl/imx-hdmi.c @@ -59,7 +59,7 @@ static int imx_hdmi_hw_params(struct snd_pcm_substream *substream, return 0; } -static struct snd_soc_ops imx_hdmi_ops = { +static const struct snd_soc_ops imx_hdmi_ops = { .hw_params = imx_hdmi_hw_params, }; -- 2.33.0

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[PATCH v3 0/3] Convert name-prefix doc to json-schema
by Sameer Pujar 21 Sep '21

21 Sep '21

Following are the changes: - Add json-schema for 'sound-name-prefix' documentation under 'name-perfix.yaml' - Use schema references wherever needed. - Remove txt based doc Changelog ========= v2 -> v3 -------- * Removed examples in patch 1/3 as suggested by Rob * Added reviewed tag from Jerome for the series * Added reviewed tag from Rob for patch 2/3 v1 -> v2 -------- * Provide top reference to name-prefix.yaml as suggested by Rob for patch 2/3 * Dropped couple of unreachable email ids from Cc list in commit message of patch 2/3 * No changes in remaining patches Sameer Pujar (3): ASoC: Add json-schema documentation for sound-name-prefix ASoC: Use schema reference for sound-name-prefix ASoC: Remove name-prefix.txt .../devicetree/bindings/sound/name-prefix.txt | 24 ---------------------- .../devicetree/bindings/sound/name-prefix.yaml | 21 +++++++++++++++++++ .../bindings/sound/nvidia,tegra186-dspk.yaml | 9 +++----- .../bindings/sound/nvidia,tegra210-dmic.yaml | 9 +++----- .../bindings/sound/nvidia,tegra210-i2s.yaml | 9 +++----- .../devicetree/bindings/sound/nxp,tfa989x.yaml | 9 +++----- Documentation/devicetree/bindings/sound/rt5659.txt | 2 +- .../bindings/sound/simple-audio-mux.yaml | 9 +++----- 8 files changed, 37 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/name-prefix.txt create mode 100644 Documentation/devicetree/bindings/sound/name-prefix.yaml -- 2.7.4

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