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

[PATCH v2] ASoC: atmel: Convert to new style DAI format definitions
by Mark Brown 16 Sep '21

16 Sep '21

Convert the Atmel drivers to use the new style defines for clocking in DAI formats. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/atmel-i2s.c | 4 ++-- sound/soc/atmel/atmel_ssc_dai.c | 18 +++++++++--------- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 8 ++++---- sound/soc/atmel/mikroe-proto.c | 4 ++-- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- 8 files changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6b3d9c05eaf2..e119abbca099 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -343,7 +343,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* codec is slave, so cpu is master */ mr |= ATMEL_I2SC_MR_MODE_MASTER; ret = atmel_i2s_get_gck_param(dev, params_rate(params)); @@ -351,7 +351,7 @@ static int atmel_i2s_hw_params(struct snd_pcm_substream *substream, return ret; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* codec is master, so cpu is slave */ mr |= ATMEL_I2SC_MR_MODE_SLAVE; dev->gck_param = NULL; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6a63e8797a0b..b34e995c59e4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -210,7 +210,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, return frame_size; switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: if ((ssc_p->dir_mask & SSC_DIR_MASK_CAPTURE) && ssc->clk_from_rk_pin) /* Receiver Frame Synchro (i.e. capture) @@ -220,7 +220,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, mck_div = 3; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: if ((ssc_p->dir_mask & SSC_DIR_MASK_PLAYBACK) && !ssc->clk_from_rk_pin) /* Transmit Frame Synchro (i.e. playback) @@ -233,7 +233,7 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: r.num = ssc_p->mck_rate / mck_div / frame_size; ret = snd_interval_ratnum(i, 1, &r, &num, &den); @@ -243,8 +243,8 @@ static int atmel_ssc_hw_rule_rate(struct snd_pcm_hw_params *params, } break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBP_CFP: t.min = 8000; t.max = ssc_p->mck_rate / mck_div / frame_size; t.openmin = t.openmax = 0; @@ -430,8 +430,8 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, static int atmel_ssc_cfs(struct atmel_ssc_info *ssc_p) { switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBP_CFC: + case SND_SOC_DAIFMT_CBC_CFC: return 1; } return 0; @@ -441,8 +441,8 @@ static int atmel_ssc_cfs(struct atmel_ssc_info *ssc_p) static int atmel_ssc_cbs(struct atmel_ssc_info *ssc_p) { switch (ssc_p->daifmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFP: + case SND_SOC_DAIFMT_CBC_CFC: return 1; } return 0; diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index 9e237580afa9..9c974c4e187d 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -66,7 +66,7 @@ static struct snd_soc_dai_link atmel_asoc_wm8904_dailink = { .stream_name = "WM8904 PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, + | SND_SOC_DAIFMT_CBP_CFP, .ops = &atmel_asoc_wm8904_ops, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 8988f024a732..67550e3d9ee3 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -350,7 +350,7 @@ static int mchp_i2s_mcc_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; /* We can't generate only FSYNC */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFS) + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBP_CFC) return -EINVAL; /* We can only reconfigure the IP when it's stopped */ @@ -547,19 +547,19 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, } switch (dev->fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* cpu is BCLK and LRC master */ mra |= MCHP_I2SMCC_MRA_MODE_MASTER; if (dev->sysclk) mra |= MCHP_I2SMCC_MRA_IMCKMODE_GEN; set_divs = 1; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: /* cpu is BCLK master */ mrb |= MCHP_I2SMCC_MRB_CLKSEL_INT; set_divs = 1; fallthrough; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: /* cpu is slave */ mra |= MCHP_I2SMCC_MRA_MODE_SLAVE; if (dev->sysclk) diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index 0be7b4221c14..f9331f7e80fe 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -129,9 +129,9 @@ static int snd_proto_probe(struct platform_device *pdev) } if (bitclkmaster) { if (codec_np == bitclkmaster) - dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; else - dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; + dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; } else { dai_fmt |= snd_soc_daifmt_parse_clock_provider_as_flag(np, NULL); } diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index ed1f69b57024..915da92e1ec8 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -126,7 +126,7 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .stream_name = "WM8731 PCM", .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, SND_SOC_DAILINK_REG(pcm), }; diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 7745250fd743..7c45dc4f8c1b 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -115,7 +115,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) dai->codecs->dai_name = "wm8731-hifi"; dai->init = sam9x5_wm8731_init; dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM; + | SND_SOC_DAIFMT_CBP_CFP; ret = snd_soc_of_parse_card_name(card, "atmel,model"); if (ret) { diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 50c3dc6936f9..1b3a31296c9b 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -304,7 +304,7 @@ static struct snd_soc_dai_link tse850_dailink = { .stream_name = "TSE-850-PCM", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFS, + | SND_SOC_DAIFMT_CBP_CFC, SND_SOC_DAILINK_REG(pcm), }; -- 2.20.1

3 3

[PATCH 0/6] ASoC: SOF: Clean up the probe support
by Peter Ujfalusi 16 Sep '21

16 Sep '21

Hi, The probe debug feature of SOF can be used to extract streams of data from a given point of a pipeline for analysis. The support is implemented by using the ALSA/ASoC compress support for the capture stream, but the code can not be used by/for a normal compressed data stream. It is a debug feature. Merge the probe implementation in the core (compress.c/h and probe.c/h) into one file: sof-probes.c/h Rename the Intel HDA specific probe implementation from hda-compressc.c to hda-probes.c We also need to add IPC logging support for the probes messages and drop the unused references to SOF compress to have reasonably clean code. Regards, Peter --- Peter Ujfalusi (5): ASoC: SOF: ipc: Add probe message logging to ipc_log_header() ASoC: SOF: pcm: Remove non existent CONFIG_SND_SOC_SOF_COMPRESS reference ASoC: SOF: probe: Merge and clean up the probe and compress files ASoC: SOF: Intel: Rename hda-compress.c to hda-probes.c ASoC: SOF: sof-probes: Correct the function names used for snd_soc_cdai_ops Ranjani Sridharan (1): ASoC: SOF: compress: move and export sof_probe_compr_ops sound/soc/sof/Makefile | 3 +- sound/soc/sof/compress.c | 147 --------- sound/soc/sof/compress.h | 32 -- sound/soc/sof/core.c | 2 +- sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/Makefile | 2 +- sound/soc/sof/intel/hda-dai.c | 16 +- .../intel/{hda-compress.c => hda-probes.c} | 0 sound/soc/sof/ipc.c | 23 ++ sound/soc/sof/pcm.c | 6 +- sound/soc/sof/probe.h | 85 ------ sound/soc/sof/sof-priv.h | 5 - sound/soc/sof/{probe.c => sof-probes.c} | 280 +++++++++++------- sound/soc/sof/sof-probes.h | 38 +++ 14 files changed, 248 insertions(+), 393 deletions(-) delete mode 100644 sound/soc/sof/compress.c delete mode 100644 sound/soc/sof/compress.h rename sound/soc/sof/intel/{hda-compress.c => hda-probes.c} (100%) delete mode 100644 sound/soc/sof/probe.h rename sound/soc/sof/{probe.c => sof-probes.c} (52%) create mode 100644 sound/soc/sof/sof-probes.h -- 2.33.0

2 7

[PATCH] ASoC: au1x: Convert to modern terminology for DAI clocking
by Mark Brown 16 Sep '21

16 Sep '21

As part of retiring the old macros defining the DAI clocking mode in the DAI format update the au1x drivers to use the new style macros. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/au1x/db1200.c | 6 +++--- sound/soc/au1x/i2sc.c | 4 ++-- sound/soc/au1x/psc-i2s.c | 8 ++++---- 3 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 5f8baad37a40..400eaf9f8b14 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -117,7 +117,7 @@ static struct snd_soc_dai_link db1200_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &db1200_i2s_wm8731_ops, SND_SOC_DAILINK_REG(db1200_i2s), }; @@ -138,7 +138,7 @@ static struct snd_soc_dai_link db1300_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &db1200_i2s_wm8731_ops, SND_SOC_DAILINK_REG(db1300_i2s), }; @@ -159,7 +159,7 @@ static struct snd_soc_dai_link db1550_i2s_dai = { .name = "WM8731", .stream_name = "WM8731 PCM", .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, + SND_SOC_DAIFMT_CBP_CFP, .ops = &db1200_i2s_wm8731_ops, SND_SOC_DAILINK_REG(db1550_i2s), }; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 65bd39f5032d..529284c1f4ad 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -119,9 +119,9 @@ static int au1xi2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) goto out; } - /* I2S controller only supports master */ + /* I2S controller only supports provider */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ + case SND_SOC_DAIFMT_CBC_CFC: /* CODEC consumer */ break; default: goto out; diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 767ce950d0da..40bbf83c492f 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -91,11 +91,11 @@ static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: /* CODEC master */ - ct |= PSC_I2SCFG_MS; /* PSC I2S slave mode */ + case SND_SOC_DAIFMT_CBP_CFP: /* CODEC provider */ + ct |= PSC_I2SCFG_MS; /* PSC I2S consumer mode */ break; - case SND_SOC_DAIFMT_CBS_CFS: /* CODEC slave */ - ct &= ~PSC_I2SCFG_MS; /* PSC I2S Master mode */ + case SND_SOC_DAIFMT_CBC_CFC: /* CODEC consumer */ + ct &= ~PSC_I2SCFG_MS; /* PSC I2S provider mode */ break; default: goto out; -- 2.20.1

1 1

[PATCH] ASoC: SOF: imx8m: add SAI1 info
by Daniel Baluta 16 Sep '21

16 Sep '21

From: Viorel Suman <viorel.suman(a)nxp.com> Add SAI1 instance to imx8m_dai array. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Reviewed-by: Paul Olaru <paul.olaru(a)oss.nxp.com> Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/soc/sof/imx/imx8m.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index b9f8a9e8cc3a..6dad87ed0bc8 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -253,6 +253,17 @@ static int imx8m_ipc_pcm_params(struct snd_sof_dev *sdev, } static struct snd_soc_dai_driver imx8m_dai[] = { +{ + .name = "sai1", + .playback = { + .channels_min = 1, + .channels_max = 32, + }, + .capture = { + .channels_min = 1, + .channels_max = 32, + }, +}, { .name = "sai3", .playback = { -- 2.27.0

2 1

[PATCH] misc: cs35l41: Remove unused pdn variable
by Charles Keepax 16 Sep '21

16 Sep '21

Remove pdn variable that was made redundant in an earlier patch. Fixes: c2f14cc2bcdd ("ASoC: cs35l41: Fix use of an uninitialised variable") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l41.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41.c b/sound/soc/codecs/cs35l41.c index ad86c030d9cb5..b16eb6610c0e9 100644 --- a/sound/soc/codecs/cs35l41.c +++ b/sound/soc/codecs/cs35l41.c @@ -564,7 +564,6 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, struct cs35l41_private *cs35l41 = snd_soc_component_get_drvdata(component); unsigned int val; int ret = 0; - bool pdn; switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -582,7 +581,6 @@ static int cs35l41_main_amp_event(struct snd_soc_dapm_widget *w, regmap_update_bits(cs35l41->regmap, CS35L41_PWR_CTRL1, CS35L41_GLOBAL_EN_MASK, 0); - pdn = false; ret = regmap_read_poll_timeout(cs35l41->regmap, CS35L41_IRQ1_STATUS1, val, val & CS35L41_PDN_DONE_MASK, 1000, 100000); -- 2.11.0

2 1

[PATCH] ASoC: SOF: loader: Re-phrase the missing firmware error to avoid duplication
by Peter Ujfalusi 16 Sep '21

16 Sep '21

In case the firmware is missing we will have the following in the kernel log: 1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2 2 | error: request firmware intel/sof/sof-tgl-h.ri failed err: -2 3 | you may need to download the firmware from https://github.com/thesofproject/sof-bin/ 4 | error: failed to load DSP firmware -2 5 | error: sof_probe_work failed err: -2 The first line is the standard, request_firmware() warning. The second and third line is printed in snd_sof_load_firmware_raw() Note that the first and second line is mostly identical. With this patch the log will be changed to: 1 | Direct firmware load for intel/sof/sof-tgl-h.ri failed with error -2 2 | error: sof firmware file is missing, you might need to 3 | download it from https://github.com/thesofproject/sof-bin/ 4 | error: failed to load DSP firmware -2 5 | error: sof_probe_work failed err: -2 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> --- sound/soc/sof/loader.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 2b38a77cd594..46f3900487ee 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev) ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev); if (ret < 0) { - dev_err(sdev->dev, "error: request firmware %s failed err: %d\n", - fw_filename, ret); dev_err(sdev->dev, - "you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n"); + "error: sof firmware file is missing, you might need to\n"); + dev_err(sdev->dev, + " download it from https://github.com/thesofproject/sof-bin/\n"); goto err; } else { dev_dbg(sdev->dev, "request_firmware %s successful\n", -- 2.33.0

2 1

[PATCH v1 0/1] ASoC: cs42l42: Implement Manual Type detection as fallback
by Vitaly Rodionov 16 Sep '21

16 Sep '21

For some headsets CS42L42 autodetect mode is not working correctly. They will be detected as unknown types or as headphones. According to the CS42L42 datasheet, if the headset autodetect failed, then the driver should switch to manual mode and perform a manual steps sequence. These steps were missing in the current driver code. This patch will add manual mode fallback steps in case autodetect failed. The default behavior is not affected, manual mode runs only when autodetect failed. Tested for regression with autodetect with all known headsets - no regression. Tested with all headsets customers reported as false detected: Gumdrop DropTech B1 - detected as headset OK HUAWEI AM115 - detected as headset OK UGREEN EP103 - detected as headset OK HONOR AM116 - detected as headset OK Stefan Binding (1): ASoC: cs42l42: Implement Manual Type detection as fallback sound/soc/codecs/cs42l42.c | 104 ++++++++++++++++++++++++++++++++----- sound/soc/codecs/cs42l42.h | 54 +++++++++++++++++++ 2 files changed, 146 insertions(+), 12 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: cs42l42: Minor fix all errors reported by checkpatch.pl script
by Vitaly Rodionov 16 Sep '21

16 Sep '21

Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index c586ebff45f7..d5e1e5228b5f 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -851,7 +851,7 @@ static int cs42l42_pcm_hw_params(struct snd_pcm_substream *substream, if (params_width(params) == 24) cs42l42->bclk = (cs42l42->bclk / 3) * 4; - switch(substream->stream) { + switch (substream->stream) { case SNDRV_PCM_STREAM_CAPTURE: if (channels == 2) { val |= CS42L42_ASP_TX_CH2_AP_MASK; @@ -935,7 +935,7 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) CS42L42_HP_ANA_BMUTE_MASK); cs42l42->stream_use &= ~(1 << stream); - if(!cs42l42->stream_use) { + if (!cs42l42->stream_use) { /* * Switch to the internal oscillator. * SCLK must remain running until after this clock switch. @@ -1015,7 +1015,7 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) #define CS42L42_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S24_LE |\ - SNDRV_PCM_FMTBIT_S32_LE ) + SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops cs42l42_ops = { .startup = cs42l42_dai_startup, @@ -1572,7 +1572,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) if ((~masks[5]) & irq_params_table[5].mask) { if (stickies[5] & CS42L42_HSDET_AUTO_DONE_MASK) { cs42l42_process_hs_type_detect(cs42l42); - switch(cs42l42->hs_type){ + switch (cs42l42->hs_type) { case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: snd_soc_jack_report(cs42l42->jack, SND_JACK_HEADSET, @@ -1604,7 +1604,7 @@ static irqreturn_t cs42l42_irq_thread(int irq, void *data) cs42l42->plug_state = CS42L42_TS_UNPLUG; cs42l42_cancel_hs_type_detect(cs42l42); - switch(cs42l42->hs_type){ + switch (cs42l42->hs_type) { case CS42L42_PLUG_CTIA: case CS42L42_PLUG_OMTP: snd_soc_jack_report(cs42l42->jack, 0, SND_JACK_HEADSET); -- 2.25.1

2 1

[PATCH] ASoC: cs35l41: Binding fixes
by David Rhodes 16 Sep '21

16 Sep '21

Fix warnings and errors in DT bindings Add newline at end of file Replace 'unevaluatedProperties' with 'additionalProperties' Add spi context to DT example Add #sound-dai-cells to DT example Rename to 'cirrus,cs35l41.yaml' Signed-off-by: David Rhodes <drhodes(a)opensource.cirrus.com> --- .../{cs35l41.yaml => cirrus,cs35l41.yaml} | 30 +++++++++++-------- 1 file changed, 18 insertions(+), 12 deletions(-) rename Documentation/devicetree/bindings/sound/{cs35l41.yaml => cirrus,cs35l41.yaml} (87%) diff --git a/Documentation/devicetree/bindings/sound/cs35l41.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml similarity index 87% rename from Documentation/devicetree/bindings/sound/cs35l41.yaml rename to Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml index fde78c850286..3235702ce402 100644 --- a/Documentation/devicetree/bindings/sound/cs35l41.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l41.yaml @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause %YAML 1.2 --- -$id: http://devicetree.org/schemas/sound/cs35l41.yaml# +$id: http://devicetree.org/schemas/sound/cirrus,cs35l41.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# title: Cirrus Logic CS35L41 Speaker Amplifier @@ -135,17 +135,23 @@ required: - cirrus,boost-ind-nanohenry - cirrus,boost-cap-microfarad -unevaluatedProperties: false +additionalProperties: false examples: - | - cs35l41: cs35l41@2 { - compatible = "cirrus,cs35l41"; - reg = <2>; - VA-supply = <&dummy_vreg>; - VP-supply = <&dummy_vreg>; - reset-gpios = <&gpio 110 0>; - cirrus,boost-peak-milliamp = <4500>; - cirrus,boost-ind-nanohenry = <1000>; - cirrus,boost-cap-microfarad = <15>; - }; \ No newline at end of file + spi { + #address-cells = <1>; + #size-cells = <0>; + + cs35l41: cs35l41@2 { + #sound-dai-cells = <1>; + compatible = "cirrus,cs35l41"; + reg = <2>; + VA-supply = <&dummy_vreg>; + VP-supply = <&dummy_vreg>; + reset-gpios = <&gpio 110 0>; + cirrus,boost-peak-milliamp = <4500>; + cirrus,boost-ind-nanohenry = <1000>; + cirrus,boost-cap-microfarad = <15>; + }; + }; -- 2.25.1

2 1

[PATCH] ALSA: hda/cs8409: Setup Dolphin Headset Mic as Phantom Jack
by Vitaly Rodionov 16 Sep '21

16 Sep '21

From: Stefan Binding <sbinding(a)opensource.cirrus.com> Dell's requirement to have headset mic as phantom jack on this specific dolphin hardware platform. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_cs8409.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c index 3c7ef55d016e..31ff11ab868e 100644 --- a/sound/pci/hda/patch_cs8409.c +++ b/sound/pci/hda/patch_cs8409.c @@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true, SND_JACK_HEADPHONE, NULL); + snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true, + SND_JACK_MICROPHONE, NULL); + cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID); cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID); -- 2.25.1

2 1