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[PATCH v2 0/3] Fix SND_HDA_PREALLOC issue
by Amadeusz Sławiński 19 Mar '21

19 Mar '21

For context it started with user reporting failures when running arecord without any error or warning in dmesg (after fixing some configuration problems thet they had). https://bugzilla.kernel.org/show_bug.cgi?id=201251#c279 After spending time investigating the issue it was narrowed to quite big setting of CONFIG_SND_HDA_PREALLOC_SIZE (4096). When looking at code https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… there is a limit of memory per card: max_alloc_per_card = 32UL * 1024UL * 1024UL When SND_HDA_PREALLOC_SIZE is set to 4096 it only has memory to alloc for 8 frontends, while Skylake HDA card has 10 of them (6 playback and 4 capture), so preallocated memory is exhausted while probing. In consequence 2 of FEs end without allocated memory. It can be workarounded on user side with setting SND_HDA_PREALLOC_SIZE to lower value, other is changing memory limit per card. However in order to not waste user memory, change maximum allocation size on HDA controller to 4MB and force automatical memory allocation insted of preallocated one. First patch adds prints, so similar issues can be easily identified in the future. Second changes maximum size of hda buffer to reasonable value of 4MB. And last one reverts patch which allowed setting prealloc size on X86 platforms. Amadeusz Sławiński (3): ALSA: pcm: Add debug print on memory allocation failure ALSA: hda: Change AZX_MAX_BUF_SIZE from 1GB to 4MB ALSA: hda: Revert "ALSA: hda: Allow setting preallocation again for x86" include/sound/hda_register.h | 8 ++++++-- sound/core/pcm_memory.c | 8 ++++++++ sound/hda/Kconfig | 7 +++---- 3 files changed, 17 insertions(+), 6 deletions(-) -- 2.25.1

2 4

[PATCH v4 1/3] ASoC: Add Rockchip rk817 audio CODEC support
by Chris Morgan 19 Mar '21

19 Mar '21

Add support for the Rockchip rk817 audio codec integrated into the rk817 PMIC. This is based on the sources provided by Rockchip from their BSP kernel. Changes from v3: Created separate set_pll() call. Created user visible gain control for microphone. Check for return value of clk_prepare_enable(). Remove duplicate clk_prepare_enable(). Moved documentation to a separate commit. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rk817_codec.c | 706 +++++++++++++++++++++++++++++++++ sound/soc/codecs/rk817_codec.h | 94 +++++ 4 files changed, 805 insertions(+) create mode 100644 sound/soc/codecs/rk817_codec.c create mode 100644 sound/soc/codecs/rk817_codec.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e4cf14e66a51..d835145ad65e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -155,6 +155,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_PCM512x_I2C imply SND_SOC_PCM512x_SPI imply SND_SOC_RK3328 + imply SND_SOC_RK817 imply SND_SOC_RT274 imply SND_SOC_RT286 imply SND_SOC_RT298 @@ -1059,6 +1060,11 @@ config SND_SOC_RK3328 tristate "Rockchip RK3328 audio CODEC" select REGMAP_MMIO +config SND_SOC_RK817 + tristate "Rockchip RK817 audio CODEC" + depends on MFD_RK808 + select REGMAP_I2C + config SND_SOC_RL6231 tristate default y if SND_SOC_RT5514=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81357dc62ea0..4e5e9b28775c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -165,6 +165,7 @@ snd-soc-pcm512x-objs := pcm512x.o snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rk3328-objs := rk3328_codec.o +snd-soc-rk817-objs := rk817_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o @@ -479,6 +480,7 @@ obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o +obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c new file mode 100644 index 000000000000..045acf05da5c --- /dev/null +++ b/sound/soc/codecs/rk817_codec.c @@ -0,0 +1,706 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// rk817 ALSA SoC Audio driver +// +// Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/mfd/rk808.h> +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "rk817_codec.h" + +struct rk817_codec_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct rk808 *rk817; + struct clk *mclk; + unsigned int stereo_sysclk; + bool mic_in_differential; +}; + +static const struct reg_default rk817_reg_defaults[] = { + { RK817_CODEC_DTOP_VUCTL, 0x03 }, + { RK817_CODEC_DTOP_VUCTIME, 0x00 }, + { RK817_CODEC_DTOP_LPT_SRST, 0x00 }, + { RK817_CODEC_DTOP_DIGEN_CLKE, 0x00 }, + /* from vendor driver, CODEC_AREF_RTCFG0 not defined in data sheet */ + { RK817_CODEC_AREF_RTCFG0, 0x00 }, + { RK817_CODEC_AREF_RTCFG1, 0x06 }, + { RK817_CODEC_AADC_CFG0, 0xc8 }, + /* from vendor driver, CODEC_AADC_CFG1 not defined in data sheet */ + { RK817_CODEC_AADC_CFG1, 0x00 }, + { RK817_CODEC_DADC_VOLL, 0x00 }, + { RK817_CODEC_DADC_VOLR, 0x00 }, + { RK817_CODEC_DADC_SR_ACL0, 0x00 }, + { RK817_CODEC_DADC_ALC1, 0x00 }, + { RK817_CODEC_DADC_ALC2, 0x00 }, + { RK817_CODEC_DADC_NG, 0x00 }, + { RK817_CODEC_DADC_HPF, 0x00 }, + { RK817_CODEC_DADC_RVOLL, 0xff }, + { RK817_CODEC_DADC_RVOLR, 0xff }, + { RK817_CODEC_AMIC_CFG0, 0x70 }, + { RK817_CODEC_AMIC_CFG1, 0x00 }, + { RK817_CODEC_DMIC_PGA_GAIN, 0x66 }, + { RK817_CODEC_DMIC_LMT1, 0x00 }, + { RK817_CODEC_DMIC_LMT2, 0x00 }, + { RK817_CODEC_DMIC_NG1, 0x00 }, + { RK817_CODEC_DMIC_NG2, 0x00 }, + /* from vendor driver, CODEC_ADAC_CFG0 not defined in data sheet */ + { RK817_CODEC_ADAC_CFG0, 0x00 }, + { RK817_CODEC_ADAC_CFG1, 0x07 }, + { RK817_CODEC_DDAC_POPD_DACST, 0x82 }, + { RK817_CODEC_DDAC_VOLL, 0x00 }, + { RK817_CODEC_DDAC_VOLR, 0x00 }, + { RK817_CODEC_DDAC_SR_LMT0, 0x00 }, + { RK817_CODEC_DDAC_LMT1, 0x00 }, + { RK817_CODEC_DDAC_LMT2, 0x00 }, + { RK817_CODEC_DDAC_MUTE_MIXCTL, 0xa0 }, + { RK817_CODEC_DDAC_RVOLL, 0xff }, + { RK817_CODEC_DDAC_RVOLR, 0xff }, + { RK817_CODEC_AHP_ANTI0, 0x00 }, + { RK817_CODEC_AHP_ANTI1, 0x00 }, + { RK817_CODEC_AHP_CFG0, 0xe0 }, + { RK817_CODEC_AHP_CFG1, 0x1f }, + { RK817_CODEC_AHP_CP, 0x09 }, + { RK817_CODEC_ACLASSD_CFG1, 0x69 }, + { RK817_CODEC_ACLASSD_CFG2, 0x44 }, + { RK817_CODEC_APLL_CFG0, 0x04 }, + { RK817_CODEC_APLL_CFG1, 0x00 }, + { RK817_CODEC_APLL_CFG2, 0x30 }, + { RK817_CODEC_APLL_CFG3, 0x19 }, + { RK817_CODEC_APLL_CFG4, 0x65 }, + { RK817_CODEC_APLL_CFG5, 0x01 }, + { RK817_CODEC_DI2S_CKM, 0x01 }, + { RK817_CODEC_DI2S_RSD, 0x00 }, + { RK817_CODEC_DI2S_RXCR1, 0x00 }, + { RK817_CODEC_DI2S_RXCR2, 0x17 }, + { RK817_CODEC_DI2S_RXCMD_TSD, 0x00 }, + { RK817_CODEC_DI2S_TXCR1, 0x00 }, + { RK817_CODEC_DI2S_TXCR2, 0x17 }, + { RK817_CODEC_DI2S_TXCR3_TXCMD, 0x00 }, +}; + +static bool rk817_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_LPT_SRST: + return true; + default: + return false; + } +} + +static bool rk817_codec_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RK817_CODEC_DTOP_VUCTL: + case RK817_CODEC_DTOP_VUCTIME: + case RK817_CODEC_DTOP_LPT_SRST: + case RK817_CODEC_DTOP_DIGEN_CLKE: + case RK817_CODEC_AREF_RTCFG0: + case RK817_CODEC_AREF_RTCFG1: + case RK817_CODEC_AADC_CFG0: + case RK817_CODEC_AADC_CFG1: + case RK817_CODEC_DADC_VOLL: + case RK817_CODEC_DADC_VOLR: + case RK817_CODEC_DADC_SR_ACL0: + case RK817_CODEC_DADC_ALC1: + case RK817_CODEC_DADC_ALC2: + case RK817_CODEC_DADC_NG: + case RK817_CODEC_DADC_HPF: + case RK817_CODEC_DADC_RVOLL: + case RK817_CODEC_DADC_RVOLR: + case RK817_CODEC_AMIC_CFG0: + case RK817_CODEC_AMIC_CFG1: + case RK817_CODEC_DMIC_PGA_GAIN: + case RK817_CODEC_DMIC_LMT1: + case RK817_CODEC_DMIC_LMT2: + case RK817_CODEC_DMIC_NG1: + case RK817_CODEC_DMIC_NG2: + case RK817_CODEC_ADAC_CFG0: + case RK817_CODEC_ADAC_CFG1: + case RK817_CODEC_DDAC_POPD_DACST: + case RK817_CODEC_DDAC_VOLL: + case RK817_CODEC_DDAC_VOLR: + case RK817_CODEC_DDAC_SR_LMT0: + case RK817_CODEC_DDAC_LMT1: + case RK817_CODEC_DDAC_LMT2: + case RK817_CODEC_DDAC_MUTE_MIXCTL: + case RK817_CODEC_DDAC_RVOLL: + case RK817_CODEC_DDAC_RVOLR: + case RK817_CODEC_AHP_ANTI0: + case RK817_CODEC_AHP_ANTI1: + case RK817_CODEC_AHP_CFG0: + case RK817_CODEC_AHP_CFG1: + case RK817_CODEC_AHP_CP: + case RK817_CODEC_ACLASSD_CFG1: + case RK817_CODEC_ACLASSD_CFG2: + case RK817_CODEC_APLL_CFG0: + case RK817_CODEC_APLL_CFG1: + case RK817_CODEC_APLL_CFG2: + case RK817_CODEC_APLL_CFG3: + case RK817_CODEC_APLL_CFG4: + case RK817_CODEC_APLL_CFG5: + case RK817_CODEC_DI2S_CKM: + case RK817_CODEC_DI2S_RSD: + case RK817_CODEC_DI2S_RXCR1: + case RK817_CODEC_DI2S_RXCR2: + case RK817_CODEC_DI2S_RXCMD_TSD: + case RK817_CODEC_DI2S_TXCR1: + case RK817_CODEC_DI2S_TXCR2: + case RK817_CODEC_DI2S_TXCR3_TXCMD: + return true; + default: + return false; + } +} + +/* + * This sets the codec up with the values defined in the default implementation including the APLL + * from the Rockchip vendor kernel. I do not know if these values are universal despite differing + * from the default values defined above and taken from the datasheet, or implementation specific. + * I don't have another implementation to compare from the Rockchip sources. Hard-coding for now. + */ + +static int rk817_init(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_write(component, RK817_CODEC_DDAC_POPD_DACST, 0x02); + snd_soc_component_write(component, RK817_CODEC_DDAC_SR_LMT0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DADC_SR_ACL0, 0x02); + snd_soc_component_write(component, RK817_CODEC_DTOP_VUCTIME, 0xf4); + if (rk817->mic_in_differential) { + snd_soc_component_update_bits(component, RK817_CODEC_AMIC_CFG0, MIC_DIFF_MASK, + MIC_DIFF_EN); + }; + return 0; +} + +static int rk817_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + snd_soc_component_write(component, RK817_CODEC_APLL_CFG1, 0x58); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG2, 0x2d); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG3, 0x0c); + snd_soc_component_write(component, RK817_CODEC_APLL_CFG4, 0xa5); + + return 0; +}; + +/* + * DDAC/DADC L/R volume setting + * 0db~-95db, 0.375db/step, for example: + * 0x00: 0dB + * 0xff: -95dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_vol_tlv, -9500, 0); + +/* + * PGA GAIN L/R volume setting + * 27db~-18db, 3db/step, for example: + * 0x0: -18dB + * 0xf: 27dB + */ + +static const DECLARE_TLV_DB_MINMAX(rk817_gain_tlv, -1800, 2700); + +static const struct snd_kcontrol_new rk817_volume_controls[] = { + SOC_DOUBLE_R_RANGE_TLV("Master Playback Volume", RK817_CODEC_DDAC_VOLL, + RK817_CODEC_DDAC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_R_RANGE_TLV("Master Capture Volume", RK817_CODEC_DADC_VOLL, + RK817_CODEC_DADC_VOLR, 0, 0x00, 0xff, 1, rk817_vol_tlv), + SOC_DOUBLE_TLV("Mic Capture Gain", RK817_CODEC_DMIC_PGA_GAIN, 4, 0, 0xf, 0, + rk817_gain_tlv), +}; + +static const char *dac_mux_text[] = { + "HP", + "SPK", +}; + +static SOC_ENUM_SINGLE_VIRT_DECL(dac_enum, dac_mux_text); + +static const struct snd_kcontrol_new dac_mux = + SOC_DAPM_ENUM("Playback Mux", dac_enum); + +static const struct snd_soc_dapm_widget rk817_dapm_widgets[] = { + + /* capture/playback common */ + SND_SOC_DAPM_SUPPLY("LDO Regulator", RK817_CODEC_AREF_RTCFG1, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("IBIAS Block", RK817_CODEC_AREF_RTCFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("VAvg Buffer", RK817_CODEC_AREF_RTCFG1, 1, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL Power", RK817_CODEC_APLL_CFG5, 0, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX1 Transfer Start", RK817_CODEC_DI2S_RXCMD_TSD, 5, 0, NULL, 0), + + /* capture path common */ + SND_SOC_DAPM_SUPPLY("ADC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 6, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC Power On", RK817_CODEC_AMIC_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Transfer Start", RK817_CODEC_DI2S_TXCR3_TXCMD, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S TX3 Right Justified", RK817_CODEC_DI2S_TXCR3_TXCMD, 3, 0, NULL, 0), + + /* capture path L */ + SND_SOC_DAPM_ADC("ADC L", "Capture", RK817_CODEC_AADC_CFG0, 7, 1), + SND_SOC_DAPM_SUPPLY("PGA L Power On", RK817_CODEC_AMIC_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost L2", RK817_CODEC_AMIC_CFG0, 2, 0, NULL, 0), + + /* capture path R */ + SND_SOC_DAPM_ADC("ADC R", "Capture", RK817_CODEC_AADC_CFG0, 6, 1), + SND_SOC_DAPM_SUPPLY("PGA R Power On", RK817_CODEC_AMIC_CFG0, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R1", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Boost R2", RK817_CODEC_AMIC_CFG0, 3, 0, NULL, 0), + + /* playback path common */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Clock", RK817_CODEC_DTOP_DIGEN_CLKE, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S RX Channel Enable", RK817_CODEC_DTOP_DIGEN_CLKE, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Bias", RK817_CODEC_ADAC_CFG1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC Mute Off", RK817_CODEC_DDAC_MUTE_MIXCTL, 0, 1, NULL, 0), + + /* playback path speaker */ + SND_SOC_DAPM_SUPPLY("Class D Mode", RK817_CODEC_DDAC_MUTE_MIXCTL, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("High Pass Filter", RK817_CODEC_DDAC_MUTE_MIXCTL, 7, 0, NULL, 0), + SND_SOC_DAPM_DAC("SPK DAC", "Playback", RK817_CODEC_ADAC_CFG1, 2, 1), + SND_SOC_DAPM_SUPPLY("Enable Class D", RK817_CODEC_ACLASSD_CFG1, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Disable Class D Mute Ramp", RK817_CODEC_ACLASSD_CFG1, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 1", RK817_CODEC_ACLASSD_CFG1, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D Mute Rate 2", RK817_CODEC_ACLASSD_CFG1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 2", RK817_CODEC_ACLASSD_CFG2, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPP 3", RK817_CODEC_ACLASSD_CFG2, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 2", RK817_CODEC_ACLASSD_CFG2, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Class D OCPN 3", RK817_CODEC_ACLASSD_CFG2, 0, 0, NULL, 0), + + /* playback path headphones */ + SND_SOC_DAPM_SUPPLY("Headphone Charge Pump", RK817_CODEC_AHP_CP, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone CP Discharge LDO", RK817_CODEC_AHP_CP, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone OStage", RK817_CODEC_AHP_CFG0, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("Headphone Pre Amp", RK817_CODEC_AHP_CFG0, 5, 1, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", "Playback", RK817_CODEC_ADAC_CFG1, 1, 1), + SND_SOC_DAPM_DAC("DAC R", "Playback", RK817_CODEC_ADAC_CFG1, 0, 1), + + /* Mux for input/output path selection */ + SND_SOC_DAPM_MUX("Playback Mux", SND_SOC_NOPM, 1, 0, &dac_mux), + + /* Pins for Simple Card Bindings */ + SND_SOC_DAPM_INPUT("MICL"), + SND_SOC_DAPM_INPUT("MICR"), + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("SPKO"), +}; + +static const struct snd_soc_dapm_route rk817_dapm_routes[] = { + + /* capture path */ + /* left mic */ + {"ADC L", NULL, "LDO Regulator"}, + {"ADC L", NULL, "IBIAS Block"}, + {"ADC L", NULL, "VAvg Buffer"}, + {"ADC L", NULL, "PLL Power"}, + {"ADC L", NULL, "ADC Clock"}, + {"ADC L", NULL, "I2S TX Clock"}, + {"ADC L", NULL, "ADC Channel Enable"}, + {"ADC L", NULL, "I2S TX Channel Enable"}, + {"ADC L", NULL, "I2S TX1 Transfer Start"}, + {"MICL", NULL, "MIC Power On"}, + {"MICL", NULL, "PGA L Power On"}, + {"MICL", NULL, "Mic Boost L1"}, + {"MICL", NULL, "Mic Boost L2"}, + {"MICL", NULL, "I2S TX3 Transfer Start"}, + {"MICL", NULL, "I2S TX3 Right Justified"}, + {"ADC L", NULL, "MICL"}, + + /* right mic */ + {"ADC R", NULL, "LDO Regulator"}, + {"ADC R", NULL, "IBIAS Block"}, + {"ADC R", NULL, "VAvg Buffer"}, + {"ADC R", NULL, "PLL Power"}, + {"ADC R", NULL, "ADC Clock"}, + {"ADC R", NULL, "I2S TX Clock"}, + {"ADC R", NULL, "ADC Channel Enable"}, + {"ADC R", NULL, "I2S TX Channel Enable"}, + {"ADC R", NULL, "I2S TX1 Transfer Start"}, + {"MICR", NULL, "MIC Power On"}, + {"MICR", NULL, "PGA R Power On"}, + {"MICR", NULL, "Mic Boost R1"}, + {"MICR", NULL, "Mic Boost R2"}, + {"MICR", NULL, "I2S TX3 Transfer Start"}, + {"MICR", NULL, "I2S TX3 Right Justified"}, + {"ADC R", NULL, "MICR"}, + + /* playback path */ + /* speaker path */ + {"SPK DAC", NULL, "LDO Regulator"}, + {"SPK DAC", NULL, "IBIAS Block"}, + {"SPK DAC", NULL, "VAvg Buffer"}, + {"SPK DAC", NULL, "PLL Power"}, + {"SPK DAC", NULL, "I2S TX1 Transfer Start"}, + {"SPK DAC", NULL, "DAC Clock"}, + {"SPK DAC", NULL, "I2S RX Clock"}, + {"SPK DAC", NULL, "DAC Channel Enable"}, + {"SPK DAC", NULL, "I2S RX Channel Enable"}, + {"SPK DAC", NULL, "Class D Mode"}, + {"SPK DAC", NULL, "DAC Bias"}, + {"SPK DAC", NULL, "DAC Mute Off"}, + {"SPK DAC", NULL, "Enable Class D"}, + {"SPK DAC", NULL, "Disable Class D Mute Ramp"}, + {"SPK DAC", NULL, "Class D Mute Rate 1"}, + {"SPK DAC", NULL, "Class D Mute Rate 2"}, + {"SPK DAC", NULL, "Class D OCPP 2"}, + {"SPK DAC", NULL, "Class D OCPP 3"}, + {"SPK DAC", NULL, "Class D OCPN 2"}, + {"SPK DAC", NULL, "Class D OCPN 3"}, + {"SPK DAC", NULL, "High Pass Filter"}, + + /* headphone path L */ + {"DAC L", NULL, "LDO Regulator"}, + {"DAC L", NULL, "IBIAS Block"}, + {"DAC L", NULL, "VAvg Buffer"}, + {"DAC L", NULL, "PLL Power"}, + {"DAC L", NULL, "I2S TX1 Transfer Start"}, + {"DAC L", NULL, "DAC Clock"}, + {"DAC L", NULL, "I2S RX Clock"}, + {"DAC L", NULL, "DAC Channel Enable"}, + {"DAC L", NULL, "I2S RX Channel Enable"}, + {"DAC L", NULL, "DAC Bias"}, + {"DAC L", NULL, "DAC Mute Off"}, + {"DAC L", NULL, "Headphone Charge Pump"}, + {"DAC L", NULL, "Headphone CP Discharge LDO"}, + {"DAC L", NULL, "Headphone OStage"}, + {"DAC L", NULL, "Headphone Pre Amp"}, + + /* headphone path R */ + {"DAC R", NULL, "LDO Regulator"}, + {"DAC R", NULL, "IBIAS Block"}, + {"DAC R", NULL, "VAvg Buffer"}, + {"DAC R", NULL, "PLL Power"}, + {"DAC R", NULL, "I2S TX1 Transfer Start"}, + {"DAC R", NULL, "DAC Clock"}, + {"DAC R", NULL, "I2S RX Clock"}, + {"DAC R", NULL, "DAC Channel Enable"}, + {"DAC R", NULL, "I2S RX Channel Enable"}, + {"DAC R", NULL, "DAC Bias"}, + {"DAC R", NULL, "DAC Mute Off"}, + {"DAC R", NULL, "Headphone Charge Pump"}, + {"DAC R", NULL, "Headphone CP Discharge LDO"}, + {"DAC R", NULL, "Headphone OStage"}, + {"DAC R", NULL, "Headphone Pre Amp"}, + + /* mux path for output selection */ + {"Playback Mux", "HP", "DAC L"}, + {"Playback Mux", "HP", "DAC R"}, + {"Playback Mux", "SPK", "SPK DAC"}, + {"SPKO", NULL, "Playback Mux"}, + {"HPOL", NULL, "Playback Mux"}, + {"HPOR", NULL, "Playback Mux"}, +}; + +static int rk817_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_component *component = codec_dai->component; + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + rk817->stereo_sysclk = freq; + + return 0; +} + +static int rk817_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_component *component = codec_dai->component; + unsigned int i2s_mst = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + i2s_mst |= RK817_I2S_MODE_SLV; + break; + case SND_SOC_DAIFMT_CBM_CFM: + i2s_mst |= RK817_I2S_MODE_MST; + break; + default: + dev_err(component->dev, "%s : set master mask failed!\n", __func__); + return -EINVAL; + } + + snd_soc_component_update_bits(component, RK817_CODEC_DI2S_CKM, + RK817_I2S_MODE_MASK, i2s_mst); + + return 0; +} + +static int rk817_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_16BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_16BITS); + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + snd_soc_component_write(component, RK817_CODEC_DI2S_RXCR2, + VDW_RX_24BITS); + snd_soc_component_write(component, RK817_CODEC_DI2S_TXCR2, + VDW_TX_24BITS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int rk817_digital_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_component *component = dai->component; + + if (mute) + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_ENABLE); + else + snd_soc_component_update_bits(component, + RK817_CODEC_DDAC_MUTE_MIXCTL, + DACMT_MASK, DACMT_DISABLE); + + return 0; +} + +#define RK817_PLAYBACK_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_CAPTURE_RATES (SNDRV_PCM_RATE_8000 |\ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define RK817_FORMATS (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops rk817_dai_ops = { + .hw_params = rk817_hw_params, + .set_fmt = rk817_set_dai_fmt, + .set_sysclk = rk817_set_dai_sysclk, + .mute_stream = rk817_digital_mute, + .no_capture_mute = 1, +}; + +static struct snd_soc_dai_driver rk817_dai[] = { + { + .name = "rk817-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 8, + .rates = RK817_PLAYBACK_RATES, + .formats = RK817_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RK817_CAPTURE_RATES, + .formats = RK817_FORMATS, + }, + .ops = &rk817_dai_ops, + }, +}; + +static int rk817_probe(struct snd_soc_component *component) +{ + struct rk817_codec_priv *rk817 = snd_soc_component_get_drvdata(component); + + snd_soc_component_init_regmap(component, rk817->regmap); + rk817->component = component; + + snd_soc_component_write(component, RK817_CODEC_DTOP_LPT_SRST, 0x40); + + rk817_init(component); + + /* setting initial pll values so that we can continue to leverage simple-audio-card. + * The values aren't important since no parameters are used. + */ + + snd_soc_component_set_pll(component, 0, 0, 0, 0); + + return 0; +} + +static void rk817_remove(struct snd_soc_component *component) +{ + snd_soc_component_exit_regmap(component); +} + +static const struct snd_soc_component_driver soc_codec_dev_rk817 = { + .probe = rk817_probe, + .remove = rk817_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, + .controls = rk817_volume_controls, + .num_controls = ARRAY_SIZE(rk817_volume_controls), + .dapm_routes = rk817_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rk817_dapm_routes), + .dapm_widgets = rk817_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rk817_dapm_widgets), + .set_pll = rk817_set_component_pll, +}; + +static int rk817_codec_parse_dt_property(struct device *dev, + struct rk817_codec_priv *rk817) +{ + struct device_node *node = dev->parent->of_node; + + if (!node) { + dev_err(dev, "%s() dev->parent->of_node is NULL\n", + __func__); + return -ENODEV; + } + + node = of_get_child_by_name(dev->parent->of_node, "codec"); + if (!node) { + dev_err(dev, "%s() Can not get child: codec\n", + __func__); + return -ENODEV; + } + + rk817->mic_in_differential = + of_property_read_bool(node, "mic-in-differential"); + + return 0; +} + +static const struct regmap_config rk817_codec_regmap_config = { + .name = "rk817-codec", + .reg_bits = 8, + .val_bits = 8, + .reg_stride = 1, + .max_register = 0x4f, + .cache_type = REGCACHE_NONE, + .volatile_reg = rk817_volatile_register, + .writeable_reg = rk817_codec_register, + .readable_reg = rk817_codec_register, + .reg_defaults = rk817_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rk817_reg_defaults), +}; + +static int rk817_platform_probe(struct platform_device *pdev) +{ + struct rk808 *rk817 = dev_get_drvdata(pdev->dev.parent); + struct rk817_codec_priv *rk817_codec_data; + int ret; + + if (!rk817) { + dev_err(&pdev->dev, "%s : rk817 is NULL\n", __func__); + return -EINVAL; + } + + rk817_codec_data = devm_kzalloc(&pdev->dev, + sizeof(struct rk817_codec_priv), + GFP_KERNEL); + if (!rk817_codec_data) + return -ENOMEM; + + platform_set_drvdata(pdev, rk817_codec_data); + + ret = rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); + if (ret < 0) { + dev_err(&pdev->dev, "%s() parse device tree property error %d\n", + __func__, ret); + goto err_; + } + + rk817_codec_data->regmap = devm_regmap_init_i2c(rk817->i2c, + &rk817_codec_regmap_config); + if (IS_ERR(rk817_codec_data->regmap)) { + ret = PTR_ERR(rk817_codec_data->regmap); + dev_err(&pdev->dev, "failed to allocate register map: %d\n", + ret); + goto err_; + } + + rk817_codec_data->mclk = devm_clk_get(&pdev->dev, "mclk"); + if (IS_ERR(rk817_codec_data->mclk)) { + dev_err(&pdev->dev, "Unable to get mclk\n"); + ret = -ENXIO; + goto err_; + } + + ret = clk_prepare_enable(rk817_codec_data->mclk); + if (ret < 0) { + dev_err(&pdev->dev, "%s() clock prepare error %d\n", + __func__, ret); + goto err_; + } + + ret = devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_rk817, + rk817_dai, ARRAY_SIZE(rk817_dai)); + if (ret < 0) { + dev_err(&pdev->dev, "%s() register codec error %d\n", + __func__, ret); + goto err_; + } + + return 0; +err_: + + return ret; +} + +static int rk817_platform_remove(struct platform_device *pdev) +{ + struct rk817_codec_priv *rk817 = platform_get_drvdata(pdev); + + clk_disable_unprepare(rk817->mclk); + + return 0; +} + +MODULE_DEVICE_TABLE(of, rk817_codec_dt_ids); + +static struct platform_driver rk817_codec_driver = { + .driver = { + .name = "rk817-codec", + }, + .probe = rk817_platform_probe, + .remove = rk817_platform_remove, +}; + +module_platform_driver(rk817_codec_driver); + +MODULE_DESCRIPTION("ASoC RK817 codec driver"); +MODULE_AUTHOR("binyuan <kevan.lan(a)rock-chips.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rk817_codec.h b/sound/soc/codecs/rk817_codec.h new file mode 100644 index 000000000000..f3bc59d498b4 --- /dev/null +++ b/sound/soc/codecs/rk817_codec.h @@ -0,0 +1,94 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * rk817 ALSA SoC Audio driver + * + Copyright (c) 2018, Fuzhou Rockchip Electronics Co., Ltd All rights reserved. + */ + +#ifndef __RK817_CODEC_H__ +#define __RK817_CODEC_H__ + +/* codec register */ +#define RK817_CODEC_BASE 0x0000 + +#define RK817_CODEC_DTOP_VUCTL (RK817_CODEC_BASE + 0x12) +#define RK817_CODEC_DTOP_VUCTIME (RK817_CODEC_BASE + 0x13) +#define RK817_CODEC_DTOP_LPT_SRST (RK817_CODEC_BASE + 0x14) +#define RK817_CODEC_DTOP_DIGEN_CLKE (RK817_CODEC_BASE + 0x15) +#define RK817_CODEC_AREF_RTCFG0 (RK817_CODEC_BASE + 0x16) +#define RK817_CODEC_AREF_RTCFG1 (RK817_CODEC_BASE + 0x17) +#define RK817_CODEC_AADC_CFG0 (RK817_CODEC_BASE + 0x18) +#define RK817_CODEC_AADC_CFG1 (RK817_CODEC_BASE + 0x19) +#define RK817_CODEC_DADC_VOLL (RK817_CODEC_BASE + 0x1a) +#define RK817_CODEC_DADC_VOLR (RK817_CODEC_BASE + 0x1b) +#define RK817_CODEC_DADC_SR_ACL0 (RK817_CODEC_BASE + 0x1e) +#define RK817_CODEC_DADC_ALC1 (RK817_CODEC_BASE + 0x1f) +#define RK817_CODEC_DADC_ALC2 (RK817_CODEC_BASE + 0x20) +#define RK817_CODEC_DADC_NG (RK817_CODEC_BASE + 0x21) +#define RK817_CODEC_DADC_HPF (RK817_CODEC_BASE + 0x22) +#define RK817_CODEC_DADC_RVOLL (RK817_CODEC_BASE + 0x23) +#define RK817_CODEC_DADC_RVOLR (RK817_CODEC_BASE + 0x24) +#define RK817_CODEC_AMIC_CFG0 (RK817_CODEC_BASE + 0x27) +#define RK817_CODEC_AMIC_CFG1 (RK817_CODEC_BASE + 0x28) +#define RK817_CODEC_DMIC_PGA_GAIN (RK817_CODEC_BASE + 0x29) +#define RK817_CODEC_DMIC_LMT1 (RK817_CODEC_BASE + 0x2a) +#define RK817_CODEC_DMIC_LMT2 (RK817_CODEC_BASE + 0x2b) +#define RK817_CODEC_DMIC_NG1 (RK817_CODEC_BASE + 0x2c) +#define RK817_CODEC_DMIC_NG2 (RK817_CODEC_BASE + 0x2d) +#define RK817_CODEC_ADAC_CFG0 (RK817_CODEC_BASE + 0x2e) +#define RK817_CODEC_ADAC_CFG1 (RK817_CODEC_BASE + 0x2f) +#define RK817_CODEC_DDAC_POPD_DACST (RK817_CODEC_BASE + 0x30) +#define RK817_CODEC_DDAC_VOLL (RK817_CODEC_BASE + 0x31) +#define RK817_CODEC_DDAC_VOLR (RK817_CODEC_BASE + 0x32) +#define RK817_CODEC_DDAC_SR_LMT0 (RK817_CODEC_BASE + 0x35) +#define RK817_CODEC_DDAC_LMT1 (RK817_CODEC_BASE + 0x36) +#define RK817_CODEC_DDAC_LMT2 (RK817_CODEC_BASE + 0x37) +#define RK817_CODEC_DDAC_MUTE_MIXCTL (RK817_CODEC_BASE + 0x38) +#define RK817_CODEC_DDAC_RVOLL (RK817_CODEC_BASE + 0x39) +#define RK817_CODEC_DDAC_RVOLR (RK817_CODEC_BASE + 0x3a) +#define RK817_CODEC_AHP_ANTI0 (RK817_CODEC_BASE + 0x3b) +#define RK817_CODEC_AHP_ANTI1 (RK817_CODEC_BASE + 0x3c) +#define RK817_CODEC_AHP_CFG0 (RK817_CODEC_BASE + 0x3d) +#define RK817_CODEC_AHP_CFG1 (RK817_CODEC_BASE + 0x3e) +#define RK817_CODEC_AHP_CP (RK817_CODEC_BASE + 0x3f) +#define RK817_CODEC_ACLASSD_CFG1 (RK817_CODEC_BASE + 0x40) +#define RK817_CODEC_ACLASSD_CFG2 (RK817_CODEC_BASE + 0x41) +#define RK817_CODEC_APLL_CFG0 (RK817_CODEC_BASE + 0x42) +#define RK817_CODEC_APLL_CFG1 (RK817_CODEC_BASE + 0x43) +#define RK817_CODEC_APLL_CFG2 (RK817_CODEC_BASE + 0x44) +#define RK817_CODEC_APLL_CFG3 (RK817_CODEC_BASE + 0x45) +#define RK817_CODEC_APLL_CFG4 (RK817_CODEC_BASE + 0x46) +#define RK817_CODEC_APLL_CFG5 (RK817_CODEC_BASE + 0x47) +#define RK817_CODEC_DI2S_CKM (RK817_CODEC_BASE + 0x48) +#define RK817_CODEC_DI2S_RSD (RK817_CODEC_BASE + 0x49) +#define RK817_CODEC_DI2S_RXCR1 (RK817_CODEC_BASE + 0x4a) +#define RK817_CODEC_DI2S_RXCR2 (RK817_CODEC_BASE + 0x4b) +#define RK817_CODEC_DI2S_RXCMD_TSD (RK817_CODEC_BASE + 0x4c) +#define RK817_CODEC_DI2S_TXCR1 (RK817_CODEC_BASE + 0x4d) +#define RK817_CODEC_DI2S_TXCR2 (RK817_CODEC_BASE + 0x4e) +#define RK817_CODEC_DI2S_TXCR3_TXCMD (RK817_CODEC_BASE + 0x4f) + +/* RK817_CODEC_DI2S_CKM */ +#define RK817_I2S_MODE_MASK (0x1 << 0) +#define RK817_I2S_MODE_MST (0x1 << 0) +#define RK817_I2S_MODE_SLV (0x0 << 0) + +/* RK817_CODEC_DDAC_MUTE_MIXCTL */ +#define DACMT_MASK (0x1 << 0) +#define DACMT_ENABLE (0x1 << 0) +#define DACMT_DISABLE (0x0 << 0) + +/* RK817_CODEC_DI2S_RXCR2 */ +#define VDW_RX_24BITS (0x17) +#define VDW_RX_16BITS (0x0f) + +/* RK817_CODEC_DI2S_TXCR2 */ +#define VDW_TX_24BITS (0x17) +#define VDW_TX_16BITS (0x0f) + +/* RK817_CODEC_AMIC_CFG0 */ +#define MIC_DIFF_MASK (0x1 << 7) +#define MIC_DIFF_DIS (0x0 << 7) +#define MIC_DIFF_EN (0x1 << 7) + +#endif /* __RK817_CODEC_H__ */ -- 2.25.1

2 7

[PATCH 1/2] ASoC: rt715: add main capture switch and main capture volume
by Jack Yu 19 Mar '21

19 Mar '21

Add main capture switch and main capture volume control. Main capture control has its own channel value respectivelly. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt715.c | 392 ++++++++++++++++++++++++++++++++------- 1 file changed, 323 insertions(+), 69 deletions(-) diff --git a/sound/soc/codecs/rt715.c b/sound/soc/codecs/rt715.c index f419d001a3fb..c461a4c23fd3 100644 --- a/sound/soc/codecs/rt715.c +++ b/sound/soc/codecs/rt715.c @@ -80,85 +80,102 @@ static int rt715_set_amp_gain_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mc = (struct soc_mixer_control *)kcontrol->private_value; struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; unsigned int addr_h, addr_l, val_h, val_ll, val_lr; - unsigned int read_ll, read_rl; - int i; - - /* Can't use update bit function, so read the original value first */ - addr_h = mc->reg; - addr_l = mc->rreg; - if (mc->shift == RT715_DIR_OUT_SFT) /* output */ - val_h = 0x80; - else /* input */ - val_h = 0x0; - - rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); - - /* L Channel */ - if (mc->invert) { - /* for mute */ - val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; - /* keep gain */ - read_ll = read_ll & 0x7f; - val_ll |= read_ll; - } else { - /* for gain */ - val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); - if (val_ll > mc->max) - val_ll = mc->max; - /* keep mute status */ - read_ll = read_ll & 0x80; - val_ll |= read_ll; - } - - /* R Channel */ - if (mc->invert) { - regmap_write(rt715->regmap, - RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); - /* for mute */ - val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; - /* keep gain */ - read_rl = read_rl & 0x7f; - val_lr |= read_rl; - } else { - /* for gain */ - val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); - if (val_lr > mc->max) - val_lr = mc->max; - /* keep mute status */ - read_rl = read_rl & 0x80; - val_lr |= read_rl; - } + unsigned int read_ll, read_rl, i, j, loop_cnt; - for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (strstr(ucontrol->id.name, "Main Capture Switch") || + strstr(ucontrol->id.name, "Main Capture Volume")) + loop_cnt = 4; + else + loop_cnt = 1; - if (val_ll == val_lr) { - /* Set both L/R channels at the same time */ - val_h = (1 << mc->shift) | (3 << 4); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_ll)); + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + if (loop_cnt == 1) { + addr_h = mc->reg; + addr_l = mc->rreg; } else { - /* Lch*/ - val_h = (1 << mc->shift) | (1 << 5); - regmap_write(rt715->regmap, addr_h, - (val_h << 8 | val_ll)); - /* Rch */ - val_h = (1 << mc->shift) | (1 << 4); - regmap_write(rt715->regmap, addr_l, - (val_h << 8 | val_lr)); + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; } - /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ val_h = 0x80; else /* input */ val_h = 0x0; - rt715_get_gain(rt715, addr_h, addr_l, val_h, - &read_rl, &read_ll); - if (read_rl == val_lr && read_ll == val_ll) - break; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + if (mc->invert) { + /* for mute */ + val_ll = (mc->max - ucontrol->value.integer.value[0]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + } else { + /* for gain */ + val_ll = ((ucontrol->value.integer.value[0]) & 0x7f); + if (val_ll > mc->max) + val_ll = mc->max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + } + + /* R Channel */ + if (mc->invert) { + /* for mute */ + val_lr = (mc->max - ucontrol->value.integer.value[1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + } else { + /* for gain */ + val_lr = ((ucontrol->value.integer.value[1]) & 0x7f); + if (val_lr > mc->max) + val_lr = mc->max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + } + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << mc->shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << mc->shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << mc->shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + /* check result */ + if (mc->shift == RT715_DIR_OUT_SFT) /* output */ + val_h = 0x80; + else /* input */ + val_h = 0x0; + + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } } /* D0:power on state, D3: power saving mode */ if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) @@ -201,9 +218,227 @@ static int rt715_set_amp_gain_get(struct snd_kcontrol *kcontrol, return 0; } +static int rt715_set_main_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + unsigned int k_shift = RT715_DIR_IN_SFT; + + for (j = 0; j < loop_cnt; j++) { + /* Can't use update bit function, so read the original value first */ + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for mute */ + val_ll = (!ucontrol->value.integer.value[j * 2]) << 7; + /* keep gain */ + val_ll |= read_ll & 0x7f; + + /* R Channel */ + /* for mute */ + val_lr = (!ucontrol->value.integer.value[j * 2 + 1]) << 7; + /* keep gain */ + val_lr |= read_rl & 0x7f; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_main_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = !(read_ll & 0x80); + ucontrol->value.integer.value[i * 2 + 1] = !(read_rl & 0x80); + } + + return 0; +} + +static int rt715_set_main_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, val_ll, val_lr; + unsigned int read_ll, read_rl, i, j, loop_cnt = 4; + unsigned int k_shift = RT715_DIR_IN_SFT, k_max = 0x3f; + + for (j = 0; j < loop_cnt; j++) { + addr_h = capture_reg_H[j]; + addr_l = capture_reg_L[j]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D0); + + /* L Channel */ + /* for gain */ + val_ll = ((ucontrol->value.integer.value[j * 2]) & 0x7f); + if (val_ll > k_max) + val_ll = k_max; + /* keep mute status */ + val_ll |= read_ll & 0x80; + + /* R Channel */ + /* for gain */ + val_lr = ((ucontrol->value.integer.value[j * 2 + 1]) & 0x7f); + if (val_lr > k_max) + val_lr = k_max; + /* keep mute status */ + val_lr |= read_rl & 0x80; + + for (i = 0; i < 3; i++) { /* retry 3 times at most */ + if (val_ll == val_lr) { + /* Set both L/R channels at the same time */ + val_h = (1 << k_shift) | (3 << 4); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_ll); + } else { + /* Lch*/ + val_h = (1 << k_shift) | (1 << 5); + regmap_write(rt715->regmap, addr_h, + (val_h << 8) | val_ll); + /* Rch */ + val_h = (1 << k_shift) | (1 << 4); + regmap_write(rt715->regmap, addr_l, + (val_h << 8) | val_lr); + } + val_h = 0x0; + rt715_get_gain(rt715, addr_h, addr_l, val_h, + &read_rl, &read_ll); + if (read_rl == val_lr && read_ll == val_ll) + break; + } + } + /* D0:power on state, D3: power saving mode */ + if (dapm->bias_level <= SND_SOC_BIAS_STANDBY) + regmap_write(rt715->regmap, + RT715_SET_AUDIO_POWER_STATE, AC_PWRST_D3); + return 0; +} + +static int rt715_set_main_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt715_priv *rt715 = snd_soc_component_get_drvdata(component); + unsigned int capture_reg_H[] = {RT715_SET_GAIN_MIC_ADC_H, + RT715_SET_GAIN_LINE_ADC_H, RT715_SET_GAIN_MIX_ADC_H, + RT715_SET_GAIN_MIX_ADC2_H}; + unsigned int capture_reg_L[] = {RT715_SET_GAIN_MIC_ADC_L, + RT715_SET_GAIN_LINE_ADC_L, RT715_SET_GAIN_MIX_ADC_L, + RT715_SET_GAIN_MIX_ADC2_L}; + unsigned int addr_h, addr_l, val_h = 0x0, i, loop_cnt = 4; + unsigned int read_ll, read_rl; + + for (i = 0; i < loop_cnt; i++) { + addr_h = capture_reg_H[i]; + addr_l = capture_reg_L[i]; + rt715_get_gain(rt715, addr_h, addr_l, val_h, &read_rl, &read_ll); + + ucontrol->value.integer.value[i * 2] = read_ll & 0x7f; + ucontrol->value.integer.value[i * 2 + 1] = read_rl & 0x7f; + } + + return 0; +} + static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); +static int rt715_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int rt715_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 8; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 0x3f; + return 0; +} + #define SOC_DOUBLE_R_EXT(xname, reg_left, reg_right, xshift, xmax, xinvert,\ xhandler_get, xhandler_put) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -212,6 +447,21 @@ static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define RT715_MAIN_SWITCH_EXT(xname, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt715_switch_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + +#define RT715_MAIN_VOL_EXT_TLV(xname, xhandler_get, xhandler_put, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt715_vol_info, \ + .get = xhandler_get, .put = xhandler_put, \ +} + static const struct snd_kcontrol_new rt715_snd_controls[] = { /* Capture switch */ SOC_DOUBLE_R_EXT("ADC 07 Capture Switch", RT715_SET_GAIN_MIC_ADC_H, @@ -226,6 +476,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { SOC_DOUBLE_R_EXT("ADC 27 Capture Switch", RT715_SET_GAIN_MIX_ADC2_H, RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 1, 1, rt715_set_amp_gain_get, rt715_set_amp_gain_put), + RT715_MAIN_SWITCH_EXT("Main Capture Switch", + rt715_set_main_switch_get, rt715_set_main_switch_put), /* Volume Control */ SOC_DOUBLE_R_EXT_TLV("ADC 07 Capture Volume", RT715_SET_GAIN_MIC_ADC_H, RT715_SET_GAIN_MIC_ADC_L, RT715_DIR_IN_SFT, 0x3f, 0, @@ -243,6 +495,8 @@ static const struct snd_kcontrol_new rt715_snd_controls[] = { RT715_SET_GAIN_MIX_ADC2_L, RT715_DIR_IN_SFT, 0x3f, 0, rt715_set_amp_gain_get, rt715_set_amp_gain_put, in_vol_tlv), + RT715_MAIN_VOL_EXT_TLV("Main Capture Volume", + rt715_set_main_vol_get, rt715_set_main_vol_put, in_vol_tlv), /* MIC Boost Control */ SOC_DOUBLE_R_EXT_TLV("DMIC1 Boost", RT715_SET_GAIN_DMIC1_H, RT715_SET_GAIN_DMIC1_L, RT715_DIR_IN_SFT, 3, 0, -- 2.29.0

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aplay: avoid any further PCM writing if in abort
by GitHub pull_request - opened 19 Mar '21

19 Mar '21

alsa-project/alsa-utils pull request #81 was opened from aditpape: Terminating stream with CTRL-C will set in_aborting flag which is used to leave any write/read loop on the ALSA device. After ending the read/write loop aplay tries to drain the stream which is not required and can also lead to malfunctions: -If user interrupts a blocked/non responsive PCM (e.g. usb uac2 gadget which does not consume data due to stream stopped by host) it will successfully terminate the write loop but will hang again in drain call. This would require to hit CTRL-C again to unblock which should be avoided. Aplay currently anyhow allows signal handler to get invoked only once. Signed-off-by: Andreas Pape <apape(a)de.adit-jv.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/81 Patch URL : https://github.com/alsa-project/alsa-utils/pull/81.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ALSA: hda: Fix spelling mistakes
by huangjianghui 19 Mar '21

19 Mar '21

Signed-off-by: huangjianghui <huangjianghui(a)uniontech.com> --- sound/hda/hdac_stream.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c index a6ed3dc35f7e..1eb8563db2df 100644 --- a/sound/hda/hdac_stream.c +++ b/sound/hda/hdac_stream.c @@ -618,7 +618,7 @@ void snd_hdac_stream_sync_trigger(struct hdac_stream *azx_dev, bool set, EXPORT_SYMBOL_GPL(snd_hdac_stream_sync_trigger); /** - * snd_hdac_stream_sync - sync with start/strop trigger operation + * snd_hdac_stream_sync - sync with start/stop trigger operation * @azx_dev: HD-audio core stream (master stream) * @start: true = start, false = stop * @streams: bit flags of streams to sync -- 2.20.1

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[PATCH] ALSA: hda/realtek: add more delay time before determine_headset_type
by Hui Wang 19 Mar '21

19 Mar '21

We found a recording issue on the headset-mic recently, sometimes users plug a headset and select headset-mic from UI, but can't record any sound from headset-mic. The root cause is the determine_headset_type() returns a wrong type, e.g. users plug a ctia type headset, but that function returns omtp type. In the past, determine_headset_type() worked well because the internal mic is connected to the codec, so the "Input Source" or "Capture Source" is internal mic by default when users plug a headset, the determine_headset_type() will not be called unless users select headset-mic from UI, when users select headset-mic, the plugging action already finished and the headset is completely plugged into the jack, so determine_headset_type() could return a correct type. But more and more machines connect the internal mic to the PCH now, and the "Input Source" is headset mic by default, when users plug a headset, the determine_headset_type() will be called immediately, if the headset is not completely plugged in, it will return a wrong type. Here add 2s delay before calling determine_headset_type(), and since there is a pop-up dialogue when users do plugging action, to avoid freezing the UI, use the deleyed_work to call that function. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 31 ++++++++++++++++++++++++------- 1 file changed, 24 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b47504fa8dfd..1f6fc8addf3e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -127,6 +127,8 @@ struct alc_spec { unsigned int coef0; struct input_dev *kb_dev; u8 alc_mute_keycode_map[1]; + struct hda_codec *codec; + struct delayed_work headset_set_work; }; /* @@ -1160,6 +1162,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) kfree(spec); return err; } + spec->codec = codec; return 0; } @@ -5363,6 +5366,21 @@ static void alc_determine_headset_type(struct hda_codec *codec) spec->current_headset_type = is_ctia ? ALC_HEADSET_TYPE_CTIA : ALC_HEADSET_TYPE_OMTP; } +static void alc_headset_check_and_set(struct work_struct *work) +{ + struct alc_spec *spec = container_of(work, struct alc_spec, + headset_set_work.work); + struct hda_codec *codec = spec->codec; + + if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) + alc_determine_headset_type(codec); + if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) + alc_headset_mode_ctia(codec); + else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) + alc_headset_mode_omtp(codec); + spec->gen.hp_jack_present = true; +} + static void alc_update_headset_mode(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -5386,6 +5404,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) return; } + cancel_delayed_work_sync(&spec->headset_set_work); switch (new_headset_mode) { case ALC_HEADSET_MODE_UNPLUGGED: alc_headset_mode_unplugged(codec); @@ -5394,13 +5413,7 @@ static void alc_update_headset_mode(struct hda_codec *codec) spec->gen.hp_jack_present = false; break; case ALC_HEADSET_MODE_HEADSET: - if (spec->current_headset_type == ALC_HEADSET_TYPE_UNKNOWN) - alc_determine_headset_type(codec); - if (spec->current_headset_type == ALC_HEADSET_TYPE_CTIA) - alc_headset_mode_ctia(codec); - else if (spec->current_headset_type == ALC_HEADSET_TYPE_OMTP) - alc_headset_mode_omtp(codec); - spec->gen.hp_jack_present = true; + schedule_delayed_work(&spec->headset_set_work, msecs_to_jiffies(2000)); break; case ALC_HEADSET_MODE_MIC: alc_headset_mode_mic_in(codec, hp_pin, spec->headphone_mic_pin); @@ -5466,6 +5479,7 @@ static void alc_fixup_headset_mode(struct hda_codec *codec, spec->parse_flags |= HDA_PINCFG_HEADSET_MIC | HDA_PINCFG_HEADPHONE_MIC; break; case HDA_FIXUP_ACT_PROBE: + INIT_DELAYED_WORK(&spec->headset_set_work, alc_headset_check_and_set); alc_probe_headset_mode(codec); break; case HDA_FIXUP_ACT_INIT: @@ -5475,6 +5489,9 @@ static void alc_fixup_headset_mode(struct hda_codec *codec, } alc_update_headset_mode(codec); break; + case HDA_FIXUP_ACT_FREE: + cancel_delayed_work_sync(&spec->headset_set_work); + break; } } -- 2.25.1

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Very unstable and broken after audio device switching, reboots, multimedia app usage etc etc etc
by GitHub issues - edited 19 Mar '21

19 Mar '21

alsa-project/alsa-lib issue #127 was edited from oliiix: To me this seems to be a huge issue for years now. Running Ubuntu 20.10 very latest. I'm using an Astro A50 and two HDMI Displays for playing audio plus there's two more outputs I'm not really using. But as soon as the computer ran a little long (yes, that's actually enough to cause problems, probably due to certain apps like Unreal engine and games!), came out of sleep or I switch playback devices (which I'm doing very often and that shouldnt be ANY issue), things get seriously mixed up. Suddenly the routing goes all wrong (i select one output device, another plays instead, or just no sound at all). Sometimes "no sound" problem gets fixed by just switching devices about 10-20! times until sound is back (it either works or doesn't but that it starts working again after ten times switching devices is very weird behavior). Reinstalling as well as resetting alsa doesn't do anything to remedy no sound, restarting computer doesn't fix sound problems, only switching devices ten times does! Alsa Mixer doesn't show any controls for my A50, yet it often works even if some profiles are to low volume and the one that has good volume there I can't hear audio that comes from behind me (I have the no sound issue also on my HDMI outputs, just saying). Pavucontrol on the other hand lists my A50 but doesn't show one of my HDMI displays. I mean what is going wrong here (for years)??? I mean I found forum posts about these issues that reach back years in time -.- When will we have proper working audio where audio device switching on linux is possible without any problems and the sound doesn't just "give up" after some time? maybe someone should create one all including audio software system structure instead of several packages that are unable to work together properly... can't be that two apps (pavucontrol and pulseaudio) that work on top on alsa show different devices and settings to begin with Issue URL : https://github.com/alsa-project/alsa-lib/issues/127 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Very unstable and broken after audio device switching, reboots, multimedia app usage etc etc etc
by GitHub issues - opened 19 Mar '21

19 Mar '21

alsa-project/alsa-lib issue #127 was opened from oliiix: To me this seems to be a huge issue for years now. Running Ubuntu 20.10 very latest. I'm using an Astro A50 and two HDMI Displays for playing audio plus there's two more outputs I'm not really using. But as soon as the computer ran a little long (yes, that's actually enough to cause problems, probably due to certain apps like Unreal engine and games!), came out of sleep or I switch playback devices (which I'm doing very often and that shouldnt be ANY issue), things get seriously mixed up. Suddenly the routing goes all wrong (i select one output device, another plays instead, or just no sound at all). Sometimes "no sound" problem gets fixed by just switching devices about 10-20! times until sound is back (it either works or doesn't but that it starts working again after ten times switching devices is very weird behavior). Reinstalling as well as resetting alsa doesn't do anything to remedy no sound, restarting computer doesn't fix sound problems, only switching devices ten times does! Alsa Mixer doesn't show any controls for my A50, yet it often works even if some profiles are to low volume and the one that has good volume there I can't hear audio that comes from behind me (I have the no sound issue also on my HDMI outputs, just saying). Pavucontrol on the other hand lists my A50 but doesn't show one of my HDMI displays. I mean what is going wrong here (for years)??? I mean I found forum posts about these issues that reach back years in time -.- When will we have proper working audio where audio device switching on linux is possible without any problems and the sound doesn't just "give up" after some time? Issue URL : https://github.com/alsa-project/alsa-lib/issues/127 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: fsl_sai: remove reset code from dai_probe
by Shengjiu Wang 19 Mar '21

19 Mar '21

From: Viorel Suman <viorel.suman(a)nxp.com> SAI software reset is done in runtime resume, there is no need to do it in fsl_sai_dai_probe. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 6ef2ce348232..9e7893f91882 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -711,13 +711,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); unsigned int ofs = sai->soc_data->reg_offset; - /* Software Reset for both Tx and Rx */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), FSL_SAI_CSR_SR); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), FSL_SAI_CSR_SR); - /* Clear SR bit to finish the reset */ - regmap_write(sai->regmap, FSL_SAI_TCSR(ofs), 0); - regmap_write(sai->regmap, FSL_SAI_RCSR(ofs), 0); - regmap_update_bits(sai->regmap, FSL_SAI_TCR1(ofs), FSL_SAI_CR1_RFW_MASK(sai->soc_data->fifo_depth), sai->soc_data->fifo_depth - FSL_SAI_MAXBURST_TX); -- 2.27.0

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[PATCH] docs: devicetree: bindings: sound: Fix a typo in the file rt5682.txt
by Bhaskar Chowdhury 18 Mar '21

18 Mar '21

s/drving/driving/ Signed-off-by: Bhaskar Chowdhury <unixbhaskar(a)gmail.com> --- Documentation/devicetree/bindings/sound/rt5682.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rt5682.txt b/Documentation/devicetree/bindings/sound/rt5682.txt index 9c5fadb6ac82..cd8c53d8497e 100644 --- a/Documentation/devicetree/bindings/sound/rt5682.txt +++ b/Documentation/devicetree/bindings/sound/rt5682.txt @@ -44,7 +44,7 @@ Optional properties: - realtek,dmic-delay-ms : Set the delay time (ms) for the requirement of the particular DMIC. -- realtek,dmic-clk-driving-high : Set the high drving of the DMIC clock out. +- realtek,dmic-clk-driving-high : Set the high driving of the DMIC clock out. Pins on the device (for linking into audio routes) for RT5682: -- 2.26.2

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