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[PATCH 5/5] ASoC: cs34l45: Hibernation support
by Vlad Karpovich 02 Mar '23

02 Mar '23

From: "Vlad.Karpovich" <vkarpovi(a)opensource.cirrus.com> Adds support for the amplifier hibernation controlled by DSP firmware. Signed-off-by: Vlad Karpovich <vkarpovi(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45-i2c.c | 2 + sound/soc/codecs/cs35l45-spi.c | 1 + sound/soc/codecs/cs35l45-tables.c | 6 +++ sound/soc/codecs/cs35l45.c | 80 +++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l45.h | 29 +++++++++++ 5 files changed, 118 insertions(+) diff --git a/sound/soc/codecs/cs35l45-i2c.c b/sound/soc/codecs/cs35l45-i2c.c index 5224aaf584b8..1c644ddf66bc 100644 --- a/sound/soc/codecs/cs35l45-i2c.c +++ b/sound/soc/codecs/cs35l45-i2c.c @@ -33,6 +33,8 @@ static int cs35l45_i2c_probe(struct i2c_client *client) cs35l45->dev = dev; cs35l45->irq = client->irq; + cs35l45->bus_type = CONTROL_BUS_I2C; + cs35l45->i2c_addr = client->addr; return cs35l45_probe(cs35l45); } diff --git a/sound/soc/codecs/cs35l45-spi.c b/sound/soc/codecs/cs35l45-spi.c index bfb79255783e..3ea2e6fd2b88 100644 --- a/sound/soc/codecs/cs35l45-spi.c +++ b/sound/soc/codecs/cs35l45-spi.c @@ -36,6 +36,7 @@ static int cs35l45_spi_probe(struct spi_device *spi) cs35l45->dev = dev; cs35l45->irq = spi->irq; + cs35l45->bus_type = CONTROL_BUS_SPI; return cs35l45_probe(cs35l45); } diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index 6b68773979ec..212d9cb5a4fb 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -47,6 +47,8 @@ static const struct reg_default cs35l45_defaults[] = { { CS35L45_INTB_GPIO2_MCLK_REF, 0x00000005 }, { CS35L45_GPIO3, 0x00000005 }, { CS35L45_PWRMGT_CTL, 0x00000000 }, + { CS35L45_WAKESRC_CTL, 0x00000008 }, + { CS35L45_WKI2C_CTL, 0x00000030 }, { CS35L45_REFCLK_INPUT, 0x00000510 }, { CS35L45_GLOBAL_SAMPLE_RATE, 0x00000003 }, { CS35L45_ASP_ENABLES1, 0x00000000 }, @@ -126,6 +128,9 @@ static bool cs35l45_readable_reg(struct device *dev, unsigned int reg) case CS35L45_INTB_GPIO2_MCLK_REF: case CS35L45_GPIO3: case CS35L45_PWRMGT_CTL: + case CS35L45_WAKESRC_CTL: + case CS35L45_WKI2C_CTL: + case CS35L45_PWRMGT_STS: case CS35L45_REFCLK_INPUT: case CS35L45_GLOBAL_SAMPLE_RATE: case CS35L45_ASP_ENABLES1: @@ -210,6 +215,7 @@ static bool cs35l45_volatile_reg(struct device *dev, unsigned int reg) case CS35L45_GLOBAL_ENABLES: case CS35L45_ERROR_RELEASE: case CS35L45_AMP_PCM_HPF_TST: /* not cachable */ + case CS35L45_PWRMGT_STS: case CS35L45_IRQ1_STATUS: case CS35L45_IRQ1_EINT_1 ... CS35L45_IRQ1_EINT_18: case CS35L45_IRQ1_STS_1 ... CS35L45_IRQ1_STS_18: diff --git a/sound/soc/codecs/cs35l45.c b/sound/soc/codecs/cs35l45.c index c5de2c8a9fe3..a2bab613daa2 100644 --- a/sound/soc/codecs/cs35l45.c +++ b/sound/soc/codecs/cs35l45.c @@ -36,6 +36,8 @@ static bool cs35l45_check_cspl_mbox_sts(const enum cs35l45_cspl_mboxcmd cmd, return (sts == CSPL_MBOX_STS_RUNNING); case CSPL_MBOX_CMD_STOP_PRE_REINIT: return (sts == CSPL_MBOX_STS_RDY_FOR_REINIT); + case CSPL_MBOX_CMD_HIBERNATE: + return (sts == CSPL_MBOX_STS_HIBERNATE); default: return false; } @@ -745,11 +747,81 @@ static const struct snd_soc_component_driver cs35l45_component = { .endianness = 1, }; +static void cs35l45_setup_hibernate(struct cs35l45_private *cs35l45) +{ + unsigned int wksrc; + + if (cs35l45->bus_type == CONTROL_BUS_I2C) + wksrc = CS35L45_WKSRC_I2C; + else + wksrc = CS35L45_WKSRC_SPI; + + regmap_update_bits(cs35l45->regmap, CS35L45_WAKESRC_CTL, + CS35L45_WKSRC_EN_MASK, + wksrc << CS35L45_WKSRC_EN_SHIFT); + + regmap_set_bits(cs35l45->regmap, CS35L45_WAKESRC_CTL, + CS35L45_UPDT_WKCTL_MASK); + + regmap_update_bits(cs35l45->regmap, CS35L45_WKI2C_CTL, + CS35L45_WKI2C_ADDR_MASK, cs35l45->i2c_addr); + + regmap_set_bits(cs35l45->regmap, CS35L45_WKI2C_CTL, + CS35L45_UPDT_WKI2C_MASK); +} + +static int cs35l45_enter_hibernate(struct cs35l45_private *cs35l45) +{ + dev_dbg(cs35l45->dev, "Enter hibernate\n"); + + cs35l45_setup_hibernate(cs35l45); + + // Don't wait for ACK since bus activity would wake the device + regmap_write(cs35l45->regmap, CS35L45_DSP_VIRT1_MBOX_1, CSPL_MBOX_CMD_HIBERNATE); + + return 0; +} + +static int cs35l45_exit_hibernate(struct cs35l45_private *cs35l45) +{ + const int wake_retries = 20; + const int sleep_retries = 5; + int ret, i, j; + + for (i = 0; i < sleep_retries; i++) { + dev_dbg(cs35l45->dev, "Exit hibernate\n"); + + for (j = 0; j < wake_retries; j++) { + ret = cs35l45_set_cspl_mbox_cmd(cs35l45, cs35l45->regmap, + CSPL_MBOX_CMD_OUT_OF_HIBERNATE); + if (!ret) { + dev_dbg(cs35l45->dev, "Wake success at cycle: %d\n", j); + return 0; + } + usleep_range(100, 200); + } + + dev_err(cs35l45->dev, "Wake failed, re-enter hibernate: %d\n", ret); + + cs35l45_setup_hibernate(cs35l45); + } + + dev_err(cs35l45->dev, "Timed out waking device\n"); + + return -ETIMEDOUT; +} + static int __maybe_unused cs35l45_runtime_suspend(struct device *dev) { struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); + if (!cs35l45->dsp.preloaded || !cs35l45->dsp.cs_dsp.running) + return 0; + + cs35l45_enter_hibernate(cs35l45); + regcache_cache_only(cs35l45->regmap, true); + regcache_mark_dirty(cs35l45->regmap); dev_dbg(cs35l45->dev, "Runtime suspended\n"); @@ -761,9 +833,17 @@ static int __maybe_unused cs35l45_runtime_resume(struct device *dev) struct cs35l45_private *cs35l45 = dev_get_drvdata(dev); int ret; + if (!cs35l45->dsp.preloaded || !cs35l45->dsp.cs_dsp.running) + return 0; + dev_dbg(cs35l45->dev, "Runtime resume\n"); regcache_cache_only(cs35l45->regmap, false); + + ret = cs35l45_exit_hibernate(cs35l45); + if (ret) + return ret; + ret = regcache_sync(cs35l45->regmap); if (ret != 0) dev_warn(cs35l45->dev, "regcache_sync failed: %d\n", ret); diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index 87032619b341..0da28439f628 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -30,6 +30,9 @@ #define CS35L45_INTB_GPIO2_MCLK_REF 0x00002434 #define CS35L45_GPIO3 0x00002438 #define CS35L45_PWRMGT_CTL 0x00002900 +#define CS35L45_WAKESRC_CTL 0x00002904 +#define CS35L45_WKI2C_CTL 0x00002908 +#define CS35L45_PWRMGT_STS 0x0000290C #define CS35L45_REFCLK_INPUT 0x00002C04 #define CS35L45_GLOBAL_SAMPLE_RATE 0x00002C0C #define CS35L45_BOOST_CCM_CFG 0x00003808 @@ -348,6 +351,25 @@ #define CS35L45_POST_GLOBAL_EN_US 5000 #define CS35L45_PRE_GLOBAL_DIS_US 3000 +/* WAKESRC_CTL */ +#define CS35L45_WKSRC_SYNC_GPIO1 BIT(0) +#define CS35L45_WKSRC_INT_GPIO2 BIT(1) +#define CS35L45_WKSRC_GPIO3 BIT(2) +#define CS35L45_WKSRC_SPI BIT(3) +#define CS35L45_WKSRC_I2C BIT(4) +#define CS35L45_UPDT_WKCTL_SHIFT 15 +#define CS35L45_UPDT_WKCTL_MASK BIT(15) +#define CS35L45_WKSRC_EN_SHIFT 8 +#define CS35L45_WKSRC_EN_MASK GENMASK(12, 8) +#define CS35L45_WKSRC_POL_SHIFT 0 +#define CS35L45_WKSRC_POL_MASK GENMASK(3, 0) + +/* WAKEI2C_CTL */ +#define CS35L45_UPDT_WKI2C_SHIFT 15 +#define CS35L45_UPDT_WKI2C_MASK BIT(15) +#define CS35L45_WKI2C_ADDR_SHIFT 0 +#define CS35L45_WKI2C_ADDR_MASK GENMASK(6, 0) + #define CS35L45_SPI_MAX_FREQ 4000000 enum cs35l45_cspl_mboxstate { @@ -369,6 +391,11 @@ enum cs35l45_cspl_mboxcmd { CSPL_MBOX_CMD_INVALID_SEQUENCE = -2, }; +enum control_bus_type { + CONTROL_BUS_I2C = 0, + CONTROL_BUS_SPI = 1, +}; + #define CS35L45_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_3LE| \ SNDRV_PCM_FMTBIT_S24_LE) @@ -439,6 +466,8 @@ struct cs35l45_private { u8 slot_count; int irq_invert; int irq; + unsigned int i2c_addr; + enum control_bus_type bus_type; struct regmap_irq_chip_data *irq_data; }; -- 2.25.1

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ALC4080 on MSI PRO X670-P WIFI
by GitHub issues - opened 02 Mar '23

02 Mar '23

alsa-project/alsa-ucm-conf issue #285 was opened from Onyx47: Hello! I have the board from the title. Out of the box, the audio does not work at all, however the following change to `alsa/ucm2/USB-Audio/USB-Audio.conf` gets the rear panel working: ``` Regex "USB((0414:a00e)|(0b05:(1996|1a(16|2[07])))|(0db0(005a|151f|1feb|419c|82c7|a073|a47c|b202|d6e7|3130)))" ``` (USB id of the card is `0db0:3130`, just added `3130` to the regex). No luck on the front panel however, tried following some discussions and figuring it out to make a complete patch on my own but it's way over my head :) Here is my `alsa-info` output: http://alsa-project.org/db/?f=22aed0d76705b354a59364e6e8e41a4fde20e4da Let me know if there is any other info I need to provide. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/285 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH V2] ucm2: Rockchip: rk817: Add ALSA UCM support
by Chris Morgan 02 Mar '23

02 Mar '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add ALSA-UCM support for the Rockchip RK817 audio codec. This codec is typically configured either with an internal or external amplifier as reflected by the longname. This configuration has been tested on the Anbernic RG353P (rk817_ext) and the Odroid Go Advance (rk817_int). Changes from V1: - Use a constant to control if the output mux is set to HP or SPK. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- ucm2/Rockchip/rk817-sound/HiFi.conf | 68 ++++++++++++++++++++++ ucm2/Rockchip/rk817-sound/rk817-sound.conf | 6 ++ ucm2/conf.d/simple-card/rk817_ext.conf | 1 + ucm2/conf.d/simple-card/rk817_int.conf | 1 + 4 files changed, 76 insertions(+) create mode 100644 ucm2/Rockchip/rk817-sound/HiFi.conf create mode 100644 ucm2/Rockchip/rk817-sound/rk817-sound.conf create mode 120000 ucm2/conf.d/simple-card/rk817_ext.conf create mode 120000 ucm2/conf.d/simple-card/rk817_int.conf diff --git a/ucm2/Rockchip/rk817-sound/HiFi.conf b/ucm2/Rockchip/rk817-sound/HiFi.conf new file mode 100644 index 0000000..8d90567 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/HiFi.conf @@ -0,0 +1,68 @@ +Define.pbk_mux "SPK" + +If.1 { + Condition { + Type ControlExists + Control "name='Internal Speakers Switch'" + } + + True { + Define.pbk_mux "HP" + SectionDevice."Speaker".EnableSequence [ + cset "name='Internal Speakers Switch' on" + ] + + SectionDevice."Speaker".DisableSequence [ + cset "name='Internal Speakers Switch' off" + ] + } +} + +SectionDevice."Speaker" { + Comment "Internal Speaker" + + EnableSequence [ + cset "name='Playback Mux' ${var:pbk_mux}" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + CaptureMixerElem "Mic Capture Gain" + CaptureMasterElem "Master Capture Volume" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='Playback Mux' HP" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphones Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] +} diff --git a/ucm2/Rockchip/rk817-sound/rk817-sound.conf b/ucm2/Rockchip/rk817-sound/rk817-sound.conf new file mode 100644 index 0000000..90ec794 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/rk817-sound.conf @@ -0,0 +1,6 @@ +Syntax 4 + +SectionUseCase."HiFi" { + File "/Rockchip/rk817-sound/HiFi.conf" + Comment "Play HiFi quality music" +} diff --git a/ucm2/conf.d/simple-card/rk817_ext.conf b/ucm2/conf.d/simple-card/rk817_ext.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_ext.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file diff --git a/ucm2/conf.d/simple-card/rk817_int.conf b/ucm2/conf.d/simple-card/rk817_int.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_int.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file -- 2.34.1

2 1

X13s
by GitHub pull_request - opened 02 Mar '23

02 Mar '23

alsa-project/alsa-ucm-conf pull request #284 was opened from Srinivas-Kandagatla: This patchset adds support for X13s ucm which includes support for Speakers, 2 DMICs on Panel and Headset with Mic. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/284 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/284.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Question: is it possible to subscribe a port to system announcer ?
by GitHub issues - opened 01 Mar '23

01 Mar '23

alsa-project/alsa-python issue #7 was opened from Houston4444: Hi. I try to add ALSA support to [Patchance](https://github.com/Houston4444/Patchance). I 'd like to create a port capable to receive events as SEQ_EVENT_PORT_START or SEQ_EVENT_PORT_EXIT in order to get a complete graph. I see in patchage source code (C++) that it uses 'snd_seq_connect_from' function to enable this events. Is it possible in python with this lib ? Thanks. Issue URL : https://github.com/alsa-project/alsa-python/issues/7 Repository URL: https://github.com/alsa-project/alsa-python

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[PATCH v2 0/2] ASoC: mediatek: add missing initialization
by Trevor Wu 01 Mar '23

01 Mar '23

Fix dts parsing problem in mt8188 and mt8195 etdm dai drivers. Changes since v1: - split the patch into two commits and add Fixes tag. Trevor Wu (2): ASoC: mediatek: mt8188: add missing initialization ASoC: mediatek: mt8195: add missing initialization sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 3 +++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 3 +++ 2 files changed, 6 insertions(+) -- 2.18.0

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[PATCH] ASoC: mediatek: add missing initialization
by Trevor Wu 01 Mar '23

01 Mar '23

In etdm dai driver, dai_etdm_parse_of() function is used to parse dts properties to get parameters. There are two for-loops which are sepearately for all etdm and etdm input only cases. In etdm in only loop, dai_id is not initialized, so it keeps the value intiliazed in another loop. In the patch, add the missing initialization to fix the unexpected parsing problem. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-dai-etdm.c | 3 +++ sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 3 +++ 2 files changed, 6 insertions(+) diff --git a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c index 1c53d4cb19bb..7a37752d4244 100644 --- a/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c +++ b/sound/soc/mediatek/mt8188/mt8188-dai-etdm.c @@ -2498,6 +2498,9 @@ static void mt8188_dai_etdm_parse_of(struct mtk_base_afe *afe) /* etdm in only */ for (i = 0; i < 2; i++) { + dai_id = ETDM_TO_DAI_ID(i); + etdm_data = afe_priv->dai_priv[dai_id]; + snprintf(prop, sizeof(prop), "mediatek,%s-chn-disabled", of_afe_etdms[i].name); diff --git a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c index c2e268054773..f2c9a1fdbe0d 100644 --- a/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c +++ b/sound/soc/mediatek/mt8195/mt8195-dai-etdm.c @@ -2567,6 +2567,9 @@ static void mt8195_dai_etdm_parse_of(struct mtk_base_afe *afe) /* etdm in only */ for (i = 0; i < 2; i++) { + dai_id = ETDM_TO_DAI_ID(i); + etdm_data = afe_priv->dai_priv[dai_id]; + ret = snprintf(prop, sizeof(prop), "mediatek,%s-chn-disabled", of_afe_etdms[i].name); -- 2.18.0

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[Patch] sound/aoa: Add support for iBook G4 (powerbook6,5)
by Thaison Nguyen 01 Mar '23

01 Mar '23

The audio on my iBook G4 (powerbook65) has not been working. This is because the ids of the device were missing in the aoa audio driver. With this patch, the IDs of Powerbook6,4 and Powerbook6,5 are added to the driver. Signed-off-by: Thaison Nguyen <mail(a)thaison-nguyen.de> --- sound/aoa/fabrics/layout.c | 15 +++++++++++++++ sound/aoa/soundbus/i2sbus/core.c | 2 +- 2 files changed, 16 insertions(+), 1 deletions(-) diff a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -111,7 +111,9 @@ MODULE_ALIAS("aoa-device-id-14"); MODULE_ALIAS("aoa-device-id-22"); MODULE_ALIAS("aoa-device-id-31"); MODULE_ALIAS("aoa-device-id-35"); +MODULE_ALIAS("aoa-device-id-38"); +MODULE_ALIAS("aoa-device-id-40"); MODULE_ALIAS("aoa-device-id-44"); /* onyx with all but microphone connected */ @@ -368,7 +368,20 @@ .connections = tas_connections_nolineout, }, }, + /* PowerBook6,4 */ + { .device_id = 40, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, /* PowerBook6,5 */ + { .device_id = 38, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, { .device_id = 44, .codecs[0] = { .name = "tas", diff a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -197,7 +197,7 @@ * so restrict to those we do handle for now. */ if (id && (*id == 22 || *id == 14 || *id == 35 || - *id == 31 || *id == 44)) { + *id == 31 || *id == 44 || *id == 40 || *id == 38)) { snprintf(dev->sound.modalias, 32, "aoa-device-id-%d", *id); ok = 1;

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Re: Patch for audio on iBook G4 (powerbook6,5)
by Christophe Leroy 01 Mar '23

01 Mar '23

Le 28/02/2023 à 22:17, mail(a)thaison-nguyen.de a écrit : > Hi > > I have added a bug report and a patch regarding audio for iBook G4 > (powerbook6,5) on kernel.org bugzilla, but nobody has picked it up. > https://bugzilla.kernel.org/show_bug.cgi?id=216504 > > Now I am trying to submit the patch here. > Please read and follow the process to submit a patch : https://docs.kernel.org/process/5.Posting.html Patches must be in the core part of the mail and not as an attached file. Patches must have a description of the change. Patches must be signed Patches must be sent to the relevant maintainer(s) based on the modified files. As far as I can see your patch is for the sound subsystem. Christophe

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[PATCH v3 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 28 Feb '23

28 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, component DAI driver sets its struct snd_soc_component_driver::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. Thank you, Claudiu Beznea Changes in v3: - update the commit message of patch 2/3 to be more descriptive - in patch 3/3 initialize dd->startup_delay_us before calling of_property_read_u32() and don't check its return value; property is optional and the the default value is 150ms Changes in v2: - patch 1/3 from v1 is now "ASoC: soc-pcm: add option to start DMA after DAI" - pass start_dma_last from component DAI driver object (struct snd_soc_component_driver::start_dma_last); adapt patch 3/3 after this; - in patch 1/3 s/Do we need to start dma first/Do we need to start dma last in comment from soc_pcm_trigger() - collect review tag from Krzysztof Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 +++ include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 53 +++++++++++++++++-- sound/soc/soc-pcm.c | 27 ++++++++-- 4 files changed, 78 insertions(+), 10 deletions(-) -- 2.34.1

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