July 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/6] ASoC: Add support to WCD9340/WCD9341 codec
by Srinivas Kandagatla 23 Jul '19

23 Jul '19

This patchset adds support to Qualcomm WCD9340/WCD9341 Codec which is a standalone Hi-Fi audio codec IC. This codec supports both I2S/I2C and SLIMbus audio interfaces. On slimbus interface it supports two data lanes; 16 Tx ports and 8 Rx ports. It has Five DACs and seven dedicated interpolators, Multibutton headset control (MBHC), Active noise cancellation, Sidetone paths, MAD (mic activity detection) and codec processing engine. It supports Class-H differential earpiece out and stereo single ended headphones out. This codec also has integrated SoundWire controller. Patchset for this is already sent for review at https://lwn.net/Articles/790651/ This patchset has been tested on SDM845 based DragonBoard DB845c with WSA881x smart speaker amplifiers via soundwire and 4 DMICs. Thanks, srini Srinivas Kandagatla (6): ASoC: dt-bindings: add dt bindings for WCD9340/WCD9341 audio codec ASoC: wcd934x: add support to wcd9340/wcd9341 codec ASoC: wcd934x: add basic controls ASoC: wcd934x: add playback dapm widgets ASoC: wcd934x: add capture dapm widgets ASoC: wcd934x: add audio routings .../bindings/sound/qcom,wcd934x.txt | 188 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wcd934x.c | 5979 +++++++++++++++++ sound/soc/codecs/wcd934x.h | 426 ++ 5 files changed, 6605 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd934x.txt create mode 100644 sound/soc/codecs/wcd934x.c create mode 100644 sound/soc/codecs/wcd934x.h -- 2.21.0

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[alsa-devel] [PATCH] ASoC: intel: Add Broadwell rt5650 machine driver
by Jon Flatley 23 Jul '19

23 Jul '19

From: Ben Zhang <benzh(a)chromium.org> Add machine driver for Broadwell + rt5650. Signed-off-by: Jon Flatley <jflat(a)chromium.org> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bdw-rt5650.c | 305 ++++++++++++++++++ .../common/soc-acpi-intel-hsw-bdw-match.c | 5 + 4 files changed, 323 insertions(+) create mode 100644 sound/soc/intel/boards/bdw-rt5650.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 50bf149818b5..c111ae177b4a 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -31,6 +31,17 @@ endif ## SND_SOC_INTEL_HASWELL if SND_SOC_INTEL_HASWELL || SND_SOC_SOF_BROADWELL +config SND_SOC_INTEL_BDW_RT5650_MACH + tristate "Broadwell with RT5650 codec" + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC + select SND_COMPRESS_OFFLOAD + select SND_SOC_RT5645 + help + This adds the ASoC machine driver for Intel Broadwell platforms with + the RT5650 codec. + Say Y if you have such a device + If unsure select "N". + config SND_SOC_INTEL_BDW_RT5677_MACH tristate "Broadwell with RT5677 codec" depends on I2C diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 6445f90ea542..b5e2619607be 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -2,6 +2,7 @@ snd-soc-sst-haswell-objs := haswell.o snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o +snd-soc-sst-bdw-rt5650-mach-objs := bdw-rt5650.o snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_ obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5650_MACH) += snd-soc-sst-bdw-rt5650-mach.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c new file mode 100644 index 000000000000..cb875eeab055 --- /dev/null +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -0,0 +1,305 @@ +// SPDX-License-Identifier: GPL-2.0-only +/* + * ASoC machine driver for Intel Broadwell platforms with RT5650 codec + * + * Copyright 2019, The Chromium OS Authors. All rights reserved. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/gpio/consumer.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> + +#include "../common/sst-dsp.h" +#include "../haswell/sst-haswell-ipc.h" + +#include "../../codecs/rt5645.h" + +struct bdw_rt5650_priv { + struct gpio_desc *gpio_hp_en; + struct snd_soc_component *component; +}; + +static const struct snd_soc_dapm_widget bdw_rt5650_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("DMIC Pair1", NULL), + SND_SOC_DAPM_MIC("DMIC Pair2", NULL), +}; + +static const struct snd_soc_dapm_route bdw_rt5650_map[] = { + /* Speakers */ + {"Speaker", NULL, "SPOL"}, + {"Speaker", NULL, "SPOR"}, + + /* Headset jack connectors */ + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + + /* Digital MICs + * DMIC Pair1 are the two DMICs connected on the DMICN1 connector. + * DMIC Pair2 are the two DMICs connected on the DMICN2 connector. + * Facing the camera, DMIC Pair1 are on the left side, DMIC Pair2 + * are on the right side. + */ + {"DMIC L1", NULL, "DMIC Pair1"}, + {"DMIC R1", NULL, "DMIC Pair1"}, + {"DMIC L2", NULL, "DMIC Pair2"}, + {"DMIC R2", NULL, "DMIC Pair2"}, + + /* CODEC BE connections */ + {"SSP0 CODEC IN", NULL, "AIF1 Capture"}, + {"AIF1 Playback", NULL, "SSP0 CODEC OUT"}, +}; + +static const struct snd_kcontrol_new bdw_rt5650_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("DMIC Pair1"), + SOC_DAPM_PIN_SWITCH("DMIC Pair2"), +}; + + +static struct snd_soc_jack headphone_jack; +static struct snd_soc_jack mic_jack; + +static struct snd_soc_jack_pin headphone_jack_pin = { + .pin = "Headphone", + .mask = SND_JACK_HEADPHONE, +}; + +static struct snd_soc_jack_pin mic_jack_pin = { + .pin = "Headset Mic", + .mask = SND_JACK_MICROPHONE, +}; + +static int broadwell_ssp0_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* The ADSP will covert the FE rate to 48k, max 4-channels */ + rate->min = rate->max = 48000; + channels->min = 2; + channels->max = 4; + + /* set SSP0 to 24 bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Workaround: set codec PLL to 19.2MHz that PLL source is + * from MCLK(24MHz) to conform 2.4MHz DMIC clock. + */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + 24000000, 19200000); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + /* The actual MCLK freq is 24MHz. The codec is told that MCLK is + * 24.576MHz to satisfy the requirement of rl6231_get_clk_info. + * ASRC is enabled on AD and DA filters to ensure good audio quality. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, 24576000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk configuration\n"); + return ret; + } + + return ret; +} + +static struct snd_soc_ops bdw_rt5650_ops = { + .hw_params = bdw_rt5650_hw_params, +}; + +static int bdw_rt5650_rtd_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, DRV_NAME); + struct sst_pdata *pdata = dev_get_platdata(component->dev); + struct sst_hsw *broadwell = pdata->dsp; + int ret; + + /* Set ADSP SSP port settings + * clock_divider = 4 means BCLK = MCLK/5 = 24MHz/5 = 4.8MHz + */ + ret = sst_hsw_device_set_config(broadwell, SST_HSW_DEVICE_SSP_0, + SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, + SST_HSW_DEVICE_TDM_CLOCK_MASTER, 4); + if (ret < 0) { + dev_err(rtd->dev, "error: failed to set device config\n"); + return ret; + } + + return 0; +} + +static int bdw_rt5650_init(struct snd_soc_pcm_runtime *rtd) +{ + struct bdw_rt5650_priv *bdw_rt5650 = + snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* Enable codec ASRC function for Stereo DAC/Stereo1 ADC/DMIC/I2S1. + * The ASRC clock source is clk_i2s1_asrc. + */ + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER | + RT5645_AD_STEREO_FILTER | + RT5645_AD_MONO_L_FILTER | + RT5645_AD_MONO_R_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); + + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* Create and initialize headphone jack */ + if (snd_soc_card_jack_new(rtd->card, "Headphone Jack", + SND_JACK_HEADPHONE, &headphone_jack, + &headphone_jack_pin, 1)) { + dev_err(component->dev, "Can't create headphone jack\n"); + } + + /* Create and initialize mic jack */ + if (snd_soc_card_jack_new(rtd->card, "Mic Jack", SND_JACK_MICROPHONE, + &mic_jack, &mic_jack_pin, 1)) { + dev_err(component->dev, "Can't create mic jack\n"); + } + + rt5645_set_jack_detect(component, &headphone_jack, &mic_jack, NULL); + + bdw_rt5650->component = component; + + return 0; +} + +/* broadwell digital audio interface glue - connects codec <--> CPU */ +SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); +SND_SOC_DAILINK_DEF(fe, DAILINK_COMP_ARRAY(COMP_CPU("System Pin"))); +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("haswell-pcm-audio"))); +SND_SOC_DAILINK_DEF(be, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5650:00", "rt5645-aif1"))); + +static struct snd_soc_dai_link bdw_rt5650_dais[] = { + /* Front End DAI links */ + { + .name = "System PCM", + .stream_name = "System Playback", + .dynamic = 1, + .init = bdw_rt5650_rtd_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + .dpcm_capture = 1, + SND_SOC_DAILINK_REG(fe, dummy, platform), + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "Codec", + .id = 0, + .no_pcm = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = broadwell_ssp0_fixup, + .ops = &bdw_rt5650_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = bdw_rt5650_init, + SND_SOC_DAILINK_REG(dummy, be, dummy) + }, +}; + +/* ASoC machine driver for Broadwell DSP + RT5650 */ +static struct snd_soc_card bdw_rt5650_card = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + .dai_link = bdw_rt5650_dais, + .num_links = ARRAY_SIZE(bdw_rt5650_dais), + .dapm_widgets = bdw_rt5650_widgets, + .num_dapm_widgets = ARRAY_SIZE(bdw_rt5650_widgets), + .dapm_routes = bdw_rt5650_map, + .num_dapm_routes = ARRAY_SIZE(bdw_rt5650_map), + .controls = bdw_rt5650_controls, + .num_controls = ARRAY_SIZE(bdw_rt5650_controls), + .fully_routed = true, +}; + +static int bdw_rt5650_probe(struct platform_device *pdev) +{ + struct bdw_rt5650_priv *bdw_rt5650; + + bdw_rt5650_card.dev = &pdev->dev; + + /* Allocate driver private struct */ + bdw_rt5650 = devm_kzalloc(&pdev->dev, sizeof(struct bdw_rt5650_priv), + GFP_KERNEL); + if (!bdw_rt5650) + return -ENOMEM; + snd_soc_card_set_drvdata(&bdw_rt5650_card, bdw_rt5650); + + return snd_soc_register_card(&bdw_rt5650_card); +} + +static int bdw_rt5650_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bdw_rt5650_card); + return 0; +} + +static struct platform_driver bdw_rt5650_audio = { + .probe = bdw_rt5650_probe, + .remove = bdw_rt5650_remove, + .driver = { + .name = "bdw-rt5650", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bdw_rt5650_audio) + +/* Module information */ +MODULE_AUTHOR("Ben Zhang <benzh(a)chromium.org>"); +MODULE_DESCRIPTION("Intel Broadwell RT5650 machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bdw-rt5650"); diff --git a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c index d27853e7a369..ba3d25658436 100644 --- a/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-hsw-bdw-match.c @@ -29,6 +29,11 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_broadwell_machines[] = { .sof_fw_filename = "sof-bdw.ri", .sof_tplg_filename = "sof-bdw-rt286.tplg", }, + { + .id = "10EC5650", + .drv_name = "bdw-rt5650", + .fw_filename = "intel/IntcSST2.bin", + }, { .id = "RT5677CE", .drv_name = "bdw-rt5677", -- 2.22.0.657.g960e92d24f-goog

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[alsa-devel] [PATCH] ASoC: rt5677: remove superfluous set
by Curtis Malainey 22 Jul '19

22 Jul '19

We define this variable with the same value, no need to set it twice Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index c779dc3474f9e..deaed5132dc95 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5098,7 +5098,6 @@ static irqreturn_t rt5677_irq(int unused, void *data) goto exit; } - irq_fired = false; for (i = 0; i < RT5677_IRQ_NUM; i++) { if (reg_irq & rt5677_irq_descs[i].status_mask) { irq_fired = true; -- 2.22.0.410.gd8fdbe21b5-goog

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[alsa-devel] [PATCH] ASoC: SOF: use __u32 instead of uint32_t in uapi headers
by Masahiro Yamada 22 Jul '19

22 Jul '19

When CONFIG_UAPI_HEADER_TEST=y, exported headers are compile-tested to make sure they can be included from user-space. Currently, header.h and fw.h are excluded from the test coverage. To make them join the compile-test, we need to fix the build errors attached below. For a case like this, we decided to use __u{8,16,32,64} variable types in this discussion: https://lkml.org/lkml/2019/6/5/18 Build log: CC usr/include/sound/sof/header.h.s CC usr/include/sound/sof/fw.h.s In file included from <command-line>:32:0: ./usr/include/sound/sof/header.h:19:2: error: unknown type name ‘uint32_t’ uint32_t magic; /**< 'S', 'O', 'F', '\0' */ ^~~~~~~~ ./usr/include/sound/sof/header.h:20:2: error: unknown type name ‘uint32_t’ uint32_t type; /**< component specific type */ ^~~~~~~~ ./usr/include/sound/sof/header.h:21:2: error: unknown type name ‘uint32_t’ uint32_t size; /**< size in bytes of data excl. this struct */ ^~~~~~~~ ./usr/include/sound/sof/header.h:22:2: error: unknown type name ‘uint32_t’ uint32_t abi; /**< SOF ABI version */ ^~~~~~~~ ./usr/include/sound/sof/header.h:23:2: error: unknown type name ‘uint32_t’ uint32_t reserved[4]; /**< reserved for future use */ ^~~~~~~~ ./usr/include/sound/sof/header.h:24:2: error: unknown type name ‘uint32_t’ uint32_t data[0]; /**< Component data - opaque to core */ ^~~~~~~~ In file included from <command-line>:32:0: ./usr/include/sound/sof/fw.h:49:2: error: unknown type name ‘uint32_t’ uint32_t size; /* bytes minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:50:2: error: unknown type name ‘uint32_t’ uint32_t offset; /* offset from base */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:64:2: error: unknown type name ‘uint32_t’ uint32_t size; /* bytes minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:65:2: error: unknown type name ‘uint32_t’ uint32_t num_blocks; /* number of blocks */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:73:2: error: unknown type name ‘uint32_t’ uint32_t file_size; /* size of file minus this header */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:74:2: error: unknown type name ‘uint32_t’ uint32_t num_modules; /* number of modules */ ^~~~~~~~ ./usr/include/sound/sof/fw.h:75:2: error: unknown type name ‘uint32_t’ uint32_t abi; /* version of header format */ ^~~~~~~~ Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- include/uapi/sound/sof/fw.h | 16 +++++++++------- include/uapi/sound/sof/header.h | 14 ++++++++------ 2 files changed, 17 insertions(+), 13 deletions(-) diff --git a/include/uapi/sound/sof/fw.h b/include/uapi/sound/sof/fw.h index 1afca973eb09..e9f697467a86 100644 --- a/include/uapi/sound/sof/fw.h +++ b/include/uapi/sound/sof/fw.h @@ -13,6 +13,8 @@ #ifndef __INCLUDE_UAPI_SOF_FW_H__ #define __INCLUDE_UAPI_SOF_FW_H__ +#include <linux/types.h> + #define SND_SOF_FW_SIG_SIZE 4 #define SND_SOF_FW_ABI 1 #define SND_SOF_FW_SIG "Reef" @@ -46,8 +48,8 @@ enum snd_sof_fw_blk_type { struct snd_sof_blk_hdr { enum snd_sof_fw_blk_type type; - uint32_t size; /* bytes minus this header */ - uint32_t offset; /* offset from base */ + __u32 size; /* bytes minus this header */ + __u32 offset; /* offset from base */ } __packed; /* @@ -61,8 +63,8 @@ enum snd_sof_fw_mod_type { struct snd_sof_mod_hdr { enum snd_sof_fw_mod_type type; - uint32_t size; /* bytes minus this header */ - uint32_t num_blocks; /* number of blocks */ + __u32 size; /* bytes minus this header */ + __u32 num_blocks; /* number of blocks */ } __packed; /* @@ -70,9 +72,9 @@ struct snd_sof_mod_hdr { */ struct snd_sof_fw_header { unsigned char sig[SND_SOF_FW_SIG_SIZE]; /* "Reef" */ - uint32_t file_size; /* size of file minus this header */ - uint32_t num_modules; /* number of modules */ - uint32_t abi; /* version of header format */ + __u32 file_size; /* size of file minus this header */ + __u32 num_modules; /* number of modules */ + __u32 abi; /* version of header format */ } __packed; #endif diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h index 7868990b0d6f..5f4518e7a972 100644 --- a/include/uapi/sound/sof/header.h +++ b/include/uapi/sound/sof/header.h @@ -9,6 +9,8 @@ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ #define __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ +#include <linux/types.h> + /* * Header for all non IPC ABI data. * @@ -16,12 +18,12 @@ * Used by any bespoke component data structures or binary blobs. */ struct sof_abi_hdr { - uint32_t magic; /**< 'S', 'O', 'F', '\0' */ - uint32_t type; /**< component specific type */ - uint32_t size; /**< size in bytes of data excl. this struct */ - uint32_t abi; /**< SOF ABI version */ - uint32_t reserved[4]; /**< reserved for future use */ - uint32_t data[0]; /**< Component data - opaque to core */ + __u32 magic; /**< 'S', 'O', 'F', '\0' */ + __u32 type; /**< component specific type */ + __u32 size; /**< size in bytes of data excl. this struct */ + __u32 abi; /**< SOF ABI version */ + __u32 reserved[4]; /**< reserved for future use */ + __u32 data[0]; /**< Component data - opaque to core */ } __packed; #endif -- 2.17.1

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[alsa-devel] [PATCH 00/20] ALSA: firewire-lib: use packet descriptor to represent sequence of packet
by Takashi Sakamoto 22 Jul '19

22 Jul '19

Hi, This patchset is for Linux kernel v5.4. In current implementation of ALSA IEC 61883-1/6 packet streaming engine, one callback of isochronous context consists three steps to process an isochronous packet: Callback of isochronous context - For each of packets 1.parse context header and decide the amount of PCM frames 2.process data blocks (in each unit driver) 3.queue packet The result of step 1 affects the rest, thus the above steps can be changed for below process: Callback of isochronous context 1.parse context header and decide the amount of PCM frames - For each of packets 2.process data blocks (in each unit driver) - For each of packets 3.queue packet - For each of packets Especially it's convenient to decide the amount of PCM frames multiplexed into each packet as batch calculation before processing payloads of these packets. Additionally it allows each unit driver to process data blocks for these packet by one call. This patchset uses list of 'struct pkt_desc' as an intermediate representation of packet parameters between these three steps. Here, I note my future plan for ALSA firewire stack for your motivation to review this patchset: * v5.4 * Handle several IT/IR context in one callback of IT context. This idea will introduce AMDTP domain structure with relationship of IRQ master/slave between the IT/IR contexts. * As a result, both of playback/capture PCM substreams run on the same hardware interrupt. Thus the drivers just support batch PCM operation and irq-based programming model[1]. * v5.5 * Some (p.s. not all) supported devices don't follow packet sequences transferred by the drivers. They require clock recovery in driver side to reduce playback noise. Enhance the AMDTP domain structure to have clock recovery target and fill list of 'struct pkt_desc' according to the result of clock recovery. My work for libhinoko[2] allows me to sniff actual packet transmission between devices and drivers in Windows/Mac OS for long period (e.g. 1 hour). Then I can classify some cases and some devices for the requirement of clock recovery. (I note that no specification describes this mechanism as long as I know. In the specification, clock recovery is one-way from transmitter to receivers, thus recovered clock is not necessarily used for transmission from the receiver to the transmitter.) [1] https://git.alsa-project.org/?p=alsa-utils.git;a=blob;f=axfer/axfer-transfe… [2] https://github.com/takaswie/libhinoko Takashi Sakamoto (20): ALSA: firewire-lib: obsolete ctx_data.tx.first_dbc with CIP_UNALIGHED_DBC flag ALSA: firewire-lib: pass data block count as an argument to tracepoints event ALSA: firewire-lib: pass data block counter to data block processing layer ALSA: firewire-lib: operate data block counter in top level of processing for IT context ALSA: firewire-lib: operate data block counter in top level of processing for IR context ALSA: firewire-lib: add syt_override member for some protocols ALSA: firewire-lib: pass no syt information to data block processing layer ALSA: firewire-lib: add list of packet descriptor ALSA: firewire-lib: use packet descriptor for IT context ALSA: firewire-lib: use packet descriptor for IR context ALSA: firewire-lib: code refactoring to process PCM substream ALSA: firewire-lib: code refactoring to process context payloads ALSA: firewire-lib: pass packet descriptor to data block processing layer ALSA: firewire-lib: code refactoring for AM824 data block processing layer ALSA: firewire-digi00x: code refactoring for DOT data block processing layer ALSA: firewire-tascam: code refactoring for TASCAM data block processing layer ALSA: firewire-motu: code refactoring for MOTU data block processing layer ALSA: fireface: code refactoring for FF data block processing layer ALSA: firewire-lib: process payload of isoc context according to packet descriptors ALSA: firewire-motu: more code refactoring for MOTU data block processing layer sound/firewire/amdtp-am824.c | 134 ++++++---- sound/firewire/amdtp-stream-trace.h | 6 +- sound/firewire/amdtp-stream.c | 264 ++++++++++++-------- sound/firewire/amdtp-stream.h | 27 +- sound/firewire/digi00x/amdtp-dot.c | 112 +++++---- sound/firewire/fireface/amdtp-ff.c | 105 ++++---- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/motu/amdtp-motu.c | 155 ++++++++---- sound/firewire/tascam/amdtp-tascam.c | 115 +++++---- 9 files changed, 570 insertions(+), 350 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 0/3] i.MX8M support for FSL SAI
by Lucas Stach 22 Jul '19

22 Jul '19

Hi all, those are the changes that are absolutely needed to support the i.MX8M SAI block. The most noticable difference is the moved register layout, due to the hardware guys adding two version registers at the start of the register map and considerable bigger FIFO sizes. Downstream has a lot more changes to support new features, but those 3 patches at least allows me to play back audio on i.MX8M based systems. Regards, Lucas Lucas Stach (3): ASoC: fsl_sai: add of_match data ASoC: fsl_sai: derive TX FIFO watermark from FIFO depth ASoC: fsl_sai: add i.MX8M support .../devicetree/bindings/sound/fsl-sai.txt | 2 +- sound/soc/fsl/fsl_sai.c | 45 +++++++++++++++---- sound/soc/fsl/fsl_sai.h | 8 +++- 3 files changed, 44 insertions(+), 11 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] SoC: rockchip: rockchip_max98090: Enable MICBIAS for headset keypress detection
by Enric Balletbo i Serra 22 Jul '19

22 Jul '19

The TS3A227E says that the headset keypress detection needs the MICBIAS power in order to report the key events to ensure proper operation The headset keypress detection needs the MICBIAS power in order to report the key events all the time as long as MIC is present. So MICBIAS pin is forced on when a MICROPHONE is detected. On Veyron Minnie I observed that if the MICBIAS power is not present and the key press detection is activated (just because it is enabled when you insert a headset), it randomly reports a keypress on insert. E.g. (KEY_PLAYPAUSE) Event: (SW_HEADPHONE_INSERT), value 1 Event: (SW_MICROPHONE_INSERT), value 1 Event: -------------- SYN_REPORT ------------ Event: (KEY_PLAYPAUSE), value 1 Userspace thinks that KEY_PLAYPAUSE is pressed and produces the annoying effect that the media player starts a play/pause loop. Note that, although most of the time the key reported is the one associated with BTN_0, not always this is true. On my tests I also saw different keys reported Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- Some notes about the patch Steps to test (userspace GNOME3): 1. Play audio using a media player 2. Make sure the Internal MIC is selected as audio input device and doesn't switches when you insert the headset. 3. Insert a headset (with buttons) Audio switches to headphones and you can hear that enters on a loop play/pause if the KEY_PLAYPAUSE was reported. Also you can check that press to any headset button doesn't work. The part where the datasheet says that power must be supplied in order to have keypress detection work is in page 44 of [1] " The TS3A227E can monitor the microphone line of a 4-pole headset to detect up to 4 key presses/releases and report the key press events back to the host. The key press detection must be activated manually by setting the KP Enable bit of the Device Settings 2 register. To ensure proper operation the MICBIAS voltage must be applied to MICP before enabling key press detection. " [1] http://www.ti.com/lit/ds/symlink/ts3a227e.pdf sound/soc/rockchip/rockchip_max98090.c | 32 ++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/rockchip/rockchip_max98090.c b/sound/soc/rockchip/rockchip_max98090.c index c5fc24675a33..782e534d4c0d 100644 --- a/sound/soc/rockchip/rockchip_max98090.c +++ b/sound/soc/rockchip/rockchip_max98090.c @@ -61,6 +61,37 @@ static const struct snd_kcontrol_new rk_mc_controls[] = { SOC_DAPM_PIN_SWITCH("Speaker"), }; +static int rk_jack_event(struct notifier_block *nb, unsigned long event, + void *data) +{ + struct snd_soc_jack *jack = (struct snd_soc_jack *)data; + struct snd_soc_dapm_context *dapm = &jack->card->dapm; + + if (event & SND_JACK_MICROPHONE) + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + else + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct notifier_block rk_jack_nb = { + .notifier_call = rk_jack_event, +}; + +static int rk_init(struct snd_soc_pcm_runtime *runtime) +{ + /* + * The jack has already been created in the rk_98090_headset_init() + * function. + */ + snd_soc_jack_notifier_register(&headset_jack, &rk_jack_nb); + + return 0; +} + static int rk_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -119,6 +150,7 @@ SND_SOC_DAILINK_DEFS(hifi, static struct snd_soc_dai_link rk_dailink = { .name = "max98090", .stream_name = "Audio", + .init = rk_init, .ops = &rk_aif1_ops, /* set max98090 as slave */ .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | -- 2.20.1

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[alsa-devel] [PATCH] ASoC: dapm: Fix handling of custom_stop_condition on DAPM graph walks
by Charles Keepax 22 Jul '19

22 Jul '19

DPCM uses snd_soc_dapm_dai_get_connected_widgets to build a list of the widgets connected to a specific front end DAI so it can search through this list for available back end DAIs. The custom_stop_condition was added to is_connected_ep to facilitate this list not containing more widgets than is necessary. Doing so both speeds up the DPCM handling as less widgets need to be searched and avoids issues with CODEC to CODEC links as these would be confused with back end DAIs if they appeared in the list of available widgets. custom_stop_condition was implemented by aborting the graph walk when the condition is triggered, however there is an issue with this approach. Whilst walking the graph is_connected_ep should update the endpoints cache on each widget, if the walk is aborted the number of attached end points is unknown for that sub-graph. When the stop condition triggered, the original patch ignored the triggering widget and returned zero connected end points; a later patch updated this to set the triggering widget's cache to 1 and return that. Both of these approaches result in inaccurate values being stored in various end point caches as the values propagate back through the graph, which can result in later issues with widgets powering/not powering unexpectedly. As the original goal was to reduce the size of the widget list passed to the DPCM code, the simplest solution is to limit the functionality of the custom_stop_condition to the widget list. This means the rest of the graph will still be processed resulting in correct end point caches, but only widgets up to the stop condition will be added to the returned widget list. Fixes: 6742064aef7f ("ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets") Fixes: 5fdd022c2026 ("ASoC: dpcm: play nice with CODEC<->CODEC links") Fixes: 09464974eaa8 ("ASoC: dapm: Fix to return correct path list in is_connected_ep.") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/soc-dapm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f013b24c050a1..f73882cf0031d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1157,8 +1157,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, list_add_tail(&widget->work_list, list); if (custom_stop_condition && custom_stop_condition(widget, dir)) { - widget->endpoints[dir] = 1; - return widget->endpoints[dir]; + list = NULL; + custom_stop_condition = NULL; } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { @@ -1195,8 +1195,8 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, * * Optionally, can be supplied with a function acting as a stopping condition. * This function takes the dapm widget currently being examined and the walk - * direction as an arguments, it should return true if the walk should be - * stopped and false otherwise. + * direction as an arguments, it should return true if widgets from that point + * in the graph onwards should not be added to the widget list. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, -- 2.11.0

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[alsa-devel] [PATCH] ASoC: cs42xx8: Fix MFREQ selection issue for async mode
by shengjiu.wang＠nxp.com 22 Jul '19

22 Jul '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> When sample rate of TX is different with sample rate of RX in async mode, the MFreq selection will be wrong. For example, sysclk = 24.576MHz, TX rate = 96000Hz, RX rate = 48000Hz. Then ratio of TX = 256, ratio of RX = 512, For MFreq is shared by TX and RX instance, the correct value of MFreq is 2 for both TX and RX. But original method will cause MFreq = 0 for TX, MFreq = 2 for RX. If TX is started after RX, RX will be impacted, RX work abnormal with MFreq = 0. This patch is to select proper MFreq value according to TX rate and RX rate. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 116 +++++++++++++++++++++++++++++++------ 1 file changed, 97 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index b377cddaf2e6..804969495754 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -47,6 +47,7 @@ struct cs42xx8_priv { unsigned long sysclk; u32 tx_channels; struct gpio_desc *gpiod_reset; + u32 rate[2]; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -176,21 +177,27 @@ static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { }; struct cs42xx8_ratios { - unsigned int ratio; - unsigned char speed; - unsigned char mclk; + unsigned int mfreq; + unsigned int min_mclk; + unsigned int max_mclk; + unsigned int ratio[3]; }; +/* + * According to reference mannual, define the cs42xx8_ratio struct + * MFreq2 | MFreq1 | MFreq0 | Description | SSM | DSM | QSM | + * 0 | 0 | 0 |1.029MHz to 12.8MHz | 256 | 128 | 64 | + * 0 | 0 | 1 |1.536MHz to 19.2MHz | 384 | 192 | 96 | + * 0 | 1 | 0 |2.048MHz to 25.6MHz | 512 | 256 | 128 | + * 0 | 1 | 1 |3.072MHz to 38.4MHz | 768 | 384 | 192 | + * 1 | x | x |4.096MHz to 51.2MHz |1024 | 512 | 256 | + */ static const struct cs42xx8_ratios cs42xx8_ratios[] = { - { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, - { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, - { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, - { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, - { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, - { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, - { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, - { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, - { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } + { 0, 1029000, 12800000, {256, 128, 64} }, + { 2, 1536000, 19200000, {384, 192, 96} }, + { 4, 2048000, 25600000, {512, 256, 128} }, + { 6, 3072000, 38400000, {768, 384, 192} }, + { 8, 4096000, 51200000, {1024, 512, 256} }, }; static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, @@ -257,14 +264,68 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - u32 ratio = cs42xx8->sysclk / params_rate(params); - u32 i, fm, val, mask; + u32 ratio[2]; + u32 rate[2]; + u32 fm[2]; + u32 i, val, mask; + bool condition1, condition2; if (tx) cs42xx8->tx_channels = params_channels(params); + rate[tx] = params_rate(params); + rate[!tx] = cs42xx8->rate[!tx]; + + ratio[tx] = rate[tx] > 0 ? cs42xx8->sysclk / rate[tx] : 0; + ratio[!tx] = rate[!tx] > 0 ? cs42xx8->sysclk / rate[!tx] : 0; + + /* Get functional mode for tx and rx according to rate */ + for (i = 0; i < 2; i++) { + if (cs42xx8->slave_mode) { + fm[i] = CS42XX8_FM_AUTO; + } else { + if (rate[i] < 50000) { + fm[i] = CS42XX8_FM_SINGLE; + } else if (rate[i] > 50000 && rate[i] < 100000) { + fm[i] = CS42XX8_FM_DOUBLE; + } else if (rate[i] > 100000 && rate[i] < 200000) { + fm[i] = CS42XX8_FM_QUAD; + } else { + dev_err(component->dev, + "unsupported sample rate\n"); + return -EINVAL; + } + } + } + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { - if (cs42xx8_ratios[i].ratio == ratio) + /* Is the ratio[tx] valid ? */ + condition1 = ((fm[tx] == CS42XX8_FM_AUTO) ? + (cs42xx8_ratios[i].ratio[0] == ratio[tx] || + cs42xx8_ratios[i].ratio[1] == ratio[tx] || + cs42xx8_ratios[i].ratio[2] == ratio[tx]) : + (cs42xx8_ratios[i].ratio[fm[tx]] == ratio[tx])) && + cs42xx8->sysclk >= cs42xx8_ratios[i].min_mclk && + cs42xx8->sysclk <= cs42xx8_ratios[i].max_mclk; + + if (!ratio[tx]) + condition1 = true; + + /* Is the ratio[!tx] valid ? */ + condition2 = ((fm[!tx] == CS42XX8_FM_AUTO) ? + (cs42xx8_ratios[i].ratio[0] == ratio[!tx] || + cs42xx8_ratios[i].ratio[1] == ratio[!tx] || + cs42xx8_ratios[i].ratio[2] == ratio[!tx]) : + (cs42xx8_ratios[i].ratio[fm[!tx]] == ratio[!tx])); + + if (!ratio[!tx]) + condition2 = true; + + /* + * Both ratio[tx] and ratio[!tx] is valid, then we get + * a proper MFreq. + */ + if (condition1 && condition2) break; } @@ -273,15 +334,31 @@ static int cs42xx8_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - mask = CS42XX8_FUNCMOD_MFREQ_MASK; - val = cs42xx8_ratios[i].mclk; + cs42xx8->rate[tx] = params_rate(params); - fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mfreq; regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, - CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + CS42XX8_FUNCMOD_xC_FM(tx, fm[tx]) | val); + + return 0; +} + +static int cs42xx8_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct cs42xx8_priv *cs42xx8 = snd_soc_component_get_drvdata(component); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + /* Clear stored rate */ + cs42xx8->rate[tx] = 0; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx), + CS42XX8_FUNCMOD_xC_FM(tx, CS42XX8_FM_AUTO)); return 0; } @@ -302,6 +379,7 @@ static const struct snd_soc_dai_ops cs42xx8_dai_ops = { .set_fmt = cs42xx8_set_dai_fmt, .set_sysclk = cs42xx8_set_dai_sysclk, .hw_params = cs42xx8_hw_params, + .hw_free = cs42xx8_hw_free, .digital_mute = cs42xx8_digital_mute, }; -- 2.21.0

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[alsa-devel] [PATCH] ASoC: codecs: ad193x: Use regmap_multi_reg_write() when initializing
by Codrin Ciubotariu 22 Jul '19

22 Jul '19

Using regmap_multi_reg_write() when we set the default values for our registers makes the code smaller and easier to read. Suggested-by: Tzung-Bi Shih <tzungbi(a)google.com> Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> --- sound/soc/codecs/ad193x.c | 19 +++++-------------- 1 file changed, 5 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 80dab5df9633..fb04c9379b71 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -413,15 +413,10 @@ static struct snd_soc_dai_driver ad193x_no_adc_dai = { .ops = &ad193x_dai_ops, }; -struct ad193x_reg_default { - unsigned int reg; - unsigned int val; -}; - /* codec register values to set after reset */ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { - const struct ad193x_reg_default reg_init[] = { + const struct reg_sequence reg_init[] = { { 0, 0x99 }, /* PLL_CLK_CTRL0: pll input: mclki/xi 12.288Mhz */ { 1, 0x04 }, /* PLL_CLK_CTRL1: no on-chip Vref */ { 2, 0x40 }, /* DAC_CTRL0: TDM mode */ @@ -437,21 +432,17 @@ static void ad193x_reg_default_init(struct ad193x_priv *ad193x) { 12, 0x00 }, /* DAC_L4_VOL: no attenuation */ { 13, 0x00 }, /* DAC_R4_VOL: no attenuation */ }; - const struct ad193x_reg_default reg_adc_init[] = { + const struct reg_sequence reg_adc_init[] = { { 14, 0x03 }, /* ADC_CTRL0: high-pass filter enable */ { 15, 0x43 }, /* ADC_CTRL1: sata delay=1, adc aux mode */ { 16, 0x00 }, /* ADC_CTRL2: reset */ }; - int i; - for (i = 0; i < ARRAY_SIZE(reg_init); i++) - regmap_write(ad193x->regmap, reg_init[i].reg, reg_init[i].val); + regmap_multi_reg_write(ad193x->regmap, reg_init, ARRAY_SIZE(reg_init)); if (ad193x_has_adc(ad193x)) { - for (i = 0; i < ARRAY_SIZE(reg_adc_init); i++) { - regmap_write(ad193x->regmap, reg_adc_init[i].reg, - reg_adc_init[i].val); - } + regmap_multi_reg_write(ad193x->regmap, reg_adc_init, + ARRAY_SIZE(reg_adc_init)); } } -- 2.20.1

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