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[PATCH -next] ALSA: info: Remove unused function declarations
by Yue Haibing 08 Aug '23

08 Aug '23

These declarations is never used since beginning of git history. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- include/sound/info.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/include/sound/info.h b/include/sound/info.h index 7c13bf52cc81..adbc506860d6 100644 --- a/include/sound/info.h +++ b/include/sound/info.h @@ -118,8 +118,6 @@ struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent); void snd_info_free_entry(struct snd_info_entry *entry); -int snd_info_store_text(struct snd_info_entry *entry); -int snd_info_restore_text(struct snd_info_entry *entry); int snd_info_card_create(struct snd_card *card); int snd_info_card_register(struct snd_card *card); -- 2.34.1

2 1

[PATCH v2 1/2] ALSA: pcmtest: Move buffer iterator initialization to prepare callback
by Ivan Orlov 08 Aug '23

08 Aug '23

Trigger callback is not the best place for buffer iterator initialization, so move it out to the prepare callback, where it have to be. Minor enhancement: remove blank line. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- V1 -> V2: - Split changes in the patch into two different patches sound/drivers/pcmtest.c | 33 ++++++++++++++++----------------- 1 file changed, 16 insertions(+), 17 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 08e14b5eb772..dc7c9c758537 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -92,7 +92,6 @@ MODULE_PARM_DESC(inject_trigger_err, "Inject EINVAL error in the 'trigger' callb module_param(inject_open_err, bool, 0600); MODULE_PARM_DESC(inject_open_err, "Inject EBUSY error in the 'open' callback"); - struct pcmtst { struct snd_pcm *pcm; struct snd_card *card; @@ -405,25 +404,9 @@ static int snd_pcmtst_pcm_close(struct snd_pcm_substream *substream) static int snd_pcmtst_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_pcm_runtime *runtime = substream->runtime; - struct pcmtst_buf_iter *v_iter = runtime->private_data; - if (inject_trigger_err) return -EINVAL; - v_iter->sample_bytes = runtime->sample_bits / 8; - v_iter->period_bytes = frames_to_bytes(runtime, runtime->period_size); - if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || - runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { - v_iter->chan_block = runtime->dma_bytes / runtime->channels; - v_iter->interleaved = false; - } else { - v_iter->interleaved = true; - } - // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes - v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; - v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; - return 0; } @@ -454,8 +437,24 @@ static void pcmtst_pdev_release(struct device *dev) static int snd_pcmtst_pcm_prepare(struct snd_pcm_substream *substream) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct pcmtst_buf_iter *v_iter = runtime->private_data; + if (inject_prepare_err) return -EINVAL; + + v_iter->sample_bytes = samples_to_bytes(runtime, 1); + v_iter->period_bytes = snd_pcm_lib_period_bytes(substream); + v_iter->interleaved = true; + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) { + v_iter->chan_block = snd_pcm_lib_buffer_bytes(substream) / runtime->channels; + v_iter->interleaved = false; + } + // We want to record RATE * ch_cnt samples per sec, it is rate * sample_bytes * ch_cnt bytes + v_iter->s_rw_ch = runtime->rate / TIMER_PER_SEC; + v_iter->b_rw = v_iter->s_rw_ch * v_iter->sample_bytes * runtime->channels; + return 0; } -- 2.34.1

2 2

[PATCH v2 00/28] Add support for QMC HDLC, framer infrastruture and PEF2256 framer
by Herve Codina 08 Aug '23

08 Aug '23

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2, 3, 4) 2) Introduce the QMC HDLC driver (patches 5, 6, 7) 3) Add QMC dynamic timeslot support (patches 8 - 18) 4) Add timeslots change support in QMC HDLC (patch 19) 5) Introduce framer infrastructure (patch 20) 6) Add PEF2256 framer provider (patches 11, 22, 23, 24, 25) 7) Add framer codec as a framer consumer (patch 26) 8) Add framer support as a framer consumer in QMC HDLC (patch 27, 28) The series contains the full story and detailed modifications. If needed, the series can be split and/or commmits can be squashed. Let me know. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230725092417.43706-1-herve.codina@bo… This v2 series mainly: - Adds 2 patches to fix some compilation warning present on existing files (__iomem addresses declaration issues). - Fixes compilation warning on some patches. - Reword a commit log. Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Changes v1 -> v2 - Patches 1, 2 (New in v2) Fix __iomem addresses declaration - Patch 19 (17 in v1) Fix a compilation warning - Patch 26 (24 in v1) Fix a typo in Kconfig file Fix issues raised by sparse (make C=1) Herve Codina (28): soc: fsl: cpm1: tsa: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix __iomem addresses declaration soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status dt-bindings: net: Add support for QMC HDLC net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer mfd: core: Ensure disabled devices are skiped without aborting net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec dt-bindings: net: fsl,qmc-hdlc: Add framer support net: wan: fsl_qmc_hdlc: Add framer support .../devicetree/bindings/net/fsl,qmc-hdlc.yaml | 46 + .../bindings/net/lantiq,pef2256.yaml | 226 +++++ MAINTAINERS | 17 + drivers/mfd/mfd-core.c | 18 +- drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 35 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 935 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 +++++++++++++++ drivers/pinctrl/Kconfig | 14 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256-regs.h | 65 ++ drivers/pinctrl/pinctrl-pef2256.c | 310 ++++++ drivers/soc/fsl/qe/qmc.c | 501 ++++++++-- drivers/soc/fsl/qe/tsa.c | 22 +- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 215 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 25 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 423 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 27 files changed, 4957 insertions(+), 122 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/fsl,qmc-hdlc.yaml create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256-regs.h create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

8 90

[PATCH 00/27] ASoC: Map missing jack kcontrols
by Alper Nebi Yasak 08 Aug '23

08 Aug '23

This patchset adds missing jack kcontrols for each independently detectable audio peripheral and maps jack pins to those kcontrols accordingly, with the primary intent to enable/improve jack detection handling in PulseAudio and PipeWire through JackControl UCM values. Usually it's just splitting a joint anything-is-connected "Headset Jack" kcontrol (from [1]) into those like "Headphone Jack" and "Headset Mic" (similar to a previous series for Intel Chromebooks [2]). This split is important to avoid automatically switching to a nonexistent external microphone when a headphone-only device is connected. When the underlying hardware seems to support it, this also adds a "Line Out" kcontrol. This is important in case the hardware can actually support a line-level connection via a different configuration (bypassing output amplifiers?), or simply for userspace to display "Line Out" instead of "Headphones" to the user for connected line-out devices. Beyond the mappings, I had to add PIN_SWITCH card kcontrols and DAPM widgets to avoid "unknown pin" errors on my devices, so tried to do them for all. For Intel devices I saw a pattern of routing things to "Platform Clock" and added to that as well. Looking at patch 5/7 of a Mediatek-related series [3], I can only guess that routes could be further improved, but don't know exactly how for each device. And one more concern is I don't know if the names conflict with any controls from codecs, although I tried to keep to names of existing widgets. As far as I can tell, the root cause for most of why these are missing originates to things being developed for ChromeOS, whose userspace reads the jack input device and doesn't care for these kcontrols. There's non-ChromeOS cases as well, maybe things got copy-pasted around and people didn't need or couldn't figure out how to get more specific than a single jack kcontrol. The secondary intent in this patchset is to fix this *everywhere*, so future copy-pastes result in the right behaviour. For more context also see: [1] ASoC: soc-card: Create jack kcontrol without pins https://lore.kernel.org/alsa-devel/20220408041114.6024-1-akihiko.odaki@gmai… [2] ASoC: Intel: Chromebooks: remap jack pins https://lore.kernel.org/alsa-devel/20220616214055.134943-1-pierre-louis.bos… [3] ASoC: mediatek: Allow separate handling of headphone and headset mic jack https://lore.kernel.org/alsa-devel/20220922235951.252532-1-nfraprado@collab… [4] ASoC: rk3399_gru_sound: Add DAPM pins, kcontrols for jack detection https://lore.kernel.org/alsa-devel/20200721182709.6895-1-alpernebiyasak@gma… This applies onto next-20230802. Unfortunately most of it is untested except for a few Chromebooks I have (Kevin, Lick, Hana, Cozmo), because I'm intentionally generalizing to everything. Alper Nebi Yasak (27): ASoC: amd: acp-da7219-max98357a: Map missing jack kcontrols ASoC: amd: acp-rt5645: Map missing jack kcontrols ASoC: amd: acp: Map missing jack kcontrols ASoC: amd: acp3x-rt5682-max9836: Map missing jack kcontrols ASoC: Intel: avs: da7219: Map missing jack kcontrols ASoC: Intel: bxt_da7219_max98357a: Map missing Line Out jack kcontrol ASoC: Intel: bytcr_wm5102: Map missing Line Out jack kcontrol ASoC: Intel: kbl_da7219_max98357a: Map missing Line Out jack kcontrol ASoC: Intel: kbl_da7219_max98927: Map missing Line Out jack kcontrol ASoC: Intel: sof_da7219_max98373: Map missing Line Out jack kcontrol ASoC: imx-es8328: Map missing jack kcontrols ASoC: mediatek: mt8173-max98090: Configure jack as a Headset jack ASoC: mediatek: mt8173-rt5650-rt5514: Map missing jack kcontrols ASoC: mediatek: mt8173-rt5650-rt5676: Map missing jack kcontrols ASoC: mediatek: mt8173-rt5650: Map missing jack kcontrols ASoC: mediatek: mt8183-da7219-max98357: Map missing jack kcontrols ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: Map missing jack kcontrols ASoC: mediatek: mt8186-mt6366-da7219-max98357: Map missing jack kcontrols ASoC: qcom: apq8016_sbc: Map missing jack kcontrols ASoC: qcom: sc7180: Map missing jack kcontrols ASoC: qcom: sc7280: Map missing jack kcontrols ASoC: qcom: sdm845: Map missing jack kcontrols ASoC: rk3399-gru-sound: Map missing Line Out jack kcontrol ASoC: rockchip: rockchip_rt5645: Map missing jack kcontrols ASoC: samsung: littlemill: Map missing jack kcontrols ASoC: samsung: lowland: Split Line Out jack kcontrol from Headphone ASoC: samsung: midas_wm1811: Map missing jack kcontrols sound/soc/amd/acp-da7219-max98357a.c | 41 +++- sound/soc/amd/acp-rt5645.c | 22 +- sound/soc/amd/acp/acp-mach-common.c | 226 ++++++++++++++---- sound/soc/amd/acp3x-rt5682-max9836.c | 23 +- sound/soc/fsl/imx-es8328.c | 25 +- sound/soc/intel/avs/boards/da7219.c | 34 ++- sound/soc/intel/boards/bxt_da7219_max98357a.c | 7 + sound/soc/intel/boards/bytcr_wm5102.c | 7 + sound/soc/intel/boards/kbl_da7219_max98357a.c | 7 + sound/soc/intel/boards/kbl_da7219_max98927.c | 7 + sound/soc/intel/boards/sof_da7219_max98373.c | 10 + sound/soc/mediatek/mt8173/mt8173-max98090.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 23 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 23 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 23 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 43 +++- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 53 +++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 6 + sound/soc/qcom/apq8016_sbc.c | 37 ++- sound/soc/qcom/sc7180.c | 40 +++- sound/soc/qcom/sc7280.c | 34 ++- sound/soc/qcom/sdm845.c | 32 ++- sound/soc/rockchip/rk3399_gru_sound.c | 7 +- sound/soc/rockchip/rockchip_rt5645.c | 22 +- sound/soc/samsung/littlemill.c | 27 ++- sound/soc/samsung/lowland.c | 8 +- sound/soc/samsung/midas_wm1811.c | 25 +- 27 files changed, 668 insertions(+), 146 deletions(-) base-commit: 626c67169f9972fffcdf3bc3864de421f162ebf5 -- 2.40.1

4 34

Re: linux-next: Tree for Aug 7 (sound/soc/fsl/)
by Randy Dunlap 08 Aug '23

08 Aug '23

On 8/6/23 22:47, Stephen Rothwell wrote: > Hi all, > > Changes since 20230804: > on PPC32: WARNING: unmet direct dependencies detected for SND_SOC_MPC5200_AC97 Depends on [n]: SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_MPC52xx [=y] && PPC_BESTCOMM [=n] Selected by [y]: - SND_MPC52xx_SOC_PCM030 [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_MPC5200_SIMPLE [=y] - SND_MPC52xx_SOC_EFIKA [=y] && SOUND [=y] && SND [=y] && SND_SOC [=y] && SND_POWERPC_SOC [=y] && PPC_EFIKA [=y] powerpc-linux-ld: sound/soc/fsl/mpc5200_psc_ac97.o: in function `psc_ac97_of_remove': mpc5200_psc_ac97.c:(.text.psc_ac97_of_remove+0x1c): undefined reference to `mpc5200_audio_dma_destroy' powerpc-linux-ld: sound/soc/fsl/mpc5200_psc_ac97.o: in function `psc_ac97_of_probe': mpc5200_psc_ac97.c:(.text.psc_ac97_of_probe+0x20): undefined reference to `mpc5200_audio_dma_create' Full randconfig file is attached. -- ~Randy

1 0

[PATCH] ASoC: lower "no backend DAIs enabled for ... Port" log severity
by Hans de Goede 07 Aug '23

07 Aug '23

If SNDRV_PCM_IOCTL_PREPARE is called when the mixer settings linking frontend and backend have not been setup yet this results in e.g. the following errors getting logged: [ 43.244549] Baytrail Audio Port: ASoC: no backend DAIs enabled for Baytrail Audio Port [ 43.244744] Baytrail Audio Port: ASoC: error at dpcm_fe_dai_prepare on Baytrail Audio Port: -22 pipewire triggers this leading to 96 repeats of this in the log after the user has logged into a GNOME session. IMHO userspace should not be able to get the kernel to spam dmesg like this. Lower the severity of the "no backend DAIs enabled" log message for dai-s with the dynamic flag set to avoid this. And also changes _soc_pcm_ret() to not log for -EINVAL errors, to fix the other error. Userspace passing wrong parameters should not lead to dmesg messages. Link: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3407 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/soc-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8896227e4fb7..fdc1a1c654d6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -38,6 +38,7 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, switch (ret) { case -EPROBE_DEFER: case -ENOTSUPP: + case -EINVAL: break; default: dev_err(rtd->dev, @@ -2466,7 +2467,11 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) /* there is no point preparing this FE if there are no BEs */ if (list_empty(&fe->dpcm[stream].be_clients)) { - dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + if (fe->dai_link->dynamic) + dev_dbg(fe->dev, "ASoC: no backend DAIs enabled for %s\n", + fe->dai_link->name); + else + dev_err(fe->dev, "ASoC: no backend DAIs enabled for %s\n", fe->dai_link->name); ret = -EINVAL; goto out; -- 2.41.0

4 9

[PATCH V4 0/3] ASoC: codecs: Add awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 07 Aug '23

07 Aug '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier v3 -> v4: Modify the dev_err_probe usage Changed the use of sizeof in kzalloc Changed the function name Merge aw88261_device.c into aw88261.c Weidong Wang (3): ASoC: dt-bindings: Add schema for "awinic,aw88261" ASoC: codecs: Add code for bin parsing compatible with aw88261 ASoC: codecs: Add aw88261 amplifier driver .../bindings/sound/awinic,aw88395.yaml | 4 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw88261.c | 1298 +++++++++++++++++ sound/soc/codecs/aw88261.h | 459 ++++++ sound/soc/codecs/aw88395/aw88395_lib.c | 193 ++- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 7 files changed, 1954 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/aw88261.c create mode 100644 sound/soc/codecs/aw88261.h base-commit: c1a515d3c0270628df8ae5f5118ba859b85464a2 -- 2.41.0

2 4

[PATCH] ASoC: jz4740: Update to use maple tree register cache
by Mark Brown 07 Aug '23

07 Aug '23

The maple tree register cache is based on a much more modern data structure than the rbtree cache and makes optimisation choices which are probably more appropriate for modern systems than those made by the rbtree cache. In v6.5 it has also acquired the ability to generate multi-register writes in sync operations, bringing performance up to parity with the rbtree cache there. Update the jz4740 driver to use the more modern data structure. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/jz4740.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/jz4740.c b/sound/soc/codecs/jz4740.c index 7c25acf6ff0d..d1cea93bdb59 100644 --- a/sound/soc/codecs/jz4740.c +++ b/sound/soc/codecs/jz4740.c @@ -301,7 +301,7 @@ static const struct regmap_config jz4740_codec_regmap_config = { .reg_defaults = jz4740_codec_reg_defaults, .num_reg_defaults = ARRAY_SIZE(jz4740_codec_reg_defaults), - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static int jz4740_codec_probe(struct platform_device *pdev) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230630-asoc-jz-maple-aacbd3a12bb6 Best regards, -- Mark Brown <broonie(a)kernel.org>

1 1

[PATCH 00/16] ASoC: SoundWire codecs: improve pm_runtime handling
by Pierre-Louis Bossart 07 Aug '23

07 Aug '23

This patchset improves the pm_runtime behavior in rare corner cases identified by the Intel CI in the last 6 months. a) in stress-tests, it's not uncommon to see the following type of warnings when the codec reports as ATTACHED "rt711 sdw:0:025d:0711:00: runtime PM trying to activate child device sdw:0:025d:0711:00 but parent (sdw-master-0) is not active" This warning was not correlated with any functional issue, but it exposed a design issue on when to enable pm_runtime. The recommended practice in the pm_runtime documentation is to keep the devices in 'suspended' mode and mark them as 'active' when they are really functional. b) enabling pm_runtime when the codec reports as ATTACHED also creates a problematic case when the ASoC pm_runtime_get_sync() will silently fail due to the -EACCESS error handling. This can happen when playback starts before the codec is enumerated. This patchset modifies the initial stages so that codecs are pm_runtime enabled in the .probe() callback, but become pm_runtime 'active' only when they report present. This is better aligned with the design of the pm_runtime helpers and improved CI results significantly. This patchset modifies all existing SoundWire codecs (except Qualcomm ones), but the pattern of changes is exactly the same in all patches. Pierre-Louis Bossart (16): ASoC: SoundWire codecs: return error status in probe ASoC: SoundWire codecs: make regmap cache-only in probe ASoC: rt5682-sdw: make regmap cache-only in probe ASoC: rt711: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt711-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt700: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1712-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt712-sdca-dmic: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt715-sdca: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1308-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1316-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt1318-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: rt5682-sdw: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98363: enable pm_runtime in probe, keep status as 'suspended' ASoC: max98373-sdw: enable pm_runtime in probe, keep status as 'suspended' sound/soc/codecs/max98363.c | 41 +++++++++++++--------- sound/soc/codecs/max98373-sdw.c | 47 ++++++++++++++----------- sound/soc/codecs/rt1308-sdw.c | 54 +++++++++++++++------------- sound/soc/codecs/rt1316-sdw.c | 44 ++++++++++++++--------- sound/soc/codecs/rt1318-sdw.c | 43 ++++++++++++++--------- sound/soc/codecs/rt5682-sdw.c | 56 ++++++++++++++++++------------ sound/soc/codecs/rt5682.c | 3 ++ sound/soc/codecs/rt700-sdw.c | 7 ++-- sound/soc/codecs/rt700.c | 46 +++++++++++++++--------- sound/soc/codecs/rt711-sdca-sdw.c | 3 +- sound/soc/codecs/rt711-sdca.c | 48 +++++++++++++++++-------- sound/soc/codecs/rt711-sdw.c | 7 ++-- sound/soc/codecs/rt711.c | 47 ++++++++++++++++--------- sound/soc/codecs/rt712-sdca-dmic.c | 47 +++++++++++++++---------- sound/soc/codecs/rt712-sdca-sdw.c | 3 +- sound/soc/codecs/rt712-sdca.c | 45 ++++++++++++++++-------- sound/soc/codecs/rt715-sdca-sdw.c | 5 +-- sound/soc/codecs/rt715-sdca.c | 41 ++++++++++++++++------ sound/soc/codecs/rt715-sdw.c | 9 ++--- sound/soc/codecs/rt715.c | 41 +++++++++++++++------- 20 files changed, 392 insertions(+), 245 deletions(-) -- 2.39.2

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[PATCH v2 1/3] ASoC: dt-bindings: Convert maxim,max98925 to DT schema
by Krzysztof Kozlowski 07 Aug '23

07 Aug '23

Convert the Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier bindings to DT schema format. Changes during conversion: 1. Add "sound-dai-cells", already used by DTS. 2. Use "maxim,interleave-mode" instead previous "interleave-mode" and undocumented but used interleave_mode. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Use maxim,interleave-mode as bool. Ryan, As original author, I added you as bindings maintainer. Is that okay or maybe someone else from Maxim should take care about the bindings? --- .../devicetree/bindings/sound/max9892x.txt | 44 --------- .../bindings/sound/maxim,max98925.yaml | 98 +++++++++++++++++++ 2 files changed, 98 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98925.yaml diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index 98cb9ba5b328..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Optional properties: - - reset-gpios : GPIO to reset the device - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..32fd86204a7a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,98 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + maxim,interleave-mode: + type: boolean + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. When enabled, the Digital Audio + Interleave mode provides a means to output VMON and IMON data from two + devices on a single DOUT line when running smaller frames sizes such as + 32 BCLKS per LRCLK or 48 BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; -- 2.34.1

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