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[RESEND PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 18 Aug '22

18 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

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[PATCH] ALSA: hda/realtek: Add quirk for Clevo NS50PU, NS70PU
by Werner Sembach 18 Aug '22

18 Aug '22

From: Christoffer Sandberg <cs(a)tuxedo.de> Fixes headset microphone detection on Clevo NS50PU and NS70PU. Signed-off-by: Christoffer Sandberg <cs(a)tuxedo.de> Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- 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 fd630d62b5a0..fceac608214e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9389,6 +9389,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x70f4, "Clevo NH77EPY", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x70f6, "Clevo NH77DPQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7716, "Clevo NS50PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x7717, "Clevo NS70PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x7718, "Clevo L140PU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8228, "Clevo NR40BU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8520, "Clevo NH50D[CD]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.34.1

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HDA ATI HDMI not supported
by GitHub issues - edited 18 Aug '22

18 Aug '22

alsa-project/alsa-ucm-conf issue #194 was edited from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… The `alsaucm reload` command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` the HDMI works but the I2S and HDMI all use fallback ucm config. despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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RE: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp
by Ki-Seok Jo 18 Aug '22

18 Aug '22

Dear Mark Brown, Thank you for your detailed feedback. I will apply what you told me as soon as possible and request it again. I'll ask you again if there's anything wrong with the correction. I appreciate your quick response. Thank you very much for your time Best Regards, Kiseok Jo Iron Device Corporation Tel.: +82-2-541-2896 Mobile: +82-10-3320-0376 Email: kiseok.jo(a)irondevice.com The information in this email and any attachment is confidential and may be legally protected. It is intended solely for the addressee. Access to this email by anyone else is unauthorised. If you are not the intended recipient, any disclosure or actions taken as a result of the information in this email is prohibited and may be unlawful. If you are not the intended recipient, please notify us immediately and then delete this e-mail and any attachment. Thank you. -----Original Message----- From: Mark Brown <broonie(a)kernel.org> Sent: Wednesday, August 17, 2022 9:09 PM To: Ki-Seok Jo <kiseok.jo(a)irondevice.com> Cc: Gyu-Hwa Park <gyuhwa.park(a)irondevice.com>; alsa-devel(a)alsa-project.org; Suk-Min Kang <sukmin.kang(a)irondevice.com> Subject: Re: [PATCH 1/2] ASoC: sma1303: Add driver for Iron Device SMA1303 Amp On Wed, Aug 17, 2022 at 12:29:37PM +0900, Kiseok Jo wrote: I got part way through reviewing this driver. The top level feedback here is that the driver appears to be exposing everything in the device to userspace as enumeration controls rather than using standard features like DAPM, or appropriate other control types where there's something that's a better fit than an enum. If there's a strong reason for the driver to do something so unusual then theere should be something in here which explains what's going on. > +// SPDX-License-Identifier: GPL-2.0-or-later > +/* sma1303.c -- sma1303 ALSA SoC Audio driver > + * > + * Copyright 2019 Iron Device Corporation 2019? Please make the entire block a C++ one so things look more intentional. > +static int power_up_down_control_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + if (sel && !(sma1303->force_amp_power_down)) > + sma1303_startup(component); > + else > + sma1303_shutdown(component); > + > + return 0; > +} This and all the other controls need to return 1 on change, please make sure a card with your driver your driver passes mixer-test cleanly - it will spot this and other problems. However I wouldn't expect the power controls to exist at all, generally power is managed via DAPM rather than userspace. > +static int force_sdo_bypass_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) { > + struct snd_soc_component *component = > + snd_soc_kcontrol_component(kcontrol); > + struct sma1303_priv *sma1303 = snd_soc_component_get_drvdata(component); > + int sel = (int)ucontrol->value.integer.value[0]; > + > + sma1303->sdo_bypass_flag = (bool)sel; > + > + return 0; > +} What is this bypassing? > +static const char * const sma1303_input_format_text[] = { > + "I2S", "LJ", "Reserved", "Reserved", > + "RJ_16", "RJ_18", "RJ_20", "RJ_24"}; > + > +static const struct soc_enum sma1303_input_format_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_format_text), > + sma1303_input_format_text); This should be controlled by the machine driver through set_dai_fmt() and hw_params(). > +static const char * const sma1303_pcm_n_slot_text[] = { > + "Slot_1", "Slot_2", "Slot_3", "Slot_4", "Slot_5", "Slot_6", > + "Slot_7", "Slot_8", "Slot_9", "Slot_10", "Slot_11", "Slot_12", > + "Slot_13", "Slot_14", "Slot_15", "Slot_16"}; > + > +static const struct soc_enum sma1303_pcm_n_slot_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_pcm_n_slot_text), > + sma1303_pcm_n_slot_text); set_tdm_slot() > +static const char * const sma1303_port_config_text[] = { > + "IN_Port", "Reserved", "OUT_Port", "Reserved"}; This should be DT properties, possibly using the pinctrl schema depending on what's actually going on. > +static const char * const sma1303_port_out_sel_text[] = { > + "Disable", "Format_C", "Mixer_out", "After_DSP", > + "Postscaler", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_port_out_sel_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_port_out_sel_text), > + sma1303_port_out_sel_text); This looks like it should be a DAPM routing control. > +static const char * const sma1303_spk_off_slope_text[] = { > + "00", "01", "10", "11"}; > + > +static const struct soc_enum sma1303_spk_off_slope_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spk_off_slope_text), > + sma1303_spk_off_slope_text); Why make this an enum and not just a numerical control with values 0-3? > +static const char * const sma1303_spkmode_text[] = { > + "Off", "Mono", "Reserved", "Reserved", > + "Stereo", "Reserved", "Reserved", "Reserved"}; > + > +static const struct soc_enum sma1303_spkmode_enum = > + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_spkmode_text), > + sma1303_spkmode_text); Use a value enum to eliminate the reserved values from user selection, however if this is controlling if the device has mono or stereo speakers it should be a DT property since that's unlikely to change at runtime. > +static const char * const sma1303_input_gain_text[] = { > + "Gain_0dB", "Gain_M6dB", "Gain_M12dB", "Gain_MInf"}; > + > +static const struct soc_enum sma1303_input_gain_enum = > +SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(sma1303_input_gain_text), > + sma1303_input_gain_text); This should just be a normal volume control. I've stopped reviewing at this point.

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snd_hda_intel: Woke up but no data
by GitHub issues - edited 18 Aug '22

18 Aug '22

alsa-project/alsa-lib issue #262 was edited from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. alsa-lib 1.2.7.2 alsa-utils 1.2.7 alsa-plugins 1.2.2 Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

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snd_hda_intel: Woke up but no data
by GitHub issues - opened 17 Aug '22

17 Aug '22

alsa-project/alsa-lib issue #262 was opened from shmohawk: ``` Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Aug 17 01:11:23 hellokitty pulseaudio[12938]: [alsa-sink-CX20590 Analog] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. ``` It is symptomous with a crackling sound for ~5 seconds after any playback starts. Issue URL : https://github.com/alsa-project/alsa-lib/issues/262 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: fsl_sai: Remove unnecessary FIFO reset in ISR
by Shengjiu Wang 17 Aug '22

17 Aug '22

The FIFO reset drops the words in the FIFO, which may cause channel swap when SAI module is running, especially when the DMA speed is low. So it is not good to do FIFO reset in ISR, then remove the operation. Fixes: e2681a1bf5ae ("ASoC: fsl_sai: Add isr to deal with error flag") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d430eece1d6b..a7fa6f0bf83d 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -114,11 +114,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Tx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Transmit underrun detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled transmit FIFO is empty\n"); @@ -148,11 +145,8 @@ static irqreturn_t fsl_sai_isr(int irq, void *devid) if (flags & FSL_SAI_CSR_SEF) dev_dbg(dev, "isr: Rx Frame sync error detected\n"); - if (flags & FSL_SAI_CSR_FEF) { + if (flags & FSL_SAI_CSR_FEF) dev_dbg(dev, "isr: Receive overflow detected\n"); - /* FIFO reset for safety */ - xcsr |= FSL_SAI_CSR_FR; - } if (flags & FSL_SAI_CSR_FWF) dev_dbg(dev, "isr: Enabled receive FIFO is full\n"); -- 2.34.1

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Re: [PATCH] ASoC: soc-utils-test: Add test for snd_soc_params_to_bclk()
by Mark Brown 17 Aug '22

17 Aug '22

On Wed, 17 Aug 2022 13:55:08 +0100, Richard Fitzgerald wrote: > snd_soc_params_to_bclk() calculates the BCLK from only the information in > snd_pcm_hw_params. It is therefore a subset of the functionality of > snd_soc_tdm_params_to_bclk() so can use a subset of the test case table. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: soc-utils-test: Add test for snd_soc_params_to_bclk() commit: e32e23a2b588424aec0c4c4435530f8022318b8f All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

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[PATCH 0/5] ASoC: cs42l42: Some small code improvements
by Richard Fitzgerald 17 Aug '22

17 Aug '22

This is a collection of minor improvements to the code or comments. Richard Fitzgerald (5): ASoC: cs42l42: Don't include kernel.h ASoC: cs42l42: Add include dependencies to cs42l42.h ASoC: cs42l42: Move cs42l42_supply_names to .c file ASoC: cs42l42: Fix comment typo in cs42l42_slow_start_put() ASoC: cs42l42: Use snd_soc_tdm_params_to_bclk() sound/soc/codecs/cs42l42.c | 24 ++++++++++++++++-------- sound/soc/codecs/cs42l42.h | 13 +++++-------- 2 files changed, 21 insertions(+), 16 deletions(-) -- 2.30.2

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[PATCH 0/4] ASoC/soundwire: log actual PING status on resume issues
by Bard Liao 17 Aug '22

17 Aug '22

we've been stuck with problems in the dual-amplifier configurations where one of the two devices seems to become UNATTACHED and never regains sync, see https://github.com/thesofproject/linux/issues/3638. This is a rather infrequent issue that may happen once or twice per month, but still it remains a concern. One possibility is that the device does lose sync but somehow our hardware detection fails to see it resync. This series just adds a basic read directly from the PING frames to help confirm if yes/no the device regain sync. The change is mainly on soundwire. @Mark, Could you ack the ASoC patch if it looks good to you? Pierre-Louis Bossart (4): soundwire: add read_ping_status helper definition in manager ops soundwire: intel/cadence: expose PING status in manager ops soundwire: add sdw_show_ping_status() helper ASoC: codecs: show PING status on resume failures drivers/soundwire/bus.c | 32 ++++++++++++++++++++++++++++++ drivers/soundwire/cadence_master.c | 8 ++++++++ drivers/soundwire/cadence_master.h | 2 ++ drivers/soundwire/intel.c | 1 + include/linux/soundwire/sdw.h | 5 +++++ sound/soc/codecs/max98373-sdw.c | 2 ++ sound/soc/codecs/rt1308-sdw.c | 2 ++ sound/soc/codecs/rt1316-sdw.c | 2 ++ sound/soc/codecs/rt5682-sdw.c | 2 ++ sound/soc/codecs/rt700-sdw.c | 2 ++ sound/soc/codecs/rt711-sdca-sdw.c | 2 ++ sound/soc/codecs/rt715-sdca-sdw.c | 2 ++ sound/soc/codecs/rt715-sdw.c | 2 ++ 13 files changed, 64 insertions(+) -- 2.25.1

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[PATCH -next] ASoC: codecs: es8326: change es8326_regmap_config to static
by Yang Yingliang 17 Aug '22

17 Aug '22

es8326_regmap_config is only used in es8326.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/es8326.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 975302b2a61d..87c1cc16592b 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -207,7 +207,7 @@ static const struct regmap_access_table es8326_volatile_table = { .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), }; -const struct regmap_config es8326_regmap_config = { +static const struct regmap_config es8326_regmap_config = { .reg_bits = 8, .val_bits = 8, .max_register = 0xff, -- 2.25.1

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[PATCH 0/4] Enable timestamp support for SOF compress driver
by Daniel Baluta 17 Aug '22

17 Aug '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> This patchseries computes pcm_io_frames from the DAI position reported by SOF firmware. Using pcm_io_frames userspace applications can later compute timestamps for compressed stream. Daniel Baluta (2): ASoC: SOF: compress: Remove byte offset computation ASoC: SOF: compress: Introduce sof_compr_stream Laurentiu Mihalcea (2): ASoC: SOF: compress: Save channel count and sample bytes ASoC: SOF: compress: Add support for computing timestamps sound/soc/sof/compress.c | 54 ++++++++++++++++++++++++---------------- sound/soc/sof/sof-priv.h | 7 ++++++ 2 files changed, 40 insertions(+), 21 deletions(-) -- 2.27.0

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[PATCH] ASoC: qcom: qdsp6: q6prm: add new clocks
by Srinivas Kandagatla 17 Aug '22

17 Aug '22

Add support to new clocks that are added in Q6DSP as part of newer version of LPASS support on SM8450 and SC8280XP. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../sound/qcom,q6dsp-lpass-ports.h | 18 ++++++++++++++++++ sound/soc/qcom/qdsp6/q6prm-clocks.c | 9 +++++++++ sound/soc/qcom/qdsp6/q6prm.h | 19 +++++++++++++++++++ 3 files changed, 46 insertions(+) diff --git a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h index 0d3276c8fc11..9f7c5103bc82 100644 --- a/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h +++ b/include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h @@ -193,6 +193,24 @@ #define LPASS_CLK_ID_RX_CORE_MCLK 59 #define LPASS_CLK_ID_RX_CORE_NPL_MCLK 60 #define LPASS_CLK_ID_VA_CORE_2X_MCLK 61 +/* Clock ID for MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_MCLK 62 +/* Clock ID for NPL MCLK for WSA2 core */ +#define LPASS_CLK_ID_WSA2_CORE_2X_MCLK 63 +/* Clock ID for RX Core TX MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_MCLK 64 +/* Clock ID for RX CORE TX 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 65 +/* Clock ID for WSA core TX MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_MCLK 66 +/* Clock ID for WSA core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 67 +/* Clock ID for WSA2 core TX MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_MCLK 68 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 69 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 70 #define LPASS_HW_AVTIMER_VOTE 101 #define LPASS_HW_MACRO_VOTE 102 diff --git a/sound/soc/qcom/qdsp6/q6prm-clocks.c b/sound/soc/qcom/qdsp6/q6prm-clocks.c index a26cda5140c1..73b0cbac73d4 100644 --- a/sound/soc/qcom/qdsp6/q6prm-clocks.c +++ b/sound/soc/qcom/qdsp6/q6prm-clocks.c @@ -50,6 +50,15 @@ static const struct q6dsp_clk_init q6prm_clks[] = { Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK), Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_NPL_MCLK), Q6PRM_CLK(LPASS_CLK_ID_VA_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK), + Q6PRM_CLK(LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK), Q6DSP_VOTE_CLK(LPASS_HW_MACRO_VOTE, Q6PRM_HW_CORE_ID_LPASS, "LPASS_HW_MACRO"), Q6DSP_VOTE_CLK(LPASS_HW_DCODEC_VOTE, Q6PRM_HW_CORE_ID_DCODEC, diff --git a/sound/soc/qcom/qdsp6/q6prm.h b/sound/soc/qcom/qdsp6/q6prm.h index fea4d1954bc1..a988a32086fe 100644 --- a/sound/soc/qcom/qdsp6/q6prm.h +++ b/sound/soc/qcom/qdsp6/q6prm.h @@ -64,6 +64,25 @@ #define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK 0x30e #define Q6PRM_LPASS_CLK_ID_RX_CORE_NPL_MCLK 0x30f +/* Clock ID for MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_MCLK 0x310 +/* Clock ID for NPL MCLK for WSA2 core */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_2X_MCLK 0x311 +/* Clock ID for RX Core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_MCLK 0x312 +/* Clock ID for RX CORE TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_TX_2X_MCLK 0x313 +/* Clock ID for WSA core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_MCLK 0x314 +/* Clock ID for WSA core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA_CORE_TX_2X_MCLK 0x315 +/* Clock ID for WSA2 core TX MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_MCLK 0x316 +/* Clock ID for WSA2 core TX 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_WSA2_CORE_TX_2X_MCLK 0x317 +/* Clock ID for RX CORE MCLK2 2X MCLK */ +#define Q6PRM_LPASS_CLK_ID_RX_CORE_MCLK2_2X_MCLK 0x318 + #define Q6PRM_LPASS_CLK_SRC_INTERNAL 1 #define Q6PRM_LPASS_CLK_ROOT_DEFAULT 0 #define Q6PRM_HW_CORE_ID_LPASS 1 -- 2.21.0

2 1

Re: [PATCH] ASoC: core: remove setting platform_max in kcontrol macros
by Mark Brown 17 Aug '22

17 Aug '22

On Tue, 16 Aug 2022 18:21:29 +0100, Srinivas Kandagatla wrote: > platform_max should not be set by the driver, its intended for machine drivers > or DT to override the max value for platform specific reasons. > > So remove setting this from Kcontrol macros. > > Setting this to max in these macros would limit the range when min > value is less then zero. > > [...] Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: core: remove setting platform_max in kcontrol macros commit: 26bdcc4ba12351642cd94339aa6996f96434dd47 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

Re: [PATCH 0/4] ALSA: hda: Minor cleanups
by Cezary Rojewski 17 Aug '22

17 Aug '22

On 2022-08-17 3:23 PM, Cezary Rojewski wrote: > On 2022-08-17 3:11 PM, Amadeusz Sławiński wrote: >> First two patches are small refactoring of code to use poll macros >> instead of open coding register checks. >> Following two patches remove unused defines from code. >> >> Amadeusz Sławiński (4): >> ALSA: hda: Move stream-register polling macros >> ALSA: hda: Rework snd_hdac_stream_reset() to use macros >> ALSA: hda: Remove unused MAX_PIN_CONFIGS constant >> ALSA: hda: Remove unused defines > > > Reviewed by me before sending, plus from the test results patchset > causes no regression in the avs-driver CI. > > Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> RESEND. Patch was blocked so my tag did not make it to the list.

1 0

[RESEND][PATCH] ALSA: info: Fix llseek return value when using callback
by Amadeusz Sławiński 17 Aug '22

17 Aug '22

When using callback there was a flow of ret = -EINVAL if (callback) { offset = callback(); goto out; } ... offset = some other value in case of no callback; ret = offset; out: return ret; which causes the snd_info_entry_llseek() to return -EINVAL when there is callback handler. Fix this by setting "ret" directly to callback return value before jumping to "out". 73029e0ff18d ("ALSA: info - Implement common llseek for binary mode") Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/core/info.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/info.c b/sound/core/info.c index b8058b341178..0b2f04dcb589 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -111,9 +111,9 @@ static loff_t snd_info_entry_llseek(struct file *file, loff_t offset, int orig) entry = data->entry; mutex_lock(&entry->access); if (entry->c.ops->llseek) { - offset = entry->c.ops->llseek(entry, - data->file_private_data, - file, offset, orig); + ret = entry->c.ops->llseek(entry, + data->file_private_data, + file, offset, orig); goto out; } -- 2.25.1

3 4

[PATCH 0/4] ALSA: hda: Minor cleanups
by Amadeusz Sławiński 17 Aug '22

17 Aug '22

First two patches are small refactoring of code to use poll macros instead of open coding register checks. Following two patches remove unused defines from code. Amadeusz Sławiński (4): ALSA: hda: Move stream-register polling macros ALSA: hda: Rework snd_hdac_stream_reset() to use macros ALSA: hda: Remove unused MAX_PIN_CONFIGS constant ALSA: hda: Remove unused defines include/sound/hdaudio.h | 6 ++++++ include/sound/hdaudio_ext.h | 6 ------ sound/hda/hdac_stream.c | 27 +++++++-------------------- sound/pci/hda/hda_intel.c | 7 ------- sound/pci/hda/hda_sysfs.c | 2 -- 5 files changed, 13 insertions(+), 35 deletions(-) -- 2.25.1

1 4

[PATCH] ASoC: Variable type completion
by Xin Gao 17 Aug '22

17 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/soc-ac97.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 5f49e3dec3fc..32c5be61e2ec 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -57,7 +57,7 @@ static inline struct snd_soc_component *gpio_to_component(struct gpio_chip *chip return gpio_priv->component; } -static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned int offset) { if (offset >= AC97_NUM_GPIOS) return -EINVAL; @@ -66,7 +66,7 @@ static int snd_soc_ac97_gpio_request(struct gpio_chip *chip, unsigned offset) } static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, - unsigned offset) + unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); @@ -75,7 +75,7 @@ static int snd_soc_ac97_gpio_direction_in(struct gpio_chip *chip, 1 << offset, 1 << offset); } -static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) +static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned int offset) { struct snd_soc_component *component = gpio_to_component(chip); int ret; @@ -88,7 +88,7 @@ static int snd_soc_ac97_gpio_get(struct gpio_chip *chip, unsigned offset) return !!(ret & (1 << offset)); } -static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned offset, +static void snd_soc_ac97_gpio_set(struct gpio_chip *chip, unsigned int offset, int value) { struct snd_ac97_gpio_priv *gpio_priv = gpiochip_get_data(chip); -- 2.30.2

2 1

[PATCH] ASoC: qcom: sm8250: add missing module owner
by Srinivas Kandagatla 17 Aug '22

17 Aug '22

Add missing module owner to able to build and load this driver as module. Fixes: aa2e2785545a ("ASoC: qcom: sm8250: add sound card qrb5165-rb5 support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/sm8250.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index ce4a5713386a..98a2fde9e004 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -270,6 +270,7 @@ static int sm8250_platform_probe(struct platform_device *pdev) if (!card) return -ENOMEM; + card->owner = THIS_MODULE; /* Allocate the private data */ data = devm_kzalloc(dev, sizeof(*data), GFP_KERNEL); if (!data) -- 2.21.0

2 1

Channel mapping randomly change on usb soundcard and sucpicious dmesg log
by GitHub issues - opened 17 Aug '22

17 Aug '22

alsa-project/alsa-lib issue #261 was opened from sprudent: Hello erveryone, I am getting random channel swap when using speaker test to output sequencially a pink noise on every channel of the soundcard. I tested two distinct usb soundcard getting the same issue. I could identify a suspicious log showing at the same time the channels swapped. Tested and reproduced on a recent archlinux image (see system information). ``` [Aug16 18:26] xhci_hcd 0000:00:14.0: ERROR Transfer event TRB DMA ptr not part of current TD ep_index 11 comp_code 1 [ +0.000134] xhci_hcd 0000:00:14.0: Looking for event-dma 0000000014ed98e0 trb-start 0000000014ed98f0 trb-end 0000000014ed98f0 seg-start 0000000014ed9000 seg-end 0000000014ed9ff0 ``` ### system information ``` uname -r 5.18.7-arch1-1 ``` ``` aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Audio [Intel HDMI/DP LPE Audio], device 0: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [Intel HDMI/DP LPE Audio], device 1: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audio [Intel HDMI/DP LPE Audio], device 2: HdmiLpeAudio [Intel HDMI/DP LPE Audi] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: ICUSBAUDIO7D [ICUSBAUDIO7D], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: bytchtes8316 [bytcht-es8316], device 0: Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: bytchtes8316 [bytcht-es8316], device 1: Deep-Buffer Audio (*) [] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/9358483/alsa-… ### Test with dmix #### Description First test was made with alsa dmix plugin. Multiple processes running in parrallel and outputing to dmix pcm device. #### Alsa configuration ``` pcm.play_8 { @args [ CARD ] @args.CARD { type string } type dmix ipc_key 5678 slave { pcm $CARD channels 8 period_time 0 period_size 1024 buffer_size 4096 rate 44100 } bindings { 0 0 1 1 2 2 3 3 4 4 5 5 6 6 7 7 } } ``` #### Step to reproduce Run this script to simulate multiple processes writing to dmix device at the same time ``` for i in $(seq 20) do speaker-test -r44100 -D"play_8:\"hw:1\"" -c8 -t pink & done ``` Then on another tty, start a speaker-test command outputing a sine wave in order to identify channel mapping by correlating speaker test output with audio output. `speaker-test -r44100 -D"play_8:\"hw:1\"" -c8 -t sine` ``` 0 - Front Left 4 - Front Center 1 - Front Right 3 - Side Right 7 - Side Right 2 - Side Left 6 - Side Left 5 - LFE Time per period = 21.998519 ... ``` #### Expected result Audio output to the same channels during all the test #### Actual result Random audio channel swap correlated with dmesg log mentioned above. ### Second test with direct output #### Alsa configuration No specific configuration required #### Step to reproduce Run speaker-following test command and wait for channel swap `speaker-test -r44100 -Dsurround71:ICUSBAUDIO7D -c8` #### Expected result Audio output to the same channels during all the test #### Actual result Random audio channel swap correlated with dmesg log mentioned above. The chanel swap happen far less often than with dmix. ### Interpretation The DMA transfer ring buffer get messed up after a while (see dmesg log) and interleaved audio frames get shifted in soundcard buffer. If I unplug/replug usb soundcard and try again, channel mapping goes back to the correct order. My alsa, kernel and usb driver knowledge is limited, so I come to you for a little help. Tell me if you need more information to reproduce. Thank's, Sylvain Issue URL : https://github.com/alsa-project/alsa-lib/issues/261 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ALSA: hda/cs8409: Support new Dolphin Variants
by Stefan Binding 17 Aug '22

17 Aug '22

Add 4 new Dolphin Systems, same configuration as older systems. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409-tables.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_cs8409-tables.c b/sound/pci/hda/patch_cs8409-tables.c index e0d3a8be2e38..b288874e401e 100644 --- a/sound/pci/hda/patch_cs8409-tables.c +++ b/sound/pci/hda/patch_cs8409-tables.c @@ -546,6 +546,10 @@ const struct snd_pci_quirk cs8409_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0BD6, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD7, "Dolphin", CS8409_DOLPHIN), SND_PCI_QUIRK(0x1028, 0x0BD8, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C43, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C50, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C51, "Dolphin", CS8409_DOLPHIN), + SND_PCI_QUIRK(0x1028, 0x0C52, "Dolphin", CS8409_DOLPHIN), {} /* terminator */ }; -- 2.34.1

2 1

[PATCH] ALSA: hda/realtek: Add CLSA0101 to serial-multi-instantiate.
by Cameron Berkenpas 17 Aug '22

17 Aug '22

Added CLSA0101 to serial-multi-instantiate.c. Signed-off-by: Cameron Berkenpas <cam(a)neo-zeon.de> --- drivers/platform/x86/serial-multi-instantiate.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/platform/x86/serial-multi-instantiate.c b/drivers/platform/x86/serial-multi-instantiate.c index 1e8063b7c169..bc9bf591dd10 100644 --- a/drivers/platform/x86/serial-multi-instantiate.c +++ b/drivers/platform/x86/serial-multi-instantiate.c @@ -327,8 +327,9 @@ static const struct acpi_device_id smi_acpi_ids[] = { { "BSG2150", (unsigned long)&bsg2150_data }, { "INT3515", (unsigned long)&int3515_data }, { "CSC3551", (unsigned long)&cs35l41_hda }, - /* Non-conforming _HID for Cirrus Logic already released */ + /* Non-conforming _HIDs for Cirrus Logic already released */ { "CLSA0100", (unsigned long)&cs35l41_hda }, + { "CLSA0101", (unsigned long)&cs35l41_hda }, { } }; MODULE_DEVICE_TABLE(acpi, smi_acpi_ids); -- 2.34.1

2 3

Re: [PATCH v2 linux-next] Remove Unnecessary TODO
by Pierre-Louis Bossart 17 Aug '22

17 Aug '22

On 8/16/22 19:59, Khalid Masum wrote: > The capabilities enabled for multi-link are required as part of the > programming sequences, even when a stream uses a single link we still > use the syncArm/syncGo sequences. Therefore the TODO is no longer > necessary. > > Suggested-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> the title of your patch should be something like soundwire: intel: remove unnecessary TODO so that the patch scope and context is clearly identified. > --- > Previous discussions: > - v1 Link: https://lore.kernel.org/lkml/20220814080416.7531-1-khalid.masum.92@gmail.co… > > Changes since v1: > - Remove the todo, leave rest of the code as is > - Update commit message > > drivers/soundwire/bus.c | 1 - > 1 file changed, 1 deletion(-) > > diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c > index 8d4000664fa3..18158825a6d8 100644 > --- a/drivers/soundwire/bus.c > +++ b/drivers/soundwire/bus.c > @@ -75,7 +75,6 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, > > /* > * Initialize multi_link flag > - * TODO: populate this flag by reading property from FW node > */ > bus->multi_link = false; > if (bus->ops->read_prop) {

2 1

bat: Add 'readcapture' option to support analyzing external audio
by GitHub pull_request - opened 16 Aug '22

16 Aug '22

alsa-project/alsa-utils pull request #166 was opened from tbird20d: If audio data is captured on another device (ie we are NOT using loopback mode), then allow alsabat to analyze that data, by passing a filename reference on the command line. Add the '--readcapture' option to the argument parser. When this option is specified, avoid doing a local capture, and instead just read the audio data from the indicated file, and analyze that. Signed-off-by: Tim Bird <tim.bird(a)sony.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/166 Patch URL : https://github.com/alsa-project/alsa-utils/pull/166.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Re: Rockchip I2S commit possibly ignored
by Mark Brown 16 Aug '22

16 Aug '22

On Tue, Aug 16, 2022 at 02:54:22PM -0300, Geraldo Nascimento wrote: > On Tue, Aug 16, 2022 at 04:22:37PM +0100, Mark Brown wrote: > > There's absolutely no problem with reposting someone else's patch - just > > add your Signed-off-by at the end of the signoff chain. > Mark, I'm in no position to lecture anyone, least of who, you, hard-working > ASoC maintainer that you are. But there are *tons* of problems with > reposting someone else's patch, even if they had been previously given > the green-light but misteriously vanished. > You are assuming responsibility for someone else's work! Let's see in Oh, sure - but TBH if you're chasing a patch via e-mail you're pretty much going to be in a similar situation if that results in the patch getting applied. A big part of the goal behind getting things reposted is to push them through the normal test/review cycle properly so if there was some reason for it to not get applied the first time around it's more likely that someone will notice than if it's just pulled out of list archives or whatever. > My main point is that without adding "resets" and "reset-names" to *at > least* every Rockchip device tree that enables sound over HDMI, just an > example, you get systems with non-working HDMI. I just tested it, I > omitted "resets" and "reset-names" from my device tree and then had > to SSH into the black screen to revert the changes to my boot partition. > So it's not trivial to RESEND this. It amounts to device tree overhaul > of arch/arm64/boot/dts/rockchip > This would require community effort of say, equivalent to LibreELEC > resources, which are not many, but they have enough patience to test > every proposed change to Rockchip device trees, and could help upstream > this. Right, that's a problem. Even with those changes if we start requiring new properties that'd be an ABI break anyway which isn't something we want for DT - ideally the patch should be reworked so that existing systems continue to work even without DT updates.

1 0

[PATCH] ASoC: Variable type completion
by Xin Gao 16 Aug '22

16 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/ti/davinci-evm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c index 68d69e32681a..392bf17564bf 100644 --- a/sound/soc/ti/davinci-evm.c +++ b/sound/soc/ti/davinci-evm.c @@ -23,7 +23,7 @@ struct snd_soc_card_drvdata_davinci { struct clk *mclk; - unsigned sysclk; + unsigned int sysclk; }; static int evm_startup(struct snd_pcm_substream *substream) @@ -57,7 +57,7 @@ static int evm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); struct snd_soc_card *soc_card = rtd->card; int ret = 0; - unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *) + unsigned int sysclk = ((struct snd_soc_card_drvdata_davinci *) snd_soc_card_get_drvdata(soc_card))->sysclk; /* set the codec system clock */ -- 2.30.2

1 0

[PATCH] ASoC: Variable type completion
by Xin Gao 16 Aug '22

16 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/codecs/pcm3008.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/pcm3008.h b/sound/soc/codecs/pcm3008.h index f7f4fbbd89db..45270bcf1c02 100644 --- a/sound/soc/codecs/pcm3008.h +++ b/sound/soc/codecs/pcm3008.h @@ -10,10 +10,10 @@ #define __LINUX_SND_SOC_PCM3008_H struct pcm3008_setup_data { - unsigned dem0_pin; - unsigned dem1_pin; - unsigned pdad_pin; - unsigned pdda_pin; + unsigned int dem0_pin; + unsigned int dem1_pin; + unsigned int pdad_pin; + unsigned int pdda_pin; }; #endif -- 2.30.2

1 0

ASoC: snd_soc_info_volsw and platfrom_max
by Srinivas Kandagatla 16 Aug '22

16 Aug '22

Hi Mark, After patch 30ac49841386 (ASoC: ops: Don't modify the driver's plaform_max when reading state) all the controls that are using signed TLV range like below /* -84dB min - 40dB max */ SOC_SINGLE_S8_TLV("RX0 Digital Volume", WCD934X_CDC_RX0_RX_VOL_CTL, -84, 40, digital_gain), reports max value as 40 instead of 124. before this patch the controls max value was calculated considering the min value, but with this patch this calculation has changed resulting in low volume on most of the codecs that are using SOC_SINGLE_S8_TLV. snd_soc_put_volsw does the right thing by considering mc->min, but info_volsw does it differently. Below change fixes the issue for me, but I am bit confused with the first line of this function that calculates max value as max = mc->max - mc->min and then limits it to platform_max. diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index bd88de056358..49fb34609202 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -196,7 +196,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; - uinfo->value.integer.max = max; + uinfo->value.integer.max = max - mc->min; return 0; } Or should we fix the macro to set platform_max to be max - min. thanks, Srini

2 5

[PATCH 1/2] ASoC: Intel: common: add ACPI matching tables for Raptor Lake
by Kai Vehmanen 16 Aug '22

16 Aug '22

Initial support for RPL w/ RT711 Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Tested-by: Gopal Vamshi Krishna <vamshi.krishna.gopal(a)intel.com> --- include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 2 +- .../intel/common/soc-acpi-intel-rpl-match.c | 51 +++++++++++++++++++ 3 files changed, 54 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-rpl-match.c diff --git a/include/sound/soc-acpi-intel-match.h b/include/sound/soc-acpi-intel-match.h index bc7fd46ec2bc..ac750afa7bc6 100644 --- a/include/sound/soc-acpi-intel-match.h +++ b/include/sound/soc-acpi-intel-match.h @@ -30,6 +30,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_ehl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cnl_sdw_machines[]; @@ -38,6 +39,7 @@ extern struct snd_soc_acpi_mach snd_soc_acpi_intel_cml_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_icl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_tgl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_adl_sdw_machines[]; +extern struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[]; extern struct snd_soc_acpi_mach snd_soc_acpi_intel_mtl_sdw_machines[]; /* diff --git a/sound/soc/intel/common/Makefile b/sound/soc/intel/common/Makefile index 8ca8f872ec80..41054cf09ec9 100644 --- a/sound/soc/intel/common/Makefile +++ b/sound/soc/intel/common/Makefile @@ -9,7 +9,7 @@ snd-soc-acpi-intel-match-objs := soc-acpi-intel-byt-match.o soc-acpi-intel-cht-m soc-acpi-intel-cml-match.o soc-acpi-intel-icl-match.o \ soc-acpi-intel-tgl-match.o soc-acpi-intel-ehl-match.o \ soc-acpi-intel-jsl-match.o soc-acpi-intel-adl-match.o \ - soc-acpi-intel-mtl-match.o \ + soc-acpi-intel-rpl-match.o soc-acpi-intel-mtl-match.o \ soc-acpi-intel-hda-match.o \ soc-acpi-intel-sdw-mockup-match.o diff --git a/sound/soc/intel/common/soc-acpi-intel-rpl-match.c b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c new file mode 100644 index 000000000000..0b77401e4e6f --- /dev/null +++ b/sound/soc/intel/common/soc-acpi-intel-rpl-match.c @@ -0,0 +1,51 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * soc-apci-intel-rpl-match.c - tables and support for RPL ACPI enumeration. + * + * Copyright (c) 2022 Intel Corporation. + */ + +#include <sound/soc-acpi.h> +#include <sound/soc-acpi-intel-match.h> + +static const struct snd_soc_acpi_endpoint single_endpoint = { + .num = 0, + .aggregated = 0, + .group_position = 0, + .group_id = 0, +}; + +static const struct snd_soc_acpi_adr_device rt711_0_adr[] = { + { + .adr = 0x000020025D071100ull, + .num_endpoints = 1, + .endpoints = &single_endpoint, + .name_prefix = "rt711" + } +}; + +static const struct snd_soc_acpi_link_adr rpl_rvp[] = { + { + .mask = BIT(0), + .num_adr = ARRAY_SIZE(rt711_0_adr), + .adr_d = rt711_0_adr, + }, + {} +}; + +struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_machines[] = { + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_machines); + +/* this table is used when there is no I2S codec present */ +struct snd_soc_acpi_mach snd_soc_acpi_intel_rpl_sdw_machines[] = { + { + .link_mask = 0x1, /* link0 required */ + .links = rpl_rvp, + .drv_name = "sof_sdw", + .sof_tplg_filename = "sof-rpl-rt711.tplg", + }, + {}, +}; +EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_rpl_sdw_machines); base-commit: efe30e2cb6ff87467389cece8ce604f6601059f9 -- 2.37.2

2 2

[PATCH 0/4] ASoC: Intel: HSW and BDW updates
by Cezary Rojewski 16 Aug '22

16 Aug '22

Sharing this PR as it touches on cross-driver subjects. Four commits yet two subject. Given the small delta, decided to combine within single PR here. I plan to send these subjects separately to the mailing list though. 1. Rename hsw_rt5640 and bdw_rt286 module names 2. Drop SND_SOC_INTEL_ACPI_MATCH dependency for catpt-driver In regard to 1), this is a followup to discussion that has already happened on upstream [1]. Goal is to align kernel module names for hsw and bdw boards with their siblings found in /intel/boards and /intel/avs/boards. In regard to 2), this is yet another followup [2]. Given the fact that /intel/avs/boards are already their on upstream, intention is to align catpt-driver to approach found in the avs-driver. This is also my suggestion for existing sof-boards: most should be listed under /sof/intel/boards/ as they're not common, really - they're are SOF specific. I believe that in time /intel/common and /intel/boards should be either limited or completely removed. As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. [1]: https://lore.kernel.org/alsa-devel/54fcdedc-278c-06b8-5402-28880597d3f4@int… [2]: https://lore.kernel.org/alsa-devel/d9fd60e0-a157-bda7-a918-5548918d1552@int… [3]: https://github.com/thesofproject/linux/pull/3799 Cezary Rojewski (4): ASoC: Intel: hsw_rt5640: Rename module ASoC: Intel: bdw_rt286: Rename module ASoC: Intel: catpt: Drop SND_SOC_ACPI_INTEL_MATCH dependency ASoC: Intel: Drop legacy HSW/BDW board-match information include/sound/soc-acpi-intel-match.h | 1 - sound/soc/intel/Kconfig | 2 +- sound/soc/intel/boards/Makefile | 8 ++--- sound/soc/intel/catpt/device.c | 33 +++++++++++++++++-- .../common/soc-acpi-intel-hsw-bdw-match.c | 15 --------- 5 files changed, 35 insertions(+), 24 deletions(-) -- 2.25.1

2 6

Rockchip I2S commit possibly ignored
by Geraldo Nascimento 16 Aug '22

16 Aug '22

Hi Mark, I was looking at Rockchip I2S commits and it seems "ASoC: rockchip: i2s: Reset the controller if soft reset failed" was supposed to have been merged to your sound.git but never was. I don't know if this was intentional or a real miss but in any case I'm letting you know. According to 515b436be291ff197c52198282bbb19e79c9d197 'Merge series "Patches to update for rockchip i2s" from Sugar Zhang <sugar.zhang(a)rock-chips.com>:' it should have been merged to your tree. In rockchip-i2s.yaml there's even documentation that references the missing commit. However in the alsa-devel archives, https://mailman.alsa-project.org/pipermail/alsa-devel/2021-August/189050.ht… I see there was no commit info for the unmerged patch. Perhaps this caused it to be black-holed? Thanks, Geraldo Nascimento

2 4

Re: [PATCH v6 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Mark Brown 16 Aug '22

16 Aug '22

On Tue, 16 Aug 2022 10:44:55 +0800, Zhu Ning wrote: > Add device tree binding documentation for Everest ES8326 > > ---- > v5 tested by dtschema > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC commit: 8c6789f4e2d4ee7d6c8c60daa88ea7a4c4cf6779 [2/2] ASoC: codecs: add support for ES8326 commit: 5c439937775d77a334696a98fb2a25dee72ffa2d All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

[RESEND v4 0/2] Add support for linked list to store acp_stream and tdm support.
by Venkata Prasad Potturu 16 Aug '22

16 Aug '22

This patch is to add support for linked list to store acp_stream instead static array and add tdm support for acp I2S stream. Ajit Kumar Pandey (1): ASoC: amd: acp: Initialize list to store acp_stream during pcm_open Venkata Prasad Potturu (1): ASoC: amd: acp: Add TDM support for acp i2s stream Changes since v3: -- Modify to spin_lock() in irq handler and spin_lock_irq() in dma open and close functions instead of spin_lock_irqsave(). -- Add tdm support patch to this series. Changes since v2: -- Add locking mechanism in irq handler, linked list updatation and deletion. Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-i2s.c | 80 +++++++++++++++++++++++++++++++- sound/soc/amd/acp/acp-platform.c | 40 ++++++++-------- sound/soc/amd/acp/amd.h | 16 ++++++- 3 files changed, 113 insertions(+), 23 deletions(-) -- 2.25.1

2 3

[PATCH v5 1/2] ASoC: dt-bindings: Add Everest ES8326 audio CODEC
by Zhu Ning 16 Aug '22

16 Aug '22

Add device tree binding documentation for Everest ES8326 Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ---- v5 tested by dtschema --- .../bindings/sound/everest,es8326.yaml | 116 ++++++++++++++++++ 1 file changed, 116 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8326.yaml diff --git a/Documentation/devicetree/bindings/sound/everest,es8326.yaml b/Documentation/devicetree/bindings/sound/everest,es8326.yaml new file mode 100755 index 000000000000..07781408e788 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/everest,es8326.yaml @@ -0,0 +1,116 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/everest,es8326.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Everest ES8326 audio CODEC + +maintainers: + - David Yang <yangxiaohua(a)everest-semi.com> + +properties: + compatible: + const: everest,es8326 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock for master clock (MCLK) + + clock-names: + items: + - const: mclk + + "#sound-dai-cells": + const: 0 + + everest,jack-pol: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + just the value of reg 57. Bit(3) decides whether the jack polarity is inverted. + Bit(2) decides whether the button on the headset is inverted. + Bit(1)/(0) decides the mic properity to be OMTP/CTIA or auto. + minimum: 0x00 + maximum: 0x0f + default: 0x0f + + everest,mic1-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset plugged. + minimum: 0x00 + maximum: 0x77 + default: 0x22 + + everest,mic2-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: + the value of reg 2A when headset unplugged. + minimum: 0x00 + maximum: 0x77 + default: 0x44 + + everest,jack-detect-inverted: + $ref: /schemas/types.yaml#/definitions/flag + description: + Defined to invert the jack detection. + + everest,interrupt-src: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x58, Defines the interrupt source. + Bit(2) 1 means button press triggers irq, 0 means not. + Bit(3) 1 means PIN9 is the irq source for jack detection. When set to 0, + bias change on PIN9 do not triggers irq. + Bit(4) 1 means PIN27 is the irq source for jack detection. + Bit(5) 1 means PIN9 is the irq source after MIC detect. + Bit(6) 1 means PIN27 is the irq source after MIC detect. + minimum: 0 + maximum: 0x3c + default: 0x08 + + everest,interrupt-clk: + $ref: /schemas/types.yaml#/definitions/uint8 + description: | + value of reg 0x59, Defines the interrupt output behavior. + Bit(0-3) 0 means irq pulse equals 512*internal clock + 1 means irq pulse equals 1024*internal clock + 2 means ... + 7 means irq pulse equals 65536*internal clock + 8 means irq mutes PA + 9 means irq mutes PA and DAC output + Bit(4) 1 means we invert the interrupt output. + Bit(6) 1 means the chip do not detect jack type after button released. + 0 means the chip detect jack type again after button released. + minimum: 0 + maximum: 0x7f + default: 0x45 + +required: + - compatible + - reg + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + es8326: codec@19 { + compatible = "everest,es8326"; + reg = <0x19>; + clocks = <&clks 10>; + clock-names = "mclk"; + #sound-dai-cells = <0>; + everest,mic1-src = [22]; + everest,mic2-src = [44]; + everest,jack-pol = [0e]; + everest,interrupt-src = [08]; + everest,interrupt-clk = [45]; + }; + }; -- 2.36.1

4 5

[PATCH v2] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 16 Aug '22

16 Aug '22

In commit c3ecef21c3f26 ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- V2: rebase against latest version --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 7523bb944b21..d430eece1d6b 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1306,7 +1306,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

3 2

[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Yoga7 14IAL7
by Takashi Iwai 16 Aug '22

16 Aug '22

Lenovo Yoga7 14IAL7 requires the same quirk as Lenovo Yoga9 14IAP7 for fixing the bass speaker problems. Reported-by: Pascal Gross <baipush(a)tutanota.com> Cc: <stable(a)vger.kernel.org> Link: https://lore.kernel.org/r/N9_CjBz--3-2@tutanota.com 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 f544761eb11b..b42496c01c43 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9492,6 +9492,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3855, "Legion 7 16ITHG6", ALC287_FIXUP_LEGION_16ITHG6), + SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.35.3

1 0

[GIT PULL] ASoC fixes for v6.0-rc1
by Mark Brown 16 Aug '22

16 Aug '22

The following changes since commit 40d060b8158e26438398bf1132925f666e3b6480: ASoC: q6asm: use kcalloc() instead of kzalloc() (2022-07-28 11:59:10 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.0-rc1 for you to fetch changes up to b4b5f29a076e52181f63e45a2ad1bc88593072e3: ASoC: codec: tlv320aic32x4: fix mono playback via I2S (2022-08-10 14:52:05 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.0 A relatively large batch of fixes that came in since my pull request, none of them too major and mostly device specific apart from a series of security/robustness improvements from Takashi. ---------------------------------------------------------------- Biju Das (1): ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path Justin Stitt (1): ASoC: SOF: ipc3-topology: Fix clang -Wformat warning Mark Brown (2): ASoC: Fix theoretical buffer overflow by snprintf() TAS2770 fixes Martin Povišer (4): ASoC: tas2770: Set correct FSYNC polarity ASoC: tas2770: Allow mono streams ASoC: tas2770: Drop conflicting set_bias_level power setting ASoC: tas2770: Fix handling of mute/unmute Oder Chiou (1): ASoC: rt5640: Fix the JD voltage dropping issue Philipp Zabel (1): ASoC: codec: tlv320aic32x4: fix mono playback via I2S Pierre-Louis Bossart (1): ASoC: Intel: fix sof_es8336 probe Takashi Iwai (4): ASoC: Intel: avs: Fix potential buffer overflow by snprintf() ASoC: SOF: debug: Fix potential buffer overflow by snprintf() ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf() ASoC: DPCM: Don't pick up BE without substream syed sabakareem (1): ASoC: amd: yc: Update DMI table entries for AMD platforms sound/soc/amd/yc/acp6x-mach.c | 28 +++++++++++ sound/soc/codecs/rt5640.c | 5 +- sound/soc/codecs/tas2770.c | 98 ++++++++++++++++--------------------- sound/soc/codecs/tas2770.h | 5 ++ sound/soc/codecs/tlv320aic32x4.c | 9 ++++ sound/soc/intel/avs/pcm.c | 4 +- sound/soc/intel/boards/sof_es8336.c | 4 +- sound/soc/sh/rz-ssi.c | 26 +++++----- sound/soc/soc-pcm.c | 3 ++ sound/soc/sof/debug.c | 6 +-- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/ipc3-topology.c | 2 +- 12 files changed, 116 insertions(+), 76 deletions(-)

2 1

[PATCH v2] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 16 Aug '22

16 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..15e2a0009080 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.\n", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

2 1

[PATCH v2 0/6] ALSA: hda: Unify codec construction
by Cezary Rojewski 16 Aug '22

16 Aug '22

A follow up to the recent HDAudio fixes series [1]. Given the recently reported regression [2], before the page fault occurring on codec shutdown can be fixed, codec construction procedure needs to be updated for skylake and sof-intel drivers. Drivers: pci-hda and avs need no changes - already making use of snd_hda_codec_device_init(). As subject touches code used by the sof-driver, additional review has been conducted on thesofproject/linux [3]. Changes in v2: - dropped snd_hda_ext_core <-> snd_hda_codec dependency by calling snd_hda_codec_device_init() directly in skylake and sof drivers probe enumeration routines, as suggested by Takashi - skylake/sof portion of the change has been split into two separate patches - new functions that aim to replace hdac_ext codec init & exit functionality are added first - for skylake and sof drivers both - third patch in the series now combines the "field -> pointer" change for hdac_hda_priv->codec plus the codec-enumeration adjustments for skylake and sof drivers Both above are here to keep git bisect happy, as suggested by Pierre [1]: https://lore.kernel.org/alsa-devel/20220706120230.427296-7-cezary.rojewski@… [2]: https://lore.kernel.org/alsa-devel/3c40df55-3aee-1e08-493b-7b30cd84dc00@lin… [3]: https://github.com/thesofproject/linux/pull/3775 Cezary Rojewski (6): ASoC: Intel: Skylake: Introduce HDA codec init and exit routines ASoC: SOF: Intel: Introduce HDA codec init and exit routines ASoC: Intel: Drop hdac_ext usage for codec device creation ALSA: hda: Always free codec on the device release ALSA: hda: Remove codec init and exit routines ALSA: hda: Fix page fault in snd_hda_codec_shutdown() include/sound/hda_codec.h | 2 - include/sound/hdaudio_ext.h | 3 -- sound/hda/ext/hdac_ext_bus.c | 53 ------------------- sound/pci/hda/hda_codec.c | 49 ++++++++--------- sound/soc/codecs/hdac_hda.c | 26 ++++----- sound/soc/codecs/hdac_hda.h | 2 +- sound/soc/intel/boards/hda_dsp_common.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- sound/soc/intel/skylake/skl.c | 53 ++++++++++++++----- sound/soc/sof/intel/hda-codec.c | 55 ++++++++++++++------ 10 files changed, 113 insertions(+), 134 deletions(-) -- 2.25.1

4 10

[PATCH] ASoC: Fix the include guard used for include/sound/wm8904.h
by Christophe JAILLET 16 Aug '22

16 Aug '22

__MFD_WM8994_PDATA_H__ is already used for: include/linux/mfd/wm8994/pdata.h Based on file names, use __MFD_WM8904_PDATA_H__ instead here. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Not sure of if a Fixes: tag is needed. If yes, I would use: 9e50108668a7 ("mfd: Add initial WM8994 support") which introduced the conflict, even if: a91eb199e4dc8 ("ASoC: Initial WM8904 CODEC driver") which introduced the typo is more logical. No strong opinion on it. --- include/sound/wm8904.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/wm8904.h b/include/sound/wm8904.h index 88ac1870510e..f74293a83a17 100644 --- a/include/sound/wm8904.h +++ b/include/sound/wm8904.h @@ -7,8 +7,8 @@ * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> */ -#ifndef __MFD_WM8994_PDATA_H__ -#define __MFD_WM8994_PDATA_H__ +#ifndef __MFD_WM8904_PDATA_H__ +#define __MFD_WM8904_PDATA_H__ /* Used to enable configuration of a GPIO to all zeros */ #define WM8904_GPIO_NO_CONFIG 0x8000 -- 2.34.1

2 1

[PATCH] ASoC: intel: Replace scnprintf() calls with sysfs_emit_at()
by zhaoxiao 16 Aug '22

16 Aug '22

For sysfs outputs, it's safer to use a new helper, sysfs_emit(), instead of the raw sprintf() & co. This patch replaces the open-code with a new helper, sysfs_emit_at(), by passing the string offset. Signed-off-by: zhaoxiao <zhaoxiao(a)uniontech.com> --- sound/soc/sof/intel/hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 6d4ecbe14adf..917d639671a7 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -574,7 +574,7 @@ static void hda_dsp_dump_ext_rom_status(struct snd_sof_dev *sdev, const char *le chip = get_chip_info(sdev->pdata); for (i = 0; i < HDA_EXT_ROM_STATUS_SIZE; i++) { value = snd_sof_dsp_read(sdev, HDA_DSP_BAR, chip->rom_status_reg + i * 0x4); - len += scnprintf(msg + len, sizeof(msg) - len, " 0x%x", value); + len += sysfs_emit_at(msg, len, " 0x%x", value); } dev_printk(level, sdev->dev, "extended rom status: %s", msg); -- 2.20.1

1 0

[PATCH] ASoC: Intel: sof_rt5682: Add support for jsl_rt5682_rt1019
by Sean Hong 16 Aug '22

16 Aug '22

This patch adds the driver data for rt5682 support jsl_rt5682_rt1019. Signed-off-by: Sean Hong <sean.hong(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 12 ++++++++++++ 2 files changed, 21 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312..3a840f3a9 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1100,6 +1100,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1019", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); 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 b95c4b2cd..e374bf163 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -29,6 +29,11 @@ static const struct snd_soc_acpi_codecs rt1015p_spk = { .codecs = {"RTL1015"} }; +static struct snd_soc_acpi_codecs rt1019p_spk = { + .num_codecs = 1, + .codecs = {"RTL1019"} +}; + static const struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -78,6 +83,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .comp_ids = &rt5682_rt5682s_hp, + .drv_name = "jsl_rt5682_rt1019", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1019p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1019.tplg", + }, { .id = "10134242", .drv_name = "jsl_cs4242_mx98360a", -- 2.25.1

2 1

Re: [PATCH 07/13] ASoC: amd: add acp6.2 pdm driver dma ops
by Syed Saba Kareem 16 Aug '22

16 Aug '22

On 8/12/22 19:50, Amadeusz Sławiński wrote: > [CAUTION: External Email] > > On 8/12/2022 2:07 PM, Syed Saba kareem wrote: >> This patch adds PDM driver DMA operations for Pink Sardine Platform. >> >> Signed-off-by: Syed Saba Kareem <Syed.SabaKareem(a)amd.com> >> --- >> sound/soc/amd/ps/acp62.h | 41 +++++ >> sound/soc/amd/ps/ps-pdm-dma.c | 310 ++++++++++++++++++++++++++++++++++ >> 2 files changed, 351 insertions(+) >> >> diff --git a/sound/soc/amd/ps/acp62.h b/sound/soc/amd/ps/acp62.h >> index 563252834b09..525e8bd225a2 100644 >> --- a/sound/soc/amd/ps/acp62.h >> +++ b/sound/soc/amd/ps/acp62.h >> @@ -27,6 +27,31 @@ >> #define ACP_EXT_INTR_STAT_CLEAR_MASK 0xFFFFFFFF >> #define PDM_DMA_STAT 0x10 >> >> +#define PDM_DMA_INTR_MASK 0x10000 >> +#define ACP_ERROR_STAT 29 >> +#define PDM_DECIMATION_FACTOR 2 >> +#define ACP_PDM_CLK_FREQ_MASK 7 >> +#define ACP_WOV_MISC_CTRL_MASK 0x10 >> +#define ACP_PDM_ENABLE 1 >> +#define ACP_PDM_DISABLE 0 >> +#define ACP_PDM_DMA_EN_STATUS 2 >> +#define TWO_CH 2 >> +#define DELAY_US 5 >> +#define ACP_COUNTER 20000 >> + >> +#define ACP_SRAM_PTE_OFFSET 0x03800000 >> +#define PAGE_SIZE_4K_ENABLE 2 >> +#define PDM_PTE_OFFSET 0 >> +#define PDM_MEM_WINDOW_START 0x4000000 >> + >> +#define CAPTURE_MIN_NUM_PERIODS 4 >> +#define CAPTURE_MAX_NUM_PERIODS 4 >> +#define CAPTURE_MAX_PERIOD_SIZE 8192 >> +#define CAPTURE_MIN_PERIOD_SIZE 4096 >> + >> +#define MAX_BUFFER (CAPTURE_MAX_PERIOD_SIZE * CAPTURE_MAX_NUM_PERIODS) >> +#define MIN_BUFFER MAX_BUFFER >> + >> enum acp_config { >> ACP_CONFIG_0 = 0, >> ACP_CONFIG_1, >> @@ -46,6 +71,22 @@ enum acp_config { >> ACP_CONFIG_15, >> }; >> >> +struct pdm_stream_instance { >> + u16 num_pages; >> + u16 channels; >> + dma_addr_t dma_addr; >> + u64 bytescount; >> + void __iomem *acp62_base; >> +}; >> + >> +union acp_pdm_dma_count { >> + struct { >> + u32 low; >> + u32 high; >> + } bcount; > > Fix indentation. > Also you can remove struct name, and access fields directly, so instead > of accessing somevariable.bcount.low, you would use somevariable.low, it > would probably be more readable. > Okay Will fix it. >> + u64 bytescount; >> +}; >> + >> struct pdm_dev_data { >> u32 pdm_irq; >> void __iomem *acp62_base; >> diff --git a/sound/soc/amd/ps/ps-pdm-dma.c >> b/sound/soc/amd/ps/ps-pdm-dma.c >> index bca2ac3e81f5..a98a2015015d 100644 > > ... > >> + >> +static int acp62_pdm_dma_open(struct snd_soc_component *component, >> + struct snd_pcm_substream *substream) >> +{ >> + struct snd_pcm_runtime *runtime; >> + struct pdm_dev_data *adata; >> + struct pdm_stream_instance *pdm_data; >> + int ret; >> + >> + runtime = substream->runtime; >> + adata = dev_get_drvdata(component->dev); >> + pdm_data = kzalloc(sizeof(*pdm_data), GFP_KERNEL); >> + if (!pdm_data) >> + return -EINVAL; >> + >> + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) >> + runtime->hw = acp62_pdm_hardware_capture; >> + >> + ret = snd_pcm_hw_constraint_integer(runtime, >> + SNDRV_PCM_HW_PARAM_PERIODS); >> + if (ret < 0) { >> + dev_err(component->dev, "set integer constraint >> failed\n"); >> + kfree(pdm_data); >> + return ret; >> + } >> + >> + acp62_enable_pdm_interrupts(adata->acp62_base); >> + >> + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) >> + adata->capture_stream = substream; >> + >> + pdm_data->acp62_base = adata->acp62_base; >> + runtime->private_data = pdm_data; > > Should probably kfree(runtime->private_data) in acp62_pdm_dma_close(), > otherwise won't there be a memory leak? > Will fix it. >> + return ret; >> +} >> + > > ... >

1 0

[PATCH 00/13] Add Pink Sardine platform ASoC driver
by Syed Saba kareem 16 Aug '22

16 Aug '22

Pink Sardine platform is new APU series based on acp6.2 design. This patch set adds an ASoC driver for the ACP (Audio CoProcessor) block on AMD Pink Sardine APU with DMIC endpoint support. Syed Saba kareem (13): ASoC: amd: add Pink Sardine platform ACP IP register header ASoC: amd: add Pink Sardine ACP PCI driver ASoC: amd: add acp6.2 init/de-init functions ASoC: amd: add platform devices for acp6.2 pdm driver and dmic driver ASoC: amd: add acp6.2 pdm platform driver ASoC: amd: add acp6.2 irq handler ASoC: amd: add acp6.2 pdm driver dma ops ASoC: amd: add acp6.2 pci driver pm ops ASoC: amd: add acp6.2 pdm driver pm ops ASoC: amd: enable Pink Sardine acp6.2 drivers build ASoC: amd: create platform device for acp6.2 machine driver ASoC: amd: add Pink Sardine machine driver using dmic ASoC: amd: enable Pink sardine platform machine driver build. include/sound/acp62_chip_offset_byte.h | 444 ++++++++++++++++++++++++ sound/soc/amd/Kconfig | 21 ++ sound/soc/amd/Makefile | 1 + sound/soc/amd/ps/Makefile | 9 + sound/soc/amd/ps/acp62.h | 107 ++++++ sound/soc/amd/ps/pci-ps.c | 356 ++++++++++++++++++++ sound/soc/amd/ps/ps-mach.c | 79 +++++ sound/soc/amd/ps/ps-pdm-dma.c | 449 +++++++++++++++++++++++++ 8 files changed, 1466 insertions(+) create mode 100644 include/sound/acp62_chip_offset_byte.h create mode 100644 sound/soc/amd/ps/Makefile create mode 100644 sound/soc/amd/ps/acp62.h create mode 100644 sound/soc/amd/ps/pci-ps.c create mode 100644 sound/soc/amd/ps/ps-mach.c create mode 100644 sound/soc/amd/ps/ps-pdm-dma.c -- 2.25.1

4 22

6.0-RC1 new module for lenovo yoga
by Pascal Gross 16 Aug '22

16 Aug '22

Hello everybody I've seen that there's a module in the new Kernel-RC for bass speaker problems with new Lenovo (written by Philipp Jungkamp): SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN) Could you please add the exact same line for a similar model with the same Problem? SND_PCI_QUIRK(0x17aa, 0x3869, "Lenovo Yoga7 14IAL7", ALC287_FIXUP_YOGA7_14IAL7_BASS_SPK_PIN) I'm neither a developer nor a coder so I don't know how to contribute to the Kernel, I just played around with Philipps patch, changed the lines according to my model and it works now. I would really appreciate your help and of course, if you have any questions feel free to contact me. Thanks in advance, Pascal

1 0

UAC gadget audio sample rates
by Matt Flax 16 Aug '22

16 Aug '22

Hi there, Is it possible to have a USB audio gadget which supports multiple/arbitrary sample rates ? Matt

1 0

[PATCH v6 2/2] ASoC: codecs: add support for ES8326
by Zhu Ning 16 Aug '22

16 Aug '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ------ v6 remove rate 96000 remove HEX suffix --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 905 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 182 ++++++++ 4 files changed, 1094 insertions(+) create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..9465992c2d68 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -913,6 +914,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..81e00863d992 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -455,6 +456,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..975302b2a61d --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,905 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct delayed_work button_press_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling or during an irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + u8 interrupt_src; + u8 interrupt_clk; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL, ES8326_ADC2_VOL, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFP: + snd_soc_component_update_bits(component, ES8326_RESET, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_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 es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(component, ES8326_FMT, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR, + coeff_div[coeff].regb); + } else { + dev_warn(component->dev, "Clock coefficients do not match"); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + regmap_write(es8326->regmap, ES8326_RESET, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x11); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +/* + * For button detection, set the following in soundcard + * snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + */ +static void es8326_jack_button_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, button_press_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + static int button_to_report, press_count; + static int prev_button, cur_button; + + if (!(es8326->jack->status & SND_JACK_HEADSET)) /* Jack unplugged */ + return; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + switch (iface) { + case 0x93: + /* pause button detected */ + cur_button = SND_JACK_BTN_0; + break; + case 0x6f: + /* button volume up */ + cur_button = SND_JACK_BTN_1; + break; + case 0x27: + /* button volume down */ + cur_button = SND_JACK_BTN_2; + break; + case 0x1e: + /* button released or not pressed */ + cur_button = 0; + break; + default: + break; + } + + if ((prev_button == cur_button) && (cur_button != 0)) { + press_count++; + if (press_count > 10) { + /* report a press every 500ms */ + snd_soc_jack_report(es8326->jack, cur_button, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + press_count = 0; + } + button_to_report = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else if (prev_button != cur_button) { + /* mismatch, detect again */ + prev_button = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else { + /* released or no pressed */ + if (button_to_report != 0) { + snd_soc_jack_report(es8326->jack, button_to_report, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + snd_soc_jack_report(es8326->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + button_to_report = 0; + } + } + mutex_unlock(&es8326->lock); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + /* Jack unplugged or spurious IRQ */ + dev_dbg(comp->dev, "No headset detected"); + if (es8326->jack->status & SND_JACK_HEADPHONE) { + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC, es8326->mic2_src); + es8326_disable_micbias(comp); + } + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if (es8326->jack->status & SND_JACK_HEADSET) { + /* detect button */ + queue_delayed_work(system_wq, &es8326->button_press_work, 10); + } else { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, + ES8326_ADC1_SRC, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } + } + mutex_unlock(&es8326->lock); +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + if (es8326->jack->status & SND_JACK_HEADSET) + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(10)); + else + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + unsigned int reg; + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE, es8326->interrupt_src); + regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); + regmap_write(es8326->regmap, ES8326_RESET, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + regmap_read(es8326->regmap, ES8326_CHIP_VERSION, &reg); + if ((reg & ES8326_VERSION_B) == 1) { + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS, 0xDD); + regmap_write(es8326->regmap, ES8326_ANA_VSEL, 0x7F); + regmap_write(es8326->regmap, ES8326_VMIDLOW, 0x0F); + /* enable button detect */ + regmap_write(es8326->regmap, ES8326_HP_DRIVER, 0xA0); + } + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&es8326->jack_detect_work); + es8326_disable_micbias(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-src return %d", ret); + es8326->interrupt_src = ES8326_HP_DET_SRC_PIN9; + } + dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); + + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-clk return %d", ret); + es8326->interrupt_clk = 0x45; + } + dev_dbg(component->dev, "interrupt-clk %x", es8326->interrupt_clk); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + INIT_DELAYED_WORK(&es8326->button_press_work, + es8326_jack_button_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to request IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, + .id_table = es8326_i2c_id, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..8e5ffe5ee10d --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,182 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ +#define ES8326_RESET 0x00 +#define ES8326_CLK_CTL 0x01 +#define ES8326_CLK_INV 0x02 +#define ES8326_CLK_RESAMPLE 0x03 +#define ES8326_CLK_DIV1 0x04 +#define ES8326_CLK_DIV2 0x05 +#define ES8326_CLK_DLL 0x06 +#define ES8326_CLK_MUX 0x07 +#define ES8326_CLK_ADC_SEL 0x08 +#define ES8326_CLK_DAC_SEL 0x09 +#define ES8326_CLK_ADC_OSR 0x0a +#define ES8326_CLK_DAC_OSR 0x0b +#define ES8326_CLK_DIV_CPC 0x0c +#define ES8326_CLK_DIV_BCLK 0x0d +#define ES8326_CLK_TRI 0x0e +#define ES8326_CLK_DIV_LRCK 0x0f +#define ES8326_CLK_VMIDS1 0x10 +#define ES8326_CLK_VMIDS2 0x11 +#define ES8326_CLK_CAL_TIME 0x12 +#define ES8326_FMT 0x13 + +#define ES8326_DAC_MUTE 0x14 +#define ES8326_ADC_MUTE 0x15 +#define ES8326_ANA_PDN 0x16 +#define ES8326_PGA_PDN 0x17 +#define ES8326_VMIDSEL 0x18 +#define ES8326_ANA_LP 0x19 +#define ES8326_ANA_DMS 0x1a +#define ES8326_ANA_MICBIAS 0x1b +#define ES8326_ANA_VSEL 0x1c +#define ES8326_SYS_BIAS 0x1d +#define ES8326_BIAS_SW1 0x1e +#define ES8326_BIAS_SW2 0x1f +#define ES8326_BIAS_SW3 0x20 +#define ES8326_BIAS_SW4 0x21 +#define ES8326_VMIDLOW 0x22 +#define ES8326_PGAGAIN 0x23 +#define ES8326_HP_DRIVER 0x24 +#define ES8326_DAC2HPMIX 0x25 +#define ES8326_HP_VOL 0x26 +#define ES8326_HP_CAL 0x27 +#define ES8326_HP_DRIVER_REF 0x28 +#define ES8326_ADC_SCALE 0x29 +#define ES8326_ADC1_SRC 0x2a +#define ES8326_ADC2_SRC 0x2b +#define ES8326_ADC1_VOL 0x2c +#define ES8326_ADC2_VOL 0x2d +#define ES8326_ADC_RAMPRATE 0x2e +#define ES8326_ALC_RECOVERY 0x32 +#define ES8326_ALC_LEVEL 0x33 +#define ES8326_ADC_HPFS1 0x34 +#define ES8326_ADC_HPFS2 0x35 +#define ES8326_ADC_EQ 0x36 +#define ES8326_HP_OFFSET_CAL 0x4A +#define ES8326_HPL_OFFSET_INI 0x4B +#define ES8326_HPR_OFFSET_INI 0x4C +#define ES8326_DAC_DSM 0x4D +#define ES8326_DAC_RAMPRATE 0x4E +#define ES8326_DAC_VPPSCALE 0x4F +#define ES8326_DAC_VOL 0x50 +#define ES8326_DRC_RECOVERY 0x53 +#define ES8326_DRC_WINSIZE 0x54 +#define ES8326_HPJACK_TIMER 0x56 +#define ES8326_HP_DET 0x57 +#define ES8326_INT_SOURCE 0x58 +#define ES8326_INTOUT_IO 0x59 +#define ES8326_SDINOUT1_IO 0x5A +#define ES8326_SDINOUT23_IO 0x5B +#define ES8326_JACK_PULSE 0x5C + +#define ES8326_PULLUP_CTL 0xF9 +#define ES8326_HP_DETECT 0xFB +#define ES8326_CHIP_ID1 0xFD +#define ES8326_CHIP_ID2 0xFE +#define ES8326_CHIP_VERSION 0xFF + +/* ES8326_RESET */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

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Re: [PATCH v2 3/6] ASoC: Intel: Drop hdac_ext usage for codec device creation
by Mark Brown 16 Aug '22

16 Aug '22

On Mon, Aug 15, 2022 at 07:42:24PM +0200, Cezary Rojewski wrote: > To make snd_hda_codec_device_init() the only constructor for struct > hda_codec instances remaining tasks are: Acked-by: Mark Brown <broonie(a)kernel.org>

1 0

[PATCH 5.19 0894/1157] ASoC: max98390: use linux/gpio/consumer.h to fix build
by Greg Kroah-Hartman 15 Aug '22

15 Aug '22

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit aa7407f807b250eca7697e5fe9a699bc6c2fab71 ] Change the header file to fix build errors in max98390.c: ../sound/soc/codecs/max98390.c: In function 'max98390_i2c_probe': ../sound/soc/codecs/max98390.c:1076:22: error: implicit declaration of function 'devm_gpiod_get_optional'; did you mean 'devm_regulator_get_optional'? [-Werror=implicit-function-declaration] 1076 | reset_gpio = devm_gpiod_get_optional(&i2c->dev, ../sound/soc/codecs/max98390.c:1077:55: error: 'GPIOD_OUT_HIGH' undeclared (first use in this function); did you mean 'GPIOF_INIT_HIGH'? 1077 | "reset", GPIOD_OUT_HIGH); ../sound/soc/codecs/max98390.c:1077:55: note: each undeclared identifier is reported only once for each function it appears in ../sound/soc/codecs/max98390.c:1083:17: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1083 | gpiod_set_value_cansleep(reset_gpio, 0); Fixes: 397ff0249606 ("ASoC: max98390: Add reset gpio control") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Steve Lee <steve.lee.analog(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20220605163123.23537-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/max98390.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98390.c b/sound/soc/codecs/max98390.c index 2a6b1648c884..d83f81d9ff4e 100644 --- a/sound/soc/codecs/max98390.c +++ b/sound/soc/codecs/max98390.c @@ -10,7 +10,7 @@ #include <linux/cdev.h> #include <linux/dmi.h> #include <linux/firmware.h> -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_gpio.h> -- 2.35.1

1 0

[PATCH] ASoC: cs42l42: Only report button state if there was a button interrupt
by Richard Fitzgerald 15 Aug '22

15 Aug '22

Only report a button state change if the interrupt status shows that there was a button event. Previously the code would always drop into the button reporting at the end of interrupt handling if the jack was present. If neither of the button report interrupts were pending it would report all buttons released. This could then lead to a button being reported as released while it is still pressed. Fixes: c5b8ee0879bc ("ASoC: cs42l42: Report jack and button detection") Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index d545a593a251..daafd4251ce6 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1617,7 +1617,6 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) unsigned int current_plug_status; unsigned int current_button_status; unsigned int i; - int report = 0; mutex_lock(&cs42l42->irq_lock); if (cs42l42->suspended) { @@ -1711,13 +1710,15 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) if (current_button_status & CS42L42_M_DETECT_TF_MASK) { dev_dbg(cs42l42->dev, "Button released\n"); - report = 0; + snd_soc_jack_report(cs42l42->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); } else if (current_button_status & CS42L42_M_DETECT_FT_MASK) { - report = cs42l42_handle_button_press(cs42l42); - + snd_soc_jack_report(cs42l42->jack, + cs42l42_handle_button_press(cs42l42), + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); } - snd_soc_jack_report(cs42l42->jack, report, SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); } } -- 2.30.2

2 1

trying to open ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf for PinePhonePro
by GitHub issues - edited 15 Aug '22

15 Aug '22

alsa-project/alsa-ucm-conf issue #199 was edited from ollieparanoid: Hi, we are trying to narrow down why alsa-lib tries to open `/usr/share/alsa/ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf` instead of the proper path `/usr/share/alsa/ucm2/PinePhonePro/PINE64-PinePhonePro-.conf` with custom (to-be-upstreamed to alsa-ucm-conf.git) ucm configurations we are using in postmarketOS for the PinePhonePro: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/community/dev… It only happens for some users, but not for everybody. I've taken a quick look at the code but didn't see right away where the wrong path is built, `git grep UnknownProduct` gives no results. Could somebody give us a hint where to look? I don't really have a clue how this works. Thanks! Downstream discussion: https://gitlab.com/postmarketOS/pmaports/-/merge_requests/3356 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/199 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

trying to open ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf for PinePhonePro
by GitHub issues - opened 15 Aug '22

15 Aug '22

alsa-project/alsa-ucm-conf issue #199 was opened from ollieparanoid: Hi, we are trying to narrow down why alsa-lib tries to open `/usr/share/alsa/ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf` instead of the proper path `/usr/share/alsa/ucm2/PinePhonePro/PINE64-PinePhonePro-.conf` with custom (to-be-upstreamed to alsa-ucm-conf.git) ucm configurations we are using in postmarketOS for the PinePhonePro: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/community/dev… It only happens for some users, but not for everybody. I've taken a quick look at the code but didn't see right away where the wrong path is built, `git grep UnknownProduct` gives no results. Could somebody give us a hint where to look? I don't really have a clue how this works. Thanks! Downstream discussion: https://gitlab.com/postmarketOS/pmaports/-/merge_requests/3356 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/199 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

trying to open ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf for PinePhonePro
by GitHub issues - opened 15 Aug '22

15 Aug '22

alsa-project/alsa-lib issue #260 was opened from ollieparanoid: Hi, we are trying to narrow down why alsa-lib tries to open `/usr/share/alsa/ucm2/conf.d/simple-card/Unknown-UnknownProduct-.conf` instead of the proper path `/usr/share/alsa/ucm2/PinePhonePro/PINE64-PinePhonePro-.conf` with custom (to-be-upstreamed to alsa-ucm-conf.git) ucm configurations we are using in postmarketOS for the PinePhonePro: https://gitlab.com/postmarketOS/pmaports/-/tree/master/device/community/dev… It only happens for some users, but not for everybody. I've taken a quick look at the code but didn't see right away where the wrong path is built, `git grep UnknownProduct` gives no results. Could somebody give us a hint where to look? I don't really have a clue how this works. Thanks! Downstream discussion: https://gitlab.com/postmarketOS/pmaports/-/merge_requests/3356 Issue URL : https://github.com/alsa-project/alsa-lib/issues/260 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: v4: * updated the id to have the correct path v3: * Renamed to include ti, * fixed license to include BSD-2-Clause * removed generic description * renamed the node in the example v2: * Changed the maintainer .../devicetree/bindings/sound/ti,src4xxx.yaml | 48 +++++++++++++++++++ 1 file changed, 48 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,src4xxx.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000000..9681b72b4918 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax(a)flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... -- 2.34.1

2 1

[PATCH] ASoC: mt6359: fix tests for platform_get_irq() failure
by Dan Carpenter 15 Aug '22

15 Aug '22

The platform_get_irq() returns negative error codes. It can't actually return zero, but if it did that should be treated as success. Fixes: eef07b9e0925 ("ASoC: mediatek: mt6359: add MT6359 accdet jack driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/codecs/mt6359-accdet.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/mt6359-accdet.c b/sound/soc/codecs/mt6359-accdet.c index c190628e2905..7f624854948c 100644 --- a/sound/soc/codecs/mt6359-accdet.c +++ b/sound/soc/codecs/mt6359-accdet.c @@ -965,7 +965,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) mutex_init(&priv->res_lock); priv->accdet_irq = platform_get_irq(pdev, 0); - if (priv->accdet_irq) { + if (priv->accdet_irq >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_irq, NULL, mt6359_accdet_irq, IRQF_TRIGGER_HIGH | IRQF_ONESHOT, @@ -979,7 +979,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) if (priv->caps & ACCDET_PMIC_EINT0) { priv->accdet_eint0 = platform_get_irq(pdev, 1); - if (priv->accdet_eint0) { + if (priv->accdet_eint0 >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_eint0, NULL, mt6359_accdet_irq, @@ -994,7 +994,7 @@ static int mt6359_accdet_probe(struct platform_device *pdev) } } else if (priv->caps & ACCDET_PMIC_EINT1) { priv->accdet_eint1 = platform_get_irq(pdev, 2); - if (priv->accdet_eint1) { + if (priv->accdet_eint1 >= 0) { ret = devm_request_threaded_irq(&pdev->dev, priv->accdet_eint1, NULL, mt6359_accdet_irq, -- 2.35.1

2 1

[PATCH v3] ASoC: codecs: add support for the TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

The src4xxx keyword is used for future capability to integrate other codecs similar to the src4392 to the same code base. Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: V3 changes : * converted header to C++ comments * added break to default switch cases * updated SRC vol to SRC volume, scanned for other name issues * now using yaml for the dt binding documentation (separate patch) * Fixed Kconfig to select the _I2C module in menuconfig V2 changes : * Fixed indents in Kconfig * Comment header of source code is now one block * Documented the devicetree binding * sample rate converter MCLK src is non-user settable * hushed spurious dev_info output * src4xxx_remove was empty is removed. * Using switch in hw_params to select port reg. for clarity * Removed interrupt reg. settings - will re-intro. properly later * probe function is now EXPORT_SYMBOL_GPL sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/src4xxx-i2c.c | 47 +++ sound/soc/codecs/src4xxx.c | 513 +++++++++++++++++++++++++++++++++ sound/soc/codecs/src4xxx.h | 113 ++++++++ 5 files changed, 690 insertions(+) create mode 100644 sound/soc/codecs/src4xxx-i2c.c create mode 100644 sound/soc/codecs/src4xxx.c create mode 100644 sound/soc/codecs/src4xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..fe445465f877 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -205,6 +205,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SPDIF + imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 imply SND_SOC_SSM2602_SPI @@ -1471,6 +1472,18 @@ config SND_SOC_SIMPLE_MUX config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SRC4XXX_I2C + tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs" + depends on I2C + select SND_SOC_SRC4XXX + help + Enable support for the TI SRC4XXX family of codecs. These include the + scr4392 which has digital receivers, transmitters, and + a sample rate converter, including numerous ports. + +config SND_SOC_SRC4XXX + tristate + config SND_SOC_SSM2305 tristate "Analog Devices SSM2305 Class-D Amplifier" help diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..a599b727c65b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -231,6 +231,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-src4xxx-objs := src4xxx.o +snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o @@ -579,6 +581,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o +obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c new file mode 100644 index 000000000000..43daa9dc8ab5 --- /dev/null +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -0,0 +1,47 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Driver for SRC4XXX codecs +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt Flax <flatmax(a)flatmax.com> + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "src4xxx.h" + +static int src4xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return src4xxx_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &src4xxx_regmap_config), NULL); +} + +static const struct i2c_device_id src4xxx_i2c_ids[] = { + { "src4392", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); + +static const struct of_device_id src4xxx_of_match[] = { + { .compatible = "ti,src4392", }, + { } +}; +MODULE_DEVICE_TABLE(of, src4xxx_of_match); + + +static struct i2c_driver src4xxx_i2c_driver = { + .driver = { + .name = "src4xxx", + .of_match_table = of_match_ptr(src4xxx_of_match), + }, + .probe = src4xxx_i2c_probe, + .id_table = src4xxx_i2c_ids, +}; +module_i2c_driver(src4xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC SRC4392 CODEC I2C driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c new file mode 100644 index 000000000000..a8f143057b41 --- /dev/null +++ b/sound/soc/codecs/src4xxx.c @@ -0,0 +1,513 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// TI SRC4xxx Audio Codec driver +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt Flax <flatmax(a)flatmax.com> + +#include <linux/module.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "src4xxx.h" + +struct src4xxx { + struct regmap *regmap; + bool master[2]; + int mclk_hz; + struct device *dev; +}; + +enum {SRC4XXX_PORTA, SRC4XXX_PORTB}; + +/* SRC attenuation */ +static const DECLARE_TLV_DB_SCALE(src_tlv, -12750, 50, 0); + +static const struct snd_kcontrol_new src4xxx_controls[] = { + SOC_DOUBLE_R_TLV("SRC Volume", + SRC4XXX_SCR_CTL_30, SRC4XXX_SCR_CTL_31, 0, 255, 1, src_tlv), +}; + +/* I2S port control */ +static const char * const port_out_src_text[] = { + "loopback", "other_port", "DIR", "SRC" +}; +static SOC_ENUM_SINGLE_DECL(porta_out_src_enum, SRC4XXX_PORTA_CTL_03, 4, + port_out_src_text); +static SOC_ENUM_SINGLE_DECL(portb_out_src_enum, SRC4XXX_PORTB_CTL_05, 4, + port_out_src_text); +static const struct snd_kcontrol_new porta_out_control = + SOC_DAPM_ENUM("Port A source select", porta_out_src_enum); +static const struct snd_kcontrol_new portb_out_control = + SOC_DAPM_ENUM("Port B source select", portb_out_src_enum); + +/* Digital audio transmitter control */ +static const char * const dit_mux_text[] = {"Port A", "Port B", "DIR", "SRC"}; +static SOC_ENUM_SINGLE_DECL(dit_mux_enum, SRC4XXX_TX_CTL_07, 3, dit_mux_text); +static const struct snd_kcontrol_new dit_mux_control = + SOC_DAPM_ENUM("DIT source", dit_mux_enum); + +/* SRC control */ +static const char * const src_in_text[] = {"Port A", "Port B", "DIR"}; +static SOC_ENUM_SINGLE_DECL(src_in_enum, SRC4XXX_SCR_CTL_2D, 0, src_in_text); +static const struct snd_kcontrol_new src_in_control = + SOC_DAPM_ENUM("SRC source select", src_in_enum); + +/* DIR control */ +static const char * const dir_in_text[] = {"Ch 1", "Ch 2", "Ch 3", "Ch 4"}; +static SOC_ENUM_SINGLE_DECL(dir_in_enum, SRC4XXX_RCV_CTL_0D, 0, dir_in_text); +static const struct snd_kcontrol_new dir_in_control = + SOC_DAPM_ENUM("Digital Input", dir_in_enum); + +static const struct snd_soc_dapm_widget src4xxx_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("loopback_A"), + SND_SOC_DAPM_INPUT("other_port_A"), + SND_SOC_DAPM_INPUT("DIR_A"), + SND_SOC_DAPM_INPUT("SRC_A"), + SND_SOC_DAPM_MUX("Port A source", + SND_SOC_NOPM, 0, 0, &porta_out_control), + + SND_SOC_DAPM_INPUT("loopback_B"), + SND_SOC_DAPM_INPUT("other_port_B"), + SND_SOC_DAPM_INPUT("DIR_B"), + SND_SOC_DAPM_INPUT("SRC_B"), + SND_SOC_DAPM_MUX("Port B source", + SND_SOC_NOPM, 0, 0, &portb_out_control), + + SND_SOC_DAPM_INPUT("Port_A"), + SND_SOC_DAPM_INPUT("Port_B"), + SND_SOC_DAPM_INPUT("DIR_"), + + /* Digital audio receivers and transmitters */ + SND_SOC_DAPM_OUTPUT("DIR_OUT"), + SND_SOC_DAPM_OUTPUT("SRC_OUT"), + SND_SOC_DAPM_MUX("DIT Out Src", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIT_SHIFT, 1, &dit_mux_control), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF_A_RX", "Playback A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_A_TX", "Capture A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_IN("AIF_B_RX", "Playback B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_B_TX", "Capture B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + + SND_SOC_DAPM_MUX("SRC source", SND_SOC_NOPM, 0, 0, &src_in_control), + + SND_SOC_DAPM_INPUT("MCLK"), + SND_SOC_DAPM_INPUT("RXMCLKI"), + SND_SOC_DAPM_INPUT("RXMCLKO"), + + SND_SOC_DAPM_INPUT("RX1"), + SND_SOC_DAPM_INPUT("RX2"), + SND_SOC_DAPM_INPUT("RX3"), + SND_SOC_DAPM_INPUT("RX4"), + SND_SOC_DAPM_MUX("Digital Input", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIR_SHIFT, 1, &dir_in_control), +}; + +static const struct snd_soc_dapm_route src4xxx_audio_routes[] = { + /* I2S Input to Output Routing */ + {"Port A source", "loopback", "loopback_A"}, + {"Port A source", "other_port", "other_port_A"}, + {"Port A source", "DIR", "DIR_A"}, + {"Port A source", "SRC", "SRC_A"}, + {"Port B source", "loopback", "loopback_B"}, + {"Port B source", "other_port", "other_port_B"}, + {"Port B source", "DIR", "DIR_B"}, + {"Port B source", "SRC", "SRC_B"}, + /* DIT muxing */ + {"DIT Out Src", "Port A", "Capture A"}, + {"DIT Out Src", "Port B", "Capture B"}, + {"DIT Out Src", "DIR", "DIR_OUT"}, + {"DIT Out Src", "SRC", "SRC_OUT"}, + + /* SRC input selection */ + {"SRC source", "Port A", "Port_A"}, + {"SRC source", "Port B", "Port_B"}, + {"SRC source", "DIR", "DIR_"}, + /* SRC mclk selection */ + {"SRC mclk source", "Master (MCLK)", "MCLK"}, + {"SRC mclk source", "Master (RXCLKI)", "RXMCLKI"}, + {"SRC mclk source", "Recovered receiver clk", "RXMCLKO"}, + /* DIR input selection */ + {"Digital Input", "Ch 1", "RX1"}, + {"Digital Input", "Ch 2", "RX2"}, + {"Digital Input", "Ch 3", "RX3"}, + {"Digital Input", "Ch 4", "RX4"}, +}; + + +static const struct snd_soc_component_driver src4xxx_driver = { + .controls = src4xxx_controls, + .num_controls = ARRAY_SIZE(src4xxx_controls), + + .dapm_widgets = src4xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(src4xxx_dapm_widgets), + .dapm_routes = src4xxx_audio_routes, + .num_dapm_routes = ARRAY_SIZE(src4xxx_audio_routes), +}; + +static int src4xxx_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = SRC4XXX_BUS_MASTER; + src4xxx->master[dai->id] = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + src4xxx->master[dai->id] = false; + break; + default: + return -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= SRC4XXX_BUS_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= SRC4XXX_BUS_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= SRC4XXX_BUS_RIGHT_J_24; + break; + default: + return -EINVAL; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + break; + } + + regmap_update_bits(src4xxx->regmap, SRC4XXX_BUS_FMT(dai->id), + SRC4XXX_BUS_FMT_MS_MASK, ctrl); + + return 0; +} + +static int src4xxx_set_mclk_hz(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + + dev_info(component->dev, "changing mclk rate from %d to %d Hz\n", + src4xxx->mclk_hz, freq); + src4xxx->mclk_hz = freq; + + return 0; +} + +static int src4xxx_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 src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int mclk_div; + int val, pj, jd, d; + int reg; + int ret; + + switch (dai->id) { + case SRC4XXX_PORTB: + reg = SRC4XXX_PORTB_CTL_06; + break; + default: + reg = SRC4XXX_PORTA_CTL_04; + break; + } + + if (src4xxx->master[dai->id]) { + mclk_div = src4xxx->mclk_hz/params_rate(params); + if (src4xxx->mclk_hz != mclk_div*params_rate(params)) { + dev_err(component->dev, + "mclk %d / rate %d has a remainder.\n", + src4xxx->mclk_hz, params_rate(params)); + return -EINVAL; + } + + val = ((int)mclk_div - 128) / 128; + if ((val < 0) | (val > 3)) { + dev_err(component->dev, + "div register setting %d is out of range\n", + val); + dev_err(component->dev, + "unsupported sample rate %d Hz for the master clock of %d Hz\n", + params_rate(params), src4xxx->mclk_hz); + return -EINVAL; + } + + /* set the TX DIV */ + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + /* set the PLL for the digital receiver */ + switch (src4xxx->mclk_hz) { + case 24576000: + pj = 0x22; + jd = 0x00; + d = 0x00; + break; + case 22579200: + pj = 0x22; + jd = 0x1b; + d = 0xa3; + break; + default: + /* don't error out here, + * other parts of the chip are still functional + */ + dev_info(component->dev, + "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + break; + } + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_0F); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_10); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_11); + + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret < 0) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + return regmap_update_bits(src4xxx->regmap, reg, + SRC4XXX_MCLK_DIV_MASK, val); + } else { + dev_info(dai->dev, "not setting up MCLK as not master\n"); + } + + return 0; +}; + +static const struct snd_soc_dai_ops src4xxx_dai_ops = { + .hw_params = src4xxx_hw_params, + .set_sysclk = src4xxx_set_mclk_hz, + .set_fmt = src4xxx_set_dai_fmt, +}; + +#define SRC4XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define SRC4XXX_RATES (SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000|\ + SNDRV_PCM_RATE_88200|\ + SNDRV_PCM_RATE_96000|\ + SNDRV_PCM_RATE_176400|\ + SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver src4xxx_dai_driver[] = { + { + .id = SRC4XXX_PORTA, + .name = "src4xxx-portA", + .playback = { + .stream_name = "Playback A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, + { + .id = SRC4XXX_PORTB, + .name = "src4xxx-portB", + .playback = { + .stream_name = "Playback B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, +}; + +static const struct reg_default src4xxx_reg_defaults[] = { + { SRC4XXX_PWR_RST_01, 0x00 }, /* all powered down intially */ + { SRC4XXX_PORTA_CTL_03, 0x00 }, + { SRC4XXX_PORTA_CTL_04, 0x00 }, + { SRC4XXX_PORTB_CTL_05, 0x00 }, + { SRC4XXX_PORTB_CTL_06, 0x00 }, + { SRC4XXX_TX_CTL_07, 0x00 }, + { SRC4XXX_TX_CTL_08, 0x00 }, + { SRC4XXX_TX_CTL_09, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MSK_0B, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MODE_0C, 0x00 }, + { SRC4XXX_RCV_CTL_0D, 0x00 }, + { SRC4XXX_RCV_CTL_0E, 0x00 }, + { SRC4XXX_RCV_PLL_0F, 0x00 }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_10, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_11, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RVC_IRQ_MSK_16, 0x00 }, + { SRC4XXX_RVC_IRQ_MSK_17, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_18, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_19, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_1A, 0x00 }, + { SRC4XXX_GPIO_1_1B, 0x00 }, + { SRC4XXX_GPIO_2_1C, 0x00 }, + { SRC4XXX_GPIO_3_1D, 0x00 }, + { SRC4XXX_GPIO_4_1E, 0x00 }, + { SRC4XXX_SCR_CTL_2D, 0x00 }, + { SRC4XXX_SCR_CTL_2E, 0x00 }, + { SRC4XXX_SCR_CTL_2F, 0x00 }, + { SRC4XXX_SCR_CTL_30, 0x00 }, + { SRC4XXX_SCR_CTL_31, 0x00 }, +}; + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)) +{ + struct src4xxx *src4xxx; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + src4xxx = devm_kzalloc(dev, sizeof(*src4xxx), GFP_KERNEL); + if (!src4xxx) + return -ENOMEM; + + src4xxx->regmap = regmap; + src4xxx->dev = dev; + src4xxx->mclk_hz = 0; /* mclk has not been configured yet */ + dev_set_drvdata(dev, src4xxx); + + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_RESET); + if (ret < 0) + dev_err(dev, "Failed to issue reset: %d\n", ret); + usleep_range(1, 500); /* sleep for more then 500 ns */ + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_POWER_DOWN); + if (ret < 0) + dev_err(dev, "Failed to decommission reset: %d\n", ret); + usleep_range(500, 1000); /* sleep for 500 us or more */ + + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_PWR_RST_01, + SRC4XXX_POWER_ENABLE, SRC4XXX_POWER_ENABLE); + if (ret < 0) + dev_err(dev, "Failed to port A and B : %d\n", ret); + + /* set receiver to use master clock (rcv mclk is most likely jittery) */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0D, + SRC4XXX_RXCLK_MCLK, SRC4XXX_RXCLK_MCLK); + if (ret < 0) + dev_err(dev, + "Failed to enable mclk as the PLL1 DIR reference : %d\n", ret); + + /* default to leaving the PLL2 running on loss of lock, divide by 8 */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0E, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL); + if (ret < 0) + dev_err(dev, "Failed to enable mclk rec and div : %d\n", ret); + + ret = devm_snd_soc_register_component(dev, &src4xxx_driver, + src4xxx_dai_driver, ARRAY_SIZE(src4xxx_dai_driver)); + if (ret == 0) + dev_info(dev, "src4392 probe ok %d\n", ret); + return ret; +} +EXPORT_SYMBOL_GPL(src4xxx_probe); + +static bool src4xxx_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SRC4XXX_RES_00: + case SRC4XXX_GLOBAL_ITR_STS_02: + case SRC4XXX_SRC_DIT_STS_0A: + case SRC4XXX_NON_AUDIO_D_12: + case SRC4XXX_RVC_STS_13: + case SRC4XXX_RVC_STS_14: + case SRC4XXX_RVC_STS_15: + case SRC4XXX_SUB_CODE_1F: + case SRC4XXX_SUB_CODE_20: + case SRC4XXX_SUB_CODE_21: + case SRC4XXX_SUB_CODE_22: + case SRC4XXX_SUB_CODE_23: + case SRC4XXX_SUB_CODE_24: + case SRC4XXX_SUB_CODE_25: + case SRC4XXX_SUB_CODE_26: + case SRC4XXX_SUB_CODE_27: + case SRC4XXX_SUB_CODE_28: + case SRC4XXX_PC_PREAMBLE_HI_29: + case SRC4XXX_PC_PREAMBLE_LO_2A: + case SRC4XXX_PD_PREAMBLE_HI_2B: + case SRC4XXX_PC_PREAMBLE_LO_2C: + case SRC4XXX_IO_RATIO_32: + case SRC4XXX_IO_RATIO_33: + return true; + } + + if (reg > SRC4XXX_IO_RATIO_33 && reg < SRC4XXX_PAGE_SEL_7F) + return true; + + return false; +} + +const struct regmap_config src4xxx_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + .max_register = SRC4XXX_IO_RATIO_33, + + .reg_defaults = src4xxx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(src4xxx_reg_defaults), + .volatile_reg = src4xxx_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(src4xxx_regmap_config); + +MODULE_DESCRIPTION("ASoC SRC4XXX CODEC driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.h b/sound/soc/codecs/src4xxx.h new file mode 100644 index 000000000000..5bf778fb9945 --- /dev/null +++ b/sound/soc/codecs/src4xxx.h @@ -0,0 +1,113 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// src4xxx.h -- SRC4XXX ALSA SoC audio driver +// +// Copyright 2021-2022 Deqx Pty Ltd +// Author: Matt R Flax <flatmax(a)flatmax.com> + +#ifndef __SRC4XXX_H__ +#define __SRC4XXX_H__ + +#define SRC4XXX_RES_00 0x00 +#define SRC4XXX_PWR_RST_01 0x01 +#define SRC4XXX_RESET 0x80 +#define SRC4XXX_POWER_DOWN 0x00 +#define SRC4XXX_POWER_ENABLE 0x20 +#define SRC4XXX_ENABLE_SRC 0x1 +#define SRC4XXX_ENABLE_SRC_SHIFT 0 +#define SRC4XXX_ENABLE_DIR 0x2 +#define SRC4XXX_ENABLE_DIR_SHIFT 1 +#define SRC4XXX_ENABLE_DIT 0x4 +#define SRC4XXX_ENABLE_DIT_SHIFT 2 +#define SRC4XXX_ENABLE_PORT_B 0x8 +#define SRC4XXX_ENABLE_PORT_B_SHIFT 3 +#define SRC4XXX_ENABLE_PORT_A 0x10 +#define SRC4XXX_ENABLE_PORT_A_SHIFT 4 + +#define SRC4XXX_PORTA_CTL_03 0x03 +#define SRC4XXX_BUS_MASTER 0x8 +#define SRC4XXX_BUS_LEFT_J 0x0 +#define SRC4XXX_BUS_I2S 0x1 +#define SRC4XXX_BUS_RIGHT_J_16 0x4 +#define SRC4XXX_BUS_RIGHT_J_18 0x5 +#define SRC4XXX_BUS_RIGHT_J_20 0x6 +#define SRC4XXX_BUS_RIGHT_J_24 0x7 +#define SRC4XXX_BUS_FMT_MS_MASK 0xf + +#define SRC4XXX_PORTA_CTL_04 0x04 +#define SRC4XXX_MCLK_DIV_MASK 0x3 + +#define SRC4XXX_BUS_FMT(id) (SRC4XXX_PORTA_CTL_03+2*id) +#define SRC4XXX_BUS_CLK(id) (SRC4XXX_PORTA_CTL_04+2*id) + +#define SRC4XXX_PORTB_CTL_05 0x05 +#define SRC4XXX_PORTB_CTL_06 0x06 + +#define SRC4XXX_TX_CTL_07 0x07 +#define SRC4XXX_TX_MCLK_DIV_MASK 0x60 +#define SRC4XXX_TX_MCLK_DIV_SHIFT 5 + +#define SRC4XXX_TX_CTL_08 0x08 +#define SRC4XXX_TX_CTL_09 0x09 +#define SRC4XXX_SRC_DIT_IRQ_MSK_0B 0x0B +#define SRC4XXX_SRC_BTI_EN 0x01 +#define SRC4XXX_SRC_TSLIP_EN 0x02 +#define SRC4XXX_SRC_DIT_IRQ_MODE_0C 0x0C +#define SRC4XXX_RCV_CTL_0D 0x0D +#define SRC4XXX_RXCLK_RXCKI 0x0 +#define SRC4XXX_RXCLK_MCLK 0x8 +#define SRC4XXX_RCV_CTL_0E 0x0E +#define SRC4XXX_REC_MCLK_EN 0x1 +#define SRC4XXX_PLL2_DIV_0 (0x0<<1) +#define SRC4XXX_PLL2_DIV_2 (0x1<<1) +#define SRC4XXX_PLL2_DIV_4 (0x2<<1) +#define SRC4XXX_PLL2_DIV_8 (0x3<<1) +#define SRC4XXX_PLL2_LOL 0x8 +#define SRC4XXX_RCV_PLL_0F 0x0F +#define SRC4XXX_RCV_PLL_10 0x10 +#define SRC4XXX_RCV_PLL_11 0x11 +#define SRC4XXX_RVC_IRQ_MSK_16 0x16 +#define SRC4XXX_RVC_IRQ_MSK_17 0x17 +#define SRC4XXX_RVC_IRQ_MODE_18 0x18 +#define SRC4XXX_RVC_IRQ_MODE_19 0x19 +#define SRC4XXX_RVC_IRQ_MODE_1A 0x1A +#define SRC4XXX_GPIO_1_1B 0x1B +#define SRC4XXX_GPIO_2_1C 0x1C +#define SRC4XXX_GPIO_3_1D 0x1D +#define SRC4XXX_GPIO_4_1E 0x1E +#define SRC4XXX_SCR_CTL_2D 0x2D +#define SRC4XXX_SCR_CTL_2E 0x2E +#define SRC4XXX_SCR_CTL_2F 0x2F +#define SRC4XXX_SCR_CTL_30 0x30 +#define SRC4XXX_SCR_CTL_31 0x31 +#define SRC4XXX_PAGE_SEL_7F 0x7F + +// read only registers +#define SRC4XXX_GLOBAL_ITR_STS_02 0x02 +#define SRC4XXX_SRC_DIT_STS_0A 0x0A +#define SRC4XXX_NON_AUDIO_D_12 0x12 +#define SRC4XXX_RVC_STS_13 0x13 +#define SRC4XXX_RVC_STS_14 0x14 +#define SRC4XXX_RVC_STS_15 0x15 +#define SRC4XXX_SUB_CODE_1F 0x1F +#define SRC4XXX_SUB_CODE_20 0x20 +#define SRC4XXX_SUB_CODE_21 0x21 +#define SRC4XXX_SUB_CODE_22 0x22 +#define SRC4XXX_SUB_CODE_23 0x23 +#define SRC4XXX_SUB_CODE_24 0x24 +#define SRC4XXX_SUB_CODE_25 0x25 +#define SRC4XXX_SUB_CODE_26 0x26 +#define SRC4XXX_SUB_CODE_27 0x27 +#define SRC4XXX_SUB_CODE_28 0x28 +#define SRC4XXX_PC_PREAMBLE_HI_29 0x29 +#define SRC4XXX_PC_PREAMBLE_LO_2A 0x2A +#define SRC4XXX_PD_PREAMBLE_HI_2B 0x2B +#define SRC4XXX_PC_PREAMBLE_LO_2C 0x2C +#define SRC4XXX_IO_RATIO_32 0x32 +#define SRC4XXX_IO_RATIO_33 0x33 + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)); +extern const struct regmap_config src4xxx_regmap_config; + +#endif /* __SRC4XXX_H__ */ -- 2.34.1

2 2

[PATCH v2] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: v2: * Added myself as the maintainer .../devicetree/bindings/sound/src4xxx.yaml | 50 +++++++++++++++++++ 1 file changed, 50 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/src4xxx.yaml diff --git a/Documentation/devicetree/bindings/sound/src4xxx.yaml b/Documentation/devicetree/bindings/sound/src4xxx.yaml new file mode 100644 index 000000000000..368bb4ba1a77 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/src4xxx.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax(a)flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + description: + The I2C address of the device + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + src4392@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... -- 2.34.1

3 2

[PATCH v3 0/3] DT binding for sample format conversion
by Sameer Pujar 15 Aug '22

15 Aug '22

DT binding properties are available to fixup rate and channel parameters of a DAI. This series extends this to sample format conversion as well. With this now DAI PCM parameters (channels, sample rate and sample format) can be fixed up as necessary in an audio path. Changelog: ========== v2->v3: ------- * Fix DT binding errors in simple-card.yaml * Drop simple-card binding changes from the series since there is no usage requirement. It can be extended later when necessary. * Use definitions instead of properties in the common schema and re-use this in audio-graph related bindings. v1->v2: ------- * Move DAI params properties to a new schema and re-use this for simple-card and audio-graph-card. * Use string type for DAI format binding as suggested by Krzysztof. Sameer Pujar (3): ASoC: dt-bindings: Definitions for DAI params ASoC: dt-bindings: Add sample format conversion ASoC: simple-card-utils: Fixup DAI sample format .../bindings/sound/audio-graph-port.yaml | 17 ++++----- .../devicetree/bindings/sound/audio-graph.yaml | 9 ++--- .../devicetree/bindings/sound/dai-params.yaml | 40 ++++++++++++++++++++++ include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 34 ++++++++++++++++++ 5 files changed, 89 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/dai-params.yaml -- 2.7.4

3 5

[PATCH] ASoC: Variable type completion
by Xin Gao 15 Aug '22

15 Aug '22

'unsigned int' is better than 'unsigned'. Signed-off-by: Xin Gao <gaoxin(a)cdjrlc.com> --- sound/soc/codecs/uda134x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 037833c509f7..607897f1f357 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -450,7 +450,7 @@ static int uda134x_soc_probe(struct snd_soc_component *component) struct uda134x_priv *uda134x = snd_soc_component_get_drvdata(component); struct uda134x_platform_data *pd = uda134x->pd; const struct snd_soc_dapm_widget *widgets; - unsigned num_widgets; + unsigned int num_widgets; int ret; printk(KERN_INFO "UDA134X SoC Audio Codec\n"); -- 2.30.2

2 1

[PATCH] ASoC: sam9g20_wm8731: Simplify some error message
by Christophe JAILLET 15 Aug '22

15 Aug '22

dev_err_probe() already prints the error code in a human readable way, so there is no need to duplicate it as a numerical value at the end of the message. Fixes: 29f4078f777f ("ASoC: sam9g20_wm8731: Use dev_err_probe() for snd_soc_register_card()") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 4d25fb61c652..1430642c8433 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -172,7 +172,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) ret = snd_soc_register_card(card); if (ret) { dev_err_probe(&pdev->dev, ret, - "snd_soc_register_card() failed: %d\n", ret); + "snd_soc_register_card() failed\n"); goto err; } -- 2.34.1

2 1

[PATCH] ASoC: tlv320adcx140: Fix a typo in a comment
by Christophe JAILLET 15 Aug '22

15 Aug '22

s/TLV320ADCX104/TLV320ADCX140/ Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/tlv320adcx140.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index d7d4e3a88b5c..795b5f7194e6 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -1,5 +1,5 @@ // SPDX-License-Identifier: GPL-2.0 -// TLV320ADCX104 Sound driver +// TLV320ADCX140 Sound driver // Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com/ #ifndef _TLV320ADCX140_H -- 2.34.1

2 1

[PATCH 0/8] ASoC: Replace sprintf() with sysfs_emit()
by Takashi Iwai 15 Aug '22

15 Aug '22

Hi, this is a patch set for rather simple conversions from the plain sprintf() & co to the new helpers, sysfs_emit() and sysfs_emit_at(). No functional changes are expected. Takashi === Takashi Iwai (8): ASoC: cs43130: Replace scnprintf() with sysfs_emit() ASoC: tlv320aic26: Replace sprintf() with sysfs_emit() ASoC: Intel: sst: Replace sprintf() with sysfs_emit() ASoC: Intel: catpt: Replace sprintf() with sysfs_emit() ASoC: Intel: skylake: Replace sprintf() with sysfs_emit() ASoC: core: Replace sprintf() with sysfs_emit() ASoC: DAPM: Replace sprintf() calls with sysfs_emit_at() ASoC: omap: Replace sprintf() with sysfs_emit() sound/soc/codecs/cs43130.c | 11 +++++------ sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/intel/atom/sst/sst.c | 8 ++++---- sound/soc/intel/catpt/sysfs.c | 6 +++--- sound/soc/intel/skylake/skl-nhlt.c | 2 +- sound/soc/soc-core.c | 2 +- sound/soc/soc-dapm.c | 9 ++++----- sound/soc/ti/omap-mcbsp-st.c | 6 +++--- sound/soc/ti/omap-mcbsp.c | 8 ++++---- 9 files changed, 26 insertions(+), 28 deletions(-) -- 2.35.3

4 12

[PATCH v2] ASoC: imx-rpmsg: Support configure sysclk for codec dai
by Shengjiu Wang 15 Aug '22

15 Aug '22

Some codecs need to configure the sysclk even with slave mode, otherwise it may not work properly with some case. wm8960 is the one that need sysclk be configured, so add late_probe() to call the snd_soc_dai_set_sysclk() of codec Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2: - use unsigned long for sysclk sound/soc/fsl/imx-rpmsg.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..4d99f4858a14 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned long sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using -- 2.34.1

2 1

[PATCH] ASoC: mediatek: mt8186: remove unnecessary NULL check
by Dan Carpenter 15 Aug '22

15 Aug '22

The "i2s_priv" pointer cannot be NULL. Some NULL checks were deleted in commit d7bffbe9cbd3 ("ASoC: mediatek: mt8186: remove unnecessary judgments") but this one was accidentally left behind. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c index ec79e2f2a54d..d7a227169548 100644 --- a/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c +++ b/sound/soc/mediatek/mt8186/mt8186-dai-i2s.c @@ -968,7 +968,7 @@ static int mtk_dai_i2s_config(struct mtk_base_afe *afe, } /* set share i2s */ - if (i2s_priv && i2s_priv->share_i2s_id >= 0) { + if (i2s_priv->share_i2s_id >= 0) { ret = mtk_dai_i2s_config(afe, params, i2s_priv->share_i2s_id); if (ret) return ret; -- 2.35.1

2 1

[PATCH v4] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 15 Aug '22

15 Aug '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Beside conversion, 'fsl,shared-interrupt' and '#sound-dai-cells' are added for they are already used by some dts. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v4 - remove $ref for boolean properties changes in v3 - address comments from Krzysztof changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 216 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 216 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..70c4111d59c7 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,216 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + lsb-first: + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + type: boolean + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + description: | + SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + type: boolean + + fsl,sai-asynchronous: + description: | + SAI will work in the asynchronous mode, which means both transmitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + type: boolean + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + maxItems: 16 + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,shared-interrupt: + description: Interrupt is shared with other modules. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + dma-names: + items: + - const: rx + - const: tx + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

3 2

[PATCH v3] ASoC: Intel: cirrus-common: Use UID to map correct amp to prefix
by Stefan Binding 15 Aug '22

15 Aug '22

Since the order of the amps in the ACPI determines the device name, and the ACPI order may change depending on hardware configuration, use UID to dynamically compute the dai links, allowing dynamic assignment of the name_prefix. The UIDs for these amps in ACPI are fixed, and map to a name_prefix, where: UID 0x0 -> WL UID 0x1 -> WR UID 0x2 -> TL UID 0x3 -> TR Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 92 ++++++++++++---------- 1 file changed, 49 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..6e39eda77385 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,10 +140,51 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +/* + * Expected UIDs are integers (stored as strings). + * UID Mapping is fixed: + * UID 0x0 -> WL + * UID 0x1 -> WR + * UID 0x2 -> TL + * UID 0x3 -> TR + * Note: If there are less than 4 Amps, UIDs still map to WL/WR/TL/TR. Dynamic code will only create + * dai links for UIDs which exist, and ignore non-existant ones. Only 2 or 4 amps are expected. + * Return number of codecs found. + */ +static int cs35l41_compute_codec_conf(void) +{ + const char * const uid_strings[] = { "0", "1", "2", "3" }; + unsigned int uid, sz = 0; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, uid_strings[uid], -1); + if (!adev) { + pr_devel("Cannot find match for HID %s UID %u (%s)\n", CS35L41_HID, uid, + cs35l41_name_prefixes[uid]); + continue; + } + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[sz].name = dev_name(physdev); + cs35l41_components[sz].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[sz].dlc.name = dev_name(physdev); + cs35l41_codec_conf[sz].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + sz++; + } + + if (sz != 2 && sz != 4) + pr_warn("Invalid number of cs35l41 amps found: %d, expected 2 or 4\n", sz); + return sz; +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + link->num_codecs = cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; - link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; link->ops = &cs35l41_ops; } -- 2.34.1

3 2

[PATCH] ASoC: imx-rpmsg: Support configure sysclk for codec dai
by Shengjiu Wang 15 Aug '22

15 Aug '22

Some codecs need to configure the sysclk even with slave mode, otherwise it may not work properly with some case. wm8960 is the one that need sysclk be configured, so add late_probe() to call the snd_soc_dai_set_sysclk() of codec Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-rpmsg.c | 29 +++++++++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/sound/soc/fsl/imx-rpmsg.c b/sound/soc/fsl/imx-rpmsg.c index 2e117311e582..bef4bd997c98 100644 --- a/sound/soc/fsl/imx-rpmsg.c +++ b/sound/soc/fsl/imx-rpmsg.c @@ -19,6 +19,7 @@ struct imx_rpmsg { struct snd_soc_dai_link dai; struct snd_soc_card card; + unsigned int sysclk; }; static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { @@ -28,6 +29,27 @@ static const struct snd_soc_dapm_widget imx_rpmsg_dapm_widgets[] = { SND_SOC_DAPM_MIC("Main MIC", NULL), }; +static int imx_rpmsg_late_probe(struct snd_soc_card *card) +{ + struct imx_rpmsg *data = snd_soc_card_get_drvdata(card); + struct snd_soc_pcm_runtime *rtd = list_first_entry(&card->rtd_list, + struct snd_soc_pcm_runtime, list); + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + struct device *dev = card->dev; + int ret; + + if (!data->sysclk) + return 0; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, data->sysclk, SND_SOC_CLOCK_IN); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set sysclk in %s\n", __func__); + return ret; + } + + return 0; +} + static int imx_rpmsg_probe(struct platform_device *pdev) { struct snd_soc_dai_link_component *dlc; @@ -72,12 +94,18 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->dai.codecs->dai_name = "snd-soc-dummy-dai"; data->dai.codecs->name = "snd-soc-dummy"; } else { + struct clk *clk; + data->dai.codecs->of_node = args.np; ret = snd_soc_get_dai_name(&args, &data->dai.codecs->dai_name); if (ret) { dev_err(&pdev->dev, "Unable to get codec_dai_name\n"); goto fail; } + + clk = devm_get_clk_from_child(&pdev->dev, args.np, NULL); + if (!IS_ERR(clk)) + data->sysclk = clk_get_rate(clk); } data->dai.cpus->dai_name = dev_name(&rpmsg_pdev->dev); @@ -103,6 +131,7 @@ static int imx_rpmsg_probe(struct platform_device *pdev) data->card.owner = THIS_MODULE; data->card.dapm_widgets = imx_rpmsg_dapm_widgets; data->card.num_dapm_widgets = ARRAY_SIZE(imx_rpmsg_dapm_widgets); + data->card.late_probe = imx_rpmsg_late_probe; /* * Inoder to use common api to get card name and audio routing. * Use parent of_node for this device, revert it after finishing using -- 2.34.1

4 4

[PATCH v1] ALSA: hda: cs35l41: Clarify support for CSC3551 without _DSD Properties
by Stefan Binding 15 Aug '22

15 Aug '22

For devices which use HID CSC3551, correct ACPI _DSD properties are required to be able support those systems. Add error message to clarify this. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 129bffb431c2..b9d94f3a5b99 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1163,6 +1163,11 @@ static int cs35l41_no_acpi_dsd(struct cs35l41_hda *cs35l41, struct device *physd hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; } else { + /* + * Note: CLSA010(0/1) are special cases which use a slightly different design. + * All other HIDs e.g. CSC3551 require valid ACPI _DSD properties to be supported. + */ + dev_err(cs35l41->dev, "Error: ACPI _DSD Properties are missing for HID %s.", hid); hw_cfg->valid = false; hw_cfg->gpio1.valid = false; hw_cfg->gpio2.valid = false; -- 2.34.1

1 0

Re: [PATCH v1] ALSA: hda/realtek: Add quirks for ASUS Zenbooks using CS35L41
by Takashi Iwai 15 Aug '22

15 Aug '22

On Mon, 15 Aug 2022 16:19:53 +0200, Stefan Binding wrote: > > These Asus Zenbook laptop use Realtek HDA codec combined with > 2xCS35L41 Amplifiers using SPI. > > Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Thanks, applied. Takashi

1 0

[RFC PATCH] Soundwire: Initialize multi_link with fwnode props
by Khalid Masum 15 Aug '22

15 Aug '22

According to the TODO, In sw_bus_master_add, bus->multi_link is to be populated with properties from FW node props. Make this happen by creating a new fwnode_handle flag FWNODE_FLAG_MULTI_LINKED and use the flag to store the multi_link value from intel_link_startup. Use this flag to initialize bus->multi_link. Signed-off-by: Khalid Masum <khalid.masum.92(a)gmail.com> --- I do not think adding a new flag for fwnode_handle is a good idea. So, what would be the best way to initialize bus->multilink with fwnode props? -- Khalid Masum drivers/soundwire/bus.c | 4 ++-- drivers/soundwire/intel.c | 1 + include/linux/fwnode.h | 1 + 3 files changed, 4 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index a2bfb0434a67..80df1672c60b 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -74,9 +74,9 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* * Initialize multi_link flag - * TODO: populate this flag by reading property from FW node */ - bus->multi_link = false; + bus->multi_link = (fwnode->flags & FWNODE_FLAG_MULTI_LINKED) + == FWNODE_FLAG_MULTI_LINKED; if (bus->ops->read_prop) { ret = bus->ops->read_prop(bus); if (ret < 0) { diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index 505c5ef061e3..034d1c523ddf 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -1347,6 +1347,7 @@ int intel_link_startup(struct auxiliary_device *auxdev) */ bus->multi_link = true; bus->hw_sync_min_links = 1; + dev->fwnode->flags |= FWNODE_FLAG_MULTI_LINKED; } /* Initialize shim, controller */ diff --git a/include/linux/fwnode.h b/include/linux/fwnode.h index 9a81c4410b9f..446a52744953 100644 --- a/include/linux/fwnode.h +++ b/include/linux/fwnode.h @@ -32,6 +32,7 @@ struct device; #define FWNODE_FLAG_NOT_DEVICE BIT(1) #define FWNODE_FLAG_INITIALIZED BIT(2) #define FWNODE_FLAG_NEEDS_CHILD_BOUND_ON_ADD BIT(3) +#define FWNODE_FLAG_MULTI_LINKED BIT(4) struct fwnode_handle { struct fwnode_handle *secondary; -- 2.37.1

5 8

[PATCH] ASoC: soc-utils: Improve kerneldoc for snd_soc_tdm_params_to_bclk()
by Richard Fitzgerald 15 Aug '22

15 Aug '22

The statement that snd_soc_tdm_params_to_bclk() is equivalent to snd_soc_params_to_bclk() if tdm_width==tdm_slots==0 is not accurate, it is only true is slot_multiple is also <2. However, the description of special-case behaviour in terms of pairs of tdm_width and tdm_slot values is not particularly helpful so we might as well take the opportunity to rework the description to say the same thing in a simpler way. The behaviour of a pair of values is obvious from a description of each argument. At the same time make a few edits to clarify the rest of the description. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/soc-utils.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 70c380c0ac7b..a3b6df2378b4 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -56,23 +56,24 @@ EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk); /** * snd_soc_tdm_params_to_bclk - calculate bclk from params and tdm slot info. * - * Calculate the bclk from the params sample rate and the tdm slot count and - * tdm slot width. Either or both of tdm_width and tdm_slots can be 0. + * Calculate the bclk from the params sample rate, the tdm slot count and the + * tdm slot width. Optionally round-up the slot count to a given multiple. + * Either or both of tdm_width and tdm_slots can be 0. * - * If tdm_width == 0 and tdm_slots > 0: the params_width will be used. - * If tdm_width > 0 and tdm_slots == 0: the params_channels will be used - * as the slot count. - * Both tdm_width and tdm_slots are 0: this is equivalent to calling - * snd_soc_params_to_bclk(). + * If tdm_width == 0: use params_width() as the slot width. + * If tdm_slots == 0: use params_channels() as the slot count. * - * If slot_multiple > 1 the slot count (or params_channels if tdm_slots == 0) - * will be rounded up to a multiple of this value. This is mainly useful for + * If slot_multiple > 1 the slot count (or params_channels() if tdm_slots == 0) + * will be rounded up to a multiple of slot_multiple. This is mainly useful for * I2S mode, which has a left and right phase so the number of slots is always * a multiple of 2. * + * If tdm_width == 0 && tdm_slots == 0 && slot_multiple < 2, this is equivalent + * to calling snd_soc_params_to_bclk(). + * * @params: Pointer to struct_pcm_hw_params. - * @tdm_width: Width in bits of the tdm slots. - * @tdm_slots: Number of tdm slots per frame. + * @tdm_width: Width in bits of the tdm slots. Must be >= 0. + * @tdm_slots: Number of tdm slots per frame. Must be >= 0. * @slot_multiple: If >1 roundup slot count to a multiple of this value. * * Return: bclk frequency in Hz, else a negative error code if params format -- 2.30.2

1 0

[PATCH] ASoC: Intel: sof_rt5682: Add support for jsl_rt5682_rt1019
by Sean Hong 15 Aug '22

15 Aug '22

This patch adds the driver data for rt5682s on SSP0 for ADL platform Signed-off-by: Sean Hong <sean.hong(a)quanta.corp-partner.google.com> --- sound/soc/intel/boards/sof_rt5682.c | 9 +++++++++ sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 13 +++++++++++++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 045965312245b..3a840f3a9f5d2 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -1100,6 +1100,15 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(1) | SOF_RT5682_NUM_HDMIDEV(4)), }, + { + .name = "jsl_rt5682_rt1019", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT1019_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); 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 b95c4b2cda947..139d2468f5f70 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -29,6 +29,11 @@ static const struct snd_soc_acpi_codecs rt1015p_spk = { .codecs = {"RTL1015"} }; +static struct snd_soc_acpi_codecs rt1019p_spk = { + .num_codecs = 1, + .codecs = {"RTL1019"} +} + static const struct snd_soc_acpi_codecs mx98360a_spk = { .num_codecs = 1, .codecs = {"MX98360A"} @@ -78,6 +83,14 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { .quirk_data = &mx98360a_spk, .sof_tplg_filename = "sof-jsl-rt5682-mx98360a.tplg", }, + { + .comp_ids = &rt5682_rt5682s_hp, + .drv_name = "jsl_rt5682_rt1019", + .sof_fw_filename = "sof-jsl.ri", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt1019p_spk, + .sof_tplg_filename = "sof-jsl-rt5682-rt1015.tplg", + }, { .id = "10134242", .drv_name = "jsl_cs4242_mx98360a", -- 2.25.1

4 3

[PATCH v3] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Matt Flax 15 Aug '22

15 Aug '22

Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- Notes: v3: * Renamed to include ti, * fixed license to include BSD-2-Clause * removed generic description * renamed the node in the example v2: * Changed the maintainer .../devicetree/bindings/sound/ti,src4xxx.yaml | 48 +++++++++++++++++++ 1 file changed, 48 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,src4xxx.yaml diff --git a/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml new file mode 100644 index 000000000000..807b9e35b9b9 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,src4xxx.yaml @@ -0,0 +1,48 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/src4xxx.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments SRC4392 Device Tree Bindings + +description: | + The SRC4392 is a digital audio codec that can be connected via + I2C or SPI. Currently, only I2C bus is supported. + +maintainers: + - Matt Flax <flatmax(a)flatmax.com> + +allOf: + - $ref: name-prefix.yaml# + +properties: + compatible: + const: ti,src4392 + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@70 { + #sound-dai-cells = <0>; + compatible = "ti,src4392"; + reg = <0x70>; + }; + }; +... -- 2.34.1

1 0

[PATCH AUTOSEL 4.9 4/8] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 31079ea5e484..d5d0e5e53920 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -457,4 +457,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index f2e8226c88fb..efe26b8ca57f 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -153,3 +154,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 5/9] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index c333ceb80d5f..720d63574bfb 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -75,7 +75,7 @@ struct snd_timer_user { unsigned int filter; struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1295,7 +1295,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1332,7 +1332,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1399,7 +1399,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1465,6 +1465,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (tu->timeri) snd_timer_close(tu->timeri); mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2017,7 +2018,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 4.14 4/9] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 4104a9d1001f..9d04e700b855 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -442,4 +442,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 0f818d593c9e..d100feba26b5 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -160,3 +161,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 09/14] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 4920ec4f4594..f0e8b98f346e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -75,7 +75,7 @@ struct snd_timer_user { unsigned int filter; struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1306,7 +1306,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1343,7 +1343,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1410,7 +1410,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1476,6 +1476,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (tu->timeri) snd_timer_close(tu->timeri); mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2027,7 +2028,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 4.19 08/14] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 36a5934cf4b1..b5a8cc4d02cc 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -444,4 +444,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 0f818d593c9e..d100feba26b5 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -25,6 +25,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -160,3 +161,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 11/16] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index d684aa4150aa..420cc07a7f88 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -61,7 +61,7 @@ struct snd_timer_user { unsigned int filter; struct timespec tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1317,7 +1317,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1354,7 +1354,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1421,7 +1421,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1487,6 +1487,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) if (tu->timeri) snd_timer_close(tu->timeri); mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2050,7 +2051,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.4 10/16] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index ee238f100f73..8a80121811d9 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -440,4 +440,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 3579dd7a161f..c3f3d94b5197 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 13/19] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 77d9fa10812d..41bd72ffd232 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -103,7 +103,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index 3b44378b9dec..732eb515d2f5 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -121,6 +121,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -175,7 +176,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -1941,7 +1942,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2015,7 +2016,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 12/19] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index 04cd8953605a..764d2b19344e 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.10 11/19] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 0462c577d7a3..85610ede9ea0 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -446,4 +446,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 3579dd7a161f..c3f3d94b5197 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 21/28] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..a1fc7e0a47d9 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -109,7 +109,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..f66fe4be30d3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -127,6 +127,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -181,7 +182,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -2002,7 +2003,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2170,7 +2171,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 20/28] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b3214baa8919..e08a37c23add 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 19/28] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..39cee40ac22e 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -501,4 +501,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 50e4aaa6270d..d32a19976a2b 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.15 08/28] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl()
by Sasha Levin 14 Aug '22

14 Aug '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit ef30911d3c39fd57884c348c29b9cbff88def155 ] Before, ssiu.c didn't care SSI5-8, thus, commit b1384d4c95088d0 ("ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()") cares it for status clear. But we should care it for error irq handling, too. This patch cares it. Reported-by: Nguyen Bao Nguyen <nguyen.nguyen.yj(a)renesas.com> Reported-by: Nishiyama Kunihiko <kunihiko.nishiyama.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/871quocio1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..d7f4646ee029 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -67,6 +67,8 @@ static void rsnd_ssiu_busif_err_irq_ctrl(struct rsnd_mod *mod, int enable) shift = 1; offset = 1; break; + default: + return; } for (i = 0; i < 4; i++) { -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 32/39] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..a1fc7e0a47d9 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -109,7 +109,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..f66fe4be30d3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -127,6 +127,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -181,7 +182,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -2002,7 +2003,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2170,7 +2171,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 31/39] ALSA: pcm: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 96b097091c66df4f6fbf5cbff21df6cc02a2f055 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from timer API. Note that it's merely a workaround. Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Reported-by: syzbot+669c9abf11a6a011dd09(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/pcm.h | 2 +- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 4 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..6987110843f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -399,7 +399,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ - struct fasync_struct *fasync; + struct snd_fasync *fasync; bool stop_operating; /* sync_stop will be called */ struct mutex buffer_mutex; /* protect for buffer changes */ atomic_t buffer_accessing; /* >0: in r/w operation, <0: blocked */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 977d54320a5c..c917ac84a7e5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1005,6 +1005,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->runtime = NULL; } mutex_destroy(&runtime->buffer_mutex); + snd_fasync_free(runtime->fasync); kfree(runtime); put_pid(substream->pid); substream->pid = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1fc7c50ffa62..40751e5aff09 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1822,7 +1822,7 @@ void snd_pcm_period_elapsed_under_stream_lock(struct snd_pcm_substream *substrea snd_timer_interrupt(substream->timer, 1); #endif _end: - kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_kill_fasync(runtime->fasync, SIGIO, POLL_IN); } EXPORT_SYMBOL(snd_pcm_period_elapsed_under_stream_lock); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..16fcf57c6f03 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3945,7 +3945,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) runtime = substream->runtime; if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; - return fasync_helper(fd, file, on, &runtime->fasync); + return snd_fasync_helper(fd, file, on, &runtime->fasync); } /* -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 30/39] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b3214baa8919..e08a37c23add 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 29/39] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..39cee40ac22e 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -501,4 +501,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 50e4aaa6270d..d32a19976a2b 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 26/39] ASoC: codecs: va-macro: use fsgen as clock
by Sasha Levin 14 Aug '22

14 Aug '22

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> [ Upstream commit 30097967e0566cac817273ef76add100f6b0f463 ] VA Macro fsgen clock is supplied to other LPASS Macros using proper clock apis, however the internal user uses the registers directly without clk apis. This approch has race condition where in external users of the clock might cut the clock while VA macro is actively using this. Moving the internal usage to clk apis would provide a proper refcounting and avoid such race conditions. This issue was noticed while headset was pulled out while recording is in progress and shifting record patch to DMIC. Reported-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Tested-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Link: https://lore.kernel.org/r/20220727124749.4604-1-srinivas.kandagatla@linaro.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index d18b56e60433..1ea10dc70748 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -199,6 +199,7 @@ struct va_macro { struct clk *mclk; struct clk *macro; struct clk *dcodec; + struct clk *fsgen; struct clk_hw hw; struct lpass_macro *pds; @@ -467,9 +468,9 @@ static int va_macro_mclk_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - return va_macro_mclk_enable(va, true); + return clk_prepare_enable(va->fsgen); case SND_SOC_DAPM_POST_PMD: - return va_macro_mclk_enable(va, false); + clk_disable_unprepare(va->fsgen); } return 0; @@ -1473,6 +1474,12 @@ static int va_macro_probe(struct platform_device *pdev) if (ret) goto err_clkout; + va->fsgen = clk_hw_get_clk(&va->hw, "fsgen"); + if (IS_ERR(va->fsgen)) { + ret = PTR_ERR(va->fsgen); + goto err_clkout; + } + ret = devm_snd_soc_register_component(dev, &va_macro_component_drv, va_macro_dais, ARRAY_SIZE(va_macro_dais)); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 20/39] ASoC: Intel: sof_nau8825: Move quirk check to the front in late probe
by Sasha Levin 14 Aug '22

14 Aug '22

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 5b56db90bbaf9d8581e5e6268727d8ad706555e4 ] The sof_rt5682_quirk check was placed in the middle of hdmi handling code, move it to the front to be consistent with sof_rt5682.c/sof_card_late_probe(). Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-11-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_nau8825.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 33de043b66c6..b6826d6572b8 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -177,11 +177,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct sof_hdmi_pcm *pcm; int err; - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); - if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { /* Disable Left and Right Spk pin after boot */ snd_soc_dapm_disable_pin(dapm, "Left Spk"); @@ -191,6 +186,11 @@ static int sof_card_late_probe(struct snd_soc_card *card) return err; } + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); } -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 19/39] ASoC: Intel: sof_es8336: ignore GpioInt when looking for speaker/headset GPIO lines
by Sasha Levin 14 Aug '22

14 Aug '22

From: Andrey Turkin <andrey.turkin(a)gmail.com> [ Upstream commit 751e77011f7a43a204bf2a5d02fbf5f8219bc531 ] This fixes speaker GPIO detection on machines those ACPI tables list their jack detection GpioInt before output GpioIo. GpioInt entry can never be the speaker/headphone amplifier control so it makes sense to only look for GpioIo entries when looking for them. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Andrey Turkin <andrey.turkin(a)gmail.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-5-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_es8336.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 81e12f03ec97..dade1749c444 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -57,23 +57,23 @@ static const struct acpi_gpio_params enable_gpio0 = { 0, 0, true }; static const struct acpi_gpio_params enable_gpio1 = { 1, 0, true }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio0[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio1[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, }; static const struct acpi_gpio_mapping acpi_enable_both_gpios[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, - { "headphone-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, - { "headphone-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 18/39] ASoC: Intel: sof_es8336: Fix GPIO quirks set via module option
by Sasha Levin 14 Aug '22

14 Aug '22

From: Andrey Turkin <andrey.turkin(a)gmail.com> [ Upstream commit 5e60f1cfb830342304200437121f440b72b54f54 ] The two GPIO quirk bits only affected actual GPIO selection when set by the quirks table. They were reported as being in effect when set via module options but actually did nothing. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Andrey Turkin <andrey.turkin(a)gmail.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-4-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_es8336.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 9d617831dd20..81e12f03ec97 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -77,8 +77,6 @@ static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { { } }; -static const struct acpi_gpio_mapping *gpio_mapping = acpi_speakers_enable_gpio0; - static void log_quirks(struct device *dev) { dev_info(dev, "quirk mask %#lx\n", quirk); @@ -272,15 +270,6 @@ static int sof_es8336_quirk_cb(const struct dmi_system_id *id) { quirk = (unsigned long)id->driver_data; - if (quirk & SOF_ES8336_HEADPHONE_GPIO) { - if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) - gpio_mapping = acpi_enable_both_gpios; - else - gpio_mapping = acpi_enable_both_gpios_rev_order; - } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { - gpio_mapping = acpi_speakers_enable_gpio1; - } - return 1; } @@ -529,6 +518,7 @@ static int sof_es8336_probe(struct platform_device *pdev) struct acpi_device *adev; struct snd_soc_dai_link *dai_links; struct device *codec_dev; + const struct acpi_gpio_mapping *gpio_mapping; unsigned int cnt = 0; int dmic_be_num = 0; int hdmi_num = 3; @@ -635,6 +625,17 @@ static int sof_es8336_probe(struct platform_device *pdev) } /* get speaker enable GPIO */ + if (quirk & SOF_ES8336_HEADPHONE_GPIO) { + if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + gpio_mapping = acpi_enable_both_gpios; + else + gpio_mapping = acpi_enable_both_gpios_rev_order; + } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { + gpio_mapping = acpi_speakers_enable_gpio1; + } else { + gpio_mapping = acpi_speakers_enable_gpio0; + } + ret = devm_acpi_dev_add_driver_gpios(codec_dev, gpio_mapping); if (ret) dev_warn(codec_dev, "unable to add GPIO mapping table\n"); -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 17/39] ASoC: SOF: Intel: hda: add sanity check on SSP index reported by NHLT
by Sasha Levin 14 Aug '22

14 Aug '22

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit e51699505042fb365df3a0ce68b850ccd9ad0108 ] We should have a limited trust in the BIOS and verify that the SSP index reported in NHLT is valid for each platform. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725195343.145603-2-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 9c97c80a7f48..b688caa34774 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1375,6 +1375,7 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) if (mach->tplg_quirk_mask & SND_SOC_ACPI_TPLG_INTEL_SSP_NUMBER && mach->mach_params.i2s_link_mask) { + const struct sof_intel_dsp_desc *chip = get_chip_info(sdev->pdata); int ssp_num; if (hweight_long(mach->mach_params.i2s_link_mask) > 1 && @@ -1384,6 +1385,12 @@ struct snd_soc_acpi_mach *hda_machine_select(struct snd_sof_dev *sdev) /* fls returns 1-based results, SSPs indices are 0-based */ ssp_num = fls(mach->mach_params.i2s_link_mask) - 1; + if (ssp_num >= chip->ssp_count) { + dev_err(sdev->dev, "Invalid SSP %d, max on this platform is %d\n", + ssp_num, chip->ssp_count); + return NULL; + } + tplg_filename = devm_kasprintf(sdev->dev, GFP_KERNEL, "%s%s%d", sof_pdata->tplg_filename, -- 2.35.1

1 0

[PATCH AUTOSEL 5.18 16/39] ALSA: hda/realtek: Enable speaker and mute LEDs for HP laptops
by Sasha Levin 14 Aug '22

14 Aug '22

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit c578d5da10dc429c6676ab09f3fec0b79b31633a ] Two more HP laptops that use cs35l41 AMP for speaker and GPIO for mute LEDs. So use the existing quirk to enable them accordingly. [ Sort the entries at the SSID order by tiwai ] Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Reviewed-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220719142015.244426-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 b7f1f2fb60cb..ebd970961e00 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9166,6 +9166,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.35.1

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[PATCH AUTOSEL 5.18 10/39] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl()
by Sasha Levin 14 Aug '22

14 Aug '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit ef30911d3c39fd57884c348c29b9cbff88def155 ] Before, ssiu.c didn't care SSI5-8, thus, commit b1384d4c95088d0 ("ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()") cares it for status clear. But we should care it for error irq handling, too. This patch cares it. Reported-by: Nguyen Bao Nguyen <nguyen.nguyen.yj(a)renesas.com> Reported-by: Nishiyama Kunihiko <kunihiko.nishiyama.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/871quocio1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..d7f4646ee029 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -67,6 +67,8 @@ static void rsnd_ssiu_busif_err_irq_ctrl(struct rsnd_mod *mod, int enable) shift = 1; offset = 1; break; + default: + return; } for (i = 0; i < 4; i++) { -- 2.35.1

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[PATCH AUTOSEL 5.18 08/39] ALSA: hda: Fix page fault in snd_hda_codec_shutdown()
by Sasha Levin 14 Aug '22

14 Aug '22

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 980b3a8790b402e959a6d773b38b771019682be1 ] If early probe of HDAudio bus driver fails e.g.: due to missing firmware file, snd_hda_codec_shutdown() ends in manipulating uninitialized codec->pcm_list_head causing page fault. Iinitialization of HDAudio codec in ASoC is split in two: - snd_hda_codec_device_init() - snd_hda_codec_device_new() snd_hda_codec_device_init() is called during probe_codecs() by HDAudio bus driver while snd_hda_codec_device_new() is called by codec-component's ->probe(). The second call will not happen until all components required by related sound card are present within the ASoC framework. With firmware failing to load during the PCI's deferred initialization i.e.: probe_work(), no platform components are ever registered. HDAudio codec enumeration is done at that point though, so the codec components became registered to ASoC framework, calling snd_hda_codec_device_init() in the process. Now, during platform reboot snd_hda_codec_shutdown() is called for every codec found on the HDAudio bus causing oops if any of them has not completed both of their initialization steps. Relocating field initialization fixes the issue. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220706120230.427296-7-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_codec.c | 41 +++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5cbac315dbe1..527616b39043 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,8 +931,28 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -980,29 +1000,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; - codec->card = card; codec->addr = codec_addr; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- 2.35.1

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[PATCH AUTOSEL 5.19 39/48] ALSA: control: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 4a971e84a7ae10a38d875cd2d4e487c8d1682ca3 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from the control API. Note that it's merely a workaround. Another note: although we haven't received reports about the deadlock with the control API, the deadlock is still potentially possible, and it's better to align the behavior with other core APIs (PCM and timer); so let's move altogether. Link: https://lore.kernel.org/r/20220728125945.29533-5-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/control.h | 2 +- sound/core/control.c | 7 ++++--- 2 files changed, 5 insertions(+), 4 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 985c51a8fb74..a1fc7e0a47d9 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -109,7 +109,7 @@ struct snd_ctl_file { int preferred_subdevice[SND_CTL_SUBDEV_ITEMS]; wait_queue_head_t change_sleep; spinlock_t read_lock; - struct fasync_struct *fasync; + struct snd_fasync *fasync; int subscribed; /* read interface is activated */ struct list_head events; /* waiting events for read */ }; diff --git a/sound/core/control.c b/sound/core/control.c index a25c0d64d104..f66fe4be30d3 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -127,6 +127,7 @@ static int snd_ctl_release(struct inode *inode, struct file *file) if (control->vd[idx].owner == ctl) control->vd[idx].owner = NULL; up_write(&card->controls_rwsem); + snd_fasync_free(ctl->fasync); snd_ctl_empty_read_queue(ctl); put_pid(ctl->pid); kfree(ctl); @@ -181,7 +182,7 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask, _found: wake_up(&ctl->change_sleep); spin_unlock(&ctl->read_lock); - kill_fasync(&ctl->fasync, SIGIO, POLL_IN); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_IN); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); } @@ -2002,7 +2003,7 @@ static int snd_ctl_fasync(int fd, struct file * file, int on) struct snd_ctl_file *ctl; ctl = file->private_data; - return fasync_helper(fd, file, on, &ctl->fasync); + return snd_fasync_helper(fd, file, on, &ctl->fasync); } /* return the preferred subdevice number if already assigned; @@ -2170,7 +2171,7 @@ static int snd_ctl_dev_disconnect(struct snd_device *device) read_lock_irqsave(&card->ctl_files_rwlock, flags); list_for_each_entry(ctl, &card->ctl_files, list) { wake_up(&ctl->change_sleep); - kill_fasync(&ctl->fasync, SIGIO, POLL_ERR); + snd_kill_fasync(ctl->fasync, SIGIO, POLL_ERR); } read_unlock_irqrestore(&card->ctl_files_rwlock, flags); -- 2.35.1

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[PATCH AUTOSEL 5.19 38/48] ALSA: pcm: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 96b097091c66df4f6fbf5cbff21df6cc02a2f055 ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from timer API. Note that it's merely a workaround. Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Reported-by: syzbot+669c9abf11a6a011dd09(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/pcm.h | 2 +- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- 4 files changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 6b99310b5b88..6987110843f0 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -399,7 +399,7 @@ struct snd_pcm_runtime { snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ - struct fasync_struct *fasync; + struct snd_fasync *fasync; bool stop_operating; /* sync_stop will be called */ struct mutex buffer_mutex; /* protect for buffer changes */ atomic_t buffer_accessing; /* >0: in r/w operation, <0: blocked */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 977d54320a5c..c917ac84a7e5 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1005,6 +1005,7 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) substream->runtime = NULL; } mutex_destroy(&runtime->buffer_mutex); + snd_fasync_free(runtime->fasync); kfree(runtime); put_pid(substream->pid); substream->pid = NULL; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1fc7c50ffa62..40751e5aff09 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1822,7 +1822,7 @@ void snd_pcm_period_elapsed_under_stream_lock(struct snd_pcm_substream *substrea snd_timer_interrupt(substream->timer, 1); #endif _end: - kill_fasync(&runtime->fasync, SIGIO, POLL_IN); + snd_kill_fasync(runtime->fasync, SIGIO, POLL_IN); } EXPORT_SYMBOL(snd_pcm_period_elapsed_under_stream_lock); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 4adaee62ef33..16fcf57c6f03 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3945,7 +3945,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) runtime = substream->runtime; if (runtime->status->state == SNDRV_PCM_STATE_DISCONNECTED) return -EBADFD; - return fasync_helper(fd, file, on, &runtime->fasync); + return snd_fasync_helper(fd, file, on, &runtime->fasync); } /* -- 2.35.1

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[PATCH AUTOSEL 5.19 37/48] ALSA: timer: Use deferred fasync helper
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit 95cc637c1afd83fb7dd3d7c8a53710488f4caf9c ] For avoiding the potential deadlock via kill_fasync() call, use the new fasync helpers to defer the invocation from PCI API. Note that it's merely a workaround. Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com Reported-by: syzbot+49b10793b867871ee26f(a)syzkaller.appspotmail.com Reported-by: syzbot+8285e973a41b5aa68902(a)syzkaller.appspotmail.com Link: https://lore.kernel.org/r/20220728125945.29533-3-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/core/timer.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/core/timer.c b/sound/core/timer.c index b3214baa8919..e08a37c23add 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -83,7 +83,7 @@ struct snd_timer_user { unsigned int filter; struct timespec64 tstamp; /* trigger tstamp */ wait_queue_head_t qchange_sleep; - struct fasync_struct *fasync; + struct snd_fasync *fasync; struct mutex ioctl_lock; }; @@ -1345,7 +1345,7 @@ static void snd_timer_user_interrupt(struct snd_timer_instance *timeri, } __wake: spin_unlock(&tu->qlock); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1383,7 +1383,7 @@ static void snd_timer_user_ccallback(struct snd_timer_instance *timeri, spin_lock_irqsave(&tu->qlock, flags); snd_timer_user_append_to_tqueue(tu, &r1); spin_unlock_irqrestore(&tu->qlock, flags); - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1453,7 +1453,7 @@ static void snd_timer_user_tinterrupt(struct snd_timer_instance *timeri, spin_unlock(&tu->qlock); if (append == 0) return; - kill_fasync(&tu->fasync, SIGIO, POLL_IN); + snd_kill_fasync(tu->fasync, SIGIO, POLL_IN); wake_up(&tu->qchange_sleep); } @@ -1521,6 +1521,7 @@ static int snd_timer_user_release(struct inode *inode, struct file *file) snd_timer_instance_free(tu->timeri); } mutex_unlock(&tu->ioctl_lock); + snd_fasync_free(tu->fasync); kfree(tu->queue); kfree(tu->tqueue); kfree(tu); @@ -2135,7 +2136,7 @@ static int snd_timer_user_fasync(int fd, struct file * file, int on) struct snd_timer_user *tu; tu = file->private_data; - return fasync_helper(fd, file, on, &tu->fasync); + return snd_fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, -- 2.35.1

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[PATCH AUTOSEL 5.19 36/48] ALSA: core: Add async signal helpers
by Sasha Levin 14 Aug '22

14 Aug '22

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit ef34a0ae7a2654bc9e58675e36898217fb2799d8 ] Currently the call of kill_fasync() from an interrupt handler might lead to potential spin deadlocks, as spotted by syzkaller. Unfortunately, it's not so trivial to fix this lock chain as it's involved with the tasklist_lock that is touched in allover places. As a temporary workaround, this patch provides the way to defer the async signal notification in a work. The new helper functions, snd_fasync_helper() and snd_kill_faync() are replacements for fasync_helper() and kill_fasync(), respectively. In addition, snd_fasync_free() needs to be called at the destructor of the relevant file object. Link: https://lore.kernel.org/r/20220728125945.29533-2-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/core.h | 8 ++++ sound/core/misc.c | 94 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 102 insertions(+) diff --git a/include/sound/core.h b/include/sound/core.h index 6d4cc49584c6..39cee40ac22e 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -501,4 +501,12 @@ snd_pci_quirk_lookup_id(u16 vendor, u16 device, } #endif +/* async signal helpers */ +struct snd_fasync; + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp); +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll); +void snd_fasync_free(struct snd_fasync *fasync); + #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 50e4aaa6270d..d32a19976a2b 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -10,6 +10,7 @@ #include <linux/time.h> #include <linux/slab.h> #include <linux/ioport.h> +#include <linux/fs.h> #include <sound/core.h> #ifdef CONFIG_SND_DEBUG @@ -145,3 +146,96 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) } EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif + +/* + * Deferred async signal helpers + * + * Below are a few helper functions to wrap the async signal handling + * in the deferred work. The main purpose is to avoid the messy deadlock + * around tasklist_lock and co at the kill_fasync() invocation. + * fasync_helper() and kill_fasync() are replaced with snd_fasync_helper() + * and snd_kill_fasync(), respectively. In addition, snd_fasync_free() has + * to be called at releasing the relevant file object. + */ +struct snd_fasync { + struct fasync_struct *fasync; + int signal; + int poll; + int on; + struct list_head list; +}; + +static DEFINE_SPINLOCK(snd_fasync_lock); +static LIST_HEAD(snd_fasync_list); + +static void snd_fasync_work_fn(struct work_struct *work) +{ + struct snd_fasync *fasync; + + spin_lock_irq(&snd_fasync_lock); + while (!list_empty(&snd_fasync_list)) { + fasync = list_first_entry(&snd_fasync_list, struct snd_fasync, list); + list_del_init(&fasync->list); + spin_unlock_irq(&snd_fasync_lock); + if (fasync->on) + kill_fasync(&fasync->fasync, fasync->signal, fasync->poll); + spin_lock_irq(&snd_fasync_lock); + } + spin_unlock_irq(&snd_fasync_lock); +} + +static DECLARE_WORK(snd_fasync_work, snd_fasync_work_fn); + +int snd_fasync_helper(int fd, struct file *file, int on, + struct snd_fasync **fasyncp) +{ + struct snd_fasync *fasync = NULL; + + if (on) { + fasync = kzalloc(sizeof(*fasync), GFP_KERNEL); + if (!fasync) + return -ENOMEM; + INIT_LIST_HEAD(&fasync->list); + } + + spin_lock_irq(&snd_fasync_lock); + if (*fasyncp) { + kfree(fasync); + fasync = *fasyncp; + } else { + if (!fasync) { + spin_unlock_irq(&snd_fasync_lock); + return 0; + } + *fasyncp = fasync; + } + fasync->on = on; + spin_unlock_irq(&snd_fasync_lock); + return fasync_helper(fd, file, on, &fasync->fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_helper); + +void snd_kill_fasync(struct snd_fasync *fasync, int signal, int poll) +{ + unsigned long flags; + + if (!fasync || !fasync->on) + return; + spin_lock_irqsave(&snd_fasync_lock, flags); + fasync->signal = signal; + fasync->poll = poll; + list_move(&fasync->list, &snd_fasync_list); + schedule_work(&snd_fasync_work); + spin_unlock_irqrestore(&snd_fasync_lock, flags); +} +EXPORT_SYMBOL_GPL(snd_kill_fasync); + +void snd_fasync_free(struct snd_fasync *fasync) +{ + if (!fasync) + return; + fasync->on = 0; + flush_work(&snd_fasync_work); + kfree(fasync); +} +EXPORT_SYMBOL_GPL(snd_fasync_free); -- 2.35.1

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[PATCH AUTOSEL 5.19 27/48] ASoC: Intel: sof_nau8825: Move quirk check to the front in late probe
by Sasha Levin 14 Aug '22

14 Aug '22

From: Yong Zhi <yong.zhi(a)intel.com> [ Upstream commit 5b56db90bbaf9d8581e5e6268727d8ad706555e4 ] The sof_rt5682_quirk check was placed in the middle of hdmi handling code, move it to the front to be consistent with sof_rt5682.c/sof_card_late_probe(). Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-11-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_nau8825.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index 97dcd204a246..9b3a2ff4d9cd 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -177,11 +177,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) struct sof_hdmi_pcm *pcm; int err; - if (list_empty(&ctx->hdmi_pcm_list)) - return -EINVAL; - - pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); - if (sof_nau8825_quirk & SOF_MAX98373_SPEAKER_AMP_PRESENT) { /* Disable Left and Right Spk pin after boot */ snd_soc_dapm_disable_pin(dapm, "Left Spk"); @@ -191,6 +186,11 @@ static int sof_card_late_probe(struct snd_soc_card *card) return err; } + if (list_empty(&ctx->hdmi_pcm_list)) + return -EINVAL; + + pcm = list_first_entry(&ctx->hdmi_pcm_list, struct sof_hdmi_pcm, head); + return hda_dsp_hdmi_build_controls(card, pcm->codec_dai->component); } -- 2.35.1

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[PATCH AUTOSEL 5.19 26/48] ASoC: Intel: sof_es8336: ignore GpioInt when looking for speaker/headset GPIO lines
by Sasha Levin 14 Aug '22

14 Aug '22

From: Andrey Turkin <andrey.turkin(a)gmail.com> [ Upstream commit 751e77011f7a43a204bf2a5d02fbf5f8219bc531 ] This fixes speaker GPIO detection on machines those ACPI tables list their jack detection GpioInt before output GpioIo. GpioInt entry can never be the speaker/headphone amplifier control so it makes sense to only look for GpioIo entries when looking for them. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Andrey Turkin <andrey.turkin(a)gmail.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-5-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_es8336.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 4d0c361fc277..d70d8255b8c7 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -57,23 +57,23 @@ static const struct acpi_gpio_params enable_gpio0 = { 0, 0, true }; static const struct acpi_gpio_params enable_gpio1 = { 1, 0, true }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio0[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio1[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, }; static const struct acpi_gpio_mapping acpi_enable_both_gpios[] = { - { "speakers-enable-gpios", &enable_gpio0, 1 }, - { "headphone-enable-gpios", &enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { - { "speakers-enable-gpios", &enable_gpio1, 1 }, - { "headphone-enable-gpios", &enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, + { "headphone-enable-gpios", &enable_gpio0, 1, ACPI_GPIO_QUIRK_ONLY_GPIOIO }, { } }; -- 2.35.1

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[PATCH AUTOSEL 5.19 25/48] ASoC: Intel: sof_es8336: Fix GPIO quirks set via module option
by Sasha Levin 14 Aug '22

14 Aug '22

From: Andrey Turkin <andrey.turkin(a)gmail.com> [ Upstream commit 5e60f1cfb830342304200437121f440b72b54f54 ] The two GPIO quirk bits only affected actual GPIO selection when set by the quirks table. They were reported as being in effect when set via module options but actually did nothing. Reviewed-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Signed-off-by: Andrey Turkin <andrey.turkin(a)gmail.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220725194909.145418-4-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_es8336.c | 23 ++++++++++++----------- 1 file changed, 12 insertions(+), 11 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 23d03e0f7759..4d0c361fc277 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -77,8 +77,6 @@ static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { { } }; -static const struct acpi_gpio_mapping *gpio_mapping = acpi_speakers_enable_gpio0; - static void log_quirks(struct device *dev) { dev_info(dev, "quirk mask %#lx\n", quirk); @@ -272,15 +270,6 @@ static int sof_es8336_quirk_cb(const struct dmi_system_id *id) { quirk = (unsigned long)id->driver_data; - if (quirk & SOF_ES8336_HEADPHONE_GPIO) { - if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) - gpio_mapping = acpi_enable_both_gpios; - else - gpio_mapping = acpi_enable_both_gpios_rev_order; - } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { - gpio_mapping = acpi_speakers_enable_gpio1; - } - return 1; } @@ -529,6 +518,7 @@ static int sof_es8336_probe(struct platform_device *pdev) struct acpi_device *adev; struct snd_soc_dai_link *dai_links; struct device *codec_dev; + const struct acpi_gpio_mapping *gpio_mapping; unsigned int cnt = 0; int dmic_be_num = 0; int hdmi_num = 3; @@ -635,6 +625,17 @@ static int sof_es8336_probe(struct platform_device *pdev) } /* get speaker enable GPIO */ + if (quirk & SOF_ES8336_HEADPHONE_GPIO) { + if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + gpio_mapping = acpi_enable_both_gpios; + else + gpio_mapping = acpi_enable_both_gpios_rev_order; + } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { + gpio_mapping = acpi_speakers_enable_gpio1; + } else { + gpio_mapping = acpi_speakers_enable_gpio0; + } + ret = devm_acpi_dev_add_driver_gpios(codec_dev, gpio_mapping); if (ret) dev_warn(codec_dev, "unable to add GPIO mapping table\n"); -- 2.35.1

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[PATCH AUTOSEL 5.19 23/48] ALSA: hda/realtek: Enable speaker and mute LEDs for HP laptops
by Sasha Levin 14 Aug '22

14 Aug '22

From: Kai-Heng Feng <kai.heng.feng(a)canonical.com> [ Upstream commit c578d5da10dc429c6676ab09f3fec0b79b31633a ] Two more HP laptops that use cs35l41 AMP for speaker and GPIO for mute LEDs. So use the existing quirk to enable them accordingly. [ Sort the entries at the SSID order by tiwai ] Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> Reviewed-by: Lucas Tanure <tanureal(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/20220719142015.244426-1-kai.heng.feng@canonical.c… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- 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 2f55bc43bfa9..8da712c5d743 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9114,6 +9114,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa8, "HP EliteBook 640 G9 (MB 8AA6)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aab, "HP EliteBook 650 G9 (MB 8AA9)", ALC236_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad1, "HP EliteBook 840 14 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8ad2, "HP EliteBook 860 16 inch G9 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x1043, 0x103e, "ASUS X540SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), -- 2.35.1

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[PATCH AUTOSEL 5.19 22/48] ASoC: Intel: avs: Use lookup table to create modules
by Sasha Levin 14 Aug '22

14 Aug '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 1e744351bcb9c4cee81300de5a6097100d835386 ] As reported by Nathan, when building avs driver using clang with: CONFIG_COMPILE_TEST=y CONFIG_FORTIFY_SOURCE=y CONFIG_KASAN=y CONFIG_PCI=y CONFIG_SOUND=y CONFIG_SND=y CONFIG_SND_SOC=y CONFIG_SND_SOC_INTEL_AVS=y there are reports of too big stack use, like: sound/soc/intel/avs/path.c:815:18: error: stack frame size (2176) exceeds limit (2048) in 'avs_path_create' [-Werror,-Wframe-larger-than] struct avs_path *avs_path_create(struct avs_dev *adev, u32 dma_id, ^ 1 error generated. This is apparently caused by inlining many calls to guid_equal which inlines fortified memcpy, using 2 size_t variables. Instead of hardcoding many calls to guid_equal, use lookup table with one call, this improves stack usage. Link: https://lore.kernel.org/alsa-devel/YtlzY9aYdbS4Y3+l@dev-arch.thelio-3990X/T/ Link: https://github.com/ClangBuiltLinux/linux/issues/1642 Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Reported-by: Nathan Chancellor <nathan(a)kernel.org> Build-tested-by: Nathan Chancellor <nathan(a)kernel.org> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220722111959.2588597-1-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/path.c | 54 ++++++++++++++++++++------------------ 1 file changed, 29 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/avs/path.c b/sound/soc/intel/avs/path.c index 3d46dd5e5bc4..ce157a8d6552 100644 --- a/sound/soc/intel/avs/path.c +++ b/sound/soc/intel/avs/path.c @@ -449,35 +449,39 @@ static int avs_modext_create(struct avs_dev *adev, struct avs_path_module *mod) return ret; } +static int avs_probe_create(struct avs_dev *adev, struct avs_path_module *mod) +{ + dev_err(adev->dev, "Probe module can't be instantiated by topology"); + return -EINVAL; +} + +struct avs_module_create { + guid_t *guid; + int (*create)(struct avs_dev *adev, struct avs_path_module *mod); +}; + +static struct avs_module_create avs_module_create[] = { + { &AVS_MIXIN_MOD_UUID, avs_modbase_create }, + { &AVS_MIXOUT_MOD_UUID, avs_modbase_create }, + { &AVS_KPBUFF_MOD_UUID, avs_modbase_create }, + { &AVS_COPIER_MOD_UUID, avs_copier_create }, + { &AVS_MICSEL_MOD_UUID, avs_micsel_create }, + { &AVS_MUX_MOD_UUID, avs_mux_create }, + { &AVS_UPDWMIX_MOD_UUID, avs_updown_mix_create }, + { &AVS_SRCINTC_MOD_UUID, avs_src_create }, + { &AVS_AEC_MOD_UUID, avs_aec_create }, + { &AVS_ASRC_MOD_UUID, avs_asrc_create }, + { &AVS_INTELWOV_MOD_UUID, avs_wov_create }, + { &AVS_PROBE_MOD_UUID, avs_probe_create }, +}; + static int avs_path_module_type_create(struct avs_dev *adev, struct avs_path_module *mod) { const guid_t *type = &mod->template->cfg_ext->type; - if (guid_equal(type, &AVS_MIXIN_MOD_UUID) || - guid_equal(type, &AVS_MIXOUT_MOD_UUID) || - guid_equal(type, &AVS_KPBUFF_MOD_UUID)) - return avs_modbase_create(adev, mod); - if (guid_equal(type, &AVS_COPIER_MOD_UUID)) - return avs_copier_create(adev, mod); - if (guid_equal(type, &AVS_MICSEL_MOD_UUID)) - return avs_micsel_create(adev, mod); - if (guid_equal(type, &AVS_MUX_MOD_UUID)) - return avs_mux_create(adev, mod); - if (guid_equal(type, &AVS_UPDWMIX_MOD_UUID)) - return avs_updown_mix_create(adev, mod); - if (guid_equal(type, &AVS_SRCINTC_MOD_UUID)) - return avs_src_create(adev, mod); - if (guid_equal(type, &AVS_AEC_MOD_UUID)) - return avs_aec_create(adev, mod); - if (guid_equal(type, &AVS_ASRC_MOD_UUID)) - return avs_asrc_create(adev, mod); - if (guid_equal(type, &AVS_INTELWOV_MOD_UUID)) - return avs_wov_create(adev, mod); - - if (guid_equal(type, &AVS_PROBE_MOD_UUID)) { - dev_err(adev->dev, "Probe module can't be instantiated by topology"); - return -EINVAL; - } + for (int i = 0; i < ARRAY_SIZE(avs_module_create); i++) + if (guid_equal(type, avs_module_create[i].guid)) + return avs_module_create[i].create(adev, mod); return avs_modext_create(adev, mod); } -- 2.35.1

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[PATCH AUTOSEL 5.19 19/48] ASoC: nau8821: Don't unconditionally free interrupt
by Sasha Levin 14 Aug '22

14 Aug '22

From: Mark Brown <broonie(a)kernel.org> [ Upstream commit 2d86cef353b8f3d20b16f8c5615742fd6938c801 ] The remove() operation unconditionally frees the interrupt for the device but we may not actually have an interrupt so there might be nothing to free. Since the interrupt is requested after all other resources we don't need the explicit free anyway, unwinding is guaranteed to be safe, so just delete the remove() function and let devm take care of things. Reported-by: Zheyu Ma <zheyuma97(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Tested-by: Zheyu Ma <zheyuma97(a)gmail.com> Link: https://lore.kernel.org/r/20220718140405.57233-1-broonie@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/nau8821.c | 10 ---------- 1 file changed, 10 deletions(-) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index ce4e7f46bb06..e078d2ffb3f6 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -1665,15 +1665,6 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) 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 }, { } @@ -1703,7 +1694,6 @@ static struct i2c_driver nau8821_driver = { .acpi_match_table = ACPI_PTR(nau8821_acpi_match), }, .probe_new = nau8821_i2c_probe, - .remove = nau8821_i2c_remove, .id_table = nau8821_i2c_ids, }; module_i2c_driver(nau8821_driver); -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 15/48] ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl()
by Sasha Levin 14 Aug '22

14 Aug '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit ef30911d3c39fd57884c348c29b9cbff88def155 ] Before, ssiu.c didn't care SSI5-8, thus, commit b1384d4c95088d0 ("ASoC: rsnd: care default case on rsnd_ssiu_busif_err_status_clear()") cares it for status clear. But we should care it for error irq handling, too. This patch cares it. Reported-by: Nguyen Bao Nguyen <nguyen.nguyen.yj(a)renesas.com> Reported-by: Nishiyama Kunihiko <kunihiko.nishiyama.dn(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/871quocio1.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/ssiu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index 4b8a63e336c7..d7f4646ee029 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -67,6 +67,8 @@ static void rsnd_ssiu_busif_err_irq_ctrl(struct rsnd_mod *mod, int enable) shift = 1; offset = 1; break; + default: + return; } for (i = 0; i < 4; i++) { -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 14/48] ASoC: SOF: sof-client-probes: Only load the driver if IPC3 is used
by Sasha Levin 14 Aug '22

14 Aug '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit 9b93eda355089b36482f7a2f134bdd24be70f907 ] The current implementation of probes only supports IPC3 and should not be loaded for other IPC implementation. 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> Link: https://lore.kernel.org/r/20220712131022.1124-1-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/sof-client-probes.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index 34e6bd356e71..60e4250fac87 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -693,6 +693,10 @@ static int sof_probes_client_probe(struct auxiliary_device *auxdev, if (!sof_probes_enabled) return -ENXIO; + /* only ipc3 is supported */ + if (sof_client_get_ipc_type(cdev) != SOF_IPC) + return -ENXIO; + if (!dev->platform_data) { dev_err(dev, "missing platform data\n"); return -ENODEV; -- 2.35.1

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[PATCH AUTOSEL 5.19 13/48] ASoC: SOF: Intel: hda-ipc: Do not process IPC reply before firmware boot
by Sasha Levin 14 Aug '22

14 Aug '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit 499cc881b09c8283ab5e75b0d6d21cb427722161 ] It is not yet clear, but it is possible to create a firmware so broken that it will send a reply message before a FW_READY message (it is not yet clear if FW_READY will arrive later). Since the reply_data is allocated only after the FW_READY message, this will lead to a NULL pointer dereference if not filtered out. The issue was reported with IPC4 firmware but the same condition is present for IPC3. Reported-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220712122357.31282-3-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/intel/hda-ipc.c | 39 ++++++++++++++++++++++------------- 1 file changed, 25 insertions(+), 14 deletions(-) diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index f08011249955..65e688f749ea 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -148,17 +148,23 @@ irqreturn_t hda_dsp_ipc4_irq_thread(int irq, void *context) if (primary & SOF_IPC4_MSG_DIR_MASK) { /* Reply received */ - struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; - data->primary = primary; - data->extension = extension; + data->primary = primary; + data->extension = extension; - spin_lock_irq(&sdev->ipc_lock); + spin_lock_irq(&sdev->ipc_lock); - snd_sof_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, data->primary); + snd_sof_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, data->primary); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, + "IPC reply before FW_READY: %#x|%#x\n", + primary, extension); + } } else { /* Notification received */ @@ -225,16 +231,21 @@ irqreturn_t hda_dsp_ipc_irq_thread(int irq, void *context) * place, the message might not yet be marked as expecting a * reply. */ - spin_lock_irq(&sdev->ipc_lock); + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core */ - hda_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, msg); + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); - /* set the done bit */ - hda_dsp_ipc_dsp_done(sdev); + /* set the done bit */ + hda_dsp_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_READY: %#x\n", + msg); + } ipc_irq = true; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 12/48] ASoC: SOF: Intel: cnl: Do not process IPC reply before firmware boot
by Sasha Levin 14 Aug '22

14 Aug '22

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit acacd9eefd0def5a83244d88e5483b5f38ee7287 ] It is not yet clear, but it is possible to create a firmware so broken that it will send a reply message before a FW_READY message (it is not yet clear if FW_READY will arrive later). Since the reply_data is allocated only after the FW_READY message, this will lead to a NULL pointer dereference if not filtered out. The issue was reported with IPC4 firmware but the same condition is present for IPC3. Reported-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20220712122357.31282-2-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/intel/cnl.c | 37 ++++++++++++++++++++++++------------- 1 file changed, 24 insertions(+), 13 deletions(-) diff --git a/sound/soc/sof/intel/cnl.c b/sound/soc/sof/intel/cnl.c index cd6e5f8a5eb4..6c98f65635fc 100644 --- a/sound/soc/sof/intel/cnl.c +++ b/sound/soc/sof/intel/cnl.c @@ -60,17 +60,23 @@ irqreturn_t cnl_ipc4_irq_thread(int irq, void *context) if (primary & SOF_IPC4_MSG_DIR_MASK) { /* Reply received */ - struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + struct sof_ipc4_msg *data = sdev->ipc->msg.reply_data; - data->primary = primary; - data->extension = extension; + data->primary = primary; + data->extension = extension; - spin_lock_irq(&sdev->ipc_lock); + spin_lock_irq(&sdev->ipc_lock); - snd_sof_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, data->primary); + snd_sof_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, data->primary); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, + "IPC reply before FW_READY: %#x|%#x\n", + primary, extension); + } } else { /* Notification received */ notification_data.primary = primary; @@ -124,15 +130,20 @@ irqreturn_t cnl_ipc_irq_thread(int irq, void *context) CNL_DSP_REG_HIPCCTL, CNL_DSP_REG_HIPCCTL_DONE, 0); - spin_lock_irq(&sdev->ipc_lock); + if (likely(sdev->fw_state == SOF_FW_BOOT_COMPLETE)) { + spin_lock_irq(&sdev->ipc_lock); - /* handle immediate reply from DSP core */ - hda_dsp_ipc_get_reply(sdev); - snd_sof_ipc_reply(sdev, msg); + /* handle immediate reply from DSP core */ + hda_dsp_ipc_get_reply(sdev); + snd_sof_ipc_reply(sdev, msg); - cnl_ipc_dsp_done(sdev); + cnl_ipc_dsp_done(sdev); - spin_unlock_irq(&sdev->ipc_lock); + spin_unlock_irq(&sdev->ipc_lock); + } else { + dev_dbg_ratelimited(sdev->dev, "IPC reply before FW_READY: %#x\n", + msg); + } ipc_irq = true; } -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 10/48] ALSA: hda: Fix page fault in snd_hda_codec_shutdown()
by Sasha Levin 14 Aug '22

14 Aug '22

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 980b3a8790b402e959a6d773b38b771019682be1 ] If early probe of HDAudio bus driver fails e.g.: due to missing firmware file, snd_hda_codec_shutdown() ends in manipulating uninitialized codec->pcm_list_head causing page fault. Iinitialization of HDAudio codec in ASoC is split in two: - snd_hda_codec_device_init() - snd_hda_codec_device_new() snd_hda_codec_device_init() is called during probe_codecs() by HDAudio bus driver while snd_hda_codec_device_new() is called by codec-component's ->probe(). The second call will not happen until all components required by related sound card are present within the ASoC framework. With firmware failing to load during the PCI's deferred initialization i.e.: probe_work(), no platform components are ever registered. HDAudio codec enumeration is done at that point though, so the codec components became registered to ASoC framework, calling snd_hda_codec_device_init() in the process. Now, during platform reboot snd_hda_codec_shutdown() is called for every codec found on the HDAudio bus causing oops if any of them has not completed both of their initialization steps. Relocating field initialization fixes the issue. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220706120230.427296-7-cezary.rojewski@intel.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_codec.c | 41 +++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7579a6982f47..d0cdb24ef3f3 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,8 +931,28 @@ snd_hda_codec_device_init(struct hda_bus *bus, unsigned int codec_addr, } codec->bus = bus; + codec->depop_delay = -1; + codec->fixup_id = HDA_FIXUP_ID_NOT_SET; + codec->core.dev.release = snd_hda_codec_dev_release; + codec->core.exec_verb = codec_exec_verb; codec->core.type = HDA_DEV_LEGACY; + mutex_init(&codec->spdif_mutex); + mutex_init(&codec->control_mutex); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); + snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); + snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); + snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); + INIT_LIST_HEAD(&codec->conn_list); + INIT_LIST_HEAD(&codec->pcm_list_head); + INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); + refcount_set(&codec->pcm_ref, 1); + init_waitqueue_head(&codec->remove_sleep); + return codec; } EXPORT_SYMBOL_GPL(snd_hda_codec_device_init); @@ -980,29 +1000,8 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) return -EINVAL; - codec->core.dev.release = snd_hda_codec_dev_release; - codec->core.exec_verb = codec_exec_verb; - codec->card = card; codec->addr = codec_addr; - mutex_init(&codec->spdif_mutex); - mutex_init(&codec->control_mutex); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); - snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); - snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); - snd_array_init(&codec->spdif_out, sizeof(struct hda_spdif_out), 16); - snd_array_init(&codec->jacktbl, sizeof(struct hda_jack_tbl), 16); - snd_array_init(&codec->verbs, sizeof(struct hda_verb *), 8); - INIT_LIST_HEAD(&codec->conn_list); - INIT_LIST_HEAD(&codec->pcm_list_head); - refcount_set(&codec->pcm_ref, 1); - init_waitqueue_head(&codec->remove_sleep); - - INIT_DELAYED_WORK(&codec->jackpoll_work, hda_jackpoll_work); - codec->depop_delay = -1; - codec->fixup_id = HDA_FIXUP_ID_NOT_SET; #ifdef CONFIG_PM codec->power_jiffies = jiffies; -- 2.35.1

1 0

[PATCH AUTOSEL 5.19 09/48] ASoC: Intel: avs: Set max DMA segment size
by Sasha Levin 14 Aug '22

14 Aug '22

From: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> [ Upstream commit 8544eebc78c96f1834a46b26ade3e7ebe785d10c ] Apparently it is possible for code to allocate large buffers which may cause warnings as reported in [1]. This was fixed for HDA, SOF and skylake in patchset [2], fix it also for avs driver. [1] https://github.com/thesofproject/linux/issues/3430 [2] https://lore.kernel.org/all/20220215132756.31236-1-tiwai@suse.de/ Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://lore.kernel.org/r/20220707124153.1858249-8-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/avs/core.c b/sound/soc/intel/avs/core.c index 3a0997c3af2b..cf373969bb69 100644 --- a/sound/soc/intel/avs/core.c +++ b/sound/soc/intel/avs/core.c @@ -445,6 +445,7 @@ static int avs_pci_probe(struct pci_dev *pci, const struct pci_device_id *id) dma_set_mask(dev, DMA_BIT_MASK(32)); dma_set_coherent_mask(dev, DMA_BIT_MASK(32)); } + dma_set_max_seg_size(dev, UINT_MAX); ret = avs_hdac_bus_init_streams(bus); if (ret < 0) { -- 2.35.1

1 0

[PATCH] soundwire: qcom: Check device status before reading devid
by Srinivas Kandagatla 13 Aug '22

13 Aug '22

As per hardware datasheet its recommended that we check the device status before reading devid assigned by auto-enumeration. Without this patch we see SoundWire devices with invalid enumeration addresses on the bus. Cc: stable(a)vger.kernel.org Fixes: a6e6581942ca ("soundwire: qcom: add auto enumeration support") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index dbf793e7e5d7..9df970eeca45 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -480,6 +480,10 @@ static int qcom_swrm_enumerate(struct sdw_bus *bus) char *buf1 = (char *)&val1, *buf2 = (char *)&val2; for (i = 1; i <= SDW_MAX_DEVICES; i++) { + /* do not continue if the status is Not Present */ + if (!ctrl->status[i]) + continue; + /*SCP_Devid5 - Devid 4*/ ctrl->reg_read(ctrl, SWRM_ENUMERATOR_SLAVE_DEV_ID_1(i), &val1); -- 2.25.1

3 3

snd_pcm_readi(),return -11,Resource temporarily unavailable
by GitHub issues - opened 13 Aug '22

13 Aug '22

alsa-project/alsa-lib issue #259 was opened from xukawendi: i called the function snd_pcm_readi() to capture pcm data, but the function returns -11,and the snd_strerror(-11) shows "Resource temporarily unavailable",How can I resolve this problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/259 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2 0/5] iio/hwmon/mfd/leds/net/power/ASoC: dt-bindings: few stale maintainers cleanup
by Krzysztof Kozlowski 12 Aug '22

12 Aug '22

Hi, Changes since v1 ================ 1. Patch #5: Drop also Ricardo Rivera-Matos and assign TI bindings to Andrew Davis 2. Add acks. A question ========== Several of the bindings here had only one maintainer and history does not always point to a new one (although I did not perform extensive digging). I added subsystem maintainer, because dtschema requires an entry with valid email address. This is not the best choice as simply subsystem maintainer might not have the actual device (or its datasheets or any interest in it). Maybe we could add some "orphaned" entry in such case? Best regards, Krzysztof Krzysztof Kozlowski (5): dt-bindings: iio: Drop Joachim Eastwood dt-bindings: iio: Drop Bogdan Pricop dt-bindings: Drop Beniamin Bia and Stefan Popa dt-bindings: Drop Robert Jones dt-bindings: Drop Dan Murphy and Ricardo Rivera-Matos Documentation/devicetree/bindings/hwmon/adi,adm1177.yaml | 1 - Documentation/devicetree/bindings/iio/accel/fsl,mma7455.yaml | 1 - Documentation/devicetree/bindings/iio/adc/adi,ad7091r5.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/adi,ad7606.yaml | 3 +-- Documentation/devicetree/bindings/iio/adc/nxp,lpc1850-adc.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/ti,adc108s102.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml | 2 +- .../devicetree/bindings/iio/amplifiers/adi,hmc425a.yaml | 1 - Documentation/devicetree/bindings/iio/imu/nxp,fxos8700.yaml | 2 +- .../devicetree/bindings/leds/leds-class-multicolor.yaml | 2 +- Documentation/devicetree/bindings/leds/leds-lp50xx.yaml | 2 +- Documentation/devicetree/bindings/mfd/gateworks-gsc.yaml | 1 - Documentation/devicetree/bindings/net/ti,dp83822.yaml | 2 +- Documentation/devicetree/bindings/net/ti,dp83867.yaml | 2 +- Documentation/devicetree/bindings/net/ti,dp83869.yaml | 2 +- Documentation/devicetree/bindings/power/supply/bq2515x.yaml | 3 +-- Documentation/devicetree/bindings/power/supply/bq256xx.yaml | 2 +- Documentation/devicetree/bindings/power/supply/bq25980.yaml | 3 +-- Documentation/devicetree/bindings/sound/tas2562.yaml | 2 +- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 2 +- 20 files changed, 16 insertions(+), 23 deletions(-) -- 2.34.1

3 8

Include <sys/soundcard.h> instead of <linux/soundcard.h>
by GitHub pull_request - opened 12 Aug '22

12 Aug '22

alsa-project/alsa-oss pull request #3 was opened from davidebeatrici: The `linux` folder strictly contains kernel headers, including OSSv3's. A distribution may package the OSS development header(s) separately. Request URL : https://github.com/alsa-project/alsa-oss/pull/3 Patch URL : https://github.com/alsa-project/alsa-oss/pull/3.patch Repository URL: https://github.com/alsa-project/alsa-oss

1 0

snd_pcm_sw_params_set_xfer_align() is deprecated
by GitHub issues - opened 11 Aug '22

11 Aug '22

alsa-project/alsa-oss issue #2 was opened from davidebeatrici: ```c pcm.c: In function 'oss_dsp_sw_params': pcm.c:376:17: warning: 'snd_pcm_sw_params_set_xfer_align' is deprecated [-Wdeprecated-declarations] 376 | snd_pcm_sw_params_set_xfer_align(pcm, sw, 1); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ In file included from /usr/include/alsa/asoundlib.h:54, from pcm.c:40: /usr/include/alsa/pcm.h:1368:5: note: declared here 1368 | int snd_pcm_sw_params_set_xfer_align(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, snd_pcm_uframes_t val) __attribute__((deprecated)); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ ``` Issue URL : https://github.com/alsa-project/alsa-oss/issues/2 Repository URL: https://github.com/alsa-project/alsa-oss

1 0

[PATCH v8 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 11 Aug '22

11 Aug '22

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Reviewed-by: Maxime Ripard <maxime(a)cerno.tech> --- v1->v2: 1.Fix some build errors. v2->v3: 1.Fix some build errors. v3->v4: 1.None. v4->v5: 1.Add interrupt. 2.Keep clock and reset index. v5->v6: 1.None. v6->v7: 1.None. v7->v8: 1.Fix some build errors. --- .../bindings/sound/allwinner,sun50i-h6-dmic.yaml | 79 ++++++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 0000000..0cfc07f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... -- 2.7.4

1 0

snd_pcm_hw_params_current() always returning zero for periods and buffer_time
by GitHub issues - opened 11 Aug '22

11 Aug '22

alsa-project/alsa-lib issue #258 was opened from ossilator: as can be reproduced rather easily using pyalsaaudio's pcm.dumpinfo(), snd_pcm_hw_params_current() seems to always return zero for periods and buffer_time, at least for playback streams (i didn't test capture). the sequence is basically this: ``` snd_pcm_hw_params_any(...); ... unsigned periods = 4; snd_pcm_hw_params_set_periods_near(..., &periods, ...); // periods is still fine here snd_pcm_hw_params(...); snd_pcm_hw_params_current(...); snd_pcm_hw_params_get_periods(..., &periods, ...); // nope, zero periods ``` i tested this with two completely different setups on two different machines (one a raw driver, another one with pipewire), so it doesn't appear related to the driver. from the other returned parameters it can be inferred that the period count is actually fine. one thing i noticed is that snd_pcm_hw_params_current() uses snd_interval_copy() instead of snd_interval_set_value() for exactly these two parameters. but min & max are both zero. i wasn't able to trace it down so far; both the library and the kernel code look fine. Issue URL : https://github.com/alsa-project/alsa-lib/issues/258 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: hda/cirrus - support for iMac 12,1 model
by Allen Ballway 11 Aug '22

11 Aug '22

The 12,1 model requires the same configuration as the 12,2 model to enable headphones but has a different codec SSID. Adds 12,1 SSID for matching quirk. Signed-off-by: Allen Ballway <ballway(a)chromium.org> --- sound/pci/hda/patch_cirrus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 678fbcaf2a3b..e1055d7d9be6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -396,6 +396,7 @@ static const struct snd_pci_quirk cs420x_fixup_tbl[] = { /* codec SSID */ SND_PCI_QUIRK(0x106b, 0x0600, "iMac 14,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x1c00, "MacBookPro 8,1", CS420X_MBP81), + SND_PCI_QUIRK(0x106b, 0x0900, "iMac 12,1", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2000, "iMac 12,2", CS420X_IMAC27_122), SND_PCI_QUIRK(0x106b, 0x2800, "MacBookPro 10,1", CS420X_MBP101), SND_PCI_QUIRK(0x106b, 0x5600, "MacBookAir 5,2", CS420X_MBP81), -- 2.37.1.559.g78731f0fdb-goog

2 1

[PATCH] ALSA: hda: Fix crash due to jack poll in suspend
by Mohan Kumar 11 Aug '22

11 Aug '22

With jackpoll_in_suspend flag set, there is a possibility that jack poll worker thread will run even after system suspend was completed. Any register access after system pm callback flow will result in kernel crash as still jack poll worker thread tries to access registers. To fix the crash issue during system flow, cancel the jack poll worker thread during system pm prepare callback and cancel the worker thread at start of runtime suspend callback and re-schedule at last to avoid any unwarranted access of register by worker thread during suspend flow. Signed-off-by: Mohan Kumar <mkumard(a)nvidia.com> --- sound/pci/hda/hda_codec.c | 14 +++++++++----- 1 file changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 7b2e62fa82d5..384426d7e9dd 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2940,8 +2940,7 @@ static int hda_codec_runtime_suspend(struct device *dev) if (!codec->card) return 0; - if (!codec->bus->jackpoll_in_suspend) - cancel_delayed_work_sync(&codec->jackpoll_work); + cancel_delayed_work_sync(&codec->jackpoll_work); state = hda_call_codec_suspend(codec); if (codec->link_down_at_suspend || @@ -2949,6 +2948,11 @@ static int hda_codec_runtime_suspend(struct device *dev) (state & AC_PWRST_CLK_STOP_OK))) snd_hdac_codec_link_down(&codec->core); snd_hda_codec_display_power(codec, false); + + if (codec->bus->jackpoll_in_suspend && + (dev->power.power_state.event != PM_EVENT_SUSPEND)) + schedule_delayed_work(&codec->jackpoll_work, + codec->jackpoll_interval); return 0; } @@ -2972,6 +2976,9 @@ static int hda_codec_runtime_resume(struct device *dev) #ifdef CONFIG_PM_SLEEP static int hda_codec_pm_prepare(struct device *dev) { + struct hda_codec *codec = dev_to_hda_codec(dev); + + cancel_delayed_work_sync(&codec->jackpoll_work); dev->power.power_state = PMSG_SUSPEND; return pm_runtime_suspended(dev); } @@ -2991,9 +2998,6 @@ static void hda_codec_pm_complete(struct device *dev) static int hda_codec_pm_suspend(struct device *dev) { - struct hda_codec *codec = dev_to_hda_codec(dev); - - cancel_delayed_work_sync(&codec->jackpoll_work); dev->power.power_state = PMSG_SUSPEND; return pm_runtime_force_suspend(dev); } -- 2.17.1

1 0

[PATCH] ASoC: Intel: kbl_rt5663_max98927: Simplify clk_get() usage
by Christophe JAILLET 10 Aug '22

10 Aug '22

If clk_get() returns -ENOENT, there is no need to defer the driver, -ENOENT will be returned the same for each retries. So, return the error code directly instead of -EPROBE_DEFER. Remove this special case and use dev_err_probe() to simplify code. It will also be less verbose if the clk is really deferred. Fixes: f7f61e08fe58 ("ASoC: Intel: kbl: Enable mclk and ssp sclk early") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This is based on my understanding of clk_get(). Review with care. Not sure the Fixes tag is needed. The patch does not fix anything. If devm_clk_get() returns -ENOENT, it will just loop several time until the framework gives up. If it returns -EPROBE_DEFER, this case is already handled by the "return ret;" So this patch should be a no-op, just a clean-up. --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 31 ++++---------------- 1 file changed, 6 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index 2d4224c5b152..07b00af2fa3c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -989,7 +989,6 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_rt5663_private *ctx; struct snd_soc_acpi_mach *mach; - int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -1009,32 +1008,14 @@ static int kabylake_audio_probe(struct platform_device *pdev) &constraints_dmic_2ch : &constraints_dmic_channels; ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->mclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->mclk), + "Failed to get ssp1_mclk\n"); ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->sclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->sclk), + "Failed to get ssp1_sclk\n"); return devm_snd_soc_register_card(&pdev->dev, kabylake_audio_card); } -- 2.34.1

3 4

[PATCH] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 10 Aug '22

10 Aug '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,sai.yaml | 175 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 ---------- 2 files changed, 175 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..adcd77531eba --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,175 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + minItems: 1 + maxItems: 2 + items: + enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - fsl,imx8ulp-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + minItems: 2 + maxItems: 2 + description: + Must contain a list of pairs of references to DMA specifiers, one for + transmission, and one for reception. + + dma-names: + minItems: 2 + maxItems: 2 + items: + enum: + - tx + - rx + + clocks: + minItems: 4 + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted, compatible for old dts) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + + clock-names: + minItems: 4 + maxItems: 7 + items: + enum: + - bus + - mclk0 + - mclk1 + - mclk2 + - mclk3 + - pll8k + - pll11k + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + big-endian: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Boolean property, required if all the SAI + registers are big-endian rather than little-endian. + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating + that SAI will work in the synchronous mode (sync Tx + with Rx) which means both the transmitter and the + receiver will send and receive data by following + receiver's bit clocks and frame sync clocks. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, indicating + that SAI will work in the asynchronous mode, which + means both transmitter and receiver will send and + receive data by following their own bit clocks and + frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + configure the dataline. it has 3 value for each configuration + first one means the type: I2S(1) or PDM(2) + second one is dataline mask for 'rx' + third one is dataline mask for 'tx'. + for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type + rx mask is 0xff, tx mask is 0x11 (dataline 1 and 5 enabled). + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, + indicates that SAI will output the SAI MCLK clock. + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: | + This is a boolean property. If present, + indicates that interrupt is shared with other modules. + + "#sound-dai-cells": + const: 0 + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "rx", "tx"; + dmas = <&edma0 0 20>, + <&edma0 0 21>; + big-endian; + lsb-first; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

4 5

Re: [PATCH] ASoC: codec: tlv320aic32x4: fix mono playback via I2S
by Mark Brown 10 Aug '22

10 Aug '22

On Wed, 10 Aug 2022 12:41:56 +0200, Philipp Zabel wrote: > The two commits referenced below break mono playback via I2S DAI because > they set BCLK to half the required speed. For PCM transport over I2S, the > number of transmitted channels is always 2, even for mono playback. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: codec: tlv320aic32x4: fix mono playback via I2S commit: b4b5f29a076e52181f63e45a2ad1bc88593072e3 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

ALSA: Setting FIFO type as the slave for dsnoop
by GitHub issues - opened 10 Aug '22

10 Aug '22

alsa-project/alsa-lib issue #257 was opened from cnkoder: **Problem** I am facing problems in using "dsnoop" to share/divide the audio stream into two. When dsnoop plugins' slave is set to be a FIFO it throws an error. executed: sudo arecord -D default -f cd defRecording.wav -c 1 -r 32000 error: ALSA lib pcm_direct.c:1809:(_snd_pcm_direct_get_slave_ipc_offset) Invalid type 'fifo' for slave PCM arecord: main:828: audio open error: Invalid argument This is the current asound.conf settings asound.conf   pcm.!default { type asym playback.pcm "playback" capture.pcm "capture1 " } pcm.playback { type plug slave.pcm "eci" } # Stream Output 1: Final pcm.capture1 { type plug slave.pcm "array" } # Stream Output 2: Final pcm.capture2 { type plug slave.pcm "array" } # Used to share the record device pcm.array { type dsnoop slave { pcm "eco" channels 1 } ipc_key 666666 } # Writes audio coming from any sort of player to ec.input, this is read by the echo # cancellation software. pcm.eci { type plug slave { format S16_LE rate 32000 channels 1 pcm { type file slave.pcm null file "/tmp/ec.input" format "raw" } } } # Read FIFO output which contains echo cancelled audio pcm.eco { type plug slave.pcm { type fifo infile "/tmp/ec.output" rate 32000 format S16_LE channels 1 } #ipc_key 666666 }  Note: eco is used to read the FIFO file which contains the echo canceled audio coming in from cancellation software. This software's input is hw:0 and records audio directly from the microphone, and then processes and passes this over to ec.output Dsnoop works well when the slave.pcm is a hardware device but as soon as I point to something else it fails. Is there a workaround or any other solution to tackle this problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/257 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Re: [PATCH v2] ASoC: amd: acp: Add TDM support for acp i2s stream
by Venkata Prasad Potturu 10 Aug '22

10 Aug '22

On 8/7/22 12:35, kernel test robot wrote: > Hi Venkata, > > Thank you for the patch! Yet something to improve: > > [auto build test ERROR on broonie-sound/for-next] > [also build test ERROR on linus/master next-20220805] > [cannot apply to v5.19] > [If your patch is applied to the wrong git tree, kindly drop us a note. > And when submitting patch, we suggest to use '--base' as documented in > https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgit-scm.c…] > > url:https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgithu… > base:https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgit.… for-next > config: x86_64-allmodconfig (https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fdownload.…) > compiler: gcc-11 (Debian 11.3.0-3) 11.3.0 > reproduce (this is a W=1 build): > #https://nam11.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgithub.com%2Fintel-lab-lkp%2Flinux%2Fcommit%2Fc603f9570621784501d7ce7f5099be37983cb810&data=05%7C01%7Cvenkataprasad.potturu%40amd.com%7C06397243ad964e430c5a08da784359bd%7C3dd8961fe4884e608e11a82d994e183d%7C0%7C0%7C637954527920715155%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000%7C%7C%7C&sdata=QaOUzq7KDkNDmDtTbawTnpB46weD9fXj8rnZmxUC9kI%3D&reserved=0 > git remote add linux-reviewhttps://nam11.safelinks.protection.outlook.com/?url=https%3A%2F… > git fetch --no-tags linux-review Venkata-Prasad-Potturu/ASoC-amd-acp-Add-TDM-support-for-acp-i2s-stream/20220805-140248 > git checkout c603f9570621784501d7ce7f5099be37983cb810 > # save the config file > mkdir build_dir && cp config build_dir/.config > make W=1 O=build_dir ARCH=x86_64 SHELL=/bin/bash sound/soc/ > > If you fix the issue, kindly add following tag where applicable > Reported-by: kernel test robot<lkp(a)intel.com> > > All errors (new ones prefixed by >>): > > sound/soc/amd/acp/acp-i2s.c: In function 'acp_i2s_set_tdm_slot': >>> sound/soc/amd/acp/acp-i2s.c:74:29: error: 'struct acp_dev_data' has no member named 'acp_lock' > 74 | spin_lock_irq(&adata->acp_lock); > | ^~ > In file included from include/linux/kernel.h:21, > from arch/x86/include/asm/percpu.h:27, > from arch/x86/include/asm/current.h:6, > from include/linux/sched.h:12, > from include/linux/ratelimit.h:6, > from include/linux/dev_printk.h:16, > from include/linux/device.h:15, > from include/linux/platform_device.h:13, > from sound/soc/amd/acp/acp-i2s.c:15: > sound/soc/amd/acp/acp-i2s.c:75:43: error: 'struct acp_dev_data' has no member named 'stream_list' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~ > include/linux/container_of.h:18:33: note: in definition of macro 'container_of' > 18 | void *__mptr = (void *)(ptr); \ > | ^~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' > 674 | for (pos = list_first_entry(head, typeof(*pos), member); \ > | ^~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:9: note: in expansion of macro 'list_for_each_entry' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~~~~~~~~~~~~~~~~~~ > In file included from include/linux/bits.h:22, > from include/linux/ratelimit_types.h:5, > from include/linux/ratelimit.h:5, > from include/linux/dev_printk.h:16, > from include/linux/device.h:15, > from include/linux/platform_device.h:13, > from sound/soc/amd/acp/acp-i2s.c:15: > sound/soc/amd/acp/acp-i2s.c:75:43: error: 'struct acp_dev_data' has no member named 'stream_list' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~ > include/linux/build_bug.h:78:56: note: in definition of macro '__static_assert' > 78 | #define __static_assert(expr, msg, ...) _Static_assert(expr, msg) > | ^~~~ > include/linux/container_of.h:19:9: note: in expansion of macro 'static_assert' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~~~ > include/linux/container_of.h:19:23: note: in expansion of macro '__same_type' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~ > include/linux/list.h:520:9: note: in expansion of macro 'container_of' > 520 | container_of(ptr, type, member) > | ^~~~~~~~~~~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' > 674 | for (pos = list_first_entry(head, typeof(*pos), member); \ > | ^~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:9: note: in expansion of macro 'list_for_each_entry' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~~~~~~~~~~~~~~~~~~ > include/linux/container_of.h:19:54: error: 'struct acp_stream' has no member named 'list' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~ > include/linux/build_bug.h:78:56: note: in definition of macro '__static_assert' > 78 | #define __static_assert(expr, msg, ...) _Static_assert(expr, msg) > | ^~~~ > include/linux/container_of.h:19:9: note: in expansion of macro 'static_assert' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~~~ > include/linux/container_of.h:19:23: note: in expansion of macro '__same_type' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~ > include/linux/list.h:520:9: note: in expansion of macro 'container_of' > 520 | container_of(ptr, type, member) > | ^~~~~~~~~~~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' > 674 | for (pos = list_first_entry(head, typeof(*pos), member); \ > | ^~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:9: note: in expansion of macro 'list_for_each_entry' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~~~~~~~~~~~~~~~~~~ > sound/soc/amd/acp/acp-i2s.c:75:43: error: 'struct acp_dev_data' has no member named 'stream_list' > 75 | list_for_each_entry(stream, &adata->stream_list, list) { > | ^~ > include/linux/build_bug.h:78:56: note: in definition of macro '__static_assert' > 78 | #define __static_assert(expr, msg, ...) _Static_assert(expr, msg) > | ^~~~ > include/linux/container_of.h:19:9: note: in expansion of macro 'static_assert' > 19 | static_assert(__same_type(*(ptr), ((type *)0)->member) || \ > | ^~~~~~~~~~~~~ > include/linux/container_of.h:20:23: note: in expansion of macro '__same_type' > 20 | __same_type(*(ptr), void), \ > | ^~~~~~~~~~~ > include/linux/list.h:520:9: note: in expansion of macro 'container_of' > 520 | container_of(ptr, type, member) > | ^~~~~~~~~~~~ > include/linux/list.h:531:9: note: in expansion of macro 'list_entry' > 531 | list_entry((ptr)->next, type, member) > | ^~~~~~~~~~ > include/linux/list.h:674:20: note: in expansion of macro 'list_first_entry' This patch is dependent on below patch, we will send these two patches in one series. depends on :- https://patchwork.kernel.org/project/alsa-devel/patch/20220804072556.601396… > > > vim +74 sound/soc/amd/acp/acp-i2s.c > > 47 > 48 static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask, > 49 int slots, int slot_width) > 50 { > 51 struct device *dev = dai->component->dev; > 52 struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); > 53 struct acp_stream *stream; > 54 int slot_len; > 55 > 56 switch (slot_width) { > 57 case SLOT_WIDTH_8: > 58 slot_len = 8; > 59 break; > 60 case SLOT_WIDTH_16: > 61 slot_len = 16; > 62 break; > 63 case SLOT_WIDTH_24: > 64 slot_len = 24; > 65 break; > 66 case SLOT_WIDTH_32: > 67 slot_len = 0; > 68 break; > 69 default: > 70 dev_err(dev, "Unsupported bitdepth %d\n", slot_width); > 71 return -EINVAL; > 72 } > 73 > > 74 spin_lock_irq(&adata->acp_lock); > 75 list_for_each_entry(stream, &adata->stream_list, list) { > 76 if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) > 77 adata->tdm_tx_fmt[stream->dai_id - 1] = > 78 FRM_LEN | (slots << 15) | (slot_len << 18); > 79 else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) > 80 adata->tdm_rx_fmt[stream->dai_id - 1] = > 81 FRM_LEN | (slots << 15) | (slot_len << 18); > 82 } > 83 spin_unlock_irq(&adata->acp_lock); > 84 return 0; > 85 } > 86 > > --

1 0

Re: [PATCH] ASoC: codecs: dt bind. doc for the new TI SRC4392 codec
by Mark Brown 10 Aug '22

10 Aug '22

On Wed, Aug 10, 2022 at 11:35:02AM +1000, Matt Flax wrote: > +maintainers: > + - Mark Brown <broonie(a)kernel.org> Please don't add me as a maintainer for device specific bindings that I'm not working on, the expectation is that people listed will know something about the device. If the schema requires an address and you don't want to take it on I guess alsa-devel would work?

1 0

Re: [PATCH v2 5/5] dt-bindings: Drop Dan Murphy and Ricardo Rivera-Matos
by Sebastian Reichel 10 Aug '22

10 Aug '22

Hi, On Tue, Aug 09, 2022 at 07:27:52PM +0300, Krzysztof Kozlowski wrote: > Emails to Dan Murphy and Ricardo Rivera-Matos bounce ("550 Invalid > recipient"). Andrew Davis agreed to take over the bindings. > > Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> > > --- > > Changes since v1: > 1. Add Andrew Davis instead. > 2. Not adding accumulated ack due to change above. > --- ... > Documentation/devicetree/bindings/power/supply/bq2515x.yaml | 3 +-- > Documentation/devicetree/bindings/power/supply/bq256xx.yaml | 2 +- > Documentation/devicetree/bindings/power/supply/bq25980.yaml | 3 +-- Acked-by: Sebastian Reichel <sebastian.reichel(a)collabora.com> -- Sebastian > [...] > diff --git a/Documentation/devicetree/bindings/power/supply/bq2515x.yaml b/Documentation/devicetree/bindings/power/supply/bq2515x.yaml > index 27db38577822..1a1b240034ef 100644 > --- a/Documentation/devicetree/bindings/power/supply/bq2515x.yaml > +++ b/Documentation/devicetree/bindings/power/supply/bq2515x.yaml > @@ -8,8 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: TI bq2515x 500-mA Linear charger family > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > - - Ricardo Rivera-Matos <r-rivera-matos(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The BQ2515x family is a highly integrated battery charge management IC that > diff --git a/Documentation/devicetree/bindings/power/supply/bq256xx.yaml b/Documentation/devicetree/bindings/power/supply/bq256xx.yaml > index 91abe5733c41..82f382a7ffb3 100644 > --- a/Documentation/devicetree/bindings/power/supply/bq256xx.yaml > +++ b/Documentation/devicetree/bindings/power/supply/bq256xx.yaml > @@ -8,7 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: TI bq256xx Switch Mode Buck Charger > > maintainers: > - - Ricardo Rivera-Matos <r-rivera-matos(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The bq256xx devices are a family of highly-integrated battery charge > diff --git a/Documentation/devicetree/bindings/power/supply/bq25980.yaml b/Documentation/devicetree/bindings/power/supply/bq25980.yaml > index 4883527ab5c7..b687b8bcd705 100644 > --- a/Documentation/devicetree/bindings/power/supply/bq25980.yaml > +++ b/Documentation/devicetree/bindings/power/supply/bq25980.yaml > @@ -8,8 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: TI BQ25980 Flash Charger > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > - - Ricardo Rivera-Matos <r-rivera-matos(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The BQ25980, BQ25975, and BQ25960 are a series of flash chargers intended > [...]

1 0

[PATCH 0/4] TAS2770 fixes
by Martin Povišer 10 Aug '22

10 Aug '22

The first two fixes should be straightforward. The latter two clean up what looks to me like a mess in the setting of power levels. However we settle it, we should then do the same changes to TAS2764, which has the same template (and maybe there are other drivers). Martin Povišer (4): ASoC: tas2770: Set correct FSYNC polarity ASoC: tas2770: Allow mono streams ASoC: tas2770: Drop conflicting set_bias_level power setting ASoC: tas2770: Fix handling of mute/unmute sound/soc/codecs/tas2770.c | 98 +++++++++++++++++--------------------- sound/soc/codecs/tas2770.h | 5 ++ 2 files changed, 48 insertions(+), 55 deletions(-) -- 2.33.0

2 6

Re: [PATCH v2 5/5] dt-bindings: Drop Dan Murphy and Ricardo Rivera-Matos
by Mark Brown 10 Aug '22

10 Aug '22

On Tue, Aug 09, 2022 at 07:27:52PM +0300, Krzysztof Kozlowski wrote: > Emails to Dan Murphy and Ricardo Rivera-Matos bounce ("550 Invalid > recipient"). Andrew Davis agreed to take over the bindings. Acked-by: Mark Brown <broonie(a)kernel.org>

1 0

Re: [PATCH v2 5/5] dt-bindings: Drop Dan Murphy and Ricardo Rivera-Matos
by Jonathan Cameron 10 Aug '22

10 Aug '22

On Tue, 9 Aug 2022 19:27:52 +0300 Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> wrote: > Emails to Dan Murphy and Ricardo Rivera-Matos bounce ("550 Invalid > recipient"). Andrew Davis agreed to take over the bindings. > > Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> > > --- > > Changes since v1: > 1. Add Andrew Davis instead. > 2. Not adding accumulated ack due to change above. > --- > Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml | 2 +- > .../devicetree/bindings/leds/leds-class-multicolor.yaml | 2 +- > Documentation/devicetree/bindings/leds/leds-lp50xx.yaml | 2 +- > Documentation/devicetree/bindings/net/ti,dp83822.yaml | 2 +- > Documentation/devicetree/bindings/net/ti,dp83867.yaml | 2 +- > Documentation/devicetree/bindings/net/ti,dp83869.yaml | 2 +- > Documentation/devicetree/bindings/power/supply/bq2515x.yaml | 3 +-- > Documentation/devicetree/bindings/power/supply/bq256xx.yaml | 2 +- > Documentation/devicetree/bindings/power/supply/bq25980.yaml | 3 +-- > Documentation/devicetree/bindings/sound/tas2562.yaml | 2 +- > Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 2 +- > 11 files changed, 11 insertions(+), 13 deletions(-) > > diff --git a/Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml b/Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml > index 9f5e96439c01..2e6abc9d746a 100644 > --- a/Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml > +++ b/Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml > @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: Texas Instruments' ads124s08 and ads124s06 ADC chip > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> For this one. Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com> > > properties: > compatible: > diff --git a/Documentation/devicetree/bindings/leds/leds-class-multicolor.yaml b/Documentation/devicetree/bindings/leds/leds-class-multicolor.yaml > index 12693483231f..31840e33dcf5 100644 > --- a/Documentation/devicetree/bindings/leds/leds-class-multicolor.yaml > +++ b/Documentation/devicetree/bindings/leds/leds-class-multicolor.yaml > @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: Common properties for the multicolor LED class. > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > Bindings for multi color LEDs show how to describe current outputs of > diff --git a/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml b/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml > index e0b658f07973..63da380748bf 100644 > --- a/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml > +++ b/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml > @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: LED driver for LP50XX RGB LED from Texas Instruments. > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The LP50XX is multi-channel, I2C RGB LED Drivers that can group RGB LEDs into > diff --git a/Documentation/devicetree/bindings/net/ti,dp83822.yaml b/Documentation/devicetree/bindings/net/ti,dp83822.yaml > index 75e8712e903a..f2489a9c852f 100644 > --- a/Documentation/devicetree/bindings/net/ti,dp83822.yaml > +++ b/Documentation/devicetree/bindings/net/ti,dp83822.yaml > @@ -8,7 +8,7 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" > title: TI DP83822 ethernet PHY > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The DP83822 is a low-power, single-port, 10/100 Mbps Ethernet PHY. It > diff --git a/Documentation/devicetree/bindings/net/ti,dp83867.yaml b/Documentation/devicetree/bindings/net/ti,dp83867.yaml > index 76ff08a477ba..b8c0e4b5b494 100644 > --- a/Documentation/devicetree/bindings/net/ti,dp83867.yaml > +++ b/Documentation/devicetree/bindings/net/ti,dp83867.yaml > @@ -11,7 +11,7 @@ allOf: > - $ref: "ethernet-controller.yaml#" > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The DP83867 device is a robust, low power, fully featured Physical Layer > diff --git a/Documentation/devicetree/bindings/net/ti,dp83869.yaml b/Documentation/devicetree/bindings/net/ti,dp83869.yaml > index 1b780dce61ab..b04ff0014a59 100644 > --- a/Documentation/devicetree/bindings/net/ti,dp83869.yaml > +++ b/Documentation/devicetree/bindings/net/ti,dp83869.yaml > @@ -11,7 +11,7 @@ allOf: > - $ref: "ethernet-phy.yaml#" > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The DP83869HM device is a robust, fully-featured Gigabit (PHY) transceiver > diff --git a/Documentation/devicetree/bindings/power/supply/bq2515x.yaml b/Documentation/devicetree/bindings/power/supply/bq2515x.yaml > index 27db38577822..1a1b240034ef 100644 > --- a/Documentation/devicetree/bindings/power/supply/bq2515x.yaml > +++ b/Documentation/devicetree/bindings/power/supply/bq2515x.yaml > @@ -8,8 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: TI bq2515x 500-mA Linear charger family > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > - - Ricardo Rivera-Matos <r-rivera-matos(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The BQ2515x family is a highly integrated battery charge management IC that > diff --git a/Documentation/devicetree/bindings/power/supply/bq256xx.yaml b/Documentation/devicetree/bindings/power/supply/bq256xx.yaml > index 91abe5733c41..82f382a7ffb3 100644 > --- a/Documentation/devicetree/bindings/power/supply/bq256xx.yaml > +++ b/Documentation/devicetree/bindings/power/supply/bq256xx.yaml > @@ -8,7 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: TI bq256xx Switch Mode Buck Charger > > maintainers: > - - Ricardo Rivera-Matos <r-rivera-matos(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The bq256xx devices are a family of highly-integrated battery charge > diff --git a/Documentation/devicetree/bindings/power/supply/bq25980.yaml b/Documentation/devicetree/bindings/power/supply/bq25980.yaml > index 4883527ab5c7..b687b8bcd705 100644 > --- a/Documentation/devicetree/bindings/power/supply/bq25980.yaml > +++ b/Documentation/devicetree/bindings/power/supply/bq25980.yaml > @@ -8,8 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: TI BQ25980 Flash Charger > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > - - Ricardo Rivera-Matos <r-rivera-matos(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The BQ25980, BQ25975, and BQ25960 are a series of flash chargers intended > diff --git a/Documentation/devicetree/bindings/sound/tas2562.yaml b/Documentation/devicetree/bindings/sound/tas2562.yaml > index 5f7dd5d6cbca..30f6b029ac08 100644 > --- a/Documentation/devicetree/bindings/sound/tas2562.yaml > +++ b/Documentation/devicetree/bindings/sound/tas2562.yaml > @@ -8,7 +8,7 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" > title: Texas Instruments TAS2562 Smart PA > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The TAS2562 is a mono, digital input Class-D audio amplifier optimized for > diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml > index bc2fb1a80ed7..ee698614862e 100644 > --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml > +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml > @@ -8,7 +8,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# > title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter > > maintainers: > - - Dan Murphy <dmurphy(a)ti.com> > + - Andrew Davis <afd(a)ti.com> > > description: | > The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital

1 0

[PATCH] aplay: Add support for G711 A law on .au files
by Christophe Leroy 10 Aug '22

10 Aug '22

.au files support G711 A law as well. Add it to aplay. Signed-off-by: Christophe Leroy <christophe.leroy(a)csgroup.eu> --- aplay/aplay.c | 9 +++++++++ aplay/formats.h | 1 + 2 files changed, 10 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index 6ce7191..c004bf9 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -1210,6 +1210,12 @@ static int test_au(int fd, void *buffer) fprintf(stderr, _("Warning: format is changed to S16_BE\n")); hwparams.format = SND_PCM_FORMAT_S16_BE; break; + case AU_FMT_ALAW: + if (hwparams.format != DEFAULT_FORMAT && + hwparams.format != SND_PCM_FORMAT_A_LAW) + fprintf(stderr, _("Warning: format is changed to A_LAW\n")); + hwparams.format = SND_PCM_FORMAT_A_LAW; + break; default: return -1; } @@ -2735,6 +2741,9 @@ static void begin_au(int fd, size_t cnt) case SND_PCM_FORMAT_S16_BE: ah.encoding = BE_INT(AU_FMT_LIN16); break; + case SND_PCM_FORMAT_A_LAW: + ah.encoding = BE_INT(AU_FMT_ALAW); + break; default: error(_("Sparc Audio doesn't support %s format..."), snd_pcm_format_name(hwparams.format)); prg_exit(EXIT_FAILURE); diff --git a/aplay/formats.h b/aplay/formats.h index ac0a2b0..cbc99fe 100644 --- a/aplay/formats.h +++ b/aplay/formats.h @@ -121,6 +121,7 @@ typedef struct { #define AU_FMT_ULAW 1 #define AU_FMT_LIN8 2 #define AU_FMT_LIN16 3 +#define AU_FMT_ALAW 27 typedef struct au_header { u_int magic; /* '.snd' */ -- 2.37.1

1 0

[PATCH] ASoC: rt5640: Fix the JD voltage dropping issue
by Oder Chiou 09 Aug '22

09 Aug '22

The patch fixes the JD voltage dropping issue in the HDA JD using. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Mohan Kumar D <mkumard(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 38ab8d4291c..5a844329800 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1986,7 +1986,7 @@ static int rt5640_set_bias_level(struct snd_soc_component *component, snd_soc_component_write(component, RT5640_PWR_MIXER, 0x0000); if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) snd_soc_component_write(component, RT5640_PWR_ANLG1, - 0x0018); + 0x2818); else snd_soc_component_write(component, RT5640_PWR_ANLG1, 0x0000); @@ -2600,7 +2600,8 @@ static void rt5640_enable_hda_jack_detect( snd_soc_component_update_bits(component, RT5640_DUMMY1, 0x400, 0x0); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, - RT5640_PWR_VREF2, RT5640_PWR_VREF2); + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG, + RT5640_PWR_VREF2 | RT5640_PWR_MB | RT5640_PWR_BG); usleep_range(10000, 15000); snd_soc_component_update_bits(component, RT5640_PWR_ANLG1, RT5640_PWR_FV2, RT5640_PWR_FV2); -- 2.34.1

2 1

[PATCH 0/5] iio/hwmon/mfd/leds/net/power/ASoC: dt-bindings: few stale maintainers cleanup
by Krzysztof Kozlowski 09 Aug '22

09 Aug '22

Hi, A question: Several of the bindings here had only one maintainer and history does not always point to a new one (although I did not perform extensive digging). I added subsystem maintainer, because dtschema requires such entry. This is not the best choice as simply subsystem maintainer might not have the actual device (or its datasheets or any interest in it). However dtschema requires a maintainer. Maybe we could add some "orphaned" entry in such case? Best regards, Krzysztof Krzysztof Kozlowski (5): dt-bindings: iio: Drop Joachim Eastwood dt-bindings: iio: Drop Bogdan Pricop dt-bindings: Drop Beniamin Bia and Stefan Popa dt-bindings: Drop Robert Jones dt-bindings: Drop Dan Murphy Documentation/devicetree/bindings/hwmon/adi,adm1177.yaml | 1 - Documentation/devicetree/bindings/iio/accel/fsl,mma7455.yaml | 1 - Documentation/devicetree/bindings/iio/adc/adi,ad7091r5.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/adi,ad7606.yaml | 3 +-- Documentation/devicetree/bindings/iio/adc/nxp,lpc1850-adc.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/ti,adc108s102.yaml | 2 +- Documentation/devicetree/bindings/iio/adc/ti,ads124s08.yaml | 2 +- .../devicetree/bindings/iio/amplifiers/adi,hmc425a.yaml | 1 - Documentation/devicetree/bindings/iio/imu/nxp,fxos8700.yaml | 2 +- .../devicetree/bindings/leds/leds-class-multicolor.yaml | 2 +- Documentation/devicetree/bindings/leds/leds-lp50xx.yaml | 2 +- Documentation/devicetree/bindings/mfd/gateworks-gsc.yaml | 1 - Documentation/devicetree/bindings/net/ti,dp83822.yaml | 2 +- Documentation/devicetree/bindings/net/ti,dp83867.yaml | 2 +- Documentation/devicetree/bindings/net/ti,dp83869.yaml | 2 +- Documentation/devicetree/bindings/power/supply/bq2515x.yaml | 1 - Documentation/devicetree/bindings/power/supply/bq25980.yaml | 1 - Documentation/devicetree/bindings/sound/tas2562.yaml | 2 +- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 2 +- 19 files changed, 13 insertions(+), 20 deletions(-) -- 2.34.1

5 16

Re: [PATCH 0/5] iio/hwmon/mfd/leds/net/power/ASoC: dt-bindings: few stale maintainers cleanup
by Jakub Kicinski 09 Aug '22

09 Aug '22

On Tue, 9 Aug 2022 08:25:29 +0300 Krzysztof Kozlowski wrote: > On 08/08/2022 21:52, Jakub Kicinski wrote: > > Integrating it with MAINTAINERS would be another option worth exploring > > although slightly tangential. > > > > How do you want this merged? It's all over the place subsystem-wise. > > I was thinking this could go via Rob's tree as fixes for current cycle, > so your Ack would be great Sounds good!

1 0

[PATCH] ALSA: usb-audio: make read-only array marker static const
by Colin Ian King 09 Aug '22

09 Aug '22

Don't populate the read-only array marker on the stack but instead make it static const. Also makes the object code a little smaller. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/usb/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index e692ae04436a..d45d1d7e6664 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1269,7 +1269,7 @@ static inline void fill_playback_urb_dsd_dop(struct snd_usb_substream *subs, unsigned int wrap = subs->buffer_bytes; u8 *dst = urb->transfer_buffer; u8 *src = runtime->dma_area; - u8 marker[] = { 0x05, 0xfa }; + static const u8 marker[] = { 0x05, 0xfa }; unsigned int queued = 0; /* -- 2.35.3

1 0

Behringer UMC204HD audio delay
by GitHub issues - opened 09 Aug '22

09 Aug '22

alsa-project/alsa-ucm-conf issue #198 was opened from vostok01: After update from 1.2.6.3 to 1.2.7.2, there was a desynchronization of audio and video. The sound is delayed by about 400 ms. Sometimes the out-of-sync disappears, but then it reappears. The sound is output via pulseaudio. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/198 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: [PATCH v2 5/5] dt-bindings: Drop Dan Murphy and Ricardo Rivera-Matos
by Andrew Davis 09 Aug '22

09 Aug '22

On 8/9/22 11:27 AM, Krzysztof Kozlowski wrote: > Emails to Dan Murphy and Ricardo Rivera-Matos bounce ("550 Invalid > recipient"). Andrew Davis agreed to take over the bindings. > > Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> > > --- Acked-by: Andrew Davis <afd(a)ti.com>

1 0

[PATCH] ALSA: hda/realtek: Add a quirk for HP OMEN 15 (8786) mute LED
by Bedant 09 Aug '22

09 Aug '22

From: Bedant Patnaik <bedant.patnaik(a)gmail.com> Board ID 8786 seems to be another variant of the Omen 15 that needs ALC285_FIXUP_HP_MUTE_LED for working mute LED. Signed-off-by: Bedant Patnaik <bedant.patnaik(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8a57636f6..258f4e490 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9203,6 +9203,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8783, "HP ZBook Fury 15 G7 Mobile Workstation", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x8786, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8787, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8788, "HP OMEN 15", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x87c8, "HP", ALC287_FIXUP_HP_GPIO_LED), -- 2.37.1

2 1

[PATCH] ALSA: scarlett2: Add Focusrite Clarett+ 8Pre support
by Geoffrey D. Bennett 09 Aug '22

09 Aug '22

From: Christian Colglazier <christian(a)cacolglazier.com> The Focusrite Clarett+ 8Pre uses the same protocol as the Scarlett Gen 2 and Gen 3 product range. This patch adds support for the Clarett+ 8Pre by adding appropriate entries to the scarlett2 driver. The Clarett+ 2Pre and 4Pre, and the Clarett USB product line presumably use the same protocol as well, so support for them can easily be added if someone can test. Signed-off-by: Christian Colglazier <christian(a)cacolglazier.com> Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_quirks.c | 1 + sound/usb/mixer_scarlett_gen2.c | 91 ++++++++++++++++++++++++++++++++- 2 files changed, 90 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index d35cf54cab33..4a6ee50f69c4 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3238,6 +3238,7 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x8213): /* Focusrite Scarlett 8i6 3rd Gen */ case USB_ID(0x1235, 0x8214): /* Focusrite Scarlett 18i8 3rd Gen */ case USB_ID(0x1235, 0x8215): /* Focusrite Scarlett 18i20 3rd Gen */ + case USB_ID(0x1235, 0x820c): /* Focusrite Clarett+ 8Pre */ err = snd_scarlett_gen2_init(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 69a2cd429ee2..9d11bb08667e 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -1,13 +1,15 @@ // SPDX-License-Identifier: GPL-2.0 /* - * Focusrite Scarlett Gen 2/3 Driver for ALSA + * Focusrite Scarlett Gen 2/3 and Clarett+ Driver for ALSA * * Supported models: * - 6i6/18i8/18i20 Gen 2 * - Solo/2i2/4i4/8i6/18i8/18i20 Gen 3 + * - Clarett+ 8Pre * * Copyright (c) 2018-2022 by Geoffrey D. Bennett <g at b4.vu> * Copyright (c) 2020-2021 by Vladimir Sadovnikov <sadko4u(a)gmail.com> + * Copyright (c) 2022 by Christian Colglazier <christian(a)cacolglazier.com> * * Based on the Scarlett (Gen 1) Driver for ALSA: * @@ -51,6 +53,9 @@ * Support for phantom power, direct monitoring, speaker switching, * and talkback added in May-June 2021. * + * Support for Clarett+ 8Pre added in Aug 2022 by Christian + * Colglazier. + * * This ALSA mixer gives access to (model-dependent): * - input, output, mixer-matrix muxes * - mixer-matrix gain stages @@ -203,7 +208,8 @@ enum { SCARLETT2_CONFIG_SET_NO_MIXER = 0, SCARLETT2_CONFIG_SET_GEN_2 = 1, SCARLETT2_CONFIG_SET_GEN_3 = 2, - SCARLETT2_CONFIG_SET_COUNT = 3 + SCARLETT2_CONFIG_SET_CLARETT = 3, + SCARLETT2_CONFIG_SET_COUNT = 4 }; /* Hardware port types: @@ -841,6 +847,61 @@ static const struct scarlett2_device_info s18i20_gen3_info = { } }, }; +static const struct scarlett2_device_info clarett_8pre_info = { + .usb_id = USB_ID(0x1235, 0x820c), + + .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .line_out_hw_vol = 1, + .level_input_count = 2, + .air_input_count = 8, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + NULL, + NULL, + NULL, + NULL, + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 8, 10 }, + [SCARLETT2_PORT_TYPE_SPDIF] = { 2, 2 }, + [SCARLETT2_PORT_TYPE_ADAT] = { 8, 8 }, + [SCARLETT2_PORT_TYPE_MIX] = { 10, 18 }, + [SCARLETT2_PORT_TYPE_PCM] = { 20, 18 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_PCM, 0, 18 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 10 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_ADAT, 0, 8 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 14 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 10 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_ADAT, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 12 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 10 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 22 }, + { 0, 0, 0 }, + } }, +}; + static const struct scarlett2_device_info *scarlett2_devices[] = { /* Supported Gen 2 devices */ &s6i6_gen2_info, @@ -855,6 +916,9 @@ static const struct scarlett2_device_info *scarlett2_devices[] = { &s18i8_gen3_info, &s18i20_gen3_info, + /* Supported Clarett+ devices */ + &clarett_8pre_info, + /* End of list */ NULL }; @@ -1047,6 +1111,29 @@ static const struct scarlett2_config [SCARLETT2_CONFIG_TALKBACK_MAP] = { .offset = 0xb0, .size = 16, .activate = 10 }, + +/* Clarett+ 8Pre */ +}, { + [SCARLETT2_CONFIG_DIM_MUTE] = { + .offset = 0x31, .size = 8, .activate = 2 }, + + [SCARLETT2_CONFIG_LINE_OUT_VOLUME] = { + .offset = 0x34, .size = 16, .activate = 1 }, + + [SCARLETT2_CONFIG_MUTE_SWITCH] = { + .offset = 0x5c, .size = 8, .activate = 1 }, + + [SCARLETT2_CONFIG_SW_HW_SWITCH] = { + .offset = 0x66, .size = 8, .activate = 3 }, + + [SCARLETT2_CONFIG_LEVEL_SWITCH] = { + .offset = 0x7c, .size = 8, .activate = 7 }, + + [SCARLETT2_CONFIG_AIR_SWITCH] = { + .offset = 0x95, .size = 8, .activate = 8 }, + + [SCARLETT2_CONFIG_STANDALONE_SWITCH] = { + .offset = 0x8d, .size = 8, .activate = 6 }, } }; /* proprietary request/response format */ -- 2.37.1

2 1

[PATCH] ALSA: usb-audio: More comprehensive mixer map for ASUS ROG Zenith II
by Takashi Iwai 09 Aug '22

09 Aug '22

ASUS ROG Zenith II has two USB interfaces, one for the front headphone and another for the rest I/O. Currently we provided the mixer mapping for the latter but with an incomplete form. This patch corrects and provides more comprehensive mixer mapping, as well as providing the proper device names for both the front headphone and main audio. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=211005 Fixes: 2a48218f8e23 ("ALSA: usb-audio: Add mixer workaround for TRX40 and co") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/card.c | 8 ++++++++ sound/usb/mixer_maps.c | 34 +++++++++++++++++++++++++--------- 2 files changed, 33 insertions(+), 9 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 0fff96a5d3ab..d356743de2ff 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -387,6 +387,14 @@ static const struct usb_audio_device_name usb_audio_names[] = { DEVICE_NAME(0x05e1, 0x0408, "Syntek", "STK1160"), DEVICE_NAME(0x05e1, 0x0480, "Hauppauge", "Woodbury"), + /* ASUS ROG Zenith II: this machine has also two devices, one for + * the front headphone and another for the rest + */ + PROFILE_NAME(0x0b05, 0x1915, "ASUS", "Zenith II Front Headphone", + "Zenith-II-Front-Headphone"), + PROFILE_NAME(0x0b05, 0x1916, "ASUS", "Zenith II Main Audio", + "Zenith-II-Main-Audio"), + /* ASUS ROG Strix */ PROFILE_NAME(0x0b05, 0x1917, "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"), diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 3c795675f048..f4bd1e8ae4b6 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -374,13 +374,28 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { { 0 } }; -/* Some mobos shipped with a dummy HD-audio show the invalid GET_MIN/GET_MAX - * response for Input Gain Pad (id=19, control=12) and the connector status - * for SPDIF terminal (id=18). Skip them. - */ -static const struct usbmix_name_map asus_rog_map[] = { - { 18, NULL }, /* OT, connector control */ - { 19, NULL, 12 }, /* FU, Input Gain Pad */ +/* ASUS ROG Zenith II with Realtek ALC1220-VB */ +static const struct usbmix_name_map asus_zenith_ii_map[] = { + { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ + { 16, "Speaker" }, /* OT */ + { 22, "Speaker Playback" }, /* FU */ + { 7, "Line" }, /* IT */ + { 19, "Line Capture" }, /* FU */ + { 8, "Mic" }, /* IT */ + { 20, "Mic Capture" }, /* FU */ + { 9, "Front Mic" }, /* IT */ + { 21, "Front Mic Capture" }, /* FU */ + { 17, "IEC958" }, /* OT */ + { 23, "IEC958 Playback" }, /* FU */ + {} +}; + +static const struct usbmix_connector_map asus_zenith_ii_connector_map[] = { + { 10, 16 }, /* (Back) Speaker */ + { 11, 17 }, /* SPDIF */ + { 13, 7 }, /* Line */ + { 14, 8 }, /* Mic */ + { 15, 9 }, /* Front Mic */ {} }; @@ -611,9 +626,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = gigabyte_b450_map, .connector_map = gigabyte_b450_connector_map, }, - { /* ASUS ROG Zenith II */ + { /* ASUS ROG Zenith II (main audio) */ .id = USB_ID(0x0b05, 0x1916), - .map = asus_rog_map, + .map = asus_zenith_ii_map, + .connector_map = asus_zenith_ii_connector_map, }, { /* ASUS ROG Strix */ .id = USB_ID(0x0b05, 0x1917), -- 2.35.3

1 0

Re: [PATCH 5/5] dt-bindings: Drop Dan Murphy
by Andrew Lunn 08 Aug '22

08 Aug '22

> I'm not seeing his name in the latest MAINTAINERS, seem they all got > dropped in 57a9006240b2. Ah, great. And there does not appear to be a MAINTAINERS entry for any of the TI PHYs, so giving the correct impression they are without a Maintainer. Andrew

1 0

[PATCH] ASoC: codecs: add support for the TI SRC4392 codec
by Matt Flax 08 Aug '22

08 Aug '22

The src4xxx keyword is used for future capability to integrate other codecs similar to the src4392 to the same code base. Signed-off-by: Matt Flax <flatmax(a)flatmax.com> --- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/src4xxx-i2c.c | 55 ++++ sound/soc/codecs/src4xxx.c | 538 +++++++++++++++++++++++++++++++++ sound/soc/codecs/src4xxx.h | 115 +++++++ 5 files changed, 725 insertions(+) create mode 100644 sound/soc/codecs/src4xxx-i2c.c create mode 100644 sound/soc/codecs/src4xxx.c create mode 100644 sound/soc/codecs/src4xxx.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d16b4efb88a7..44c19d973ef3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -205,6 +205,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_SIMPLE_AMPLIFIER imply SND_SOC_SIMPLE_MUX imply SND_SOC_SPDIF + imply SND_SOC_SRC4XXX_I2C imply SND_SOC_SSM2305 imply SND_SOC_SSM2518 imply SND_SOC_SSM2602_SPI @@ -1471,6 +1472,18 @@ config SND_SOC_SIMPLE_MUX config SND_SOC_SPDIF tristate "S/PDIF CODEC" +config SND_SOC_SRC4XXX_I2C + tristate + depends on I2C + select SND_SOC_SRC4XXX + +config SND_SOC_SRC4XXX + tristate "Texas Instruments SRC4XXX DIR/DIT and SRC codecs" + help + Enable support for the TI SRC4XXX family of codecs. These include the + scr4392 which has digital receivers, transmitters, and + a sample rate converter, including numerous ports. + config SND_SOC_SSM2305 tristate "Analog Devices SSM2305 Class-D Amplifier" help diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 92fd441d426a..a599b727c65b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -231,6 +231,8 @@ snd-soc-sigmadsp-regmap-objs := sigmadsp-regmap.o snd-soc-si476x-objs := si476x.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o +snd-soc-src4xxx-objs := src4xxx.o +snd-soc-src4xxx-i2c-objs := src4xxx-i2c.o snd-soc-ssm2305-objs := ssm2305.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o @@ -579,6 +581,8 @@ obj-$(CONFIG_SND_SOC_SIGMADSP_I2C) += snd-soc-sigmadsp-i2c.o obj-$(CONFIG_SND_SOC_SIGMADSP_REGMAP) += snd-soc-sigmadsp-regmap.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o +obj-$(CONFIG_SND_SOC_SRC4XXX) += snd-soc-src4xxx.o +obj-$(CONFIG_SND_SOC_SRC4XXX_I2C) += snd-soc-src4xxx-i2c.o obj-$(CONFIG_SND_SOC_SSM2305) += snd-soc-ssm2305.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o diff --git a/sound/soc/codecs/src4xxx-i2c.c b/sound/soc/codecs/src4xxx-i2c.c new file mode 100644 index 000000000000..4547078a076b --- /dev/null +++ b/sound/soc/codecs/src4xxx-i2c.c @@ -0,0 +1,55 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * Driver for SRC4XXX codecs + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Natt Flax <flatmax(a)flatmax.com> + */ + +#include <linux/i2c.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> + +#include "src4xxx.h" + +static int src4xxx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + return src4xxx_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &src4xxx_regmap_config), NULL); +} + +static int src4xxx_i2c_remove(struct i2c_client *i2c) +{ + src4xxx_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id src4xxx_i2c_ids[] = { + { "src4392", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, src4xxx_i2c_ids); + +static const struct of_device_id src4xxx_of_match[] = { + { .compatible = "ti,src4392", }, + { } +}; +MODULE_DEVICE_TABLE(of, src4xxx_of_match); + + +static struct i2c_driver src4xxx_i2c_driver = { + .driver = { + .name = "src4xxx", + .of_match_table = of_match_ptr(src4xxx_of_match), + }, + .probe = src4xxx_i2c_probe, + .remove = src4xxx_i2c_remove, + .id_table = src4xxx_i2c_ids, +}; +module_i2c_driver(src4xxx_i2c_driver); + +MODULE_DESCRIPTION("ASoC SRC4392 CODEC I2C driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.c b/sound/soc/codecs/src4xxx.c new file mode 100644 index 000000000000..11ad6cb5b086 --- /dev/null +++ b/sound/soc/codecs/src4xxx.c @@ -0,0 +1,538 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * TI SRC4xxx Audio Codec driver + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt Flax <flatmax(a)flatmax.com> + */ + +#include <linux/module.h> + +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "src4xxx.h" + +struct src4xxx { + struct regmap *regmap; + bool master[2]; + int mclk_hz; + struct device *dev; +}; + +enum {SRC4XXX_PORTA, SRC4XXX_PORTB}; + +/* SRC attenuation */ +static const DECLARE_TLV_DB_SCALE(src_tlv, -12750, 50, 0); + +static const struct snd_kcontrol_new src4xxx_controls[] = { + SOC_DOUBLE_R_TLV("SRC vol", + SRC4XXX_SCR_CTL_30, SRC4XXX_SCR_CTL_31, 0, 255, 1, src_tlv), +}; + +/* I2S port control */ +static const char * const port_out_src_text[] = { + "loopback", "other_port", "DIR", "SRC" +}; +static SOC_ENUM_SINGLE_DECL(porta_out_src_enum, SRC4XXX_PORTA_CTL_03, 4, + port_out_src_text); +static SOC_ENUM_SINGLE_DECL(portb_out_src_enum, SRC4XXX_PORTB_CTL_05, 4, + port_out_src_text); +static const struct snd_kcontrol_new porta_out_control = + SOC_DAPM_ENUM("Port A source select", porta_out_src_enum); +static const struct snd_kcontrol_new portb_out_control = + SOC_DAPM_ENUM("Port B source select", portb_out_src_enum); + +/* Digital audio transmitter control */ +static const char * const dit_mux_text[] = {"Port A", "Port B", "DIR", "SRC"}; +static SOC_ENUM_SINGLE_DECL(dit_mux_enum, SRC4XXX_TX_CTL_07, 3, dit_mux_text); +static const struct snd_kcontrol_new dit_mux_control = + SOC_DAPM_ENUM("Dig. Out src", dit_mux_enum); + +/* SRC control */ +static const char * const src_in_text[] = {"Port A", "Port B", "DIR"}; +static SOC_ENUM_SINGLE_DECL(src_in_enum, SRC4XXX_SCR_CTL_2D, 0, src_in_text); +static const struct snd_kcontrol_new src_in_control = + SOC_DAPM_ENUM("SRC source select", src_in_enum); + +static const char * const src_mclk_text[] = { + "Master (MCLK)", "Master (RXCLKI)", "Recovered receiver clk", +}; +static SOC_ENUM_SINGLE_DECL(src_mclk_enum, SRC4XXX_SCR_CTL_2D, 2, + src_mclk_text); +static const struct snd_kcontrol_new src_mclk_control = + SOC_DAPM_ENUM("SRC master clock select", src_mclk_enum); + +/* DIR control */ +static const char * const dir_in_text[] = {"Ch 1", "Ch 2", "Ch 3", "Ch 4"}; +static SOC_ENUM_SINGLE_DECL(dir_in_enum, SRC4XXX_RCV_CTL_0D, 0, dir_in_text); +static const struct snd_kcontrol_new dir_in_control = + SOC_DAPM_ENUM("Digital Input", dir_in_enum); + +static const struct snd_soc_dapm_widget src4xxx_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("loopback_A"), + SND_SOC_DAPM_INPUT("other_port_A"), + SND_SOC_DAPM_INPUT("DIR_A"), + SND_SOC_DAPM_INPUT("SRC_A"), + SND_SOC_DAPM_MUX("Port A source", + SND_SOC_NOPM, 0, 0, &porta_out_control), + + SND_SOC_DAPM_INPUT("loopback_B"), + SND_SOC_DAPM_INPUT("other_port_B"), + SND_SOC_DAPM_INPUT("DIR_B"), + SND_SOC_DAPM_INPUT("SRC_B"), + SND_SOC_DAPM_MUX("Port B source", + SND_SOC_NOPM, 0, 0, &portb_out_control), + + SND_SOC_DAPM_INPUT("Port_A"), + SND_SOC_DAPM_INPUT("Port_B"), + SND_SOC_DAPM_INPUT("DIR_"), + + /* Digital audio receivers and transmitters */ + SND_SOC_DAPM_OUTPUT("DIR_OUT"), + SND_SOC_DAPM_OUTPUT("SRC_OUT"), + SND_SOC_DAPM_MUX("DIT Out Src", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIT_SHIFT, 1, &dit_mux_control), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF_A_RX", "Playback A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_A_TX", "Capture A", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_A_SHIFT, 1), + SND_SOC_DAPM_AIF_IN("AIF_B_RX", "Playback B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF_B_TX", "Capture B", 0, + SRC4XXX_PWR_RST_01, SRC4XXX_ENABLE_PORT_B_SHIFT, 1), + + SND_SOC_DAPM_MUX("SRC source", SND_SOC_NOPM, 0, 0, &src_in_control), + + SND_SOC_DAPM_INPUT("MCLK"), + SND_SOC_DAPM_INPUT("RXMCLKI"), + SND_SOC_DAPM_INPUT("RXMCLKO"), + SND_SOC_DAPM_MUX("SRC mclk source", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_SRC_SHIFT, 1, &src_mclk_control), + + SND_SOC_DAPM_INPUT("RX1"), + SND_SOC_DAPM_INPUT("RX2"), + SND_SOC_DAPM_INPUT("RX3"), + SND_SOC_DAPM_INPUT("RX4"), + SND_SOC_DAPM_MUX("Digital Input", SRC4XXX_PWR_RST_01, + SRC4XXX_ENABLE_DIR_SHIFT, 1, &dir_in_control), +}; + +static const struct snd_soc_dapm_route src4xxx_audio_routes[] = { + /* I2S Input to Output Routing */ + {"Port A source", "loopback", "loopback_A"}, + {"Port A source", "other_port", "other_port_A"}, + {"Port A source", "DIR", "DIR_A"}, + {"Port A source", "SRC", "SRC_A"}, + {"Port B source", "loopback", "loopback_B"}, + {"Port B source", "other_port", "other_port_B"}, + {"Port B source", "DIR", "DIR_B"}, + {"Port B source", "SRC", "SRC_B"}, + /* DIT muxing */ + {"DIT Out Src", "Port A", "Capture A"}, + {"DIT Out Src", "Port B", "Capture B"}, + {"DIT Out Src", "DIR", "DIR_OUT"}, + {"DIT Out Src", "SRC", "SRC_OUT"}, + + /* SRC input selection */ + {"SRC source", "Port A", "Port_A"}, + {"SRC source", "Port B", "Port_B"}, + {"SRC source", "DIR", "DIR_"}, + /* SRC mclk selection */ + {"SRC mclk source", "Master (MCLK)", "MCLK"}, + {"SRC mclk source", "Master (RXCLKI)", "RXMCLKI"}, + {"SRC mclk source", "Recovered receiver clk", "RXMCLKO"}, + /* DIR input selection */ + {"Digital Input", "Ch 1", "RX1"}, + {"Digital Input", "Ch 2", "RX2"}, + {"Digital Input", "Ch 3", "RX3"}, + {"Digital Input", "Ch 4", "RX4"}, +}; + + +static const struct snd_soc_component_driver src4xxx_driver = { + .controls = src4xxx_controls, + .num_controls = ARRAY_SIZE(src4xxx_controls), + + .dapm_widgets = src4xxx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(src4xxx_dapm_widgets), + .dapm_routes = src4xxx_audio_routes, + .num_dapm_routes = ARRAY_SIZE(src4xxx_audio_routes), +}; + +static int src4xxx_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int ctrl; + + dev_info(dai->dev, "__func__ enter 0x%x id=%d\n", + fmt, dai->id); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = SRC4XXX_BUS_MASTER; + src4xxx->master[dai->id] = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + src4xxx->master[dai->id] = false; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= SRC4XXX_BUS_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= SRC4XXX_BUS_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= SRC4XXX_BUS_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + regmap_update_bits(src4xxx->regmap, SRC4XXX_BUS_FMT(dai->id), + SRC4XXX_BUS_FMT_MS_MASK, ctrl); + + dev_info(dai->dev, "__func__ exit\n"); + return 0; +} + +static int src4xxx_set_mclk_hz(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + + dev_info(component->dev, "changing mclk rate from %d to %d Hz\n", + src4xxx->mclk_hz, freq); + src4xxx->mclk_hz = freq; + + return 0; +} + +static int src4xxx_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 src4xxx *src4xxx = snd_soc_component_get_drvdata(component); + unsigned int mclk_div; + int val, pj, jd, d; + int reg = SRC4XXX_PORTA_CTL_04; + int ret; + + if (dai->id == SRC4XXX_PORTB) + reg = SRC4XXX_PORTB_CTL_06; + + dev_info(dai->dev, "__func__ enter for dai %d\n", dai->id); + + if (src4xxx->master[dai->id]) { + mclk_div = src4xxx->mclk_hz/params_rate(params); + if (src4xxx->mclk_hz != mclk_div*params_rate(params)) { + dev_err(component->dev, + "mclk %d / rate %d has a remainder.\n", + src4xxx->mclk_hz, params_rate(params)); + return -EINVAL; + } + + dev_info(dai->dev, "__func__ mclk %d rate %d div %d\n", + src4xxx->mclk_hz, params_rate(params), mclk_div); + val = ((int)mclk_div - 128) / 128; + dev_info(dai->dev, + "__func__ register value = %d\n", val); + if ((val < 0) | (val > 3)) { + dev_err(component->dev, + "div register setting %d is out of range\n", + val); + dev_err(component->dev, + "unsupported sample rate %d Hz for the master clock of %d Hz\n", + params_rate(params), src4xxx->mclk_hz); + return -EINVAL; + } + + /* set the TX DIV */ + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + /* set the PLL for the digital receiver */ + switch (src4xxx->mclk_hz) { + case 24576000: + pj = 0x22; + jd = 0x00; + d = 0x00; + break; + case 22579200: + pj = 0x22; + jd = 0x1b; + d = 0xa3; + break; + default: + /* don't error out here, + * other parts of the chip are still functional + */ + dev_info(component->dev, + "Couldn't set the RCV PLL as this master clock rate is unknown\n"); + } + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_0F, pj); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_0F); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_10, jd); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_10); + ret = regmap_write(src4xxx->regmap, SRC4XXX_RCV_PLL_11, d); + if (ret < 0) + dev_err(component->dev, + "Failed to update PLL register 0x%x\n", + SRC4XXX_RCV_PLL_11); + + ret = regmap_update_bits(src4xxx->regmap, + SRC4XXX_TX_CTL_07, SRC4XXX_TX_MCLK_DIV_MASK, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + if (ret < 0) { + dev_err(component->dev, + "Couldn't set the TX's div register to %d << %d = 0x%x\n", + val, SRC4XXX_TX_MCLK_DIV_SHIFT, + val<<SRC4XXX_TX_MCLK_DIV_SHIFT); + return ret; + } + + return regmap_update_bits(src4xxx->regmap, reg, + SRC4XXX_MCLK_DIV_MASK, val); + } else + dev_info(dai->dev, "not setting up MCLK as not master\n"); + + return 0; +}; + +static const struct snd_soc_dai_ops src4xxx_dai_ops = { + .hw_params = src4xxx_hw_params, + .set_sysclk = src4xxx_set_mclk_hz, + .set_fmt = src4xxx_set_dai_fmt, +}; + +#define SRC4XXX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) +#define SRC4XXX_RATES (SNDRV_PCM_RATE_44100|SNDRV_PCM_RATE_48000|\ + SNDRV_PCM_RATE_88200|\ + SNDRV_PCM_RATE_96000|\ + SNDRV_PCM_RATE_176400|\ + SNDRV_PCM_RATE_192000) + +static struct snd_soc_dai_driver src4xxx_dai_driver[] = { + { + .id = SRC4XXX_PORTA, + .name = "src4xxx-portA", + .playback = { + .stream_name = "Playback A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture A", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, + { + .id = SRC4XXX_PORTB, + .name = "src4xxx-portB", + .playback = { + .stream_name = "Playback B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .capture = { + .stream_name = "Capture B", + .channels_min = 2, + .channels_max = 2, + .rates = SRC4XXX_RATES, + .formats = SRC4XXX_FORMATS, + }, + .ops = &src4xxx_dai_ops, + }, +}; + +static const struct reg_default src4xxx_reg_defaults[] = { + { SRC4XXX_PWR_RST_01, 0x00 }, /* all powered down intially */ + { SRC4XXX_PORTA_CTL_03, 0x00 }, + { SRC4XXX_PORTA_CTL_04, 0x00 }, + { SRC4XXX_PORTB_CTL_05, 0x00 }, + { SRC4XXX_PORTB_CTL_06, 0x00 }, + { SRC4XXX_TX_CTL_07, 0x00 }, + { SRC4XXX_TX_CTL_08, 0x00 }, + { SRC4XXX_TX_CTL_09, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MSK_0B, 0x00 }, + { SRC4XXX_SRC_DIT_IRQ_MODE_0C, 0x00 }, + { SRC4XXX_RCV_CTL_0D, 0x00 }, + { SRC4XXX_RCV_CTL_0E, 0x00 }, + { SRC4XXX_RCV_PLL_0F, 0x00 }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_10, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RCV_PLL_11, 0xff }, /* not spec. in the datasheet */ + { SRC4XXX_RVC_IRQ_MSK_16, 0x00 }, + { SRC4XXX_RVC_IRQ_MSK_17, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_18, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_19, 0x00 }, + { SRC4XXX_RVC_IRQ_MODE_1A, 0x00 }, + { SRC4XXX_GPIO_1_1B, 0x00 }, + { SRC4XXX_GPIO_2_1C, 0x00 }, + { SRC4XXX_GPIO_3_1D, 0x00 }, + { SRC4XXX_GPIO_4_1E, 0x00 }, + { SRC4XXX_SCR_CTL_2D, 0x00 }, + { SRC4XXX_SCR_CTL_2E, 0x00 }, + { SRC4XXX_SCR_CTL_2F, 0x00 }, + { SRC4XXX_SCR_CTL_30, 0x00 }, + { SRC4XXX_SCR_CTL_31, 0x00 }, +}; + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)) +{ + struct src4xxx *src4xxx; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + src4xxx = devm_kzalloc(dev, sizeof(*src4xxx), GFP_KERNEL); + if (!src4xxx) + return -ENOMEM; + + src4xxx->regmap = regmap; + src4xxx->dev = dev; + src4xxx->mclk_hz = 0; /* mclk has not been configured yet */ + dev_set_drvdata(dev, src4xxx); + + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_RESET); + if (ret < 0) + dev_err(dev, "Failed to issue reset: %d\n", ret); + usleep_range(1, 500); /* sleep for more then 500 ns */ + ret = regmap_write(regmap, SRC4XXX_PWR_RST_01, SRC4XXX_POWER_DOWN); + if (ret < 0) + dev_err(dev, "Failed to decommission reset: %d\n", ret); + usleep_range(500, 1000); /* sleep for 500 us or more */ + + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_PWR_RST_01, + SRC4XXX_POWER_ENABLE, SRC4XXX_POWER_ENABLE); + if (ret < 0) + dev_err(dev, "Failed to port A and B : %d\n", ret); + + /* enable the BTI and TSLIP interrupts */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_SRC_DIT_IRQ_MSK_0B, + SRC4XXX_SRC_BTI_EN | SRC4XXX_SRC_TSLIP_EN, + SRC4XXX_SRC_BTI_EN | SRC4XXX_SRC_TSLIP_EN); + if (ret < 0) + dev_err(dev, + "Failed to enable BTI and TSLIP interrupts : %d\n", + ret); + + /* set receiver to use master clock (rcv mclk is most likely jittery) */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0D, + SRC4XXX_RXCLK_MCLK, SRC4XXX_RXCLK_MCLK); + if (ret < 0) + dev_err(dev, + "Failed to enable mclk as the PLL1 DIR reference : %d\n", ret); + + /* default to leaving the PLL2 running on loss of lock, divide by 8 */ + ret = regmap_update_bits(src4xxx->regmap, SRC4XXX_RCV_CTL_0E, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL, + SRC4XXX_PLL2_DIV_8 | SRC4XXX_REC_MCLK_EN | SRC4XXX_PLL2_LOL); + if (ret < 0) + dev_err(dev, "Failed to enable mclk rec and div : %d\n", ret); + + ret = devm_snd_soc_register_component(dev, &src4xxx_driver, + src4xxx_dai_driver, ARRAY_SIZE(src4xxx_dai_driver)); + if (ret == 0) + dev_info(dev, "src4392 probe ok %d\n", ret); + return ret; +} +EXPORT_SYMBOL(src4xxx_probe); + +void src4xxx_remove(struct device *dev) +{ + dev_info(dev, "__func__\n"); +} +EXPORT_SYMBOL(src4xxx_remove); + +static bool src4xxx_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case SRC4XXX_RES_00: + case SRC4XXX_GLOBAL_ITR_STS_02: + case SRC4XXX_SRC_DIT_STS_0A: + case SRC4XXX_NON_AUDIO_D_12: + case SRC4XXX_RVC_STS_13: + case SRC4XXX_RVC_STS_14: + case SRC4XXX_RVC_STS_15: + case SRC4XXX_SUB_CODE_1F: + case SRC4XXX_SUB_CODE_20: + case SRC4XXX_SUB_CODE_21: + case SRC4XXX_SUB_CODE_22: + case SRC4XXX_SUB_CODE_23: + case SRC4XXX_SUB_CODE_24: + case SRC4XXX_SUB_CODE_25: + case SRC4XXX_SUB_CODE_26: + case SRC4XXX_SUB_CODE_27: + case SRC4XXX_SUB_CODE_28: + case SRC4XXX_PC_PREAMBLE_HI_29: + case SRC4XXX_PC_PREAMBLE_LO_2A: + case SRC4XXX_PD_PREAMBLE_HI_2B: + case SRC4XXX_PC_PREAMBLE_LO_2C: + case SRC4XXX_IO_RATIO_32: + case SRC4XXX_IO_RATIO_33: + return true; + } + + if (reg > SRC4XXX_IO_RATIO_33 && reg < SRC4XXX_PAGE_SEL_7F) + return true; + + return false; +} + +const struct regmap_config src4xxx_regmap_config = { + .val_bits = 8, + .reg_bits = 8, + .max_register = SRC4XXX_IO_RATIO_33, + + .reg_defaults = src4xxx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(src4xxx_reg_defaults), + .volatile_reg = src4xxx_volatile_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(src4xxx_regmap_config); + +MODULE_DESCRIPTION("ASoC SRC4XXX CODEC driver"); +MODULE_AUTHOR("Matt Flax <flatmax(a)flatmax.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/src4xxx.h b/sound/soc/codecs/src4xxx.h new file mode 100644 index 000000000000..5e7b468bc22e --- /dev/null +++ b/sound/soc/codecs/src4xxx.h @@ -0,0 +1,115 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * src4xxx.h -- SRC4XXX ALSA SoC audio driver + * + * Copyright 2021-2022 Deqx Pty Ltd + * Author: Matt R, Fkax <flatmax(a)flatmax.com> + */ + +#ifndef __SRC4XXX_H__ +#define __SRC4XXX_H__ + +#define SRC4XXX_RES_00 0x00 +#define SRC4XXX_PWR_RST_01 0x01 +#define SRC4XXX_RESET 0x80 +#define SRC4XXX_POWER_DOWN 0x00 +#define SRC4XXX_POWER_ENABLE 0x20 +#define SRC4XXX_ENABLE_SRC 0x1 +#define SRC4XXX_ENABLE_SRC_SHIFT 0 +#define SRC4XXX_ENABLE_DIR 0x2 +#define SRC4XXX_ENABLE_DIR_SHIFT 1 +#define SRC4XXX_ENABLE_DIT 0x4 +#define SRC4XXX_ENABLE_DIT_SHIFT 2 +#define SRC4XXX_ENABLE_PORT_B 0x8 +#define SRC4XXX_ENABLE_PORT_B_SHIFT 3 +#define SRC4XXX_ENABLE_PORT_A 0x10 +#define SRC4XXX_ENABLE_PORT_A_SHIFT 4 + +#define SRC4XXX_PORTA_CTL_03 0x03 +#define SRC4XXX_BUS_MASTER 0x8 +#define SRC4XXX_BUS_LEFT_J 0x0 +#define SRC4XXX_BUS_I2S 0x1 +#define SRC4XXX_BUS_RIGHT_J_16 0x4 +#define SRC4XXX_BUS_RIGHT_J_18 0x5 +#define SRC4XXX_BUS_RIGHT_J_20 0x6 +#define SRC4XXX_BUS_RIGHT_J_24 0x7 +#define SRC4XXX_BUS_FMT_MS_MASK 0xf + +#define SRC4XXX_PORTA_CTL_04 0x04 +#define SRC4XXX_MCLK_DIV_MASK 0x3 + +#define SRC4XXX_BUS_FMT(id) (SRC4XXX_PORTA_CTL_03+2*id) +#define SRC4XXX_BUS_CLK(id) (SRC4XXX_PORTA_CTL_04+2*id) + +#define SRC4XXX_PORTB_CTL_05 0x05 +#define SRC4XXX_PORTB_CTL_06 0x06 + +#define SRC4XXX_TX_CTL_07 0x07 +#define SRC4XXX_TX_MCLK_DIV_MASK 0x60 +#define SRC4XXX_TX_MCLK_DIV_SHIFT 5 + +#define SRC4XXX_TX_CTL_08 0x08 +#define SRC4XXX_TX_CTL_09 0x09 +#define SRC4XXX_SRC_DIT_IRQ_MSK_0B 0x0B +#define SRC4XXX_SRC_BTI_EN 0x01 +#define SRC4XXX_SRC_TSLIP_EN 0x02 +#define SRC4XXX_SRC_DIT_IRQ_MODE_0C 0x0C +#define SRC4XXX_RCV_CTL_0D 0x0D +#define SRC4XXX_RXCLK_RXCKI 0x0 +#define SRC4XXX_RXCLK_MCLK 0x8 +#define SRC4XXX_RCV_CTL_0E 0x0E +#define SRC4XXX_REC_MCLK_EN 0x1 +#define SRC4XXX_PLL2_DIV_0 (0x0<<1) +#define SRC4XXX_PLL2_DIV_2 (0x1<<1) +#define SRC4XXX_PLL2_DIV_4 (0x2<<1) +#define SRC4XXX_PLL2_DIV_8 (0x3<<1) +#define SRC4XXX_PLL2_LOL 0x8 +#define SRC4XXX_RCV_PLL_0F 0x0F +#define SRC4XXX_RCV_PLL_10 0x10 +#define SRC4XXX_RCV_PLL_11 0x11 +#define SRC4XXX_RVC_IRQ_MSK_16 0x16 +#define SRC4XXX_RVC_IRQ_MSK_17 0x17 +#define SRC4XXX_RVC_IRQ_MODE_18 0x18 +#define SRC4XXX_RVC_IRQ_MODE_19 0x19 +#define SRC4XXX_RVC_IRQ_MODE_1A 0x1A +#define SRC4XXX_GPIO_1_1B 0x1B +#define SRC4XXX_GPIO_2_1C 0x1C +#define SRC4XXX_GPIO_3_1D 0x1D +#define SRC4XXX_GPIO_4_1E 0x1E +#define SRC4XXX_SCR_CTL_2D 0x2D +#define SRC4XXX_SCR_CTL_2E 0x2E +#define SRC4XXX_SCR_CTL_2F 0x2F +#define SRC4XXX_SCR_CTL_30 0x30 +#define SRC4XXX_SCR_CTL_31 0x31 +#define SRC4XXX_PAGE_SEL_7F 0x7F + +// read only registers +#define SRC4XXX_GLOBAL_ITR_STS_02 0x02 +#define SRC4XXX_SRC_DIT_STS_0A 0x0A +#define SRC4XXX_NON_AUDIO_D_12 0x12 +#define SRC4XXX_RVC_STS_13 0x13 +#define SRC4XXX_RVC_STS_14 0x14 +#define SRC4XXX_RVC_STS_15 0x15 +#define SRC4XXX_SUB_CODE_1F 0x1F +#define SRC4XXX_SUB_CODE_20 0x20 +#define SRC4XXX_SUB_CODE_21 0x21 +#define SRC4XXX_SUB_CODE_22 0x22 +#define SRC4XXX_SUB_CODE_23 0x23 +#define SRC4XXX_SUB_CODE_24 0x24 +#define SRC4XXX_SUB_CODE_25 0x25 +#define SRC4XXX_SUB_CODE_26 0x26 +#define SRC4XXX_SUB_CODE_27 0x27 +#define SRC4XXX_SUB_CODE_28 0x28 +#define SRC4XXX_PC_PREAMBLE_HI_29 0x29 +#define SRC4XXX_PC_PREAMBLE_LO_2A 0x2A +#define SRC4XXX_PD_PREAMBLE_HI_2B 0x2B +#define SRC4XXX_PC_PREAMBLE_LO_2C 0x2C +#define SRC4XXX_IO_RATIO_32 0x32 +#define SRC4XXX_IO_RATIO_33 0x33 + +int src4xxx_probe(struct device *dev, struct regmap *regmap, + void (*switch_mode)(struct device *dev)); +void src4xxx_remove(struct device *dev); +extern const struct regmap_config src4xxx_regmap_config; + +#endif /* __SRC4XXX_H__ */ -- 2.34.1

3 2

ASUS ROG Zenith II Extreme UCM2 profile
by GitHub pull_request - opened 08 Aug '22

08 Aug '22

alsa-project/alsa-ucm-conf pull request #197 was opened from nabajour: Hi, I'm trying to get sound working on my ASUS ROG Zenith II Extreme motherboard. There is a [kernel patch](https://bugzilla.kernel.org/show_bug.cgi?id=211005) that I'm using for a while, that seems to get most ALSA stuff working. It kinda also works with the default ProAudio ACP profile (minus some case where it seems to loose synch after a while. But recording works). Now, I'm trying to get it to work correctly with my Debian unstable desktop, using PipeWire with the PA tools (mostly pavucontrol). Mostly, the routing of the inputs for capture seem to not work. It's always routing the input from `hw:0,0`, which is `Line In`. I tried making a UCM2 profile, and got some basic stuff working. but there are some things where I'm stuck. What I do is edit the UCM files, and then reload PW and WP with `systemctl --user restart pipewire pipewire-pulse wireplumber`. Then, I check with `pavucontrol` what devices appear. If there is a syntax error, it falls back to `Off` profile, or `ProAudio` profile, which was an experiment from me to see if I could just have the same pass-through behaviour as the ACP pass-through. I don't know if there is a better way to debug UCM2 profile loading, this approach leaves me pretty blind on what failed. Here are the issues I currently have: 1. surround input channels are mixed up compared to alsa channels (testing with `speaker-test` output, sound goes to wrong outputs with pw. 2. Can't get SPDIF to work. When I uncomment the SPDIF block, it doesn't load the profile. 3. Mic 1 (the rear mic) appears and reacts correctly to jack plug/unplug. But recording seems to take the input from `hw:0,0` instead of `hw:0,1` 4. uncommenting the Line In or the Mic2 (front mic) block fails. It doesn't load the profile. I don't know if the issue is still on the ALSA driver side or on the PW/PA config. I'm not a kernel programmer and don't know much about ALSA and PW, so any pointer here is welcome. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/197 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/197.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: adau1761: Reset codec at probe and shutdown
by Niklas Carlsson 08 Aug '22

08 Aug '22

From: Niklas Carlsson <niklasc(a)axis.com> ADAU1X61 doesn't have a reset pin, which leads to the codec retaining the register values after the device has been rebooted without a power cycle. This causes the driver to be out of sync with the codec. Simulating a hardware reset ensures that the codec is in a defined state. Link: https://lore.kernel.org/all/alpine.DEB.2.02.1603291544160.3594@lnxricardw1.… Link: https://lore.kernel.org/all/alpine.DEB.2.02.1604211547540.31333@lnxricardw1… Signed-off-by: Niklas Carlsson <niklasc(a)axis.com> --- sound/soc/codecs/adau1761-i2c.c | 6 +++++ sound/soc/codecs/adau1761-spi.c | 6 +++++ sound/soc/codecs/adau1761.c | 43 +++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1761.h | 1 + 4 files changed, 56 insertions(+) diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 0683caf86aea..23bffbd74469 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -36,6 +36,11 @@ static int adau1761_i2c_remove(struct i2c_client *client) return 0; } +static void adau1761_i2c_shutdown(struct i2c_client *client) +{ + adau1761_shutdown(&client->dev); +} + static const struct i2c_device_id adau1761_i2c_ids[] = { { "adau1361", ADAU1361 }, { "adau1461", ADAU1761 }, @@ -63,6 +68,7 @@ static struct i2c_driver adau1761_i2c_driver = { }, .probe_new = adau1761_i2c_probe, .remove = adau1761_i2c_remove, + .shutdown = adau1761_i2c_shutdown, .id_table = adau1761_i2c_ids, }; module_i2c_driver(adau1761_i2c_driver); diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index 7c9242c2ff94..5bbbc3d3b5be 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -50,6 +50,11 @@ static void adau1761_spi_remove(struct spi_device *spi) adau17x1_remove(&spi->dev); } +static void adau1761_spi_shutdown(struct spi_device *spi) +{ + adau1761_shutdown(&spi->dev); +} + static const struct spi_device_id adau1761_spi_id[] = { { "adau1361", ADAU1361 }, { "adau1461", ADAU1761 }, @@ -77,6 +82,7 @@ static struct spi_driver adau1761_spi_driver = { }, .probe = adau1761_spi_probe, .remove = adau1761_spi_remove, + .shutdown = adau1761_spi_shutdown, .id_table = adau1761_spi_id, }; module_spi_driver(adau1761_spi_driver); diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c index 8f887227981f..ac544e1e8dc7 100644 --- a/sound/soc/codecs/adau1761.c +++ b/sound/soc/codecs/adau1761.c @@ -974,6 +974,39 @@ static struct snd_soc_dai_driver adau1761_dai_driver = { .ops = &adau17x1_dai_ops, }; +static void adau1761_reset(struct regmap *regmap) +{ + size_t i; + + /* + * Handle the lack of a reset pin in ADAU1X61 by manually writing all + * registers to their default values. + * + * Steps for performing the reset: + * 1) Make sure that the cache is marked as dirty by writing all + * default values directly to the cache. + * + * 2) Enable the core clock which is needed for writing all registers + * except CLOCK_CONTROL. + * + * 3) Use regcache_sync() for synchronizing the dirty cache back to + * the hardware. + */ + + regcache_cache_only(regmap, true); + for (i = 0; i < ARRAY_SIZE(adau1761_reg_defaults); i++) + regmap_write(regmap, adau1761_reg_defaults[i].reg, + adau1761_reg_defaults[i].def); + regcache_cache_only(regmap, false); + + regmap_update_bits(regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN); + regcache_sync(regmap); + regmap_update_bits(regmap, ADAU17X1_CLOCK_CONTROL, + ADAU17X1_CLOCK_CONTROL_SYSCLK_EN, 0); +} + int adau1761_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev)) { @@ -997,6 +1030,8 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, if (ret) return ret; + adau1761_reset(regmap); + /* Enable cache only mode as we could miss writes before bias level * reaches standby and the core clock is enabled */ regcache_cache_only(regmap, true); @@ -1006,6 +1041,14 @@ int adau1761_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau1761_probe); +void adau1761_shutdown(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + adau1761_reset(adau->regmap); +} +EXPORT_SYMBOL_GPL(adau1761_shutdown); + const struct regmap_config adau1761_regmap_config = { .val_bits = 8, .reg_bits = 16, diff --git a/sound/soc/codecs/adau1761.h b/sound/soc/codecs/adau1761.h index 7beabf448ad1..98d698ebde27 100644 --- a/sound/soc/codecs/adau1761.h +++ b/sound/soc/codecs/adau1761.h @@ -16,6 +16,7 @@ struct device; int adau1761_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev)); +void adau1761_shutdown(struct device *dev); extern const struct regmap_config adau1761_regmap_config; -- 2.30.2

2 1

[PATCH] ALSA: hda/conexant: Add quirk for LENOVO 20149 Notebook model
by Meng Tang 08 Aug '22

08 Aug '22

There is another LENOVO 20149 (Type1Sku0) Notebook model with CX20590, the device PCI SSID is 17aa:3977, which headphones are not responding, that requires the quirk CXT_PINCFG_LENOVO_NOTEBOOK. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_conexant.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 83ae21a01bbf..7b1a30a551f6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -222,6 +222,7 @@ enum { CXT_PINCFG_LEMOTE_A1205, CXT_PINCFG_COMPAQ_CQ60, CXT_FIXUP_STEREO_DMIC, + CXT_PINCFG_LENOVO_NOTEBOOK, CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, @@ -772,6 +773,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, }, + [CXT_PINCFG_LENOVO_NOTEBOOK] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x05d71030 }, + { } + }, + .chain_id = CXT_FIXUP_STEREO_DMIC, + }, [CXT_FIXUP_INC_MIC_BOOST] = { .type = HDA_FIXUP_FUNC, .v.func = cxt5066_increase_mic_boost, @@ -971,7 +980,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3905, "Lenovo G50-30", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x390b, "Lenovo G50-80", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), - SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_PINCFG_LENOVO_NOTEBOOK), SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo G50-70", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), -- 2.20.1

2 1

[PATCH] ASoC: Intel: kbl_rt5663_rt5514_max98927: Simplify clk_get() usage
by Christophe JAILLET 07 Aug '22

07 Aug '22

If clk_get() returns -ENOENT, there is no need to defer the driver, -ENOENT will be returned the same for each retries. So, return the error code directly instead of -EPROBE_DEFER. Remove this special case and use dev_err_probe() to simplify code. It will also be less verbose if the clk is really deferred. Fixes: 47cbea216281 ("ASoC: Intel: eve: Enable mclk and ssp sclk early") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- This is based on my understanding of clk_get(). Review with care. Not sure the Fixes tag is needed. The patch does not fix anything. If devm_clk_get() returns -ENOENT, it will just loop several time until the framework gives up. If it returns -EPROBE_DEFER, this case is already handled by the "return ret;" So this patch should be a no-op, just a clean-up. --- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 31 ++++--------------- 1 file changed, 6 insertions(+), 25 deletions(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 2c79fca57b19..e23001ca2d04 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -798,7 +798,6 @@ static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; struct snd_soc_acpi_mach *mach; - int ret; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -815,32 +814,14 @@ static int kabylake_audio_probe(struct platform_device *pdev) &constraints_dmic_2ch : &constraints_dmic_channels; ctx->mclk = devm_clk_get(&pdev->dev, "ssp1_mclk"); - if (IS_ERR(ctx->mclk)) { - ret = PTR_ERR(ctx->mclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_mclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_mclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->mclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->mclk), + "Failed to get ssp1_mclk\n"); ctx->sclk = devm_clk_get(&pdev->dev, "ssp1_sclk"); - if (IS_ERR(ctx->sclk)) { - ret = PTR_ERR(ctx->sclk); - if (ret == -ENOENT) { - dev_info(&pdev->dev, - "Failed to get ssp1_sclk, defer probe\n"); - return -EPROBE_DEFER; - } - - dev_err(&pdev->dev, "Failed to get ssp1_sclk with err:%d\n", - ret); - return ret; - } + if (IS_ERR(ctx->sclk)) + return dev_err_probe(&pdev->dev, PTR_ERR(ctx->sclk), + "Failed to get ssp1_sclk\n"); return devm_snd_soc_register_card(&pdev->dev, &kabylake_audio_card); } -- 2.34.1

1 0

[PATCH] ASoC: Intel: fix sof_es8336 probe
by Pierre-Louis Bossart 05 Aug '22

05 Aug '22

Changes to add HDMI capture support broke the machine driver probe for all other platforms. The commit listed in the Fixes tag added a board_id descriptor but didn't add the default name for the sof_essx8336 machine driver. Add the missing entry and remove the now-useless platform driver alias. Reported-by: Eugene J Markow <ejmarkow(a)yahoo.com> BugLink: https://github.com/thesofproject/linux/issues/3336 Fixes: 86b1959a2ccb ("ASoC: Intel: sof_es8336: add support for HDMI_In capture") Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/sof_es8336.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index c7f33c89588e..606cc3242a60 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -759,6 +759,9 @@ static int sof_es8336_remove(struct platform_device *pdev) } static const struct platform_device_id board_ids[] = { + { + .name = "sof-essx8336", /* default quirk == 0 */ + }, { .name = "adl_es83x6_c1_h02", .driver_data = (kernel_ulong_t)(SOF_ES8336_SSP_CODEC(1) | @@ -786,5 +789,4 @@ module_platform_driver(sof_es8336_driver); MODULE_DESCRIPTION("ASoC Intel(R) SOF + ES8336 Machine driver"); MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:sof-essx8336"); MODULE_IMPORT_NS(SND_SOC_INTEL_HDA_DSP_COMMON); -- 2.34.1

2 1

[PATCH resent] ASoC: DPCM: Don't pick up BE without substream
by Takashi Iwai 05 Aug '22

05 Aug '22

When DPCM tries to add valid BE connections at dpcm_add_paths(), it doesn't check whether the picked BE actually supports for the given stream direction. Due to that, when an asymmetric BE stream is present, it picks up wrongly and this may result in a NULL dereference at a later point where the code assumes the existence of a corresponding BE substream. This patch adds the check for the presence of the substream for the target BE for avoiding the problem above. Note that we have already some fix for non-existing BE substream at commit 6246f283d5e0 ("ASoC: dpcm: skip missing substream while applying symmetry"). But the code path we've hit recently is rather happening before the previous fix. So this patch tries to fix at picking up a BE instead of parsing BE lists. Fixes: bbf7d3b1c4f4 ("ASoC: soc-pcm: align BE 'atomicity' with that of the FE") Reported-by: Alex Natalsson <harmoniesworlds(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Link: https://lore.kernel.org/r/CADs9LoPZH_D+eJ9qjTxSLE5jGyhKsjMN7g2NighZ16biVxsy… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- Seems that the previous submission was overlooked since I forgot to put Cc to Mark. Resubmitted... sound/soc/soc-pcm.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index a827cc3c158a..0c1de5624842 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1318,6 +1318,9 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + if (!snd_soc_dpcm_get_substream(be, stream)) + continue; + for_each_rtd_dais(be, i, dai) { w = snd_soc_dai_get_widget(dai, stream); -- 2.35.3

2 1

[PATCH 0/3] ASoC: Fix theoretical buffer overflow by snprintf()
by Takashi Iwai 05 Aug '22

05 Aug '22

Hi, this is a patch series to paper over the theoretical buffer overflow that might be caused by snprintf(). snprintf() is notorious for its behavior and the usage of a safer version, scnprintf(), is recommended. Takashi === Takashi Iwai (3): ASoC: Intel: avs: Fix potential buffer overflow by snprintf() ASoC: SOF: debug: Fix potential buffer overflow by snprintf() ASoC: SOF: Intel: hda: Fix potential buffer overflow by snprintf() sound/soc/intel/avs/pcm.c | 4 ++-- sound/soc/sof/debug.c | 6 +++--- sound/soc/sof/intel/hda.c | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) -- 2.35.3

3 5

[PATCH resend] ASoC: SOF: ipc3-topology: Fix clang -Wformat warning
by Justin Stitt 05 Aug '22

05 Aug '22

When building with Clang we encounter these warnings: | sound/soc/sof/ipc3-topology.c:2343:4: error: format specifies type | 'unsigned char' but the argument has type 'int' [-Werror,-Wformat] | SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); | ^~~~~~~~~~~~~~~^~~~~~~~~~~~~~~^~~~~~~~~~~~~ Use correct format specifier `%d` since args are of type int. Link: https://github.com/ClangBuiltLinux/linux/issues/378 Reported-by: Nathan Chancellor <nathan(a)kernel.org> Suggested-by: Nathan Chancellor <nathan(a)kernel.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- Note: resending with the subject corrected, this patch fixes a build warning that will break the Clang build when CONFIG_WERROR=y. Original patch: https://lore.kernel.org/all/20220721211218.4039288-1-justinstitt@google.com/ Reported by Nathan here: https://lore.kernel.org/all/YtmrCJjQrSbv8Aj1@dev-arch.thelio-3990X/ sound/soc/sof/ipc3-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index b2cc046b9f60..65923e7a5976 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2338,7 +2338,7 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, } dev_info(scomp->dev, - "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n", + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", man->priv.data[0], man->priv.data[1], man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); -- 2.37.1.359.gd136c6c3e2-goog

4 3

[PATCH v2] ASoC: sh: rz-ssi: Improve error handling in rz_ssi_probe() error path
by Biju Das 05 Aug '22

05 Aug '22

We usually do cleanup in reverse order of init. Currently in case of error rz_ssi_release_dma_channels() done in the reverse order. This patch improves error handling in rz_ssi_probe() error path. While at it, use "goto cleanup" style to reduce code duplication. Reported-by: Pavel Machek <pavel(a)denx.de> Signed-off-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- v1->v2: Replaced the label err_dma->err_reset. --- sound/soc/sh/rz-ssi.c | 26 +++++++++++++++----------- 1 file changed, 15 insertions(+), 11 deletions(-) diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index 0d0594a0e4f6..7ace0c0db5b1 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1017,32 +1017,36 @@ static int rz_ssi_probe(struct platform_device *pdev) ssi->rstc = devm_reset_control_get_exclusive(&pdev->dev, NULL); if (IS_ERR(ssi->rstc)) { - rz_ssi_release_dma_channels(ssi); - return PTR_ERR(ssi->rstc); + ret = PTR_ERR(ssi->rstc); + goto err_reset; } reset_control_deassert(ssi->rstc); pm_runtime_enable(&pdev->dev); ret = pm_runtime_resume_and_get(&pdev->dev); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); - return dev_err_probe(ssi->dev, ret, "pm_runtime_resume_and_get failed\n"); + dev_err(&pdev->dev, "pm_runtime_resume_and_get failed\n"); + goto err_pm; } ret = devm_snd_soc_register_component(&pdev->dev, &rz_ssi_soc_component, rz_ssi_soc_dai, ARRAY_SIZE(rz_ssi_soc_dai)); if (ret < 0) { - rz_ssi_release_dma_channels(ssi); - - pm_runtime_put(ssi->dev); - pm_runtime_disable(ssi->dev); - reset_control_assert(ssi->rstc); dev_err(&pdev->dev, "failed to register snd component\n"); + goto err_snd_soc; } + return 0; + +err_snd_soc: + pm_runtime_put(ssi->dev); +err_pm: + pm_runtime_disable(ssi->dev); + reset_control_assert(ssi->rstc); +err_reset: + rz_ssi_release_dma_channels(ssi); + return ret; } -- 2.25.1

2 1

[PATCH] ALSA: ice1712: remove redundant assignment to new
by Colin Ian King 05 Aug '22

05 Aug '22

The variable new is initialized with a value but it is never read. It is being re-assigned a new value in every case path in the following switch statement. The assignment is redundant and can be removed. Cleans up clang scan build warning: sound/pci/ice1712/quartet.c:569:8: warning: Although the value stored to 'new' is used in the enclosing expression, the value is never actually read from 'new' [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/ice1712/quartet.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/quartet.c b/sound/pci/ice1712/quartet.c index 0dfa093f7dca..20b3e8f94719 100644 --- a/sound/pci/ice1712/quartet.c +++ b/sound/pci/ice1712/quartet.c @@ -566,7 +566,7 @@ static int qtet_ain12_sw_put(struct snd_kcontrol *kcontrol, { struct snd_ice1712 *ice = snd_kcontrol_chip(kcontrol); unsigned int old, new, tmp, masked_old; - old = new = get_scr(ice); + old = get_scr(ice); masked_old = old & (SCR_AIN12_SEL1 | SCR_AIN12_SEL0); tmp = ucontrol->value.integer.value[0]; if (tmp == 2) -- 2.35.3

2 1

[PATCH v2] ALSA: hda/realtek: Add quirk for another Asus K42JZ model
by Meng Tang 05 Aug '22

05 Aug '22

There is another Asus K42JZ model with the PCI SSID 1043:1313 that requires the quirk ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 383a814b8539..4461237a3474 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6842,6 +6842,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, + ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -7427,6 +7428,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, }, + [ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a110f0 }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -9124,6 +9134,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.20.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for another Asus K42JZ model
by Meng Tang 05 Aug '22

05 Aug '22

There is another Asus K42JZ model with the PCI SSID 1043:1313 that requires the quirk ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE. Add the corresponding entry to the quirk table. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 383a814b8539..4461237a3474 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6842,6 +6842,7 @@ enum { ALC269_FIXUP_LIMIT_INT_MIC_BOOST, ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A, + ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, @@ -7427,6 +7428,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269VB_FIXUP_ASUS_ZENBOOK, }, + [ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a110f0 }, /* use as headset mic */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + } [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -9124,6 +9134,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12a0, "ASUS X441UV", ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.20.1

1 0

[PATCH] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 05 Aug '22

05 Aug '22

In commit <c3ecef21c3f26> ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ffc24afb5a7a..f0602077b385 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1054,7 +1054,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

1 0

[PATCH v2] ASoC: amd: acp: Add TDM support for acp i2s stream
by Venkata Prasad Potturu 05 Aug '22

05 Aug '22

Add callback and code changes to enable ACP I2S controller in TDM mode. Add new fields in acp_stream and acp_dev_data struct to configure tdm related registers for ACP i2s controllers. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Depends on:- -- https://patchwork.kernel.org/project/alsa-devel/patch/20220804072556.601396… -- https://patchwork.kernel.org/project/alsa-devel/patch/20220801063501.51439-… Changes since v1: -- Add spin lock where linked list is referring. sound/soc/amd/acp/acp-i2s.c | 80 ++++++++++++++++++++++++++++++++++++- sound/soc/amd/acp/amd.h | 12 ++++++ 2 files changed, 91 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 393f729ef561..ac416572db0d 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -25,6 +25,65 @@ #define DRV_NAME "acp_i2s_playcap" +static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + int mode; + + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (mode) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = TDM_DISABLE; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = TDM_ENABLE; + break; + default: + return -EINVAL; + } + return 0; +} + +static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask, + int slots, int slot_width) +{ + struct device *dev = dai->component->dev; + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); + struct acp_stream *stream; + int slot_len; + + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + dev_err(dev, "Unsupported bitdepth %d\n", slot_width); + return -EINVAL; + } + + spin_lock_irq(&adata->acp_lock); + list_for_each_entry(stream, &adata->stream_list, list) { + if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + adata->tdm_tx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + adata->tdm_rx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + } + spin_unlock_irq(&adata->acp_lock); + return 0; +} + static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -33,7 +92,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ struct acp_resource *rsrc; u32 val; u32 xfer_resolution; - u32 reg_val; + u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; adata = snd_soc_dai_get_drvdata(dai); @@ -62,12 +121,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_ITER; + fmt_reg = ACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_ITER; + fmt_reg = ACP_I2STDM_TXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_ITER; + fmt_reg = ACP_HSTDM_TXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -77,12 +139,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_IRER; + fmt_reg = ACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_IRER; + fmt_reg = ACP_I2STDM_RXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_IRER; + fmt_reg = ACP_HSTDM_RXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -95,6 +160,16 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ val = val | (xfer_resolution << 3); writel(val, adata->acp_base + reg_val); + if (adata->tdm_mode) { + val = readl(adata->acp_base + reg_val); + writel(val | BIT(1), adata->acp_base + reg_val); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tdm_fmt = adata->tdm_tx_fmt[dai->driver->id - 1]; + else + tdm_fmt = adata->tdm_rx_fmt[dai->driver->id - 1]; + writel(tdm_fmt, adata->acp_base + fmt_reg); + } + if (rsrc->soc_mclk) { switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -443,6 +518,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d stream->id = dai->driver->id + dir; stream->dai_id = dai->driver->id; stream->irq_bit = irq_bit; + stream->dir = substream->stream; return 0; } @@ -452,6 +528,8 @@ const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops = { .hw_params = acp_i2s_hwparams, .prepare = acp_i2s_prepare, .trigger = acp_i2s_trigger, + .set_fmt = acp_i2s_set_fmt, + .set_tdm_slot = acp_i2s_set_tdm_slot, }; EXPORT_SYMBOL_NS_GPL(asoc_acp_cpu_dai_ops, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index be8bb8247c4e..5f2119f42271 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -84,6 +84,14 @@ #define ACP_MAX_STREAM 8 +#define TDM_ENABLE 1 +#define TDM_DISABLE 0 + +#define SLOT_WIDTH_8 0x8 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + struct acp_chip_info { char *name; /* Platform name */ unsigned int acp_rev; /* ACP Revision id */ @@ -96,6 +104,7 @@ struct acp_stream { int irq_bit; int dai_id; int id; + int dir; u64 bytescount; u32 reg_offset; u32 pte_offset; @@ -120,6 +129,7 @@ struct acp_dev_data { void __iomem *acp_base; unsigned int i2s_irq; + bool tdm_mode; /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; int num_dai; @@ -134,6 +144,8 @@ struct acp_dev_data { u32 lrclk_div; struct acp_resource *rsrc; + u32 tdm_tx_fmt[3]; + u32 tdm_rx_fmt[3]; }; union acp_i2stdm_mstrclkgen { -- 2.25.1

1 0

[PATCH] ASoC: fsl_sai: fix incorrect mclk number in error message
by Pieterjan Camerlynck 05 Aug '22

05 Aug '22

In commit <c3ecef21c3f26> ("ASoC: fsl_sai: add sai master mode support") the loop was changed to start iterating from 1 instead of 0. The error message however was not updated, reporting the wrong clock to the user. Signed-off-by: Pieterjan Camerlynck <pieterjan.camerlynck(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ffc24afb5a7a..f0602077b385 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1054,7 +1054,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", - i + 1, PTR_ERR(sai->mclk_clk[i])); + i, PTR_ERR(sai->mclk_clk[i])); sai->mclk_clk[i] = NULL; } } -- 2.25.1

2 2

[PATCH] ASoC: fsl_asrc_dma: fully initialize structs
by Sascha Hauer 05 Aug '22

05 Aug '22

The driver uses two statically ininitialized struct dma_slave_config, but only one of them is initialized to zero. Initialize config_be to zero as well to make sure that no fields are filled with random values. Let the compiler do this instead of explicitly calling memset() which makes it easier to read. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_asrc_dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_asrc_dma.c b/sound/soc/fsl/fsl_asrc_dma.c index 5038faf035cba..fec56dbb64e60 100644 --- a/sound/soc/fsl/fsl_asrc_dma.c +++ b/sound/soc/fsl/fsl_asrc_dma.c @@ -138,7 +138,7 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, struct dma_chan *tmp_chan = NULL, *be_chan = NULL; struct snd_soc_component *component_be = NULL; struct fsl_asrc *asrc = pair->asrc; - struct dma_slave_config config_fe, config_be; + struct dma_slave_config config_fe = {}, config_be = {}; enum asrc_pair_index index = pair->index; struct device *dev = component->dev; struct device_node *of_dma_node; @@ -181,7 +181,6 @@ static int fsl_asrc_dma_hw_params(struct snd_soc_component *component, return -EINVAL; } - memset(&config_fe, 0, sizeof(config_fe)); ret = snd_dmaengine_pcm_prepare_slave_config(substream, params, &config_fe); if (ret) { dev_err(dev, "failed to prepare DMA config for Front-End\n"); -- 2.30.2

3 3

[PATCH 0/2] ASoC: SOF: Introduce of_machine_select
by Chunxu Li 04 Aug '22

04 Aug '22

From: "chunxu.li" <chunxu.li(a)mediatek.com> In these patches, we introduce of_machine_select for SOF Chunxu Li (2): ASoC: SOF: Introduce optional callback of_machine_select ASoC: SOF: mediatek: Add .of_machine_select field for mt8186 include/sound/sof.h | 2 ++ sound/soc/sof/mediatek/mt8186/mt8186.c | 11 +++++++++++ sound/soc/sof/ops.h | 9 +++++++++ sound/soc/sof/pcm.c | 8 +++++++- sound/soc/sof/sof-audio.c | 7 +++++++ sound/soc/sof/sof-of-dev.c | 23 +++++++++++++++++++++++ sound/soc/sof/sof-of-dev.h | 8 ++++++++ sound/soc/sof/sof-priv.h | 1 + 8 files changed, 68 insertions(+), 1 deletion(-) -- 2.25.1

3 9

[PATCH v2] dt-bindings: sound: tlv320aic3x: Convert to dtschema
by Jai Luthra 04 Aug '22

04 Aug '22

Convert bindings for TI's TLV320AIC3x audio codecs to dtschema. Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- .../bindings/sound/ti,tlv320aic3x.yaml | 158 ++++++++++++++++++ .../devicetree/bindings/sound/tlv320aic3x.txt | 97 ----------- 2 files changed, 158 insertions(+), 97 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml new file mode 100644 index 000000000000..f0ed856162ea --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic3x.yaml @@ -0,0 +1,158 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec + +description: | + TLV320AIC3x are a series of low-power stereo audio codecs with stereo + headphone amplifier, as well as multiple inputs and outputs programmable in + single-ended or fully differential configurations. + + The serial control bus supports SPI or I2C protocols, while the serial audio + data bus is programmable for I2S, left/right-justified, DSP, or TDM modes. + + The following pins can be referred in the sound node's audio routing property: + + CODEC output pins: + LLOUT + RLOUT + MONO_LOUT + HPLOUT + HPROUT + HPLCOM + HPRCOM + + CODEC input pins for TLV320AIC3104: + MIC2L + MIC2R + LINE1L + LINE1R + + CODEC input pins for other compatible codecs: + MIC3L + MIC3R + LINE1L + LINE2L + LINE1R + LINE2R + +maintainers: + - Jai Luthra <j-luthra(a)ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + gpio-reset: + maxItems: 1 + description: + Deprecated, please use reset-gpios instead. + deprecated: true + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + ai3x-ocmv = <0>; + }; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; - ai3x-ocmv = <0>; - }; -}; -- 2.17.1

3 2

[PATCH v2 0/3] DT binding for sample format conversion
by Sameer Pujar 04 Aug '22

04 Aug '22

DT binding properties are available to fixup rate and channel parameters of a DAI. This series extends this to sample format conversion as well. With this now DAI PCM parameters (channels, sample rate and sample format) can be fixed up as necessary in an audio path. Changelog: ========== v1->v2: ------- * Move DAI params properties to a new schema and re-use this for simple-card and audio-graph-card. * Use string type for DAI format binding as suggested by Krzysztof. Sameer Pujar (3): ASoC: dt-bindings: Add schema for common DAI params ASoC: dt-bindings: Add sample format conversion ASoC: simple-card-utils: Fixup DAI sample format .../bindings/sound/audio-graph-port.yaml | 17 +++++----- .../devicetree/bindings/sound/audio-graph.yaml | 9 ++--- .../devicetree/bindings/sound/dai-params.yaml | 38 ++++++++++++++++++++++ .../devicetree/bindings/sound/simple-card.yaml | 20 +++++------- include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 34 +++++++++++++++++++ 6 files changed, 95 insertions(+), 24 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/dai-params.yaml -- 2.7.4

2 9

[PATCH v4] ASoC: amd: acp: Initialize list to store acp_stream during pcm_open
by Venkata Prasad Potturu 04 Aug '22

04 Aug '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> We are currently allocating acp_stream during pcm_open and saving it in static array corresponds to array index calculated based on cpu dai->driver id. This approach will fail if we have single dai linked to multiple pcm device as we will have same dai->driver id or array index for multiple pcm open. Initialize new linked list stream_list to store opened pcm stream info dynamically. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v3: -- Modify to spin_lock() in irq handler and spin_lock_irq() in dma open and close functions instead of spin_lock_irqsave(). Changes since v2: -- Add locking mechanism in irq handler, linked list updatation and deletion. Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-platform.c | 40 +++++++++++++++----------------- sound/soc/amd/acp/amd.h | 4 +++- 2 files changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index f561d39b33e2..beee53aedeaf 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -94,7 +94,7 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; u16 i2s_flag = 0; - u32 ext_intr_stat, ext_intr_stat1, i; + u32 ext_intr_stat, ext_intr_stat1; if (!adata) return IRQ_NONE; @@ -104,14 +104,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); - for (i = 0; i < ACP_MAX_STREAM; i++) { - stream = adata->stream[i]; + spin_lock(&adata->acp_lock); + list_for_each_entry(stream, &adata->stream_list, list) { if (stream && (ext_intr_stat & stream->irq_bit)) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } if (adata->rsrc->no_of_ctrls == 2) { if (stream && (ext_intr_stat1 & stream->irq_bit)) { @@ -119,10 +118,10 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } } } + spin_unlock(&adata->acp_lock); if (i2s_flag) return IRQ_HANDLED; @@ -146,9 +145,8 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } -static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) +static void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int size) { - struct acp_stream *stream = adata->stream[cpu_id]; struct snd_pcm_substream *substream = stream->substream; struct acp_resource *rsrc = adata->rsrc; dma_addr_t addr = substream->dma_buffer.addr; @@ -174,13 +172,10 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct snd_pcm_runtime *runtime = substream->runtime; struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; int ret; stream = kzalloc(sizeof(*stream), GFP_KERNEL); @@ -188,7 +183,10 @@ static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_subs return -ENOMEM; stream->substream = substream; - adata->stream[stream_id] = stream; + + spin_lock_irq(&adata->acp_lock); + list_add_tail(&stream->list, &adata->stream_list); + spin_unlock_irq(&adata->acp_lock); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; @@ -212,16 +210,13 @@ static int acp_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); struct acp_dev_data *adata = snd_soc_component_get_drvdata(component); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct acp_stream *stream = substream->runtime->private_data; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; u64 size = params_buffer_bytes(params); /* Configure ACP DMA block with params */ config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream_id, size); + config_acp_dma(adata, stream, size); return 0; } @@ -261,16 +256,15 @@ static int acp_dma_new(struct snd_soc_component *component, static int acp_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); - struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; + struct acp_stream *stream = substream->runtime->private_data; - stream = adata->stream[stream_id]; + /* Remove entry from list */ + spin_lock_irq(&adata->acp_lock); + list_del(&stream->list); + spin_unlock_irq(&adata->acp_lock); kfree(stream); - adata->stream[stream_id] = NULL; return 0; } @@ -305,6 +299,10 @@ int acp_platform_register(struct device *dev) dev_err(dev, "Fail to register acp i2s component\n"); return status; } + + INIT_LIST_HEAD(&adata->stream_list); + spin_lock_init(&adata->acp_lock); + return 0; } EXPORT_SYMBOL_NS_GPL(acp_platform_register, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 60a7c95f134f..be8bb8247c4e 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -91,6 +91,7 @@ struct acp_chip_info { }; struct acp_stream { + struct list_head list; struct snd_pcm_substream *substream; int irq_bit; int dai_id; @@ -123,7 +124,8 @@ struct acp_dev_data { struct snd_soc_dai_driver *dai_driver; int num_dai; - struct acp_stream *stream[ACP_MAX_STREAM]; + struct list_head stream_list; + spinlock_t acp_lock; struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; -- 2.25.1

1 0

USB-Audio: Add profile for MSI MEG Z690I Unify
by GitHub pull_request - opened 04 Aug '22

04 Aug '22

alsa-project/alsa-ucm-conf pull request #196 was opened from 13r0ck: This is tested and works, though I am not a fan of my `Needle` and `Haystack` solution here, but before also making a PR to alsa-utils I wanted direction. Would it be better to add a `Type Bool` condition, or a `skip` modifier to `SectionDevice`? I prefer the later, because it skips the `If {} Condition {} true {}` lines, but I would like an alsa maintainer's opinion. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/196 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/196.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
by Ivan Hasenkampf 03 Aug '22

03 Aug '22

Fixes speaker output on HP Spectre x360 15-eb0xxx Signed-off-by: Ivan Hasenkampf <ivan.hasenkampf(a)gmail.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 2f55bc43bfa9..fe43e5807b8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9072,6 +9072,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.37.0

2 1

[PATCH] ALSA: hda/realtek: Add quirk for HP Spectre x360 15-eb0xxx
by Ivan Hasenkampf 03 Aug '22

03 Aug '22

Fixes speaker output on HP Spectre x360 15-eb0xxx Signed-off-by: Ivan Hasenkampf <ivan.hasenkampf(a)gmail.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 2f55bc43bfa9..fe43e5807b8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9072,6 +9072,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x86e7, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x86e8, "HP Spectre x360 15-eb0xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.37.0

1 0

[PATCH] dt-bindings: sound: tlv320aic3x: Convert to dtschema
by Jai Luthra 03 Aug '22

03 Aug '22

Convert bindings for TI's TLV320AIC3x audio codecs to dtschema. Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- .../devicetree/bindings/sound/tlv320aic3x.txt | 97 ------------ .../bindings/sound/tlv320aic3x.yaml | 145 ++++++++++++++++++ 2 files changed, 145 insertions(+), 97 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.yaml diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt deleted file mode 100644 index 20931a63fd64..000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ /dev/null @@ -1,97 +0,0 @@ -Texas Instruments - tlv320aic3x Codec module - -The tlv320aic3x serial control bus communicates through both I2C and SPI bus protocols - -Required properties: - -- compatible - "string" - One of: - "ti,tlv320aic3x" - Generic TLV320AIC3x device - "ti,tlv320aic33" - TLV320AIC33 - "ti,tlv320aic3007" - TLV320AIC3007 - "ti,tlv320aic3106" - TLV320AIC3106 - "ti,tlv320aic3104" - TLV320AIC3104 - - -- reg - <int> - I2C slave address - - -Optional properties: - -- reset-gpios - GPIO specification for the active low RESET input. -- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality - - Not supported on tlv320aic3104 -- ai3x-micbias-vg - MicBias Voltage required. - 1 - MICBIAS output is powered to 2.0V, - 2 - MICBIAS output is powered to 2.5V, - 3 - MICBIAS output is connected to AVDD, - If this node is not mentioned or if the value is incorrect, then MicBias - is powered down. -- ai3x-ocmv - Output Common-Mode Voltage selection: - 0 - 1.35V, - 1 - 1.5V, - 2 - 1.65V, - 3 - 1.8V -- AVDD-supply, IOVDD-supply, DRVDD-supply, DVDD-supply : power supplies for the - device as covered in Documentation/devicetree/bindings/regulator/regulator.txt - -Deprecated properties: - -- gpio-reset - gpio pin number used for codec reset - -CODEC output pins: - * LLOUT - * RLOUT - * MONO_LOUT - * HPLOUT - * HPROUT - * HPLCOM - * HPRCOM - -CODEC input pins for TLV320AIC3104: - * MIC2L - * MIC2R - * LINE1L - * LINE1R - -CODEC input pins for other compatible codecs: - * MIC3L - * MIC3R - * LINE1L - * LINE2L - * LINE1R - * LINE2R - -The pins can be used in referring sound node's audio-routing property. - -I2C example: - -#include <dt-bindings/gpio/gpio.h> - -tlv320aic3x: tlv320aic3x@1b { - compatible = "ti,tlv320aic3x"; - reg = <0x1b>; - - reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; -}; - -SPI example: - -spi0: spi@f0000000 { - tlv320aic3x: codec@0 { - compatible = "ti,tlv320aic3x"; - reg = <0>; /* CS number */ - #sound-dai-cells = <0>; - spi-max-frequency = <1000000>; - - AVDD-supply = <&regulator>; - IOVDD-supply = <&regulator>; - DRVDD-supply = <&regulator>; - DVDD-supply = <&regulator>; - ai3x-ocmv = <0>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.yaml b/Documentation/devicetree/bindings/sound/tlv320aic3x.yaml new file mode 100644 index 000000000000..6efb1d459543 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.yaml @@ -0,0 +1,145 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320aic3x.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC3x Codec Device Tree Bindings + +maintainers: + - Jai Luthra <j-luthra(a)ti.com> + +properties: + compatible: + enum: + - ti,tlv320aic3x + - ti,tlv320aic33 + - ti,tlv320aic3007 + - ti,tlv320aic3106 + - ti,tlv320aic3104 + + reg: + maxItems: 1 + description: i2c slave address + + reset-gpios: + maxItems: 1 + description: + GPIO specification for the active low RESET input. + + ai3x-gpio-func: + description: AIC3X_GPIO1 & AIC3X_GPIO2 Functionality + $ref: /schemas/types.yaml#/definitions/uint32 + minItems: 3 + maxItems: 3 + + ai3x-micbias-vg: + description: MicBias required voltage. If node is omitted then MicBias is powered down. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 1 + description: MICBIAS output is powered to 2.0V. + - const: 2 + description: MICBIAS output is powered to 2.5V. + - const: 3 + description: MICBIAS output is connected to AVDD. + + ai3x-ocmv: + description: Output Common-Mode Voltage selection. + $ref: /schemas/types.yaml#/definitions/uint32 + oneOf: + - const: 0 + description: 1.35V + - const: 1 + description: 1.5V + - const: 2 + description: 1.65V + - const: 3 + description: 1.8V + + AVDD-supply: + description: Analog DAC voltage. + + IOVDD-supply: + description: I/O voltage. + + DRVDD-supply: + description: ADC analog and output driver voltage. + + DVDD-supply: + description: Digital core voltage. + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +#The pins can be used in referring sound node's audio-routing property. + +#CODEC output pins: + # LLOUT + # RLOUT + # MONO_LOUT + # HPLOUT + # HPROUT + # HPLCOM + # HPRCOM + +#CODEC input pins for TLV320AIC3104: + # MIC2L + # MIC2R + # LINE1L + # LINE1R + +#CODEC input pins for other compatible codecs: + # MIC3L + # MIC3R + # LINE1L + # LINE2L + # LINE1R + # LINE2R + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_i2c: tlv320aic3x@1b { + compatible = "ti,tlv320aic3x"; + reg = <0x1b>; + + reset-gpios = <&gpio1 17 GPIO_ACTIVE_LOW>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + }; + }; + + - | + #include <dt-bindings/gpio/gpio.h> + spi { + #address-cells = <1>; + #size-cells = <0>; + + tlv320aic3x_spi: codec@0 { + compatible = "ti,tlv320aic3x"; + reg = <0>; /* CS number */ + #sound-dai-cells = <0>; + + AVDD-supply = <&regulator>; + IOVDD-supply = <&regulator>; + DRVDD-supply = <&regulator>; + DVDD-supply = <&regulator>; + ai3x-ocmv = <0>; + }; + }; + +... -- 2.17.1

3 3

[PATCH] soc: sof: fix clang -Wformat warnings
by Justin Stitt 03 Aug '22

03 Aug '22

When building with Clang we encounter these warnings: | sound/soc/sof/ipc3-topology.c:2343:4: error: format specifies type | 'unsigned char' but the argument has type 'int' [-Werror,-Wformat] | SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); | ^~~~~~~~~~~~~~~^~~~~~~~~~~~~~~^~~~~~~~~~~~~ Use correct format specifier `%d` since args are of type int. Link: https://github.com/ClangBuiltLinux/linux/issues/378 Reported-by: Nathan Chancellor <nathan(a)kernel.org> Suggested-by: Nathan Chancellor <nathan(a)kernel.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- Reported by Nathan here: https://lore.kernel.org/all/YtmrCJjQrSbv8Aj1@dev-arch.thelio-3990X/ sound/soc/sof/ipc3-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc3-topology.c b/sound/soc/sof/ipc3-topology.c index b2cc046b9f60..65923e7a5976 100644 --- a/sound/soc/sof/ipc3-topology.c +++ b/sound/soc/sof/ipc3-topology.c @@ -2338,7 +2338,7 @@ static int sof_ipc3_parse_manifest(struct snd_soc_component *scomp, int index, } dev_info(scomp->dev, - "Topology: ABI %d:%d:%d Kernel ABI %hhu:%hhu:%hhu\n", + "Topology: ABI %d:%d:%d Kernel ABI %d:%d:%d\n", man->priv.data[0], man->priv.data[1], man->priv.data[2], SOF_ABI_MAJOR, SOF_ABI_MINOR, SOF_ABI_PATCH); -- 2.37.1.359.gd136c6c3e2-goog

3 3

[PATCH] ASoC: amd: yc: Update DMI table entries for AMD platforms
by Syed Saba Kareem 02 Aug '22

02 Aug '22

From: syed sabakareem <Syed.SabaKareem(a)amd.com> Updated DMI entries 21EM, 21EN, 21J5 and 21J6 for AMD platforms P15v Gen 3 and P14s Gen 3. Link: https://bugzilla.kernel.org/show_bug.cgi?id=216267 Signed-off-by: syed saba kareem <Syed.SabaKareem(a)amd.com> Reported-by: Pananchikkal Renjith <renjith.pananchikkal(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index ecfe7a790790..e0b24e1daef3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -143,6 +143,34 @@ static const struct dmi_system_id yc_acp_quirk_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "21CL"), } }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EM"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21EN"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21J5"), + } + }, + { + .driver_data = &acp6x_card, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_MATCH(DMI_PRODUCT_NAME, "21J6"), + } + }, {} }; -- 2.25.1

2 1

[PATCH v7 2/2] ASoC: sun50i-dmic: dt-bindings: add DT bindings for DMIC controller
by Ban Tao 02 Aug '22

02 Aug '22

DT binding documentation for this new ASoC driver. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Reviewed-by: Maxime Ripard <maxime(a)cerno.tech> --- v1->v2: 1.Fix some build errors. v2->v3: 1.Fix some build errors. v3->v4: 1.None. v4->v5: 1.Add interrupt. 2.Keep clock and reset index. v5->v6: 1.None. v6->v7: 1.None. --- .../bindings/sound/allwinner,sun50i-h6-dmic.yaml | 79 ++++++++++++++++++++++ 1 file changed, 79 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml new file mode 100644 index 0000000..185d3e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml @@ -0,0 +1,79 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun50i-h6-dmic.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner H6 DMIC Device Tree Bindings + +maintainers: + - Ban Tao <fengzheng923(a)gmail.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + const: allwinner,sun50i-h6-dmic + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: bus + - const: mod + + dmas: + items: + - description: RX DMA Channel + + dma-names: + items: + - const: rx + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + - resets + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/interrupt-controller/irq.h> + + #include <dt-bindings/clock/sun50i-h6-ccu.h> + #include <dt-bindings/reset/sun50i-h6-ccu.h> + + dmic: dmic@5095000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun50i-h6-dmic"; + reg = <0x05095000 0x400>; + interrupts = <GIC_SPI 22 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&ccu CLK_BUS_DMIC>, <&ccu CLK_DMIC>; + clock-names = "bus", "mod"; + dmas = <&dma 7>; + dma-names = "rx"; + resets = <&ccu RST_BUS_DMIC>; + }; + +... \ No newline at end of file -- 2.7.4

3 2

[PATCH v3] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 02 Aug '22

02 Aug '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Beside conversion, 'fsl,shared-interrupt' and '#sound-dai-cells' are added for they are already used by some dts. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v3 - address comments from Krzysztof changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 221 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 221 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..86ee46755b0c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,221 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - enum: + - fsl,vf610-sai + - fsl,imx6sx-sai + - fsl,imx6ul-sai + - fsl,imx7ulp-sai + - fsl,imx8mq-sai + - fsl,imx8qm-sai + - fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + type: boolean + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + type: boolean + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the asynchronous mode, which means both transmitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + type: boolean + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + maxItems: 16 + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: SAI will output the SAI MCLK clock. + type: boolean + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: Interrupt is shared with other modules. + type: boolean + + "#sound-dai-cells": + const: 0 + description: optional, some dts node didn't add it. + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + dma-names: + items: + - const: rx + - const: tx + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + #sound-dai-cells = <0>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

3 3

[PATCH 0/2] DT binding for sample format conversion
by Sameer Pujar 02 Aug '22

02 Aug '22

DT binding properties are available to fixup rate and channel parameters of a DAI. This series extends this to sample format conversion as well. With this now DAI PCM parameters (channels, sample rate and sample format) can be fixed up as necessary in an audio path. Sameer Pujar (2): ASoC: dt-bindings: Add sample format conversion ASoC: simple-card-utils: Fixup DAI sample format .../devicetree/bindings/sound/audio-graph-port.yaml | 21 +++++++++++++++++++++ .../devicetree/bindings/sound/audio-graph.yaml | 11 +++++++++++ .../devicetree/bindings/sound/simple-card.yaml | 15 +++++++++++++++ include/sound/simple_card_utils.h | 1 + sound/soc/generic/simple-card-utils.c | 12 ++++++++++++ 5 files changed, 60 insertions(+) -- 2.7.4

2 5

Discuss echo reference feature
by Song, Gongjun 02 Aug '22

02 Aug '22

Hi all, Could you please give me some suggestions on how to support loop-back in embedded DSP FW?  ------------------------------------ | CPU: SoC with embedded DSP | ------------------------- | | dai | | | PCM0P --> volume --> demux --> SSP |------> | codec: speaker amplifer | | | | | | | PCM6C <----------------+ | ------------------------- | | ------------------------------------ As the diagram shows, when using the speaker to play audio data (PCM0P), in addition to transferring the audio data to the codec, I also need to transfer this audio data as a reference stream (PCM6C) to the user space for AEC (Automatic Echo Cancellation). The speaker amplifier doesn’t have the capability to send feedback data to SoC over SSP (Synchronous Serial Protocol, I2S/PCM interface) port. So, it’s the demux widget in the embedded DSP firmware (Sound Open Firmware) that sends a copy of playback data to PCM6C as the feedback for user space to implement AEC.  When using the above topology file to verify functionality (playing audio data, and starting to record data at the same time), I get the following error: [  303.956529]  Echo-Ref: ASoC: prepare FE Echo-Ref  [  303.956531]  Echo-Ref: ASoC: no backend DAIs enabled for Echo-Ref  [  303.956532]  Echo-Ref: ASoC: dpcm_fe_dai_prepare() failed (-22)  [  303.957215]  Echo-Ref: ASoC: hw_free FE Echo-Ref  From the log we can see that the above error is caused by PCM6C not having backend dais (Echo-Ref is the name of the PCM6C).  Could you give me some suggestions to implement the above case in the kernel and topology?  Thanks, Gongjun

1 0

topology: plugins: fix off by 1 mem allocation error
by GitHub pull_request - opened 01 Aug '22

01 Aug '22

alsa-project/alsa-utils pull request #162 was opened from yongzhi1: Fix valgrind memcheck error: ==1337389== Invalid write of size 1 ==1337389== at 0x4A4AFAB: __vsnprintf_internal (vsnprintf.c:117) ==1337389== by 0x4AECF40: __snprintf_chk (snprintf_chk.c:38) ==1337389== by 0x484B870: snprintf (stdio2.h:67) ==1337389== by 0x484B870: print_as_hex_bytes (nhlt-processor.c:112) ==1337389== by 0x484B870: merge_manifest_data (nhlt-processor.c:154) ==1337389== by 0x484B870: do_nhlt (nhlt-processor.c:420) ==1337389== by 0x484B870: _snd_topology_nhlt_process (nhlt-processor.c:484) The consecutive snprintf overwrites always the previous terminator until it hits the very last call of: snprintf(dst, ALSA_BYTE_CHARS + 1, "0x%02x,", *nhlt_buffer); when the size n given to snprintf is 1 more than allocated. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/162 Patch URL : https://github.com/alsa-project/alsa-utils/pull/162.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: amd: acp: Add TDM support for acp i2s stream
by Venkata Prasad Potturu 01 Aug '22

01 Aug '22

Add callback and code changes to enable ACP I2S controller in TDM mode. Add new fields in acp_stream and acp_dev_data struct to configure tdm related registers for ACP i2s controllers. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Depends on:- -- https://patchwork.kernel.org/project/alsa-devel/patch/20220728124057.54259-… sound/soc/amd/acp/acp-i2s.c | 78 ++++++++++++++++++++++++++++++++++++- sound/soc/amd/acp/amd.h | 12 ++++++ 2 files changed, 89 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-i2s.c b/sound/soc/amd/acp/acp-i2s.c index 393f729ef561..b1257923b225 100644 --- a/sound/soc/amd/acp/acp-i2s.c +++ b/sound/soc/amd/acp/acp-i2s.c @@ -25,6 +25,63 @@ #define DRV_NAME "acp_i2s_playcap" +static int acp_i2s_set_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(cpu_dai); + int mode; + + mode = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + switch (mode) { + case SND_SOC_DAIFMT_I2S: + adata->tdm_mode = TDM_DISABLE; + break; + case SND_SOC_DAIFMT_DSP_A: + adata->tdm_mode = TDM_ENABLE; + break; + default: + return -EINVAL; + } + return 0; +} + +static int acp_i2s_set_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, u32 rx_mask, + int slots, int slot_width) +{ + struct device *dev = dai->component->dev; + struct acp_dev_data *adata = snd_soc_dai_get_drvdata(dai); + struct acp_stream *stream; + int slot_len; + + switch (slot_width) { + case SLOT_WIDTH_8: + slot_len = 8; + break; + case SLOT_WIDTH_16: + slot_len = 16; + break; + case SLOT_WIDTH_24: + slot_len = 24; + break; + case SLOT_WIDTH_32: + slot_len = 0; + break; + default: + dev_err(dev, "Unsupported bitdepth %d\n", slot_width); + return -EINVAL; + } + + list_for_each_entry(stream, &adata->stream_list, list) { + if (tx_mask && stream->dir == SNDRV_PCM_STREAM_PLAYBACK) + adata->tdm_tx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + else if (rx_mask && stream->dir == SNDRV_PCM_STREAM_CAPTURE) + adata->tdm_rx_fmt[stream->dai_id - 1] = + FRM_LEN | (slots << 15) | (slot_len << 18); + } + return 0; +} + static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -33,7 +90,7 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ struct acp_resource *rsrc; u32 val; u32 xfer_resolution; - u32 reg_val; + u32 reg_val, fmt_reg, tdm_fmt; u32 lrclk_div_val, bclk_div_val; adata = snd_soc_dai_get_drvdata(dai); @@ -62,12 +119,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_ITER; + fmt_reg = ACP_BTTDM_TXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_ITER; + fmt_reg = ACP_I2STDM_TXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_ITER; + fmt_reg = ACP_HSTDM_TXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -77,12 +137,15 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ switch (dai->driver->id) { case I2S_BT_INSTANCE: reg_val = ACP_BTTDM_IRER; + fmt_reg = ACP_BTTDM_RXFRMT; break; case I2S_SP_INSTANCE: reg_val = ACP_I2STDM_IRER; + fmt_reg = ACP_I2STDM_RXFRMT; break; case I2S_HS_INSTANCE: reg_val = ACP_HSTDM_IRER; + fmt_reg = ACP_HSTDM_RXFRMT; break; default: dev_err(dev, "Invalid dai id %x\n", dai->driver->id); @@ -95,6 +158,16 @@ static int acp_i2s_hwparams(struct snd_pcm_substream *substream, struct snd_pcm_ val = val | (xfer_resolution << 3); writel(val, adata->acp_base + reg_val); + if (adata->tdm_mode) { + val = readl(adata->acp_base + reg_val); + writel(val | BIT(1), adata->acp_base + reg_val); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tdm_fmt = adata->tdm_tx_fmt[dai->driver->id - 1]; + else + tdm_fmt = adata->tdm_rx_fmt[dai->driver->id - 1]; + writel(tdm_fmt, adata->acp_base + fmt_reg); + } + if (rsrc->soc_mclk) { switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -443,6 +516,7 @@ static int acp_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_d stream->id = dai->driver->id + dir; stream->dai_id = dai->driver->id; stream->irq_bit = irq_bit; + stream->dir = substream->stream; return 0; } @@ -452,6 +526,8 @@ const struct snd_soc_dai_ops asoc_acp_cpu_dai_ops = { .hw_params = acp_i2s_hwparams, .prepare = acp_i2s_prepare, .trigger = acp_i2s_trigger, + .set_fmt = acp_i2s_set_fmt, + .set_tdm_slot = acp_i2s_set_tdm_slot, }; EXPORT_SYMBOL_NS_GPL(asoc_acp_cpu_dai_ops, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index be8bb8247c4e..5f2119f42271 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -84,6 +84,14 @@ #define ACP_MAX_STREAM 8 +#define TDM_ENABLE 1 +#define TDM_DISABLE 0 + +#define SLOT_WIDTH_8 0x8 +#define SLOT_WIDTH_16 0x10 +#define SLOT_WIDTH_24 0x18 +#define SLOT_WIDTH_32 0x20 + struct acp_chip_info { char *name; /* Platform name */ unsigned int acp_rev; /* ACP Revision id */ @@ -96,6 +104,7 @@ struct acp_stream { int irq_bit; int dai_id; int id; + int dir; u64 bytescount; u32 reg_offset; u32 pte_offset; @@ -120,6 +129,7 @@ struct acp_dev_data { void __iomem *acp_base; unsigned int i2s_irq; + bool tdm_mode; /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; int num_dai; @@ -134,6 +144,8 @@ struct acp_dev_data { u32 lrclk_div; struct acp_resource *rsrc; + u32 tdm_tx_fmt[3]; + u32 tdm_rx_fmt[3]; }; union acp_i2stdm_mstrclkgen { -- 2.25.1

1 0

[PATCH 0/7] ALSA: Replace sprintf() with sysfs_emit()
by Takashi Iwai 01 Aug '22

01 Aug '22

Hi, this is a patch set for rather simple conversions from the plain sprintf() & co to the new helpers, sysfs_emit() and sysfs_emit_at(). No functional changes are expected. Takashi === Takashi Iwai (7): ALSA: ac97: Replace sprintf() with sysfs_emit() ALSA: aoa: Replace sprintf() with sysfs_emit() ALSA: control-led: Replace sprintf() with sysfs_emit() ALSA: core: Replace scnprintf() with sysfs_emit() ALSA: pcm: Replace sprintf() with sysfs_emit() ALSA: hda: Replace sprintf() with sysfs_emit() ALSA: line6: Replace sprintf() with sysfs_emit() sound/ac97/bus.c | 2 +- sound/aoa/soundbus/sysfs.c | 22 ++++++++------------ sound/core/control_led.c | 29 +++++++++++--------------- sound/core/init.c | 4 ++-- sound/core/pcm.c | 2 +- sound/hda/hdac_sysfs.c | 42 +++++++++++++++++++------------------- sound/pci/hda/hda_sysfs.c | 23 ++++++++++----------- sound/usb/line6/pod.c | 8 ++++---- sound/usb/line6/podhd.c | 4 ++-- 9 files changed, 62 insertions(+), 74 deletions(-) -- 2.35.3

1 7

[PATCH v1] ASoc: dt-bindings: tas27xx: update maximum I2C address and revise the format
by Raphael-Xu 01 Aug '22

01 Aug '22

1.use correct maximum I2C address 2.revise the format Signed-off-by: Raphael-Xu <13691752556(a)139.com> --- .../devicetree/bindings/sound/tas27xx.yaml | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tas27xx.yaml b/Documentation/devicetree/bindings/sound/tas27xx.yaml index 66a0df8850ea..573652577462 100644 --- a/Documentation/devicetree/bindings/sound/tas27xx.yaml +++ b/Documentation/devicetree/bindings/sound/tas27xx.yaml @@ -8,7 +8,7 @@ $schema: "http://devicetree.org/meta-schemas/core.yaml#" title: Texas Instruments TAS2764/TAS2780 Smart PA maintainers: - - Shenghao Ding <shenghao-ding(a)ti.com> + - Raphael Xu <raphael-xu(a)ti.com> description: | The TAS2764/TAS2780 is a mono, digital input Class-D audio amplifier @@ -23,17 +23,19 @@ properties: - ti,tas2780 reg: - maxItems: 1 description: | - I2C address of the device can be between 0x38 to 0x45. + I2C address of the device can be between 0x38 to 0x3f. + minimum: 0x38 + maximum: 0x3f reset-gpios: - maxItems: 1 description: GPIO used to reset the device. + maxItems: 1 shutdown-gpios: + description: | + GPIO used to control the state of the device.Only for TAS2764. maxItems: 1 - description: GPIO used to control the state of the device. interrupts: maxItems: 1 @@ -41,10 +43,14 @@ properties: ti,imon-slot-no: $ref: /schemas/types.yaml#/definitions/uint32 description: TDM TX current sense time slot. + minimum: 0 + maximum: 3 ti,vmon-slot-no: $ref: /schemas/types.yaml#/definitions/uint32 description: TDM TX voltage sense time slot. + minimum: 0 + maximum: 3 '#sound-dai-cells': const: 1 -- 2.35.1

2 1

[GIT PULL] ASoC updates for v5.20-2
by Mark Brown 01 Aug '22

01 Aug '22

The following changes since commit 1e347f861da8ddb17e1d1b3113cb6c188e0de3e5: ASoC: rockchip-i2s: Undo BCLK pinctrl changes (2022-07-14 13:25:52 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v5.20-2 for you to fetch changes up to 40d060b8158e26438398bf1132925f666e3b6480: ASoC: q6asm: use kcalloc() instead of kzalloc() (2022-07-28 11:59:10 +0100) ---------------------------------------------------------------- ASoC: More updates for v5.20 More updates that came in since the last pull request I sent, a series of driver specific changes: - Support for AMD RPL, some Intel platforms and Mediatek MT8186. ---------------------------------------------------------------- Aidan MacDonald (3): ASoC: jz4740-i2s: Remove Open Firmware dependency ASoC: jz4740-i2s: Refactor DMA channel setup ASoC: jz4740-i2s: Remove unused 'mem' resource Ajit Kumar Pandey (2): ASoC: SOF: amd: Add SOF pm ops callback for Renoir ASoC: SOF: topology: add code to parse config params for ACPDMIC dai Alexander Martinz (2): ASoC: codecs: tfa989x: Add support for tfa9890 ASoC: dt-bindings: nxp,tfa989x: Add tfa9890 support Amadeusz Sławiński (12): ASoC: Intel: avs: Add max98373 machine board ASoC: Intel: avs: Fix parsing UUIDs in topology ASoC: Remove unused hw_write_t type ASoC: Intel: avs: Set max DMA segment size ASoC: Intel: avs: Use helper function to set up DMA ASoC: codecs: rt298: Fix NULL jack in interrupt ASoC: codecs: rt298: Fix jack detection ASoC: codecs: rt286: Set component to NULL on remove ASoC: codecs: rt298: Set component to NULL on remove ASoC: codecs: rt274: Set component to NULL on remove ASoC: Makefile: Fix simultaneous build of KUNIT tests ASoC: Intel: avs: Use lookup table to create modules Andrey Turkin (2): ASoC: Intel: sof_es8336: Fix GPIO quirks set via module option ASoC: Intel: sof_es8336: ignore GpioInt when looking for speaker/headset GPIO lines Andy Shevchenko (4): ASoC: madera: Replace kernel.h with the necessary inclusions ASoC: Intel: catpt: remove duplicating driver data retrieval ASoC: SOF: Intel: bdw: remove duplicating driver data retrieval ASoC: SOF: Intel: byt: remove duplicating driver data retrieval Arnaud Ferraris (2): ASoC: dt-bindings: sun50i-codec: Add binding for internal bias ASoC: sun50i-codec-analog: Add support for internal bias Bard Liao (8): ASoC: SOF: Intel: add MeteorLake machines ASoC: SOF: IPC4: add sdw blob ASoC: SOF: ipc4-topology: add SoundWire/ALH aggregation support ASoC: SOC: Intel: introduce cl_init callback ASoC: SOF: Intel: Add support for MeteorLake (MTL) soundwire: Intel: add trigger callback ASoC: SOF: Intel: add trigger callback into sdw_callback ASoC: SOF: ipc4-topology: move ida allocate/free to widget_setup/free Brent Lu (3): ASoC: Intel: sof_cs42l42: support BT offload audio ASoC: Intel: sof_cs42l42: add adl_mx98360a_cs4242 board config ASoC: SOF: dai-intel: add SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_AON bit Bryan O'Donoghue (1): ASoC: qcom: lpass: Fix apq8016 compat string to match yaml Cezary Rojewski (47): ALSA: Add snd_pcm_direction_name() helper ASoC: codecs: Add HD-Audio codec driver ASoC: Intel: avs: Add HDAudio machine board ASoC: Intel: avs: Add DMIC machine board ASoC: Intel: avs: Add I2S-test machine board ASoC: Intel: avs: Add rt274 machine board ASoC: Intel: avs: Add rt286 machine board ASoC: Intel: avs: Add rt298 machine board ASoC: Intel: avs: Add rt5682 machine board ASoC: Intel: avs: Add nau8825 machine board ASoC: Intel: avs: Add ssm4567 machine board ASoC: Intel: avs: Add max98357a machine board ASoC: Intel: avs: Add da7219 machine board ASoC: codecs: rt274: Always init jack_detect_work ASoC: codecs: rt286: Reorganize jack detect handling ASoC: codecs: rt298: Reorganize jack detect handling ASoC: Intel: Rename haswell source file to hsw_rt5640 ASoC: Intel: hsw_rt5640: Reword prefixes of all driver members ASoC: Intel: hsw_rt5640: Reword driver name ASoC: Intel: hsw_rt5640: Update code indentation ASoC: Intel: hsw_rt5640: Update file comments ASoC: Intel: hsw_rt5640: Improve probe() function quality ASoC: Intel: hsw_rt5640: Improve hw_params() debug-ability ASoC: Intel: Rename broadwell source file to bdw_rt286 ASoC: Intel: bdw_rt286: Reword prefixes of all driver members ASoC: Intel: bdw_rt286: Reword driver name ASoC: Intel: bdw_rt286: Update code indentation ASoC: Intel: bdw_rt286: Update file comments ASoC: Intel: bdw_rt286: Improve probe() function quality ASoC: Intel: bdw_rt286: Improve hw_params() debug-ability ASoC: Intel: bdw_rt286: Improve codec_init() quality ASoC: Intel: bdw_rt286: Remove FE DAI ops ASoC: Intel: avs: Fix i2s_test card name initialization ASoC: Intel: avs: Register HDAudio ext-bus operations ASoC: Intel: avs: Assign I2S gateway when parsing topology ASoC: Intel: avs: Relax DSP core transition timings ASoC: Intel: avs: Copy only as many RX bytes as necessary ASoC: Intel: avs: Shield LARGE_CONFIG_GETs against zero payload_size ASoC: Intel: avs: Block IPC channel on suspend ASoC: Intel: avs: Recognize FW_CFG_RESERVED ASoC: Intel: avs: Replace hardcodes with SD_CTL_STREAM_RESET ASoC: Intel: avs: Lower UNLOAD_MULTIPLE_MODULES IPC timeout ASoC: Intel: avs: Update AVS_FW_INIT_TIMEOUT_US declaration ASoC: Intel: hsw_rt5640: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt286: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5650: Mark BE DAI as nonatomic ASoC: Intel: bdw_rt5677: Mark BE DAI as nonatomic Charles Keepax (170): ASoC: core: Add set_fmt_new callback that directly specifies provider ASoC: amd: vangogh: Update to use set_fmt_new callback ASoC: atmel: Update to use set_fmt_new callback ASoC: au1x: Update to use set_fmt_new callback ASoC: bcm: Update to use set_fmt_new callback ASoC: ep93xx: Update to use set_fmt_new callback ASoC: dwc: Update to use set_fmt_new callback ASoC: fsl: Update to use set_fmt_new callback ASoC: hisilicon: Update to use set_fmt_new callback ASoC: img: Update to use set_fmt_new callback ASoC: Intel: Update to use set_fmt_new callback ASoC: js4740-i2s: Update to use set_fmt_new callback ASoC: mediatek: Update to use set_fmt_new callback ASoC: meson: Update to use set_fmt_new callback ASoC: mxs-saif: Update to use set_fmt_new callback ASoC: pxa: Update to use set_fmt_new callback ASoC: qcom: Update to use set_fmt_new callback ASoC: rockchip: Update to use set_fmt_new callback ASoC: samsung: Update to use set_fmt_new callback ASoC: sh: Update to use set_fmt_new callback ASoC: stm: Update to use set_fmt_new callback ASoC: sunxi: Update to use set_fmt_new callback ASoC: tegra: Update to use set_fmt_new callback ASoC: test-component: Update to use set_fmt_new callback ASoC: ti: Update to use set_fmt_new callback ASoC: ux500: Update to use set_fmt_new callback ASoC: xtensa: Update to use set_fmt_new callback ASoC: core: Always send the CPU DAI a direct clock specifier ASoC: amd: vangogh: Rename set_fmt_new back to set_fmt ASoC: atmel: Rename set_fmt_new back to set_fmt ASoC: au1x: Rename set_fmt_new back to set_fmt ASoC: bcm: Rename set_fmt_new back to set_fmt ASoC: ep93xx: Rename set_fmt_new back to set_fmt ASoC: dwc: Rename set_fmt_new back to set_fmt ASoC: fsl: Rename set_fmt_new back to set_fmt ASoC: hisilicon: Rename set_fmt_new back to set_fmt ASoC: img: Rename set_fmt_new back to set_fmt ASoC: Intel: Rename set_fmt_new back to set_fmt ASoC: jz4740-i2s: Rename set_fmt_new back to set_fmt ASoC: mediatek: Rename set_fmt_new back to set_fmt ASoC: meson: Rename set_fmt_new back to set_fmt ASoC: mxs-saif: Rename set_fmt_new back to set_fmt ASoC: pxa: Rename set_fmt_new back to set_fmt ASoC: qcom: Rename set_fmt_new back to set_fmt ASoC: rockchip: Rename set_fmt_new back to set_fmt ASoC: samsung: Rename set_fmt_new back to set_fmt ASoC: sh: Rename set_fmt_new back to set_fmt ASoC: stm: Rename set_fmt_new back to set_fmt ASoC: sunxi: Rename set_fmt_new back to set_fmt ASoC: tegra: Rename set_fmt_new back to set_fmt ASoC: test-component: Rename set_fmt_new back to set_fmt ASoC: ti: Rename set_fmt_new back to set_fmt ASoC: ux500: Rename set_fmt_new back to set_fmt ASoC: xtensa: Rename set_fmt_new back to set_fmt ASoC: soc-dai: Remove set_fmt_new callback ASoC: simple-card-utils: Move snd_soc_component_is_codec to be local ASoC: cs42l52: Fix TLV scales for mixer controls ASoC: cs35l36: Update digital volume TLV ASoC: cs53l30: Correct number of volume levels on SX controls ASoC: cs42l52: Correct TLV for Bypass Volume ASoC: cs42l56: Correct typo in minimum level for SX volume controls ASoC: cs42l51: Correct minimum value for SX volume control ASoC: sun8i-codec: Partial revert to fix clock specifiers ASoC: mediatek: mt8186: Use new direct clock defines ASoC: cs35l45: Add endianness flag in snd_soc_component_driver ASoC: wm_adsp: Fix event for preloader ASoC: wm5110: Fix DRE control ASoC: cs35l41: Correct some control names ASoC: core: Add new SOC_DOUBLE_SX_TLV macro ASoC: dapm: Move stereo autodisable check ASoC: soc-component: Add legacy_dai_naming flag soundwire: intel: Migrate to new style legacy DAI naming flag drm/vc4: Migrate to new style legacy DAI naming flag ASoC: img: Migrate to new style legacy DAI naming flag ASoC: spear: Migrate to new style legacy DAI naming flag ASoC: jz4740-i2c: Migrate to new style legacy DAI naming flag ASoC: ep93xx: Migrate to new style legacy DAI naming flag ASoC: stm32: Migrate to new style legacy DAI naming flag ASoC: bcm: Migrate to new style legacy DAI naming flag ASoC: sh: Migrate to new style legacy DAI naming flag ASoC: tegra: Migrate to new style legacy DAI naming flag ASoC: hisilicon: Migrate to new style legacy DAI naming flag ASoC: xilinx: Migrate to new style legacy DAI naming flag ASoC: sunxi: Migrate to new style legacy DAI naming flag ASoC: Intel: Migrate to new style legacy DAI naming flag ASoC: meson: Migrate to new style legacy DAI naming flag ASoC: sti-uniperf: Migrate to new style legacy DAI naming flag ASoC: amd: Migrate to new style legacy DAI naming flag ASoC: atmel: Migrate to new style legacy DAI naming flag ASoC: fsl: Migrate to new style legacy DAI naming flag ASoC: xtensa: Migrate to new style legacy DAI naming flag ASoC: adi: Migrate to new style legacy DAI naming flag ASoC: dwc: Migrate to new style legacy DAI naming flag ASoC: qcom: Migrate to new style legacy DAI naming flag ASoC: test-component: Migrate to new style legacy DAI naming flag ASoC: rockchip: Migrate to new style legacy DAI naming flag ASoC: au1x: Migrate to new style legacy DAI naming flag ASoC: pxa: Migrate to new style legacy DAI naming flag ASoC: sof: Migrate to new style legacy DAI naming flag ASoC: ux500: Migrate to new style legacy DAI naming flag ASoC: ti: Migrate to new style legacy DAI naming flag ASoC: mxs-saif: Migrate to new style legacy DAI naming flag ASoC: samsung: Migrate to new style legacy DAI naming flag ASoC: core: Switch core to new DAI naming flag media: TDA1997x: Remove now redundant non_legacy_dai_naming flag ASoC: fsl: Remove now redundant non_legacy_dai_naming flag ASoC: meson: Remove now redundant non_legacy_dai_naming flag ASoC: pistachio: Remove now redundant non_legacy_dai_naming flag ASoC: samsung: Remove now redundant non_legacy_dai_naming flag ASoC: soc-utils: Remove now redundant non_legacy_dai_naming flag ASoC: sunxi: Remove now redundant non_legacy_dai_naming flag ASoC: tegra: Remove now redundant non_legacy_dai_naming flag ASoC: test-component: Remove now redundant non_legacy_dai_naming flag ASoC: topology: KUnit: Remove now redundant non_legacy_dai_naming flag ASoC: uniphier: Remove now redundant non_legacy_dai_naming flag ASoC: ad*: Remove now redundant non_legacy_dai_naming flag ASoC: ak*: Remove now redundant non_legacy_dai_naming flag ASoC: alc*: Remove now redundant non_legacy_dai_naming flag ASoC: cs*: Remove now redundant non_legacy_dai_naming flag ASoC: da*: Remove now redundant non_legacy_dai_naming flag ASoC: es*: Remove now redundant non_legacy_dai_naming flag ASoC: max*: Remove now redundant non_legacy_dai_naming flag ASoC: msm*: Remove now redundant non_legacy_dai_naming flag ASoC: nau*: Remove now redundant non_legacy_dai_naming flag ASoC: pcm*: Remove now redundant non_legacy_dai_naming flag ASoC: rt*: Remove now redundant non_legacy_dai_naming flag ASoC: spdif: Remove now redundant non_legacy_dai_naming flag ASoC: ssm*: Remove now redundant non_legacy_dai_naming flag ASoC: sta*: Remove now redundant non_legacy_dai_naming flag ASoC: tas*: Remove now redundant non_legacy_dai_naming flag ASoC: tfa*: Remove now redundant non_legacy_dai_naming flag ASoC: tlv320*: Remove now redundant non_legacy_dai_naming flag ASoC: twl*: Remove now redundant non_legacy_dai_naming flag ASoC: uda*: Remove now redundant non_legacy_dai_naming flag ASoC: wm*: Remove now redundant non_legacy_dai_naming flag ASoC: 88pm860x: Remove now redundant non_legacy_dai_naming flag ASoC: ab8500: Remove now redundant non_legacy_dai_naming flag ASoC: ac97: Remove now redundant non_legacy_dai_naming flag ASoC: ads117x: Remove now redundant non_legacy_dai_naming flag ASoC: bd28623: Remove now redundant non_legacy_dai_naming flag ASoC: bt-sco: Remove now redundant non_legacy_dai_naming flag ASoC: cpcap: Remove now redundant non_legacy_dai_naming flag ASoC: cq93vc: Remove now redundant non_legacy_dai_naming flag ASoC: cx20442: Remove now redundant non_legacy_dai_naming flag ASoC: dmic: Remove now redundant non_legacy_dai_naming flag ASoC: gtm601: Remove now redundant non_legacy_dai_naming flag ASoC: hdac_hdmi: Remove now redundant non_legacy_dai_naming flag ASoC: hdmi-codec: Remove now redundant non_legacy_dai_naming flag ASoC: ics43432: Remove now redundant non_legacy_dai_naming flag ASoC: inno_rk3036: Remove now redundant non_legacy_dai_naming flag ASoC: Intel: avs: Remove now redundant non_legacy_dai_naming flag ASoC: isabelle: Remove now redundant non_legacy_dai_naming flag ASoC: jz4740: Remove now redundant non_legacy_dai_naming flag ASoC: lm49453: Remove now redundant non_legacy_dai_naming flag ASoC: lochnagar: Remove now redundant non_legacy_dai_naming flag ASoC: mc13783: Remove now redundant non_legacy_dai_naming flag ASoC: ml26124: Remove now redundant non_legacy_dai_naming flag ASoC: rk817: Remove now redundant non_legacy_dai_naming flag ASoC: sgtl5000: Remove now redundant non_legacy_dai_naming flag ASoC: si476x: Remove now redundant non_legacy_dai_naming flag ASoC: stac9766: Remove now redundant non_legacy_dai_naming flag ASoC: sti-sas: Remove now redundant non_legacy_dai_naming flag ASoC: tscs42xx: Remove now redundant non_legacy_dai_naming flag ASoC: wl1273: Remove now redundant non_legacy_dai_naming flag ASoC: zl38060: Remove now redundant non_legacy_dai_naming flag ASoC: soc-component: Remove non_legacy_dai_naming flag ASoC: samsung: s3c24xx-i2s: Fix typo in DAIFMT handling ASoC: wm_adsp: Minor clean and redundant code removal firmware: cs_dsp: Add pre_stop callback firmware: cs_dsp: Add memory chunk helpers Chen-Yu Tsai (1): ASoC: rockchip: i2s: Fix crash on missing pinctrl Christophe JAILLET (3): ASoC: ux500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules ASoC: ab8500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules ASoC: qcom: q6dsp: Fix an off-by-one in q6adm_alloc_copp() Chunxu Li (1): ASoC: mediatek: mt8195: extract SOF common code Claudiu Beznea (7): ASoC: atmel: mchp-pdmc: remove space in front of mchp_pdmc_dt_init() ASoC: atmel_ssc_dai: remove #ifdef CONFIG_PM ASoC: mchp-spdifrx: use single tab indent for structure ASoC: mchp-spdiftx: return directly ret ASoC: mchp-spdiftx: add and remove black line around MODULE_DEVICE_TABLE() ASoC: mchp-spdifrx: disable end of block interrupt on failures ASoC: mchp-spdiftx: remove references to mchp_i2s_caps Colin Ian King (2): ASoC: mediatek: mt8186: Fix a handful of spelling mistakes ASoC: Intel: Skylake: remove redundant re-assignments to pointer array Conor Dooley (1): ASoC: dt-bindings: convert designware-i2s to dt-schema Cristian Ciocaltea (2): ASoC: cs35l41: Add support for CLSA3541 ACPI device ID ASoC: amd: vangogh: Use non-legacy DAI naming for cs35l41 Dan Carpenter (4): ASoC: SOF: mediatek: Fix error code in probe ASoC: SOF: ipc4-topology: Fix error code in sof_ipc4_volume_put() ASoC: amd: Fix error pointer dereference ASoC: SOF: ipc-msg-injector: fix copy in sof_msg_inject_ipc4_dfs_write() Daniel Baluta (4): ASoC: SOF: compress: Dynamically allocate pcm params struct ASoC: SOF: Copy compress parameters into extended data ASoC: SOF: compress: Prevent current kernel running with older FW uapi: sof: abi: Bump SOF ABI for ext_data_length Daniel Mack (7): ASoC: dt-bindings: max98396: add voltage supplies ASoC: dt-bindings: max98396: Document adi,bypass-slot-no ASoC: max98396: add voltage regulators ASoC: max98396: Improve some error prints ASoC: max98396: Fix register access for PCM format settings ASoC: max98396: Implement DSP speaker monitor ASoC: max98396: Fix TDM mode BSEL settings David Lin (1): ASoC: nau8825: Declare 2 channels for DAI of capture stream Fabio Estevam (1): ASoC: imx-audmux: Silence a clang warning Fei Shao (1): ASoC: mediatek: mt8186: Fix mutex double unlock in GPIO request Francesco Dolcini (1): ASoC: sgtl5000: Fix noise on shutdown/remove Gaosheng Cui (1): ASoC: codecs: wsa883x: fix warning using-module-alias-sdw.cocci Gustavo A. R. Silva (1): ASoC: q6asm: use kcalloc() instead of kzalloc() Hans de Goede (1): ASoC: Intel: bytcr_wm5102: Fix GPIO related probe-ordering problem Hui Wang (2): ASoC: nau8822: Disable internal PLL if freq_out is zero ASoC: nau8822: Add operation for internal PLL off and on Javier Martinez Canillas (6): regmap: Re-introduce bulk read support check in regmap_bulk_read() regmap: Make regmap_noinc_read() return -ENOTSUPP if map->read isn't set regmap: Wire up regmap_config provided bulk write in missed functions regmap: Re-introduce bulk read support check in regmap_bulk_read() regmap: Make regmap_noinc_read() return -ENOTSUPP if map->read isn't set regmap: Wire up regmap_config provided bulk write in missed functions Jiapeng Chong (3): ASoC: rockchip: i2s: Fix missing error code in rockchip_i2s_probe() ASoC: codecs: max98088: Clean up some inconsistent indenting ASoC: tegra20_ac97: Fix missing error code in tegra20_ac97_platform_probe() Jiasheng Jiang (1): ASoC: codecs: da7210: add check for i2c_add_driver Jiaxin Yu (22): ASoC: mediatek: mt6366: support for mt6366 codec ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: support audio clock control in platform driver ASoC: mediatek: mt8186: support gpio control in platform driver ASoC: mediatek: mt8186: add misc driver and register definitions ASoC: mediatek: mt8186: add platform driver ASoC: mediatek: mt8186: add mt8186-mt6366 common driver ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s ASoC: mediatek: mt6358: add new compatible for using mt6366 ASoC: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document ASoC: mediatek: mt8186: set the correct string to strncmp() ASoC: mediatek: mt8186: remove unnecessary judgments Judy Hsiao (6): ASoC: rockchip: i2s: switch BCLK to GPIO ASoC: rockchip: i2s: switch BCLK to GPIO ASoC: dt-bindings: rockchip: Document pinctrl-names for i2s ASoC: rockchip: i2s: Fix the debug level on missing pinctrl ASoC: rockchip: i2s: Remove unwanted dma settings in rockchip_i2s_probe ASoC: rockchip: i2s: Fix error code when fail to read I2S_CLR Julia Lawall (2): ASoC: amd: acp: fix typo in comment ASoC: stm32: dfsdm: fix typo in comment Krzysztof Kozlowski (7): ASoC: samsung: s3c-i2s-v2: Allow build for unsupported hardware ASoC: samsung: s3c24xx-i2s: Drop unneeded gpio.h include ASoC: samsung: Enable compile test ASoC: samsung: h1940_uda1380: include proepr GPIO consumer header ASoC: dt-bindings: qcom,sm8250: add SDM845 sound ASoC: dt-bindings: qcom,wcd934x: use absolute path to other schema ASoC: dt-bindings: use spi-peripheral-props.yaml Kuninori Morimoto (11): ASoC: simple-card-utils: rename asoc_simple_init_dai_link_params() to asoc_simple_init_for_codec2codec() ASoC: ak4613: cares Simple-Audio-Card case for TDM ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() ASoC: soc-core.c: share code for snd_soc_of_get_dai_link_cpus/codecs() ASoC: audio-graph-card2.c: use of_property_read_u32() for rate ASoC: audio-graph-card2.c: make Codec2Codec settings optional ASoC: audio-graph-card2.c: remove pre-alloced Codec2Codec space ASoC: audio-graph-card2-custom-sample.dtsi: add verbose explanation ASoC: simple-card-utils.c: ignore Codec2Codec setting if it already have ASoC: simple-card-utils.c: care Codec2Codec vs DPCM:BE ASoC: rsnd: care default case on rsnd_ssiu_busif_err_irq_ctrl() Li Chen (2): regmap: provide regmap_field helpers for simple bit operations ASoC: sunxi: Use {regmap/regmap_field}_{set/clear}_bits helpers Li kunyu (1): ASoC: tegra: delete a semicolon Liang He (5): ASoC: qcom: Fix missing of_node_put() in asoc_qcom_lpass_cpu_platform_probe() ASoc: audio-graph-card2: Fix refcount leak bug in __graph_get_type() ASoC: mt6359: Fix refcount leak bug ASoC: audio-graph-card: Add of_node_put() in fail path ASoC: audio-graph-card2: Add of_node_put() in fail path Libin Yang (2): ASoC: SOF: ipc4-topology: check dai->private in ipc_free() ASoC: SOF: ipc4-topology: free memories allocated in sof_ipc4_get_audio_fmt Lukas Bulwahn (1): ASoC: Intel: avs: correct config reference for I2S test board Marco Felsch (2): ASoC: fsl_sai: use local device pointer ASoC: fsl_sai: add error message in case of missing imx-pcm-dma support Marek Vasut (1): ASoC: doc: Update dead links Mario Limonciello (1): ASoC: amd: yc: Decrease level of error message Mark Brown (105): Merge existing fixes from asoc/for-5.19 into new branch ASoC: ssm2518: Use modern ASoC DAI format terminology ASoC: ssm2602: Use modern ASoC DAI format terminology ASoC: ssm4567: Use modern ASoC DAI format terminology ASoC: ops: Clarify snd_soc_info_volsw_sx() ASoC: nau8822: Don't reconfigure PLL to the same values ASoC: ops: Remove unneeded delay.h inclusion ASoC: tfa9879: Use modern ASoC DAI format terminology ASoC: max9860: Use modern ASoC DAI format terminology ASoC: wm_adsp: Fix event generation for wm_adsp_fw_put() ASoC: cx2072x: Use modern ASoC DAI format terminology ASoC: max98090: Use modern ASoC DAI format terminology ASoC: rk3328: Use modern ASoC DAI format terminology ASoC: sta32x: Use modern ASoC DAI format terminology ASoC: sta350: Use modern ASoC DAI format terminology ASoC: sti-sas: Use modern ASoC DAI format terminology ASoC: tas2552: Use modern ASoC DAI format terminology ASoC: tas2770: Use modern ASoC DAI format terminology ASoC: tas5086: Use modern ASoC DAI format terminology ASoC: tas5720: Use modern ASoC DAI format terminology ASoC: tas6424: Use modern ASoC DAI format terminology ASoC: uda1334: Use modern ASoC DAI format terminology ASoC: tlv320adc3xxx: Use modern ASoC DAI format terminology ASoC: tlv320adcx140: Use modern ASoC DAI format terminology ASoC: tlv320aic23: Use modern ASoC DAI format terminology ASoC: tlv320aic26: Use modern ASoC DAI format terminology ASoC: tlv320aic31xx: Use modern ASoC DAI format terminology ASoC: tlv320aic32x4: Use modern ASoC DAI format terminology ASoC: tlv320aic33: Use modern ASoC DAI format terminology ASoC: tlv320dac3x: Use modern ASoC DAI format terminology ALSA: hda: cirrus: Add initial DSP support and firmware loading ASoC: Intel: soc-acpi and machine driver updates ASoC: Intel: avs: Machine boards and HDA codec support OPE support on Tegra210 and later ASoC: trivial changes for cppcheck warnings Switch to use internal PLL for iMCLK ASoC: Drop some i2c noop remove callbacks Add Machine driver support for nau8825, max98560 and rt5682s, rt1019 ASoC: SOF: AMD/Mediatek updates for 5.20 ASoC: ssm: Use modern ASoC DAI format terminology ASoC: DAI clocking terminology modernisation ASoC: harden SoundWire codec/machine drivers used on Intel platforms Specify clock provider directly to CPU DAIs ASoC: codecs: Series of fixes for realtek codecs used on RVPs ASoC: codecs: qualcomm move gain to S8_TLV ASoC: SOF: ipc3-dtrace: Handle race during initialization ASoC: SOF: Add support ctx_save with IPC4 ASoC: wcd9335: Fix spurious event generation ASoC: hdmi-codec: Update to modern DAI terminology ASoC: wcd938x: Fix event generation for some controls ASoC: SOF: IPC4: Add topology, control and PCM ops ASoC: Merge fixes ASoC: ops: Don't modify the driver's plaform_max when reading state ASoC: ops: Fix boolean/integer detection for simple controls ASoC: SOF: Intel: Add support for MeteorLake ASoC: Add regmap_field helpers for simple bit operations ASoC: SOF/Intel: remove __func__ from dev_dbg() ASoC: minor kernel-doc corrections ASoC: Intel: Chromebooks: remap jack pins ASoC: SOF: Intel: fix resume from hibernate ASoC: SOF/soundwire: use resume_and_get on component probe remap: Some fixes for bulk read/write callbacks in regmap_config support ASoC: Merge fixes Add MT8186 ADSP dt-binding ASoC: rockchip: i2s: switch BCLK to GPIO ASoC: Intel: haswell and broadwell boards update ASoC: Merge up fixes ASoC: sun50i-codec-analog: Internal bias support ASoC: max98396: Some assorted fixes and additions ASoC: use pm_runtime_resume_and_get() when possible Add PDM/DSD/dataline configuration support ASoC: codecs: add WSA883x support ASoC: Refactor non_legacy_dai_naming flag ASoC: dt-bindings: Add bindings for WCD9335 DAIs ASoC: soc-core.c: fixup snd_soc_of_get_dai_link_cpus() Add support of two Audio PLL source I2S driver changes for Jadeite platform ASoC: cleanups and improvements for jz4740-i2s ASoC: Intel: sof_cs42l42: adding support for ADL configuration and BT offload ASoC: audio-graph-card2.c: make Codec2Codec settings optional ASoC: Merge up fixes ASoC: SOF: ipc4-topology: fix error and memory handling ADD legacy audio driver support for rembrandt ASoC: codecs: Series of fixes for realtek codecs used on RVPs ASoC: Intel: avs: Updates and cleanups ASoC: SOF: Intel: hda: Use cold/purge boot after firmware crash ASoC: SOF: Intel: hda-dai: No need to decouple host/link DMA twice ASoC: SOF: Intel: Do not process IPC reply before firmware boot Extend ipc stream parameters sent to DSP ASoC: SOF: Intel: hda: Correct Firmware State Register use ASoC/SoundWire: Intel: add sdw BE dai trigger ASoC: Merge up revert of v5.19 version of the rockchip BCLK management Read _SUB from ACPI to be able to identify firmware ASoC: Intel: Mark BE DAIs as nonatomic for hsw and ASoC: SOF: Intel: add support for SoundWire-based HP Omen16 ASoC: nau8821: Don't unconditionally free interrupt ASoC: SOF: mediatek: fix boot sequences ASoC: SOF: Intel: updates and cleanups ASoC: dt-bindings: atmel-classd: Convert to ASoC: mediatek: Add support for MT8186 SoC ASoC: fsl: Fix sparse warning ASoC: SOF: minor updates for 5.20 ASoC: Intel: machine driver updates for 5.20 ASoC: atmel: one fix and few cleanups ASoC: atmel: one fix and one cleanup Martin Povišer (1): ASoC: Introduce 'fixup_controls' card method Mauro Carvalho Chehab (1): ASoC: wm8731: update wlf,wm8731.yaml reference Miaoqian Lin (5): ASoC: cros_ec_codec: Fix refcount leak in cros_ec_codec_platform_probe ASoC: samsung: Fix error handling in aries_audio_probe ASoC: mediatek: mt8173: Fix refcount leak in mt8173_rt5650_rt5676_dev_probe ASoC: mt6797-mt6351: Fix refcount leak in mt6797_mt6351_dev_probe ASoC: mediatek: mt8173-rt5650: Fix refcount leak in mt8173_rt5650_dev_probe Minghao Chi (2): ASoC: mediatek: remove unnecessary check of clk_disable_unprepare ASoC: imx-audmux: remove unnecessary check of clk_disable_unprepare/clk_prepare_enable Muralidhar Reddy (3): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ADL match table ASoC: Intel: sof_es8336: add support for HDMI_In capture ASoC: Intel: sof_es8336: reset the num_links during probe Nathan Chancellor (1): ASoC: amd: acp: Fix initialization of ext_intr_stat1 in i2s_irq_handler() Oder Chiou (1): ASoC: rt5640: Add the MICBIAS1 to the dapm routing Peter Ujfalusi (29): ASoC: SOF: Intel: hda-dsp: Expose hda_dsp_core_power_up() ASoC: SOF: Intel: hda-loader: Make sure that the fw load sequence is followed ASoC: SOF: Intel: hda-loader: Clarify the cl_dsp_init() flow ASoC: SOF: ipc3-topology: Move and correct size checks in sof_ipc3_control_load_bytes() ASoC: SOF: make ctx_store and ctx_restore as optional ASoC: SOF: sof_ipc_pm_ops: Add support for DSP core power management ASoC: SOF: ipc3: Add set_core_state pm_ops implementation ASoC: SOF: ipc4: Add set_core_state pm_ops implementation ASoC: SOF: Intel: Switch to use the generic pm_ops.set_core_state ASoC: SOF: ipc4: implement pm ctx_save callback ASoC: SOF: ipc3-dtrace: Introduce SOF_DTRACE_INITIALIZING state ASoC: SOF: ipc3-dtrace: Add helper function to update the sdev->host_offset ASoC: SOF: ipc3-dtrace: Return from dtrace_read if there is no new data available ASoC: SOF: debug: Clarify the IPC timeout handling path ASoC: SOF: Compile and runtime IPC version selection ASoC: SOF: Intel: IPC4: enable IMR boot ASoC: SOF: Intel: hda: Fix compressed stream position tracking ASoC: twl4030: Drop legacy, non DT boot support ASoC: SOF: Intel: cnl: Do not process IPC reply before firmware boot ASoC: SOF: Intel: hda-ipc: Do not process IPC reply before firmware boot ASoC: SOF: Intel: mtl: Do not process IPC reply before firmware boot ASoC: SOF: Intel: hda: Introduce skip_imr_boot flag ASoC: SOF: Intel: hda: Correct the ROM/FW state reporting code ASoC: SOF: Intel: hda-dai: Drop misleading comment regarding dma_data ASoC: SOF: Intel: hda-dai: Do snd_hdac_ext_stream_decouple() only once ASoC: SOF: sof-client-probes: Only load the driver if IPC3 is used ASoC: SOF: ipc3-topology: Prevent double freeing of ipc_control_data via load_bytes ASoC: SOF: ipc3-loader: Print out the non matching ext_man magic number ASoC: SOF: Intel: atom: don't keep a temporary string in fixup_tplg_name Pierre-Louis Bossart (85): ASoC: cs35l45: typo in argument definition ASoC: cs42l42: remove redundant test ASoC: wcd-mbhc-v2: remove useless initialization ASoC: wcd9335: remove redundant tests ASoC: Intel: atom: sst: remove useless initialization ASoC: Intel: atom: sst_ipc: remove redundant test ASoC: Intel: atom: sst_ipc: remove useless initializations ASoC: Intel: atom: controls: remove useless initializations ASoC: Intel: boards: reset acpi_chan_package ASoC: Intel: sof_pcm512x: remove unnecessary init ASoC: mediatek: mt8195: simplify error handling ASoC: qcom: q6dsp: q6adm: remove useless initializations ASoC: qcom: q6dsp: remove spurious space ASoC: rockchip: simplify error handling ASoC: samsung: snow: simplify error handling ASoC: meson: remove useless initialization ASoC: Intel: sof_sdw: allow HDaudio/HDMI disable ASoC: Realtek/Maxim SoundWire codecs: disable pm_runtime on remove ASoC: rt711-sdca-sdw: fix calibrate mutex initialization ASoC: Intel: sof_sdw: handle errors on card registration ASoC: rt711: fix calibrate mutex initialization ASoC: rt7*-sdw: harden jack_detect_handler ASoC: codecs: rt700/rt711/rt711-sdca: initialize workqueues in probe ASoC: codecs: rt700/rt711/rt711-sdca: resume bus/codec in .set_jack_detect ASoC: SOF: ipc3-dtrace: use pm_runtime_resume_and_get() MAINTAINERS: update ASoC/Intel/SOF maintainers ASoC: SOF: Intel: hda-dai: enhance debug messages ASoC: SOF: reduce default verbosity of IPC logs ASoC: SOF: pm: add explicit behavior for ACPI S1 and S2 ASoC: SOF: pm: add definitions for S4 and S5 states ASoC: SOF: Intel: disable IMR boot when resuming from ACPI S4 and S5 states ASoC: cs4270: update kernel-doc ASoC: sunxi: sun4i-i2s: update kernel-doc ASoC: Intel: skl_nau88l25_max98357a: remap jack pins ASoC: Intel: skl_nau88l25_ssm4567: remap jack pins ASoC: Intel: kbl_rt5663_max98927: remap jack pins ASoC: Intel: kbl_da7219_max98357a: remap jack pins ASoC: Intel: kbl_da7219_max98927: remap jack pins ASoC: Intel: kbl_rt5663_rt5514_max98927: remap jack pins ASoC: Intel: bxt_da7219_max98357a: remap jack pins ASoC: Intel: glk_rt5682_max98357a: remap jack pins ASoC: Intel: cml_rt1011_rt5682: remap jack pins ASoC: Intel: sof_cs42l42: remap jack pins ASoC: Intel: sof_da7219_max98373: remap jack pins ASoC: Intel: sof_nau8825: remap jack pins ASoC: Intel: sof_rt5682: remap jack pins ASoC: SOF: Intel: hda-dsp: report error on power-up/down ASoC: SOF: Intel: hda-stream: report error on stream not opened ASoC: SOF: Intel: hda-dai: remove use of __func__ in dev_dbg ASoC: SOF: Intel: hda-stream: remove use of __func__ in dev_dbg ASoC: SOF: Intel: mtl: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-dtrace: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-loader: remove use of __func__ in dev_dbg ASoC: SOF: ipc3-topology: remove use of __func__ in dev_dbg ASoC: SOF: ipc4-topology remove use of __func__ in dev_dbg ASoC: SOF: sof-client: remove use of __func__ in dev_dbg ASoC: SOF: ipc4: remove use of __func__ in dev_dbg ASoC: Intel: boards: hda: remove use of __func__ in dev_dbg ASoC: Intel: boards: sof_sdw: remove use of __func__ in dev_dbg ASoC: SOF: pcm: use pm_resume_and_get() on component probe ASoC: codecs: soundwire: call pm_runtime_resume() in component probe ASoC: Intel: catpt: use pm_runtime_resume_and_get() ASoC: Intel: skylake: skl-pcm: use pm_runtime_resume_and_get() ASoC: wcd-mbhc-v2: use pm_runtime_resume_and_get() ASoC: wsa881x: use pm_runtime_resume_and_get() ASoC: rockchip: i2s_tdm: use pm_runtime_resume_and_get() ASoC: fsl: fsl_sai: use pm_runtime_resume_and_get() ASoC: img: img-i2s-out: use pm_runtime_resume_and_get() ASoC: rockchip: pdm: use pm_runtime_resume_and_get() ASoC: tas2552: use pm_runtime_resume_and_get() ASoC: ti: davinci-mcasp: use pm_runtime_resume_and_get() ASoC: amd: acp-es8336: use static variables ASoC: SOF: remove warning on ABI checks ASoC: soc-pcm: demote warnings on non-atomic BE connection ASoC: Intel: sof_sdw: avoid oops in error handling ASoC: Intel: soc-acpi: add table for HP Omen 16-k0005TX ASoC: Intel: sof_sdw: add quirk for HP Omen 16-k0005TX ASoC: SOF: Intel: enable dmic handling with 2 or fewer SoundWire links ASoC: SOF: Intel: hda: don't set fw_filename ASoC: SOF: Intel: hda: only fixup topology name if not set already ASoC: SOF: probes: rename assign/free callbacks as startup/shutdown ASoC: SOF: Intel: hda-stream: test DMA buffer first in hw_params ASoC: SOF: Intel: hda: add sanity check on SSP index reported by NHLT ASoC: Intel: sof_sdw: add support for Dell SKU 0AF0 ASoC: Intel: sof_es8336: remove hard-coded SSP selection Rander Wang (3): ASoC: SOF: ipc4-pcm: set pcm rate to dai setting ASoC: SOF: add ipc4 SRC module support ASoC: SOF: ipc4-topology: set domain bit based on dp domain type Randy Dunlap (2): ASoC: max98390: use linux/gpio/consumer.h to fix build ASoC: amd: fix Jadeite kconfig warning and build errors Ranjani Sridharan (23): ASoC: SOF: Add topology tokens for IPC4 ASoC: SOF: IPC4: Introduce topology ops ASoC: SOF: ipc4-topology: Add support for parsing AIF_IN/AIF_OUT widgets ASoC: SOF: ipc4-topology: Add support for parsing DAI_IN/DAI_OUT widgets ASoC: SOF: ipc4-topology: Add prepare op for AIF type widgets ASoC: SOF: ipc4-topology: Add prepare op for DAI type widgets ASoC: SOF: ipc4-topology: Add support for parsing and preparing pga widgets ASoC: SOF: ipc4-topology: Add support for parsing mixer widgets ASoC: SOF: ipc4-topology: Add control_setup op ASoC: SOF: ipc4-topology: Add control IO ops ASoC: SOF: IPC4: Add pcm ops ASoC: SOF: ipc4-topology: Add widget_setup/widget_free ops ASoC: SOF: ipc4-topology: Add route_setup/route_free ops ASoC: SOF: ipc4-topology: Add the dai_config op ASoC: SOF: ipc4-pcm: Expose sof_ipc4_set_pipeline_state() ASoC: SOF: IPC4: set the BE DAI ops ASoC: SOF: Add ops_free ASoC: SOF: Intel: hda: init NHLT for IPC4 ASoC: SOF: Add two new structures for topology manifest data ASoC: SOF: Add a new IPC op for parsing topology manifest ASoC: SOF: ipc4-topology: Add support for SSP/DMIC DAI's AsoC: SOF: ipc4-topology: Add dai_get_clk op ASoC: SOF: topology: remove unused variable Raphael-Xu (3): ASoC: add tas2780 driver ASoC: add tas2780 DT binding ASoC: dt-bindings: tas2780: remove tas2780 and will merge it to tas27xx Ren Zhijie (1): ASoC: mediatek: mt8186: fix Kconfig dependency Rob Herring (1): ASoC: dt-bindings: sgtl5000: Add missing type to 'micbias-voltage-m-volts' Ryan Wanner (3): ASoC: dt-bindings: atmel-classd: Convert to json-schema ASoC: dt-bindings: atmel-classd: PDMIC convert to json-schema ASoC: dt-bindings: atmel-i2s: Convert to json-schema Sameer Pujar (5): ASoC: tegra: Add binding doc for OPE module ASoC: tegra: Add Tegra210 based OPE driver ASoC: tegra: AHUB routes for OPE module ASoC: tegra: Fix clock DAI format on Tegra210 ASoC: tegra: Fix MBDRC bypass mode check Sascha Hauer (2): ASoC: fsl_asrc_dma: Use dmaengine_terminate_async() ASoC: audio_graph_card2: Fix port numbers in example Seven Lee (1): ASoC: nau8821: Add headset button detection Shengjiu Wang (30): ASoC: fsl_asrc_dma: enable dual fifo for ASRC P2P ASoC: fsl_mqs: simplify the code with adding fsl_mqs_soc_data ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode ASoC: dt-bindings: fsl,mqs: Add compatible string for i.MX93 platform ASoC: fsl_mqs: Add support for i.MX93 platform ASoC: fsl_sai: Add PDM daifmt support ASoC: fsl_sai: Add DSD bit format support ASoC: fsl_sai: Add support for more sample rates ASoc: fsl_sai: Add pinctrl operation for PDM and DSD ASoC: fsl_sai: Make res a member of struct fsl_sai ASoC: dt-bindings: fsl-sai: Add new property to configure dataline ASoC: fsl_sai: Configure dataline/FIFO information from dts property ASoC: fsl_micfil: change micfil default settings ASoC: dt-bindings: fsl,micfil: Convert format to json-schema ASoC: fsl_micfil: Add legacy_dai_naming flag ASoC: fsl_asrc_dma: Add legacy_dai_naming flag ASoC: fsl_utils: Add function to handle PLL clock source ASoC: fsl_spdif: Add support for PLL switch at runtime. ASoC: fsl_micfil: Add support for PLL switch at runtime ASoC: fsl_sai: Add support for PLL switch at runtime ASoC: dt-bindings: fsl_spdif: Add two PLL clock source ASoC: dt-bindings: fsl-sai: Add two PLL clock source ASoC: imx-card: Fix DSD/PDM mclk frequency ASoC: fsl_utils: Don't use plain integer as NULL pointer ASoC: fsl_utils: Drop usage of __clk_get_name() ASoC: fsl_sai: Don't use plain integer as NULL pointer ASoC: fsl_asrc: force cast the asrc_format type ASoC: fsl-asoc-card: force cast the asrc_format type ASoC: fsl_easrc: use snd_pcm_format_t type for sample_format ASoC: imx-card: use snd_pcm_format_t type for asrc_format Shuming Fan (1): ASoC: rt711-sdca: fix kernel NULL pointer dereference when IO error Srinivas Kandagatla (11): ASoC: codecs: msm8916-wcd-digital: move gains from SX_TLV to S8_TLV ASoC: codecs: wcd9335: move gains from SX_TLV to S8_TLV ASoC: qdsp6: q6apm-dai: unprepare stream if its already prepared MAINTAINERS: update ASoC Qualcomm maintainer email-id ASoC: dt-bindings: Add WSA883x bindings ASoC: codecs: add wsa883x amplifier support MAINTAINERS: add ASoC Qualcomm codecs ASoC: codecs: wsa883x: add control, dapm widgets and map ASoC: codecs: wsa881x: handle timeouts in resume path ASoC: codecs: wsa883x: handle timeouts in resume path ASoC: codecs: wsa883x: add missing break statement Srinivasa Rao Mandadapu (3): ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS ASoC: qcom: lpass-platform: Update VMA access permissions in mmap callback ASoC: qcom: Add driver support for audioreach solution Stefan Binding (6): ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Add common cs35l41 enter hibernate function ASoC: cs35l41: Do not print error when waking from hibernation ASoC: ops: Fix integer detection for when max possible values > 1 ACPI: utils: Add api to read _SUB from ACPI ASoC: cs35l41: Read System Name from ACPI _SUB to identify firmware Syed Saba Kareem (5): ASoC: amd: add RPL Platform acp header file ASoC: amd: add ACP PCI driver for RPL platform ASoC: amd: add RPL Platform init/de-init functions ASoC: amd: add RPL Platform pci driver pm-ops ASoC: amd: enable RPL Platform acp drivers build Tang Bin (1): ASoC: stm32: sai: Remove useless define Tinghan Shen (4): dt-bindings: dsp: mediatek: Use meaningful names for mbox firmware: mediatek: Use meaningful names for mbox dt-bindings: dsp: mediatek: Add mt8186 dsp document ASoC: SOF: mediatek: Align mt8186 clock names with dt-bindings Tom Rix (2): ASoC: samsung: change gpiod_speaker_power and rx1950_audio from global to static variables ASoC: samsung: change neo1973_audio from a global to static Uwe Kleine-König (17): ASoC: ak4642: Drop no-op remove function ASoC: da7219: Drop no-op remove function ASoC: lm49453: Drop no-op remove function ASoC: da732x: Drop no-op remove function ASoC: cs35l45: Make cs35l45_remove() return void ASoC: Intel: broadwell: Make broadwell_disable_jack() return void ASoC: simple-card-utils: Make asoc_simple_clean_reference() return void ASoC: core: Make snd_soc_unregister_card() return void ASoC: amd: acp: Fix error handling in .remove() ASoC: topology: KUnit: Followup prototype change of snd_soc_unregister_card() ASoC: xilinx: Suppress second error message about reset failure in .remove() ASoC: tegra: tegra20_das: Fold header file into only user ASoC: tegra: tegra20_das: Remove unused function tegra20_das_read ASoC: tegra: tegra20_das: Get rid of global pointer for driver data ASoC: tegra: tegra20_das: Make helper functions return void ASoC: tegra: tegra20_das: Drop write-only driver data member ASoC: rsnd: Emit useful error messages in .remove() V sujith kumar Reddy (5): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance ASoC: amd: acp: Remove rt1019_1 codec conf from machine driver ASoC: amd: acp: ACP code generic to support newer platforms ASoC: amd: acp: Add legacy audio driver support for Rembrandt platform Venkata Prasad Potturu (4): ASoC: amd: acp: Modify local variables name to generic ASoC: amd: acp: Drop superfluous mmap callback ASoC: amd: acp: Modify const resource struct variable to generic ASoC: amd: acp: Add error handling cases Vijendar Mukunda (7): ASoC: amd: add I2S MICSP instance support ASoC: amd: add Machine driver for Jadeite platform ASoC: amd: enable machine driver build for Jadeite platform ASoC: amd: fix ACPI dependency compile errors and warnings ASoC: amd: remove unused header file inclusion ASoC: amd: drop machine driver remove function ASoC: amd: fix for variable set but not used warning Wallace Lin (1): ASoC: nau8821: add new widget to control system clock YC Hung (7): ASoC: SOF: mediatek: revise mt8195 clock sequence ASoC: SOF: mediatek: Add shared_size for mediatek common chip information ASoC: SOF: mediatek: mt8195 modify dram type as non-cache ASoC: SOF: mediatek: mt8195 suspend check dsp idle ASoC: SOF: mediatek: Revise mt8195 boot flow ASoC: SOF: mediatek: fix mt8195 StatvectorSel wrong setting ASoC: SOF: Add cont_update_posn to platform parameters Yang Li (1): ASoC: amd: Remove duplicated include in acp-es8336.c Yang Yingliang (1): ASoC: fsl: pcm030-audio-fabric: use platform_device_unregsiter() Yassine Oudjana (3): ASoC: wcd9335: Remove RX channel from old list before adding it to a new one ASoC: dt-bindings: Add bindings for WCD9335 DAIs ASoC: wcd9335: Use DT bindings instead of local DAI definitions Yong Zhi (6): ASoC: Intel: boards: rename RTL1019 compatible driver to rt1019p ASoC: intel: sof_sdw: add RT711 SDCA card for MTL platform ASoC: SOF: Intel: MTL: remove redundant snd_sof_dsp_read() call ASoC: Intel: sof_rt5682: Add support for mtl_mx98357_rt5682 ASoC: Intel: sof_rt5682: Perform quirk check first in card late probe ASoC: Intel: sof_nau8825: Move quirk check to the front in late probe Zhongjun Tan (1): ASoC: mediatek: mt8186: Remove condition with no effect Zhu Ning (1): ASoC: codes: Add support for ES8316 producer mode syed sabakareem (1): ASoC: amd: yc: Update DMI table entries xliu (1): ASoC: Intel: cirrus-common: fix incorrect channel mapping zhangqilong (1): ASoC: fsl_xcvr:Fix unbalanced pm_runtime_enable in fsl_xcvr_probe .../bindings/dsp/mediatek,mt8186-dsp.yaml | 91 + .../bindings/dsp/mediatek,mt8195-dsp.yaml | 10 +- .../devicetree/bindings/sound/adi,adau1977.yaml | 7 +- .../devicetree/bindings/sound/adi,max98396.yaml | 30 +- .../sound/allwinner,sun50i-a64-codec-analog.yaml | 5 + .../bindings/sound/atmel,sama5d2-classd.yaml | 100 + .../bindings/sound/atmel,sama5d2-i2s.yaml | 85 + .../bindings/sound/atmel,sama5d2-pdmic.yaml | 98 + .../devicetree/bindings/sound/atmel-classd.txt | 55 - .../devicetree/bindings/sound/atmel-i2s.txt | 46 - .../devicetree/bindings/sound/atmel-pdmic.txt | 55 - .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 2 +- .../devicetree/bindings/sound/designware-i2s.txt | 35 - .../devicetree/bindings/sound/fsl,micfil.txt | 33 - .../devicetree/bindings/sound/fsl,micfil.yaml | 85 + .../devicetree/bindings/sound/fsl,mqs.txt | 2 +- .../devicetree/bindings/sound/fsl,spdif.yaml | 4 + .../devicetree/bindings/sound/fsl-sai.txt | 11 + Documentation/devicetree/bindings/sound/mt6358.txt | 4 +- .../devicetree/bindings/sound/mt8186-afe-pcm.yaml | 175 ++ .../sound/mt8186-mt6366-da7219-max98357.yaml | 75 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 75 + .../devicetree/bindings/sound/nau8821.txt | 2 +- .../bindings/sound/nvidia,tegra210-ahub.yaml | 4 + .../bindings/sound/nvidia,tegra210-mbdrc.yaml | 47 + .../bindings/sound/nvidia,tegra210-ope.yaml | 87 + .../bindings/sound/nvidia,tegra210-peq.yaml | 48 + .../devicetree/bindings/sound/nxp,tfa989x.yaml | 1 + .../devicetree/bindings/sound/qcom,sdm845.txt | 91 - .../devicetree/bindings/sound/qcom,sm8250.yaml | 3 + .../devicetree/bindings/sound/qcom,wcd934x.yaml | 2 +- .../devicetree/bindings/sound/qcom,wsa883x.yaml | 74 + .../devicetree/bindings/sound/rockchip-i2s.yaml | 7 + .../devicetree/bindings/sound/sgtl5000.yaml | 1 + .../bindings/sound/snps,designware-i2s.yaml | 94 + .../devicetree/bindings/sound/wlf,wm8731.yaml | 9 +- Documentation/process/kernel-docs.rst | 2 +- Documentation/sound/soc/codec.rst | 2 +- Documentation/sound/soc/platform.rst | 2 +- MAINTAINERS | 5 + drivers/acpi/utils.c | 38 + drivers/base/regmap/regmap.c | 37 +- drivers/firmware/cirrus/cs_dsp.c | 107 + drivers/firmware/mtk-adsp-ipc.c | 36 +- drivers/gpu/drm/bridge/sii902x.c | 5 +- .../gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio.c | 2 +- drivers/gpu/drm/exynos/exynos_hdmi.c | 8 +- drivers/gpu/drm/i2c/tda998x_drv.c | 6 +- drivers/gpu/drm/sti/sti_hdmi.c | 8 +- drivers/gpu/drm/vc4/vc4_hdmi.c | 1 + drivers/media/i2c/tda1997x.c | 1 - drivers/soundwire/intel.c | 14 +- drivers/soundwire/qcom.c | 10 + include/dt-bindings/sound/qcom,wcd9335.h | 15 + include/linux/acpi.h | 6 + include/linux/firmware/cirrus/cs_dsp.h | 77 +- include/linux/regmap.h | 37 + include/linux/soundwire/sdw_intel.h | 3 + include/sound/cs35l41.h | 3 + include/sound/hdmi-codec.h | 4 +- include/sound/madera-pdata.h | 2 +- include/sound/pcm.h | 19 +- include/sound/simple_card_utils.h | 5 +- include/sound/soc-acpi-intel-match.h | 2 + include/sound/soc-card.h | 1 + include/sound/soc-component.h | 7 +- include/sound/soc-dai.h | 6 + include/sound/soc.h | 15 +- include/sound/sof.h | 1 + include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai-intel.h | 2 + include/sound/sof/dai.h | 2 +- include/sound/sof/ipc4/header.h | 8 + include/sound/sof/stream.h | 6 +- include/uapi/sound/sof/abi.h | 4 +- include/uapi/sound/sof/header.h | 30 + include/uapi/sound/sof/tokens.h | 44 + sound/hda/intel-dsp-config.c | 5 + sound/pci/ens1370.c | 2 +- sound/soc/Makefile | 4 +- sound/soc/adi/axi-i2s.c | 1 + sound/soc/adi/axi-spdif.c | 1 + sound/soc/amd/Kconfig | 22 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-config.c | 30 + sound/soc/amd/acp-es8336.c | 318 +++ sound/soc/amd/acp-pcm-dma.c | 50 +- sound/soc/amd/acp.h | 13 + sound/soc/amd/acp/Kconfig | 12 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 169 +- sound/soc/amd/acp/acp-legacy-mach.c | 32 + sound/soc/amd/acp/acp-mach-common.c | 301 +- sound/soc/amd/acp/acp-mach.h | 9 + sound/soc/amd/acp/acp-pci.c | 35 +- sound/soc/amd/acp/acp-pdm.c | 10 +- sound/soc/amd/acp/acp-platform.c | 55 +- sound/soc/amd/acp/acp-rembrandt.c | 401 +++ sound/soc/amd/acp/acp-renoir.c | 48 +- sound/soc/amd/acp/acp-sof-mach.c | 30 + sound/soc/amd/acp/amd.h | 86 +- sound/soc/amd/acp/chip_offset_byte.h | 40 +- sound/soc/amd/mach-config.h | 1 + sound/soc/amd/raven/acp3x-i2s.c | 3 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 13 +- sound/soc/amd/rpl/Makefile | 5 + sound/soc/amd/rpl/rpl-pci-acp6x.c | 227 ++ sound/soc/amd/rpl/rpl_acp6x.h | 36 + sound/soc/amd/rpl/rpl_acp6x_chip_offset_byte.h | 30 + sound/soc/amd/vangogh/acp5x-i2s.c | 5 +- sound/soc/amd/vangogh/acp5x-mach.c | 3 +- sound/soc/amd/yc/acp6x-mach.c | 32 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 13 +- sound/soc/amd/yc/pci-acp6x.c | 2 +- sound/soc/atmel/atmel-classd.c | 1 + sound/soc/atmel/atmel-i2s.c | 7 +- sound/soc/atmel/atmel-pdmic.c | 1 + sound/soc/atmel/atmel_ssc_dai.c | 30 +- sound/soc/atmel/mchp-i2s-mcc.c | 11 +- sound/soc/atmel/mchp-pdmc.c | 7 +- sound/soc/atmel/mchp-spdifrx.c | 22 +- sound/soc/atmel/mchp-spdiftx.c | 25 +- sound/soc/atmel/mikroe-proto.c | 4 +- sound/soc/au1x/ac97c.c | 3 +- sound/soc/au1x/i2sc.c | 5 +- sound/soc/au1x/psc-ac97.c | 3 +- sound/soc/au1x/psc-i2s.c | 7 +- sound/soc/bcm/bcm2835-i2s.c | 23 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 1 + sound/soc/bcm/cygnus-ssp.c | 11 +- sound/soc/cirrus/ep93xx-ac97.c | 3 +- sound/soc/cirrus/ep93xx-i2s.c | 11 +- sound/soc/codecs/88pm860x-codec.c | 1 - sound/soc/codecs/Kconfig | 28 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ab8500-codec.c | 3 - sound/soc/codecs/ab8500-codec.h | 2 - sound/soc/codecs/ac97.c | 1 - sound/soc/codecs/ad1836.c | 1 - sound/soc/codecs/ad193x.c | 1 - sound/soc/codecs/ad1980.c | 1 - sound/soc/codecs/ad73311.c | 1 - sound/soc/codecs/adau1373.c | 1 - sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761.c | 1 - sound/soc/codecs/adau1781.c | 1 - sound/soc/codecs/adau1977.c | 1 - sound/soc/codecs/adau7002.c | 1 - sound/soc/codecs/adau7118.c | 1 - sound/soc/codecs/adav80x.c | 1 - sound/soc/codecs/ads117x.c | 1 - sound/soc/codecs/ak4104.c | 1 - sound/soc/codecs/ak4118.c | 1 - sound/soc/codecs/ak4375.c | 1 - sound/soc/codecs/ak4458.c | 2 - sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4554.c | 1 - sound/soc/codecs/ak4613.c | 7 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 1 - sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/ak5386.c | 1 - sound/soc/codecs/ak5558.c | 2 - sound/soc/codecs/alc5623.c | 1 - sound/soc/codecs/alc5632.c | 1 - sound/soc/codecs/bd28623.c | 1 - sound/soc/codecs/bt-sco.c | 1 - sound/soc/codecs/cpcap.c | 1 - sound/soc/codecs/cq93vc.c | 1 - sound/soc/codecs/cros_ec_codec.c | 1 + sound/soc/codecs/cs35l32.c | 1 - sound/soc/codecs/cs35l33.c | 1 - sound/soc/codecs/cs35l34.c | 1 - sound/soc/codecs/cs35l35.c | 1 - sound/soc/codecs/cs35l36.c | 1 - sound/soc/codecs/cs35l41-lib.c | 82 +- sound/soc/codecs/cs35l41-spi.c | 1 + sound/soc/codecs/cs35l41.c | 101 +- sound/soc/codecs/cs35l45-i2c.c | 4 +- sound/soc/codecs/cs35l45.c | 6 +- sound/soc/codecs/cs35l45.h | 4 +- sound/soc/codecs/cs4234.c | 1 - sound/soc/codecs/cs4265.c | 1 - sound/soc/codecs/cs4270.c | 2 - sound/soc/codecs/cs4271.c | 1 - sound/soc/codecs/cs42l42.c | 4 +- sound/soc/codecs/cs42l51.c | 1 - sound/soc/codecs/cs42l52.c | 1 - sound/soc/codecs/cs42l56.c | 1 - sound/soc/codecs/cs42l73.c | 1 - sound/soc/codecs/cs42xx8.c | 1 - sound/soc/codecs/cs43130.c | 1 - sound/soc/codecs/cs4341.c | 1 - sound/soc/codecs/cs4349.c | 1 - sound/soc/codecs/cs47l15.c | 1 - sound/soc/codecs/cs47l24.c | 1 - sound/soc/codecs/cs47l35.c | 1 - sound/soc/codecs/cs47l85.c | 1 - sound/soc/codecs/cs47l90.c | 1 - sound/soc/codecs/cs47l92.c | 1 - sound/soc/codecs/cs53l30.c | 1 - sound/soc/codecs/cx20442.c | 1 - sound/soc/codecs/cx2072x.c | 17 +- sound/soc/codecs/da7210.c | 3 +- sound/soc/codecs/da7213.c | 1 - sound/soc/codecs/da7218.c | 1 - sound/soc/codecs/da7219.c | 7 - sound/soc/codecs/da732x.c | 7 - sound/soc/codecs/da9055.c | 1 - sound/soc/codecs/dmic.c | 1 - sound/soc/codecs/es7134.c | 1 - sound/soc/codecs/es7241.c | 1 - sound/soc/codecs/es8316.c | 21 +- sound/soc/codecs/es8328.c | 1 - sound/soc/codecs/gtm601.c | 1 - sound/soc/codecs/hda-dai.c | 102 + sound/soc/codecs/hda.c | 395 +++ sound/soc/codecs/hda.h | 19 + sound/soc/codecs/hdac_hdmi.c | 1 - sound/soc/codecs/hdmi-codec.c | 19 +- sound/soc/codecs/ics43432.c | 1 - sound/soc/codecs/inno_rk3036.c | 1 - sound/soc/codecs/isabelle.c | 1 - sound/soc/codecs/jz4740.c | 2 - sound/soc/codecs/lm49453.c | 7 - sound/soc/codecs/lochnagar-sc.c | 1 - sound/soc/codecs/max98088.c | 33 +- sound/soc/codecs/max98090.c | 13 +- sound/soc/codecs/max98095.c | 1 - sound/soc/codecs/max98357a.c | 1 - sound/soc/codecs/max98371.c | 1 - sound/soc/codecs/max98373.c | 16 +- sound/soc/codecs/max98390.c | 3 +- sound/soc/codecs/max98396.c | 271 +- sound/soc/codecs/max98396.h | 10 +- sound/soc/codecs/max9850.c | 1 - sound/soc/codecs/max98520.c | 1 - sound/soc/codecs/max9860.c | 7 +- sound/soc/codecs/max9867.c | 1 - sound/soc/codecs/max98925.c | 1 - sound/soc/codecs/max98926.c | 1 - sound/soc/codecs/max98927.c | 1 - sound/soc/codecs/mc13783.c | 1 - sound/soc/codecs/ml26124.c | 1 - sound/soc/codecs/msm8916-wcd-analog.c | 1 - sound/soc/codecs/msm8916-wcd-digital.c | 47 +- sound/soc/codecs/mt6358.c | 1 + sound/soc/codecs/mt6359-accdet.c | 1 + sound/soc/codecs/mt6359.c | 1 + sound/soc/codecs/nau8315.c | 1 - sound/soc/codecs/nau8540.c | 1 - sound/soc/codecs/nau8810.c | 1 - sound/soc/codecs/nau8821.c | 76 +- sound/soc/codecs/nau8821.h | 1 + sound/soc/codecs/nau8822.c | 15 +- sound/soc/codecs/nau8822.h | 2 + sound/soc/codecs/nau8824.c | 1 - sound/soc/codecs/nau8825.c | 3 +- sound/soc/codecs/pcm1681.c | 1 - sound/soc/codecs/pcm1789.c | 1 - sound/soc/codecs/pcm179x.c | 1 - sound/soc/codecs/pcm186x.c | 2 - sound/soc/codecs/pcm3008.c | 1 - sound/soc/codecs/pcm3168a.c | 1 - sound/soc/codecs/pcm5102a.c | 1 - sound/soc/codecs/pcm512x.c | 1 - sound/soc/codecs/rk3328_codec.c | 6 +- sound/soc/codecs/rk817_codec.c | 1 - sound/soc/codecs/rt1011.c | 1 - sound/soc/codecs/rt1015.c | 1 - sound/soc/codecs/rt1015p.c | 1 - sound/soc/codecs/rt1016.c | 1 - sound/soc/codecs/rt1019.c | 1 - sound/soc/codecs/rt1305.c | 1 - sound/soc/codecs/rt1308-sdw.c | 12 + sound/soc/codecs/rt1308.c | 1 - sound/soc/codecs/rt1316-sdw.c | 12 + sound/soc/codecs/rt274.c | 11 +- sound/soc/codecs/rt286.c | 19 +- sound/soc/codecs/rt286.h | 2 - sound/soc/codecs/rt298.c | 61 +- sound/soc/codecs/rt298.h | 2 - sound/soc/codecs/rt5514.c | 1 - sound/soc/codecs/rt5616.c | 1 - sound/soc/codecs/rt5631.c | 1 - sound/soc/codecs/rt5640.c | 13 +- sound/soc/codecs/rt5645.c | 1 - sound/soc/codecs/rt5651.c | 1 - sound/soc/codecs/rt5659.c | 1 - sound/soc/codecs/rt5660.c | 1 - sound/soc/codecs/rt5663.c | 1 - sound/soc/codecs/rt5665.c | 1 - sound/soc/codecs/rt5668.c | 1 - sound/soc/codecs/rt5670.c | 1 - sound/soc/codecs/rt5677.c | 1 - sound/soc/codecs/rt5682.c | 1 - sound/soc/codecs/rt5682s.c | 1 - sound/soc/codecs/rt700.c | 5 + sound/soc/codecs/rt711-sdca.c | 5 + sound/soc/codecs/rt711.c | 5 + sound/soc/codecs/rt715-sdca.c | 12 + sound/soc/codecs/rt715.c | 12 + sound/soc/codecs/sgtl5000.c | 1 - sound/soc/codecs/si476x.c | 1 - sound/soc/codecs/spdif_receiver.c | 1 - sound/soc/codecs/spdif_transmitter.c | 1 - sound/soc/codecs/ssm2518.c | 5 +- sound/soc/codecs/ssm2602.c | 7 +- sound/soc/codecs/ssm4567.c | 5 +- sound/soc/codecs/sta32x.c | 5 +- sound/soc/codecs/sta350.c | 5 +- sound/soc/codecs/sta529.c | 1 - sound/soc/codecs/stac9766.c | 2 - sound/soc/codecs/sti-sas.c | 7 +- sound/soc/codecs/tas2552.c | 13 +- sound/soc/codecs/tas2562.c | 2 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 7 +- sound/soc/codecs/tas2780.c | 663 +++++ sound/soc/codecs/tas2780.h | 101 + sound/soc/codecs/tas5086.c | 3 +- sound/soc/codecs/tas571x.c | 1 - sound/soc/codecs/tas5720.c | 6 +- sound/soc/codecs/tas5805m.c | 1 - sound/soc/codecs/tas6424.c | 7 +- sound/soc/codecs/tfa9879.c | 5 +- sound/soc/codecs/tfa989x.c | 31 +- sound/soc/codecs/tlv320adc3xxx.c | 3 +- sound/soc/codecs/tlv320adcx140.c | 11 +- sound/soc/codecs/tlv320aic23.c | 8 +- sound/soc/codecs/tlv320aic26.c | 16 +- sound/soc/codecs/tlv320aic31xx.c | 20 +- sound/soc/codecs/tlv320aic32x4.c | 11 +- sound/soc/codecs/tlv320aic3x.c | 12 +- sound/soc/codecs/tlv320dac33.c | 13 +- sound/soc/codecs/tscs42xx.c | 1 - sound/soc/codecs/twl4030.c | 102 +- sound/soc/codecs/twl6040.c | 1 - sound/soc/codecs/uda1334.c | 3 +- sound/soc/codecs/uda134x.c | 1 - sound/soc/codecs/uda1380.c | 1 - sound/soc/codecs/wcd-mbhc-v2.c | 12 +- sound/soc/codecs/wcd9335.c | 98 +- sound/soc/codecs/wl1273.c | 1 - sound/soc/codecs/wm0010.c | 1 - sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 1 - sound/soc/codecs/wm5100.c | 1 - sound/soc/codecs/wm5102.c | 1 - sound/soc/codecs/wm5110.c | 1 - sound/soc/codecs/wm8350.c | 1 - sound/soc/codecs/wm8400.c | 1 - sound/soc/codecs/wm8510.c | 1 - sound/soc/codecs/wm8523.c | 1 - sound/soc/codecs/wm8524.c | 1 - sound/soc/codecs/wm8580.c | 1 - sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8727.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 1 - sound/soc/codecs/wm8737.c | 1 - sound/soc/codecs/wm8741.c | 1 - sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm8770.c | 1 - sound/soc/codecs/wm8776.c | 1 - sound/soc/codecs/wm8782.c | 1 - sound/soc/codecs/wm8804.c | 1 - sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 1 - sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 1 - sound/soc/codecs/wm8961.c | 1 - sound/soc/codecs/wm8962.c | 1 - sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 1 - sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 1 - sound/soc/codecs/wm8991.c | 1 - sound/soc/codecs/wm8993.c | 1 - sound/soc/codecs/wm8994.c | 1 - sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 2 - sound/soc/codecs/wm8997.c | 1 - sound/soc/codecs/wm8998.c | 1 - sound/soc/codecs/wm9081.c | 1 - sound/soc/codecs/wm9090.c | 1 - sound/soc/codecs/wm9705.c | 1 - sound/soc/codecs/wm9712.c | 1 - sound/soc/codecs/wm9713.c | 1 - sound/soc/codecs/wm_adsp.c | 25 +- sound/soc/codecs/wsa881x.c | 16 +- sound/soc/codecs/wsa883x.c | 1511 ++++++++++ sound/soc/codecs/zl38060.c | 1 - sound/soc/dwc/dwc-i2s.c | 15 +- sound/soc/fsl/Kconfig | 3 + sound/soc/fsl/fsl-asoc-card.c | 5 +- sound/soc/fsl/fsl_asrc.c | 6 +- sound/soc/fsl/fsl_asrc_dma.c | 19 +- sound/soc/fsl/fsl_aud2htx.c | 3 +- sound/soc/fsl/fsl_audmix.c | 6 +- sound/soc/fsl/fsl_easrc.c | 16 +- sound/soc/fsl/fsl_easrc.h | 2 +- sound/soc/fsl/fsl_esai.c | 11 +- sound/soc/fsl/fsl_micfil.c | 55 +- sound/soc/fsl/fsl_micfil.h | 9 + sound/soc/fsl/fsl_mqs.c | 136 +- sound/soc/fsl/fsl_rpmsg.c | 3 +- sound/soc/fsl/fsl_sai.c | 375 ++- sound/soc/fsl/fsl_sai.h | 28 +- sound/soc/fsl/fsl_spdif.c | 51 +- sound/soc/fsl/fsl_ssi.c | 23 +- sound/soc/fsl/fsl_utils.c | 69 + sound/soc/fsl/fsl_utils.h | 7 + sound/soc/fsl/fsl_xcvr.c | 12 +- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/fsl/imx-audmux.c | 24 +- sound/soc/fsl/imx-card.c | 24 +- sound/soc/fsl/mpc5200_psc_i2s.c | 3 +- sound/soc/fsl/pcm030-audio-fabric.c | 8 +- sound/soc/generic/audio-graph-card.c | 4 +- .../generic/audio-graph-card2-custom-sample.dtsi | 101 +- sound/soc/generic/audio-graph-card2.c | 78 +- sound/soc/generic/simple-card-utils.c | 44 +- sound/soc/generic/test-component.c | 20 +- sound/soc/hisilicon/hi6210-i2s.c | 19 +- sound/soc/img/img-i2s-in.c | 7 +- sound/soc/img/img-i2s-out.c | 21 +- sound/soc/img/img-parallel-out.c | 3 +- sound/soc/img/img-spdif-in.c | 3 +- sound/soc/img/img-spdif-out.c | 3 +- sound/soc/img/pistachio-internal-dac.c | 1 - sound/soc/intel/Kconfig | 5 +- sound/soc/intel/atom/sst-atom-controls.c | 8 +- sound/soc/intel/atom/sst/sst.c | 2 +- sound/soc/intel/atom/sst/sst_ipc.c | 8 +- sound/soc/intel/avs/Makefile | 3 + sound/soc/intel/avs/boards/Kconfig | 121 + sound/soc/intel/avs/boards/Makefile | 27 + sound/soc/intel/avs/boards/da7219.c | 282 ++ sound/soc/intel/avs/boards/dmic.c | 93 + sound/soc/intel/avs/boards/hdaudio.c | 294 ++ sound/soc/intel/avs/boards/i2s_test.c | 180 ++ sound/soc/intel/avs/boards/max98357a.c | 154 + sound/soc/intel/avs/boards/max98373.c | 239 ++ sound/soc/intel/avs/boards/nau8825.c | 353 +++ sound/soc/intel/avs/boards/rt274.c | 310 ++ sound/soc/intel/avs/boards/rt286.c | 281 ++ sound/soc/intel/avs/boards/rt298.c | 281 ++ sound/soc/intel/avs/boards/rt5682.c | 340 +++ sound/soc/intel/avs/boards/ssm4567.c | 271 ++ sound/soc/intel/avs/cldma.c | 12 +- sound/soc/intel/avs/core.c | 13 +- sound/soc/intel/avs/dsp.c | 11 +- sound/soc/intel/avs/ipc.c | 1 + sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 18 +- sound/soc/intel/avs/path.c | 54 +- sound/soc/intel/avs/pcm.c | 2 - sound/soc/intel/avs/topology.c | 27 + sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bdw-rt5650.c | 1 + sound/soc/intel/boards/bdw-rt5677.c | 1 + sound/soc/intel/boards/bdw_rt286.c | 280 ++ sound/soc/intel/boards/broadwell.c | 336 --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcht_nocodec.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 3 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 8 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 23 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 21 +- sound/soc/intel/boards/haswell.c | 202 -- sound/soc/intel/boards/hda_dsp_common.c | 4 +- sound/soc/intel/boards/hsw_rt5640.c | 177 ++ sound/soc/intel/boards/kbl_da7219_max98357a.c | 21 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 21 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 21 +- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 21 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 19 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 19 +- sound/soc/intel/boards/skl_rt286.c | 2 +- sound/soc/intel/boards/sof_cs42l42.c | 109 +- sound/soc/intel/boards/sof_da7219_max98373.c | 23 +- sound/soc/intel/boards/sof_es8336.c | 160 +- sound/soc/intel/boards/sof_nau8825.c | 33 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 24 +- sound/soc/intel/boards/sof_realtek_common.h | 6 +- sound/soc/intel/boards/sof_rt5682.c | 51 +- sound/soc/intel/boards/sof_sdw.c | 53 +- sound/soc/intel/boards/sof_sdw_rt711.c | 3 + sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 3 + sound/soc/intel/catpt/device.c | 5 +- sound/soc/intel/catpt/pcm.c | 26 +- sound/soc/intel/catpt/sysfs.c | 4 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-adl-match.c | 61 + .../intel/common/soc-acpi-intel-hsw-bdw-match.c | 6 +- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 89 + sound/soc/intel/keembay/kmb_platform.c | 18 +- sound/soc/intel/skylake/skl-pcm.c | 5 +- sound/soc/intel/skylake/skl-topology.c | 6 - sound/soc/jz4740/Kconfig | 2 +- sound/soc/jz4740/jz4740-i2s.c | 44 +- sound/soc/mediatek/Kconfig | 45 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/Makefile | 2 +- sound/soc/mediatek/common/mtk-dsp-sof-common.c | 196 ++ sound/soc/mediatek/common/mtk-dsp-sof-common.h | 36 + sound/soc/mediatek/common/mtk-soc-card.h | 17 + sound/soc/mediatek/mt6797/mt6797-mt6351.c | 6 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 6 +- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 9 +- sound/soc/mediatek/mt8186/Makefile | 22 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 652 +++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 106 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 195 ++ sound/soc/mediatek/mt8186/mt8186-afe-control.c | 255 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 243 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3000 ++++++++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 865 ++++++ sound/soc/mediatek/mt8186/mt8186-dai-hostless.c | 298 ++ sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 236 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1223 ++++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 418 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 695 +++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 645 +++++ sound/soc/mediatek/mt8186/mt8186-interconnection.h | 69 + sound/soc/mediatek/mt8186/mt8186-misc-control.c | 252 ++ sound/soc/mediatek/mt8186/mt8186-mt6366-common.c | 57 + sound/soc/mediatek/mt8186/mt8186-mt6366-common.h | 17 + .../mt8186/mt8186-mt6366-da7219-max98357.c | 1002 +++++++ .../mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c | 978 +++++++ sound/soc/mediatek/mt8186/mt8186-reg.h | 2913 +++++++++++++++++++ sound/soc/mediatek/mt8195/mt8195-afe-clk.c | 8 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 6 +- sound/soc/mediatek/mt8195/mt8195-dai-pcm.c | 6 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 233 +- sound/soc/meson/aiu-acodec-ctrl.c | 1 - sound/soc/meson/aiu-codec-ctrl.c | 1 - sound/soc/meson/aiu-encoder-i2s.c | 2 +- sound/soc/meson/axg-frddr.c | 3 + sound/soc/meson/axg-pdm.c | 4 +- sound/soc/meson/axg-spdifin.c | 1 + sound/soc/meson/axg-spdifout.c | 1 + sound/soc/meson/axg-tdm-interface.c | 14 +- sound/soc/meson/axg-toddr.c | 3 + sound/soc/meson/g12a-toacodec.c | 2 - sound/soc/meson/g12a-tohdmitx.c | 1 - sound/soc/meson/meson-codec-glue.c | 2 +- sound/soc/meson/t9015.c | 1 - sound/soc/mxs/mxs-saif.c | 7 +- sound/soc/pxa/magician.c | 8 +- sound/soc/pxa/mmp-sspa.c | 15 +- sound/soc/pxa/pxa-ssp.c | 43 +- sound/soc/pxa/pxa2xx-i2s.c | 27 +- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 7 + sound/soc/qcom/qdsp6/audioreach.c | 4 +- sound/soc/qcom/qdsp6/q6adm.c | 8 +- sound/soc/qcom/qdsp6/q6afe.c | 6 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 23 +- sound/soc/qcom/qdsp6/q6asm.c | 2 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sc7280.c | 33 + sound/soc/qcom/sdm845.c | 6 +- sound/soc/qcom/sm8250.c | 4 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rockchip_i2s.c | 182 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 13 +- sound/soc/rockchip/rockchip_pdm.c | 7 +- sound/soc/rockchip/rockchip_spdif.c | 1 + sound/soc/samsung/Kconfig | 20 +- sound/soc/samsung/aries_wm8994.c | 7 +- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 8 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- sound/soc/samsung/pcm.c | 7 +- sound/soc/samsung/rx1950_uda1380.c | 4 +- sound/soc/samsung/s3c-i2s-v2.c | 17 +- sound/soc/samsung/s3c2412-i2s.c | 7 +- sound/soc/samsung/s3c24xx-i2s.c | 14 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 7 +- sound/soc/sh/fsi.c | 6 +- sound/soc/sh/hac.c | 3 +- sound/soc/sh/rcar/core.c | 30 +- sound/soc/sh/rcar/ssiu.c | 2 + sound/soc/sh/rz-ssi.c | 11 +- sound/soc/sh/siu_pcm.c | 17 +- sound/soc/sh/ssi.c | 13 +- sound/soc/soc-card.c | 6 + sound/soc/soc-core.c | 167 +- sound/soc/soc-dai.c | 3 +- sound/soc/soc-dapm.c | 10 +- sound/soc/soc-ops.c | 51 +- sound/soc/soc-pcm.c | 3 +- sound/soc/soc-topology-test.c | 37 +- sound/soc/soc-utils.c | 1 - sound/soc/sof/Kconfig | 7 + sound/soc/sof/Makefile | 16 +- sound/soc/sof/amd/Kconfig | 1 + sound/soc/sof/amd/acp-dsp-offset.h | 2 + sound/soc/sof/amd/acp.c | 36 + sound/soc/sof/amd/acp.h | 4 + sound/soc/sof/amd/pci-rn.c | 4 + sound/soc/sof/amd/renoir.c | 4 + sound/soc/sof/compress.c | 73 +- sound/soc/sof/core.c | 7 +- sound/soc/sof/debug.c | 5 +- sound/soc/sof/imx/Kconfig | 1 + sound/soc/sof/intel/Kconfig | 27 + sound/soc/sof/intel/Makefile | 4 +- sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/atom.c | 16 +- sound/soc/sof/intel/bdw.c | 7 +- sound/soc/sof/intel/byt.c | 5 +- sound/soc/sof/intel/cnl.c | 39 +- sound/soc/sof/intel/hda-dai.c | 226 +- sound/soc/sof/intel/hda-dsp.c | 26 +- sound/soc/sof/intel/hda-ipc.c | 39 +- sound/soc/sof/intel/hda-loader.c | 28 +- sound/soc/sof/intel/hda-probes.c | 16 +- sound/soc/sof/intel/hda-stream.c | 21 +- sound/soc/sof/intel/hda.c | 235 +- sound/soc/sof/intel/hda.h | 70 + sound/soc/sof/intel/icl.c | 1 + sound/soc/sof/intel/mtl.c | 794 ++++++ sound/soc/sof/intel/mtl.h | 76 + sound/soc/sof/intel/pci-apl.c | 1 + sound/soc/sof/intel/pci-cnl.c | 1 + sound/soc/sof/intel/pci-icl.c | 1 + sound/soc/sof/intel/pci-mtl.c | 71 + sound/soc/sof/intel/pci-tgl.c | 1 + sound/soc/sof/intel/shim.h | 2 + sound/soc/sof/intel/tgl.c | 34 +- sound/soc/sof/ipc.c | 24 +- sound/soc/sof/ipc3-dtrace.c | 55 +- sound/soc/sof/ipc3-loader.c | 17 +- sound/soc/sof/ipc3-pcm.c | 11 +- sound/soc/sof/ipc3-topology.c | 73 +- sound/soc/sof/ipc3.c | 36 +- sound/soc/sof/ipc4-control.c | 216 ++ sound/soc/sof/ipc4-pcm.c | 234 ++ sound/soc/sof/ipc4-priv.h | 7 + sound/soc/sof/ipc4-topology.c | 1921 +++++++++++++ sound/soc/sof/ipc4-topology.h | 270 ++ sound/soc/sof/ipc4.c | 45 +- sound/soc/sof/mediatek/Kconfig | 1 + sound/soc/sof/mediatek/adsp_helper.h | 1 + sound/soc/sof/mediatek/mt8186/mt8186-clk.c | 4 +- sound/soc/sof/mediatek/mt8195/mt8195-clk.c | 7 + sound/soc/sof/mediatek/mt8195/mt8195-loader.c | 13 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 50 +- sound/soc/sof/mediatek/mt8195/mt8195.h | 5 + sound/soc/sof/ops.h | 6 + sound/soc/sof/pcm.c | 13 + sound/soc/sof/sof-audio.h | 12 + sound/soc/sof/sof-client-ipc-msg-injector.c | 29 +- sound/soc/sof/sof-client-probes.c | 13 +- sound/soc/sof/sof-client-probes.h | 8 +- sound/soc/sof/sof-client.c | 4 +- sound/soc/sof/sof-priv.h | 15 +- sound/soc/sof/topology.c | 121 +- sound/soc/spear/spdif_in.c | 3 +- sound/soc/spear/spdif_out.c | 3 +- sound/soc/sti/sti_uniperif.c | 3 +- sound/soc/stm/stm32_adfsdm.c | 3 +- sound/soc/stm/stm32_i2s.c | 9 +- sound/soc/stm/stm32_sai_sub.c | 11 +- sound/soc/stm/stm32_spdifrx.c | 1 + sound/soc/sunxi/sun4i-codec.c | 85 +- sound/soc/sunxi/sun4i-i2s.c | 23 +- sound/soc/sunxi/sun4i-spdif.c | 3 +- sound/soc/sunxi/sun50i-codec-analog.c | 8 + sound/soc/sunxi/sun8i-codec.c | 7 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra20_ac97.c | 5 +- sound/soc/tegra/tegra20_das.c | 198 +- sound/soc/tegra/tegra20_das.h | 120 - sound/soc/tegra/tegra20_i2s.c | 9 +- sound/soc/tegra/tegra20_spdif.c | 1 + sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 39 +- sound/soc/tegra/tegra210_i2s.c | 7 +- sound/soc/tegra/tegra210_mbdrc.c | 1014 +++++++ sound/soc/tegra/tegra210_mbdrc.h | 215 ++ sound/soc/tegra/tegra210_ope.c | 419 +++ sound/soc/tegra/tegra210_ope.h | 90 + sound/soc/tegra/tegra210_peq.c | 434 +++ sound/soc/tegra/tegra210_peq.h | 56 + sound/soc/tegra/tegra30_i2s.c | 9 +- sound/soc/ti/davinci-i2s.c | 35 +- sound/soc/ti/davinci-mcasp.c | 16 +- sound/soc/ti/davinci-vcif.c | 3 +- sound/soc/ti/omap-dmic.c | 3 +- sound/soc/ti/omap-hdmi.c | 1 + sound/soc/ti/omap-mcbsp.c | 15 +- sound/soc/ti/omap-mcpdm.c | 7 +- sound/soc/uniphier/evea.c | 1 - sound/soc/ux500/mop500.c | 2 - sound/soc/ux500/mop500_ab8500.c | 2 - sound/soc/ux500/mop500_ab8500.h | 2 - sound/soc/ux500/ux500_msp_dai.c | 41 +- sound/soc/ux500/ux500_msp_dai.h | 2 - sound/soc/ux500/ux500_msp_i2s.c | 2 - sound/soc/ux500/ux500_msp_i2s.h | 2 - sound/soc/ux500/ux500_pcm.c | 2 - sound/soc/ux500/ux500_pcm.h | 2 - sound/soc/xilinx/xlnx_formatter_pcm.c | 18 +- sound/soc/xilinx/xlnx_i2s.c | 1 + sound/soc/xilinx/xlnx_spdif.c | 1 + sound/soc/xtensa/xtfpga-i2s.c | 19 +- 736 files changed, 35467 insertions(+), 3857 deletions(-) create mode 100644 Documentation/devicetree/bindings/dsp/mediatek,mt8186-dsp.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-i2s.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sama5d2-pdmic.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-pdmic.txt delete mode 100644 Documentation/devicetree/bindings/sound/designware-i2s.txt delete mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,micfil.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-mbdrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ope.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-peq.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,sdm845.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml create mode 100644 Documentation/devicetree/bindings/sound/snps,designware-i2s.yaml create mode 100644 include/dt-bindings/sound/qcom,wcd9335.h create mode 100644 sound/soc/amd/acp-es8336.c create mode 100644 sound/soc/amd/acp/acp-rembrandt.c create mode 100644 sound/soc/amd/rpl/Makefile create mode 100644 sound/soc/amd/rpl/rpl-pci-acp6x.c create mode 100644 sound/soc/amd/rpl/rpl_acp6x.h create mode 100644 sound/soc/amd/rpl/rpl_acp6x_chip_offset_byte.h create mode 100644 sound/soc/codecs/hda-dai.c create mode 100644 sound/soc/codecs/hda.c create mode 100644 sound/soc/codecs/hda.h create mode 100644 sound/soc/codecs/tas2780.c create mode 100644 sound/soc/codecs/tas2780.h create mode 100644 sound/soc/codecs/wsa883x.c create mode 100644 sound/soc/intel/avs/boards/Kconfig create mode 100644 sound/soc/intel/avs/boards/Makefile create mode 100644 sound/soc/intel/avs/boards/da7219.c create mode 100644 sound/soc/intel/avs/boards/dmic.c create mode 100644 sound/soc/intel/avs/boards/hdaudio.c create mode 100644 sound/soc/intel/avs/boards/i2s_test.c create mode 100644 sound/soc/intel/avs/boards/max98357a.c create mode 100644 sound/soc/intel/avs/boards/max98373.c create mode 100644 sound/soc/intel/avs/boards/nau8825.c create mode 100644 sound/soc/intel/avs/boards/rt274.c create mode 100644 sound/soc/intel/avs/boards/rt286.c create mode 100644 sound/soc/intel/avs/boards/rt298.c create mode 100644 sound/soc/intel/avs/boards/rt5682.c create mode 100644 sound/soc/intel/avs/boards/ssm4567.c create mode 100644 sound/soc/intel/boards/bdw_rt286.c delete mode 100644 sound/soc/intel/boards/broadwell.c delete mode 100644 sound/soc/intel/boards/haswell.c create mode 100644 sound/soc/intel/boards/hsw_rt5640.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-mtl-match.c create mode 100644 sound/soc/mediatek/common/mtk-dsp-sof-common.c create mode 100644 sound/soc/mediatek/common/mtk-dsp-sof-common.h create mode 100644 sound/soc/mediatek/common/mtk-soc-card.h create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h create mode 100644 sound/soc/sof/intel/mtl.c create mode 100644 sound/soc/sof/intel/mtl.h create mode 100644 sound/soc/sof/intel/pci-mtl.c create mode 100644 sound/soc/sof/ipc4-control.c create mode 100644 sound/soc/sof/ipc4-pcm.c create mode 100644 sound/soc/sof/ipc4-topology.c create mode 100644 sound/soc/sof/ipc4-topology.h delete mode 100644 sound/soc/tegra/tegra20_das.h create mode 100644 sound/soc/tegra/tegra210_mbdrc.c create mode 100644 sound/soc/tegra/tegra210_mbdrc.h create mode 100644 sound/soc/tegra/tegra210_ope.c create mode 100644 sound/soc/tegra/tegra210_ope.h create mode 100644 sound/soc/tegra/tegra210_peq.c create mode 100644 sound/soc/tegra/tegra210_peq.h

2 2

[PATCH v2 0/4] ALSA: Defer async signal handling
by Takashi Iwai 01 Aug '22

01 Aug '22

Hi, this is a revised patch series for another attempt to address the potential deadlocks via kill_fasync() calls that have been reported by syzbot for long time. Only a missing linux/fs.h inclusion was added from v1 series[1]. Instead of the previous series to drop the async handler[2], this tries to defer the kill_fasync() call. A few new common helpers are introduced at first, then the actual usages are replaced in the following patches. The patches passed the quick tests with alsa-lib test cases. Takashi [1] https://lore.kernel.org/r/20220726153420.3403-1-tiwai@suse.de [2] https://lore.kernel.org/r/20220717070549.5993-1-tiwai@suse.de === Takashi Iwai (4): ALSA: core: Add async signal helpers ALSA: timer: Use deferred fasync helper ALSA: pcm: Use deferred fasync helper ALSA: control: Use deferred fasync helper include/sound/control.h | 2 +- include/sound/core.h | 8 ++++ include/sound/pcm.h | 2 +- sound/core/control.c | 7 +-- sound/core/misc.c | 94 +++++++++++++++++++++++++++++++++++++++++ sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/timer.c | 11 ++--- 9 files changed, 117 insertions(+), 12 deletions(-) -- 2.35.3

2 9

[PATCH v1 0/3] ADD legacy audio driver support for rembrandt
by V sujith kumar Reddy 01 Aug '22

01 Aug '22

Add Generic driver to support multiple platform, ADD HS control instance for Rembrandt platform. Add nau8825,max98560 and rt5682s,rt1019 combination support for legacy platform. v0--->v1 -- Kconfig syntax changes done -- robot errors done sound/soc/amd/acp/acp-rembrandt.c:264:5: warning: no previous prototype for 'rmb_acp_init' [-Wmissing-prototypes] sound/soc/amd/acp/acp-rembrandt.c:264:5: warning: no previous prototype for 'rmb_acp_init' [-Wmissing-prototypes] V sujith kumar Reddy (3): ASoC: amd: acp: Remove rt1019_1 codec conf from machine driver ASoC: amd: acp: ACP code generic to support newer platforms ASoC: amd: acp: Add legacy audio driver support for Rembrandt platform sound/soc/amd/acp/Kconfig | 11 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 169 +++++++++++- sound/soc/amd/acp/acp-legacy-mach.c | 32 +++ sound/soc/amd/acp/acp-mach-common.c | 104 +++++-- sound/soc/amd/acp/acp-mach.h | 6 + sound/soc/amd/acp/acp-pci.c | 6 + sound/soc/amd/acp/acp-pdm.c | 8 +- sound/soc/amd/acp/acp-platform.c | 32 ++- sound/soc/amd/acp/acp-rembrandt.c | 397 +++++++++++++++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 38 ++- sound/soc/amd/acp/amd.h | 86 +++++- sound/soc/amd/acp/chip_offset_byte.h | 40 ++- 13 files changed, 862 insertions(+), 69 deletions(-) create mode 100644 sound/soc/amd/acp/acp-rembrandt.c -- 2.25.1

4 6

[PATCH v3] ASoC: amd: acp: Initialize list to store acp_stream during pcm_open
by Venkata Prasad Potturu 01 Aug '22

01 Aug '22

From: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> We are currently allocating acp_stream during pcm_open and saving it in static array corresponds to array index calculated based on cpu dai->driver id. This approach will fail if we have single dai linked to multiple pcm device as we will have same dai->driver id or array index for multiple pcm open. Initialize new linked list stream_list to store opened pcm stream info dynamically. Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey(a)amd.com> Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Reviewed-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes since v2: -- Add locking mechanism in irq handler, linked list updatation and deletion. Changes since v1: -- Fix compile error and remove unused variable. sound/soc/amd/acp/acp-platform.c | 43 ++++++++++++++++---------------- sound/soc/amd/acp/amd.h | 4 ++- 2 files changed, 25 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp/acp-platform.c b/sound/soc/amd/acp/acp-platform.c index f561d39b33e2..4d35c75611d3 100644 --- a/sound/soc/amd/acp/acp-platform.c +++ b/sound/soc/amd/acp/acp-platform.c @@ -93,8 +93,9 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) struct acp_dev_data *adata = data; struct acp_resource *rsrc = adata->rsrc; struct acp_stream *stream; + unsigned long flags; u16 i2s_flag = 0; - u32 ext_intr_stat, ext_intr_stat1, i; + u32 ext_intr_stat, ext_intr_stat1; if (!adata) return IRQ_NONE; @@ -104,14 +105,13 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) ext_intr_stat = readl(ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); - for (i = 0; i < ACP_MAX_STREAM; i++) { - stream = adata->stream[i]; + spin_lock_irqsave(&adata->acp_lock, flags); + list_for_each_entry(stream, &adata->stream_list, list) { if (stream && (ext_intr_stat & stream->irq_bit)) { writel(stream->irq_bit, ACP_EXTERNAL_INTR_STAT(adata, rsrc->irqp_used)); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } if (adata->rsrc->no_of_ctrls == 2) { if (stream && (ext_intr_stat1 & stream->irq_bit)) { @@ -119,10 +119,10 @@ static irqreturn_t i2s_irq_handler(int irq, void *data) (rsrc->irqp_used - 1))); snd_pcm_period_elapsed(stream->substream); i2s_flag = 1; - break; } } } + spin_unlock_irqrestore(&adata->acp_lock, flags); if (i2s_flag) return IRQ_HANDLED; @@ -146,9 +146,8 @@ static void config_pte_for_stream(struct acp_dev_data *adata, struct acp_stream writel(0x01, adata->acp_base + ACPAXI2AXI_ATU_CTRL); } -static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) +static void config_acp_dma(struct acp_dev_data *adata, struct acp_stream *stream, int size) { - struct acp_stream *stream = adata->stream[cpu_id]; struct snd_pcm_substream *substream = stream->substream; struct acp_resource *rsrc = adata->rsrc; dma_addr_t addr = substream->dma_buffer.addr; @@ -174,21 +173,22 @@ static void config_acp_dma(struct acp_dev_data *adata, int cpu_id, int size) static int acp_dma_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct snd_pcm_runtime *runtime = substream->runtime; struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; int ret; + unsigned long flags; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (!stream) return -ENOMEM; stream->substream = substream; - adata->stream[stream_id] = stream; + + spin_lock_irqsave(&adata->acp_lock, flags); + list_add_tail(&stream->list, &adata->stream_list); + spin_unlock_irqrestore(&adata->acp_lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) runtime->hw = acp_pcm_hardware_playback; @@ -212,16 +212,13 @@ static int acp_dma_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); struct acp_dev_data *adata = snd_soc_component_get_drvdata(component); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct acp_stream *stream = substream->runtime->private_data; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; u64 size = params_buffer_bytes(params); /* Configure ACP DMA block with params */ config_pte_for_stream(adata, stream); - config_acp_dma(adata, stream_id, size); + config_acp_dma(adata, stream, size); return 0; } @@ -261,16 +258,16 @@ static int acp_dma_new(struct snd_soc_component *component, static int acp_dma_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *soc_runtime = asoc_substream_to_rtd(substream); - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_runtime, 0); struct device *dev = component->dev; struct acp_dev_data *adata = dev_get_drvdata(dev); - struct acp_stream *stream; - int stream_id = cpu_dai->driver->id * 2 + substream->stream; + struct acp_stream *stream = substream->runtime->private_data; + unsigned long flags; - stream = adata->stream[stream_id]; + /* Remove entry from list */ + spin_lock_irqsave(&adata->acp_lock, flags); + list_del(&stream->list); + spin_unlock_irqrestore(&adata->acp_lock, flags); kfree(stream); - adata->stream[stream_id] = NULL; return 0; } @@ -305,6 +302,10 @@ int acp_platform_register(struct device *dev) dev_err(dev, "Fail to register acp i2s component\n"); return status; } + + INIT_LIST_HEAD(&adata->stream_list); + spin_lock_init(&adata->acp_lock); + return 0; } EXPORT_SYMBOL_NS_GPL(acp_platform_register, SND_SOC_ACP_COMMON); diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 60a7c95f134f..be8bb8247c4e 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -91,6 +91,7 @@ struct acp_chip_info { }; struct acp_stream { + struct list_head list; struct snd_pcm_substream *substream; int irq_bit; int dai_id; @@ -123,7 +124,8 @@ struct acp_dev_data { struct snd_soc_dai_driver *dai_driver; int num_dai; - struct acp_stream *stream[ACP_MAX_STREAM]; + struct list_head stream_list; + spinlock_t acp_lock; struct snd_soc_acpi_mach *machines; struct platform_device *mach_dev; -- 2.25.1

4 5

[PATCH v7 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by Ban Tao 31 Jul '22

31 Jul '22

The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. v2->v3: None. v3->v4: 1.add sig_bits. v4->v5: None. v5->v6: 1.Modify RXFIFO_CTL_MODE to mode 1. v6->v7: 1.Modify dmic_rate_s to be a global variable. 2.Changed some macro names to make more sense. 3.Align code format. 4.Add a depends on PM to Kconfig entry. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e9d5b05..839f625 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -796,6 +796,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa9..8543234 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && PM && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be34..4483fe9 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 0000000..76b3378 --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMPLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMPLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MASK BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_LSB (0 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_MODE_MSB (1 << 9) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK BIT(8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16 (0 << 8) + #define SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24 (1 << 8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static struct dmic_rate dmic_rate_s[] = { + {48000, 0x0}, + {44100, 0x0}, + {32000, 0x1}, + {24000, 0x2}, + {22050, 0x2}, + {16000, 0x3}, + {12000, 0x4}, + {11025, 0x4}, + {8000, 0x5}, +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + unsigned int chan_en = (1 << channels) - 1; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, chan_en); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(chan_en)); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_16); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_MASK, + SUN50I_DMIC_RXFIFO_CTL_SAMPLE_24); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE_MASK, + SUN50I_DMIC_RXFIFO_CTL_MODE_MSB); + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMPLE_RATE_MASK, + SUN50I_DMIC_SR_SAMPLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_DMIC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_DMIC_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + + ret = clk_prepare_enable(host->bus_clk); + if (ret) { + clk_disable_unprepare(host->dmic_clk); + return ret; + } + + return 0; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, &sun50i_dmic_component, + &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_disable_runtime_pm; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_disable_runtime_pm: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.7.4

1 0

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NV45PZ
by Tim Crawford 31 Jul '22

31 Jul '22

Fixes headset detection on Clevo NV45PZ. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cca093cb643e..105468acde90 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9347,6 +9347,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x4041, "Clevo NV4[15]PZ", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40a1, "Clevo NL40GU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40c1, "Clevo NL40[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x40d1, "Clevo NL41DU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.37.1

2 1

[PATCH v2] ALSA: hda/realtek: Add quirk for Lenovo Yoga9 14IAP7
by Philipp Jungkamp 30 Jul '22

30 Jul '22

The Lenovo Yoga 9 14IAP7 is set up similarly to the Thinkpad X1 7th and 8th Gen. It also has the speakers attached to NID 0x14 and the bass speakers to NID 0x17, but here the codec misreports the NID 0x17 as unconnected. The pincfg and hda verbs connect and activate the bass speaker amplifiers, but the generic driver will connect them to NID 0x06 which has no volume control. Set connection list/preferred connections is required to gain volume control. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Philipp Jungkamp <p.jungkamp(a)gmx.net> --- The HDA verbs now use AC_VERB_SET_COEF_INDEX and AC_VERB_SET_PROC_COEF instead of 0x500 and 0x400. sound/pci/hda/patch_realtek.c | 109 ++++++++++++++++++++++++++++++++++ 1 file changed, 109 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f55bc43bfa9..6c4e9994d99c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6787,6 +6787,43 @@ static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, } } +static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* + * The Pin Complex 0x17 for the bass speakers is wrongly reported as + * unconnected. + */ + static const struct hda_pintbl pincfgs[] = { + { 0x17, 0x90170121 }, + { } + }; + /* + * Avoid DAC 0x06 and 0x08, as they have no volume controls. + * DAC 0x02 and 0x03 would be fine. + */ + static const hda_nid_t conn[] = { 0x02, 0x03 }; + /* + * Prefer both speakerbar (0x14) and bass speakers (0x17) connected to DAC 0x02. + * Headphones (0x21) are connected to DAC 0x03. + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x02, + 0x21, 0x03, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7023,6 +7060,8 @@ enum { ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, + ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, + ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -8865,6 +8904,74 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // enable left speaker + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x41 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x1a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xf }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x42 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x40 }, + { 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 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + // enable right speaker + { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xc }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x2a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xf }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x46 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x10 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x44 }, + { 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 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 }, + + { }, + }, + }, + [ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_yoga9_14iap7_bass_spk_pin, + .chained = true, + .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9315,6 +9422,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), + SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), @@ -9560,6 +9668,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.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 = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, .name = "alc287-yoga9-bass-spk-pin"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {.id = ALC285_FIXUP_HP_GPIO_AMP_INIT, .name = "alc285-hp-amp-init"}, -- 2.37.1

2 1

[PATCH v4 1/2] ASoC: codecs: add support for ES8326
by Zhu Ning 30 Jul '22

30 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ----- v4 changes since v3 1. remove sysconstrains 2. remove check for null mclk 3. add button detection 3. add properity for reg 58 & 59 ----- v3 changes since v2 1. remove .non_legacy_dai_naming flag --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 908 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 ++++++++ 4 files changed, 1102 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..ac6ed27adad7 --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,908 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling or during an irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + u8 interrupt_src; + u8 interrupt_clk; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_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 es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } else { + dev_warn(component->dev, "Clock coefficients do not match"); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +/* + * For button detection, set the following in soundcard + * snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + * snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + */ +static void es8326_jack_button_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, button_press_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + static int button_to_report, press_count; + static int prev_button, cur_button; + + if (!(es8326->jack->status & SND_JACK_HEADSET)) /* Jack unplugged */ + return; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DECTECT_FB); + switch (iface) { + case 0x93: + /* pause button detected */ + cur_button = SND_JACK_BTN_0; + break; + case 0x6f: + /* button volume up */ + cur_button = SND_JACK_BTN_1; + break; + case 0x27: + /* button volume down */ + cur_button = SND_JACK_BTN_2; + break; + case 0x1e: + /* button released or not pressed */ + cur_button = 0; + break; + default: + break; + } + + if ((prev_button == cur_button) && (cur_button != 0)) { + press_count++; + if (press_count > 10) { + /* report a press every 500ms */ + snd_soc_jack_report(es8326->jack, cur_button, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + press_count = 0; + } + button_to_report = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else if (prev_button != cur_button) { + /* mismatch, detect again */ + prev_button = cur_button; + queue_delayed_work(system_wq, &es8326->button_press_work, + msecs_to_jiffies(50)); + } else { + /* released or no pressed */ + if (button_to_report != 0) { + snd_soc_jack_report(es8326->jack, button_to_report, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + snd_soc_jack_report(es8326->jack, 0, + SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); + button_to_report = 0; + } + } + mutex_unlock(&es8326->lock); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + mutex_lock(&es8326->lock); + iface = snd_soc_component_read(comp, ES8326_HP_DECTECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + /* Jack unplugged or spurious IRQ */ + dev_dbg(comp->dev, "No headset detected"); + if (es8326->jack->status & SND_JACK_HEADPHONE) { + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if (es8326->jack->status & SND_JACK_HEADSET) { + /* detect button */ + queue_delayed_work(system_wq, &es8326->button_press_work, 10); + } else { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, + ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, + SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } + } + mutex_unlock(&es8326->lock); +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + if (es8326->jack->status & SND_JACK_HEADSET) + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(10)); + else + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, es8326->interrupt_src); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, es8326->interrupt_clk); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + if ((reg & ES8326_VERSION_B) == 1) { + regmap_write(es8326->regmap, ES8326_ANA_MICBIAS_1B, 0xDD); + regmap_write(es8326->regmap, ES8326_ANA_VSEL_1C, 0x7F); + regmap_write(es8326->regmap, ES8326_VMIDLOW_22, 0x0F); + /* enable button detect */ + regmap_write(es8326->regmap, ES8326_HP_DRIVER_24, 0xA0); + } + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&es8326->jack_detect_work); + es8326_disable_micbias(component); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + ret = device_property_read_u8(component->dev, "everest,interrupt-src", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-src return %d", ret); + es8326->interrupt_src = ES8326_INT_SRC_PIN9; + } + dev_dbg(component->dev, "interrupt-src %x", es8326->interrupt_src); + + ret = device_property_read_u8(component->dev, "everest,interrupt-clk", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "interrupt-clk return %d", ret); + es8326->interrupt_clk = 0x45; + } + dev_dbg(component->dev, "interrupt-clk %x", es8326->interrupt_clk); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to request IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, + .id_table = es8326_i2c_id, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..717f3fd42372 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

4 5

[PATCH] ALSA: hda/realtek: Add quirk for Lenovo Yoga9 14IAP7
by Philipp Jungkamp 29 Jul '22

29 Jul '22

The Lenovo Yoga 9 14IAP7 is set up similarly to the Thinkpad X1 7th and 8th Gen. It also has the speakers attached to NID 0x14 and the bass speakers to NID 0x17, but here the codec misreports the NID 0x17 as unconnected. The pincfg and hda verbs connect and activate the bass speaker amplifiers, but the generic driver will connect them to NID 0x06 which has no volume control. Set connection list/preferred connections is required to gain volume control. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208555 Signed-off-by: Philipp Jungkamp <p.jungkamp(a)gmx.net> --- sound/pci/hda/patch_realtek.c | 109 ++++++++++++++++++++++++++++++++++ 1 file changed, 109 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 2f55bc43bfa9..fe1a4212dbc3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6787,6 +6787,43 @@ static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, } } +static void alc287_fixup_yoga9_14iap7_bass_spk_pin(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + /* + * The Pin Complex 0x17 for the bass speakers is wrongly reported as + * unconnected. + */ + static const struct hda_pintbl pincfgs[] = { + { 0x17, 0x90170121 }, + { } + }; + /* + * Avoid DAC 0x06 and 0x08, as they have no volume controls. + * DAC 0x02 and 0x03 would be fine. + */ + static const hda_nid_t conn[] = { 0x02, 0x03 }; + /* + * Prefer both speakerbar (0x14) and bass speakers (0x17) connected to DAC 0x02. + * Headphones (0x21) are connected to DAC 0x03. + */ + static const hda_nid_t preferred_pairs[] = { + 0x14, 0x02, + 0x17, 0x02, + 0x21, 0x03, + 0 + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, pincfgs); + snd_hda_override_conn_list(codec, 0x17, ARRAY_SIZE(conn), conn); + spec->gen.preferred_dacs = preferred_pairs; + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -7023,6 +7060,8 @@ enum { ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, + ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, + ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -8865,6 +8904,74 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + // enable left speaker + { 0x20, 0x500, 0x24 }, + { 0x20, 0x400, 0x41 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0xc }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x1a }, + { 0x20, 0x4b0, 0x20 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0xf }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x42 }, + { 0x20, 0x4b0, 0x20 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0x10 }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x40 }, + { 0x20, 0x4b0, 0x20 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0x2 }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x4b0, 0x20 }, + + // enable righ speaker + { 0x20, 0x500, 0x24 }, + { 0x20, 0x400, 0x46 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0xc }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x2a }, + { 0x20, 0x4b0, 0x20 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0xf }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x46 }, + { 0x20, 0x4b0, 0x20 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0x10 }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x44 }, + { 0x20, 0x4b0, 0x20 }, + + { 0x20, 0x500, 0x26 }, + { 0x20, 0x400, 0x2 }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x400, 0x0 }, + { 0x20, 0x4b0, 0x20 }, + + { }, + }, + }, + [ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc287_fixup_yoga9_14iap7_bass_spk_pin, + .chained = true, + .chain_id = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9315,6 +9422,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3176, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x3178, "ThinkCentre Station", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x31af, "ThinkCentre Station", ALC623_FIXUP_LENOVO_THINKSTATION_P340), + SND_PCI_QUIRK(0x17aa, 0x3801, "Lenovo Yoga9 14IAP7", ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo Yoga DuetITL 2021", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940 / Yoga Duet 7", ALC298_FIXUP_LENOVO_C940_DUET7), @@ -9560,6 +9668,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.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 = ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, .name = "alc287-yoga9-bass-spk-pin"}, {.id = ALC623_FIXUP_LENOVO_THINKSTATION_P340, .name = "alc623-lenovo-thinkstation-p340"}, {.id = ALC255_FIXUP_ACER_HEADPHONE_AND_MIC, .name = "alc255-acer-headphone-and-mic"}, {.id = ALC285_FIXUP_HP_GPIO_AMP_INIT, .name = "alc285-hp-amp-init"}, -- 2.37.1

2 1

[PATCH v2] ASoC: Intel: cirrus-common: Use UID to map correct amp to prefix
by Stefan Binding 29 Jul '22

29 Jul '22

Since the order of the amps in the ACPI determines the device name, and the ACPI order may change depending on hardware configuration, use UID to dynamically compute the dai links, allowing dynamic assignment of the name_prefix. The UIDs for these amps in ACPI are fixed, and map to a name_prefix, where: UID 0x0 -> WL UID 0x1 -> WR UID 0x2 -> TL UID 0x3 -> TR Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/intel/boards/sof_cirrus_common.c | 89 +++++++++++----------- 1 file changed, 46 insertions(+), 43 deletions(-) diff --git a/sound/soc/intel/boards/sof_cirrus_common.c b/sound/soc/intel/boards/sof_cirrus_common.c index f4192df962d6..e74c4adbf2a5 100644 --- a/sound/soc/intel/boards/sof_cirrus_common.c +++ b/sound/soc/intel/boards/sof_cirrus_common.c @@ -10,6 +10,9 @@ #include "../../codecs/cs35l41.h" #include "sof_cirrus_common.h" +#define CS35L41_HID "CSC3541" +#define CS35L41_MAX_AMPS 4 + /* * Cirrus Logic CS35L41/CS35L53 */ @@ -35,50 +38,12 @@ static const struct snd_soc_dapm_route cs35l41_dapm_routes[] = { {"TR Spk", NULL, "TR SPK"}, }; -static struct snd_soc_dai_link_component cs35l41_components[] = { - { - .name = CS35L41_DEV0_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV1_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV2_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, - { - .name = CS35L41_DEV3_NAME, - .dai_name = CS35L41_CODEC_DAI, - }, -}; +static struct snd_soc_dai_link_component cs35l41_components[CS35L41_MAX_AMPS]; /* * Mapping between ACPI instance id and speaker position. - * - * Four speakers: - * 0: Tweeter left, 1: Woofer left - * 2: Tweeter right, 3: Woofer right */ -static struct snd_soc_codec_conf cs35l41_codec_conf[] = { - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV0_NAME), - .name_prefix = "TL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV1_NAME), - .name_prefix = "WL", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV2_NAME), - .name_prefix = "TR", - }, - { - .dlc = COMP_CODEC_CONF(CS35L41_DEV3_NAME), - .name_prefix = "WR", - }, -}; +static struct snd_soc_codec_conf cs35l41_codec_conf[CS35L41_MAX_AMPS]; static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) { @@ -117,10 +82,10 @@ static int cs35l41_init(struct snd_soc_pcm_runtime *rtd) static const struct { unsigned int rx[2]; } cs35l41_channel_map[] = { - {.rx = {0, 1}}, /* TL */ {.rx = {0, 1}}, /* WL */ - {.rx = {1, 0}}, /* TR */ {.rx = {1, 0}}, /* WR */ + {.rx = {0, 1}}, /* TL */ + {.rx = {1, 0}}, /* TR */ }; static int cs35l41_hw_params(struct snd_pcm_substream *substream, @@ -175,10 +140,48 @@ static const struct snd_soc_ops cs35l41_ops = { .hw_params = cs35l41_hw_params, }; +static const char * const cs35l41_name_prefixes[] = { "WL", "WR", "TL", "TR" }; + +/* + * Expected UIDs are integers (stored as strings). + * UID Mapping is fixed: + * UID 0x0 -> WL + * UID 0x1 -> WR + * UID 0x2 -> TL + * UID 0x3 -> TR + * Note: If there are less than 4 Amps, UIDs still map to WL/WR/TL/TR. Dynamic code will only create + * dai links for UIDs which exist, and ignore non-existant ones. + * Return number of codecs found. + */ +static int cs35l41_compute_codec_conf(void) +{ + const char * const uid_strings[] = { "0", "1", "2", "3" }; + unsigned int uid, sz = 0; + struct acpi_device *adev; + struct device *physdev; + + for (uid = 0; uid < CS35L41_MAX_AMPS; uid++) { + adev = acpi_dev_get_first_match_dev(CS35L41_HID, uid_strings[uid], -1); + if (!adev) { + pr_warn("Cannot find match for HID %s UID %u (%s)\n", CS35L41_HID, uid, + cs35l41_name_prefixes[uid]); + continue; + } + physdev = get_device(acpi_get_first_physical_node(adev)); + cs35l41_components[sz].name = dev_name(physdev); + cs35l41_components[sz].dai_name = CS35L41_CODEC_DAI; + cs35l41_codec_conf[sz].dlc.name = dev_name(physdev); + cs35l41_codec_conf[sz].name_prefix = cs35l41_name_prefixes[uid]; + acpi_dev_put(adev); + sz++; + } + return sz; +} + void cs35l41_set_dai_link(struct snd_soc_dai_link *link) { + link->num_codecs = cs35l41_compute_codec_conf(); link->codecs = cs35l41_components; - link->num_codecs = ARRAY_SIZE(cs35l41_components); link->init = cs35l41_init; link->ops = &cs35l41_ops; } -- 2.34.1

2 2

[PATCH v2] ASoC: qcom: SC7280: Add support for external DMIC bias supply
by Srinivasa Rao Mandadapu 29 Jul '22

29 Jul '22

Update SC7280 machine driver for enabling external dmic bias supply, which is required for villager evt boards. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- Changes since v1: -- Replace dapm widget SND_SOC_DAPM_MIC with SND_SOC_DAPM_REGULATOR_SUPPLY. sound/soc/qcom/sc7280.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index da7469a..4a60f34 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -356,6 +356,7 @@ static const struct snd_soc_ops sc7280_ops = { static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_REGULATOR_SUPPLY("DMICVDD", 0, 0), }; static int sc7280_snd_platform_probe(struct platform_device *pdev) -- 2.7.4

3 4

[PATCH] ASoC: amd: acp: Modify dai_id macros to be more generic
by Venkata Prasad Potturu 29 Jul '22

29 Jul '22

Update dai_id macros to make code generic for all platforms. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/acp/amd.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 148a9ab6206d..ff2fb476fa6c 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -21,9 +21,9 @@ #define ACP3X_DEV 3 #define ACP6X_DEV 6 -#define I2S_SP_INSTANCE 0x00 -#define I2S_BT_INSTANCE 0x01 -#define DMIC_INSTANCE 0x02 +#define DMIC_INSTANCE 0x00 +#define I2S_SP_INSTANCE 0x01 +#define I2S_BT_INSTANCE 0x02 #define I2S_HS_INSTANCE 0x03 #define MEM_WINDOW_START 0x4080000 -- 2.25.1

3 4

[PATCH 0/3] support SOF for mt8186
by Chunxu Li 29 Jul '22

29 Jul '22

In these patches, we add SOF support for mt8186 which have HiFi5 DSP inside. Chunxu Li (3): ASoC: mediatek: mt8186: support DSP downlink ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-rt1019-rt5682s ASoC: mediatek: mt8186: add SOF support on mt8186-mt6366-da7219-max98357 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3 + sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 12 ++ .../mt8186/mt8186-mt6366-da7219-max98357.c | 168 ++++++++++++++++-- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 168 ++++++++++++++++-- 4 files changed, 325 insertions(+), 26 deletions(-) -- 2.18.0

1 3

KASAN: vmalloc-out-of-bounds Write in snd_pcm_hw_params
by Dipanjan Das 29 Jul '22

29 Jul '22

Hi, We would like to report the following bug which has been found by our modified version of syzkaller. ====================================================== description: KASAN: vmalloc-out-of-bounds Write in snd_pcm_hw_params affected file: sound/core/pcm_native.c kernel version: 5.10.131 kernel commit: de62055f423f5dcb548f74cebd68f03c8903f73a git tree: upstream kernel config: https://syzkaller.appspot.com/x/.config?x=e49433cfed49b7d9 crash reproducer: attached ====================================================== Crash log: ====================================================== BUG: KASAN: vmalloc-out-of-bounds in memset include/linux/string.h:384 [inline] BUG: KASAN: vmalloc-out-of-bounds in snd_pcm_hw_params+0x19b0/0x1db0 sound/core/pcm_native.c:799 Write of size 2097152 at addr ffffc900113b2000 by task syz-executor.5/14437 CPU: 1 PID: 14437 Comm: syz-executor.5 Tainted: G OE 5.10.131+ #3 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.13.0-1ubuntu1.1 04/01/2014 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x107/0x163 lib/dump_stack.c:118 print_address_description.constprop.0.cold+0x5/0x4f7 mm/kasan/report.c:385 __kasan_report mm/kasan/report.c:545 [inline] kasan_report.cold+0x1f/0x37 mm/kasan/report.c:562 check_memory_region_inline mm/kasan/generic.c:194 [inline] check_memory_region+0x187/0x1e0 mm/kasan/generic.c:200 memset+0x20/0x40 mm/kasan/common.c:85 memset include/linux/string.h:384 [inline] snd_pcm_hw_params+0x19b0/0x1db0 sound/core/pcm_native.c:799 snd_pcm_kernel_ioctl+0xd1/0x240 sound/core/pcm_native.c:3401 snd_pcm_oss_change_params_locked+0x17b6/0x3aa0 sound/core/oss/pcm_oss.c:965 snd_pcm_oss_change_params+0x76/0xd0 sound/core/oss/pcm_oss.c:1107 snd_pcm_oss_make_ready+0xb7/0x170 sound/core/oss/pcm_oss.c:1166 snd_pcm_oss_set_trigger.isra.0+0x34f/0x770 sound/core/oss/pcm_oss.c:2074 snd_pcm_oss_poll+0x679/0xb40 sound/core/oss/pcm_oss.c:2858 vfs_poll include/linux/poll.h:90 [inline] do_pollfd fs/select.c:872 [inline] do_poll fs/select.c:920 [inline] do_sys_poll+0x63c/0xe40 fs/select.c:1014 __do_sys_poll fs/select.c:1079 [inline] __se_sys_poll fs/select.c:1067 [inline] __x64_sys_poll+0x18c/0x490 fs/select.c:1067 do_syscall_64+0x2d/0x70 arch/x86/entry/common.c:46 entry_SYSCALL_64_after_hwframe+0x44/0xa9 RIP: 0033:0x7f095de4f4ed Code: 02 b8 ff ff ff ff c3 66 0f 1f 44 00 00 f3 0f 1e fa 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 RSP: 002b:00007f095bdffbe8 EFLAGS: 00000246 ORIG_RAX: 0000000000000007 RAX: ffffffffffffffda RBX: 00007f095df6df60 RCX: 00007f095de4f4ed RDX: 0000000000000009 RSI: 0000000000000001 RDI: 00000000200000c0 RBP: 00007f095bdffc40 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 000000000000001d R13: 00007ffff286ceff R14: 00007f095df6df60 R15: 00007f095bdffd80 Memory state around the buggy address: ffffc900115b1d00: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ffffc900115b1d80: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 >ffffc900115b1e00: f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 ^ ffffc900115b1e80: f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 ffffc900115b1f00: f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 f8 ================================================================== -- Thanks and Regards, Dipanjan

4 8

[PATCH v3 1/2] ASoC: codecs: add support for ES8326
by Zhu Ning 29 Jul '22

29 Jul '22

The ES8326 codec is not compatible with ES8316 and requires a dedicated driver. Signed-off-by: David Yang <yangxiaohua(a)everest-semi.com> Signed-off-by: Zhu Ning <zhuning(a)everest-semi.com> ----- v3 changes since v2 1. remove .non_legacy_dai_naming flag ---- v2 changes since v1: 1. add i2c_id_table 2. add .endianness .use_pmdown_time flag 3. fix dt-binding name --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8326.c | 808 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/es8326.h | 187 +++++++++ 4 files changed, 1002 insertions(+) mode change 100644 => 100755 sound/soc/codecs/Kconfig mode change 100644 => 100755 sound/soc/codecs/Makefile create mode 100755 sound/soc/codecs/es8326.c create mode 100755 sound/soc/codecs/es8326.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig old mode 100644 new mode 100755 index 6165db92a629..b0a15cce416d --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -98,6 +98,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_DA9055 imply SND_SOC_DMIC imply SND_SOC_ES8316 + imply SND_SOC_ES8326 imply SND_SOC_ES8328_SPI imply SND_SOC_ES8328_I2C imply SND_SOC_ES7134 @@ -911,6 +912,10 @@ config SND_SOC_ES8316 tristate "Everest Semi ES8316 CODEC" depends on I2C +config SND_SOC_ES8326 + tristate "Everest Semi ES8326 CODEC" + depends on I2C + config SND_SOC_ES8328 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile old mode 100644 new mode 100755 index 28dc4edfd01f..ae9d789b5381 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -100,6 +100,7 @@ snd-soc-dmic-objs := dmic.o snd-soc-es7134-objs := es7134.o snd-soc-es7241-objs := es7241.o snd-soc-es8316-objs := es8316.o +snd-soc-es8326-objs := es8326.o snd-soc-es8328-objs := es8328.o snd-soc-es8328-i2c-objs := es8328-i2c.o snd-soc-es8328-spi-objs := es8328-spi.o @@ -452,6 +453,7 @@ obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ES7134) += snd-soc-es7134.o obj-$(CONFIG_SND_SOC_ES7241) += snd-soc-es7241.o obj-$(CONFIG_SND_SOC_ES8316) += snd-soc-es8316.o +obj-$(CONFIG_SND_SOC_ES8326) += snd-soc-es8326.o obj-$(CONFIG_SND_SOC_ES8328) += snd-soc-es8328.o obj-$(CONFIG_SND_SOC_ES8328_I2C)+= snd-soc-es8328-i2c.o obj-$(CONFIG_SND_SOC_ES8328_SPI)+= snd-soc-es8328-spi.o diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c new file mode 100755 index 000000000000..384d9cc9b2ed --- /dev/null +++ b/sound/soc/codecs/es8326.c @@ -0,0 +1,808 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// es8326.c -- es8326 ALSA SoC audio driver +// Copyright Everest Semiconductor Co., Ltd +// +// Authors: David Yang <yangxiaohua(a)everest-semi.com> +// + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include "es8326.h" + +struct es8326_priv { + struct clk *mclk; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + struct i2c_client *i2c; + struct regmap *regmap; + struct snd_soc_component *component; + struct delayed_work jack_detect_work; + struct snd_soc_jack *jack; + int irq; + /* The lock protects the situation that an irq is generated + * while enabling or disabling irq. + */ + struct mutex lock; + u8 mic1_src; + u8 mic2_src; + u8 jack_pol; + bool jd_inverted; + unsigned int sysclk; +}; + +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(dac_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_vol_tlv, -9550, 50, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_analog_pga_tlv, 0, 300, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(adc_pga_tlv, 0, 600, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(softramp_rate, 0, 100, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_target_tlv, -3200, 200, 0); +static const SNDRV_CTL_TLVD_DECLARE_DB_SCALE(drc_recovery_tlv, -125, 250, 0); + +static const char *const winsize[] = { + "0.25db/2 LRCK", + "0.25db/4 LRCK", + "0.25db/8 LRCK", + "0.25db/16 LRCK", + "0.25db/32 LRCK", + "0.25db/64 LRCK", + "0.25db/128 LRCK", + "0.25db/256 LRCK", + "0.25db/512 LRCK", + "0.25db/1024 LRCK", + "0.25db/2048 LRCK", + "0.25db/4096 LRCK", + "0.25db/8192 LRCK", + "0.25db/16384 LRCK", + "0.25db/32768 LRCK", + "0.25db/65536 LRCK", +}; + +static const char *const dacpol_txt[] = { + "Normal", "R Invert", "L Invert", "L + R Invert" }; + +static const struct soc_enum dacpol = + SOC_ENUM_SINGLE(ES8326_DAC_DSM_4D, 4, 4, dacpol_txt); +static const struct soc_enum alc_winsize = + SOC_ENUM_SINGLE(ES8326_ADC_RAMPRATE_2E, 4, 16, winsize); +static const struct soc_enum drc_winsize = + SOC_ENUM_SINGLE(ES8326_DRC_WINSIZE_54, 4, 16, winsize); + +static const struct snd_kcontrol_new es8326_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", ES8326_DAC_VOL_50, 0, 0xbf, 0, dac_vol_tlv), + SOC_ENUM("Playback Polarity", dacpol), + SOC_SINGLE_TLV("DAC Ramp Rate", ES8326_DAC_RAMPRATE_4E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE_TLV("DRC Recovery Level", ES8326_DRC_RECOVERY_53, 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("DRC Winsize", drc_winsize), + SOC_SINGLE_TLV("DRC Target Level", ES8326_DRC_WINSIZE_54, 0, 0x0f, 0, drc_target_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ES8326_ADC1_VOL_2C, ES8326_ADC2_VOL_2D, 0, 0xff, 0, + adc_vol_tlv), + SOC_DOUBLE_TLV("ADC PGA Volume", ES8326_ADC_SCALE_29, 4, 0, 5, 0, adc_pga_tlv), + SOC_SINGLE_TLV("ADC PGA Gain Volume", ES8326_PGAGAIN_23, 0, 10, 0, adc_analog_pga_tlv), + SOC_SINGLE_TLV("ADC Ramp Rate", ES8326_ADC_RAMPRATE_2E, 0, 0x0f, 0, softramp_rate), + SOC_SINGLE("ALC Capture Switch", ES8326_ALC_RECOVERY_32, 3, 1, 0), + SOC_SINGLE_TLV("ALC Capture Recovery Level", ES8326_ALC_LEVEL_33, + 0, 4, 0, drc_recovery_tlv), + SOC_ENUM("ALC Capture Winsize", alc_winsize), + SOC_SINGLE_TLV("ALC Capture Target Level", ES8326_ALC_LEVEL_33, + 0, 0x0f, 0, drc_target_tlv), + +}; + +static const struct snd_soc_dapm_widget es8326_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("MIC4"), + + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Digital Interface */ + SND_SOC_DAPM_AIF_OUT("I2S OUT", "I2S1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("I2S IN", "I2S1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* ADC Digital Mute */ + SND_SOC_DAPM_PGA("ADC L1", ES8326_ADC_MUTE_15, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R1", ES8326_ADC_MUTE_15, 1, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC L2", ES8326_ADC_MUTE_15, 2, 1, NULL, 0), + SND_SOC_DAPM_PGA("ADC R2", ES8326_ADC_MUTE_15, 3, 1, NULL, 0), + + /* Analog Power Supply*/ + SND_SOC_DAPM_DAC("Right DAC", NULL, ES8326_ANA_PDN_16, 0, 1), + SND_SOC_DAPM_DAC("Left DAC", NULL, ES8326_ANA_PDN_16, 1, 1), + SND_SOC_DAPM_SUPPLY("Analog Power", ES8326_ANA_PDN_16, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBias Power", ES8326_ANA_PDN_16, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Vref", ES8326_ANA_PDN_16, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Vref", ES8326_ANA_PDN_16, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref Power", ES8326_ANA_PDN_16, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ES8326_ANA_MICBIAS_1B, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", ES8326_ANA_MICBIAS_1B, 3, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LHPMIX", ES8326_DAC2HPMIX_25, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("RHPMIX", ES8326_DAC2HPMIX_25, 3, 0, NULL, 0), + + /* Headphone Charge Pump and Output */ + SND_SOC_DAPM_SUPPLY("HPOR Cal", ES8326_HP_CAL_27, 7, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOL Cal", ES8326_HP_CAL_27, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", ES8326_HP_DRIVER_24, + 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Driver Bias", ES8326_HP_DRIVER_24, + 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone LDO", ES8326_HP_DRIVER_24, + 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Reference", ES8326_HP_DRIVER_24, + 0, 1, NULL, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOR Supply", ES8326_HP_CAL_27, + ES8326_HPOR_SHIFT, 7, 7, 0), + SND_SOC_DAPM_REG(snd_soc_dapm_supply, "HPOL Supply", ES8326_HP_CAL_27, + 0, 7, 7, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route es8326_dapm_routes[] = { + {"ADC L1", NULL, "MIC1"}, + {"ADC R1", NULL, "MIC2"}, + {"ADC L2", NULL, "MIC3"}, + {"ADC R2", NULL, "MIC4"}, + + + {"ADC L", NULL, "ADC L1"}, + {"ADC R", NULL, "ADC R1"}, + {"ADC L", NULL, "ADC L2"}, + {"ADC R", NULL, "ADC R2"}, + + {"I2S OUT", NULL, "ADC L"}, + {"I2S OUT", NULL, "ADC R"}, + + {"I2S OUT", NULL, "Analog Power"}, + {"I2S OUT", NULL, "ADC Vref"}, + {"I2S OUT", NULL, "Vref Power"}, + {"I2S OUT", NULL, "IBias Power"}, + {"I2S IN", NULL, "Analog Power"}, + {"I2S IN", NULL, "DAC Vref"}, + {"I2S IN", NULL, "Vref Power"}, + {"I2S IN", NULL, "IBias Power"}, + + {"Right DAC", NULL, "I2S IN"}, + {"Left DAC", NULL, "I2S IN"}, + + {"LHPMIX", NULL, "Left DAC"}, + {"RHPMIX", NULL, "Right DAC"}, + + {"HPOR", NULL, "HPOR Cal"}, + {"HPOL", NULL, "HPOL Cal"}, + {"HPOR", NULL, "HPOR Supply"}, + {"HPOL", NULL, "HPOL Supply"}, + {"HPOL", NULL, "Headphone Charge Pump"}, + {"HPOR", NULL, "Headphone Charge Pump"}, + {"HPOL", NULL, "Headphone Driver Bias"}, + {"HPOR", NULL, "Headphone Driver Bias"}, + {"HPOL", NULL, "Headphone LDO"}, + {"HPOR", NULL, "Headphone LDO"}, + {"HPOL", NULL, "Headphone Reference"}, + {"HPOR", NULL, "Headphone Reference"}, + + {"HPOL", NULL, "LHPMIX"}, + {"HPOR", NULL, "RHPMIX"}, +}; + +static const struct regmap_range es8326_volatile_ranges[] = { + regmap_reg_range(ES8326_HP_DETECT_FB, ES8326_HP_DETECT_FB), +}; + +static const struct regmap_access_table es8326_volatile_table = { + .yes_ranges = es8326_volatile_ranges, + .n_yes_ranges = ARRAY_SIZE(es8326_volatile_ranges), +}; + +const struct regmap_config es8326_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0xff, + .volatile_table = &es8326_volatile_table, + .cache_type = REGCACHE_RBTREE, +}; + +struct _coeff_div { + u16 fs; + u32 rate; + u32 mclk; + u8 reg4; + u8 reg5; + u8 reg6; + u8 reg7; + u8 reg8; + u8 reg9; + u8 rega; + u8 regb; +}; + +/* codec hifi mclk clock divider coefficients */ +/* {ratio, LRCK, MCLK, REG04, REG05, REG06, REG07, REG08, REG09, REG10, REG11} */ +static const struct _coeff_div coeff_div[] = { + {32, 8000, 256000, 0x60, 0x00, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {32, 16000, 512000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {32, 44100, 1411200, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {32, 48000, 1536000, 0x00, 0x00, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {36, 8000, 288000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {36, 16000, 576000, 0x20, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {48, 8000, 384000, 0x60, 0x02, 0x1F, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {48, 16000, 768000, 0x20, 0x02, 0x0F, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {48, 48000, 2304000, 0x00, 0x02, 0x0D, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 8000, 512000, 0x60, 0x00, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {64, 16000, 1024000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + + {64, 44100, 2822400, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {64, 48000, 3072000, 0x00, 0x00, 0x11, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {72, 8000, 576000, 0x20, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {72, 16000, 1152000, 0x20, 0x00, 0x05, 0x75, 0x0A, 0x1B, 0x23, 0x47}, + {96, 8000, 768000, 0x60, 0x02, 0x1D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {96, 16000, 1536000, 0x20, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {100, 48000, 4800000, 0x04, 0x04, 0x3F, 0x6D, 0x38, 0x08, 0x4f, 0x1f}, + {125, 48000, 6000000, 0x04, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {128, 8000, 1024000, 0x60, 0x00, 0x13, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {128, 16000, 2048000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {128, 44100, 5644800, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {128, 48000, 6144000, 0x00, 0x00, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {144, 8000, 1152000, 0x20, 0x00, 0x03, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {144, 16000, 2304000, 0x20, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {192, 8000, 1536000, 0x60, 0x02, 0x0D, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {192, 16000, 3072000, 0x20, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x3F}, + {200, 48000, 9600000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {250, 48000, 12000000, 0x04, 0x04, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {256, 8000, 2048000, 0x60, 0x00, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {256, 16000, 4096000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {256, 44100, 11289600, 0x00, 0x00, 0x10, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {256, 48000, 12288000, 0x00, 0x00, 0x30, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {288, 8000, 2304000, 0x20, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x23, 0x47}, + {384, 8000, 3072000, 0x60, 0x02, 0x05, 0x75, 0x0A, 0x1B, 0x1F, 0x7F}, + {384, 16000, 6144000, 0x20, 0x02, 0x03, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {384, 48000, 18432000, 0x00, 0x02, 0x01, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {400, 48000, 19200000, 0x09, 0x04, 0x0f, 0x6d, 0x3a, 0x0A, 0x4F, 0x1F}, + {500, 48000, 24000000, 0x18, 0x04, 0x1F, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 8000, 4096000, 0x60, 0x00, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {512, 16000, 8192000, 0x20, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + + {512, 44100, 22579200, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {512, 48000, 24576000, 0x00, 0x00, 0x00, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {768, 8000, 6144000, 0x60, 0x02, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {768, 16000, 12288000, 0x20, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {800, 48000, 38400000, 0x00, 0x18, 0x13, 0x2D, 0x0A, 0x0A, 0x1F, 0x1F}, + {1024, 8000, 8192000, 0x60, 0x00, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {1024, 16000, 16384000, 0x20, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1152, 16000, 18432000, 0x20, 0x08, 0x11, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1536, 8000, 12288000, 0x60, 0x02, 0x01, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + + {1536, 16000, 24576000, 0x20, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x3F}, + {1625, 8000, 13000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {1625, 16000, 26000000, 0x0C, 0x18, 0x1F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + {2048, 8000, 16384000, 0x60, 0x00, 0x00, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {2304, 8000, 18432000, 0x40, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x5F}, + {3072, 8000, 24576000, 0x60, 0x02, 0x10, 0x35, 0x0A, 0x1B, 0x1F, 0x7F}, + {3250, 8000, 26000000, 0x0C, 0x18, 0x0F, 0x2D, 0x0A, 0x0A, 0x27, 0x27}, + +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +static unsigned int es8326_rates[] = { + 8000, 12000, 16000, 32000, 48000, 96000 +}; + +static struct snd_pcm_hw_constraint_list es8326_constraints = { + .count = ARRAY_SIZE(es8326_rates), + .list = es8326_rates, +}; + +static int es8326_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *codec = codec_dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + + es8326->sysclk = freq; + + if (freq == 0) { + es8326->sysclk_constraints->list = NULL; + es8326->sysclk_constraints->count = 0; + return 0; + } + + es8326->sysclk_constraints = &es8326_constraints; + return 0; +} + +static int es8326_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + u8 iface = 0; + + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFP: + snd_soc_component_update_bits(component, ES8326_RESET_00, + ES8326_MASTER_MODE_EN, ES8326_MASTER_MODE_EN); + break; + case SND_SOC_DAIFMT_CBC_CFC: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + dev_err(component->dev, "Codec driver does not support right justified\n"); + return -EINVAL; + case SND_SOC_DAIFMT_LEFT_J: + iface |= ES8326_DAIFMT_LEFT_J; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= ES8326_DAIFMT_DSP_A; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= ES8326_DAIFMT_DSP_B; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, ES8326_FMT_13, ES8326_DAIFMT_MASK, iface); + + return 0; +} + +static int es8326_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *codec = dai->component; + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + u8 srate = 0; + int coeff; + + coeff = get_coeff(es8326->sysclk, params_rate(params)); + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + srate |= ES8326_S16_LE; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + srate |= ES8326_S20_3_LE; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + srate |= ES8326_S18_LE; + break; + case SNDRV_PCM_FORMAT_S24_LE: + srate |= ES8326_S24_LE; + break; + case SNDRV_PCM_FORMAT_S32_LE: + srate |= ES8326_S32_LE; + break; + default: + return -EINVAL; + } + + /* set iface & srate */ + snd_soc_component_update_bits(codec, ES8326_FMT_13, ES8326_DATA_LEN_MASK, srate); + + if (coeff >= 0) { + regmap_write(es8326->regmap, ES8326_CLK_DIV1_04, + coeff_div[coeff].reg4); + regmap_write(es8326->regmap, ES8326_CLK_DIV2_05, + coeff_div[coeff].reg5); + regmap_write(es8326->regmap, ES8326_CLK_DLL_06, + coeff_div[coeff].reg6); + regmap_write(es8326->regmap, ES8326_CLK_MUX_07, + coeff_div[coeff].reg7); + regmap_write(es8326->regmap, ES8326_CLK_ADC_SEL_08, + coeff_div[coeff].reg8); + regmap_write(es8326->regmap, ES8326_CLK_DAC_SEL_09, + coeff_div[coeff].reg9); + regmap_write(es8326->regmap, ES8326_CLK_ADC_OSR_0A, + coeff_div[coeff].rega); + regmap_write(es8326->regmap, ES8326_CLK_DAC_OSR_0B, + coeff_div[coeff].regb); + } + + return 0; +} + +static int es8326_set_bias_level(struct snd_soc_component *codec, + enum snd_soc_bias_level level) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + if (!IS_ERR(es8326->mclk)) { + ret = clk_prepare_enable(es8326->mclk); + if (ret) + return ret; + } + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_PWRUP_SEQ_EN); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, + (ES8326_IO_DMIC_CLK << ES8326_SDINOUT1_SHIFT)); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_CLK_RESAMPLE_03, 0x05); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x02); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0x40); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0xAA); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + break; + case SND_SOC_BIAS_OFF: + if (es8326->mclk) + clk_disable_unprepare(es8326->mclk); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x11); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_OFF); + regmap_write(es8326->regmap, ES8326_PGA_PDN_17, 0xF8); + regmap_write(es8326->regmap, ES8326_VMIDSEL_18, 0x00); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_SDINOUT1_IO_5A, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_SDINOUT23_IO_5B, ES8326_IO_INPUT); + regmap_write(es8326->regmap, ES8326_RESET_00, + ES8326_CODEC_RESET | ES8326_PWRUP_SEQ_EN); + break; + } + + return 0; +} + +#define es8326_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops es8326_ops = { + .hw_params = es8326_pcm_hw_params, + .set_fmt = es8326_set_dai_fmt, + .set_sysclk = es8326_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver es8326_dai = { + .name = "ES8326 HiFi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = es8326_FORMATS, + }, + .ops = &es8326_ops, + .symmetric_rate = 1, +}; + +static void es8326_enable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_disable_micbias(struct snd_soc_component *component) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(component); + + snd_soc_dapm_mutex_lock(dapm); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS2"); + snd_soc_dapm_sync_unlocked(dapm); + snd_soc_dapm_mutex_unlock(dapm); +} + +static void es8326_jack_detect_handler(struct work_struct *work) +{ + struct es8326_priv *es8326 = + container_of(work, struct es8326_priv, jack_detect_work.work); + struct snd_soc_component *comp = es8326->component; + unsigned int iface; + + iface = snd_soc_component_read(comp, ES8326_HP_DETECT_FB); + dev_dbg(comp->dev, "gpio flag %#04x", iface); + if ((iface & ES8326_HPINSERT_FLAG) == 0) { + dev_dbg(comp->dev, "No headset detected"); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic2_src); + es8326_disable_micbias(comp); + } else if ((iface & ES8326_HPINSERT_FLAG) == ES8326_HPINSERT_FLAG) { + if ((iface & ES8326_HPBUTTON_FLAG) == 0x00) { + dev_dbg(comp->dev, "Headset detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADSET, SND_JACK_HEADSET); + snd_soc_component_write(comp, ES8326_ADC1_SRC_2A, es8326->mic1_src); + } else { + dev_dbg(comp->dev, "Headphone detected"); + snd_soc_jack_report(es8326->jack, SND_JACK_HEADPHONE, SND_JACK_HEADSET); + } + } +} + +static irqreturn_t es8326_irq(int irq, void *dev_id) +{ + struct es8326_priv *es8326 = dev_id; + struct snd_soc_component *comp = es8326->component; + + if (!es8326->jack) + goto out; + + es8326_enable_micbias(comp); + + queue_delayed_work(system_wq, &es8326->jack_detect_work, + msecs_to_jiffies(300)); + +out: + return IRQ_HANDLED; +} + +static int es8326_resume(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(es8326->regmap, false); + regcache_sync(es8326->regmap); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_ON); + /* Two channel ADC */ + regmap_write(es8326->regmap, ES8326_PULLUP_CTL_F9, 0x02); + regmap_write(es8326->regmap, ES8326_CLK_INV_02, 0x00); + regmap_write(es8326->regmap, ES8326_CLK_DIV_CPC_0C, 0x1F); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS1_10, 0xC8); + regmap_write(es8326->regmap, ES8326_CLK_VMIDS2_11, 0x88); + regmap_write(es8326->regmap, ES8326_CLK_CAL_TIME_12, 0x20); + regmap_write(es8326->regmap, ES8326_SYS_BIAS_1D, 0x08); + regmap_write(es8326->regmap, ES8326_DAC2HPMIX_25, 0x22); + regmap_write(es8326->regmap, ES8326_ADC1_SRC_2A, es8326->mic1_src); + regmap_write(es8326->regmap, ES8326_ADC2_SRC_2B, es8326->mic2_src); + regmap_write(es8326->regmap, ES8326_HPJACK_TIMER_56, 0x88); + regmap_write(es8326->regmap, ES8326_HP_DET_57, + ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); + regmap_write(es8326->regmap, ES8326_INT_SOURCE_58, 0x08); + regmap_write(es8326->regmap, ES8326_INTOUT_IO_59, 0x45); + regmap_write(es8326->regmap, ES8326_RESET_00, ES8326_CSM_ON); + snd_soc_component_update_bits(component, ES8326_PGAGAIN_23, + ES8326_MIC_SEL_MASK, ES8326_MIC1_SEL); + + es8326_irq(es8326->irq, es8326); + return 0; +} + +static int es8326_suspend(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + es8326_disable_micbias(component); + cancel_delayed_work_sync(&es8326->jack_detect_work); + + regmap_write(es8326->regmap, ES8326_CLK_CTL_01, ES8326_CLK_OFF); + regcache_cache_only(es8326->regmap, true); + regcache_mark_dirty(es8326->regmap); + + return 0; +} + +static int es8326_probe(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + int ret; + + es8326->component = component; + es8326->jd_inverted = device_property_read_bool(component->dev, + "everest,jack-detect-inverted"); + + ret = device_property_read_u8(component->dev, "everest,mic1-src", &es8326->mic1_src); + if (ret != 0) { + dev_dbg(component->dev, "mic1-src return %d", ret); + es8326->mic1_src = ES8326_ADC_AMIC; + } + dev_dbg(component->dev, "mic1-src %x", es8326->mic1_src); + + ret = device_property_read_u8(component->dev, "everest,mic2-src", &es8326->mic2_src); + if (ret != 0) { + dev_dbg(component->dev, "mic2-src return %d", ret); + es8326->mic2_src = ES8326_ADC_DMIC; + } + dev_dbg(component->dev, "mic2-src %x", es8326->mic2_src); + + ret = device_property_read_u8(component->dev, "everest,jack-pol", &es8326->jack_pol); + if (ret != 0) { + dev_dbg(component->dev, "jack-pol return %d", ret); + es8326->jack_pol = ES8326_HP_DET_BUTTON_POL | ES8326_HP_TYPE_OMTP; + } + dev_dbg(component->dev, "jack-pol %x", es8326->jack_pol); + + es8326_resume(component); + return 0; +} + +static void es8326_enable_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *jack) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + mutex_lock(&es8326->lock); + if (es8326->jd_inverted) + snd_soc_component_update_bits(component, ES8326_HP_DET_57, + ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); + es8326->jack = jack; + + mutex_unlock(&es8326->lock); + es8326_irq(es8326->irq, es8326); +} + +static void es8326_disable_jack_detect(struct snd_soc_component *component) +{ + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); + + dev_dbg(component->dev, "Enter into %s\n", __func__); + if (!es8326->jack) + return; /* Already disabled (or never enabled) */ + cancel_delayed_work_sync(&es8326->jack_detect_work); + + mutex_lock(&es8326->lock); + if (es8326->jack->status & SND_JACK_MICROPHONE) { + es8326_disable_micbias(component); + snd_soc_jack_report(es8326->jack, 0, SND_JACK_HEADSET); + } + es8326->jack = NULL; + mutex_unlock(&es8326->lock); +} + +static int es8326_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + if (jack) + es8326_enable_jack_detect(component, jack); + else + es8326_disable_jack_detect(component); + + return 0; +} + +static void es8326_remove(struct snd_soc_component *component) +{ + es8326_disable_jack_detect(component); + es8326_set_bias_level(component, SND_SOC_BIAS_OFF); +} + +static const struct snd_soc_component_driver soc_component_dev_es8326 = { + .probe = es8326_probe, + .remove = es8326_remove, + .resume = es8326_resume, + .suspend = es8326_suspend, + .set_bias_level = es8326_set_bias_level, + .set_jack = es8326_set_jack, + .dapm_widgets = es8326_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(es8326_dapm_widgets), + .dapm_routes = es8326_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(es8326_dapm_routes), + .controls = es8326_snd_controls, + .num_controls = ARRAY_SIZE(es8326_snd_controls), + .use_pmdown_time = 1, + .endianness = 1, +}; + +static int es8326_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct es8326_priv *es8326; + int ret; + + es8326 = devm_kzalloc(&i2c->dev, sizeof(struct es8326_priv), GFP_KERNEL); + if (!es8326) + return -ENOMEM; + + i2c_set_clientdata(i2c, es8326); + es8326->i2c = i2c; + mutex_init(&es8326->lock); + es8326->regmap = devm_regmap_init_i2c(i2c, &es8326_regmap_config); + if (IS_ERR(es8326->regmap)) { + ret = PTR_ERR(es8326->regmap); + dev_err(&i2c->dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + es8326->irq = i2c->irq; + INIT_DELAYED_WORK(&es8326->jack_detect_work, + es8326_jack_detect_handler); + /* ES8316 is level-based while ES8326 is edge-based */ + ret = devm_request_threaded_irq(&i2c->dev, es8326->irq, NULL, es8326_irq, + IRQF_TRIGGER_RISING | IRQF_ONESHOT, + "es8326", es8326); + if (ret) { + dev_warn(&i2c->dev, "Failed to reguest IRQ: %d: %d\n", + es8326->irq, ret); + es8326->irq = -ENXIO; + } + + es8326->mclk = devm_clk_get_optional(&i2c->dev, "mclk"); + if (IS_ERR(es8326->mclk)) { + dev_err(&i2c->dev, "unable to get mclk\n"); + return PTR_ERR(es8326->mclk); + } + if (!es8326->mclk) + dev_warn(&i2c->dev, "assuming static mclk\n"); + + ret = clk_prepare_enable(es8326->mclk); + if (ret) { + dev_err(&i2c->dev, "unable to enable mclk\n"); + return ret; + } + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_es8326, + &es8326_dai, 1); +} + +static const struct i2c_device_id es8326_i2c_id[] = { + {"es8326", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, es8326_i2c_id); + +#ifdef CONFIG_OF +static const struct of_device_id es8326_of_match[] = { + { .compatible = "everest,es8326", }, + {} +}; +MODULE_DEVICE_TABLE(of, es8326_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id es8326_acpi_match[] = { + {"ESSX8326", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, es8326_acpi_match); +#endif + +static struct i2c_driver es8326_i2c_driver = { + .driver = { + .name = "es8326", + .acpi_match_table = ACPI_PTR(es8326_acpi_match), + .of_match_table = of_match_ptr(es8326_of_match), + }, + .probe = es8326_i2c_probe, + .id_table = es8326_i2c_id, +}; +module_i2c_driver(es8326_i2c_driver); + +MODULE_DESCRIPTION("ASoC es8326 driver"); +MODULE_AUTHOR("David Yang <yangxiaohua(a)everest-semi.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h new file mode 100755 index 000000000000..717f3fd42372 --- /dev/null +++ b/sound/soc/codecs/es8326.h @@ -0,0 +1,187 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * es8326.h -- es8326 ALSA SoC audio driver + * Copyright Everest Semiconductor Co.,Ltd + * + * Authors: David Yang <yangxiaohua(a)everest-semi.com> + */ + +#ifndef _ES8326_H +#define _ES8326_H + +#define CONFIG_HHTECH_MINIPMP 1 + +/* ES8326 register space */ + +#define ES8326_RESET_00 0x00 +#define ES8326_CLK_CTL_01 0x01 +#define ES8326_CLK_INV_02 0x02 +#define ES8326_CLK_RESAMPLE_03 0x03 +#define ES8326_CLK_DIV1_04 0x04 +#define ES8326_CLK_DIV2_05 0x05 +#define ES8326_CLK_DLL_06 0x06 +#define ES8326_CLK_MUX_07 0x07 +#define ES8326_CLK_ADC_SEL_08 0x08 +#define ES8326_CLK_DAC_SEL_09 0x09 +#define ES8326_CLK_ADC_OSR_0A 0x0a +#define ES8326_CLK_DAC_OSR_0B 0x0b +#define ES8326_CLK_DIV_CPC_0C 0x0c +#define ES8326_CLK_DIV_BCLK_0D 0x0d +#define ES8326_CLK_TRI_0E 0x0e +#define ES8326_CLK_DIV_LRCK_0F 0x0f +#define ES8326_CLK_VMIDS1_10 0x10 +#define ES8326_CLK_VMIDS2_11 0x11 +#define ES8326_CLK_CAL_TIME_12 0x12 +#define ES8326_FMT_13 0x13 + +#define ES8326_DAC_MUTE_14 0x14 +#define ES8326_ADC_MUTE_15 0x15 +#define ES8326_ANA_PDN_16 0x16 +#define ES8326_PGA_PDN_17 0x17 +#define ES8326_VMIDSEL_18 0x18 +#define ES8326_ANA_LOWPOWER_19 0x19 +#define ES8326_ANA_DMS_1A 0x1a +#define ES8326_ANA_MICBIAS_1B 0x1b +#define ES8326_ANA_VSEL_1C 0x1c +#define ES8326_SYS_BIAS_1D 0x1d +#define ES8326_BIAS_SW1_1E 0x1e +#define ES8326_BIAS_SW2_1F 0x1f +#define ES8326_BIAS_SW3_20 0x20 +#define ES8326_BIAS_SW4_21 0x21 +#define ES8326_VMIDLOW_22 0x22 + +#define ES8326_PGAGAIN_23 0x23 +#define ES8326_HP_DRIVER_24 0x24 +#define ES8326_DAC2HPMIX_25 0x25 +#define ES8326_HP_VOL_26 0x26 +#define ES8326_HP_CAL_27 0x27 +#define ES8326_HP_DRIVER_REF_28 0x28 +#define ES8326_ADC_SCALE_29 0x29 +#define ES8326_ADC1_SRC_2A 0x2a +#define ES8326_ADC2_SRC_2B 0x2b +#define ES8326_ADC1_VOL_2C 0x2c +#define ES8326_ADC2_VOL_2D 0x2d +#define ES8326_ADC_RAMPRATE_2E 0x2e +#define ES8326_2F 0x2f +#define ES8326_30 0x30 +#define ES8326_31 0x31 +#define ES8326_ALC_RECOVERY_32 0x32 +#define ES8326_ALC_LEVEL_33 0x33 +#define ES8326_ADC_HPFS1_34 0x34 +#define ES8326_ADC_HPFS2_35 0x35 +#define ES8326_ADC_EQ_36 0x36 +#define ES8326_HP_CAL_4A 0x4A +#define ES8326_HPL_OFFSET_INI_4B 0x4B +#define ES8326_HPR_OFFSET_INI_4C 0x4C +#define ES8326_DAC_DSM_4D 0x4D +#define ES8326_DAC_RAMPRATE_4E 0x4E +#define ES8326_DAC_VPPSCALE_4F 0x4F +#define ES8326_DAC_VOL_50 0x50 +#define ES8326_DRC_RECOVERY_53 0x53 +#define ES8326_DRC_WINSIZE_54 0x54 +#define ES8326_HPJACK_TIMER_56 0x56 +#define ES8326_HP_DET_57 0x57 +#define ES8326_INT_SOURCE_58 0x58 +#define ES8326_INTOUT_IO_59 0x59 +#define ES8326_SDINOUT1_IO_5A 0x5A +#define ES8326_SDINOUT23_IO_5B 0x5B +#define ES8326_JACK_PULSE_5C 0x5C + +#define ES8326_PULLUP_CTL_F9 0xF9 +#define ES8326_HP_DETECT_FB 0xFB +#define ES8326_CHIP_ID1_FD 0xFD +#define ES8326_CHIP_ID2_FE 0xFE +#define ES8326_CHIP_VERSION_FF 0xFF + +/* ES8326_RESET_00 */ +#define ES8326_CSM_ON (1 << 7) +#define ES8326_MASTER_MODE_EN (1 << 6) +#define ES8326_PWRUP_SEQ_EN (1 << 5) +#define ES8326_CODEC_RESET (0x0f << 0) +#define ES8326_CSM_OFF (0 << 7) + +/* ES8326_CLK_CTL_01 */ +#define ES8326_CLK_ON (0x7f << 0) +#define ES8326_CLK_OFF (0 << 0) + +/* ES8326_CLK_INV_02 */ +#define ES8326_BCLK_AS_MCLK (1 << 3) + +/* ES8326_FMT_13 */ +#define ES8326_S24_LE (0 << 2) +#define ES8326_S20_3_LE (1 << 2) +#define ES8326_S18_LE (2 << 2) +#define ES8326_S16_LE (3 << 2) +#define ES8326_S32_LE (4 << 2) +#define ES8326_DATA_LEN_MASK (7 << 2) + +#define ES8326_DAIFMT_MASK ((1 << 5) | (3 << 0)) +#define ES8326_DAIFMT_I2S 0 +#define ES8326_DAIFMT_LEFT_J (1 << 0) +#define ES8326_DAIFMT_DSP_A (3 << 0) +#define ES8326_DAIFMT_DSP_B ((1 << 5) | (3 << 0)) + +/* ES8326_PGAGAIN_23 */ +#define ES8326_MIC_SEL_MASK (3 << 4) +#define ES8326_MIC1_SEL (1 << 4) +#define ES8326_MIC2_SEL (1 << 5) + +/* ES8326_HP_CAL_27 */ +#define ES8326_HPOR_SHIFT 4 + +/* ES8326_ADC1_SRC_2A */ +#define ES8326_ADC1_SHIFT 0 +#define ES8326_ADC2_SHIFT 4 +#define ES8326_ADC_SRC_ANA 0 +#define ES8326_ADC_SRC_ANA_INV_SW0 1 +#define ES8326_ADC_SRC_ANA_INV_SW1 2 +#define ES8326_ADC_SRC_DMIC_MCLK 3 +#define ES8326_ADC_SRC_DMIC_SDIN2 4 +#define ES8326_ADC_SRC_DMIC_SDIN2_INV 5 +#define ES8326_ADC_SRC_DMIC_SDIN3 6 +#define ES8326_ADC_SRC_DMIC_SDIN3_INV 7 + +#define ES8326_ADC_AMIC ((ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_ANA_INV_SW1 << ES8326_ADC1_SHIFT)) +#define ES8326_ADC_DMIC ((ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC2_SHIFT) \ + | (ES8326_ADC_SRC_DMIC_SDIN2 << ES8326_ADC1_SHIFT)) +/* ES8326_ADC2_SRC_2B */ +#define ES8326_ADC3_SHIFT 0 +#define ES8326_ADC4_SHIFT 3 + +/* ES8326_HP_DET_57 */ +#define ES8326_HP_DET_SRC_PIN27 (1 << 5) +#define ES8326_HP_DET_SRC_PIN9 (1 << 4) +#define ES8326_HP_DET_JACK_POL (1 << 3) +#define ES8326_HP_DET_BUTTON_POL (1 << 2) +#define ES8326_HP_TYPE_OMTP (3 << 0) +#define ES8326_HP_TYPE_CTIA (2 << 0) +#define ES8326_HP_TYPE_AUTO (1 << 0) +#define ES8326_HP_TYPE_AUTO_INV (0 << 0) + +/* ES8326_SDINOUT1_IO_5A */ +#define ES8326_IO_INPUT (0 << 0) +#define ES8326_IO_SDIN_SLOT0 (1 << 0) +#define ES8326_IO_SDIN_SLOT1 (2 << 0) +#define ES8326_IO_SDIN_SLOT2 (3 << 0) +#define ES8326_IO_SDIN_SLOT7 (8 << 0) +#define ES8326_IO_DMIC_CLK (9 << 0) +#define ES8326_IO_DMIC_CLK_INV (0x0a << 0) +#define ES8326_IO_SDOUT2 (0x0b << 0) +#define ES8326_IO_LOW (0x0e << 0) +#define ES8326_IO_HIGH (0x0f << 0) +#define ES8326_ADC2DAC (1 << 3) +#define ES8326_SDINOUT1_SHIFT 4 + +/* ES8326_SDINOUT23_IO_5B */ +#define ES8326_SDINOUT2_SHIFT 4 +#define ES8326_SDINOUT3_SHIFT 0 + +/* ES8326_HP_DETECT_FB */ +#define ES8326_HPINSERT_FLAG (1 << 1) +#define ES8326_HPBUTTON_FLAG (1 << 0) + +/* ES8326_CHIP_VERSION_FF 0xFF */ +#define ES8326_VERSION_B (1 << 0) + +#endif -- 2.36.1

4 8

Mega r1.1
by GitHub pull_request - opened 29 Jul '22

29 Jul '22

alsa-project/alsa-ucm-conf pull request #195 was opened from 13r0ck: requires https://github.com/pop-os/linux/pull/167 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/195 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/195.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

HDA ATI HDMI not supported
by GitHub issues - edited 29 Jul '22

29 Jul '22

alsa-project/alsa-ucm-conf issue #194 was edited from yangxiaohua2009: I am using the latest ucm2 file and alsalib alsainfo [http://alsa-project.org/db/?f=92c58f3a6b72fee05b678386858dbc816d25dbf3](htt… The `alsaucm reload` command returns error saying: `ALSA lib parser.c:244:(error_node) UCM is not supported for this HDA model (HDA ATI HDMI at 0xfeb60000 irq 28)` and the HDMI is not working. despite the fact that I already added the acpes8336.conf in conf.d and acpes8336 folder. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/194 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: dt-bindings: fsl,sai: Convert format to json-schema
by Shengjiu Wang 29 Jul '22

29 Jul '22

Convert the NXP SAI binding to DT schema format using json-schema. The Synchronous Audio Interface (SAI) provides an interface that supports full-duplex serial interfaces with frame synchronization formats such as I2S, AC97, TDM, and codec/DSP interfaces. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- changes in v2 - fix exclusive property issue - fix order issue of compatible, clock-names, dma-names .../devicetree/bindings/sound/fsl,sai.yaml | 215 ++++++++++++++++++ .../devicetree/bindings/sound/fsl-sai.txt | 95 -------- 2 files changed, 215 insertions(+), 95 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,sai.yaml delete mode 100644 Documentation/devicetree/bindings/sound/fsl-sai.txt diff --git a/Documentation/devicetree/bindings/sound/fsl,sai.yaml b/Documentation/devicetree/bindings/sound/fsl,sai.yaml new file mode 100644 index 000000000000..3e3d99febd69 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/fsl,sai.yaml @@ -0,0 +1,215 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/fsl,sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Freescale Synchronous Audio Interface (SAI). + +maintainers: + - Shengjiu Wang <shengjiu.wang(a)nxp.com> + +description: | + The SAI is based on I2S module that used communicating with audio codecs, + which provides a synchronous audio interface that supports fullduplex + serial interfaces with frame synchronization such as I2S, AC97, TDM, and + codec/DSP interfaces. + +properties: + compatible: + oneOf: + - const: fsl,vf610-sai + - const: fsl,imx6sx-sai + - const: fsl,imx6ul-sai + - const: fsl,imx7ulp-sai + - const: fsl,imx8mq-sai + - const: fsl,imx8qm-sai + - const: fsl,imx8ulp-sai + - items: + - enum: + - fsl,imx8mm-sai + - fsl,imx8mn-sai + - fsl,imx8mp-sai + - const: fsl,imx8mq-sai + + reg: + maxItems: 1 + + interrupts: + items: + - description: receive and transmit interrupt + + dmas: + maxItems: 2 + + dma-names: + maxItems: 2 + + clocks: + items: + - description: The ipg clock for register access + - description: master clock source 0 (obsoleted) + - description: master clock source 1 + - description: master clock source 2 + - description: master clock source 3 + - description: PLL clock source for 8kHz series + - description: PLL clock source for 11kHz series + minItems: 4 + + clock-names: + oneOf: + - items: + - const: bus + - const: mclk0 + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + - items: + - const: bus + - const: mclk1 + - const: mclk2 + - const: mclk3 + - const: pll8k + - const: pll11k + minItems: 4 + + lsb-first: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Configures whether the LSB or the MSB is transmitted + first for the fifo data. If this property is absent, + the MSB is transmitted first as default, or the LSB + is transmitted first. + + big-endian: + description: | + required if all the SAI registers are big-endian rather than little-endian. + type: boolean + + fsl,sai-synchronous-rx: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the synchronous mode (sync Tx with Rx) which means + both the transmitter and the receiver will send and receive data by + following receiver's bit clocks and frame sync clocks. + + fsl,sai-asynchronous: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will work in the asynchronous mode, which means both transmitter + and receiver will send and receive data by following their own bit clocks + and frame sync clocks separately. + If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the + default synchronous mode (sync Rx with Tx) will be used, which means both + transmitter and receiver will send and receive data by following clocks + of transmitter. + + fsl,dataline: + $ref: /schemas/types.yaml#/definitions/uint32-matrix + description: | + Configure the dataline. It has 3 value for each configuration + items: + items: + - description: format Default(0), I2S(1) or PDM(2) + enum: [0, 1, 2] + - description: dataline mask for 'rx' + - description: dataline mask for 'tx' + + fsl,sai-mclk-direction-output: + $ref: /schemas/types.yaml#/definitions/flag + description: | + SAI will output the SAI MCLK clock. + + fsl,shared-interrupt: + $ref: /schemas/types.yaml#/definitions/flag + description: | + Interrupt is shared with other modules. + + "#sound-dai-cells": + const: 0 + +allOf: + - if: + properties: + compatible: + contains: + const: fsl,vf610-sai + then: + properties: + dmas: + items: + - description: DMA controller phandle and request line for TX + - description: DMA controller phandle and request line for RX + dma-names: + items: + - const: tx + - const: rx + else: + properties: + dmas: + items: + - description: DMA controller phandle and request line for RX + - description: DMA controller phandle and request line for TX + dma-names: + items: + - const: rx + - const: tx + - if: + required: + - fsl,sai-asynchronous + then: + properties: + fsl,sai-synchronous-rx: false + +required: + - compatible + - reg + - interrupts + - dmas + - dma-names + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/vf610-clock.h> + sai2: sai@40031000 { + compatible = "fsl,vf610-sai"; + reg = <0x40031000 0x1000>; + interrupts = <86 IRQ_TYPE_LEVEL_HIGH>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai2_1>; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; + dma-names = "tx", "rx"; + dmas = <&edma0 0 21>, + <&edma0 0 20>; + big-endian; + lsb-first; + }; + + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/imx8mm-clock.h> + sai1: sai@30010000 { + #sound-dai-cells = <0>; + compatible = "fsl,imx8mm-sai", "fsl,imx8mq-sai"; + reg = <0x30010000 0x10000>; + interrupts = <GIC_SPI 95 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&clk IMX8MM_CLK_SAI1_IPG>, + <&clk IMX8MM_CLK_DUMMY>, + <&clk IMX8MM_CLK_SAI1_ROOT>, + <&clk IMX8MM_CLK_DUMMY>, <&clk IMX8MM_CLK_DUMMY>; + clock-names = "bus", "mclk0", "mclk1", "mclk2", "mclk3"; + dmas = <&sdma2 0 2 0>, <&sdma2 1 2 0>; + dma-names = "rx", "tx"; + fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; + }; diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt deleted file mode 100644 index fbdefc3fade7..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ /dev/null @@ -1,95 +0,0 @@ -Freescale Synchronous Audio Interface (SAI). - -The SAI is based on I2S module that used communicating with audio codecs, -which provides a synchronous audio interface that supports fullduplex -serial interfaces with frame synchronization such as I2S, AC97, TDM, and -codec/DSP interfaces. - -Required properties: - - - compatible : Compatible list, contains "fsl,vf610-sai", - "fsl,imx6sx-sai", "fsl,imx6ul-sai", - "fsl,imx7ulp-sai", "fsl,imx8mq-sai", - "fsl,imx8qm-sai", "fsl,imx8mm-sai", - "fsl,imx8mn-sai", "fsl,imx8mp-sai", or - "fsl,imx8ulp-sai". - - - reg : Offset and length of the register set for the device. - - - clocks : Must contain an entry for each entry in clock-names. - - - clock-names : Must include the "bus" for register access and - "mclk1", "mclk2", "mclk3" for bit clock and frame - clock providing. - "pll8k", "pll11k" are optional, they are the clock - source for root clock, one is for 8kHz series rates - another one is for 11kHz series rates. - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - pinctrl-names : Must contain a "default" entry. - - - pinctrl-NNN : One property must exist for each entry in - pinctrl-names. See ../pinctrl/pinctrl-bindings.txt - for details of the property values. - - - lsb-first : Configures whether the LSB or the MSB is transmitted - first for the fifo data. If this property is absent, - the MSB is transmitted first as default, or the LSB - is transmitted first. - - - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating - that SAI will work in the synchronous mode (sync Tx - with Rx) which means both the transmitter and the - receiver will send and receive data by following - receiver's bit clocks and frame sync clocks. - - - fsl,sai-asynchronous: This is a boolean property. If present, indicating - that SAI will work in the asynchronous mode, which - means both transmitter and receiver will send and - receive data by following their own bit clocks and - frame sync clocks separately. - - - fsl,dataline : configure the dataline. it has 3 value for each configuration - first one means the type: I2S(1) or PDM(2) - second one is dataline mask for 'rx' - third one is dataline mask for 'tx'. - for example: fsl,dataline = <1 0xff 0xff 2 0xff 0x11>; - it means I2S type rx mask is 0xff, tx mask is 0xff, PDM type - rx mask is 0xff, tx mask is 0x11 (dataline 1 and 4 enabled). - -Optional properties: - - - big-endian : Boolean property, required if all the SAI - registers are big-endian rather than little-endian. - -Optional properties (for mx6ul): - - - fsl,sai-mclk-direction-output: This is a boolean property. If present, - indicates that SAI will output the SAI MCLK clock. - -Note: -- If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the - default synchronous mode (sync Rx with Tx) will be used, which means both - transmitter and receiver will send and receive data by following clocks - of transmitter. -- fsl,sai-asynchronous and fsl,sai-synchronous-rx are exclusive. - -Example: -sai2: sai@40031000 { - compatible = "fsl,vf610-sai"; - reg = <0x40031000 0x1000>; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_PLATFORM_BUS>, - <&clks VF610_CLK_SAI2>, - <&clks 0>, <&clks 0>; - clock-names = "bus", "mclk1", "mclk2", "mclk3"; - dma-names = "tx", "rx"; - dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, - <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; - big-endian; - lsb-first; -}; -- 2.34.1

3 2

[GIT PULL] ASoC fixes for v5.19-rc8
by Mark Brown 28 Jul '22

28 Jul '22

The following changes since commit 1e347f861da8ddb17e1d1b3113cb6c188e0de3e5: ASoC: rockchip-i2s: Undo BCLK pinctrl changes (2022-07-14 13:25:52 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.19-rc8 for you to fetch changes up to 30097967e0566cac817273ef76add100f6b0f463: ASoC: codecs: va-macro: use fsgen as clock (2022-07-27 14:46:41 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.19 A few more small fixes, they could all wait for the merge window if you prefer. ---------------------------------------------------------------- Ryan Lee (1): ASoC: max98373: Removing 0x203E from the volatile reg Srinivas Kandagatla (1): ASoC: codecs: va-macro: use fsgen as clock Wang Jiada (1): ASoC: rsnd: ssiu: add missing .quit callback for gen2 sound/soc/codecs/lpass-va-macro.c | 11 +++++++++-- sound/soc/codecs/max98373-i2c.c | 1 - sound/soc/sh/rcar/ssiu.c | 1 + 3 files changed, 10 insertions(+), 3 deletions(-)

2 1

[PATCH] ASoC: qcom: SC7280: Add support for external DMIC bias supply
by Srinivasa Rao Mandadapu 28 Jul '22

28 Jul '22

Update SC7280 machine driver for enabling external mic bias supply, Which is required for villager rev boards. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> --- sound/soc/qcom/sc7280.c | 24 ++++++++++++++++++++++++ 1 file changed, 24 insertions(+) diff --git a/sound/soc/qcom/sc7280.c b/sound/soc/qcom/sc7280.c index da7469a..6404e94 100644 --- a/sound/soc/qcom/sc7280.c +++ b/sound/soc/qcom/sc7280.c @@ -33,6 +33,7 @@ struct sc7280_snd_data { struct snd_soc_jack hdmi_jack; bool jack_setup; bool stream_prepared[LPASS_MAX_PORTS]; + struct regulator *vdd_supply; }; static void sc7280_jack_free(struct snd_jack *jack) @@ -345,6 +346,24 @@ static int sc7280_snd_startup(struct snd_pcm_substream *substream) return ret; } +static int sc7280_dmic_micbias(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_card *card = w->dapm->card; + struct sc7280_snd_data *data = snd_soc_card_get_drvdata(card); + int ret = 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + ret = regulator_enable(data->vdd_supply); + break; + case SND_SOC_DAPM_POST_PMD: + ret = regulator_disable(data->vdd_supply); + break; + } + return ret; +} + static const struct snd_soc_ops sc7280_ops = { .startup = sc7280_snd_startup, .hw_params = sc7280_snd_hw_params, @@ -356,6 +375,7 @@ static const struct snd_soc_ops sc7280_ops = { static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("PMIC BIAS", sc7280_dmic_micbias), }; static int sc7280_snd_platform_probe(struct platform_device *pdev) @@ -389,6 +409,10 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev) link->ops = &sc7280_ops; } + data->vdd_supply = devm_regulator_get(dev, "vdd-dmic-bias"); + if (IS_ERR(data->vdd_supply) + return PTR_ERR(data->vdd_supply); + return devm_snd_soc_register_card(dev, card); } -- 2.7.4

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