February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: uniphier: evea: fix typo 'eva' -> 'evea'
by Katsuhiro Suzuki 20 Feb '18

20 Feb '18

Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index cad7e60de21d..ac2d30b55fb6 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -292,7 +292,7 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } -static const struct snd_kcontrol_new eva_controls[] = { +static const struct snd_kcontrol_new evea_controls[] = { SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, @@ -384,8 +384,8 @@ static struct snd_soc_component_driver soc_codec_evea = { .num_dapm_widgets = ARRAY_SIZE(evea_widgets), .dapm_routes = evea_routes, .num_dapm_routes = ARRAY_SIZE(evea_routes), - .controls = eva_controls, - .num_controls = ARRAY_SIZE(eva_controls), + .controls = evea_controls, + .num_controls = ARRAY_SIZE(evea_controls), .idle_bias_on = 1, .use_pmdown_time = 1, .endianness = 1, -- 2.16.1

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[alsa-devel] Need help implementing an external PCM/control plugin pair
by Dimitris Papavasiliou 20 Feb '18

20 Feb '18

Hi, I'm in the process of putting together a loudness compensation plugin for ALSA. The filtering part is implemented as an external PCM plugin and it is, more or less, ready, meaning that you can set its parameters via configuration options and it'll happily filter away in real time. I'm trying now to create a matching mixer plugin, to allow on-the-fly adjustment of the parameters, and I'm not sure what the best way of accomplishing that would be. My main concerns are: 1. I've figured out how to create a basic external control plugin, that would allow me to expose a set of controls, which the user can fiddle with, using alsamixer et al., but what would be the best way of communicating changes in the values to the PCM plugin? I could imagine using methods external to ALSA, such as a file with stored values, which the ctl plugin updates and which the PCM plugin monitors for changes, or perhaps I could try to open the mixer from inside the PCM plugin, set callbacks for each of its elements and enter a snd_mixer_wait/snd_mixer_handle_events loop, in a separate thread of course. Is there reason to prefer one way over the other? Perhaps there is a better way, I'm not aware of? 2. I'd like to support event subscription/notification, but I'm not sure how to make that work. My understanding (from searching the net and looking at ALSA's source), is that I need to provide some sort of file descriptor, which the subscribing code can poll, so as to block until there's an update. Since I have no obvious candidates for such a file, I've tried creating a pipe, handing one end to the control plugin upon initialization and then writing single bytes to the other end when there's a change in one of the controls. This works but, as far as I can see, the pipe is never actually read from, from the subscribing code, so that I would have to manually read from it somehwere, say in the read_event callback, in order to flush it. In the end, I'll probably use the descriptor from an inotify watch on the file where the plugin stores the control settings, but is this a valid approach overall? Is there some documentation specifying the exact way in which poll descriptors are handled by ALSA (in the case of an extrnal CTL plugin), that I've missed? If not, are my assumptions above (that the descriptor is only polled and that its contents are never actually read) correct? 3. Finally, and although this is not essential, I'd like to provide metadata for some controls, to support dB ranges, as that's the most natural way to set them. I see hints that this is supported for external control plugins (SND_CTL_EXT_ACCESS_TLV_READ/WRITE, the snd_ctl_ext_tlv_rw_t callback), but I'm not sure how to implement it. Is it documented somewhere? If not, could someone describe the basic approach? Thanks, Dimitris

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[alsa-devel] [PATCH v4 0/2] Add support for AK5558 ADC
by Daniel Baluta 19 Feb '18

19 Feb '18

We support normal mode, TDM mode and pm. Changes since v3: [addressed Mark's comments] * use C++ comments for SPDX identifier * removed some unused widgets and controls (channel enable MUXes, TDM, DSD related stuff) * removed set_dai_mute, set_bias_level, set_dai_sysclk as they are not used. * refactored probe/remove to have symmetric operations * use snd_soc_update_bits instead of hardcode read/write operation. Changes since v2: [addressed comments from Andy, Fabio and Rob] * sort include files * use probe_new * reword the binding document * use adc@10 instead of ak5558@10 * remove file name at the beginning of codec source code. * make i2c_probe and i2c_remove parameters naming consistent. Changes since v1: [addressed comments from Andy and Fabio] * fix GPIO polarity from active high to active low for correct documentation * fix license header by using SPDX identifier * remove debug prints at the beginning of functions. * only support auto clock switching (manual switching was dead code anyway) (in the future we could add a DT property to choose between manual and auto) * Use gpiod API * use GENMASK * introduce power_off/power_on Daniel Baluta (2): ASoC: codecs: Add support for AK5558 ADC driver ASoC: ak5558: Add bindings for AK5558 ADC Documentation/devicetree/bindings/sound/ak5558.txt | 22 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5558.c | 417 +++++++++++++++++++++ sound/soc/codecs/ak5558.h | 52 +++ 5 files changed, 499 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5558.txt create mode 100644 sound/soc/codecs/ak5558.c create mode 100644 sound/soc/codecs/ak5558.h -- 2.7.4

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[alsa-devel] [PATCH] ASoC: ADAU7002: Adding ACPI id
by Akshu Agrawal 19 Feb '18

19 Feb '18

Adding ACPI entry id for probing the driver. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- sound/soc/codecs/adau7002.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/adau7002.c b/sound/soc/codecs/adau7002.c index 49475e3..fdff868 100644 --- a/sound/soc/codecs/adau7002.c +++ b/sound/soc/codecs/adau7002.c @@ -7,6 +7,7 @@ * Licensed under the GPL-2. */ +#include <linux/acpi.h> #include <linux/init.h> #include <linux/module.h> #include <linux/of.h> @@ -69,10 +70,19 @@ static int adau7002_remove(struct platform_device *pdev) MODULE_DEVICE_TABLE(of, adau7002_dt_ids); #endif +#ifdef CONFIG_ACPI +static const struct acpi_device_id adau7002_acpi_match[] = { + { "ADAU7002", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, adau7002_acpi_match); +#endif + static struct platform_driver adau7002_driver = { .driver = { .name = "adau7002", .of_match_table = of_match_ptr(adau7002_dt_ids), + .acpi_match_table = ACPI_PTR(adau7002_acpi_match), }, .probe = adau7002_probe, .remove = adau7002_remove, -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC: tlv320aic31xx: Optimize the master mode I2S clock handling
by Peter Ujfalusi 19 Feb '18

19 Feb '18

Hi, Instead of forcing DAC/ADC to be turned on, we can use P0 R29, Bit2 to let the I2S clocks running, but we do not want it to run all the time and we do not want I2S clocks when it is not needed (analog loopback mode). Use DAPM_SUPPLY for the bit and connect it to the in/out AIF. The DAPM routes can be modify to not include the DAC/ADC. Regards, Peter --- Peter Ujfalusi (2): ASoC: tlv320aic31xx: Rename AIF_IN from 'DAC IN' to 'AIF IN' ASoC: tlv320aic31xx: Do not force power on the DAC/ADC in clock master mode sound/soc/codecs/tlv320aic31xx.c | 46 +++++++++++++++++++++++++--------------- sound/soc/codecs/tlv320aic31xx.h | 1 + 2 files changed, 30 insertions(+), 17 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH 0/4] ASoC: missing replace codec to component
by Kuninori Morimoto 19 Feb '18

19 Feb '18

Hi Mark These are new added driver on mark/for-next branch, but based on old "codec" framework. We want to convert it to component. These patches are do it Kuninori Morimoto (4): ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/codecs/ak4458.c | 105 +++++++++++++------------- sound/soc/codecs/ak5558.c | 65 ++++++++-------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 12 +-- 4 files changed, 94 insertions(+), 92 deletions(-) -- 1.9.1 Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH 0/4 v2] ASoC: missing replace codec to component
by Kuninori Morimoto 19 Feb '18

19 Feb '18

Hi Mark These are v2 of replace patch for new added driver on mark/for-next branch. The diff v1 -> v2 is that compile warning issue was solved on ak driver Kuninori Morimoto (4): ASoC: intel: kbl_da7219_max98357: replace codec to component ASoC: amd: acp-da7219-max98357: replace codec to component ASoC: ak4458: replace codec to component ASoC: ak5558: replace codec to component sound/soc/amd/acp-da7219-max98357a.c | 4 +- sound/soc/codecs/ak4458.c | 107 +++++++++++++------------- sound/soc/codecs/ak5558.c | 67 ++++++++-------- sound/soc/intel/boards/kbl_da7219_max98357a.c | 12 +-- 4 files changed, 94 insertions(+), 96 deletions(-) -- 1.9.1

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[alsa-devel] Applied "ASoC: sgtl5000: add 'Audio Switch' block implementation" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: sgtl5000: add 'Audio Switch' block implementation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 >From 7f7d5239f56cfd25ba8470354d9705815c079157 Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Mon, 19 Feb 2018 10:03:45 +0100 Subject: [PATCH] ASoC: sgtl5000: add 'Audio Switch' block implementation Add mux for choosing DAC source: ADC, I2S or DAP. DAP - digital audio processing block. Moreover create choosing mux for DAP main source (ADC or I2S) and DAP mixer source (ADC or I2S). Mixer block allows summing DAP main and mixer sources togheter with defined volumes. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 78 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 67 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 05592ddcf0e6..1724c4622052 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -216,17 +216,54 @@ static SOC_ENUM_SINGLE_DECL(adc_enum, static const struct snd_kcontrol_new adc_mux = SOC_DAPM_ENUM("Capture Mux", adc_enum); +/* input sources for headphone */ +static const char *hp_mux_text[] = { + "DAC", "LINE_IN" +}; + +static SOC_ENUM_SINGLE_DECL(hp_enum, + SGTL5000_CHIP_ANA_CTRL, 6, + hp_mux_text); + +static const struct snd_kcontrol_new hp_mux = +SOC_DAPM_ENUM("Headphone Mux", hp_enum); + /* input sources for DAC */ static const char *dac_mux_text[] = { - "DAC", "LINE_IN" + "ADC", "I2S", "Rsvrd", "DAP" }; static SOC_ENUM_SINGLE_DECL(dac_enum, - SGTL5000_CHIP_ANA_CTRL, 6, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAC_SEL_SHIFT, dac_mux_text); static const struct snd_kcontrol_new dac_mux = -SOC_DAPM_ENUM("Headphone Mux", dac_enum); +SOC_DAPM_ENUM("Digital Input Mux", dac_enum); + +/* input sources for DAP */ +static const char *dap_mux_text[] = { + "ADC", "I2S" +}; + +static SOC_ENUM_SINGLE_DECL(dap_enum, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAP_SEL_SHIFT, + dap_mux_text); + +static const struct snd_kcontrol_new dap_mux = +SOC_DAPM_ENUM("DAP Mux", dap_enum); + +/* input sources for DAP mix */ +static const char *dapmix_mux_text[] = { + "ADC", "I2S" +}; + +static SOC_ENUM_SINGLE_DECL(dapmix_enum, + SGTL5000_CHIP_SSS_CTRL, SGTL5000_DAP_MIX_SEL_SHIFT, + dapmix_mux_text); + +static const struct snd_kcontrol_new dapmix_mux = +SOC_DAPM_ENUM("DAP MIX Mux", dapmix_enum); + static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LINE_IN"), @@ -243,7 +280,12 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { SND_SOC_DAPM_PGA("LO", SGTL5000_CHIP_ANA_POWER, 0, 0, NULL, 0), SND_SOC_DAPM_MUX("Capture Mux", SND_SOC_NOPM, 0, 0, &adc_mux), - SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + SND_SOC_DAPM_MUX("Headphone Mux", SND_SOC_NOPM, 0, 0, &hp_mux), + SND_SOC_DAPM_MUX("Digital Input Mux", SND_SOC_NOPM, 0, 0, &dac_mux), + SND_SOC_DAPM_MUX("DAP Mux", SGTL5000_DAP_CTRL, 0, 0, &dap_mux), + SND_SOC_DAPM_MUX("DAP MIX Mux", SGTL5000_DAP_CTRL, 4, 0, &dapmix_mux), + SND_SOC_DAPM_MIXER("DAP", SGTL5000_CHIP_DIG_POWER, 4, 0, NULL, 0), + /* aif for i2s input */ SND_SOC_DAPM_AIF_IN("AIFIN", "Playback", @@ -270,7 +312,19 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ + {"DAP Mux", "ADC", "ADC"}, /* adc --> DAP mux */ + {"DAP Mux", NULL, "AIFIN"}, /* i2s --> DAP mux */ + {"DAP", NULL, "DAP Mux"}, /* DAP mux --> dap */ + + {"DAP MIX Mux", "ADC", "ADC"}, /* adc --> DAP MIX mux */ + {"DAP MIX Mux", NULL, "AIFIN"}, /* i2s --> DAP MIX mux */ + {"DAP", NULL, "DAP MIX Mux"}, /* DAP MIX mux --> dap */ + + {"Digital Input Mux", "ADC", "ADC"}, /* adc --> audio mux */ + {"Digital Input Mux", NULL, "AIFIN"}, /* i2s --> audio mux */ + {"Digital Input Mux", NULL, "DAP"}, /* dap --> audio mux */ + {"DAC", NULL, "Digital Input Mux"}, /* audio mux --> dac */ + {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ @@ -463,6 +517,9 @@ static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, 1, 3, TLV_DB_SCALE_ITEM(2000, 1000, 0) ); +/* tlv for DAP channels, 0% - 100% - 200% */ +static const DECLARE_TLV_DB_SCALE(dap_volume, 0, 1, 0); + /* tlv for hp volume, -51.5db to 12.0db, step .5db */ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); @@ -514,6 +571,11 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { lineout_volume), SOC_SINGLE("Lineout Playback Switch", SGTL5000_CHIP_ANA_CTRL, 8, 1, 1), + SOC_SINGLE_TLV("DAP Main channel", SGTL5000_DAP_MAIN_CHAN, + 0, 0xffff, 0, dap_volume), + + SOC_SINGLE_TLV("DAP Mix channel", SGTL5000_DAP_MIX_CHAN, + 0, 0xffff, 0, dap_volume), /* Automatic Volume Control (DAP AVC) */ SOC_SINGLE("AVC Switch", SGTL5000_DAP_AVC_CTRL, 0, 1, 0), SOC_SINGLE("AVC Hard Limiter Switch", SGTL5000_DAP_AVC_CTRL, 5, 1, 0), @@ -1217,12 +1279,6 @@ static int sgtl5000_probe(struct snd_soc_component *component) /* disable short cut detector */ snd_soc_component_write(component, SGTL5000_CHIP_SHORT_CTRL, 0); - /* - * set i2s as default input of sound switch - * TODO: add sound switch to control and dapm widge. - */ - snd_soc_component_write(component, SGTL5000_CHIP_SSS_CTRL, - SGTL5000_DAC_SEL_I2S_IN << SGTL5000_DAC_SEL_SHIFT); snd_soc_component_write(component, SGTL5000_CHIP_DIG_POWER, SGTL5000_ADC_EN | SGTL5000_DAC_EN); -- 2.16.1

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[alsa-devel] Applied "ASoC: sgtl5000: add 5 band graphic equalizer" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: sgtl5000: add 5 band graphic equalizer has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 >From 92a2742f9813ef65da19145a77e8067d6abbc822 Mon Sep 17 00:00:00 2001 From: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Date: Mon, 19 Feb 2018 10:03:46 +0100 Subject: [PATCH] ASoC: sgtl5000: add 5 band graphic equalizer Enable 5-band graphic equalizer which allows manipulation of gain in range -11.75dB to 12dB for 5 specific sound bands: 115Hz, 330Hz, 990Hz, 3000Hz, 9900Hz. Signed-off-by: Michal Oleszczyk <oleszczyk.m(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sgtl5000.c | 24 +++++++++++++++++++++--- sound/soc/codecs/sgtl5000.h | 7 +++++++ 2 files changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1724c4622052..7c1d65830c05 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -520,6 +520,9 @@ static const DECLARE_TLV_DB_RANGE(mic_gain_tlv, /* tlv for DAP channels, 0% - 100% - 200% */ static const DECLARE_TLV_DB_SCALE(dap_volume, 0, 1, 0); +/* tlv for bass bands, -11.75db to 12.0db, step .25db */ +static const DECLARE_TLV_DB_SCALE(bass_band, -1175, 25, 0); + /* tlv for hp volume, -51.5db to 12.0db, step .5db */ static const DECLARE_TLV_DB_SCALE(headphone_volume, -5150, 50, 0); @@ -585,6 +588,21 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SOC_SINGLE_EXT_TLV("AVC Threshold Volume", SGTL5000_DAP_AVC_THRESHOLD, 0, 96, 0, avc_get_threshold, avc_put_threshold, avc_threshold), + + SOC_SINGLE_TLV("BASS 0", SGTL5000_DAP_EQ_BASS_BAND0, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 1", SGTL5000_DAP_EQ_BASS_BAND1, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 2", SGTL5000_DAP_EQ_BASS_BAND2, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 3", SGTL5000_DAP_EQ_BASS_BAND3, + 0, 0x5F, 0, bass_band), + + SOC_SINGLE_TLV("BASS 4", SGTL5000_DAP_EQ_BASS_BAND4, + 0, 0x5F, 0, bass_band), }; /* mute the codec used by alsa core */ @@ -1303,11 +1321,11 @@ static int sgtl5000_probe(struct snd_soc_component *component) SGTL5000_BIAS_VOLT_MASK, sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* - * disable DAP + * enable DAP Graphic EQ * TODO: - * Enable DAP in kcontrol and dapm. + * Add control for changing between PEQ/Tone Control/GEQ */ - snd_soc_component_write(component, SGTL5000_DAP_CTRL, 0); + snd_soc_component_write(component, SGTL5000_DAP_AUDIO_EQ, SGTL5000_DAP_SEL_GEQ); /* Unmute DAC after start */ snd_soc_component_update_bits(component, SGTL5000_CHIP_ADCDAC_CTRL, diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 22f3442af982..28cf637155bb 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -399,4 +399,11 @@ #define SGTL5000_SYSCLK 0x00 #define SGTL5000_LRCLK 0x01 +/* + * SGTL5000_DAP_AUDIO_EQ + */ +#define SGTL5000_DAP_SEL_PEQ 1 +#define SGTL5000_DAP_SEL_TONE_CTRL 2 +#define SGTL5000_DAP_SEL_GEQ 3 + #endif -- 2.16.1

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[alsa-devel] Applied "ASoC: rt5651: Fix regcache sync errors on resume" to the asoc tree
by Mark Brown 19 Feb '18

19 Feb '18

The patch ASoC: rt5651: Fix regcache sync errors on resume has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git 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 >From 2d30e9494f1ea320aaaad0cff9ddd92c87eac355 Mon Sep 17 00:00:00 2001 From: Hans de Goede <hdegoede(a)redhat.com> Date: Sun, 18 Feb 2018 23:01:44 +0100 Subject: [PATCH] ASoC: rt5651: Fix regcache sync errors on resume The ALC5651 does not like multi-write accesses, avoid them. This fixes: rt5651 i2c-10EC5651:00: Unable to sync registers 0x27-0x28. -121 Errors on resume (and all registers after the registers in the error not being synced). Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/rt5651.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 831b297978a4..45a73049cf64 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1722,6 +1722,7 @@ static const struct regmap_config rt5651_regmap = { .num_reg_defaults = ARRAY_SIZE(rt5651_reg), .ranges = rt5651_ranges, .num_ranges = ARRAY_SIZE(rt5651_ranges), + .use_single_rw = true, }; #if defined(CONFIG_OF) -- 2.16.1

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