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[alsa-devel] snd_pcm_open() leaks error
by Joël Krähemann 25 Jun '16

25 Jun '16

Hi all I have some issues as opening soundcard multiple times the error is handled but it leaks memory: http://pastebin.com/aQfAQV2q The problematic function is snd_pcm_open() but I think it doesn't have any performance impact on ̀gsequencer` http://www.nongnu.org/gsequencer/ Bests, Joël -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

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[alsa-devel] [PATCH v8 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by Paul.Handrigan＠cirrus.com 24 Jun '16

24 Jun '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 48 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1314 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1590 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..b6eadce --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,48 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 60f8f13..02d2084 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -391,6 +392,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1264fa4..abf2126 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -37,6 +37,7 @@ snd-soc-arizona-objs := arizona.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -257,6 +258,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..55c1f75 --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1314 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_set_bst_ipk(struct snd_soc_codec *codec, unsigned int bst) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + int ret = 0, steps = 0; + + /* Boost current in uA */ + if (bst > 3600000 || bst < 1850000) { + dev_err(codec->dev, "Invalid boost current %d\n", bst); + ret = -EINVAL; + goto err; + } + + if (bst % 15625) { + dev_err(codec->dev, "Current not a multiple of 15625uA (%d)\n", + bst); + ret = -EINVAL; + goto err; + } + + while (bst > 1850000) { + bst -= 15625; + steps++; + } + + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + steps+0x70); + +err: + return ret; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + cs35l33_set_bst_ipk(codec, cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "cirrus,hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "cirrus,mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "cirrus,release-rate", + &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "cirrus,ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "cirrus,ldo-path-disable", + &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "cirrus,ldo-entry-delay", + &val32) >= 0) + hg_config->ldo_entry_delay = val32; + + hg_config->vp_hg_auto = of_property_read_bool(hg, + "cirrus,vp-hg-auto"); + + if (of_property_read_u32(hg, "cirrus,vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "cirrus,vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + return IRQ_NONE; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "cirrus,boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "cirrus,ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "cirrus,boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "cirrus,imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = devm_request_threaded_irq(&i2c_client->dev, i2c_client->irq, NULL, + cs35l33_irq_thread, IRQF_ONESHOT | IRQF_TRIGGER_LOW, + "cs35l33", cs35l33); + if (ret != 0) + dev_warn(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..c045737 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix the headset mic jack detection on Dell machine
by woodrow.shen＠canonical.com 24 Jun '16

24 Jun '16

From: Woodrow Shen <woodrow.shen(a)gmail.com> The new Dell laptop with codec 3246 can't detect headset mic when headset was inserted on the machine. So adding pin configurations into quirk table makes headset mic work correctly. Codec: Realtek ALC3246 Vendor Id: 0x10ec0256 Subsystem Id: 0x10280781 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1f652ca1..ac0d5bf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5847,6 +5847,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170120}, {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x12, 0x90a60170}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), + SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, ALC256_STANDARD_PINS), SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4, {0x12, 0x90a60130}, -- 2.7.4

3 2

[alsa-devel] sound: use-after-free in hrtimer_cancel
by Dmitry Vyukov 24 Jun '16

24 Jun '16

Hello, The following program triggers use-after-free: // autogenerated by syzkaller (http://github.com/google/syzkaller) #include <pthread.h> #include <stdint.h> #include <string.h> #include <sys/syscall.h> #include <unistd.h> long r[20]; void* thr(void* arg) { switch ((long)arg) { case 0: break; case 1: r[2] = syscall(SYS_open, "/dev/audio", 0xa40ul, 0, 0, 0); break; case 2: *(uint32_t*)0x2001dde8 = r[2]; *(uint16_t*)0x2001ddec = (uint16_t)0x0; *(uint16_t*)0x2001ddee = (uint16_t)0x2; *(uint32_t*)0x2001ddf0 = r[2]; *(uint16_t*)0x2001ddf4 = (uint16_t)0xfffffffff3a15aea; *(uint16_t*)0x2001ddf6 = (uint16_t)0x1; *(uint32_t*)0x2001ddf8 = r[2]; *(uint16_t*)0x2001ddfc = (uint16_t)0x836a; *(uint16_t*)0x2001ddfe = (uint16_t)0x1ff; *(uint32_t*)0x2001de00 = r[2]; *(uint16_t*)0x2001de04 = (uint16_t)0x0; *(uint16_t*)0x2001de06 = (uint16_t)0x20; r[15] = syscall(SYS_poll, 0x2001dde8ul, 0x4ul, 0x8ul, 0, 0, 0); break; case 3: *(uint64_t*)0x20dc13c0 = (uint64_t)0x20dc1f27; *(uint64_t*)0x20dc13c8 = (uint64_t)0xd9; r[18] = syscall(SYS_readv, r[2], 0x20dc13c0ul, 0x1ul, 0, 0, 0); break; case 4: r[19] = syscall(SYS_read, r[2], 0x20dbefe0ul, 0x20ul, 0, 0, 0); break; } return 0; } int main() { long i; pthread_t th[10]; syscall(SYS_mmap, 0x20000000ul, 0xde0000ul, 0x3ul, 0x32ul, 0xfffffffffffffffful, 0x0ul); srand(getpid()); memset(r, -1, sizeof(r)); for (i = 0; i < 5; i++) { pthread_create(&th[i], 0, thr, (void*)i); usleep(rand()%5000); } for (i = 0; i < 5; i++) { pthread_create(&th[5+i], 0, thr, (void*)i); if (rand()%2) usleep(rand()%5000); } for (i = 0; i < 10; i++) { pthread_join(th[i], 0); } return 0; } Here are 2 reports (note they are slightly different): ================================================================== BUG: KASAN: use-after-free in rb_erase+0x1b17/0x2010 at addr ffff88005e5b6f68 Read of size 8 by task syz-executor/8984 ============================================================================= BUG kmalloc-192 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446705582212484632 cpu=2172593693 pid=-1 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] ___slab_alloc+0x55d/0x5a0 mm/slub.c:2476 [< none >] __slab_alloc+0x68/0xc0 mm/slub.c:2505 [< inline >] slab_alloc_node mm/slub.c:2568 [< inline >] slab_alloc mm/slub.c:2610 [< none >] kmem_cache_alloc_trace+0x263/0x3d0 mm/slub.c:2627 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] dummy_pcm_open+0xcd/0x5e0 sound/drivers/dummy.c:574 [< none >] snd_pcm_open_substream+0x188/0x430 sound/core/pcm_native.c:2276 [< inline >] snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2346 [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1110 sound/core/oss/pcm_oss.c:2428 [< none >] snd_pcm_oss_open+0x35/0x50 sound/core/oss/pcm_oss.c:2392 [< none >] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3238 [< none >] path_openat+0x51bb/0x5ce0 fs/namei.c:3374 [< none >] do_filp_open+0x18e/0x250 fs/namei.c:3409 [< none >] do_sys_open+0x1fc/0x420 fs/open.c:1020 INFO: Freed in 0xfffd8e09 age=18446705496313138713 cpu=2164287125 pid=-1 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] __slab_free+0x1e8/0x300 mm/slub.c:2687 [< inline >] slab_free mm/slub.c:2840 [< none >] kfree+0x2fc/0x370 mm/slub.c:3691 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] dummy_pcm_close+0x9c/0xd0 sound/drivers/dummy.c:607 [< none >] snd_pcm_release_substream.part.37+0x169/0x2f0 sound/core/pcm_native.c:2241 [< none >] snd_pcm_release_substream+0x59/0x70 sound/core/pcm_native.c:2251 [< none >] snd_pcm_oss_release_file+0x7b/0xb0 sound/core/oss/pcm_oss.c:2305 [< none >] snd_pcm_oss_release+0xfa/0x280 sound/core/oss/pcm_oss.c:2485 [< none >] __fput+0x236/0x780 fs/file_table.c:208 [< none >] ____fput+0x15/0x20 fs/file_table.c:244 [< none >] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [< none >] do_exit+0x874/0x2d80 kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 kernel/exit.c:878 [< none >] get_signal+0x634/0x15e0 kernel/signal.c:2307 [< none >] do_signal+0x7f/0x1cf0 arch/x86/kernel/signal.c:784 INFO: Indirect in 0xfffd8e17 age=4294807066 cpu=0 pid=0 INFO: Slab 0xffffea0001796d00 objects=24 used=15 fp=0xffff88005e5b71e8 flags=0x4fffe0000004080 INFO: Object 0xffff88005e5b6f40 @offset=12096 fp=0xbbbbbbbbbbbbbbbb CPU: 1 PID: 8984 Comm: syz-executor Tainted: G B 4.6.0-rc6+ #355 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff880059fd7378 ffffffff82c9062f ffffffff5e5b4000 fffffbfff0fd64b8 ffff88003e804d40 ffff88005e5b6f40 ffffea0001796d00 ffff88005e5b4000 0000000000000000 ffff880059fd73a8 ffffffff81793e1d Call Trace: [<ffffffff8179e59e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:333 [< inline >] rb_set_parent include/linux/rbtree_augmented.h:111 [< inline >] __rb_erase_augmented include/linux/rbtree_augmented.h:218 [<ffffffff82ca5787>] rb_erase+0x1b17/0x2010 lib/rbtree.c:427 [<ffffffff82cb02e8>] timerqueue_del+0x78/0x170 lib/timerqueue.c:86 [<ffffffff814d0c80>] __remove_hrtimer+0x90/0x220 kernel/time/hrtimer.c:903 [< inline >] remove_hrtimer kernel/time/hrtimer.c:945 [<ffffffff814d23da>] hrtimer_try_to_cancel+0x22a/0x570 kernel/time/hrtimer.c:1046 [<ffffffff814d2742>] hrtimer_cancel+0x22/0x40 kernel/time/hrtimer.c:1066 [<ffffffff85420531>] dummy_hrtimer_stop+0x91/0xb0 sound/drivers/dummy.c:417 [<ffffffff854228bf>] dummy_pcm_trigger+0x17f/0x1e0 sound/drivers/dummy.c:507 [<ffffffff85392170>] snd_pcm_do_stop+0x160/0x1b0 sound/core/pcm_native.c:1106 [<ffffffff85391b26>] snd_pcm_action_single+0x76/0x120 sound/core/pcm_native.c:956 [<ffffffff85391e01>] snd_pcm_action+0x231/0x290 sound/core/pcm_native.c:974 [< inline >] snd_pcm_stop sound/core/pcm_native.c:1139 [<ffffffff8539754d>] snd_pcm_drop+0x12d/0x1d0 sound/core/pcm_native.c:1784 [<ffffffff8539d3be>] snd_pcm_common_ioctl1+0xfae/0x2150 sound/core/pcm_native.c:2805 [<ffffffff8539ee91>] snd_pcm_capture_ioctl1+0x2a1/0x5e0 sound/core/pcm_native.c:2976 [<ffffffff8539f2ec>] snd_pcm_kernel_ioctl+0x11c/0x160 sound/core/pcm_native.c:3020 [<ffffffff853d9a44>] snd_pcm_oss_sync+0x3a4/0xa30 sound/core/oss/pcm_oss.c:1693 [<ffffffff853da27d>] snd_pcm_oss_release+0x1ad/0x280 sound/core/oss/pcm_oss.c:2483 [<ffffffff817fc066>] __fput+0x236/0x780 fs/file_table.c:208 [<ffffffff817fc635>] ____fput+0x15/0x20 fs/file_table.c:244 [<ffffffff813c91d0>] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [<ffffffff81373e64>] do_exit+0x874/0x2d80 kernel/exit.c:748 [<ffffffff813764e8>] do_group_exit+0x108/0x330 kernel/exit.c:878 [<ffffffff81399674>] get_signal+0x634/0x15e0 kernel/signal.c:2307 [<ffffffff811f40af>] do_signal+0x7f/0x1cf0 arch/x86/kernel/signal.c:784 [<ffffffff81006695>] exit_to_usermode_loop+0x1a5/0x210 arch/x86/entry/common.c:229 [< inline >] prepare_exit_to_usermode arch/x86/entry/common.c:264 [<ffffffff8100868f>] syscall_return_slowpath+0x2bf/0x340 arch/x86/entry/common.c:329 [<ffffffff867c56dc>] entry_SYSCALL_64_fastpath+0xbf/0xc1 arch/x86/entry/entry_64.S:241 ================================================================== ================================================================== BUG: KASAN: use-after-free in timerqueue_add+0x29a/0x2a0 at addr ffff880062134580 Read of size 8 by task a.out/12711 ============================================================================= BUG kmalloc-192 (Not tainted): kasan: bad access detected ----------------------------------------------------------------------------- Disabling lock debugging due to kernel taint INFO: Allocated in 0xbbbbbbbbbbbbbbbb age=18446689703718641197 cpu=2172593693 pid=-1 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] ___slab_alloc+0x55d/0x5a0 mm/slub.c:2476 [< none >] __slab_alloc+0x68/0xc0 mm/slub.c:2505 [< inline >] slab_alloc_node mm/slub.c:2568 [< inline >] slab_alloc mm/slub.c:2610 [< none >] kmem_cache_alloc_trace+0x263/0x3d0 mm/slub.c:2627 [< inline >] kmalloc include/linux/slab.h:478 [< inline >] kzalloc include/linux/slab.h:622 [< none >] dummy_hrtimer_create+0x49/0x1a0 sound/drivers/dummy.c:464 [< none >] dummy_pcm_open+0xcd/0x5e0 sound/drivers/dummy.c:574 [< none >] snd_pcm_open_substream+0x188/0x430 sound/core/pcm_native.c:2276 [< inline >] snd_pcm_oss_open_file sound/core/oss/pcm_oss.c:2346 [< none >] snd_pcm_oss_open.part.17+0x5a4/0x1110 sound/core/oss/pcm_oss.c:2428 [< none >] snd_pcm_oss_open+0x35/0x50 sound/core/oss/pcm_oss.c:2392 [< none >] soundcore_open+0x30f/0x640 sound/sound_core.c:639 [< none >] chrdev_open+0x22a/0x4c0 fs/char_dev.c:388 [< none >] do_dentry_open+0x6a2/0xcb0 fs/open.c:736 [< none >] vfs_open+0x17b/0x1f0 fs/open.c:853 [< inline >] do_last fs/namei.c:3238 [< none >] path_openat+0x51bb/0x5ce0 fs/namei.c:3374 [< none >] do_filp_open+0x18e/0x250 fs/namei.c:3409 [< none >] do_sys_open+0x1fc/0x420 fs/open.c:1020 INFO: Freed in 0x100015e1b age=18446689836862627373 cpu=0 pid=0 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] __slab_free+0x1e8/0x300 mm/slub.c:2687 [< inline >] slab_free mm/slub.c:2840 [< none >] kfree+0x2fc/0x370 mm/slub.c:3691 [< none >] dummy_hrtimer_free+0x68/0x80 sound/drivers/dummy.c:481 [< none >] dummy_pcm_close+0x9c/0xd0 sound/drivers/dummy.c:607 [< none >] snd_pcm_release_substream.part.37+0x169/0x2f0 sound/core/pcm_native.c:2241 [< none >] snd_pcm_release_substream+0x59/0x70 sound/core/pcm_native.c:2251 [< none >] snd_pcm_oss_release_file+0x7b/0xb0 sound/core/oss/pcm_oss.c:2305 [< none >] snd_pcm_oss_release+0xfa/0x280 sound/core/oss/pcm_oss.c:2485 [< none >] __fput+0x236/0x780 fs/file_table.c:208 [< none >] ____fput+0x15/0x20 fs/file_table.c:244 [< none >] task_work_run+0x170/0x210 kernel/task_work.c:115 [< inline >] exit_task_work include/linux/task_work.h:21 [< none >] do_exit+0x874/0x2d80 kernel/exit.c:748 [< none >] do_group_exit+0x108/0x330 kernel/exit.c:878 [< inline >] SYSC_exit_group kernel/exit.c:889 [< none >] SyS_exit_group+0x1d/0x20 kernel/exit.c:887 [< none >] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 INFO: Indirect in 0x100015e2e age=4295056943 cpu=0 pid=0 INFO: Slab 0xffffea0001884d00 objects=24 used=10 fp=0xffff880062135a48 flags=0x4fffe0000004080 INFO: Object 0xffff880062134540 @offset=1344 fp=0xbbbbbbbbbbbbbbbb CPU: 2 PID: 12711 Comm: a.out Tainted: G B 4.6.0-rc6+ #355 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88006d407c60 ffffffff82c9062f ffffffff62134000 fffffbfff0fd64b8 ffff88003e804d40 ffff880062134540 ffffea0001884d00 ffff880062134000 ffff88006d417f10 ffff88006d407c90 ffffffff81793e1d Call Trace: [<ffffffff8179e59e>] __asan_report_load8_noabort+0x3e/0x40 mm/kasan/report.c:333 [<ffffffff82cb022a>] timerqueue_add+0x29a/0x2a0 lib/timerqueue.c:51 [<ffffffff814d1066>] enqueue_hrtimer+0x116/0x3d0 kernel/time/hrtimer.c:879 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1257 [<ffffffff814d1c5e>] __hrtimer_run_queues+0x93e/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d4022>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa62>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c7f59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c62ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [<ffffffff813c93d3>] __kernel_text_address+0x73/0xa0 kernel/extable.c:103 [<ffffffff811fd73e>] print_context_stack+0x6e/0xc0 arch/x86/kernel/dumpstack.c:107 [<ffffffff811fc984>] dump_trace+0x124/0x320 arch/x86/kernel/dumpstack_64.c:243 [<ffffffff8121d396>] save_stack_trace+0x26/0x50 arch/x86/kernel/stacktrace.c:67 [< inline >] __save_stack_trace mm/kmemleak.c:530 [<ffffffff817e2467>] create_object+0x137/0x2d0 mm/kmemleak.c:585 [<ffffffff867a7d43>] kmemleak_alloc+0x63/0xa0 mm/kmemleak.c:915 [< inline >] kmemleak_alloc_recursive include/linux/kmemleak.h:47 [< inline >] slab_post_alloc_hook mm/slab.h:406 [< inline >] slab_alloc_node mm/slub.c:2602 [< inline >] slab_alloc mm/slub.c:2610 [<ffffffff81798236>] kmem_cache_alloc+0x166/0x3c0 mm/slub.c:2615 [<ffffffff81727df0>] ptlock_alloc+0x20/0x80 mm/memory.c:3968 [< inline >] ptlock_init include/linux/mm.h:1625 [< inline >] pgtable_page_ctor include/linux/mm.h:1659 [<ffffffff81299ba9>] pte_alloc_one+0x59/0x100 arch/x86/mm/pgtable.c:31 [<ffffffff81716c68>] __pte_alloc+0x28/0x2a0 mm/memory.c:569 [< inline >] __handle_mm_fault mm/memory.c:3470 [<ffffffff81721a6a>] handle_mm_fault+0xc0a/0x11a0 mm/memory.c:3522 [<ffffffff8128ad47>] __do_page_fault+0x457/0xbb0 arch/x86/mm/fault.c:1351 [<ffffffff8128b5ef>] trace_do_page_fault+0xdf/0x5b0 arch/x86/mm/fault.c:1444 [<ffffffff8127bd84>] do_async_page_fault+0x14/0xd0 arch/x86/kernel/kvm.c:265 [<ffffffff867c77f8>] async_page_fault+0x28/0x30 arch/x86/entry/entry_64.S:920 ================================================================== On commit 83858a701cf3271f81dd321c2a81e5666c6ca8f4 (note it is not super fresh: May 3).

2 12

[alsa-devel] [PATCH] ALSA: hda / realtek - add two more Thinkpad IDs (5050, 5053) for tpt460 fixup
by Jaroslav Kysela 24 Jun '16

24 Jun '16

See: https://bugzilla.redhat.com/show_bug.cgi?id=1349539 See: https://bugzilla.kernel.org/show_bug.cgi?id=120961 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac0d5bf..999620d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5665,6 +5665,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x503c, "Thinkpad L450", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504a, "ThinkPad X260", ALC292_FIXUP_TPT440_DOCK), SND_PCI_QUIRK(0x17aa, 0x504b, "Thinkpad", ALC293_FIXUP_LENOVO_SPK_NOISE), + SND_PCI_QUIRK(0x17aa, 0x5050, "Thinkpad T560p", ALC292_FIXUP_TPT460), + SND_PCI_QUIRK(0x17aa, 0x5053, "Thinkpad T460", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 2.5.5

2 1

Re: [alsa-devel] [PATCH 6/7] of_graph: add of_graph_get_top_port()
by Kuninori Morimoto 24 Jun '16

24 Jun '16

Hi OK, I will post v2 patch to solve this issue. But it will be after patch-review. > [auto build test WARNING on robh/for-next] > [also build test WARNING on v4.7-rc4 next-20160623] > [cannot apply to glikely/devicetree/next] > [if your patch is applied to the wrong git tree, please drop us a note to help improve the system] > > url: https://github.com/0day-ci/linux/commits/Kuninori-Morimoto/of_graph-prepare… > base: https://git.kernel.org/pub/scm/linux/kernel/git/robh/linux for-next > config: arm-at91_dt_defconfig (attached as .config) > compiler: arm-linux-gnueabi-gcc (Debian 5.3.1-8) 5.3.1 20160205 > reproduce: > wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # save the attached .config to linux build tree > make.cross ARCH=arm > > All warnings (new ones prefixed by >>): > > In file included from drivers/gpu/drm/atmel-hlcdc/atmel_hlcdc_output.c:22:0: > >> include/linux/of_graph.h:54:50: warning: 'struct device' declared inside parameter list > struct device_node *of_graph_get_top_port(struct device *dev); > ^ > >> include/linux/of_graph.h:54:50: warning: its scope is only this definition or declaration, which is probably not what you want > > vim +54 include/linux/of_graph.h > > 38 */ > 39 #define for_each_endpoint_of_node(parent, child) \ > 40 for (child = of_graph_get_next_endpoint(parent, NULL); child != NULL; \ > 41 child = of_graph_get_next_endpoint(parent, child)) > 42 > 43 #define of_graph_port_type_is_sound(n) of_graph_port_type_is(n, "sound") > 44 #define of_graph_endpoint_type_is_sound(n) of_graph_endpoint_type_is(n, "sound") > 45 #define of_graph_get_sound_endpoint_count(n) of_graph_get_endpoint_count(n, "sound") > 46 > 47 #ifdef CONFIG_OF > 48 int of_graph_parse_endpoint(const struct device_node *node, > 49 struct of_endpoint *endpoint); > 50 bool of_graph_port_type_is(struct device_node *port, char *type); > 51 bool of_graph_endpoint_type_is(struct device_node *ep, char *type); > 52 int of_graph_get_endpoint_count(const struct device_node *np, char *type); > 53 struct device_node *of_graph_get_port_by_id(struct device_node *node, u32 id); > > 54 struct device_node *of_graph_get_top_port(struct device *dev); > 55 struct device_node *of_graph_get_next_endpoint(const struct device_node *parent, > 56 struct device_node *previous); > 57 struct device_node *of_graph_get_endpoint_by_regs( > 58 const struct device_node *parent, int port_reg, int reg); > 59 struct device_node *of_graph_get_remote_endpoint( > 60 const struct device_node *node); > 61 struct device_node *of_graph_get_port_parent(struct device_node *node); > 62 struct device_node *of_graph_get_remote_port_parent( > > --- > 0-DAY kernel test infrastructure Open Source Technology Center > https://lists.01.org/pipermail/kbuild-all Intel Corporation > [2 .config.gz <application/octet-stream (base64)>] >

1 0

[alsa-devel] S/PDIF and track markers (echo3g)
by Mark Hills 24 Jun '16

24 Jun '16

Hi, Is there any provision in ALSA (either within the kernel or libasound) for extracting track marker information from S/PDIF? I have a large number of MiniDisc recordings to preserve as FLAC format. Losing track markers would be a huge inconvenience, though I don't care for the rest of the metadata (track names etc..) There's enough recordings to be worth developing specific software for the archival if it can access this information. I don't really know which path is likely to be successful: * An ALSA API to access the information; though I can't see anything specific to track markers * Whether a timecode could be used to detect track change; but I can't see any in /proc/asound/Layla3G/status * Does ALSA provide access to raw AES streams, so I could decode it in userland? I'm un-sure if this relates to hardware AC3 decoding. * Maybe this sort of status is only available to the firmware of the interface and never makes it to userland (in my case, echo3g); though I can see some constants relating to 'non audio' S/PDIF. The hardware I have here is Echo Layla 3G (snd-echo3g) and a Pioneer MJ-D707 MiniDisc recorder. Any advice appreciated -- many thanks. -- Mark

2 2

[alsa-devel] [PATCH alsa-lib] doxygen: Suppress timestamp
by Takashi Iwai 23 Jun '16

23 Jun '16

The build timestamp in the doc is bad for packaging in general, so let's disable it as default. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- doc/doxygen.cfg.in | 2 ++ 1 file changed, 2 insertions(+) diff --git a/doc/doxygen.cfg.in b/doc/doxygen.cfg.in index 92bd52ba67a2..11740d342596 100644 --- a/doc/doxygen.cfg.in +++ b/doc/doxygen.cfg.in @@ -124,3 +124,5 @@ TYPEDEF_HIDES_STRUCT = YES # needed in doxygen >= 1.5.4 #INPUT_FILTER = inputfilter #FILTER_SOURCE_FILES = YES + +HTML_TIMESTAMP = NO -- 2.9.0

1 0

[alsa-devel] [very-RFC 0/8] TSN driver for the kernel
by Henrik Austad 23 Jun '16

23 Jun '16

Hi all (series based on v4.7-rc2, now with the correct netdev) This is a *very* early RFC for a TSN-driver in the kernel. It has been floating around in my repo for a while and I would appreciate some feedback on the overall design to avoid doing some major blunders. TSN: Time Sensitive Networking, formely known as AVB (Audio/Video Bridging). There are at least one AVB-driver (the AV-part of TSN) in the kernel already, however this driver aims to solve a wider scope as TSN can do much more than just audio. A very basic ALSA-driver is added to the end that allows you to play music between 2 machines using aplay in one end and arecord | aplay on the other (some fiddling required) We have plans for doing the same for v4l2 eventually (but there are other fishes to fry first). The same goes for a TSN_SOCK type approach as well. TSN is all about providing infrastructure. Allthough there are a few very interesting uses for TSN (reliable, deterministic network for audio and video), once you have that reliable link, you can do a lot more. Some notes on the design: The driver is directed via ConfigFS as we need userspace to handle stream-reservation (MSRP), discovery and enumeration (IEEE 1722.1) and whatever other management is needed. Once we have all the required attributes, we can create link using mkdir, and use write() to set the attributes. Once ready, specify the 'shim' (basically a thin wrapper between TSN and another subsystem) and we start pushing out frames. The network part: it ties directly into the rx-handler for receive and writes skb's using netdev_start_xmit(). This could probably be improved. 2 new fields in netdev_ops have been introduced, and the Intel igb-driver has been updated (as this is available as a PCI-e card). The igb-driver works-ish What remains - tie to (g)PTP properly, currently using ktime_get() for presentation time - get time from shim into TSN and vice versa - let shim create/manage buffer Henrik Austad (8): TSN: add documentation TSN: Add the standard formerly known as AVB to the kernel Adding TSN-driver to Intel I210 controller Add TSN header for the driver Add TSN machinery to drive the traffic from a shim over the network Add TSN event-tracing AVB ALSA - Add ALSA shim for TSN MAINTAINERS: add TSN/AVB-entries Documentation/TSN/tsn.txt | 147 +++++ MAINTAINERS | 14 + drivers/media/Kconfig | 15 + drivers/media/Makefile | 3 +- drivers/media/avb/Makefile | 5 + drivers/media/avb/avb_alsa.c | 742 +++++++++++++++++++++++ drivers/media/avb/tsn_iec61883.h | 124 ++++ drivers/net/ethernet/intel/Kconfig | 18 + drivers/net/ethernet/intel/igb/Makefile | 2 +- drivers/net/ethernet/intel/igb/igb.h | 19 + drivers/net/ethernet/intel/igb/igb_main.c | 10 +- drivers/net/ethernet/intel/igb/igb_tsn.c | 396 ++++++++++++ include/linux/netdevice.h | 32 + include/linux/tsn.h | 806 ++++++++++++++++++++++++ include/trace/events/tsn.h | 349 +++++++++++ net/Kconfig | 1 + net/Makefile | 1 + net/tsn/Kconfig | 32 + net/tsn/Makefile | 6 + net/tsn/tsn_configfs.c | 623 +++++++++++++++++++ net/tsn/tsn_core.c | 975 ++++++++++++++++++++++++++++++ net/tsn/tsn_header.c | 203 +++++++ net/tsn/tsn_internal.h | 383 ++++++++++++ net/tsn/tsn_net.c | 403 ++++++++++++ 24 files changed, 5306 insertions(+), 3 deletions(-) create mode 100644 Documentation/TSN/tsn.txt create mode 100644 drivers/media/avb/Makefile create mode 100644 drivers/media/avb/avb_alsa.c create mode 100644 drivers/media/avb/tsn_iec61883.h create mode 100644 drivers/net/ethernet/intel/igb/igb_tsn.c create mode 100644 include/linux/tsn.h create mode 100644 include/trace/events/tsn.h create mode 100644 net/tsn/Kconfig create mode 100644 net/tsn/Makefile create mode 100644 net/tsn/tsn_configfs.c create mode 100644 net/tsn/tsn_core.c create mode 100644 net/tsn/tsn_header.c create mode 100644 net/tsn/tsn_internal.h create mode 100644 net/tsn/tsn_net.c -- 2.7.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

9 63

[alsa-devel] [PATCH v2] ASoC: cs53l30: Add MUTE pin control support via GPIO
by Nicolin Chen 23 Jun '16

23 Jun '16

The codec chip has a physical MUTE pin to let users control it via GPIO. So this patch add a mute control support to the driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog: v1->v2 * Revise the DT binding part regarding the active state of the mute pin .../devicetree/bindings/sound/cs53l30.txt | 4 +++ sound/soc/codecs/cs53l30.c | 30 ++++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 1 + 3 files changed, 35 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt index 18d6b99..4dbfb82 100644 --- a/Documentation/devicetree/bindings/sound/cs53l30.txt +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -13,6 +13,10 @@ Optional properties: - reset-gpios : a GPIO spec for the reset pin. + - mute-gpios : a GPIO spec for the MUTE pin. The active state can be either + GPIO_ACTIVE_HIGH or GPIO_ACTIVE_LOW, which would be handled + by the driver automatically. + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. 0 = Hi-Z 1 = 1.80 V diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 384a3f7..e46b7de 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -35,6 +35,7 @@ struct cs53l30_private { struct regulator_bulk_data supplies[CS53L30_NUM_SUPPLIES]; struct regmap *regmap; struct gpio_desc *reset_gpio; + struct gpio_desc *mute_gpio; struct clk *mclk; bool use_sdout2; u32 mclk_rate; @@ -833,6 +834,16 @@ static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + + if (priv->mute_gpio) + gpiod_set_value_cansleep(priv->mute_gpio, mute); + + return 0; +} + /* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ #define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) @@ -846,6 +857,7 @@ static const struct snd_soc_dai_ops cs53l30_ops = { .set_sysclk = cs53l30_set_sysclk, .set_tristate = cs53l30_set_tristate, .set_tdm_slot = cs53l30_set_dai_tdm_slot, + .mute_stream = cs53l30_mute_stream, }; static struct snd_soc_dai_driver cs53l30_dai = { @@ -990,6 +1002,24 @@ static int cs53l30_i2c_probe(struct i2c_client *client, cs53l30->mclk = NULL; } + /* Fetch the MUTE control */ + cs53l30->mute_gpio = devm_gpiod_get_optional(dev, "mute", + GPIOD_OUT_HIGH); + if (IS_ERR(cs53l30->mute_gpio)) { + ret = PTR_ERR(cs53l30->mute_gpio); + goto error; + } + + if (cs53l30->mute_gpio) { + /* Enable MUTE controls via MUTE pin */ + regmap_write(cs53l30->regmap, CS53L30_MUTEP_CTL1, + CS53L30_MUTEP_CTL1_MUTEALL); + /* Flip the polarity of MUTE pin */ + if (gpiod_is_active_low(cs53l30->mute_gpio)) + regmap_update_bits(cs53l30->regmap, CS53L30_MUTEP_CTL2, + CS53L30_MUTE_PIN_POLARITY, 0); + } + if (!of_property_read_u8(np, "cirrus,micbias-lvl", &val)) regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL, CS53L30_MIC_BIAS_CTRL_MASK, val); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h index 0dd4afb..5e39da5 100644 --- a/sound/soc/codecs/cs53l30.h +++ b/sound/soc/codecs/cs53l30.h @@ -253,6 +253,7 @@ #define CS53L30_MUTE_MB_ALL_PDN_MASK (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) #define CS53L30_MUTE_MB_ALL_PDN (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) +#define CS53L30_MUTEP_CTL1_MUTEALL (0xdf) #define CS53L30_MUTEP_CTL1_DEFAULT (0) /* R32 (0x20) CS53L30_MUTEP_CTL2 - MUTE Pin Control 2 */ -- 2.1.4

3 2

[alsa-devel] [PATCH 1/4] ASoC: compress: Pass error out of soc_compr_pointer
by Charles Keepax 22 Jun '16

22 Jun '16

The soc_compr_pointer does not correctly pass any errors returned by the driver callback back up the stack. This patch corrects this issue. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 875733c..d2df46c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -530,14 +530,15 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) - platform->driver->compr_ops->pointer(cstream, tstamp); + ret = platform->driver->compr_ops->pointer(cstream, tstamp); mutex_unlock(&rtd->pcm_mutex); - return 0; + return ret; } static int soc_compr_copy(struct snd_compr_stream *cstream, -- 2.1.4

4 15

[alsa-devel] [PATCH 1/2] ASoC: Intel: Kconfig: fix build when ACPI is not enabled
by Vinod Koul 22 Jun '16

22 Jun '16

Randy reported following error when ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI +which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, CONFIG_SND_SST_IPC_ACPI was already dependent upon ACPI, but that was not solving it. So move the depends up to machine drivers and remove from CONFIG_SND_SST_IPC_ACPI. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/Kconfig | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 3875425a9693..63e4c14ebe9f 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -25,7 +25,6 @@ config SND_SST_IPC_ACPI tristate select SND_SST_IPC select SND_SOC_INTEL_SST - depends on ACPI config SND_SOC_INTEL_SST tristate @@ -128,7 +127,7 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -141,7 +140,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -154,7 +153,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5670 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI @@ -167,7 +166,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5645/5650 codec" - depends on X86_INTEL_LPSS && I2C + depends on X86_INTEL_LPSS && I2C && ACPI select SND_SOC_RT5645 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI -- 1.9.1

2 3

[alsa-devel] [PATCH 1/2] ASoC: rt5514: Fix the issue that the variable dereferenced before checking
by Oder Chiou 22 Jun '16

22 Jun '16

The patch fixes the issue that variable dereferenced before checking 'rt5514_dsp->substream'. Move the assignment to after the variable checking of 'rt5514_dsp->substream'. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5514-spi.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c index 8a9382e..743f509 100644 --- a/sound/soc/codecs/rt5514-spi.c +++ b/sound/soc/codecs/rt5514-spi.c @@ -80,7 +80,7 @@ static void rt5514_spi_copy_work(struct work_struct *work) { struct rt5514_dsp *rt5514_dsp = container_of(work, struct rt5514_dsp, copy_work.work); - struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + struct snd_pcm_runtime *runtime; size_t period_bytes, truncated_bytes = 0; mutex_lock(&rt5514_dsp->dma_lock); @@ -89,6 +89,7 @@ static void rt5514_spi_copy_work(struct work_struct *work) goto done; } + runtime = rt5514_dsp->substream->runtime; period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v2 0/4] ASoC: Intel: Skylake: Disable async suspend
by Vinod Koul 22 Jun '16

22 Jun '16

Skylake does not support async suspend, so mark it and update the machine driver. Also revert older machine patch which worked around this by using pm.complete. Also update a style inconsistency while at it. Vinod Koul (4): ASoC: Intel: Revert "ASoC: Intel: Add support for PM ops in bxt-rt298" ASoC: Intel: Skylake: Disable async suspend ASoC: Intel: Skylake: Add pm ops for broxton-rt298 machine ASoC: Intel: Skylake: Update comment style sound/soc/intel/boards/bxt_rt298.c | 24 +----------------------- sound/soc/intel/skylake/skl.c | 4 +++- 2 files changed, 4 insertions(+), 24 deletions(-) Thanks -- 1.9.1

2 8

[alsa-devel] [PATCH] ASoC: max9877: Remove unused function declaration
by Helen Koike 22 Jun '16

22 Jun '16

Remove unused function declaration from header Signed-off-by: Helen Koike <helen.koike(a)collabora.co.uk> --- sound/soc/codecs/max9877.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/max9877.h b/sound/soc/codecs/max9877.h index 6da7229..368343f 100644 --- a/sound/soc/codecs/max9877.h +++ b/sound/soc/codecs/max9877.h @@ -32,6 +32,4 @@ #define MAX9877_BYPASS (1 << 6) #define MAX9877_SHDN (1 << 7) -extern int max9877_add_controls(struct snd_soc_codec *codec); - #endif -- 1.9.1

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Skylake: Add DSP muti-core support
by Vinod Koul 22 Jun '16

22 Jun '16

This series adds support for managing multiple DSP cores. SW is responsible for powering up/down cores in DSP and we do that using refcount. 1st patch introduces core infrastructure change in driver and then adds support in SKL and BXT Changes in v2: - fix error report by kbuild bot Jayachandran B (3): ASoC: Intel: Skylake: Add DSP muti-core infrastructure ASoC: Intel: Skylake: Support multi-core in Skylake ASoC: Intel: Skylake: Support multi-core in Broxton sound/soc/intel/skylake/bxt-sst.c | 160 ++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 253 ++++++++++++++++++++++++---------- sound/soc/intel/skylake/skl-sst-dsp.h | 96 ++++++++----- sound/soc/intel/skylake/skl-sst-ipc.h | 11 ++ sound/soc/intel/skylake/skl-sst.c | 91 +++++++----- 5 files changed, 425 insertions(+), 186 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH 0/3] ASoC: Intel: Skylake: Add DSP muti-core support
by Vinod Koul 22 Jun '16

22 Jun '16

This series adds support for managing multiple DSP cores. SW is responsible for powering up/down cores in DSP and we do that using refcount. 1st patch introduces core infrastructure change in driver and then adds support in SKL and BXT Jayachandran B (3): ASoC: Intel: Skylake: Add DSP muti-core infrastructure ASoC: Intel: Skylake: Support multi-core in Skylake ASoC: Intel: Skylake: Support multi-core in Broxton sound/soc/intel/skylake/bxt-sst.c | 160 ++++++++++++++------- sound/soc/intel/skylake/skl-sst-dsp.c | 252 ++++++++++++++++++++++++---------- sound/soc/intel/skylake/skl-sst-dsp.h | 96 ++++++++----- sound/soc/intel/skylake/skl-sst-ipc.h | 11 ++ sound/soc/intel/skylake/skl-sst.c | 91 +++++++----- 5 files changed, 424 insertions(+), 186 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: wm_adsp: Disable DMAs before clearing the transfer length
by Charles Keepax 22 Jun '16

22 Jun '16

From: Simon Trimmer <simont(a)opensource.wolfsonmicro.com> This patch reorders the clearing of the DMA masks to avoid potential artefacts being introduced. Signed-off-by: Simon Trimmer <simont(a)opensource.wolfsonmicro.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 430bb8b..21fbe7d 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2368,13 +2368,15 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, dsp->running = false; regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, - ADSP2_SYS_ENA | ADSP2_CORE_ENA | - ADSP2_START, 0); + ADSP2_CORE_ENA | ADSP2_START, 0); /* Make sure DMAs are quiesced */ + regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_1, 0); regmap_write(dsp->regmap, dsp->base + ADSP2_WDMA_CONFIG_2, 0); - regmap_write(dsp->regmap, dsp->base + ADSP2_RDMA_CONFIG_1, 0); + + regmap_update_bits(dsp->regmap, dsp->base + ADSP2_CONTROL, + ADSP2_SYS_ENA, 0); list_for_each_entry(ctl, &dsp->ctl_list, list) ctl->enabled = 0; -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: hda/tegra: iomem fixups for sparse warnings
by Ben Dooks 22 Jun '16

22 Jun '16

The readl/writel are not being passed __iomem annotated variables, so fix the following sparse warnings by adding __iomem in: sound/pci/hda/hda_tegra.c:120:9: warning: incorrect type in argument 2 (different address spaces) sound/pci/hda/hda_tegra.c:120:9: expected void volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:120:9: got unsigned int [usertype] *addr sound/pci/hda/hda_tegra.c:125:16: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:125:16: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:125:16: got unsigned int [usertype] *addr sound/pci/hda/hda_tegra.c:134:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:134:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:134:13: got void *dword_addr sound/pci/hda/hda_tegra.c:137:9: warning: incorrect type in argument 2 (different address spaces) sound/pci/hda/hda_tegra.c:137:9: expected void volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:137:9: got void *dword_addr sound/pci/hda/hda_tegra.c:146:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:146:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:146:13: got void *dword_addr sound/pci/hda/hda_tegra.c:156:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:156:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:156:13: got void *dword_addr sound/pci/hda/hda_tegra.c:159:9: warning: incorrect type in argument 2 (different address spaces) sound/pci/hda/hda_tegra.c:159:9: expected void volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:159:9: got void *dword_addr sound/pci/hda/hda_tegra.c:168:13: warning: incorrect type in argument 1 (different address spaces) sound/pci/hda/hda_tegra.c:168:13: expected void const volatile [noderef] <asn:2>*addr sound/pci/hda/hda_tegra.c:168:13: got void *dword_addr sound/pci/hda/hda_tegra.c:173:23: warning: incorrect type in initializer (incompatible argument 2 (different address spaces)) sound/pci/hda/hda_tegra.c:173:23: expected void ( *reg_writel )( ... ) sound/pci/hda/hda_tegra.c:173:23: got void ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:174:22: warning: incorrect type in initializer (incompatible argument 1 (different address spaces)) sound/pci/hda/hda_tegra.c:174:22: expected unsigned int ( *reg_readl )( ... ) sound/pci/hda/hda_tegra.c:174:22: got unsigned int ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:175:23: warning: incorrect type in initializer (incompatible argument 2 (different address spaces)) sound/pci/hda/hda_tegra.c:175:23: expected void ( *reg_writew )( ... ) sound/pci/hda/hda_tegra.c:175:23: got void ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:176:22: warning: incorrect type in initializer (incompatible argument 1 (different address spaces)) sound/pci/hda/hda_tegra.c:176:22: expected unsigned short ( *reg_readw )( ... ) sound/pci/hda/hda_tegra.c:176:22: got unsigned short ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:177:23: warning: incorrect type in initializer (incompatible argument 2 (different address spaces)) sound/pci/hda/hda_tegra.c:177:23: expected void ( *reg_writeb )( ... ) sound/pci/hda/hda_tegra.c:177:23: got void ( static [toplevel] *<noident> )( ... ) sound/pci/hda/hda_tegra.c:178:22: warning: incorrect type in initializer (incompatible argument 1 (different address spaces)) sound/pci/hda/hda_tegra.c:178:22: expected unsigned char ( *reg_readb )( ... ) sound/pci/hda/hda_tegra.c:178:22: got unsigned char ( static [toplevel] *<noident> )( ... ) Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Stephen Warren <swarren(a)wwwdotorg.org> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Alexandre Courbot <gnurou(a)gmail.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-tegra(a)vger.kernel.org --- sound/pci/hda/hda_tegra.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 17fd817..0621920 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -115,20 +115,20 @@ static int substream_free_pages(struct azx *chip, /* * Register access ops. Tegra HDA register access is DWORD only. */ -static void hda_tegra_writel(u32 value, u32 *addr) +static void hda_tegra_writel(u32 value, u32 __iomem *addr) { writel(value, addr); } -static u32 hda_tegra_readl(u32 *addr) +static u32 hda_tegra_readl(u32 __iomem *addr) { return readl(addr); } -static void hda_tegra_writew(u16 value, u16 *addr) +static void hda_tegra_writew(u16 value, u16 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); @@ -137,20 +137,20 @@ static void hda_tegra_writew(u16 value, u16 *addr) writel(v, dword_addr); } -static u16 hda_tegra_readw(u16 *addr) +static u16 hda_tegra_readw(u16 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); return (v >> shift) & 0xffff; } -static void hda_tegra_writeb(u8 value, u8 *addr) +static void hda_tegra_writeb(u8 value, u8 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); @@ -159,10 +159,10 @@ static void hda_tegra_writeb(u8 value, u8 *addr) writel(v, dword_addr); } -static u8 hda_tegra_readb(u8 *addr) +static u8 hda_tegra_readb(u8 __iomem *addr) { unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; - void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + void __iomem *dword_addr = (void __iomem *)((unsigned long)(addr) & ~0x3); u32 v; v = readl(dword_addr); -- 2.8.1

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Re: [alsa-devel] ASoC: nau8825: cross talk suppression measurement function
by Dan Carpenter 22 Jun '16

22 Jun '16

Hello John Hsu, The patch b50455fab459: "ASoC: nau8825: cross talk suppression measurement function" from Jun 7, 2016, leads to the following static checker warning: sound/soc/codecs/nau8825.c:265 nau8825_sema_acquire() warn: 'sem:&nau8825->xtalk_sem' is sometimes locked here and sometimes unlocked. sound/soc/codecs/nau8825.c 240 /** 241 * nau8825_sema_acquire - acquire the semaphore of nau88l25 242 * @nau8825: component to register the codec private data with 243 * @timeout: how long in jiffies to wait before failure or zero to wait 244 * until release 245 * 246 * Attempts to acquire the semaphore with number of jiffies. If no more 247 * tasks are allowed to acquire the semaphore, calling this function will 248 * put the task to sleep. If the semaphore is not released within the 249 * specified number of jiffies, this function returns. 250 * Acquires the semaphore without jiffies. If no more tasks are allowed 251 * to acquire the semaphore, calling this function will put the task to 252 * sleep until the semaphore is released. 253 * It returns if the semaphore was acquired. Sometimes it returns without the lock held. 254 */ 255 static void nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) 256 { 257 int ret; 258 259 if (timeout) 260 ret = down_timeout(&nau8825->xtalk_sem, timeout); 261 else 262 ret = down_interruptible(&nau8825->xtalk_sem); 263 264 if (ret < 0) 265 dev_warn(nau8825->dev, "Acquire semaphone fail\n"); This function makes me sad. Warn and corrupt is my least favourite anti-pattern. 266 } regards, dan carpenter

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[alsa-devel] [PATCH] ASoC: cs53l30: Add MUTE pin control support via GPIO
by Nicolin Chen 22 Jun '16

22 Jun '16

The codec chip has a physical MUTE pin to let users control it via GPIO. So this patch add a mute control support to the driver. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- .../devicetree/bindings/sound/cs53l30.txt | 2 ++ sound/soc/codecs/cs53l30.c | 30 ++++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 1 + 3 files changed, 33 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt index 18d6b99..6f7d3c8 100644 --- a/Documentation/devicetree/bindings/sound/cs53l30.txt +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -13,6 +13,8 @@ Optional properties: - reset-gpios : a GPIO spec for the reset pin. + - mute-gpios : a GPIO spec for the MUTE pin. + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. 0 = Hi-Z 1 = 1.80 V diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index b0a64a1..5988b5c 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -35,6 +35,7 @@ struct cs53l30_private { struct regulator_bulk_data supplies[CS53L30_NUM_SUPPLIES]; struct regmap *regmap; struct gpio_desc *reset_gpio; + struct gpio_desc *mute_gpio; struct clk *mclk; bool use_sdout2; u32 mclk_rate; @@ -833,6 +834,16 @@ static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int cs53l30_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + + if (priv->mute_gpio) + gpiod_set_value_cansleep(priv->mute_gpio, mute); + + return 0; +} + /* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ #define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) @@ -846,6 +857,7 @@ static const struct snd_soc_dai_ops cs53l30_ops = { .set_sysclk = cs53l30_set_sysclk, .set_tristate = cs53l30_set_tristate, .set_tdm_slot = cs53l30_set_dai_tdm_slot, + .mute_stream = cs53l30_mute_stream, }; static struct snd_soc_dai_driver cs53l30_dai = { @@ -991,6 +1003,24 @@ static int cs53l30_i2c_probe(struct i2c_client *client, cs53l30->mclk = NULL; } + /* Fetch the MUTE control */ + cs53l30->mute_gpio = devm_gpiod_get_optional(dev, "mute", + GPIOD_OUT_HIGH); + if (IS_ERR(cs53l30->mute_gpio)) { + ret = PTR_ERR(cs53l30->mute_gpio); + goto error; + } + + if (cs53l30->mute_gpio) { + /* Enable MUTE controls via MUTE pin */ + regmap_write(cs53l30->regmap, CS53L30_MUTEP_CTL1, + CS53L30_MUTEP_CTL1_MUTEALL); + /* Flip the polarity of MUTE pin */ + if (gpiod_is_active_low(cs53l30->mute_gpio)) + regmap_update_bits(cs53l30->regmap, CS53L30_MUTEP_CTL2, + CS53L30_MUTE_PIN_POLARITY, 0); + } + if (!of_property_read_u8(np, "cirrus,micbias-lvl", &val)) regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL, CS53L30_MIC_BIAS_CTRL_MASK, val); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h index 0dd4afb..5e39da5 100644 --- a/sound/soc/codecs/cs53l30.h +++ b/sound/soc/codecs/cs53l30.h @@ -253,6 +253,7 @@ #define CS53L30_MUTE_MB_ALL_PDN_MASK (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) #define CS53L30_MUTE_MB_ALL_PDN (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) +#define CS53L30_MUTEP_CTL1_MUTEALL (0xdf) #define CS53L30_MUTEP_CTL1_DEFAULT (0) /* R32 (0x20) CS53L30_MUTEP_CTL2 - MUTE Pin Control 2 */ -- 2.1.4

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[alsa-devel] [PATCH -next] ASoC: max9860: fix non static symbol warnings
by weiyj_lk＠163.com 21 Jun '16

21 Jun '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warnings: sound/soc/codecs/max9860.c:120:28: warning: symbol 'max9860_regmap' was not declared. Should it be static? sound/soc/codecs/max9860.c:596:25: warning: symbol 'max9860_pm_ops' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/max9860.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c index 2b0dd6a..68074c9 100644 --- a/sound/soc/codecs/max9860.c +++ b/sound/soc/codecs/max9860.c @@ -117,7 +117,7 @@ static bool max9860_precious(struct device *dev, unsigned int reg) return false; } -const struct regmap_config max9860_regmap = { +static const struct regmap_config max9860_regmap = { .reg_bits = 8, .val_bits = 8, @@ -593,7 +593,7 @@ static int max9860_resume(struct device *dev) } #endif -const struct dev_pm_ops max9860_pm_ops = { +static const struct dev_pm_ops max9860_pm_ops = { SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) };

2 1

[alsa-devel] [PATCH -next] ASoC: cs53l30: Fix non static symbol warnings
by weiyj_lk＠163.com 21 Jun '16

21 Jun '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fixes the following sparse warnings: sound/soc/codecs/cs53l30.c:182:20: warning: symbol 'input1_sel_values' was not declared. Should it be static? sound/soc/codecs/cs53l30.c:202:20: warning: symbol 'input2_sel_values' was not declared. Should it be static? sound/soc/codecs/cs53l30.c:734:20: warning: symbol 'cs53l30_src_rates' was not declared. Should it be static? Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/codecs/cs53l30.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 714e579..a9949ad 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -179,7 +179,7 @@ static const char * const input1_sel_text[] = { "DMIC1 Off ADC1 Off", }; -unsigned int const input1_sel_values[] = { +static unsigned int const input1_sel_values[] = { CS53L30_CH_TYPE, CS53L30_ADCxB_PDN | CS53L30_CH_TYPE, CS53L30_ADCxA_PDN | CS53L30_CH_TYPE, @@ -199,7 +199,7 @@ static const char * const input2_sel_text[] = { "DMIC2 Off ADC2 Off", }; -unsigned int const input2_sel_values[] = { +static unsigned int const input2_sel_values[] = { 0x0, CS53L30_ADCxB_PDN, CS53L30_ADCxA_PDN, @@ -731,7 +731,7 @@ static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) CS53L30_ASP_3ST_MASK, val); } -unsigned int const cs53l30_src_rates[] = { +static unsigned int const cs53l30_src_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 };

4 3

[alsa-devel] [PATCH] ASoC: cs53l30: Set idle_bias_off true
by Nicolin Chen 21 Jun '16

21 Jun '16

The driver is using the set_bias_level to control the power on and off so it should get SND_SOC_BIAS_OFF in order to proceed normal powering sequences. This patch enables the idle_bias_off option so the DAPM core will set the bias level to SND_SOC_BIAS_OFF instead of stopping at SND_SOC_BIAS_STANDBY. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index a5976f3..b0a64a1 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -879,6 +879,7 @@ static int cs53l30_codec_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver cs53l30_driver = { .probe = cs53l30_codec_probe, .set_bias_level = cs53l30_set_bias_level, + .idle_bias_off = true, .dapm_widgets = cs53l30_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), -- 2.1.4

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[alsa-devel] [PATCH] ASoC: dwc: make pcm support built-in when necessary
by Arnd Bergmann 21 Jun '16

21 Jun '16

The new PIO mode for the dwc audio driver causes a link failure when it is built as a loadable module but the audio driver is built-in: sound/built-in.o: In function `i2s_irq_handler': :(.text+0x58c64): undefined reference to `dw_pcm_push_tx' sound/built-in.o: In function `dw_i2s_probe': :(.text+0x593dc): undefined reference to `dw_pcm_register' We could link both into a single module, but apparently the author intended them to be separate, so this instead changes the Makefile to force the pcm module to be built-in if the base module is. This is a bit hacky but not as bad as trying to work around it in Kconfig language. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 79361b2b98b7 ("ASoC: dwc: Add PIO PCM extension") --- sound/soc/dwc/Makefile | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile index 1b48bcccbc51..38f1ca31c5fa 100644 --- a/sound/soc/dwc/Makefile +++ b/sound/soc/dwc/Makefile @@ -1,4 +1,5 @@ # SYNOPSYS Platform Support obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o -obj-$(CONFIG_SND_DESIGNWARE_PCM) += designware_pcm.o - +ifdef CONFIG_SND_DESIGNWARE_PCM +obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_pcm.o +endif -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: rt5670: patch reg-0x8a
by Bard Liao 21 Jun '16

21 Jun '16

reg-8a assign the tracking source for each ASRC tracker. The default value is 0x0000 which means all ASRC trackers will track LRCK1. But in most case, we wish each ASRC tracker track the corresponding LRCK. i.e. ASRC1 tracks LRCK1, ASRC2 tracks LRCK2 and so on. So, we rewrite reg-8a as 0x0123. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 0af5ddb..8ef467f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -55,6 +55,7 @@ static const struct reg_sequence init_list[] = { { RT5670_PR_BASE + 0x14, 0x9a8a }, { RT5670_PR_BASE + 0x38, 0x3ba1 }, { RT5670_PR_BASE + 0x3d, 0x3640 }, + { 0x8a, 0x0123 }, }; static const struct reg_default rt5670_reg[] = { @@ -131,7 +132,7 @@ static const struct reg_default rt5670_reg[] = { { 0x87, 0x0000 }, { 0x88, 0x0000 }, { 0x89, 0x0000 }, - { 0x8a, 0x0000 }, + { 0x8a, 0x0123 }, { 0x8b, 0x0000 }, { 0x8c, 0x0003 }, { 0x8d, 0x0000 }, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [RESEND PATCH v3 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 21 Jun '16

21 Jun '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.7-rc3 kernel. Changes in v3: - Use of_node_cmp() in device_get_named_child_node() to match DT node. Changes in v2: - Rebase to v4.7-rc1 - Small updates to codec patch based on previous reviewer comments Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/of.h | 14 +++--- include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 7 files changed, 119 insertions(+), 76 deletions(-) -- 1.9.3

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[alsa-devel] [PATCH] gcc6 reporting error building alsa-tools on PowerPC
by Michel Normand 21 Jun '16

21 Jun '16

Hello there, While trying to build alsa-tools with gcc6, many reported error that are solved by proposed patch. I do not know if sending to this mailing list is the correct way to provide it to you. Michel Normand (1): gcc6 narrowing error hdspmixer/src/channelmap.cxx | 158 +++++++++++++++++++++---------------------- us428control/Cus428Midi.cc | 30 ++++---- 2 files changed, 94 insertions(+), 94 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] INSTALL: update text and drop reference to Hg
by Henrik Austad 21 Jun '16

21 Jun '16

This is really just nitpick, but it annoyed me for a few seconds, so I thought I'd just fix it. In d7534b2ceec7 (hgcompile -> gitcompile) the filename was updated but INSTALL was not, leaving outdated docs. Signed-off-by: Henrik Austad <henrik(a)austad.us> --- INSTALL | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/INSTALL b/INSTALL index 47086e3..64f3d09 100644 --- a/INSTALL +++ b/INSTALL @@ -11,13 +11,19 @@ For installation you can use these commands: make install -Compilation from HG sources +Compilation from Git-sources --------------------------- -You need also GNU packages automake and libtool installed in your system -to compile HG (Mercurial) sources of alsa-lib package. +Requirements: + * automake + * libtool -For compilation you can use these commands: +To compile alsa-lib from source: + + ./gitcompile + +If you do not want to use the supplied gitcompile-script, you can +compile alsa-lib manually by using these steps: libtoolize --force --copy --automake aclocal @@ -27,7 +33,6 @@ For compilation you can use these commands: ./configure make -The included hgcompile script does this job for you. Note: Some automake packages have missing aclocal program. Use newer version in the case. -- 2.1.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

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Re: [alsa-devel] 答复: [PATCH 1/1] sound: Fix compile error of seq_device.c
by Takashi Iwai 21 Jun '16

21 Jun '16

On Tue, 21 Jun 2016 10:48:30 +0200, 高峰 wrote: > > The lastest CentOS7 platform, and the gcc version is gcc version 4.8.5 > 20150623 (Red Hat 4.8.5-4) (GCC). > It could pass compilation after append the lost semicolon. > > from sound/core/seq/seq_device.c:39: > include/linux/init.h:216:2: error: expected ‘,’ or ‘;’ before ‘static’ > static exitcall_t __exitcall_##fn __exit_call = fn > ^ > include/linux/init.h:279:24: note: in expansion of macro ‘__exitcall’ > #define module_exit(x) __exitcall(x); > ^ > sound/core/seq/seq_device.c:315:1: note: in expansion of macro > ‘module_exit’ > module_exit(alsa_seq_device_exit) > ^ > make[3]: *** [sound/core/seq/seq_device.o] Error 1 > make[2]: *** [sound/core/seq] Error 2 > make[1]: *** [sound/core] Error 2 > make: *** [sound] Error 2 It's a RH specific issue. The upstream code has a different definition of module_init(), thus no such an error would occur. Note that I'm fine to apply the patch, it's a trivial change. But you need to give the clear reason why to apply it. In this case, it's no fault of the upstream code. But if it would make someone's life a bit easier, it's OK to apply such a change. That being said, please resubmit the patch with a more explanation. thanks, Takashi > > -----邮件原件----- > 发件人: Takashi Iwai [mailto:tiwai@suse.de] > 发送时间: 2016年6月21日 16:45 > 收件人: fgao(a)ikuai8.com > 抄送: alsa-devel(a)alsa-project.org; perex(a)perex.cz; linux-kernel(a)vger.kernel. > org; gfree.wind(a)gmail.com > 主题: Re: [PATCH 1/1] sound: Fix compile error of seq_device.c > > On Tue, 21 Jun 2016 10:34:25 +0200, > <fgao(a)ikuai8.com> wrote: > > > > From: Gao Feng <fgao(a)ikuai8.com> > > > > Signed-off-by: Gao Feng <fgao(a)ikuai8.com> > > What compile error did you get? > > > Takashi > > > --- > > sound/core/seq/seq_device.c | 4 ++-- > > 1 file changed, 2 insertions(+), 2 deletions(-) > > > > diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c > > index c4acf17..4e859e4 100644 > > --- a/sound/core/seq/seq_device.c > > +++ b/sound/core/seq/seq_device.c > > @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) > > bus_unregister(&snd_seq_bus_type); > > } > > > > -subsys_initcall(alsa_seq_device_init) > > -module_exit(alsa_seq_device_exit) > > +subsys_initcall(alsa_seq_device_init); > > +module_exit(alsa_seq_device_exit); > > -- > > 1.9.1 > > > > > > > > > > > > >

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[alsa-devel] [PATCH RFC 0/6] Sound support for Exynos5433 TM2(E) boards
by Sylwester Nawrocki 21 Jun '16

21 Jun '16

This patch series adds sound supoort for the Exynos 5433 based TM2(E) boards. It includes minimal drivers for the Low Power Audio Subsystem (LPASS) and the MAX98504 class D speaker amplifier as well as the actual asoc machine driver. Inha Song (2): ASoC: samsung: Add Samsung Low Power Audio Subsystem driver ASoC: samsung: Add machine driver for Exynos5433 based TM2 board Sylwester Nawrocki (4): ASoC: samsung: Add DT bindings documentation for LPASS ASoC: max98504: Add max98504 speaker amplifier driver ASoC: Add DT bindings documentation for max98504 amplifier ASoC: samsung: Add DT bindings documentation for TM2 sound subsystem .../devicetree/bindings/sound/max98504.txt | 44 ++ .../bindings/sound/samsung,exynos5433-lpass.txt | 8 + .../bindings/sound/samsung,tm2-wm5110.txt | 39 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max98504.c | 241 +++++++++ sound/soc/codecs/max98504.h | 94 ++++ sound/soc/samsung/Kconfig | 13 + sound/soc/samsung/Makefile | 4 + sound/soc/samsung/lpass.c | 162 ++++++ sound/soc/samsung/lpass.h | 47 ++ sound/soc/samsung/tm2_wm5110.c | 553 +++++++++++++++++++++ 12 files changed, 1211 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,exynos5433-lpass.txt create mode 100644 Documentation/devicetree/bindings/sound/samsung,tm2-wm5110.txt create mode 100644 sound/soc/codecs/max98504.c create mode 100644 sound/soc/codecs/max98504.h create mode 100644 sound/soc/samsung/lpass.c create mode 100644 sound/soc/samsung/lpass.h create mode 100644 sound/soc/samsung/tm2_wm5110.c -- 1.9.1

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[alsa-devel] [PATCH 1/1] sound: Fix one compile error of seq_device.c
by fgao＠ikuai8.com 21 Jun '16

21 Jun '16

From: Gao Feng <fgao(a)ikuai8.com> The gcc reports the following compile error on CentOS7. include/linux/init.h:216:2: error: expected ',' or ';' before 'static' static exitcall_t __exitcall_##fn __exit_call = fn ^ include/linux/init.h:279:24: note: in expansion of macro '__exitcall' #define module_exit(x) __exitcall(x); ^ sound/core/seq/seq_device.c:315:1: note: in expansion of macro 'module_exit' module_exit(alsa_seq_device_exit) ^ make[3]: *** [sound/core/seq/seq_device.o] Error 1 make[2]: *** [sound/core/seq] Error 2 make[1]: *** [sound/core] Error 2 make: *** [sound] Error 2 Signed-off-by: Gao Feng <fgao(a)ikuai8.com> --- sound/core/seq/seq_device.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index c4acf17..4e859e4 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -subsys_initcall(alsa_seq_device_init) -module_exit(alsa_seq_device_exit) +subsys_initcall(alsa_seq_device_init); +module_exit(alsa_seq_device_exit); -- 1.9.1

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[alsa-devel] [PATCH 1/1] sound: Fix compile error of seq_device.c
by fgao＠ikuai8.com 21 Jun '16

21 Jun '16

From: Gao Feng <fgao(a)ikuai8.com> Signed-off-by: Gao Feng <fgao(a)ikuai8.com> --- sound/core/seq/seq_device.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/seq/seq_device.c b/sound/core/seq/seq_device.c index c4acf17..4e859e4 100644 --- a/sound/core/seq/seq_device.c +++ b/sound/core/seq/seq_device.c @@ -311,5 +311,5 @@ static void __exit alsa_seq_device_exit(void) bus_unregister(&snd_seq_bus_type); } -subsys_initcall(alsa_seq_device_init) -module_exit(alsa_seq_device_exit) +subsys_initcall(alsa_seq_device_init); +module_exit(alsa_seq_device_exit); -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Remove compilation warning
by Helen Koike 21 Jun '16

21 Jun '16

Remove the warning: warning: ‘err’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Helen Koike <helen.koike(a)collabora.co.uk> --- sound/hda/hdac_regmap.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 87041dd..7be6ca1 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -279,7 +279,7 @@ static int hda_reg_write(void *context, unsigned int reg, unsigned int val) { struct hdac_device *codec = context; unsigned int verb; - int i, bytes, err; + int i, bytes, err = 0; int pm_lock = 0; if (codec->caps_overwriting) -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: intel: fix build when ACPI is not enabled
by Randy Dunlap 21 Jun '16

21 Jun '16

From: Randy Dunlap <rdunlap(a)infradead.org> kconfig tools generate the following warning when CONFIG_ACPI is not enabled: warning: (SND_SOC_INTEL_BYTCR_RT5640_MACH && SND_SOC_INTEL_BYTCR_RT5651_MACH && SND_SOC_INTEL_CHT_BSW_RT5672_MACH && SND_SOC_INTEL_CHT_BSW_RT5645_MACH && SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) selects SND_SST_IPC_ACPI which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && ACPI) causing these build errors: In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:40:0: ../include/acpi/acpi_bus.h:65:20: error: conflicting types for 'acpi_evaluate_dsm' union acpi_object *acpi_evaluate_dsm(acpi_handle handle, const u8 *uuid, In file included from ../sound/soc/intel/atom/sst/sst_acpi.c:31:0: ../include/linux/acpi.h:676:34: note: previous definition of 'acpi_evaluate_dsm' was here static inline union acpi_object *acpi_evaluate_dsm(acpi_handle handle, I am told that ACPI is a requirement for these drivers, so make that explicit. Also end help text sentences with a period. Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/Kconfig | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) Should SND_SOC_INTEL_SST and possibly other similar drivers get this same change? --- linux-next-20160607.orig/sound/soc/intel/Kconfig +++ linux-next-20160607/sound/soc/intel/Kconfig @@ -128,28 +128,28 @@ config SND_SOC_INTEL_BROADWELL_MACH config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5640 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5640 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI - select SND_SOC_INTEL_SST_MATCH if ACPI + select SND_SOC_INTEL_SST_MATCH help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR platforms with RT5640 audio codec. - Say Y if you have such a device + Say Y if you have such a device. If unsure select "N". config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5651 codec" - depends on X86 && I2C + depends on X86 && I2C && ACPI select SND_SOC_RT5651 select SND_SST_MFLD_PLATFORM select SND_SST_IPC_ACPI - select SND_SOC_INTEL_SST_MATCH if ACPI + select SND_SOC_INTEL_SST_MATCH help This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR platforms with RT5651 audio codec. - Say Y if you have such a device + Say Y if you have such a device. If unsure select "N". config SND_SOC_INTEL_CHT_BSW_RT5672_MACH

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[alsa-devel] How does ALSALIB commuciates with low level kernel API
by Chemsi Mehdi 20 Jun '16

20 Jun '16

Dear experts, I am too new in ALSA, trying to read a little bit and understand. I want to underatnd how ALSA userspace API calls kernel space code ? for example when I call snd_pcm_open() which code is called in kernel sound code ? and how the link is made ? via IOCTL/write ? Please clarify Thanks a lot in advance. Mehdi

3 2

Re: [alsa-devel] [PATCH] ASoC: rt5668: add rt5668 codec driver
by Mark Brown 20 Jun '16

20 Jun '16

On Thu, Jun 16, 2016 at 11:31:07AM +0000, Jack Yu wrote: > [Mark] Given that the interrupt handler just schedules a work item there's no > need for it to be a threaded interrupt, it can just be a normal > interrupt and save the bother of masking the hardware interrupt and > running a thread. > Do you mean that it's not appropriate to use " queue_delayed_work" here, we should > use other method for handling work item ? No, when I said it shouldn't be a threaded interrupt I meant that using request_threaded_irq() doesn't seem appropriate.

1 0

[alsa-devel] [PATCH] ASoC: core: Remove selects of regmap buses
by Mark Brown 20 Jun '16

20 Jun '16

We used to have code wrapping the regmap I2C and SPI buses as part of the conversion of ASoC drivers from using the ASoC level I/O to the regmap level I/O. Since all drivers have now been converted and the core code removed we can remove the per-bus selects and instead just select the regmap core. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/Kconfig | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 182d92efc7c8..4935e23ff3d6 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,8 +7,7 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK - select REGMAP_I2C if I2C - select REGMAP_SPI if SPI_MASTER + select REGMAP ---help--- If you want ASoC support, you should say Y here and also to the -- 2.8.1

2 2

[alsa-devel] [GIT PULL]: Skylake audio firmware update
by Vinod Koul 20 Jun '16

20 Jun '16

Hi, Please PULL to receive the latest Skylake audio firmware. The following changes since commit a4bbc8112cc04fcc0e0861e8b72197b6cb83ab83: linux-firmware: Update firmware file for Intel Bluetooth 8260 (2016-06-09 08:11:22 -0400) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git skl for you to fetch changes up to 80aae2270140807c4d8e304c39def1fe9f93edb9: linux-firmware: intel: Update Skylake audio firmware (2016-06-17 14:01:52 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v948.bin | Bin 0 -> 221184 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v948.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] ASoC: Intel: Add support for PM ops in bxt-da7219_max98357a
by Vinod Koul 20 Jun '16

20 Jun '16

We need card to be early suspended and late resumed, so use prepare and complete for card suspend and resume. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Harsha Priya <harshapriya.n(a)intel.com> --- sound/soc/intel/boards/bxt_da7219_max98357a.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 3774b117d365..df5f269a9b67 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -441,11 +441,30 @@ static int broxton_audio_probe(struct platform_device *pdev) return devm_snd_soc_register_card(&pdev->dev, &broxton_audio_card); } +#ifdef CONFIG_PM_SLEEP + +static void broxton_complete(struct device *dev) +{ + snd_soc_resume(dev); +} + +#else +#define broxton_complete NULL +#endif + +static const struct dev_pm_ops broxton_pm_ops = { + .prepare = snd_soc_suspend, + .complete = broxton_complete, + .freeze = snd_soc_suspend, + .thaw = snd_soc_resume, + .poweroff = snd_soc_poweroff, + .restore = snd_soc_resume, +}; static struct platform_driver broxton_audio = { .probe = broxton_audio_probe, .driver = { .name = "bxt_da7219_max98357a_i2s", - .pm = &snd_soc_pm_ops, + .pm = &broxton_pm_ops, }, }; module_platform_driver(broxton_audio) -- 1.9.1

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[alsa-devel] [PATCH v1 1/2] ASoC: Add DT bindings documentation for max98504 amplifier
by Sylwester Nawrocki 20 Jun '16

20 Jun '16

This patch adds DT bindings documentation for Maxim MAX98504 speaker amplifier. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since initial version: - dropped all optional properties used for setting up TDM and speaker data source, - added regulator supply properties. --- .../devicetree/bindings/sound/max98504.txt | 22 ++++++++++++++++++++++ 1 file changed, 22 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98504.txt diff --git a/Documentation/devicetree/bindings/sound/max98504.txt b/Documentation/devicetree/bindings/sound/max98504.txt new file mode 100644 index 0000000..7aa5b93 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98504.txt @@ -0,0 +1,22 @@ +Maxim MAX98504 class D mono speaker amplifier + +This device supports I2C control interface and an IRQ output signal. It features +a PCM and PDM digital audio interface (DAI) and a differential analog input. + +Required properties: + + - compatible : "maxim,max98504" + - reg : should contain the I2C slave device address + - DVDD-supply, DIOVDD-supply, PVDD-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + + max98504: max98504@31 { + compatible = "maxim,max98504"; + reg = <0x31>; + + DVDD-supply = <&regulator>; + DIOVDD-supply = <&regulator>; + PVDD-supply = <&regulator>; +}; -- 1.9.1

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[alsa-devel] [PATCH] alsa-tools gcc6 narrowing error
by Michel Normand 20 Jun '16

20 Jun '16

Trying to build alsa-tools with gcc6 for PowerPC architecture. It failed as detailed in proposed patch of next mail Michel Normand (1): gcc6 narrowing error hdspmixer/src/channelmap.cxx | 158 +++++++++++++++++++++---------------------- us428control/Cus428Midi.cc | 30 ++++---- 2 files changed, 94 insertions(+), 94 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 2/2] ASoc: wm8731: add normal mode with 12MHz XTAL
by Matt Flax 20 Jun '16

20 Jun '16

I have realised that this second patch may not be necessary. It is also likely that it will skew the actual sample rate. Charles, can you please advise me on the following... I notice that I can probably get the correct sample rates by spoofing the crystal frequencies in my machine driver. For example ... The ratio between the USB and Normal mode fs multipliers with BOSR=0 is : 250/256 = 0.97656 Similarly, I can match this with the following crystal frequency ratio (crystal f in MHz) : 12/12.288 = 0.97656 By induction, I assume that I can spoof a 12.288 clock with BOSR=0 for sample rate = 48kHz, 96 kHz when using a 12 MHz crystal. Again, I can match ratios between the following crystal frequecies (in MHz) : 12/16.9344 = 0.70862 With BOSR = 1, I assume that can get 44.1 kHz and 88.2 kHz because the fs ratio between normal and usb mode is : 272/384 = 0.70833 Now these two ratios are skew by 0.41% which is equvalent to the skew in 12 MHz USB mode clocks of 44.1kHz to 44.118kHz as reported in the wm8731 datasheet. Can you advise me whether my idea of spoofing clocks in the machine driver is perhaps a better way of acquiring the correct sample rates, rather then this patch ? thanks Matt On 10/06/16 10:11, Matt Flax wrote: > This patch allows a machine to set normal mode when using a > 12 MHz crystal. In USB_MODE the 12 MHz crystal signal is output on > the BCLK pin. In NORMAL_MODE the bit clock is output on the BCLK pin. > The previously ignored direction variable input to the > wm8731_set_dai_sysclk function is used to indicate whether crystal > or bit clock is output on the BCLK codec pin. > I have ensured that this does not effect the db1200.c machine driver > (the only other driver to use a 12 MHz crystal in USB_MODE). It also > does not effect the other machine drivers which use the wm8731 codec > as they don't use USB_MODE nor 12 MHz crystals. > > When driving the wm8731 in normal mode with a 12 MHz crystal, the > DSP filters are expected to be effected. If the filters are activated > then it is expected that the filter cut offs are shifted by 2.3% at > 250 fs and 29.1% at 272 fs. The assumption is that the filters are > linear, which implies a linear shift in rate scaling effects the > filter cut off frequencies linearly. If the filters are not active, > there will be no effects. > > Signed-off-by: Matt Flax <flatmax(a)flatmax.org> > --- > sound/soc/codecs/wm8731.c | 14 ++++++++++++-- > 1 file changed, 12 insertions(+), 2 deletions(-) > > diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c > index d18261a..ec98d5a 100644 > --- a/sound/soc/codecs/wm8731.c > +++ b/sound/soc/codecs/wm8731.c > @@ -5,6 +5,7 @@ > * Copyright 2006-12 Wolfson Microelectronics, plc > * > * Author: Richard Purdie <richard(a)openedhand.com> > + * Author: Matt Flax <flatmax(a)flatmax.com> 12Mhz XTAL normal mode and 32bit mode. > * > * Based on wm8753.c by Liam Girdwood > * > @@ -53,6 +54,7 @@ struct wm8731_priv { > int sysclk_type; > int playback_fs; > bool deemph; > + bool bclk_12_mhz; > > struct mutex lock; > }; > @@ -341,8 +343,11 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, > struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); > u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; > int i = get_coeff(wm8731->sysclk, params_rate(params)); > - u16 srate = (coeff_div[i].sr << 2) | > - (coeff_div[i].bosr << 1) | coeff_div[i].usb; > + u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1); > + > + /* this determines whether to output bit or crystal clk */ > + if (wm8731->bclk_12_mhz) > + srate |= coeff_div[i].usb; > > wm8731->playback_fs = params_rate(params); > > @@ -420,6 +425,11 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, > > wm8731->sysclk = freq; > > + if (dir == SND_SOC_CLOCK_IN) /* output the 12 MHz clock */ > + wm8731->bclk_12_mhz = 1; > + else /* SND_SOC_CLOCK_OUT, output the bit clock */ > + wm8731->bclk_12_mhz = 0; > + > snd_soc_dapm_sync(dapm); > > return 0;

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[alsa-devel] [PATCH] ASoC: compress: Pass error out of soc_compr_pointer
by Charles Keepax 20 Jun '16

20 Jun '16

Both soc_compr_pointer and the platform driver pointer callback return ints but current soc_compr_pointer always returns 0. Update this so we return the actual value from the platform driver callback. This doesn't fix any issues simply makes the code more consistent. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-compress.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 875733c..d2df46c 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -530,14 +530,15 @@ static int soc_compr_pointer(struct snd_compr_stream *cstream, { struct snd_soc_pcm_runtime *rtd = cstream->private_data; struct snd_soc_platform *platform = rtd->platform; + int ret = 0; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); if (platform->driver->compr_ops && platform->driver->compr_ops->pointer) - platform->driver->compr_ops->pointer(cstream, tstamp); + ret = platform->driver->compr_ops->pointer(cstream, tstamp); mutex_unlock(&rtd->pcm_mutex); - return 0; + return ret; } static int soc_compr_copy(struct snd_compr_stream *cstream, -- 2.1.4

1 0

[alsa-devel] [PATCH 1/1] ASoC: pcm5102a: Support 352k8 and 384k sample rates
by Clive Messer 19 Jun '16

19 Jun '16

Add support for 352k8 and 384k samples rates to pcm5102a codec driver, with SNDRV_PCM_RATE_KNOT and a constraint list. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/codecs/pcm5102a.c | 38 +++++++++++++++++++++++++++++++++++++- 1 file changed, 37 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index ed51567..53dfc99 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -20,16 +20,52 @@ #include <sound/soc.h> +static const u32 pcm5102a_rates[] = { + 8000, 16000, 32000, 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000, +}; + +static const struct snd_pcm_hw_constraint_list pcm5102a_constraint_rates = { + .count = ARRAY_SIZE(pcm5102a_rates), + .list = pcm5102a_rates, +}; + +static int pcm5102a_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct pcm5102a_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "%s: set rates (8k-384k) constraint\n", __func__); + + ret = snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &pcm5102a_constraint_rates); + if (ret != 0) { + dev_err(codec->dev, "%s: Failed to set rates constraint: %d\n", + __func__, ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_dai_ops pcm5102a_dai_ops = { + .startup = pcm5102a_dai_startup, +}; + static struct snd_soc_dai_driver pcm5102a_dai = { .name = "pcm5102a-hifi", .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_KNOT, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE }, + .ops = &pcm5102a_dai_ops, }; static struct snd_soc_codec_driver soc_codec_dev_pcm5102a; -- 2.5.5

1 1

[alsa-devel] [PATCH 1/1] ASoC: pcm512x: Add 352k8 sample rate support
by Clive Messer 19 Jun '16

19 Jun '16

Add 352k8 sample rate to the slave constraint list. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/codecs/pcm512x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 047c489..eca6648 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -480,7 +480,7 @@ static unsigned long pcm512x_ncp_target(struct pcm512x_priv *pcm512x, static const u32 pcm512x_dai_rates[] = { 8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000, - 88200, 96000, 176400, 192000, 384000, + 88200, 96000, 176400, 192000, 352800, 384000, }; static const struct snd_pcm_hw_constraint_list constraints_slave = { -- 2.5.5

1 0

[alsa-devel] [asoc:for-next 8/14] cygnus-pcm.c:undefined reference to `bad_dma_ops'
by kbuild test robot 19 Jun '16

19 Jun '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 1a7d46a9dfeb578ece0a707f9d97f7bd097cd4a4 commit: 49bf948b5c0b766a8ecd30537aae302997a23487 [8/14] Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/cs53l30' and 'asoc/topic/cygnus' into asoc-next config: m32r-allyesconfig (attached as .config) compiler: m32r-linux-gcc (GCC) 4.9.0 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 49bf948b5c0b766a8ecd30537aae302997a23487 # save the attached .config to linux build tree make.cross ARCH=m32r All errors (new ones prefixed by >>): drivers/built-in.o: In function `zynq_fpga_ops_write': zynq-fpga.c:(.text+0xeff4ec): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff4f4): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff614): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcb4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcbc): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): undefined reference to `dma_common_mmap' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): relocation truncated to fit: R_M32R_26_PCREL_RELA against undefined symbol `dma_common_mmap' sound/built-in.o: In function `cygnus_pcm_preallocate_dma_buffer': >> cygnus-pcm.c:(.text+0x10d2b0): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d2b4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `cygnus_dma_free_dma_buffers': cygnus-pcm.c:(.text+0x10d46c): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d470): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d47c): undefined reference to `bad_dma_ops' sound/built-in.o:cygnus-pcm.c:(.text+0x10d480): more undefined references to `bad_dma_ops' follow --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ALSA: seq_timer: use monotonic times internally
by Arnd Bergmann 17 Jun '16

17 Jun '16

The sequencer client manager reports timestamps in units of unsigned 32-bit seconds/nanoseconds, but that does not suffer from the y2038 overflow because it stores only the delta since the 'last_update' time was recorded. However, the use of the do_gettimeofday() function is problematic and we have to replace it to avoid the overflow on on 32-bit architectures. This uses 'struct timespec64' to record 'last_update', and changes the code to use monotonic timestamps that do not suffer from leap seconds and settimeofday updates. As a side-effect, the code can now use the timespec64_sub() helper and become more readable and also avoid a multiplication to convert from microseconds to nanoseconds. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/core/seq/seq_timer.c | 23 +++++++++-------------- sound/core/seq/seq_timer.h | 2 +- 2 files changed, 10 insertions(+), 15 deletions(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index 293104926098..dcc102813aef 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -165,7 +165,7 @@ static void snd_seq_timer_interrupt(struct snd_timer_instance *timeri, snd_seq_timer_update_tick(&tmr->tick, resolution); /* register actual time of this timer update */ - do_gettimeofday(&tmr->last_update); + ktime_get_ts64(&tmr->last_update); spin_unlock_irqrestore(&tmr->lock, flags); @@ -392,7 +392,7 @@ static int seq_timer_start(struct snd_seq_timer *tmr) return -EINVAL; snd_timer_start(tmr->timeri, tmr->ticks); tmr->running = 1; - do_gettimeofday(&tmr->last_update); + ktime_get_ts64(&tmr->last_update); return 0; } @@ -420,7 +420,7 @@ static int seq_timer_continue(struct snd_seq_timer *tmr) } snd_timer_start(tmr->timeri, tmr->ticks); tmr->running = 1; - do_gettimeofday(&tmr->last_update); + ktime_get_ts64(&tmr->last_update); return 0; } @@ -444,17 +444,12 @@ snd_seq_real_time_t snd_seq_timer_get_cur_time(struct snd_seq_timer *tmr) spin_lock_irqsave(&tmr->lock, flags); cur_time = tmr->cur_time; if (tmr->running) { - struct timeval tm; - int usec; - do_gettimeofday(&tm); - usec = (int)(tm.tv_usec - tmr->last_update.tv_usec); - if (usec < 0) { - cur_time.tv_nsec += (1000000 + usec) * 1000; - cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec - 1; - } else { - cur_time.tv_nsec += usec * 1000; - cur_time.tv_sec += tm.tv_sec - tmr->last_update.tv_sec; - } + struct timespec64 tm; + + ktime_get_ts64(&tm); + tm = timespec64_sub(tm, tmr->last_update); + cur_time.tv_nsec = tm.tv_nsec; + cur_time.tv_sec = tm.tv_sec; snd_seq_sanity_real_time(&cur_time); } spin_unlock_irqrestore(&tmr->lock, flags); diff --git a/sound/core/seq/seq_timer.h b/sound/core/seq/seq_timer.h index 88dfb71805ae..9506b661fe5b 100644 --- a/sound/core/seq/seq_timer.h +++ b/sound/core/seq/seq_timer.h @@ -52,7 +52,7 @@ struct snd_seq_timer { unsigned int skew; unsigned int skew_base; - struct timeval last_update; /* time of last clock update, used for interpolation */ + struct timespec64 last_update; /* time of last clock update, used for interpolation */ spinlock_t lock; }; -- 2.9.0

2 1

[alsa-devel] [PATCH] ALSA: ctxfi: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 17 Jun '16

17 Jun '16

For readability and to allow for structure randomisation, replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/pci/ctxfi/cthw20k2.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 9dc2950..6414ecf 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1615,23 +1615,23 @@ static int hw_dac_init(struct hw *hw, const struct dac_conf *info) int i; struct regs_cs4382 cs_read = {0}; struct regs_cs4382 cs_def = { - 0x00000001, /* Mode Control 1 */ - 0x00000000, /* Mode Control 2 */ - 0x00000084, /* Mode Control 3 */ - 0x00000000, /* Filter Control */ - 0x00000000, /* Invert Control */ - 0x00000024, /* Mixing Control Pair 1 */ - 0x00000000, /* Vol Control A1 */ - 0x00000000, /* Vol Control B1 */ - 0x00000024, /* Mixing Control Pair 2 */ - 0x00000000, /* Vol Control A2 */ - 0x00000000, /* Vol Control B2 */ - 0x00000024, /* Mixing Control Pair 3 */ - 0x00000000, /* Vol Control A3 */ - 0x00000000, /* Vol Control B3 */ - 0x00000024, /* Mixing Control Pair 4 */ - 0x00000000, /* Vol Control A4 */ - 0x00000000 /* Vol Control B4 */ + .mode_control_1 = 0x00000001, /* Mode Control 1 */ + .mode_control_2 = 0x00000000, /* Mode Control 2 */ + .mode_control_3 = 0x00000084, /* Mode Control 3 */ + .filter_control = 0x00000000, /* Filter Control */ + .invert_control = 0x00000000, /* Invert Control */ + .mix_control_P1 = 0x00000024, /* Mixing Control Pair 1 */ + .vol_control_A1 = 0x00000000, /* Vol Control A1 */ + .vol_control_B1 = 0x00000000, /* Vol Control B1 */ + .mix_control_P2 = 0x00000024, /* Mixing Control Pair 2 */ + .vol_control_A2 = 0x00000000, /* Vol Control A2 */ + .vol_control_B2 = 0x00000000, /* Vol Control B2 */ + .mix_control_P3 = 0x00000024, /* Mixing Control Pair 3 */ + .vol_control_A3 = 0x00000000, /* Vol Control A3 */ + .vol_control_B3 = 0x00000000, /* Vol Control B3 */ + .mix_control_P4 = 0x00000024, /* Mixing Control Pair 4 */ + .vol_control_A4 = 0x00000000, /* Vol Control A4 */ + .vol_control_B4 = 0x00000000 /* Vol Control B4 */ }; if (hw->model == CTSB1270) { -- 1.9.1

2 1

[alsa-devel] [PATCH v2] ALSA: usb-audio: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 17 Jun '16

17 Jun '16

To allow for structure randomisation, replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- Changes in v2: -Modified commit message sound/usb/mixer_maps.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1f8fb0d9..9038b2e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -107,8 +107,10 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov <bin(a)bash.info> */ -static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ -static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; + /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; + /* just guess */ static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ -- 1.9.1

2 1

[alsa-devel] [PATCH] oss: avoid time_t usage
by Arnd Bergmann 17 Jun '16

17 Jun '16

We want to remove all time_t users from the kernel because of y2038 compatibility. This particular instance does not even use time_t to store a seconds value, so we can simply use 'unsigned int', which seems more fitting anywhere. The same code is used in two OSS files. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/oss/sound_timer.c | 2 +- sound/oss/sys_timer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/oss/sound_timer.c b/sound/oss/sound_timer.c index 8021c85f076d..3a444a6f10eb 100644 --- a/sound/oss/sound_timer.c +++ b/sound/oss/sound_timer.c @@ -17,7 +17,7 @@ #include "sound_config.h" static volatile int initialized, opened, tmr_running; -static volatile time_t tmr_offs, tmr_ctr; +static volatile unsigned int tmr_offs, tmr_ctr; static volatile unsigned long ticks_offs; static volatile int curr_tempo, curr_timebase; static volatile unsigned long curr_ticks; diff --git a/sound/oss/sys_timer.c b/sound/oss/sys_timer.c index 2226dda0eff0..d17019d25b99 100644 --- a/sound/oss/sys_timer.c +++ b/sound/oss/sys_timer.c @@ -19,7 +19,7 @@ #include "sound_config.h" static volatile int opened, tmr_running; -static volatile time_t tmr_offs, tmr_ctr; +static volatile unsigned int tmr_offs, tmr_ctr; static volatile unsigned long ticks_offs; static volatile int curr_tempo, curr_timebase; static volatile unsigned long curr_ticks; -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: max9867: Fix unix permissions for source files.
by Enric Balletbo i Serra 17 Jun '16

17 Jun '16

Change file permissions of source files max9867.c/h from 0755 to 0644. Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/codecs/max9867.c | 0 sound/soc/codecs/max9867.h | 0 2 files changed, 0 insertions(+), 0 deletions(-) mode change 100755 => 100644 sound/soc/codecs/max9867.c mode change 100755 => 100644 sound/soc/codecs/max9867.h diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c old mode 100755 new mode 100644 diff --git a/sound/soc/codecs/max9867.h b/sound/soc/codecs/max9867.h old mode 100755 new mode 100644 -- 2.1.0

1 0

[alsa-devel] [alsa-lib][PATCH v4] ctl: add test program for control element set
by Takashi Sakamoto 17 Jun '16

17 Jun '16

The feature of control element set has been abandoned for a long time since firstly introduced in 2003. Furthermore, there's few applications to utilize this feature. These situations bring a hard work to persons who need the feature. Especially, a lack of test program make it harder to fix much bugs in this feature. This commit adds a test program as a sample of the feature. This program adds element sets of each element type to 'hw:0' in this order; boolean, integer, enumerated, bytes, IEC958 and integer64. Each iteration includes below scheme: 1. add an element set with 900 elements. Each of them has maximum number of members allowed by ALSA ctl core. 2. check all of events generated by above operation. 3. retrieve information of each element, then validate it. 4. unlock each member of all elements because they're initially locked. 5. write to all of members in all elements and read. 6. check all of events generated by above operation. 7. write information for threshold level to the element set and read it. 8. check all of events generated by above operation. 9. remove the element set. 10.check all of events generated by above operation. When any of these operations fail, it means regression occurs. Then, added elements still remain in a certain sound card. In this case, unloading drivers corresponding to the card is an easy way to recover. Besides, this program doesn't perform below element operations of ALSA ctl feature: - list - lock - replace Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 573 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 577 insertions(+), 1 deletion(-) create mode 100644 test/user-ctl-element-set.c diff --git a/test/Makefile.am b/test/Makefile.am index ce6e521..cbf2ab4 100644 --- a/test/Makefile.am +++ b/test/Makefile.am @@ -3,7 +3,7 @@ SUBDIRS=. lsb check_PROGRAMS=control pcm pcm_min latency seq \ playmidi1 timer rawmidi midiloop \ oldapi queue_timer namehint client_event_filter \ - chmap audio_time + chmap audio_time user-ctl-element-set control_LDADD=../src/libasound.la pcm_LDADD=../src/libasound.la @@ -24,6 +24,9 @@ code_CFLAGS=-Wall -pipe -g -O2 chmap_LDADD=../src/libasound.la audio_time_LDADD=../src/libasound.la +user_ctl_element_set_LDADD=../src/libasound.la +user_ctl_element_set_CFLAGS=-Wall -g + AM_CPPFLAGS=-I$(top_srcdir)/include AM_CFLAGS=-Wall -pipe -g diff --git a/test/user-ctl-element-set.c b/test/user-ctl-element-set.c new file mode 100644 index 0000000..09c7929 --- /dev/null +++ b/test/user-ctl-element-set.c @@ -0,0 +1,573 @@ +/* + * user-control-element-set.c - a program to test in-kernel implementation of + * user-defined control element set. + * + * Copyright (c) 2015-2016 Takashi Sakamoto + * + * Licensed under the terms of the GNU General Public License, version 2. + */ + +#include "../include/asoundlib.h" + +struct elem_set_trial { + snd_ctl_t *handle; + + snd_ctl_elem_type_t type; + unsigned int member_count; + unsigned int element_count; + + snd_ctl_elem_id_t *id; + + int (*add_elem_set)(struct elem_set_trial *trial); + int (*check_elem_props)(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info); + void (*change_elem_members)(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data); +}; + +/* Operations for elements in an element set with boolean type. */ +static int add_bool_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_boolean_set(trial->handle, trial->id, + trial->element_count, trial->member_count); +} + +static void change_bool_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + int val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_boolean(elem_data, i); + snd_ctl_elem_value_set_boolean(elem_data, i, !val); + } +} + +/* Operations for elements in an element set with integer type. */ +static int add_int_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_integer_set(trial->handle, trial->id, + trial->element_count, trial->member_count, + 0, 99, 1); +} + +static int check_int_elem_props(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info) +{ + if (snd_ctl_elem_info_get_min(info) != 0) + return -EIO; + if (snd_ctl_elem_info_get_max(info) != 99) + return -EIO; + if (snd_ctl_elem_info_get_step(info) != 1) + return -EIO; + + return 0; +} + +static void change_int_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + long val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_integer(elem_data, i); + snd_ctl_elem_value_set_integer(elem_data, i, ++val); + } +} + +/* Operations for elements in an element set with enumerated type. */ +static const char *const labels[] = { + "trusty", + "utopic", + "vivid", + "willy", + "xenial", +}; + +static int add_enum_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_enumerated_set(trial->handle, trial->id, + trial->element_count, trial->member_count, + sizeof(labels) / sizeof(labels[0]), + labels); +} + +static int check_enum_elem_props(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info) +{ + unsigned int items; + unsigned int i; + const char *label; + int err; + + items = snd_ctl_elem_info_get_items(info); + if (items != sizeof(labels) / sizeof(labels[0])) + return -EIO; + + /* Enumerate and validate all of labels registered to this element. */ + for (i = 0; i < items; ++i) { + snd_ctl_elem_info_set_item(info, i); + err = snd_ctl_elem_info(trial->handle, info); + if (err < 0) + return err; + + label = snd_ctl_elem_info_get_item_name(info); + if (strncmp(label, labels[i], strlen(labels[i])) != 0) + return -EIO; + } + + return 0; +} + +static void change_enum_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + unsigned int val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_enumerated(elem_data, i); + snd_ctl_elem_value_set_enumerated(elem_data, i, ++val); + } +} + +/* Operations for elements in an element set with bytes type. */ +static int add_bytes_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_bytes_set(trial->handle, trial->id, + trial->element_count, trial->member_count); +} + +static void change_bytes_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + unsigned char val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_byte(elem_data, i); + snd_ctl_elem_value_set_byte(elem_data, i, ++val); + } +} + +/* Operations for elements in an element set with iec958 type. */ +static int add_iec958_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_iec958(trial->handle, trial->id); +} + +static void change_iec958_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + snd_aes_iec958_t data; + + /* To suppress GCC warnings. */ + trial->element_count = 1; + + snd_ctl_elem_value_get_iec958(elem_data, &data); + /* This is an arbitrary number. */ + data.pad = 10; + snd_ctl_elem_value_set_iec958(elem_data, &data); +} + +/* Operations for elements in an element set with integer64 type. */ +static int add_int64_elem_set(struct elem_set_trial *trial) +{ + return snd_ctl_elem_add_integer64_set(trial->handle, trial->id, + trial->element_count, trial->member_count, + 100, 10000, 30); +} + +static int check_int64_elem_props(struct elem_set_trial *trial, + snd_ctl_elem_info_t *info) +{ + if (snd_ctl_elem_info_get_min64(info) != 100) + return -EIO; + if (snd_ctl_elem_info_get_max64(info) != 10000) + return -EIO; + if (snd_ctl_elem_info_get_step64(info) != 30) + return -EIO; + + return 0; +} + +static void change_int64_elem_members(struct elem_set_trial *trial, + snd_ctl_elem_value_t *elem_data) +{ + long long val; + unsigned int i; + + for (i = 0; i < trial->member_count; ++i) { + val = snd_ctl_elem_value_get_integer64(elem_data, i); + snd_ctl_elem_value_set_integer64(elem_data, i, ++val); + } +} + +/* Common operations. */ +static int add_elem_set(struct elem_set_trial *trial) +{ + char name[64] = {0}; + + snprintf(name, 64, "userspace-control-element-%s", + snd_ctl_elem_type_name(trial->type)); + + memset(trial->id, 0, snd_ctl_elem_id_sizeof()); + snd_ctl_elem_id_set_interface(trial->id, SND_CTL_ELEM_IFACE_MIXER); + snd_ctl_elem_id_set_name(trial->id, name); + + return trial->add_elem_set(trial); +} + +static int check_event(struct elem_set_trial *trial, unsigned int mask, + unsigned int expected_count) +{ + struct pollfd pfds; + int count; + unsigned short revents; + snd_ctl_event_t *event; + int err; + + snd_ctl_event_alloca(&event); + + if (snd_ctl_poll_descriptors_count(trial->handle) != 1) + return -ENXIO; + + if (snd_ctl_poll_descriptors(trial->handle, &pfds, 1) != 1) + return -ENXIO; + + while (expected_count > 0) { + count = poll(&pfds, 1, 1000); + if (count < 0) + return errno; + /* Some events are already supplied. */ + if (count == 0) + return -ETIMEDOUT; + + err = snd_ctl_poll_descriptors_revents(trial->handle, &pfds, + count, &revents); + if (err < 0) + return err; + if (revents & POLLERR) + return -EIO; + if (!(revents & POLLIN)) + continue; + + err = snd_ctl_read(trial->handle, event); + if (err < 0) + return err; + if (snd_ctl_event_get_type(event) != SND_CTL_EVENT_ELEM) + continue; + /* + * I expect each event is generated separately to the same + * element. + */ + if (!(snd_ctl_event_elem_get_mask(event) & mask)) + continue; + --expected_count; + } + + if (expected_count != 0) + return -EIO; + + return 0; +} + +static int check_elem_set_props(struct elem_set_trial *trial) +{ + snd_ctl_elem_id_t *id; + snd_ctl_elem_info_t *info; + unsigned int numid; + unsigned int i; + int err; + + snd_ctl_elem_id_alloca(&id); + snd_ctl_elem_info_alloca(&info); + + snd_ctl_elem_info_set_id(info, trial->id); + numid = snd_ctl_elem_info_get_numid(info); + + for (i = 0; i < trial->element_count; ++i) { + snd_ctl_elem_info_set_numid(info, numid + i); + + err = snd_ctl_elem_info(trial->handle, info); + if (err < 0) + return err; + + /* Check some common properties. */ + if (snd_ctl_elem_info_get_type(info) != trial->type) + return -EIO; + if (snd_ctl_elem_info_get_count(info) != trial->member_count) + return -EIO; + + /* + * In a case of IPC, this is the others. But in this case, + * it's myself. + */ + if (snd_ctl_elem_info_get_owner(info) != getpid()) + return -EIO; + + /* + * Just adding an element set by userspace applications, + * included elements are initially locked. + */ + if (!snd_ctl_elem_info_is_locked(info)) + return -EIO; + + /* Check type-specific properties. */ + if (trial->check_elem_props != NULL) { + err = trial->check_elem_props(trial, info); + if (err < 0) + return err; + } + + snd_ctl_elem_info_get_id(info, id); + err = snd_ctl_elem_unlock(trial->handle, id); + if (err < 0) + return err; + } + + return 0; +} + +static int check_elems(struct elem_set_trial *trial) +{ + snd_ctl_elem_value_t *data; + unsigned int numid; + unsigned int i; + int err; + + snd_ctl_elem_value_alloca(&data); + + snd_ctl_elem_value_set_id(data, trial->id); + + /* + * Get the value of numid field in identical information for the first + * element of this element set. + */ + numid = snd_ctl_elem_value_get_numid(data); + + for (i = 0; i < trial->element_count; ++i) { + /* + * Here, I increment numid, while we can also increment offset + * to enumerate each element in this element set. + */ + snd_ctl_elem_value_set_numid(data, numid + i); + + err = snd_ctl_elem_read(trial->handle, data); + if (err < 0) + return err; + + /* Change members of an element in this element set. */ + trial->change_elem_members(trial, data); + + err = snd_ctl_elem_write(trial->handle, data); + if (err < 0) + return err; + } + + return 0; +} + +static int check_tlv(struct elem_set_trial *trial) +{ + unsigned int orig[8], curr[8]; + int err; + + /* + * See a layout of 'struct snd_ctl_tlv'. I don't know the reason to + * construct this buffer with the same layout. It should be abstracted + * inner userspace library... + */ + orig[0] = snd_ctl_elem_id_get_numid(trial->id); + orig[1] = 6 * sizeof(orig[0]); + orig[2] = 'a'; + orig[3] = 'b'; + orig[4] = 'c'; + orig[5] = 'd'; + orig[6] = 'e'; + orig[7] = 'f'; + + /* + * In in-kernel implementation, write and command operations are the + * same for an element set added by userspace applications. Here, I + * use write. + */ + err = snd_ctl_elem_tlv_write(trial->handle, trial->id, + (const unsigned int *)orig); + if (err < 0) + return err; + + err = snd_ctl_elem_tlv_read(trial->handle, trial->id, curr, + sizeof(curr)); + if (err < 0) + return err; + + if (memcmp(curr, orig, sizeof(orig)) != 0) + return -EIO; + + return 0; +} + +int main(void) +{ + struct elem_set_trial trial = {0}; + unsigned int i; + int err; + + snd_ctl_elem_id_alloca(&trial.id); + + err = snd_ctl_open(&trial.handle, "hw:0", 0); + if (err < 0) + return EXIT_FAILURE; + + err = snd_ctl_subscribe_events(trial.handle, 1); + if (err < 0) + return EXIT_FAILURE; + + /* Test all of types. */ + for (i = 0; i < SND_CTL_ELEM_TYPE_LAST; ++i) { + trial.type = i + 1; + + /* Assign type-dependent operations. */ + switch (trial.type) { + case SND_CTL_ELEM_TYPE_BOOLEAN: + trial.element_count = 900; + trial.member_count = 128; + trial.add_elem_set = add_bool_elem_set; + trial.check_elem_props = NULL; + trial.change_elem_members = change_bool_elem_members; + break; + case SND_CTL_ELEM_TYPE_INTEGER: + trial.element_count = 900; + trial.member_count = 128; + trial.add_elem_set = add_int_elem_set; + trial.check_elem_props = check_int_elem_props; + trial.change_elem_members = change_int_elem_members; + break; + case SND_CTL_ELEM_TYPE_ENUMERATED: + trial.element_count = 900; + trial.member_count = 128; + trial.add_elem_set = add_enum_elem_set; + trial.check_elem_props = check_enum_elem_props; + trial.change_elem_members = change_enum_elem_members; + break; + case SND_CTL_ELEM_TYPE_BYTES: + trial.element_count = 900; + trial.member_count = 512; + trial.add_elem_set = add_bytes_elem_set; + trial.check_elem_props = NULL; + trial.change_elem_members = change_bytes_elem_members; + break; + case SND_CTL_ELEM_TYPE_IEC958: + trial.element_count = 1; + trial.member_count = 1; + trial.add_elem_set = add_iec958_elem_set; + trial.check_elem_props = NULL; + trial.change_elem_members = change_iec958_elem_members; + break; + case SND_CTL_ELEM_TYPE_INTEGER64: + default: + trial.element_count = 900; + trial.member_count = 64; + trial.add_elem_set = add_int64_elem_set; + trial.check_elem_props = check_int64_elem_props; + trial.change_elem_members = change_int64_elem_members; + break; + } + + /* Test an operation to add an element set. */ + err = add_elem_set(&trial); + if (err < 0) { + printf("Fail to add an element set with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_ADD, + trial.element_count); + if (err < 0) { + printf("Fail to check some events to add elements with " + "%s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + + /* Check properties of each element in this element set. */ + err = check_elem_set_props(&trial); + if (err < 0) { + printf("Fail to check propetries of each element with " + "%s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + + /* + * Test operations to change the state of members in each + * element in the element set. + */ + err = check_elems(&trial); + if (err < 0) { + printf("Fail to change status of each element with %s " + "type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_VALUE, + trial.element_count); + if (err < 0) { + printf("Fail to check some events to change status of " + "each elements with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + + /* + * Test an operation to change threshold data of this element set, + * except for IEC958 type. + */ + if (trial.type != SND_CTL_ELEM_TYPE_IEC958) { + err = check_tlv(&trial); + if (err < 0) { + printf("Fail to change threshold level of an " + "element set with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_TLV, 1); + if (err < 0) { + printf("Fail to check an event to change " + "threshold level of an an element set " + "with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + } + + /* Test an operation to remove elements in this element set. */ + err = snd_ctl_elem_remove(trial.handle, trial.id); + if (err < 0) { + printf("Fail to remove elements with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + err = check_event(&trial, SND_CTL_EVENT_MASK_REMOVE, + trial.element_count); + if (err < 0) { + printf("Fail to check some events to remove each " + "element with %s type.\n", + snd_ctl_elem_type_name(trial.type)); + break; + } + } + + if (err < 0) { + printf("%s\n", snd_strerror(err)); + + /* To ensure. */ + snd_ctl_elem_remove(trial.handle, trial.id); + return EXIT_FAILURE; + } + + return EXIT_SUCCESS; +} -- 2.7.4

2 1

[alsa-devel] [PATCH v5] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 17 Jun '16

17 Jun '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v5: - fix the build warning for missing I2C/CLKDEV_LOOKUP dependency Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 339 ++++++++++++++++++++ 4 files changed, 367 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..6d39032 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..cc9b943 --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,339 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speakers"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_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_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + snd_soc_dapm_disable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speakers"); + snd_soc_dapm_sync(dapm); + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_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 *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_sync(dapm); + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_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 *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + int mclk, ret; + + snd_soc_dapm_disable_pin(dapm, "Speakers"); + snd_soc_dapm_disable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_sync(dapm); + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

4 5

[alsa-devel] [PATCH] ALSA: hdac_regmap - fix the register access for runtime PM
by Jaroslav Kysela 17 Jun '16

17 Jun '16

Call path: 1) snd_hdac_power_up_pm() 2) snd_hdac_power_up() 3) pm_runtime_get_sync() 4) __pm_runtime_resume() 5) rpm_resume() The rpm_resume() returns 1 when the device is already active. Because the return value is unmodified, the hdac regmap read/write functions should allow this value for the retry I/O operation, too. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/hda/hdac_regmap.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 87041dd..47a358f 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -444,7 +444,7 @@ int snd_hdac_regmap_write_raw(struct hdac_device *codec, unsigned int reg, err = reg_raw_write(codec, reg, val); if (err == -EAGAIN) { err = snd_hdac_power_up_pm(codec); - if (!err) + if (err >= 0) err = reg_raw_write(codec, reg, val); snd_hdac_power_down_pm(codec); } @@ -470,7 +470,7 @@ static int __snd_hdac_regmap_read_raw(struct hdac_device *codec, err = reg_raw_read(codec, reg, val, uncached); if (err == -EAGAIN) { err = snd_hdac_power_up_pm(codec); - if (!err) + if (err >= 0) err = reg_raw_read(codec, reg, val, uncached); snd_hdac_power_down_pm(codec); } -- 2.5.5

2 1

[alsa-devel] [PATCH] ASoC fix up SND_SOC_WM8985 dependency
by Arnd Bergmann 17 Jun '16

17 Jun '16

I just added an I2C dependency to the wm8985 driver to work around a build failure, but it turns out that was premature: we actually need to depend on SND_SOC_I2C_AND_SPI, as the driver can work with either of the two, and we only need to prevent a configuration that has I2C=m and SND_SOC_WM8985=y. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 05252513fbb9 ("ASoC: wm8985: add i2c dependency") --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d0e8150c7170..60f8f13f0d34 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -994,7 +994,7 @@ config SND_SOC_WM8983 config SND_SOC_WM8985 tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" - depends on I2C + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8988 tristate -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoc: wm8731: add 32bit mode.
by Matt Flax 17 Jun '16

17 Jun '16

This patch adds 32 bit word capability to the wm8731 driver. The wm8731 codec is capable of handling 32 bit word sizes, however that has not previously been activated in the codec driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/wm8731.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4bcf5f8..d18261a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -358,6 +358,9 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, case 24: iface |= 0x0008; break; + case 32: + iface |= 0x000c; + break; } wm8731_set_deemph(codec); @@ -541,7 +544,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream, #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { .startup = wm8731_startup, -- 2.7.4

3 2

[alsa-devel] [PATCH 0/3] ASoC: ak4613/wm8940: Regmap cache fixes
by Geert Uytterhoeven 17 Jun '16

17 Jun '16

Hi Liam, Mark, This series contains 3 fixes for regmap cache handling in the ak4613 and wm8940 drivers. All fixes are based on recent similar fixes for the ak4642 driver. The first patch fixes a crash I've seen on r8a7795/Salvator-X. The last patch is completely untested. The issue was discovered using 'git grep -Lw cache_type -- $(git grep -lw regcache_sync -- $(git grep -lw regmap_config))'' Thanks! Geert Uytterhoeven (3): ASoC: ak4613: Enable cache usage to fix crashes on resume ASoC: ak4613: Implement suspend callback ASoC: wm8940: Enable cache usage to fix crashes on resume sound/soc/codecs/ak4613.c | 13 ++++++++++++- sound/soc/codecs/wm8940.c | 1 + 2 files changed, 13 insertions(+), 1 deletion(-) -- 1.9.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

3 7

[alsa-devel] [PATCH] snd: soc: cx20442 codec: set tty->receiver_room in v253_open
by Janusz Krzysztofik 17 Jun '16

17 Jun '16

Commit 79901317ce80 ("n_tty: Don't flush buffer when closing ldisc"), introduced in v3.10, revealed a bug in the cx20442 codec driver which has never been setting tty->receive_room on line discipline open as it should from the beginning. Fix it. Created and tested on Amstrad Delta against Linux-4.7-rc3 Signed-off-by: Janusz Krzysztofik <jmkrzyszt(a)gmail.com> --- sound/soc/codecs/cx20442.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d6f4abb..fb3885f 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -226,6 +226,7 @@ static int v253_open(struct tty_struct *tty) if (!tty->disc_data) return -ENODEV; + tty->receive_room = 16; if (tty->ops->write(tty, v253_init, len) != len) { ret = -EIO; goto err; -- 2.7.3

2 2

[alsa-devel] [PATCH] ASoC: codecs: Remove "select REGMAP_SPI/REGMAP_I2C"
by Alexander Shiyan 17 Jun '16

17 Jun '16

REGMAP_SPI and REGMAP_I2C is selected automatically by SND_SOC, so no additional actions is required. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/codecs/Kconfig | 21 --------------------- 1 file changed, 21 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58..1e03fa0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -288,12 +288,10 @@ config SND_SOC_ADAU1761 config SND_SOC_ADAU1761_I2C tristate select SND_SOC_ADAU1761 - select REGMAP_I2C config SND_SOC_ADAU1761_SPI tristate select SND_SOC_ADAU1761 - select REGMAP_SPI config SND_SOC_ADAU1781 select SND_SOC_ADAU17X1 @@ -302,12 +300,10 @@ config SND_SOC_ADAU1781 config SND_SOC_ADAU1781_I2C tristate select SND_SOC_ADAU1781 - select REGMAP_I2C config SND_SOC_ADAU1781_SPI tristate select SND_SOC_ADAU1781 - select REGMAP_SPI config SND_SOC_ADAU1977 tristate @@ -315,12 +311,10 @@ config SND_SOC_ADAU1977 config SND_SOC_ADAU1977_SPI tristate select SND_SOC_ADAU1977 - select REGMAP_SPI config SND_SOC_ADAU1977_I2C tristate select SND_SOC_ADAU1977 - select REGMAP_I2C config SND_SOC_ADAV80X tristate @@ -403,7 +397,6 @@ config SND_SOC_CS42L73 config SND_SOC_CS4265 tristate "Cirrus Logic CS4265 CODEC" depends on I2C - select REGMAP_I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 @@ -425,13 +418,11 @@ config SND_SOC_CS4271_I2C tristate "Cirrus Logic CS4271 CODEC (I2C)" depends on I2C select SND_SOC_CS4271 - select REGMAP_I2C config SND_SOC_CS4271_SPI tristate "Cirrus Logic CS4271 CODEC (SPI)" depends on SPI_MASTER select SND_SOC_CS4271 - select REGMAP_SPI config SND_SOC_CS42XX8 tristate @@ -440,7 +431,6 @@ config SND_SOC_CS42XX8_I2C tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" depends on I2C select SND_SOC_CS42XX8 - select REGMAP_I2C # Cirrus Logic CS4349 HiFi DAC config SND_SOC_CS4349 @@ -580,13 +570,11 @@ config SND_SOC_PCM3168A_I2C tristate "Texas Instruments PCM3168A CODEC - I2C" depends on I2C select SND_SOC_PCM3168A - select REGMAP_I2C config SND_SOC_PCM3168A_SPI tristate "Texas Instruments PCM3168A CODEC - SPI" depends on SPI_MASTER select SND_SOC_PCM3168A - select REGMAP_SPI config SND_SOC_PCM5102A tristate @@ -598,13 +586,11 @@ config SND_SOC_PCM512x_I2C tristate "Texas Instruments PCM512x CODECs - I2C" depends on I2C select SND_SOC_PCM512x - select REGMAP_I2C config SND_SOC_PCM512x_SPI tristate "Texas Instruments PCM512x CODECs - SPI" depends on SPI_MASTER select SND_SOC_PCM512x - select REGMAP_SPI config SND_SOC_RL6231 tristate @@ -668,7 +654,6 @@ config SND_SOC_RT5670 config SND_SOC_RT5677 tristate - select REGMAP_I2C select REGMAP_IRQ config SND_SOC_RT5677_SPI @@ -715,13 +700,11 @@ config SND_SOC_SSM2602_SPI tristate "Analog Devices SSM2602 CODEC - SPI" depends on SPI_MASTER select SND_SOC_SSM2602 - select REGMAP_SPI config SND_SOC_SSM2602_I2C tristate "Analog Devices SSM2602 CODEC - I2C" depends on I2C select SND_SOC_SSM2602 - select REGMAP_I2C config SND_SOC_SSM4567 tristate "Analog Devices ssm4567 amplifier driver support" @@ -730,7 +713,6 @@ config SND_SOC_SSM4567 config SND_SOC_STA32X tristate "STA326, STA328 and STA329 speaker amplifier" depends on I2C - select REGMAP_I2C config SND_SOC_STA350 tristate "STA350 speaker amplifier" @@ -788,7 +770,6 @@ config SND_SOC_TLV320AIC26 config SND_SOC_TLV320AIC31XX tristate "Texas Instruments TLV320AIC31xx CODECs" depends on I2C - select REGMAP_I2C config SND_SOC_TLV320AIC32X4 tristate @@ -918,13 +899,11 @@ config SND_SOC_WM8804_I2C tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver I2C" depends on I2C select SND_SOC_WM8804 - select REGMAP_I2C config SND_SOC_WM8804_SPI tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver SPI" depends on SPI_MASTER select SND_SOC_WM8804 - select REGMAP_SPI config SND_SOC_WM8900 tristate -- 2.4.9

3 2

[alsa-devel] [alsa-lib][PATCH 0/9 v3] ctl: add APIs for control element set
by Takashi Sakamoto 16 Jun '16

16 Jun '16

Hi, This patchset is a revised version of below one: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109041.html Changes: - Drop a patch for deprecating old APIs. - In detail, see our discussion: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-June/109117.html - Revise comments for TLV feature. - I realized it better to use vendor-neutral representation, after reading this discussion: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069483.h… Again, ALSA developers in PulseAudio side, test program in the last patch causes an abort to pulseaudio process as a result of kicking assertion of hcontrol in alsa-lib. In detail, please refer to my previous patchset. Regards Takashi Sakamoto (9): ctl: add an overview for design of ALSA control interface ctl: improve comments for handling element data ctl: add functions to add an element set ctl: improve comments for API to add an element of IEC958 type ctl: change former APIs as wrapper functions of element set APIs pcm: use new APIs to add a control element set for softvol plugin ctl: add explanation about threshold level feature ctl: improve API documentation for threshold level operations ctl: add test program for control element set include/control.h | 21 ++ src/control/control.c | 813 ++++++++++++++++++++++++++++++++------------ src/pcm/pcm_softvol.c | 7 +- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 561 ++++++++++++++++++++++++++++++ 5 files changed, 1189 insertions(+), 218 deletions(-) create mode 100644 test/user-ctl-element-set.c -- 2.7.4

2 12

[alsa-devel] [PATCH v3 0/2] ASoC: Add support to Qualcomm msm8916-wcd codec
by Srinivas Kandagatla 16 Jun '16

16 Jun '16

This patchset aims at adding msm8916-wcd codec support. msm8916-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.6-rc2 with lpass driver capture support patches for Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Changes since RFC v2: - Fixed error handling on regluator_enable suggested by Kenneth. - Fixed compatible strings suggested by Kenneth - make max/min rates inline with actual supported rates, spotted by Kenneth - Few minor suggestions by Kenneth. Changes since RFC v1 (https://lkml.org/lkml/2016/2/16/710) - Lot of code cleanup as suggest by Mark Brown. - Fixed bindings as suggested by Mark Rutland. - merged all the patches to one as spliting the patches did not help review. Srinivas Kandagatla (2): ASoC: msm8916: Add codec Device Tree bindings. ASoC: msm8916: Add msm8916-wcd codec driver .../devicetree/bindings/sound/qcom,msm8916-wcd.txt | 103 ++ include/dt-bindings/sound/msm8916-wcd.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 +- sound/soc/codecs/msm8916-wcd-registers.h | 710 +++++++++ sound/soc/codecs/msm8916-wcd.c | 1575 ++++++++++++++++++++ sound/soc/codecs/msm8916-wcd.h | 308 ++++ 7 files changed, 2709 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd.txt create mode 100644 include/dt-bindings/sound/msm8916-wcd.h create mode 100644 sound/soc/codecs/msm8916-wcd-registers.h create mode 100644 sound/soc/codecs/msm8916-wcd.c create mode 100644 sound/soc/codecs/msm8916-wcd.h -- 2.8.3

3 11

Re: [alsa-devel] ASoC: rt5514: add rt5514 SPI driver
by Dan Carpenter 16 Jun '16

16 Jun '16

Hello Oder Chiou, This is a semi-automatic email about new static checker warnings. The patch 6eebf35b0e4a: "ASoC: rt5514: add rt5514 SPI driver" from Jun 6, 2016, leads to the following Smatch complaint: sound/soc/codecs/rt5514-spi.c:87 rt5514_spi_copy_work() warn: variable dereferenced before check 'rt5514_dsp->substream' (see line 83) sound/soc/codecs/rt5514-spi.c 82 container_of(work, struct rt5514_dsp, copy_work.work); 83 struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; ^^^^^^^^^^^^^^^^^^^^^^^ Dereference. 84 size_t period_bytes, truncated_bytes = 0; 85 86 mutex_lock(&rt5514_dsp->dma_lock); 87 if (!rt5514_dsp->substream) { ^^^^^^^^^^^^^^^^^^^^^ Checked too late. 88 dev_err(rt5514_dsp->dev, "No pcm substream\n"); 89 goto done; regards, dan carpenter

1 0

[alsa-devel] [CX2075] No sound after suspend/resume from front speaker
by Max Zhao 16 Jun '16

16 Jun '16

Hi, I have recently been using Archlinux with standard kernel 4.6.2 on a Toshiba Z20T-B (Intel Broadwell). After using suspend to RAM (S3) no sound is output from the laptop speakers. (systemctl suspend) The headphone jack behaves normally. Even after suspend. Restarting pulseaudio, killing it etc does not work. Pulseaudio registers no errors and tries to output sound to speakers as normal. aplay -D front:PCH also reports no errors, but fails silently. Suspend on Win10 works with sound resume, so it should not be a hardware or UEFI bug. Best regards, Max Zhao lspci -vvnn 00:1b.0 Audio device [0403]: Intel Corporation Wildcat Point-LP High Definition Audio Controller [8086:9ca0] (rev 03) Subsystem: Toshiba America Info Systems Device [1179:0001] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 49 Region 0: Memory at f1234000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:03.0 Audio device [0403]: Intel Corporation Broadwell-U Audio Controller [8086:160c] (rev 09) Subsystem: Toshiba America Info Systems Device [1179:0001] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 50 Region 0: Memory at f1230000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd_hda_intel 00:1b.0 should be the main audio card. 00:03.0 controls hdmi output. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA Intel HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: HDMI [HDA Intel HDMI], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: PCH [HDA Intel PCH], device 0: CX20753/4 Analog [CX20753/4 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 aplay -L null Discard all samples (playback) or generate zero samples (capture) pulse PulseAudio Sound Server default Default ALSA Output (currently PulseAudio Sound Server) hdmi:CARD=HDMI,DEV=0 HDA Intel HDMI, HDMI 0 HDMI Audio Output hdmi:CARD=HDMI,DEV=1 HDA Intel HDMI, HDMI 1 HDMI Audio Output hdmi:CARD=HDMI,DEV=2 HDA Intel HDMI, HDMI 2 HDMI Audio Output sysdefault:CARD=PCH HDA Intel PCH, CX20753/4 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog Front speakers surround21:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 2.1 Surround output to Front and Subwoofer speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, CX20753/4 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers amixer -c1 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 65 [88%] [-9.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] [0.00dB] Front Right: 0 [0%] [0.00dB] Simple mixer control 'Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 7 Mono: Playback 4 [57%] [-12.00dB] [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 80 Front Left: Capture 80 [100%] [6.00dB] [on] Front Right: Capture 80 [100%] [6.00dB] [on] Simple mixer control 'Auto-Mute Mode',0 Capabilities: enum Items: 'Disabled' 'Enabled' Item0: 'Disabled' Simple mixer control 'Internal Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 2 [67%] [24.00dB] Front Right: 2 [67%] [24.00dB] /proc/asound/card1/codec#0 Codec: Conexant CX20753/4 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f15111 Subsystem Id: 0x11790694 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold S3D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=3, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20753/4 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=0, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x12 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x04] Node 0x13 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Device: name="CX20753/4 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] [0x50 0x50] [0x50 0x50] [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x18 0x1a* 0x19 0x1e Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xa]: 16 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 4 0x19* 0x1a 0x15 0x1e Node 0x15 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x16 [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x04211040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x17 [Pin Complex] wcaps 0x400501: Stereo Pincap 0x00000010: OUT Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x18 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001124: IN Detect Vref caps: HIZ 80 Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x19 [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00001124: IN Detect Vref caps: HIZ 80 Pin Default 0x04a11030: [Jack] Mic at Ext Right Conn = 1/8, Color = Black DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1a [Pin Complex] wcaps 0x40048b: Stereo Amp-In Control: name="Internal Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x02 0x02] Pincap 0x00000020: IN Pin Default 0x95a60120: [Fixed] Mic at Int Top Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x1e [Pin Complex] wcaps 0x40048b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000024: IN Detect Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0

1 0

Re: [alsa-devel] [PATCH v5] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 16 Jun '16

16 Jun '16

// + Adam, On 2016年06月16日 03:43, Steve Barber wrote: > On Mon, Jun 13, 2016 at 1:42 AM, Xing Zheng<zhengxing(a)rock-chips.com> wrote: >> + >> +static int rockchip_sound_da7219_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 *cpu_dai = rtd->cpu_dai; >> + struct snd_soc_dai *codec_dai = rtd->codec_dai; >> + struct snd_soc_dapm_context *dapm =&rtd->card->dapm; >> + int mclk, ret; >> + >> + snd_soc_dapm_disable_pin(dapm, "Speakers"); >> + snd_soc_dapm_disable_pin(dapm, "Int Mic"); >> + snd_soc_dapm_enable_pin(dapm, "Headphones"); >> + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); >> + snd_soc_dapm_sync(dapm); >> + >> + /* in bypass mode, the mclk has to be one of the frequencies below */ >> + switch (params_rate(params)) { >> + case 8000: >> + case 16000: >> + case 24000: >> + case 32000: >> + case 48000: >> + case 64000: >> + case 96000: >> + mclk = 12288000; >> + break; >> + case 11025: >> + case 22050: >> + case 44100: >> + case 88200: >> + mclk = 11289600; >> + break; >> + default: >> + return -EINVAL; >> + } >> + >> + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, >> + SND_SOC_CLOCK_OUT); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); >> + return ret; >> + } >> + >> + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, >> + SND_SOC_CLOCK_IN); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); >> + return ret; >> + } >> + >> + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); >> + if (ret< 0) { >> + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); >> + return ret; >> + } >> + >> + return 0; >> +} >> + >> +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) >> +{ >> + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; >> + int ret; >> + >> + /* Enable Headset and 4 Buttons Jack detection */ >> + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", >> + SND_JACK_HEADSET | > Should this also include SND_JACK_LINEOUT? da7219 differentiates > between the two so not including lineout means, for example, we can > miss jack insertion/removal events. > Hi Adam, Could you please confirm Steve says that about SND_JACK_LINEOUT? Thanks. -- - Xing Zheng

2 1

[alsa-devel] [PATCH] ALSA: hda: fix some klockwork scan warnings
by libin.yang＠linux.intel.com 16 Jun '16

16 Jun '16

From: Libin Yang <libin.yang(a)linux.intel.com> This patch fixes some warnings from klockwork. These warnings are not the real issues. The patch adds the sanity check. Signed-off-by: Libin Yang <libin.yang(a)linux.intel.com> --- sound/hda/hdmi_chmap.c | 28 +++++++++++++++++++++++++--- sound/pci/hda/hda_codec.c | 8 +++++++- sound/pci/hda/patch_hdmi.c | 5 +++++ 3 files changed, 37 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdmi_chmap.c b/sound/hda/hdmi_chmap.c index c6c75e7..81acc20 100644 --- a/sound/hda/hdmi_chmap.c +++ b/sound/hda/hdmi_chmap.c @@ -353,7 +353,8 @@ static void hdmi_std_setup_channel_mapping(struct hdac_chmap *chmap, int hdmi_slot = 0; /* fill actual channel mappings in ALSA channel (i) order */ for (i = 0; i < ch_alloc->channels; i++) { - while (!ch_alloc->speakers[7 - hdmi_slot] && !WARN_ON(hdmi_slot >= 8)) + while (!WARN_ON(hdmi_slot >= 8) && + !ch_alloc->speakers[7 - hdmi_slot]) hdmi_slot++; /* skip zero slots */ hdmi_channel_mapping[ca][i] = (i << 4) | hdmi_slot++; @@ -430,6 +431,12 @@ static int to_cea_slot(int ordered_ca, unsigned char pos) int mask = snd_hdac_chmap_to_spk_mask(pos); int i; + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (ordered_ca >= ARRAY_SIZE(channel_allocations)) + return -1; + if (mask) { for (i = 0; i < 8; i++) { if (channel_allocations[ordered_ca].speakers[7 - i] == mask) @@ -456,7 +463,15 @@ EXPORT_SYMBOL_GPL(snd_hdac_spk_to_chmap); /* from CEA slot to ALSA API channel position */ static int from_cea_slot(int ordered_ca, unsigned char slot) { - int mask = channel_allocations[ordered_ca].speakers[7 - slot]; + int mask; + + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (slot >= 8) + return 0; + + mask = channel_allocations[ordered_ca].speakers[7 - slot]; return snd_hdac_spk_to_chmap(mask); } @@ -523,7 +538,8 @@ static void hdmi_setup_fake_chmap(unsigned char *map, int ca) int ordered_ca = get_channel_allocation_order(ca); for (i = 0; i < 8; i++) { - if (i < channel_allocations[ordered_ca].channels) + if (ordered_ca < ARRAY_SIZE(channel_allocations) && + i < channel_allocations[ordered_ca].channels) map[i] = from_cea_slot(ordered_ca, hdmi_channel_mapping[ca][i] & 0x0f); else map[i] = 0; @@ -551,6 +567,12 @@ int snd_hdac_get_active_channels(int ca) { int ordered_ca = get_channel_allocation_order(ca); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (ordered_ca >= ARRAY_SIZE(channel_allocations)) + ordered_ca = 0; + return channel_allocations[ordered_ca].channels; } EXPORT_SYMBOL_GPL(snd_hdac_get_active_channels); diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8374188..9913be8 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3584,6 +3584,12 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, bool reset; spdif = snd_hda_spdif_out_of_nid(codec, nid); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (WARN_ON(spdif == NULL)) + return; + curr_fmt = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_STREAM_FORMAT, 0); reset = codec->spdif_status_reset && @@ -3768,7 +3774,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, spdif = snd_hda_spdif_out_of_nid(codec, mout->dig_out_nid); if (mout->dig_out_nid && mout->share_spdif && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { - if (chs == 2 && + if (chs == 2 && spdif != NULL && snd_hda_is_supported_format(codec, mout->dig_out_nid, format) && !(spdif->status & IEC958_AES0_NONAUDIO)) { diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index d0d5ad8..56e5204 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1680,6 +1680,11 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid) mutex_lock(&codec->spdif_mutex); spdif = snd_hda_spdif_out_of_nid(codec, cvt_nid); + /* Add sanity check to pass klockwork check. + * This should never happen. + */ + if (WARN_ON(spdif == NULL)) + return true; non_pcm = !!(spdif->status & IEC958_AES0_NONAUDIO); mutex_unlock(&codec->spdif_mutex); return non_pcm; -- 1.9.1

2 1

[alsa-devel] Wolfson 5102 on Beaglebone Black using ALSA
by Clark Dunson 15 Jun '16

15 Jun '16

Hey everybody! We have been working on integrating the Wolfson 5102 codec with the Beaglebone Black. We’ve made pretty decent progress, got the arizona code in kernel 4.4 registering the wm5102 as ALSA sound card (both player and recorder), have McAsp0 as 24.576 MHz clock source, and have 12 IRQs showing up in /proc/interrupts. We have been working with sound{compatible = "simple-audio-card”}, but now have hit our first real question. sysclk is hard coded to ‘0’ under simple-audio-card, and we had to hack the kernel and change that to ‘1’ to achieve ALSA sound card registration. We’d all like for this work to proceed without having to hack the kernel, and wondered if one of you might recommend another approach or selection for device tree sound{compatible} that might work out better? Thanks! Clark

3 10

[alsa-devel] [PATCH] ASoC: dapm: Fix to return correct path list in is_connected_ep.
by jeeja.kp＠intel.com 15 Jun '16

15 Jun '16

From: Jeeja KP <jeeja.kp(a)intel.com> In is_connected_ep, when custom_stop_condition is true, need to return the correct paths instead of con which is 0. Fixes: 6742064aef7f('ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets') Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/soc-dapm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index db781f6..3c3f027 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1092,8 +1092,10 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (list) list_add_tail(&widget->work_list, list); - if (custom_stop_condition && custom_stop_condition(widget, dir)) - return con; + if (custom_stop_condition && custom_stop_condition(widget, dir)) { + widget->endpoints[dir] = 1; + return widget->endpoints[dir]; + } if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); -- 2.5.0

2 1

[alsa-devel] [PATCH 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Clemens Gruber 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. [I rebased Eric's patches from Feb 2015, fixed a few warnings, URLs, etc. and squashed two patches into one. - Clemens] Clemens Gruber (1): ASoC: sgtl5000: Remove misleading comment Eric Nelson (5): ASoC: sgtl5000: Fix regulator support ASoC: sgtl5000: Write all default registers ASoC: sgtl5000: Initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: Disable internal PLL early ASoC: sgtl5000: Do not disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 421 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 321 deletions(-) -- 2.8.3

4 17

[alsa-devel] [PATCH RFC 0/6] ASoC: sgtl5000: fix use of regulators and internal LDO
by Eric Nelson 15 Jun '16

15 Jun '16

This patch set addresses a structural problem in the handling of regulators for VDDIO, VDDA, and VDDD in the SGTL5000 driver. The first two of these power rails must be powered on prior to any I2C communication, and yet the regulators were tied to the codec, which is instantiated only after a fair amount of I2C communication takes place. In other words, these regulators could never have function, and we can surmise that no user of this driver has switched power supply rails connected to them. The third power rail (VDDD) can be derived internally (by using I2C registers) though the data sheet says that if an external VDDD is used, it should be enabled before MCLK is started and I2C activity begins. Patch 1 moves the regulators to the I2C device and initializes them before I2C activity. Patch 2 addresses an issue found during development of the patch and is somewhat unrelated to regulators. It is included for discussion Patch 3 removes the root cause of patch 2 Patch 4 adds return value checks when setting the LINREG and ANA_POWER registers Patch 5 addresses the cause of follow-on failures and the source of the error in patch 3. Patch 6 removes the naive use of the supplies to lower power consumption and replaces it with something more modest. The patch set is being sent as an RFC for discussion, and should probably be squashed into two or three patches covering the regulator updates (patches 1, 3-5 and perhaps 6), initialization error handling (patch 2). Eric Nelson (6): ASoC: sgtl5000: fix regulator support ASoC: sgtl5000: write all default registers ASoC: sgtl5000: initialize CHIP_ANA_POWER to power-on defaults ASoC: sgtl5000: check return values ASoC: sgtl5000: disable internal PLL early ASoC: sgtl5000: Don't disable regulators in SND_SOC_BIAS_OFF sound/soc/codecs/sgtl5000.c | 417 +++++++++++--------------------------------- sound/soc/codecs/sgtl5000.h | 2 + 2 files changed, 102 insertions(+), 317 deletions(-) -- 1.9.1

3 28

[alsa-devel] [PATCH v2] ASoC: adau17x1: Add support for specifying the MCLK using the CCF
by Lars-Peter Clausen 15 Jun '16

15 Jun '16

The devices from the ADAU17X1 family all have a MCLK clock input which supplies the master clock for the device. The master clock is used as the input clock for the PLL. Currently the MCLK rate as well as the desired PLL output frequency need to be supplied by calling snd_soc_dai_set_pll() form a machine driver. Add support for specifying the MCLK using the common clock framework. In addition to that also automatically configure the PLL to a suitable rate if the master clock was provided using the CCW. This allows to use the CODEC driver without any special configuration requirements from the machine driver. While the PLL output frequency can be configured over a (more or less) continuous range the narrowness of the range and the other constraints of the clocking tree usually only result in two output frequencies that will actually be chosen. One for 44.1kHz based rates and one for 48kHz based rates, these are the rates that the automatic PLL configuration will use. For the rare case where a non-standard setup is required a machine driver can disable the auto-configuration and configure a custom frequency using the existing mechanisms. If the common clock framework is not enabled clk_get() will return NULL and the driver will function as before and the clock rate needs to be configured manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Change numerical value of ADAU17X1_CLK_SRC_PLL_AUTO to 0, since that is what the simple-card driver will call snd_soc_dai_set_sysclk() with when the CODEC node has a clock property. So this is really the only way to make this work with the simple-card driver. --- .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 219 +++++++++++++++------ sound/soc/codecs/adau17x1.h | 6 + 7 files changed, 179 insertions(+), 62 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt index 8dbce0e..1447dec 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -13,6 +13,11 @@ Required properties: - reg: The i2c address. Value depends on the state of ADDR0 and ADDR1, as wired in hardware. +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + Examples: #include <dt-bindings/sound/adau17x1.h> @@ -20,5 +25,8 @@ Examples: adau1361@38 { compatible = "adi,adau1761"; reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; }; }; diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 8de010f..9e7f257 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client, static int adau1761_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index d917124..a0b214b 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi) static int adau1761_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 06cbca8..7b9d180 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client, static int adau1781_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c index 3d965a0..9b23354 100644 --- a/sound/soc/codecs/adau1781-spi.c +++ b/sound/soc/codecs/adau1781-spi.c @@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi) static int adau1781_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 66a6e06..439aa3f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> @@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + bool is_pll; + bool was_pll; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + is_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL_AUTO: + if (!adau->mclk) + return -EINVAL; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: + is_pll = true; + break; + default: + return -EINVAL; + } + + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_MCLK: + was_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL: + case ADAU17X1_CLK_SRC_PLL_AUTO: + was_pll = true; + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (is_pll != was_pll) { + if (is_pll) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_auto_pll(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct adau *adau = snd_soc_dai_get_drvdata(dai); + unsigned int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 96000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + case 88200: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK, + clk_get_rate(adau->mclk), pll_rate); +} + static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret; - if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_PLL_AUTO: + ret = adau17x1_auto_pll(dai, params); + if (ret) + return ret; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; - else + break; + default: freq = adau->sysclk; + break; + } if (freq % params_rate(params) != 0) return -EINVAL; @@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); } -static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct snd_soc_codec *codec = dai->codec; - struct adau *adau = snd_soc_codec_get_drvdata(codec); - int ret; - - if (freq_in < 8000000 || freq_in > 27000000) - return -EINVAL; - - ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); - if (ret < 0) - return ret; - - /* The PLL register is 6 bytes long and can only be written at once. */ - ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, - adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); - if (ret) - return ret; - - adau->pll_freq = freq_out; - - return 0; -} - -static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); - struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - - switch (clk_id) { - case ADAU17X1_CLK_SRC_MCLK: - case ADAU17X1_CLK_SRC_PLL: - break; - default: - return -EINVAL; - } - - adau->sysclk = freq; - - if (adau->clk_src != clk_id) { - if (clk_id == ADAU17X1_CLK_SRC_PLL) { - snd_soc_dapm_add_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } else { - snd_soc_dapm_del_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } - } - - adau->clk_src = clk_id; - - return 0; -} - static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } + + if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK) + snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1); + return ret; } EXPORT_SYMBOL_GPL(adau17x1_add_routes); @@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, const char *firmware_name) { struct adau *adau; + int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; + adau->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(adau->mclk)) { + if (PTR_ERR(adau->mclk) != -ENOENT) + return PTR_ERR(adau->mclk); + /* Clock is optional (for the driver) */ + adau->mclk = NULL; + } else if (adau->mclk) { + adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; + + /* + * Any valid PLL output rate will work at this point, use one + * that is likely to be chosen later as well. The register will + * be written when the PLL is powered up for the first time. + */ + ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024, + adau->pll_regs); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(adau->mclk); + if (ret) + return ret; + } + adau->regmap = regmap; adau->switch_mode = switch_mode; adau->type = type; @@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau17x1_probe); +void adau17x1_remove(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + if (adau->mclk) + clk_disable_unprepare(adau->mclk); +} +EXPORT_SYMBOL_GPL(adau17x1_remove); + MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 5ae87a0..bf04b7e 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -22,13 +22,18 @@ enum adau17x1_pll_src { }; enum adau17x1_clk_src { + /* Automatically configure PLL based on the sample rate */ + ADAU17X1_CLK_SRC_PLL_AUTO, ADAU17X1_CLK_SRC_MCLK, ADAU17X1_CLK_SRC_PLL, }; +struct clk; + struct adau { unsigned int sysclk; unsigned int pll_freq; + struct clk *mclk; enum adau17x1_clk_src clk_src; enum adau17x1_type type; @@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name); +void adau17x1_remove(struct device *dev); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/usb/mixer_maps.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 1f8fb0d9..9038b2e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -107,8 +107,10 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov <bin(a)bash.info> */ -static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ -static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_1 = {.min = -4781, .max = 0}; + /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {.min = -1781, .max = 618}; + /* just guess */ static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: adau17x1: Add support for specifying the MCLK using the CCF
by Lars-Peter Clausen 15 Jun '16

15 Jun '16

The devices from the ADAU17X1 family all have a MCLK clock input which supplies the master clock for the device. The master clock is used as the input clock for the PLL. Currently the MCLK rate as well as the desired PLL output frequency need to be supplied by calling snd_soc_dai_set_pll() form a machine driver. Add support for specifying the MCLK using the common clock framework. In addition to that also automatically configure the PLL to a suitable rate if the master clock was provided using the CCW. This allows to use the CODEC driver without any special configuration requirements from the machine driver. While the PLL output frequency can be configured over a (more or less) continuous range the narrowness of the range and the other constraints of the clocking tree usually only result in two output frequencies that will actually be chosen. One for 44.1kHz based rates and one for 48kHz based rates, these are the rates that the automatic PLL configuration will use. For the rare case where a non-standard setup is required a machine driver can disable the auto-configuration and configure a custom frequency using the existing mechanisms. If the common clock framework is not enabled clk_get() will return NULL and the driver will function as before and the clock rate needs to be configured manually. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/adi,adau17x1.txt | 8 + sound/soc/codecs/adau1761-i2c.c | 2 +- sound/soc/codecs/adau1761-spi.c | 2 +- sound/soc/codecs/adau1781-i2c.c | 2 +- sound/soc/codecs/adau1781-spi.c | 2 +- sound/soc/codecs/adau17x1.c | 219 +++++++++++++++------ sound/soc/codecs/adau17x1.h | 6 + 7 files changed, 179 insertions(+), 62 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt index 8dbce0e..1447dec 100644 --- a/Documentation/devicetree/bindings/sound/adi,adau17x1.txt +++ b/Documentation/devicetree/bindings/sound/adi,adau17x1.txt @@ -13,6 +13,11 @@ Required properties: - reg: The i2c address. Value depends on the state of ADDR0 and ADDR1, as wired in hardware. +Optional properties: + - clock-names: If provided must be "mclk". + - clocks: phandle + clock-specifiers for the clock that provides + the audio master clock for the device. + Examples: #include <dt-bindings/sound/adau17x1.h> @@ -20,5 +25,8 @@ Examples: adau1361@38 { compatible = "adi,adau1761"; reg = <0x38>; + + clock-names = "mclk"; + clocks = <&audio_clock>; }; }; diff --git a/sound/soc/codecs/adau1761-i2c.c b/sound/soc/codecs/adau1761-i2c.c index 8de010f..9e7f257 100644 --- a/sound/soc/codecs/adau1761-i2c.c +++ b/sound/soc/codecs/adau1761-i2c.c @@ -31,7 +31,7 @@ static int adau1761_i2c_probe(struct i2c_client *client, static int adau1761_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1761-spi.c b/sound/soc/codecs/adau1761-spi.c index d917124..a0b214b 100644 --- a/sound/soc/codecs/adau1761-spi.c +++ b/sound/soc/codecs/adau1761-spi.c @@ -48,7 +48,7 @@ static int adau1761_spi_probe(struct spi_device *spi) static int adau1761_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-i2c.c b/sound/soc/codecs/adau1781-i2c.c index 06cbca8..7b9d180 100644 --- a/sound/soc/codecs/adau1781-i2c.c +++ b/sound/soc/codecs/adau1781-i2c.c @@ -31,7 +31,7 @@ static int adau1781_i2c_probe(struct i2c_client *client, static int adau1781_i2c_remove(struct i2c_client *client) { - snd_soc_unregister_codec(&client->dev); + adau17x1_remove(&client->dev); return 0; } diff --git a/sound/soc/codecs/adau1781-spi.c b/sound/soc/codecs/adau1781-spi.c index 3d965a0..9b23354 100644 --- a/sound/soc/codecs/adau1781-spi.c +++ b/sound/soc/codecs/adau1781-spi.c @@ -48,7 +48,7 @@ static int adau1781_spi_probe(struct spi_device *spi) static int adau1781_spi_remove(struct spi_device *spi) { - snd_soc_unregister_codec(&spi->dev); + adau17x1_remove(&spi->dev); return 0; } diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index 66a6e06..439aa3f 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -9,6 +9,7 @@ #include <linux/module.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> @@ -303,6 +304,116 @@ bool adau17x1_has_dsp(struct adau *adau) } EXPORT_SYMBOL_GPL(adau17x1_has_dsp); +static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau *adau = snd_soc_codec_get_drvdata(codec); + int ret; + + if (freq_in < 8000000 || freq_in > 27000000) + return -EINVAL; + + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; + + /* The PLL register is 6 bytes long and can only be written at once. */ + ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, + adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); + if (ret) + return ret; + + adau->pll_freq = freq_out; + + return 0; +} + +static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); + struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); + bool is_pll; + bool was_pll; + + switch (clk_id) { + case ADAU17X1_CLK_SRC_MCLK: + is_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL_AUTO: + if (!adau->mclk) + return -EINVAL; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: + is_pll = true; + break; + default: + return -EINVAL; + } + + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_MCLK: + was_pll = false; + break; + case ADAU17X1_CLK_SRC_PLL: + case ADAU17X1_CLK_SRC_PLL_AUTO: + was_pll = true; + break; + default: + return -EINVAL; + } + + adau->sysclk = freq; + + if (is_pll != was_pll) { + if (is_pll) { + snd_soc_dapm_add_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } else { + snd_soc_dapm_del_routes(dapm, + &adau17x1_dapm_pll_route, 1); + } + } + + adau->clk_src = clk_id; + + return 0; +} + +static int adau17x1_auto_pll(struct snd_soc_dai *dai, + struct snd_pcm_hw_params *params) +{ + struct adau *adau = snd_soc_dai_get_drvdata(dai); + unsigned int pll_rate; + + switch (params_rate(params)) { + case 48000: + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 96000: + pll_rate = 48000 * 1024; + break; + case 44100: + case 7350: + case 11025: + case 14700: + case 22050: + case 29400: + case 88200: + pll_rate = 44100 * 1024; + break; + default: + return -EINVAL; + } + + return adau17x1_set_dai_pll(dai, ADAU17X1_PLL, ADAU17X1_PLL_SRC_MCLK, + clk_get_rate(adau->mclk), pll_rate); +} + static int adau17x1_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -312,10 +423,19 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, unsigned int freq; int ret; - if (adau->clk_src == ADAU17X1_CLK_SRC_PLL) + switch (adau->clk_src) { + case ADAU17X1_CLK_SRC_PLL_AUTO: + ret = adau17x1_auto_pll(dai, params); + if (ret) + return ret; + /* Fall-through */ + case ADAU17X1_CLK_SRC_PLL: freq = adau->pll_freq; - else + break; + default: freq = adau->sysclk; + break; + } if (freq % params_rate(params) != 0) return -EINVAL; @@ -387,62 +507,6 @@ static int adau17x1_hw_params(struct snd_pcm_substream *substream, ADAU17X1_SERIAL_PORT1_DELAY_MASK, val); } -static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, - int source, unsigned int freq_in, unsigned int freq_out) -{ - struct snd_soc_codec *codec = dai->codec; - struct adau *adau = snd_soc_codec_get_drvdata(codec); - int ret; - - if (freq_in < 8000000 || freq_in > 27000000) - return -EINVAL; - - ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); - if (ret < 0) - return ret; - - /* The PLL register is 6 bytes long and can only be written at once. */ - ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, - adau->pll_regs, ARRAY_SIZE(adau->pll_regs)); - if (ret) - return ret; - - adau->pll_freq = freq_out; - - return 0; -} - -static int adau17x1_set_dai_sysclk(struct snd_soc_dai *dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(dai->codec); - struct adau *adau = snd_soc_codec_get_drvdata(dai->codec); - - switch (clk_id) { - case ADAU17X1_CLK_SRC_MCLK: - case ADAU17X1_CLK_SRC_PLL: - break; - default: - return -EINVAL; - } - - adau->sysclk = freq; - - if (adau->clk_src != clk_id) { - if (clk_id == ADAU17X1_CLK_SRC_PLL) { - snd_soc_dapm_add_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } else { - snd_soc_dapm_del_routes(dapm, - &adau17x1_dapm_pll_route, 1); - } - } - - adau->clk_src = clk_id; - - return 0; -} - static int adau17x1_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) { @@ -827,6 +891,10 @@ int adau17x1_add_routes(struct snd_soc_codec *codec) ret = snd_soc_dapm_add_routes(dapm, adau17x1_no_dsp_dapm_routes, ARRAY_SIZE(adau17x1_no_dsp_dapm_routes)); } + + if (adau->clk_src != ADAU17X1_CLK_SRC_MCLK) + snd_soc_dapm_add_routes(dapm, &adau17x1_dapm_pll_route, 1); + return ret; } EXPORT_SYMBOL_GPL(adau17x1_add_routes); @@ -849,6 +917,7 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, const char *firmware_name) { struct adau *adau; + int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -857,6 +926,30 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, if (!adau) return -ENOMEM; + adau->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(adau->mclk)) { + if (PTR_ERR(adau->mclk) != -ENOENT) + return PTR_ERR(adau->mclk); + /* Clock is optional (for the driver) */ + adau->mclk = NULL; + } else if (adau->mclk) { + adau->clk_src = ADAU17X1_CLK_SRC_PLL_AUTO; + + /* + * Any valid PLL output rate will work at this point, use one + * that is likely to be chosen later as well. The register will + * be written when the PLL is powered up for the first time. + */ + ret = adau_calc_pll_cfg(clk_get_rate(adau->mclk), 48000 * 1024, + adau->pll_regs); + if (ret < 0) + return ret; + + ret = clk_prepare_enable(adau->mclk); + if (ret) + return ret; + } + adau->regmap = regmap; adau->switch_mode = switch_mode; adau->type = type; @@ -880,6 +973,16 @@ int adau17x1_probe(struct device *dev, struct regmap *regmap, } EXPORT_SYMBOL_GPL(adau17x1_probe); +void adau17x1_remove(struct device *dev) +{ + struct adau *adau = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + if (adau->mclk) + clk_disable_unprepare(adau->mclk); +} +EXPORT_SYMBOL_GPL(adau17x1_remove); + MODULE_DESCRIPTION("ASoC ADAU1X61/ADAU1X81 common code"); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau17x1.h b/sound/soc/codecs/adau17x1.h index 5ae87a0..e9fbc3b 100644 --- a/sound/soc/codecs/adau17x1.h +++ b/sound/soc/codecs/adau17x1.h @@ -24,11 +24,16 @@ enum adau17x1_pll_src { enum adau17x1_clk_src { ADAU17X1_CLK_SRC_MCLK, ADAU17X1_CLK_SRC_PLL, + /* Automatically configure PLL based on the sample rate */ + ADAU17X1_CLK_SRC_PLL_AUTO, }; +struct clk; + struct adau { unsigned int sysclk; unsigned int pll_freq; + struct clk *mclk; enum adau17x1_clk_src clk_src; enum adau17x1_type type; @@ -52,6 +57,7 @@ int adau17x1_add_routes(struct snd_soc_codec *codec); int adau17x1_probe(struct device *dev, struct regmap *regmap, enum adau17x1_type type, void (*switch_mode)(struct device *dev), const char *firmware_name); +void adau17x1_remove(struct device *dev); int adau17x1_set_micbias_voltage(struct snd_soc_codec *codec, enum adau17x1_micbias_voltage micbias); bool adau17x1_readable_register(struct device *dev, unsigned int reg); -- 2.1.4

1 0

[alsa-devel] [PATCH] ALSA: seq_oss: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Also, removed some unnecessary comments. Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/core/seq/oss/seq_oss_synth.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index b16dbef..cd0e0eb 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -70,11 +70,11 @@ struct seq_oss_synth { static int max_synth_devs; static struct seq_oss_synth *synth_devs[SNDRV_SEQ_OSS_MAX_SYNTH_DEVS]; static struct seq_oss_synth midi_synth_dev = { - -1, /* seq_device */ - SYNTH_TYPE_MIDI, /* synth_type */ - 0, /* synth_subtype */ - 16, /* nr_voices */ - "MIDI", /* name */ + .seq_device = -1, + .synth_type = SYNTH_TYPE_MIDI, + .synth_subtype = 0, + .nr_voices = 16, + .name = "MIDI", }; static DEFINE_SPINLOCK(register_lock); -- 1.9.1

2 1

Re: [alsa-devel] Support Dell headset mode for ALC891
by Takashi Iwai 15 Jun '16

15 Jun '16

On Wed, 08 Jun 2016 09:17:12 +0200, Kailang wrote: > > Hi Takashi, > > This patch is supported Dell headset mode for ALC891. > It is only support I-phone type headset. > I think this function is only support for DELL. > So, headset mode and platform support is merged. > > This patch is test passed by Ubuntu team. > > Many Thanks. (snip) > Date: Tue, 7 Jun 2016 11:31:34 +0800 > Subject: [PATCH] ALSA: hda/realtek - ALC891 headset mode for Dell > > New headset mode of ALC891 for Dell. > > Signed-off-by: Kailang Yang <kailang(a)realtek.com> Could you put more comments in the changelog, as what you gave in the mail? > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 7960316..9dc641e 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -3718,6 +3718,9 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec) > case 0x10ec0295: > alc_process_coef_fw(codec, coef0225); > break; > + case 0x10ec0867: > + alc_update_coefex_idx(codec, 0x57, 0x5, 1<<14, 0); > + break; > } > codec_dbg(codec, "Headset jack set to unplugged mode.\n"); > } > @@ -3805,6 +3808,8 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin, > alc_process_coef_fw(codec, coef0293); > snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50); > break; > + case 0x10ec0867: > + alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14); > case 0x10ec0662: No break between 0867 and 0662? thanks, Takashi

2 5

[alsa-devel] [alsa-lib][PATCH 00/10 v2] ctl: add APIs for control element set
by Takashi Sakamoto 15 Jun '16

15 Jun '16

Hi, This patchset is revised version of below one: [alsa-devel] [alsa-lib][PATCH 0/6] control: add APIs for control element set http://mailman.alsa-project.org/pipermail/alsa-devel/2016-February/104795.h… In this time, I add a program to test the feature. Changes: - Use a model which consists of element set/element/member/value. - Add an description about the design of ALSA ctl feature. - Obsolete old APIs due to the lack of const semantics. - Add an description about threshold level feature and topology framework. - Improve comments to threshold level APIs. - Add a test program. Before starting any discussions, please read description of the model, to stand on the same basis of understanding. I note that execution of the test program causes an abort to PulseAudio. I can see this line below. pulseaudio: hcontrol.c:745: snd_hctl_handle_event: Assertion `res >= 0 && dir == 0' failed. I guess that quick addition/removal of control element set causes this assertion, but I have little time to investigate the cause. I wish to get helps of ALSA developers in PulseAudio side. If you need an practical example of the feature, please test one of Python 3 samples in my 'alsa-gi'. https://github.com/takaswie/alsa-gi/ This works without alsa-lib, by using kernel/userspace interface directly. Therefore, this patchet doesn't affect the implementation. Takashi Sakamoto (10): ctl: add an overview for design of ALSA control interface ctl: improve comments for handling element data ctl: add functions to add an element set ctl: improve comments for API to add an element of IEC958 type ctl: change former APIs as wrapper functions of element set APIs ctl: deprecate APIs to add an element set with a single element pcm: use new APIs to add a control element set for softvol plugin ctl: add explaination about threshold level feature ctl: improve API documentation for threshold level operations ctl: add test program for control element set include/control.h | 36 +- src/control/control.c | 829 ++++++++++++++++++++++++++++++++------------ src/pcm/pcm_softvol.c | 7 +- test/Makefile.am | 5 +- test/user-ctl-element-set.c | 561 ++++++++++++++++++++++++++++++ 5 files changed, 1216 insertions(+), 222 deletions(-) create mode 100644 test/user-ctl-element-set.c -- 2.7.4

2 16

[alsa-devel] [PATCH] sound: aedsp16: Change structure initialisation to C99 style
by Amitoj Kaur Chawla 15 Jun '16

15 Jun '16

Replace the in order struct initialisation style with explicit field style. The Coccinelle semantic patch used to make this change is as follows: @decl@ identifier i1,fld; type T; field list[n] fs; @@ struct i1 { fs T fld; ...}; @@ identifier decl.i1,i2,decl.fld; expression e; position bad.p, bad.fix; @@ struct i1 i2@p = { ..., + .fld = e - e@fix ,...}; Signed-off-by: Amitoj Kaur Chawla <amitoj1606(a)gmail.com> --- sound/oss/aedsp16.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/oss/aedsp16.c b/sound/oss/aedsp16.c index 35b5912..bb477d5 100644 --- a/sound/oss/aedsp16.c +++ b/sound/oss/aedsp16.c @@ -482,13 +482,13 @@ static struct orVals orDMA[] __initdata = { }; static struct aedsp16_info ae_config = { - DEF_AEDSP16_IOB, - DEF_AEDSP16_IRQ, - DEF_AEDSP16_MRQ, - DEF_AEDSP16_DMA, - -1, - -1, - INIT_NONE + .base_io = DEF_AEDSP16_IOB, + .irq = DEF_AEDSP16_IRQ, + .mpu_irq = DEF_AEDSP16_MRQ, + .dma = DEF_AEDSP16_DMA, + .mss_base = -1, + .mpu_base = -1, + .init = INIT_NONE }; /* -- 1.9.1

2 1

[alsa-devel] [PATCH v2 1/2] sound: lpass-cpu: add module licence and description
by Srinivas Kandagatla 14 Jun '16

14 Jun '16

This patch adds module licence to lpass-cpu driver, without this patch lpass-cpu module would taint with below error: snd_soc_lpass_cpu: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint snd_soc_lpass_cpu: Unknown symbol regmap_write (err 0) snd_soc_lpass_cpu: Unknown symbol devm_kmalloc (err 0) ... Acked-by: Kenneth Westfield <kwestfie(a)codeaurora.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 3cde9fb..eff3f9a 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -586,3 +586,6 @@ int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev) return 0; } EXPORT_SYMBOL_GPL(asoc_qcom_lpass_cpu_platform_remove); + +MODULE_DESCRIPTION("QTi LPASS CPU Driver"); +MODULE_LICENSE("GPL v2"); -- 2.8.3

2 4

[alsa-devel] [PATCH v3 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 14 Jun '16

14 Jun '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.7-rc3 kernel. Changes in v3: - Use of_node_cmp() in device_get_named_child_node() to match DT node. Changes in v2: - Rebase to v4.7-rc1 - Small updates to codec patch based on previous reviewer comments Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/of.h | 14 +++--- include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 7 files changed, 119 insertions(+), 76 deletions(-) -- 1.9.3

3 4

[alsa-devel] [PATCH 1/2] ASoC: remove one extraneous 'const'
by Arnd Bergmann 14 Jun '16

14 Jun '16

A recent commit made a few arrays 'const', but also added the same attribute to a function return type, where it makes no sense, and we get a warning when building with W=1: sound/soc/codecs/arizona.c:1725:27: error: type qualifiers ignored on function return type [-Werror=ignored-qualifiers] static const char * const arizona_dai_clk_str(int clk_id) This removes it again. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: a3178a3ed798 ("ASoC: arizona: Add a couple of missing consts") --- This is one of two patches for warnings that were introduced first in today's linux-next. diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ed0425763755..ecfdbfcae366 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1722,7 +1722,7 @@ restore_aif: return ret; } -static const char * const arizona_dai_clk_str(int clk_id) +static const char *arizona_dai_clk_str(int clk_id) { switch (clk_id) { case ARIZONA_CLK_SYSCLK:

3 4

[alsa-devel] [PATCH 0/7] ASoC: Intel: Update Bxt machine and pm support
by Vinod Koul 14 Jun '16

14 Jun '16

This series first adds constraints and other updates on Bxt-rt298 machine for various scenarios. Then it updates platform driver for PM support to enable firmware reload on suspend and fixes dma id for purge request, updates stall bits on stall/unstall commands and finally adds PM support in machine. Jayachandran B (2): ASoC: Intel: Skylake: Enable firmware reload in suspend ASoC: Intel: Skylake: Update DSP stall bits Jeeja KP (1): ASoC: Intel: Add support for PM ops in bxt-rt298 Senthilnathan Veppur (4): ASoC: Intel: Add DMIC 4 channel support for bxt machine ASoC: Intel: Add FE rate & channel constraints for bxt-rt298 ASoC: Intel: Update ignore suspend for bxt-rt298 ASoC: Intel: Skylake: Update FW purge for Broxton sound/soc/intel/boards/bxt_rt298.c | 140 ++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/bxt-sst.c | 25 +++++- sound/soc/intel/skylake/skl-sst-dsp.c | 15 ++-- sound/soc/intel/skylake/skl-sst-dsp.h | 2 + sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-sst.c | 1 + sound/soc/intel/skylake/skl.c | 1 + 7 files changed, 176 insertions(+), 11 deletions(-) -- 1.9.1

2 14

[alsa-devel] [PATCH v2 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 14 Jun '16

14 Jun '16

This patch set converts the da7219 codec driver to use device/fwnode functions to access properties, instead of the DT only of_* functions, allowing ACPI to be used as well. The DT bindings for da7219 have a device node for the main codec properties, and a named child node (da7219_aad), which contains all of the accessory detection related properties for the device. ACPI also supports this kind of FW hierarchy (data only sub-nodes), but some support in the kernel needs to be added to take make use of this in driver code. The first patch adds functions to allow searching for a named child node of a device, for both DT and ACPI, and the second patch updates the codec driver to use the standard device/fwnode calls, including this new function. These changes are based on the v4.7-rc1 kernel. Changes in v2: - Rebase to v4.7-rc1 - Small updates to codec patch based on previous reviewer comments Adam Thomson (2): device property: Add function to search for named child of device ASoC: da7219: Convert driver to use generic device/fwnode functions drivers/base/property.c | 28 ++++++++++++ include/acpi/acpi_bus.h | 7 +++ include/linux/acpi.h | 6 +++ include/linux/property.h | 3 ++ sound/soc/codecs/da7219-aad.c | 103 +++++++++++++++++++++--------------------- sound/soc/codecs/da7219.c | 34 +++++++------- 6 files changed, 112 insertions(+), 69 deletions(-) -- 1.9.3

5 8

[alsa-devel] ASoC: Machine Driver Interaction with Codec Drivers
by Jeremy McDermond 14 Jun '16

14 Jun '16

I apologize in advance for the probably idiotic question. I’m working with writing a machine driver for our product and dealing with some potential user interface issues surrounding the number of controls being presented by the codec driver in the mixer. Is there a way to “hard-wire” kcontrols in the codec from the machine driver and hide them in the mixer? For example, our hardware doesn’t wire the IN2_L and IN2_R pins to anywhere, so the codec controls that determine their connections into the input mixer do nothing on our hardware and I’d like to hide them from the user in the mixer. Am I missing something or is there just no way to get there from here? -- Jeremy McDermond (NH6Z) Xenotropic Systems mcdermj(a)xenotropic.com

4 6

[alsa-devel] [PATCH] ASoC: rt298: Add jack suspend and resume functions.
by Bard Liao 13 Jun '16

13 Jun '16

We need to disable jack detection before entering suspend and enable it after resume. This was essential as headphone jack widget will always remain on if a jack is inserted and results in no audio after system resumes from a suspend. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 20 ++++++++++++++++++++ sound/soc/codecs/rt298.h | 2 ++ 2 files changed, 22 insertions(+) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index d403c20..629b038 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1270,6 +1270,26 @@ static int rt298_i2c_remove(struct i2c_client *i2c) return 0; } +void rt298_jack_suspend(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm; + + dapm = snd_soc_codec_get_dapm(rt298->codec); + snd_soc_dapm_disable_pin(dapm, "LDO1"); + snd_soc_dapm_sync(dapm); +} +EXPORT_SYMBOL_GPL(rt298_jack_suspend); + +void rt298_jack_resume(struct snd_soc_codec *codec) +{ + struct rt298_priv *rt298 = snd_soc_codec_get_drvdata(codec); + bool hp = false; + bool mic = false; + + rt298_jack_detect(rt298, &hp, &mic); +} +EXPORT_SYMBOL_GPL(rt298_jack_resume); static struct i2c_driver rt298_i2c_driver = { .driver = { diff --git a/sound/soc/codecs/rt298.h b/sound/soc/codecs/rt298.h index 3638f3d..2de342f 100644 --- a/sound/soc/codecs/rt298.h +++ b/sound/soc/codecs/rt298.h @@ -211,6 +211,8 @@ enum { }; int rt298_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack); +void rt298_jack_suspend(struct snd_soc_codec *codec); +void rt298_jack_resume(struct snd_soc_codec *codec); #endif /* __RT298_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] ASoC: wm8985: add i2c dependency
by Arnd Bergmann 13 Jun '16

13 Jun '16

The wm8985 driver is now user-selectable, but building it with I2C disabled results in a link failure: sound/built-in.o: In function `wm8985_i2c_probe': :(.text+0x44914): undefined reference to `__devm_regmap_init_i2c' sound/built-in.o: In function `wm8985_exit': :(.exit.text+0x3d8): undefined reference to `i2c_del_driver' sound/built-in.o: In function `wm8985_modinit': :(.init.text+0x1454): undefined reference to `i2c_register_driver' This adds a Kconfig dependency the way that the other codec drivers have it. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 811e66de2241 ("ASoC: wm8985: add support for WM8758") --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8fe2690..f555a5a004f7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -986,6 +986,7 @@ config SND_SOC_WM8983 config SND_SOC_WM8985 tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" + depends on I2C config SND_SOC_WM8988 tristate -- 2.7.0

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[alsa-devel] [PATCH 0/2 v9] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 13 Jun '16

13 Jun '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Changes v8 -> v9: * Use defines in IRQ handling Changes v7 -> v8: * Build PIO PCM as module * Always unmask interrupts even when in DMA mode * Fallback to PIO mode only if ALSA DMA engine probe fails Changes v6 -> v7: * Discard the use of memcpy * Report IRQ_HANDLED only when there is an IRQ * Use interrupts to check if PIO mode is in use * Unmask interrupts only when in PIO mode * Remove empty functions Changes v5 -> v6: * Use SNDRV_DMA_TYPE_CONTINUOUS Changes v4 -> v5: * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh(a)kernel.org> Cc: Alexey Brodkin <abrodkin(a)synopsys.com> Cc: linux-snps-arc(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Jose Abreu (2): ASoC: dwc: Add PIO PCM extension ASoC: dwc: Add irq parameter to DOCUMENTATION .../devicetree/bindings/sound/designware-i2s.txt | 4 + sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware_i2s.c | 161 +++++++-------- sound/soc/dwc/designware_pcm.c | 225 +++++++++++++++++++++ sound/soc/dwc/local.h | 128 ++++++++++++ 6 files changed, 440 insertions(+), 88 deletions(-) create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/local.h -- 1.9.1

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[alsa-devel] [PATCH 0/4 v7] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 13 Jun '16

13 Jun '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Changes v6 -> v7: * Discard the use of memcpy * Report IRQ_HANDLED only when there is an IRQ * Use interrupts to check if PIO mode is in use * Unmask interrupts only when in PIO mode * Remove empty functions Changes v5 -> v6: * Use SNDRV_DMA_TYPE_CONTINUOUS Changes v4 -> v5: * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh(a)kernel.org> Cc: Alexey Brodkin <abrodkin(a)synopsys.com> Cc: linux-snps-arc(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: devicetree(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Jose Abreu (4): ASoC: dwc: Add helper functions to disable/enable irqs ASoC: dwc: Do not use devm_clk_get() if using platform data ASoC: dwc: Add PIO PCM extension ASoC: dwc: Add irq parameter to DOCUMENTATION .../devicetree/bindings/sound/designware-i2s.txt | 4 + sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 5 +- sound/soc/dwc/designware_i2s.c | 229 ++++++++++----------- sound/soc/dwc/designware_pcm.c | 220 ++++++++++++++++++++ sound/soc/dwc/local.h | 122 +++++++++++ 6 files changed, 467 insertions(+), 122 deletions(-) create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/local.h -- 1.9.1

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[alsa-devel] [PATCH] ASoC: cs53l30: Correct clock inversion check
by Nicolin Chen 13 Jun '16

13 Jun '16

SND_SOC_DAIFMT_IB_NF = 0x3 (11b) | SND_SOC_DAIFMT_IB_IF = 0x4 (100b) creates a mask 0x7 (111b) which also includes SND_SOC_DAIFMT_IB_NF = 0x2 (10b). So this patch uses the traditional way to check the clock inversion. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ac90dd7..62daa0c 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -598,8 +598,14 @@ static int cs53l30_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* Check to see if the SCLK is inverted */ - if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + case SND_SOC_DAIFMT_IB_IF: aspcfg ^= CS53L30_ASP_SCLK_INV; + break; + default: + break; + } regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL, CS53L30_ASP_MS | CS53L30_ASP_SCLK_INV, aspcfg); -- 2.1.4

2 2

[alsa-devel] [PATCH v3 0/4] Propagate errors out from compressed streams
by Charles Keepax 13 Jun '16

13 Jun '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v2: - Make naming of stop function more generic and allow it to specify target state Thanks, Charles Charles Keepax (4): ALSA: compress: Replace complex if statement with switch ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: compress: Pass error out of soc_compr_pointer include/sound/compress_driver.h | 4 +++ sound/core/compress_offload.c | 69 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 +++++-- sound/soc/soc-compress.c | 5 +-- 4 files changed, 81 insertions(+), 8 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v7 0/3] Propagate errors out from compressed streams
by Charles Keepax 13 Jun '16

13 Jun '16

If the DSP suffers an unrecoverable error, the driver likely knows about this, however the framework may not get informed because errors returned from pointer requests are ignored within the framework. Things work out fine if user-space is doing a read as reads return error status back to user-space so the user can find out that things have gone bad. However, if user-space is doing an avail request there is no path for the error to come back up to user-space. The pointer request returns zero available data, so a read never happens and we basically just end up sitting waiting for data on a stream that we know full well has died. This patch set attempts to address this and ensure that errors are fully propagated to user-space and we don't ever end up wait for data that will never come. Changes since v6: - Series rebased on top of my other series fixing the return values from snd_compr_poll These are based on Takashi's tree, as all the ADSP conflicts are now straightened out, although there is a reasonable chance we will do more ADSP during this cycle. Thanks, Charles Charles Keepax (3): ALSA: compress: Add function to indicate the stream has gone bad ASoC: wm_adsp: Use new snd_compr_stop_error to signal stream failure ASoC: wm_adsp: Treat missing compressed buffer as a fatal error include/sound/compress_driver.h | 5 +++ sound/core/compress_offload.c | 67 +++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm_adsp.c | 21 ++++++------- 3 files changed, 80 insertions(+), 13 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH] ASoC: nau8825: cross talk suppression measurement function
by John Hsu 13 Jun '16

13 Jun '16

The cross talk measurement function can reduce cross talk across the JKTIP HPL) and JKR1(HPR) outputs which measures the cross talk signal level to determine what is the cross talk reduction gain. This system works by sending a 23Hz -24dBV sine wave into the headset output DAC and through the PGA. The output of the PGA is then connected to an internal current sense which measures the attenuated 23Hz signal and passing the output to an ADC which converts the measurement to a binary code. With two separated measurement, one for JKR1(HPR) and the other JKTIP(HPL), measurement data can be separated read in IMM_RMS_L for HSR and HSL after each measurement. Thus, the measurement function has four states to complete whole sequence. (1)Prepare state : Prepare the resource for detection and transfer to HPR IMM stat to make JKR1(HPR) impedance measure. (2)HPR IMM state : Read out orignal signal level of JKR1(HPR) and transfer to HPL IMM state to make JKTIP(HPL) impedance measure. (3)HPL IMM state : Read out cross talk signal level of JKTIP(HPL) and transfer to IMM state to determine suppression sidetone gain. (4)IMM state : Computes cross talk suppression sidetone gain with orignal and cross talk signal level. Apply this gain and then restore codec con- figuration. Then transfer to Done state for ending. In order to get the cross talk suppression sidetone gain, we need the function to compute log10 value and the result is round off to 3 decimal. This function takes reference to dvb-math. The source code locates as the following. "Linux/drivers/media/dvb-core/dvb_math.c" Then, the orignal and cross talk signal vlues need to be characterized. The sidetone value can be converted to decibel with the equation below. sidetone = 20 * log (original signal level / crosstalk signal level) Besides, the state machine for cross talk process needs interruptions to trigger worked. We have the RMS intrruption enabled with the internal VCO clock when headset connected. In the interrupt handler, the driver will judge the headset is high impedance or not. If yes, the cross talk supp- ression shouldn't apply and do nothing but relieve the protection raised before. Otherwise, apply the cross talk suppression in the headset and start the process. Because the process spends a lot of time, there is an resource race issue easily between the application and interruption. They will control codec power and clock concurrently. In one situaiton, the jack is inserted when playback, and then the application changes to headset device. The applica- tion prepares the playback and interrupt handler raises work for cross talk process together. For this case, the solution is that driver delays soc jack report until cross talk process completes. The mechanism can avoid application to do playback preparation before cross talk detection is still working. In another situaiton, the system suspends when playback. After resume, the system restarts playback, and meanwhile jack detection restarts. The play- back preparation and cross talk process triggered by interruptions happens concurrently. For the case, the driver provides the semaphone to syn- chronize the playback and interrupt handler. In order to avoid the play- back interfered by cross talk process, the driver make the playback prepa- ration halted until cross talk process finish. After codec resume, the driver finds the codec dai is active, and then the driver raises the pro- tection for cross talk function to avoid the playback recovers before cross talk process finish. The driver also provides cancel method to forcely cancel the cross talk task and restores the configuration to original status. Before the codec remove, ejection, or suspend, the driver is obliged to cancel the cross talk detection process. It can reduce the risk of failure when quickly and continually doing jack insertion and ejection. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 785 ++++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/nau8825.h | 82 ++++- 2 files changed, 865 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 43cb677..4b0a1b8 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -18,6 +18,7 @@ #include <linux/clk.h> #include <linux/acpi.h> #include <linux/math64.h> +#include <linux/semaphore.h> #include <sound/initval.h> #include <sound/tlv.h> @@ -37,6 +38,12 @@ #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +/* cross talk suppression detection */ +#define LOG10_MAGIC 646456993 +#define GAIN_AUGMENT 22500 +#define SIDETONE_BASE 207000 + + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq); @@ -162,6 +169,661 @@ static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_CHARGE_PUMP, 0x0 }, }; +/* register backup table when cross talk detection */ +static struct reg_default nau8825_xtalk_baktab[] = { + { NAU8825_REG_ADC_DGAIN_CTRL, 0 }, + { NAU8825_REG_HSVOL_CTRL, 0 }, + { NAU8825_REG_DACL_CTRL, 0 }, + { NAU8825_REG_DACR_CTRL, 0 }, +}; + +static const unsigned short logtable[256] = { + 0x0000, 0x0171, 0x02e0, 0x044e, 0x05ba, 0x0725, 0x088e, 0x09f7, + 0x0b5d, 0x0cc3, 0x0e27, 0x0f8a, 0x10eb, 0x124b, 0x13aa, 0x1508, + 0x1664, 0x17bf, 0x1919, 0x1a71, 0x1bc8, 0x1d1e, 0x1e73, 0x1fc6, + 0x2119, 0x226a, 0x23ba, 0x2508, 0x2656, 0x27a2, 0x28ed, 0x2a37, + 0x2b80, 0x2cc8, 0x2e0f, 0x2f54, 0x3098, 0x31dc, 0x331e, 0x345f, + 0x359f, 0x36de, 0x381b, 0x3958, 0x3a94, 0x3bce, 0x3d08, 0x3e41, + 0x3f78, 0x40af, 0x41e4, 0x4319, 0x444c, 0x457f, 0x46b0, 0x47e1, + 0x4910, 0x4a3f, 0x4b6c, 0x4c99, 0x4dc5, 0x4eef, 0x5019, 0x5142, + 0x526a, 0x5391, 0x54b7, 0x55dc, 0x5700, 0x5824, 0x5946, 0x5a68, + 0x5b89, 0x5ca8, 0x5dc7, 0x5ee5, 0x6003, 0x611f, 0x623a, 0x6355, + 0x646f, 0x6588, 0x66a0, 0x67b7, 0x68ce, 0x69e4, 0x6af8, 0x6c0c, + 0x6d20, 0x6e32, 0x6f44, 0x7055, 0x7165, 0x7274, 0x7383, 0x7490, + 0x759d, 0x76aa, 0x77b5, 0x78c0, 0x79ca, 0x7ad3, 0x7bdb, 0x7ce3, + 0x7dea, 0x7ef0, 0x7ff6, 0x80fb, 0x81ff, 0x8302, 0x8405, 0x8507, + 0x8608, 0x8709, 0x8809, 0x8908, 0x8a06, 0x8b04, 0x8c01, 0x8cfe, + 0x8dfa, 0x8ef5, 0x8fef, 0x90e9, 0x91e2, 0x92db, 0x93d2, 0x94ca, + 0x95c0, 0x96b6, 0x97ab, 0x98a0, 0x9994, 0x9a87, 0x9b7a, 0x9c6c, + 0x9d5e, 0x9e4f, 0x9f3f, 0xa02e, 0xa11e, 0xa20c, 0xa2fa, 0xa3e7, + 0xa4d4, 0xa5c0, 0xa6ab, 0xa796, 0xa881, 0xa96a, 0xaa53, 0xab3c, + 0xac24, 0xad0c, 0xadf2, 0xaed9, 0xafbe, 0xb0a4, 0xb188, 0xb26c, + 0xb350, 0xb433, 0xb515, 0xb5f7, 0xb6d9, 0xb7ba, 0xb89a, 0xb97a, + 0xba59, 0xbb38, 0xbc16, 0xbcf4, 0xbdd1, 0xbead, 0xbf8a, 0xc065, + 0xc140, 0xc21b, 0xc2f5, 0xc3cf, 0xc4a8, 0xc580, 0xc658, 0xc730, + 0xc807, 0xc8de, 0xc9b4, 0xca8a, 0xcb5f, 0xcc34, 0xcd08, 0xcddc, + 0xceaf, 0xcf82, 0xd054, 0xd126, 0xd1f7, 0xd2c8, 0xd399, 0xd469, + 0xd538, 0xd607, 0xd6d6, 0xd7a4, 0xd872, 0xd93f, 0xda0c, 0xdad9, + 0xdba5, 0xdc70, 0xdd3b, 0xde06, 0xded0, 0xdf9a, 0xe063, 0xe12c, + 0xe1f5, 0xe2bd, 0xe385, 0xe44c, 0xe513, 0xe5d9, 0xe69f, 0xe765, + 0xe82a, 0xe8ef, 0xe9b3, 0xea77, 0xeb3b, 0xebfe, 0xecc1, 0xed83, + 0xee45, 0xef06, 0xefc8, 0xf088, 0xf149, 0xf209, 0xf2c8, 0xf387, + 0xf446, 0xf505, 0xf5c3, 0xf680, 0xf73e, 0xf7fb, 0xf8b7, 0xf973, + 0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47 +}; + +static struct snd_soc_dai *nau8825_get_codec_dai(struct nau8825 *nau8825) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); + struct snd_soc_component *component = &codec->component; + struct snd_soc_dai *codec_dai, *_dai; + + list_for_each_entry_safe(codec_dai, _dai, &component->dai_list, list) { + if (!strncmp(codec_dai->name, NUVOTON_CODEC_DAI, + strlen(NUVOTON_CODEC_DAI))) + return codec_dai; + } + return NULL; +} + +static bool nau8825_dai_is_active(struct nau8825 *nau8825) +{ + struct snd_soc_dai *codec_dai = nau8825_get_codec_dai(nau8825); + + if (codec_dai) { + if (codec_dai->playback_active || codec_dai->capture_active) + return true; + } + return false; +} + +/** + * nau8825_sema_acquire - acquire the semaphore of nau88l25 + * @nau8825: component to register the codec private data with + * @timeout: how long in jiffies to wait before failure or zero to wait + * until release + * + * Attempts to acquire the semaphore with number of jiffies. If no more + * tasks are allowed to acquire the semaphore, calling this function will + * put the task to sleep. If the semaphore is not released within the + * specified number of jiffies, this function returns. + * Acquires the semaphore without jiffies. If no more tasks are allowed + * to acquire the semaphore, calling this function will put the task to + * sleep until the semaphore is released. + * It returns if the semaphore was acquired. + */ +static void nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) +{ + int ret; + + if (timeout) + ret = down_timeout(&nau8825->xtalk_sem, timeout); + else + ret = down_interruptible(&nau8825->xtalk_sem); + + if (ret < 0) + dev_warn(nau8825->dev, "Acquire semaphone fail\n"); +} + +/** + * nau8825_sema_release - release the semaphore of nau88l25 + * @nau8825: component to register the codec private data with + * + * Release the semaphore which may be called from any context and + * even by tasks which have never called down(). + */ +static inline void nau8825_sema_release(struct nau8825 *nau8825) +{ + up(&nau8825->xtalk_sem); +} + +/** + * nau8825_sema_reset - reset the semaphore for nau88l25 + * @nau8825: component to register the codec private data with + * + * Reset the counter of the semaphore. Call this function to restart + * a new round task management. + */ +static inline void nau8825_sema_reset(struct nau8825 *nau8825) +{ + nau8825->xtalk_sem.count = 1; +} + +/** + * Ramp up the headphone volume change gradually to target level. + * + * @nau8825: component to register the codec private data with + * @vol_from: the volume to start up + * @vol_to: the target volume + * @step: the volume span to move on + * + * The headphone volume is from 0dB to minimum -54dB and -1dB per step. + * If the volume changes sharp, there is a pop noise heard in headphone. We + * provide the function to ramp up the volume up or down by delaying 10ms + * per step. + */ +static void nau8825_hpvol_ramp(struct nau8825 *nau8825, + unsigned int vol_from, unsigned int vol_to, unsigned int step) +{ + unsigned int value, volume, ramp_up, from, to; + + if (vol_from == vol_to || step == 0) { + return; + } else if (vol_from < vol_to) { + ramp_up = true; + from = vol_from; + to = vol_to; + } else { + ramp_up = false; + from = vol_to; + to = vol_from; + } + /* only handle volume from 0dB to minimum -54dB */ + if (to > NAU8825_HP_VOL_MIN) + to = NAU8825_HP_VOL_MIN; + + for (volume = from; volume < to; volume += step) { + if (ramp_up) + value = volume; + else + value = to - volume + from; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HPL_VOL_MASK | NAU8825_HPR_VOL_MASK, + (value << NAU8825_HPL_VOL_SFT) | value); + usleep_range(10000, 10500); + } + if (ramp_up) + value = to; + else + value = from; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HPL_VOL_MASK | NAU8825_HPR_VOL_MASK, + (value << NAU8825_HPL_VOL_SFT) | value); +} + +/** + * Computes log10 of a value; the result is round off to 3 decimal. This func- + * tion takes reference to dvb-math. The source code locates as the following. + * Linux/drivers/media/dvb-core/dvb_math.c + * + * return log10(value) * 1000 + */ +static u32 nau8825_intlog10_dec3(u32 value) +{ + u32 msb, logentry, significand, interpolation, log10val; + u64 log2val; + + /* first detect the msb (count begins at 0) */ + msb = fls(value) - 1; + /** + * now we use a logtable after the following method: + * + * log2(2^x * y) * 2^24 = x * 2^24 + log2(y) * 2^24 + * where x = msb and therefore 1 <= y < 2 + * first y is determined by shifting the value left + * so that msb is bit 31 + * 0x00231f56 -> 0x8C7D5800 + * the result is y * 2^31 -> "significand" + * then the highest 9 bits are used for a table lookup + * the highest bit is discarded because it's always set + * the highest nine bits in our example are 100011000 + * so we would use the entry 0x18 + */ + significand = value << (31 - msb); + logentry = (significand >> 23) & 0xff; + /** + * last step we do is interpolation because of the + * limitations of the log table the error is that part of + * the significand which isn't used for lookup then we + * compute the ratio between the error and the next table entry + * and interpolate it between the log table entry used and the + * next one the biggest error possible is 0x7fffff + * (in our example it's 0x7D5800) + * needed value for next table entry is 0x800000 + * so the interpolation is + * (error / 0x800000) * (logtable_next - logtable_current) + * in the implementation the division is moved to the end for + * better accuracy there is also an overflow correction if + * logtable_next is 256 + */ + interpolation = ((significand & 0x7fffff) * + ((logtable[(logentry + 1) & 0xff] - + logtable[logentry]) & 0xffff)) >> 15; + + log2val = ((msb << 24) + (logtable[logentry] << 8) + interpolation); + /** + * log10(x) = log2(x) * log10(2) + */ + log10val = (log2val * LOG10_MAGIC) >> 31; + /** + * the result is round off to 3 decimal + */ + return log10val / ((1 << 24) / 1000); +} + +/** + * computes cross talk suppression sidetone gain. + * + * @sig_org: orignal signal level + * @sig_cros: cross talk signal level + * + * The orignal and cross talk signal vlues need to be characterized. + * Once these values have been characterized, this sidetone value + * can be converted to decibel with the equation below. + * sidetone = 20 * log (original signal level / crosstalk signal level) + * + * return cross talk sidetone gain + */ +static u32 nau8825_xtalk_sidetone(u32 sig_org, u32 sig_cros) +{ + u32 gain, sidetone; + + if (unlikely(sig_org == 0) || unlikely(sig_cros == 0)) { + WARN_ON(1); + return 0; + } + + sig_org = nau8825_intlog10_dec3(sig_org); + sig_cros = nau8825_intlog10_dec3(sig_cros); + if (sig_org >= sig_cros) + gain = (sig_org - sig_cros) * 20 + GAIN_AUGMENT; + else + gain = (sig_cros - sig_org) * 20 + GAIN_AUGMENT; + sidetone = SIDETONE_BASE - gain * 2; + sidetone /= 1000; + + return sidetone; +} + +static int nau8825_xtalk_baktab_index_by_reg(unsigned int reg) +{ + int index; + + for (index = 0; index < ARRAY_SIZE(nau8825_xtalk_baktab); index++) + if (nau8825_xtalk_baktab[index].reg == reg) + return index; + return -EINVAL; +} + +static void nau8825_xtalk_backup(struct nau8825 *nau8825) +{ + int i; + + /* Backup some register values to backup table */ + for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) + regmap_read(nau8825->regmap, nau8825_xtalk_baktab[i].reg, + &nau8825_xtalk_baktab[i].def); +} + +static void nau8825_xtalk_restore(struct nau8825 *nau8825) +{ + int i, volume; + + /* Restore register values from backup table; When the driver restores + * the headphone volumem, it needs recover to original level gradually + * with 3dB per step for less pop noise. + */ + for (i = 0; i < ARRAY_SIZE(nau8825_xtalk_baktab); i++) { + if (nau8825_xtalk_baktab[i].reg == NAU8825_REG_HSVOL_CTRL) { + /* Ramping up the volume change to reduce pop noise */ + volume = nau8825_xtalk_baktab[i].def & + NAU8825_HPR_VOL_MASK; + nau8825_hpvol_ramp(nau8825, 0, volume, 3); + continue; + } + regmap_write(nau8825->regmap, nau8825_xtalk_baktab[i].reg, + nau8825_xtalk_baktab[i].def); + } +} + +static void nau8825_xtalk_prepare_dac(struct nau8825 *nau8825) +{ + /* Enable power of DAC path */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR | NAU8825_ENABLE_DACL | + NAU8825_ENABLE_ADC | NAU8825_ENABLE_ADC_CLK | + NAU8825_ENABLE_DAC_CLK, NAU8825_ENABLE_DACR | + NAU8825_ENABLE_DACL | NAU8825_ENABLE_ADC | + NAU8825_ENABLE_ADC_CLK | NAU8825_ENABLE_DAC_CLK); + /* Prevent startup click by letting charge pump to ramp up and + * change bump enable + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN); + /* Enable clock sync of DAC and DAC clock */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN | + NAU8825_RDAC_FS_BCLK_ENB, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN); + /* Power up output driver with 2 stage */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L, + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L); + /* HP outputs not shouted to ground */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, 0); + /* Enable HP boost driver */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, NAU8825_HP_BOOST_DIS); + /* Enable class G compare path to supply 1.8V or 0.9V. */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLASSG_CTRL, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN, + NAU8825_CLASSG_LDAC_EN | NAU8825_CLASSG_RDAC_EN); +} + +static void nau8825_xtalk_prepare_adc(struct nau8825 *nau8825) +{ + /* Power up left ADC and raise 5dB than Vmid for Vref */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ANALOG_ADC_2, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB); +} + +static void nau8825_xtalk_clock(struct nau8825 *nau8825) +{ + /* Recover FLL default value */ + regmap_write(nau8825->regmap, NAU8825_REG_FLL1, 0x0); + regmap_write(nau8825->regmap, NAU8825_REG_FLL2, 0x3126); + regmap_write(nau8825->regmap, NAU8825_REG_FLL3, 0x0008); + regmap_write(nau8825->regmap, NAU8825_REG_FLL4, 0x0010); + regmap_write(nau8825->regmap, NAU8825_REG_FLL5, 0x0); + regmap_write(nau8825->regmap, NAU8825_REG_FLL6, 0x6000); + /* Enable internal VCO clock for detection signal generated */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + /* Given specific clock frequency of internal clock to + * generate signal. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(nau8825->regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); +} + +static void nau8825_xtalk_prepare(struct nau8825 *nau8825) +{ + int volume, index; + + /* Backup those registers changed by cross talk detection */ + nau8825_xtalk_backup(nau8825); + /* Config IIS as master to output signal by codec */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_MASTER | + (0x2 << NAU8825_I2S_DRV_SFT) | 0x1); + /* Ramp up headphone volume to 0dB to get better performance and + * avoid pop noise in headphone. + */ + index = nau8825_xtalk_baktab_index_by_reg(NAU8825_REG_HSVOL_CTRL); + if (index != -EINVAL) { + volume = nau8825_xtalk_baktab[index].def & + NAU8825_HPR_VOL_MASK; + nau8825_hpvol_ramp(nau8825, volume, 0, 3); + } + nau8825_xtalk_clock(nau8825); + nau8825_xtalk_prepare_dac(nau8825); + nau8825_xtalk_prepare_adc(nau8825); + /* Config channel path and digital gain */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_MASK | NAU8825_DACL_CH_VOL_MASK, + NAU8825_DACL_CH_SEL_L | 0xab); + regmap_update_bits(nau8825->regmap, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_MASK | NAU8825_DACR_CH_VOL_MASK, + NAU8825_DACR_CH_SEL_R | 0xab); + /* Config cross talk parameters and generate the 23Hz sine wave with + * 1/16 full scale of signal level for impedance measurement. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, + NAU8825_IMM_THD_MASK | NAU8825_IMM_GEN_VOL_MASK | + NAU8825_IMM_CYC_MASK | NAU8825_IMM_DAC_SRC_MASK, + (0x9 << NAU8825_IMM_THD_SFT) | NAU8825_IMM_GEN_VOL_1_16th | + NAU8825_IMM_CYC_8192 | NAU8825_IMM_DAC_SRC_SIN); + /* RMS intrruption enable */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_INTERRUPT_MASK, NAU8825_IRQ_RMS_EN, 0); + /* Power up left and right DAC */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, 0); +} + +static void nau8825_xtalk_clean_dac(struct nau8825 *nau8825) +{ + /* Disable HP boost driver */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_HP_BOOST_DIS, 0); + /* HP outputs shouted to ground */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSD_CTRL, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + /* Power down left and right DAC */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL, + NAU8825_POWER_DOWN_DACR | NAU8825_POWER_DOWN_DACL); + /* Enable the TESTDAC and disable L/R HP impedance */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP | + NAU8825_BIAS_TESTDAC_EN, NAU8825_BIAS_TESTDAC_EN); + /* Power down output driver with 2 stage */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_HP_DRV_R | NAU8825_POWERUP_HP_DRV_L, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_POWER_UP_CONTROL, + NAU8825_POWERUP_INTEGR_R | NAU8825_POWERUP_INTEGR_L | + NAU8825_POWERUP_DRV_IN_R | NAU8825_POWERUP_DRV_IN_L, 0); + /* Disable clock sync of DAC and DAC clock */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_RDAC, + NAU8825_RDAC_EN | NAU8825_RDAC_CLK_EN, 0); + /* Disable charge pump ramp up function and change bump */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW | NAU8825_CHANRGE_PUMP_EN, 0); + /* Disable power of DAC path */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR | NAU8825_ENABLE_DACL | + NAU8825_ENABLE_ADC_CLK | NAU8825_ENABLE_DAC_CLK, 0); + if (!nau8825->irq) + regmap_update_bits(nau8825->regmap, + NAU8825_REG_ENA_CTRL, NAU8825_ENABLE_ADC, 0); +} + +static void nau8825_xtalk_clean_adc(struct nau8825 *nau8825) +{ + /* Power down left ADC and restore voltage to Vmid */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ANALOG_ADC_2, + NAU8825_POWERUP_ADCL | NAU8825_ADC_VREFSEL_MASK, 0); +} + +static void nau8825_xtalk_clean(struct nau8825 *nau8825) +{ + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + nau8825_xtalk_clean_dac(nau8825); + nau8825_xtalk_clean_adc(nau8825); + /* Clear cross talk parameters and disable */ + regmap_write(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, 0); + /* RMS intrruption disable */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_RMS_EN, NAU8825_IRQ_RMS_EN); + /* Recover default value for IIS */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK | NAU8825_I2S_DRV_MASK | + NAU8825_I2S_BLK_DIV_MASK, NAU8825_I2S_MS_SLAVE); + /* Restore value of specific register for cross talk */ + nau8825_xtalk_restore(nau8825); +} + +static void nau8825_xtalk_imm_start(struct nau8825 *nau8825, int vol) +{ + /* Apply ADC volume for better cross talk performance */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ADC_DGAIN_CTRL, + NAU8825_ADC_DIG_VOL_MASK, vol); + /* Disables JKTIP(HPL) DAC channel for right to left measurement. + * Do it before sending signal in order to erase pop noise. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_TESTDACR_EN | NAU8825_BIAS_TESTDACL_EN, + NAU8825_BIAS_TESTDACL_EN); + switch (nau8825->xtalk_state) { + case NAU8825_XTALK_HPR_R2L: + /* Enable right headphone impedance */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP, + NAU8825_BIAS_HPR_IMP); + break; + case NAU8825_XTALK_HPL_R2L: + /* Enable left headphone impedance */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_HPR_IMP | NAU8825_BIAS_HPL_IMP, + NAU8825_BIAS_HPL_IMP); + break; + default: + break; + } + msleep(100); + /* Impedance measurement mode enable */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_IMM_MODE_CTRL, + NAU8825_IMM_EN, NAU8825_IMM_EN); +} + +static void nau8825_xtalk_imm_stop(struct nau8825 *nau8825) +{ + /* Impedance measurement mode disable */ + regmap_update_bits(nau8825->regmap, + NAU8825_REG_IMM_MODE_CTRL, NAU8825_IMM_EN, 0); +} + +/* The cross talk measurement function can reduce cross talk across the + * JKTIP(HPL) and JKR1(HPR) outputs which measures the cross talk signal + * level to determine what cross talk reduction gain is. This system works by + * sending a 23Hz -24dBV sine wave into the headset output DAC and through + * the PGA. The output of the PGA is then connected to an internal current + * sense which measures the attenuated 23Hz signal and passing the output to + * an ADC which converts the measurement to a binary code. With two separated + * measurement, one for JKR1(HPR) and the other JKTIP(HPL), measurement data + * can be separated read in IMM_RMS_L for HSR and HSL after each measurement. + * Thus, the measurement function has four states to complete whole sequence. + * 1. Prepare state : Prepare the resource for detection and transfer to HPR + * IMM stat to make JKR1(HPR) impedance measure. + * 2. HPR IMM state : Read out orignal signal level of JKR1(HPR) and transfer + * to HPL IMM state to make JKTIP(HPL) impedance measure. + * 3. HPL IMM state : Read out cross talk signal level of JKTIP(HPL) and + * transfer to IMM state to determine suppression sidetone gain. + * 4. IMM state : Computes cross talk suppression sidetone gain with orignal + * and cross talk signal level. Apply this gain and then restore codec + * configuration. Then transfer to Done state for ending. + */ +static void nau8825_xtalk_measure(struct nau8825 *nau8825) +{ + u32 sidetone; + + switch (nau8825->xtalk_state) { + case NAU8825_XTALK_PREPARE: + /* In prepare state, set up clock, intrruption, DAC path, ADC + * path and cross talk detection parameters for preparation. + */ + nau8825_xtalk_prepare(nau8825); + msleep(280); + /* Trigger right headphone impedance detection */ + nau8825->xtalk_state = NAU8825_XTALK_HPR_R2L; + nau8825_xtalk_imm_start(nau8825, 0x00d2); + break; + case NAU8825_XTALK_HPR_R2L: + /* In right headphone IMM state, read out right headphone + * impedance measure result, and then start up left side. + */ + regmap_read(nau8825->regmap, NAU8825_REG_IMM_RMS_L, + &nau8825->imp_rms[NAU8825_XTALK_HPR_R2L]); + dev_dbg(nau8825->dev, "HPR_R2L imm: %x\n", + nau8825->imp_rms[NAU8825_XTALK_HPR_R2L]); + /* Disable then re-enable IMM mode to update */ + nau8825_xtalk_imm_stop(nau8825); + /* Trigger left headphone impedance detection */ + nau8825->xtalk_state = NAU8825_XTALK_HPL_R2L; + nau8825_xtalk_imm_start(nau8825, 0x00ff); + break; + case NAU8825_XTALK_HPL_R2L: + /* In left headphone IMM state, read out left headphone + * impedance measure result, and delay some time to wait + * detection sine wave output finish. Then, we can calculate + * the cross talk suppresstion side tone according to the L/R + * headphone imedance. + */ + regmap_read(nau8825->regmap, NAU8825_REG_IMM_RMS_L, + &nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + dev_dbg(nau8825->dev, "HPL_R2L imm: %x\n", + nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + nau8825_xtalk_imm_stop(nau8825); + msleep(150); + nau8825->xtalk_state = NAU8825_XTALK_IMM; + break; + case NAU8825_XTALK_IMM: + /* In impedance measure state, the orignal and cross talk + * signal level vlues are ready. The side tone gain is deter- + * mined with these signal level. After all, restore codec + * configuration. + */ + sidetone = nau8825_xtalk_sidetone( + nau8825->imp_rms[NAU8825_XTALK_HPR_R2L], + nau8825->imp_rms[NAU8825_XTALK_HPL_R2L]); + dev_dbg(nau8825->dev, "cross talk sidetone: %x\n", sidetone); + regmap_write(nau8825->regmap, NAU8825_REG_DAC_DGAIN_CTRL, + (sidetone << 8) | sidetone); + nau8825_xtalk_clean(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_DONE; + break; + default: + break; + } +} + +static void nau8825_xtalk_work(struct work_struct *work) +{ + struct nau8825 *nau8825 = container_of( + work, struct nau8825, xtalk_work); + + nau8825_xtalk_measure(nau8825); + /* To determine the cross talk side tone gain when reach + * the impedance measure state. + */ + if (nau8825->xtalk_state == NAU8825_XTALK_IMM) + nau8825_xtalk_measure(nau8825); + + /* Delay jack report until cross talk detection process + * completed. It can avoid application to do playback + * preparation before cross talk detection is still working. + * Meanwhile, the protection of the cross talk detection + * is released. + */ + if (nau8825->xtalk_state == NAU8825_XTALK_DONE) { + snd_soc_jack_report(nau8825->jack, nau8825->xtalk_event, + nau8825->xtalk_event_mask); + nau8825_sema_release(nau8825); + nau8825->xtalk_protect = false; + } +} + +static void nau8825_xtalk_cancel(struct nau8825 *nau8825) +{ + /* If the xtalk_protect is true, that means the process is still + * on going. The driver forces to cancel the cross talk task and + * restores the configuration to original status. + */ + if (nau8825->xtalk_protect) { + cancel_work_sync(&nau8825->xtalk_work); + nau8825_xtalk_clean(nau8825); + } + /* Reset parameters for cross talk suppression function */ + nau8825_sema_reset(nau8825); + nau8825->xtalk_state = NAU8825_XTALK_DONE; + nau8825->xtalk_protect = false; +} + static bool nau8825_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -722,6 +1384,9 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) struct snd_soc_dapm_context *dapm = nau8825->dapm; struct regmap *regmap = nau8825->regmap; + /* Force to cancel the cross talk detection process */ + nau8825_xtalk_cancel(nau8825); + snd_soc_dapm_disable_pin(dapm, "SAR"); snd_soc_dapm_disable_pin(dapm, "MICBIAS"); /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ @@ -826,6 +1491,11 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); mic_detected = (jack_status_reg >> 10) & 3; + /* The JKSLV and JKR2 all detected in high impedance headset */ + if (mic_detected == 0x3) + nau8825->high_imped = true; + else + nau8825->high_imped = false; switch (mic_detected) { case 0: @@ -923,6 +1593,33 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { if (nau8825_is_jack_inserted(regmap)) { event |= nau8825_jack_insert(nau8825); + if (!nau8825->high_imped) { + /* Apply the cross talk suppression in the + * headset without high impedance. + */ + if (!nau8825->xtalk_protect) { + /* Raise protection for cross talk de- + * tection if no protection before. + * The driver has to cancel the pro- + * cess and restore changes if process + * is ongoing when ejection. + */ + nau8825->xtalk_protect = true; + nau8825_sema_acquire(nau8825, 0); + } + /* Startup cross talk detection process */ + nau8825->xtalk_state = NAU8825_XTALK_PREPARE; + schedule_work(&nau8825->xtalk_work); + } else { + /* The cross talk suppression shouldn't apply + * in the headset with high impedance. Thus, + * relieve the protection raised before. + */ + if (nau8825->xtalk_protect) { + nau8825_sema_release(nau8825); + nau8825->xtalk_protect = false; + } + } } else { dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); nau8825_eject_jack(nau8825); @@ -930,6 +1627,17 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + /* Record the interruption report event for driver to report + * the event later. The jack report will delay until cross + * talk detection process is done. + */ + if (nau8825->xtalk_state == NAU8825_XTALK_PREPARE) { + nau8825->xtalk_event = event; + nau8825->xtalk_event_mask = event_mask; + } + } else if (active_irq & NAU8825_IMPEDANCE_MEAS_IRQ) { + schedule_work(&nau8825->xtalk_work); + clear_irq = NAU8825_IMPEDANCE_MEAS_IRQ; } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == NAU8825_JACK_INSERTION_DETECTED) { /* One more step to check GPIO status directly. Thus, the @@ -957,7 +1665,12 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) /* clears the rightmost interruption */ regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); - if (event_mask) + /* Delay jack report until cross talk detection is done. It can avoid + * application to do playback preparation when cross talk detection + * process is still working. Otherwise, the resource like clock and + * power will be issued by them at the same time and conflict happens. + */ + if (event_mask && nau8825->xtalk_state == NAU8825_XTALK_DONE) snd_soc_jack_report(nau8825->jack, event, event_mask); return IRQ_HANDLED; @@ -1122,6 +1835,16 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) return 0; } +static int nau8825_codec_remove(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + /* Cancel and reset cross tak suppresstion detection funciton */ + nau8825_xtalk_cancel(nau8825); + + return 0; +} + /** * nau8825_calc_fll_param - Calculate FLL parameters. * @fll_in: external clock provided to codec. @@ -1308,10 +2031,19 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_MCLK: + /* Acquire the semaphone to synchronize the playback and + * interrupt handler. In order to avoid the playback inter- + * fered by cross talk process, the driver make the playback + * preparation halted until cross talk process finish. + */ + nau8825_sema_acquire(nau8825, 2 * HZ); nau8825_configure_mclk_as_sysclk(regmap); /* MCLK not changed by clock tree */ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + ret = nau8825_mclk_prepare(nau8825, freq); if (ret) return ret; @@ -1344,16 +2076,34 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_FLL_MCLK: + /* Acquire the semaphone to synchronize the playback and + * interrupt handler. In order to avoid the playback inter- + * fered by cross talk process, the driver make the playback + * preparation halted until cross talk process finish. + */ + nau8825_sema_acquire(nau8825, 2 * HZ); regmap_update_bits(regmap, NAU8825_REG_FLL3, NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_MCLK); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + ret = nau8825_mclk_prepare(nau8825, freq); if (ret) return ret; break; case NAU8825_CLK_FLL_BLK: + /* Acquire the semaphone to synchronize the playback and + * interrupt handler. In order to avoid the playback inter- + * fered by cross talk process, the driver make the playback + * preparation halted until cross talk process finish. + */ + nau8825_sema_acquire(nau8825, 2 * HZ); regmap_update_bits(regmap, NAU8825_REG_FLL3, NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_BLK); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1361,8 +2111,17 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_FLL_FS: + /* Acquire the semaphone to synchronize the playback and + * interrupt handler. In order to avoid the playback inter- + * fered by cross talk process, the driver make the playback + * preparation halted until cross talk process finish. + */ + nau8825_sema_acquire(nau8825, 2 * HZ); regmap_update_bits(regmap, NAU8825_REG_FLL3, NAU8825_FLL_CLK_SRC_MASK, NAU8825_FLL_CLK_SRC_FS); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1440,6 +2199,8 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: + /* Cancel and reset cross talk detection funciton */ + nau8825_xtalk_cancel(nau8825); /* Turn off all interruptions before system shutdown. Keep the * interruption quiet before resume setup completes. */ @@ -1474,6 +2235,20 @@ static int nau8825_resume(struct snd_soc_codec *codec) regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); + if (nau8825_is_jack_inserted(nau8825->regmap)) { + /* If the jack is inserted, we need to check whether the play- + * back is active before suspend. If active, the driver has to + * raise the protection for cross talk function to avoid the + * playback recovers before cross talk process finish. Other- + * wise, the playback will be interfered by cross talk func- + * tion. It is better to apply hardware related parameters + * before starting playback or record. + */ + if (nau8825_dai_is_active(nau8825)) { + nau8825->xtalk_protect = true; + nau8825_sema_acquire(nau8825, 0); + } + } enable_irq(nau8825->irq); return 0; @@ -1485,6 +2260,7 @@ static int nau8825_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, + .remove = nau8825_codec_remove, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, @@ -1622,6 +2398,13 @@ static int nau8825_i2c_probe(struct i2c_client *i2c, return PTR_ERR(nau8825->regmap); nau8825->dev = dev; nau8825->irq = i2c->irq; + /* Initiate parameters, semaphone and work queue which are needed in + * cross talk suppression measurment function. + */ + nau8825->xtalk_state = NAU8825_XTALK_DONE; + nau8825->xtalk_protect = false; + sema_init(&nau8825->xtalk_sem, 1); + INIT_WORK(&nau8825->xtalk_work, nau8825_xtalk_work); nau8825_print_device_properties(nau8825); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 25aae5c..1c63e2a 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -101,8 +101,13 @@ #define NAU8825_ENABLE_DACR_SFT 10 #define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) #define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_DACL (1 << NAU8825_ENABLE_DACL_SFT) #define NAU8825_ENABLE_ADC_SFT 8 #define NAU8825_ENABLE_ADC (1 << NAU8825_ENABLE_ADC_SFT) +#define NAU8825_ENABLE_ADC_CLK_SFT 7 +#define NAU8825_ENABLE_ADC_CLK (1 << NAU8825_ENABLE_ADC_CLK_SFT) +#define NAU8825_ENABLE_DAC_CLK_SFT 6 +#define NAU8825_ENABLE_DAC_CLK (1 << NAU8825_ENABLE_DAC_CLK_SFT) #define NAU8825_ENABLE_SAR_SFT 1 /* CLK_DIVIDER (0x3) */ @@ -158,6 +163,7 @@ /* INTERRUPT_MASK (0xf) */ #define NAU8825_IRQ_OUTPUT_EN (1 << 11) #define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_RMS_EN (1 << 8) #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) @@ -232,10 +238,13 @@ /* I2S_PCM_CTRL2 (0x1d) */ #define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - normal mode, 1 - Hi-Z output */ +#define NAU8825_I2S_DRV_SFT 12 +#define NAU8825_I2S_DRV_MASK (0x3 << NAU8825_I2S_DRV_SFT) #define NAU8825_I2S_MS_SFT 3 #define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_BLK_DIV_MASK 0x7 /* BIQ_CTRL (0x20) */ #define NAU8825_BIQ_WRT_SFT 4 @@ -262,28 +271,72 @@ #define NAU8825_DAC_OVERSAMPLE_128 2 #define NAU8825_DAC_OVERSAMPLE_32 4 +/* ADC_DGAIN_CTRL (0x30) */ +#define NAU8825_ADC_DIG_VOL_MASK 0xff + /* MUTE_CTRL (0x31) */ #define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) #define NAU8825_DAC_SOFT_MUTE (1 << 9) /* HSVOL_CTRL (0x32) */ #define NAU8825_HP_MUTE (1 << 15) +#define NAU8825_HP_MUTE_AUTO (1 << 14) +#define NAU8825_HPL_MUTE (1 << 13) +#define NAU8825_HPR_MUTE (1 << 12) +#define NAU8825_HPL_VOL_SFT 6 +#define NAU8825_HPL_VOL_MASK (0x3f << NAU8825_HPL_VOL_SFT) +#define NAU8825_HPR_VOL_SFT 0 +#define NAU8825_HPR_VOL_MASK (0x3f << NAU8825_HPR_VOL_SFT) +#define NAU8825_HP_VOL_MIN 0x36 /* DACL_CTRL (0x33) */ #define NAU8825_DACL_CH_SEL_SFT 9 #define NAU8825_DACL_CH_SEL_MASK (0x1 << NAU8825_DACL_CH_SEL_SFT) #define NAU8825_DACL_CH_SEL_L (0x0 << NAU8825_DACL_CH_SEL_SFT) #define NAU8825_DACL_CH_SEL_R (0x1 << NAU8825_DACL_CH_SEL_SFT) +#define NAU8825_DACL_CH_VOL_MASK 0xff /* DACR_CTRL (0x34) */ #define NAU8825_DACR_CH_SEL_SFT 9 #define NAU8825_DACR_CH_SEL_MASK (0x1 << NAU8825_DACR_CH_SEL_SFT) #define NAU8825_DACR_CH_SEL_L (0x0 << NAU8825_DACR_CH_SEL_SFT) #define NAU8825_DACR_CH_SEL_R (0x1 << NAU8825_DACR_CH_SEL_SFT) +#define NAU8825_DACR_CH_VOL_MASK 0xff + +/* IMM_MODE_CTRL (0x4C) */ +#define NAU8825_IMM_THD_SFT 8 +#define NAU8825_IMM_THD_MASK (0x3f << NAU8825_IMM_THD_SFT) +#define NAU8825_IMM_GEN_VOL_SFT 6 +#define NAU8825_IMM_GEN_VOL_MASK (0x3 << NAU8825_IMM_GEN_VOL_SFT) +#define NAU8825_IMM_GEN_VOL_1_2nd (0x0 << NAU8825_IMM_GEN_VOL_SFT) +#define NAU8825_IMM_GEN_VOL_1_4th (0x1 << NAU8825_IMM_GEN_VOL_SFT) +#define NAU8825_IMM_GEN_VOL_1_8th (0x2 << NAU8825_IMM_GEN_VOL_SFT) +#define NAU8825_IMM_GEN_VOL_1_16th (0x3 << NAU8825_IMM_GEN_VOL_SFT) + +#define NAU8825_IMM_CYC_SFT 4 +#define NAU8825_IMM_CYC_MASK (0x3 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_1024 (0x0 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_2048 (0x1 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_4096 (0x2 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_CYC_8192 (0x3 << NAU8825_IMM_CYC_SFT) +#define NAU8825_IMM_EN (1 << 3) +#define NAU8825_IMM_DAC_SRC_MASK 0x7 +#define NAU8825_IMM_DAC_SRC_BIQ 0x0 +#define NAU8825_IMM_DAC_SRC_DRC 0x1 +#define NAU8825_IMM_DAC_SRC_MIX 0x2 +#define NAU8825_IMM_DAC_SRC_SIN 0x3 /* CLASSG_CTRL (0x50) */ #define NAU8825_CLASSG_TIMER_SFT 8 #define NAU8825_CLASSG_TIMER_MASK (0x3f << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_1ms (0x1 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_2ms (0x2 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_8ms (0x4 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_16ms (0x8 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_32ms (0x10 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_TIMER_64ms (0x20 << NAU8825_CLASSG_TIMER_SFT) +#define NAU8825_CLASSG_LDAC_EN (0x1 << 2) +#define NAU8825_CLASSG_RDAC_EN (0x1 << 1) #define NAU8825_CLASSG_EN (1 << 0) /* I2C_DEVICE_ID (0x58) */ @@ -292,7 +345,12 @@ #define NAU8825_SOFTWARE_ID_NAU8825 0x0 /* BIAS_ADJ (0x66) */ -#define NAU8825_BIAS_TESTDAC_EN (0x3 << 8) +#define NAU8825_BIAS_HPR_IMP (1 << 15) +#define NAU8825_BIAS_HPL_IMP (1 << 14) +#define NAU8825_BIAS_TESTDAC_SFT 8 +#define NAU8825_BIAS_TESTDAC_EN (0x3 << NAU8825_BIAS_TESTDAC_SFT) +#define NAU8825_BIAS_TESTDACR_EN (0x2 << NAU8825_BIAS_TESTDAC_SFT) +#define NAU8825_BIAS_TESTDACL_EN (0x1 << NAU8825_BIAS_TESTDAC_SFT) #define NAU8825_BIAS_VMID (1 << 6) #define NAU8825_BIAS_VMID_SEL_SFT 4 #define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) @@ -311,6 +369,11 @@ #define NAU8825_POWERUP_ADCL (1 << 6) /* RDAC (0x73) */ +#define NAU8825_RDAC_FS_BCLK_ENB (1 << 15) +#define NAU8825_RDAC_EN_SFT 12 +#define NAU8825_RDAC_EN (0x3 << NAU8825_RDAC_EN_SFT) +#define NAU8825_RDAC_CLK_EN_SFT 8 +#define NAU8825_RDAC_CLK_EN (0x3 << NAU8825_RDAC_CLK_EN_SFT) #define NAU8825_RDAC_CLK_DELAY_SFT 4 #define NAU8825_RDAC_CLK_DELAY_MASK (0x7 << NAU8825_RDAC_CLK_DELAY_SFT) #define NAU8825_RDAC_VREF_SFT 2 @@ -355,12 +418,23 @@ enum { NAU8825_CLK_FLL_FS, }; +/* Cross talk detection state */ +enum { + NAU8825_XTALK_PREPARE = 0, + NAU8825_XTALK_HPR_R2L, + NAU8825_XTALK_HPL_R2L, + NAU8825_XTALK_IMM, + NAU8825_XTALK_DONE, +}; + struct nau8825 { struct device *dev; struct regmap *regmap; struct snd_soc_dapm_context *dapm; struct snd_soc_jack *jack; struct clk *mclk; + struct work_struct xtalk_work; + struct semaphore xtalk_sem; int irq; int mclk_freq; /* 0 - mclk is disabled */ int button_pressed; @@ -379,6 +453,12 @@ struct nau8825 { int key_debounce; int jack_insert_debounce; int jack_eject_debounce; + int high_imped; + int xtalk_state; + int xtalk_event; + int xtalk_event_mask; + bool xtalk_protect; + int imp_rms[NAU8825_XTALK_IMM]; }; int nau8825_enable_jack_detect(struct snd_soc_codec *codec, -- 2.6.4

2 1

[alsa-devel] [PATCH 1/4] ASoC: arizona: Add a couple of missing consts
by Charles Keepax 13 Jun '16

13 Jun '16

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 549161b..8d6dbf3 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -630,7 +630,7 @@ const struct soc_enum arizona_asrc_rate1 = arizona_rate_text, arizona_rate_val); EXPORT_SYMBOL_GPL(arizona_asrc_rate1); -static const char *arizona_vol_ramp_text[] = { +static const char * const arizona_vol_ramp_text[] = { "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", "15ms/6dB", "30ms/6dB", }; @@ -659,7 +659,7 @@ SOC_ENUM_SINGLE_DECL(arizona_out_vi_ramp, arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_out_vi_ramp); -static const char *arizona_lhpf_mode_text[] = { +static const char * const arizona_lhpf_mode_text[] = { "Low-pass", "High-pass" }; @@ -687,7 +687,7 @@ SOC_ENUM_SINGLE_DECL(arizona_lhpf4_mode, arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); -static const char *arizona_ng_hold_text[] = { +static const char * const arizona_ng_hold_text[] = { "30ms", "120ms", "250ms", "500ms", }; @@ -1764,7 +1764,7 @@ restore_aif: return ret; } -static const char *arizona_dai_clk_str(int clk_id) +static const char * const arizona_dai_clk_str(int clk_id) { switch (clk_id) { case ARIZONA_CLK_SYSCLK: -- 2.1.4

2 7

[alsa-devel] [PATCH] ASoC: sti: fix return value check in uni_player_parse_dt_audio_glue()
by weiyj_lk＠163.com 13 Jun '16

13 Jun '16

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> In case of error, the function syscon_regmap_lookup_by_phandle() returns ERR_PTR() and never returns NULL. The NULL test in the return value check should be replaced with IS_ERR(). Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/soc/sti/uniperif_player.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index ee1c7c2..1ac2db2 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -1029,9 +1029,9 @@ static int uni_player_parse_dt_audio_glue(struct platform_device *pdev, regmap = syscon_regmap_lookup_by_phandle(node, "st,syscfg"); - if (!regmap) { + if (IS_ERR(regmap)) { dev_err(&pdev->dev, "sti-audio-clk-glue syscf not found\n"); - return -EINVAL; + return PTR_ERR(regmap); } player->clk_sel = regmap_field_alloc(regmap, regfield[0]);

2 1

Re: [alsa-devel] dmix produces garbled sound on ARM
by Ihar Filipau 13 Jun '16

13 Jun '16

On 6/9/16, Takashi Iwai <tiwai(a)suse.de> wrote: > On Thu, 09 Jun 2016 16:05:42 +0200, > Ihar Filipau wrote: >> >> The only thing so far I have found is `aplay --dump-hw-params`, but it >> seem to display the allowed ranges, not the defaults/the params which >> are used. > > You just need to check what "aplay -v ..." shows. For example, > % aplay -Dplug:dmix -v foo.wav I have just retested it with the "aloop", completely without hardware. The garbled sound comes from the software, from the dmix/libasound. The problem is not related to (sound) hardware. I can upload the garbled output which was recorded by "arecord", if it would be helpful in analyzing the problem. (12s, ~2MB wav.) Regards.

2 2

[alsa-devel] [PATCH] ASoC: adau: Factor out shared PLL configuration code
by Lars-Peter Clausen 13 Jun '16

13 Jun '16

Multiple devices from the ADAU family share the same PLL structure and configuration register layout. Introduce a new helper module that can be used to calculated the PLL configuration registers based on a specified input frequency and the desired output frequency of the PLL. The ADAU1761/ADAU1781 and ADAU1373 drivers are updated to make use of this new helper module. But future drivers for additional devices from the ADAU family are also expected to make use of it. In anticipation of sharing more infrastructure code between different devices from the ADAU family the new module is called adau-utils. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/Kconfig | 5 ++++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/adau-utils.c | 61 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adau-utils.h | 7 +++++ sound/soc/codecs/adau1373.c | 38 +++++++-------------------- sound/soc/codecs/adau17x1.c | 38 +++------------------------ 6 files changed, 89 insertions(+), 62 deletions(-) create mode 100644 sound/soc/codecs/adau-utils.c create mode 100644 sound/soc/codecs/adau-utils.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..df56003 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -272,8 +272,12 @@ config SND_SOC_AD1980 config SND_SOC_AD73311 tristate +config SND_SOC_ADAU_UTILS + tristate + config SND_SOC_ADAU1373 tristate + select SND_SOC_ADAU_UTILS config SND_SOC_ADAU1701 tristate "Analog Devices ADAU1701 CODEC" @@ -283,6 +287,7 @@ config SND_SOC_ADAU1701 config SND_SOC_ADAU17X1 tristate select SND_SOC_SIGMADSP_REGMAP + select SND_SOC_ADAU_UTILS config SND_SOC_ADAU1761 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..c0805a6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,7 @@ snd-soc-ad193x-spi-objs := ad193x-spi.o snd-soc-ad193x-i2c-objs := ad193x-i2c.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o +snd-soc-adau-utils-objs := adau-utils.o snd-soc-adau1373-objs := adau1373.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau17x1-objs := adau17x1.o @@ -223,6 +224,7 @@ obj-$(CONFIG_SND_SOC_AD193X_SPI) += snd-soc-ad193x-spi.o obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU_UTILS) += snd-soc-adau-utils.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAU17X1) += snd-soc-adau17x1.o diff --git a/sound/soc/codecs/adau-utils.c b/sound/soc/codecs/adau-utils.c new file mode 100644 index 0000000..19d6a6f --- /dev/null +++ b/sound/soc/codecs/adau-utils.c @@ -0,0 +1,61 @@ +/* + * Shared helper functions for devices from the ADAU family + * + * Copyright 2011-2016 Analog Devices Inc. + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/gcd.h> +#include <linux/kernel.h> +#include <linux/module.h> + +#include "adau-utils.h" + +int adau_calc_pll_cfg(unsigned int freq_in, unsigned int freq_out, + uint8_t regs[5]) +{ + unsigned int r, n, m, i, j; + unsigned int div; + + if (!freq_out) { + r = 0; + n = 0; + m = 0; + div = 0; + } else { + if (freq_out % freq_in != 0) { + div = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= div; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + div--; + } else { + r = freq_out / freq_in; + n = 0; + m = 0; + div = 0; + } + if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2) + return -EINVAL; + } + + regs[0] = m >> 8; + regs[1] = m & 0xff; + regs[2] = n >> 8; + regs[3] = n & 0xff; + regs[4] = (r << 3) | (div << 1); + if (m != 0) + regs[4] |= 1; /* Fractional mode */ + + return 0; +} +EXPORT_SYMBOL_GPL(adau_calc_pll_cfg); + +MODULE_DESCRIPTION("ASoC ADAU audio CODECs shared helper functions"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/adau-utils.h b/sound/soc/codecs/adau-utils.h new file mode 100644 index 0000000..939b5f3 --- /dev/null +++ b/sound/soc/codecs/adau-utils.h @@ -0,0 +1,7 @@ +#ifndef SOUND_SOC_CODECS_ADAU_PLL_H +#define SOUND_SOC_CODECS_ADAU_PLL_H + +int adau_calc_pll_cfg(unsigned int freq_in, unsigned int freq_out, + uint8_t regs[5]); + +#endif diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index fe1353a..1556b36 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -23,6 +23,7 @@ #include <sound/adau1373.h> #include "adau1373.h" +#include "adau-utils.h" struct adau1373_dai { unsigned int clk_src; @@ -1254,7 +1255,8 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, { struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); unsigned int dpll_div = 0; - unsigned int x, r, n, m, i, j, mode; + uint8_t pll_regs[5]; + int ret; switch (pll_id) { case ADAU1373_PLL1: @@ -1295,27 +1297,8 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, dpll_div++; } - if (freq_out % freq_in != 0) { - /* fout = fin * (r + (n/m)) / x */ - x = DIV_ROUND_UP(freq_in, 13500000); - freq_in /= x; - r = freq_out / freq_in; - i = freq_out % freq_in; - j = gcd(i, freq_in); - n = i / j; - m = freq_in / j; - x--; - mode = 1; - } else { - /* fout = fin / r */ - r = freq_out / freq_in; - n = 0; - m = 0; - x = 0; - mode = 0; - } - - if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff) + ret = adau_calc_pll_cfg(freq_in, freq_out, pll_regs); + if (ret) return -EINVAL; if (dpll_div) { @@ -1330,12 +1313,11 @@ static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, regmap_write(adau1373->regmap, ADAU1373_DPLL_CTRL(pll_id), (source << 4) | dpll_div); - regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); - regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); - regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); - regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); - regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), - (r << 3) | (x << 1) | mode); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL1(pll_id), pll_regs[0]); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL2(pll_id), pll_regs[1]); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL3(pll_id), pll_regs[2]); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL4(pll_id), pll_regs[3]); + regmap_write(adau1373->regmap, ADAU1373_PLL_CTRL5(pll_id), pll_regs[4]); /* Set sysclk to pll_rate / 4 */ regmap_update_bits(adau1373->regmap, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); diff --git a/sound/soc/codecs/adau17x1.c b/sound/soc/codecs/adau17x1.c index fcf05b2..66a6e06 100644 --- a/sound/soc/codecs/adau17x1.c +++ b/sound/soc/codecs/adau17x1.c @@ -23,6 +23,7 @@ #include "sigmadsp.h" #include "adau17x1.h" +#include "adau-utils.h" static const char * const adau17x1_capture_mixer_boost_text[] = { "Normal operation", "Boost Level 1", "Boost Level 2", "Boost Level 3", @@ -391,45 +392,14 @@ static int adau17x1_set_dai_pll(struct snd_soc_dai *dai, int pll_id, { struct snd_soc_codec *codec = dai->codec; struct adau *adau = snd_soc_codec_get_drvdata(codec); - unsigned int r, n, m, i, j; - unsigned int div; int ret; if (freq_in < 8000000 || freq_in > 27000000) return -EINVAL; - if (!freq_out) { - r = 0; - n = 0; - m = 0; - div = 0; - } else { - if (freq_out % freq_in != 0) { - div = DIV_ROUND_UP(freq_in, 13500000); - freq_in /= div; - r = freq_out / freq_in; - i = freq_out % freq_in; - j = gcd(i, freq_in); - n = i / j; - m = freq_in / j; - div--; - } else { - r = freq_out / freq_in; - n = 0; - m = 0; - div = 0; - } - if (n > 0xffff || m > 0xffff || div > 3 || r > 8 || r < 2) - return -EINVAL; - } - - adau->pll_regs[0] = m >> 8; - adau->pll_regs[1] = m & 0xff; - adau->pll_regs[2] = n >> 8; - adau->pll_regs[3] = n & 0xff; - adau->pll_regs[4] = (r << 3) | (div << 1); - if (m != 0) - adau->pll_regs[4] |= 1; /* Fractional mode */ + ret = adau_calc_pll_cfg(freq_in, freq_out, adau->pll_regs); + if (ret < 0) + return ret; /* The PLL register is 6 bytes long and can only be written at once. */ ret = regmap_raw_write(adau->regmap, ADAU17X1_PLL_CONTROL, -- 2.1.4

2 4

[alsa-devel] [very-RFC 0/8] TSN driver for the kernel
by Henrik Austad 12 Jun '16

12 Jun '16

Hi all (series based on v4.7-rc2) This is a *very* early RFC for a TSN-driver in the kernel. It has been floating around in my repo for a while and I would appreciate some feedback on the overall design to avoid doing some major blunders. TSN: Time Sensitive Networking, formely known as AVB (Audio/Video Bridging). There are at least one AVB-driver (the AV-part of TSN) in the kernel already, however this driver aims to solve a wider scope as TSN can do much more than just audio. A very basic ALSA-driver is added to the end that allows you to play music between 2 machines using aplay in one end and arecord | aplay on the other (some fiddling required) We have plans for doing the same for v4l2 eventually (but there are other fishes to fry first). The same goes for a TSN_SOCK type approach as well. TSN is all about providing infrastructure. Allthough there are a few very interesting uses for TSN (reliable, deterministic network for audio and video), once you have that reliable link, you can do a lot more. Some notes on the design: The driver is directed via ConfigFS as we need userspace to handle stream-reservation (MSRP), discovery and enumeration (IEEE 1722.1) and whatever other management is needed. Once we have all the required attributes, we can create link using mkdir, and use write() to set the attributes. Once ready, specify the 'shim' (basically a thin wrapper between TSN and another subsystem) and we start pushing out frames. The network part: it ties directly into the rx-handler for receive and writes skb's using netdev_start_xmit(). This could probably be improved. 2 new fields in netdev_ops have been introduced, and the Intel igb-driver has been updated (as this is available as a PCI-e card). The igb-driver works-ish What remains - tie to (g)PTP properly, currently using ktime_get() for presentation time - get time from shim into TSN and vice versa - let shim create/manage buffer Henrik Austad (8): TSN: add documentation TSN: Add the standard formerly known as AVB to the kernel Adding TSN-driver to Intel I210 controller Add TSN header for the driver Add TSN machinery to drive the traffic from a shim over the network Add TSN event-tracing AVB ALSA - Add ALSA shim for TSN MAINTAINERS: add TSN/AVB-entries Documentation/TSN/tsn.txt | 147 +++++ MAINTAINERS | 14 + drivers/media/Kconfig | 15 + drivers/media/Makefile | 3 +- drivers/media/avb/Makefile | 5 + drivers/media/avb/avb_alsa.c | 742 +++++++++++++++++++++++ drivers/media/avb/tsn_iec61883.h | 124 ++++ drivers/net/ethernet/intel/Kconfig | 18 + drivers/net/ethernet/intel/igb/Makefile | 2 +- drivers/net/ethernet/intel/igb/igb.h | 19 + drivers/net/ethernet/intel/igb/igb_main.c | 10 +- drivers/net/ethernet/intel/igb/igb_tsn.c | 396 ++++++++++++ include/linux/netdevice.h | 32 + include/linux/tsn.h | 806 ++++++++++++++++++++++++ include/trace/events/tsn.h | 349 +++++++++++ net/Kconfig | 1 + net/Makefile | 1 + net/tsn/Kconfig | 32 + net/tsn/Makefile | 6 + net/tsn/tsn_configfs.c | 623 +++++++++++++++++++ net/tsn/tsn_core.c | 975 ++++++++++++++++++++++++++++++ net/tsn/tsn_header.c | 203 +++++++ net/tsn/tsn_internal.h | 383 ++++++++++++ net/tsn/tsn_net.c | 403 ++++++++++++ 24 files changed, 5306 insertions(+), 3 deletions(-) create mode 100644 Documentation/TSN/tsn.txt create mode 100644 drivers/media/avb/Makefile create mode 100644 drivers/media/avb/avb_alsa.c create mode 100644 drivers/media/avb/tsn_iec61883.h create mode 100644 drivers/net/ethernet/intel/igb/igb_tsn.c create mode 100644 include/linux/tsn.h create mode 100644 include/trace/events/tsn.h create mode 100644 net/tsn/Kconfig create mode 100644 net/tsn/Makefile create mode 100644 net/tsn/tsn_configfs.c create mode 100644 net/tsn/tsn_core.c create mode 100644 net/tsn/tsn_header.c create mode 100644 net/tsn/tsn_internal.h create mode 100644 net/tsn/tsn_net.c -- 2.7.4 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

3 22

[alsa-devel] [PATCH 1/1] alsabat: fix a missing break in switch
by han.lu＠intel.com 12 Jun '16

12 Jun '16

From: "Lu, Han" <han.lu(a)intel.com> Add the break line for OPT_ROUNDUPLATENCY case. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/bat.c b/bat/bat.c index 72ab8b2..8645770 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -420,6 +420,7 @@ static void parse_arguments(struct bat *bat, int argc, char *argv[]) break; case OPT_ROUNDTRIPLATENCY: bat->roundtriplatency = true; + break; case OPT_SNRTHD_DB: get_snr_thd_db(bat, optarg); break; -- 2.5.0

2 1

[alsa-devel] [asoc:for-next 6/11] cygnus-pcm.c:undefined reference to `bad_dma_ops'
by kbuild test robot 11 Jun '16

11 Jun '16

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: dfc5f51c33fb2dcc27bca3fce73a8bfc215716f3 commit: 1b02d429bfc97f490072f02502ac442c1176e670 [6/11] Merge remote-tracking branches 'asoc/topic/arizona', 'asoc/topic/atmel', 'asoc/topic/cs53l30', 'asoc/topic/cygnus' and 'asoc/topic/davinci' into asoc-next config: m32r-allyesconfig (attached as .config) compiler: m32r-linux-gcc (GCC) 4.9.0 reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 1b02d429bfc97f490072f02502ac442c1176e670 # save the attached .config to linux build tree make.cross ARCH=m32r All errors (new ones prefixed by >>): drivers/built-in.o: In function `zynq_fpga_ops_write': zynq-fpga.c:(.text+0xeff3b4): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff3bc): undefined reference to `bad_dma_ops' zynq-fpga.c:(.text+0xeff4dc): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcb4): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcbc): undefined reference to `bad_dma_ops' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): undefined reference to `dma_common_mmap' sound/built-in.o: In function `snd_pcm_lib_default_mmap': (.text+0xfcd4): relocation truncated to fit: R_M32R_26_PCREL_RELA against undefined symbol `dma_common_mmap' sound/built-in.o: In function `cygnus_pcm_preallocate_dma_buffer': >> cygnus-pcm.c:(.text+0x10d048): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d04c): undefined reference to `bad_dma_ops' sound/built-in.o: In function `cygnus_dma_free_dma_buffers': cygnus-pcm.c:(.text+0x10d204): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d208): undefined reference to `bad_dma_ops' cygnus-pcm.c:(.text+0x10d214): undefined reference to `bad_dma_ops' sound/built-in.o:cygnus-pcm.c:(.text+0x10d218): more undefined references to `bad_dma_ops' follow --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] AMD Carrizo HDMI audio
by João Paulo Rechi Vita 11 Jun '16

11 Jun '16

Hello Alex and Maruthi, I'm trying to get HDMI audio working on a AMD CARRIZO laptop, which I believe need the ACP changes which got merged into Linux 4.6. But testing the v4.6 tag, amd_acp_hw_init() returns -ENODEV, because it succeeds the "if (acp_mode != ACP_MODE_I2S)" test in acp_hw.c. Does this mean this chip does not uses ACP, or is there anything else missing? lspci shows me the following gfx/audio devices (I believe this is a dual-gpu machine): 00:01.0 VGA compatible controller: Advanced Micro Devices [AMD] nee ATI Device 9874 (rev c5) (prog-if 00 [VGA controller]) 00:01.1 Audio device: Advanced Micro Devices [AMD] nee ATI Device 9840 00:09.2 Audio device: Advanced Micro Devices [AMD] Device 157a 03:00.0 Display controller: Advanced Micro Devices [AMD] nee ATI Sun [Radeon HD 8600M Series] (rev ff) Both amdgpu and radeon get automatically probed. Sorry if I'm missing something obvious here, I'm quite new to dual-gpu machines. Thanks and best regards, -- João Paulo Rechi Vita http://about.me/jprvita

3 4

[alsa-devel] [PATCH v5 0/7] ARM/ASoC: OMAP3: Fix McBSP2/3 sidetone support
by Peter Ujfalusi 10 Jun '16

10 Jun '16

Hi, Changes since v4: - Moved DTS patch as the first in the series - rebased on top of 4.7-rc1 Changes since v3: - Rebased on next-20160525 [1] to apply cleanly - Fixed subject line for patch5 - Added Mark's acked-by to the ASoC patches [1] "ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3" is gone missing from linux-next - it was in next-20160517 and I'm not sure when it got dropped. Tony: since Paul did not replied to the series, I'm sending the v4 now, I hope he will have time to look at this. Commit message from v3: Based on the ongoing discussion on v2 (ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone) I have dropped the removal of the sidetone hwmod and only corrected it. The series now includes all related changes needed to have correct sidetone support whenever we boot in legacy or in DT mode. This time the ASoC part have dependency on earlier patches so they can not be applied separately. Regards, Peter --- Peter Ujfalusi (7): ARM: dts: omap3: Add clocks to McBSP nodes ARM: OMAP3: hwmod data: Fix McBSP2/3 sidetone data ARM: OMAP3: McBSP: New callback for McBSP2/3 ICLK idle configuration ARM: OMAP3: pdata-quirks: Add support for McBSP2/3 sidetone handling ASoC: omap-mcbsp: Rename omap_mcbsp_sysfs_remove() to omap_mcbsp_cleanup() ASoC: omap-mcbsp: sidetone: Use the new callback for iclk handling ARM: OMAP2+: McBSP: Remove the old iclk allow/deny idle code arch/arm/boot/dts/omap3.dtsi | 10 ++++++++++ arch/arm/mach-omap2/mcbsp.c | 31 ++++++++++++++++------------- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 24 ++++------------------ arch/arm/mach-omap2/pdata-quirks.c | 18 +++++++++++++++++ include/linux/platform_data/asoc-ti-mcbsp.h | 4 +++- sound/soc/omap/mcbsp.c | 21 +++++++++++++------ sound/soc/omap/mcbsp.h | 3 ++- sound/soc/omap/omap-mcbsp.c | 5 ++++- 8 files changed, 73 insertions(+), 43 deletions(-) -- 2.8.3

2 9

Re: [alsa-devel] dmix produces garbled sound on ARM
by Ihar Filipau 10 Jun '16

10 Jun '16

On 6/9/16, Takashi Iwai <tiwai(a)suse.de> wrote: > On Thu, 09 Jun 2016 16:05:42 +0200, > Ihar Filipau wrote: >> >> The only thing so far I have found is `aplay --dump-hw-params`, but it >> seem to display the allowed ranges, not the defaults/the params which >> are used. > > You just need to check what "aplay -v ..." shows. For example, > % aplay -Dplug:dmix -v foo.wav [ Replying back to the list, since exchange of the lengthy outputs is over for now. ] My parameters are nearly identical to yours (except for the differences which could be attributed to my 32-bit vs your 64-bit systems, and the hardware S16_LE vs S32_LE). Yet the problem persists. I have checked the Raspberry Pi forums/mail list and it seems that "dmix" works fine there. Thus, the problem might be related to the hw driver. How can I localize the problem? Is there any way to intercept the data stream which goes to the sound card, and compare the content between normal direct playback vs dmix? P.S. Though the window is closing for my investigations. I'm starting to prepare a workaround: applications shouldn't keep pcm open all the time, should open pcm when needed, and they should retry with a delay if opening pcm failed.

1 0

[alsa-devel] [PATCH 0/3] ASoC: da7219: Add ACPI initialisation support to driver
by Adam Thomson 10 Jun '16

10 Jun '16

This patch set updates the driver to use generic device property & fwnode related functions to read in either DT and ACPI data for driver initialisation. Changes are based on v4.6-rc6 Linux Kernel. Adam Thomson (3): ASoC: da7219: Convert driver to use generic device/fwnode functions ASoC: da7219: Add ACPI parsing support ASoC: da7219: Add initial ACPI id for device sound/soc/codecs/da7219-aad.c | 96 +++++++++++++++++++++++++++++++------------ sound/soc/codecs/da7219.c | 35 ++++++++++------ 2 files changed, 91 insertions(+), 40 deletions(-) -- 1.9.3

4 13

[alsa-devel] [PATCH v2] drm: msm: Add ASoC generic hdmi audio codec support.
by Srinivas Kandagatla 10 Jun '16

10 Jun '16

This patch adds support to generic audio codec via ASoC hdmi-codec infrastucture which is merged recently. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: -Fixed typo for 44.1Kz case spotted by Jyri Sarha drivers/gpu/drm/msm/Kconfig | 1 + drivers/gpu/drm/msm/hdmi/hdmi.c | 120 +++++++++++++++++++++++++++++++++++++++- drivers/gpu/drm/msm/hdmi/hdmi.h | 14 +++++ 3 files changed, 134 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/msm/Kconfig b/drivers/gpu/drm/msm/Kconfig index 167a497..7c7a031 100644 --- a/drivers/gpu/drm/msm/Kconfig +++ b/drivers/gpu/drm/msm/Kconfig @@ -10,6 +10,7 @@ config DRM_MSM select SHMEM select TMPFS select QCOM_SCM + select SND_SOC_HDMI_CODEC if SND_SOC default y help DRM/KMS driver for MSM/snapdragon. diff --git a/drivers/gpu/drm/msm/hdmi/hdmi.c b/drivers/gpu/drm/msm/hdmi/hdmi.c index 51b9ea5..aada355 100644 --- a/drivers/gpu/drm/msm/hdmi/hdmi.c +++ b/drivers/gpu/drm/msm/hdmi/hdmi.c @@ -19,6 +19,7 @@ #include <linux/of_irq.h> #include <linux/of_gpio.h> +#include <sound/hdmi-codec.h> #include "hdmi.h" void msm_hdmi_set_mode(struct hdmi *hdmi, bool power_on) @@ -434,6 +435,114 @@ static int msm_hdmi_get_gpio(struct device_node *of_node, const char *name) return gpio; } +/* + * HDMI audio codec callbacks + */ +static int msm_hdmi_audio_hw_params(struct device *dev, + struct hdmi_codec_daifmt *daifmt, + struct hdmi_codec_params *params) +{ + struct hdmi *hdmi = dev_get_drvdata(dev); + unsigned int chan; + unsigned int channel_allocation = 0; + unsigned int rate; + unsigned int level_shift = 0; /* 0dB */ + bool down_mix = false; + + dev_dbg(dev, "%u Hz, %d bit, %d channels\n", params->sample_rate, + params->sample_width, params->cea.channels); + + switch (params->cea.channels) { + case 2: + /* FR and FL speakers */ + channel_allocation = 0; + chan = MSM_HDMI_AUDIO_CHANNEL_2; + break; + case 4: + /* FC, LFE, FR and FL speakers */ + channel_allocation = 0x3; + chan = MSM_HDMI_AUDIO_CHANNEL_4; + break; + case 6: + /* RR, RL, FC, LFE, FR and FL speakers */ + channel_allocation = 0x0B; + chan = MSM_HDMI_AUDIO_CHANNEL_6; + break; + case 8: + /* FRC, FLC, RR, RL, FC, LFE, FR and FL speakers */ + channel_allocation = 0x1F; + chan = MSM_HDMI_AUDIO_CHANNEL_8; + break; + default: + return -EINVAL; + } + + switch (params->sample_rate) { + case 32000: + rate = HDMI_SAMPLE_RATE_32KHZ; + break; + case 44100: + rate = HDMI_SAMPLE_RATE_44_1KHZ; + break; + case 48000: + rate = HDMI_SAMPLE_RATE_48KHZ; + break; + case 88200: + rate = HDMI_SAMPLE_RATE_88_2KHZ; + break; + case 96000: + rate = HDMI_SAMPLE_RATE_96KHZ; + break; + case 176400: + rate = HDMI_SAMPLE_RATE_176_4KHZ; + break; + case 192000: + rate = HDMI_SAMPLE_RATE_192KHZ; + break; + default: + dev_err(dev, "rate[%d] not supported!\n", + params->sample_rate); + return -EINVAL; + } + + msm_hdmi_audio_set_sample_rate(hdmi, rate); + msm_hdmi_audio_info_setup(hdmi, 1, chan, channel_allocation, + level_shift, down_mix); + + return 0; +} + +static void msm_hdmi_audio_shutdown(struct device *dev) +{ + struct hdmi *hdmi = dev_get_drvdata(dev); + + msm_hdmi_audio_info_setup(hdmi, 0, 0, 0, 0, 0); +} + +static const struct hdmi_codec_ops msm_hdmi_audio_codec_ops = { + .hw_params = msm_hdmi_audio_hw_params, + .audio_shutdown = msm_hdmi_audio_shutdown, +}; + +static struct hdmi_codec_pdata codec_data = { + .ops = &msm_hdmi_audio_codec_ops, + .max_i2s_channels = 8, + .i2s = 1, +}; + +static int msm_hdmi_register_audio_driver(struct hdmi *hdmi, struct device *dev) +{ + hdmi->audio_pdev = platform_device_register_data(dev, + HDMI_CODEC_DRV_NAME, + PLATFORM_DEVID_AUTO, + &codec_data, + sizeof(codec_data)); + if (IS_ERR(hdmi->audio_pdev)) + return PTR_ERR(hdmi->audio_pdev); + + return 0; +} + static int msm_hdmi_bind(struct device *dev, struct device *master, void *data) { struct drm_device *drm = dev_get_drvdata(master); @@ -441,7 +550,7 @@ static int msm_hdmi_bind(struct device *dev, struct device *master, void *data) static struct hdmi_platform_config *hdmi_cfg; struct hdmi *hdmi; struct device_node *of_node = dev->of_node; - int i; + int i, err; hdmi_cfg = (struct hdmi_platform_config *) of_device_get_match_data(dev); @@ -468,6 +577,12 @@ static int msm_hdmi_bind(struct device *dev, struct device *master, void *data) return PTR_ERR(hdmi); priv->hdmi = hdmi; + err = msm_hdmi_register_audio_driver(hdmi, dev); + if (err) { + DRM_ERROR("Failed to attach an audio codec %d\n", err); + hdmi->audio_pdev = NULL; + } + return 0; } @@ -477,6 +592,9 @@ static void msm_hdmi_unbind(struct device *dev, struct device *master, struct drm_device *drm = dev_get_drvdata(master); struct msm_drm_private *priv = drm->dev_private; if (priv->hdmi) { + if (priv->hdmi->audio_pdev) + platform_device_unregister(priv->hdmi->audio_pdev); + msm_hdmi_destroy(priv->hdmi); priv->hdmi = NULL; } diff --git a/drivers/gpu/drm/msm/hdmi/hdmi.h b/drivers/gpu/drm/msm/hdmi/hdmi.h index bc7ba0b..accc9a6 100644 --- a/drivers/gpu/drm/msm/hdmi/hdmi.h +++ b/drivers/gpu/drm/msm/hdmi/hdmi.h @@ -50,6 +50,7 @@ struct hdmi_hdcp_ctrl; struct hdmi { struct drm_device *dev; struct platform_device *pdev; + struct platform_device *audio_pdev; const struct hdmi_platform_config *config; @@ -210,6 +211,19 @@ static inline int msm_hdmi_pll_8996_init(struct platform_device *pdev) /* * audio: */ +/* Supported HDMI Audio channels and rates */ +#define MSM_HDMI_AUDIO_CHANNEL_2 0 +#define MSM_HDMI_AUDIO_CHANNEL_4 1 +#define MSM_HDMI_AUDIO_CHANNEL_6 2 +#define MSM_HDMI_AUDIO_CHANNEL_8 3 + +#define HDMI_SAMPLE_RATE_32KHZ 0 +#define HDMI_SAMPLE_RATE_44_1KHZ 1 +#define HDMI_SAMPLE_RATE_48KHZ 2 +#define HDMI_SAMPLE_RATE_88_2KHZ 3 +#define HDMI_SAMPLE_RATE_96KHZ 4 +#define HDMI_SAMPLE_RATE_176_4KHZ 5 +#define HDMI_SAMPLE_RATE_192KHZ 6 int msm_hdmi_audio_update(struct hdmi *hdmi); int msm_hdmi_audio_info_setup(struct hdmi *hdmi, bool enabled, -- 2.8.3

1 0

[alsa-devel] [PATCH 0/4] ASoC: sunxi: Add i2s controller support
by Maxime Ripard 10 Jun '16

10 Jun '16

Hi everyone, This is the first version of the I2S support for the controller found in the Allwinner A10 and later SoCs. Playback has been tested with an UDA1380 on an A20-Olinuxino. Capture is not implemented yet, but will come eventually. Let me know what you think, Maxime Emilio López (1): ARM: sun7i: Add mod1 clock nodes Maxime Ripard (3): dt-bindings: Add A10 I2S controller binding documentation ASoC: sunxi: Add Allwinner A10 Digital Audio driver ARM: sun7i: Add DAI nodes .../devicetree/bindings/sound/sun4i-i2s.txt | 33 + arch/arm/boot/dts/sun7i-a20.dtsi | 87 ++- sound/soc/sunxi/Kconfig | 10 + sound/soc/sunxi/Makefile | 2 +- sound/soc/sunxi/sun4i-i2s.c | 714 +++++++++++++++++++++ 5 files changed, 843 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-i2s.txt create mode 100644 sound/soc/sunxi/sun4i-i2s.c -- 2.8.3

4 13

[alsa-devel] Raspberry Pi sound card
by Matt Flax 10 Jun '16

10 Jun '16

If anyone is tinkering with the Raspberry Pi and wants a neat sound card, check out the Audio Injector which uses the previous patches to run the wm8731 codec with a 12 MHz crystal in normal mode : https://www.kickstarter.com/projects/1250664710/audio-injector-sound-card-f… http://www.audioinjector.net/ thanks Matt

1 0

[alsa-devel] [PATCH v2] ASoC: rsnd: adg :: AUDIO-CLKOUTn asynchronizes support
by Kuninori Morimoto 09 Jun '16

09 Jun '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> AUDIO-CLKOUTn can asynchronizes with L/R clock. AUDIO-CLKOUTn synchronizes with L/R clock is now default behavior. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - default is synchronizes, asynchronizes is option .../devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ sound/soc/sh/rcar/adg.c | 18 ++++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index c7b29df..15a7316 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -373,6 +373,8 @@ Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout it must be 1 if your system has audio_clkout0/1/2/3 - clock-frequency : for all audio_clkout0/1/2/3 +- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn + is asynchronizes with lr-clock. SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 49354d1..7d3e0e4 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -33,11 +33,15 @@ struct rsnd_adg { struct clk *clkout[CLKOUTMAX]; struct clk_onecell_data onecell; struct rsnd_mod mod; + u32 flags; int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ }; +#define LRCLK_ASYNC (1 << 0) +#define adg_mode_flags(adg) (adg->flags) + #define for_each_rsnd_clk(pos, adg, i) \ for (i = 0; \ (i < CLKMAX) && \ @@ -355,6 +359,16 @@ found_clock: rsnd_adg_set_ssi_clk(ssi_mod, data); + if (!(adg_mode_flags(adg) & LRCLK_ASYNC)) { + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + u32 ckr = 0; + + if (0 == (rate % 8000)) + ckr = 0x80000000; + + rsnd_mod_bset(adg_mod, SSICKR, 0x80000000, ckr); + } + dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n", rsnd_mod_name(ssi_mod), rsnd_mod_id(ssi_mod), data, rate); @@ -532,6 +546,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *np = dev->of_node; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -545,6 +560,9 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); + if (of_get_property(np, "clkout-lr-asynchronous", NULL)) + adg->flags = LRCLK_ASYNC; + priv->adg = adg; return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoc: wm8731: add 32bit mode and normal mode with 12MHz XTAL
by Matt Flax 09 Jun '16

09 Jun '16

This patch adds the 32 bit capability and updates the WM8731_IFACE as required. This patch also allows the machine to set normal mode when using a 12 MHz crystal. In USB_MODE the 12 MHz crystal signal is output on the BCLK pin. In NORMAL_MODE the bit clock is output on the BCLK pin. The previously ignored direction variable input to the wm8731_set_dai_sysclk function is used to indicate whether crystal or bit clock is output on the BCLK codec pin. I have ensured that this does not effect the db1200.c machine driver (the only other driver to use a 12 MHz crystal in USB_MODE). It also does not effect the other machine drivers which use the wm8731 codec as they don't use USB_MODE nor 12 MHz crystals. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/wm8731.c | 21 +++++++++++++++++---- 1 file changed, 17 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4bcf5f8..b155cf6 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -5,6 +5,7 @@ * Copyright 2006-12 Wolfson Microelectronics, plc * * Author: Richard Purdie <richard(a)openedhand.com> + * Author: Matt Flax <flatmax(a)flatmax.com> 12Mhz XTAL normal mode and 32bit mode. * * Based on wm8753.c by Liam Girdwood * @@ -53,6 +54,7 @@ struct wm8731_priv { int sysclk_type; int playback_fs; bool deemph; + bool bclk_12_mhz; struct mutex lock; }; @@ -192,7 +194,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_insel_enum); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { -SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ACTIVE", WM8731_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], @@ -341,8 +343,11 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); u16 iface = snd_soc_read(codec, WM8731_IFACE) & 0xfff3; int i = get_coeff(wm8731->sysclk, params_rate(params)); - u16 srate = (coeff_div[i].sr << 2) | - (coeff_div[i].bosr << 1) | coeff_div[i].usb; + u16 srate = (coeff_div[i].sr << 2) | (coeff_div[i].bosr << 1); + + /* this determines whether to output bit or crystal clk */ + if (wm8731->bclk_12_mhz) + srate |= coeff_div[i].usb; wm8731->playback_fs = params_rate(params); @@ -358,6 +363,9 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, case 24: iface |= 0x0008; break; + case 32: + iface |= 0x000c; + break; } wm8731_set_deemph(codec); @@ -417,6 +425,11 @@ static int wm8731_set_dai_sysclk(struct snd_soc_dai *codec_dai, wm8731->sysclk = freq; + if (dir == SND_SOC_CLOCK_IN) /* output the 12 MHz clock */ + wm8731->bclk_12_mhz = 1; + else /* SND_SOC_CLOCK_OUT, output the bit clock */ + wm8731->bclk_12_mhz = 0; + snd_soc_dapm_sync(dapm); return 0; @@ -541,7 +554,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream, #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { .startup = wm8731_startup, -- 2.5.0

3 2

[alsa-devel] dmix produces garbled sound on ARM
by Ihar Filipau 09 Jun '16

09 Jun '16

Hi All! When using the dmix on ARM, the sound comes out garbled, with as if overlapping, increasing-in-number echoes. That happens even with a single aplay instance. In a forum discussion, it was suggested that ARM lacks optimized dmix implementation. But that implies that the generic dmix code is broken, since it is what is used on ARM/etc, but not on i386/amd64. (The forum discussion about my particular issue: http://unix.stackexchange.com/questions/288451/sound-corruption-echo-when-t… ) Does anybody has any suggestions how to fix/avoid the dmix issue on ARM? Thanks. P.S. Versions: Linux v3.18.24, alsa-lib/-utils v1.0.29. The hardware: FreeScale i.MX35 (ARM1136) with SGTL5000 sound chip (used in 48KHz mode). P.P.S. The generic code has outdated functions for the "concurrent" implementation with cmpxchg. The code probably could be updated, and revised to use the C11's stdatomic.h when available.

2 3

[alsa-devel] [PATCH] ALSA: hda - fix whitespaces
by Vinod Koul 09 Jun '16

09 Jun '16

Some trailing whitespaces seems to have crept in the hda_intel.c, so fix them up Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/pci/hda/hda_intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 94089fc71884..5520082ad770 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -383,8 +383,8 @@ static char *driver_short_names[] = { [AZX_DRIVER_SIS] = "HDA SIS966", [AZX_DRIVER_ULI] = "HDA ULI M5461", [AZX_DRIVER_NVIDIA] = "HDA NVidia", - [AZX_DRIVER_TERA] = "HDA Teradici", - [AZX_DRIVER_CTX] = "HDA Creative", + [AZX_DRIVER_TERA] = "HDA Teradici", + [AZX_DRIVER_CTX] = "HDA Creative", [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", -- 1.9.1

2 2

[alsa-devel] [PATCH] conf/ucm: ROCKCHIP-I2S: rename to VEYRON-I2S
by Heiko Stübner 09 Jun '16

09 Jun '16

Commit a192f52fc63a introduced an ucm profile for Rockchip Veyron- Chromebooks by taking the ucm profile from the ChromeOS userspace. But similarly to DAISY-I2S, PAZ00 and most other profiles, the audio setup is pretty specific to a board type, so hogging the Rockchip name will make it harder for future Rockchip based boards to fit in nicely. And while Veyron also is a family of boards, all of them share the same audio setup. The ucm profile was not released with any official alsa release and the audio setup also isn't in the mainline kernel yet, so such a rename should be easily possible. Fixes: a192f52fc63a ("conf/ucm: ROCKCHIP-I2S: add Rockchip I2S UCM config.") Signed-off-by: Heiko Stuebner <heiko(a)sntech.de> --- src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/ROCKCHIP-I2S/Makefile.am | 4 ---- src/conf/ucm/{ROCKCHIP-I2S => VEYRON-I2S}/HiFi.conf | 14 +++++++------- src/conf/ucm/VEYRON-I2S/Makefile.am | 4 ++++ .../ROCKCHIP-I2S.conf => VEYRON-I2S/VEYRON-I2S.conf} | 2 +- 5 files changed, 13 insertions(+), 13 deletions(-) delete mode 100644 src/conf/ucm/ROCKCHIP-I2S/Makefile.am rename src/conf/ucm/{ROCKCHIP-I2S => VEYRON-I2S}/HiFi.conf (91%) create mode 100644 src/conf/ucm/VEYRON-I2S/Makefile.am rename src/conf/ucm/{ROCKCHIP-I2S/ROCKCHIP-I2S.conf => VEYRON-I2S/VEYRON-I2S.conf} (67%) diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index d1d51e5..3cce36a 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -1 +1 @@ -SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 ROCKCHIP-I2S +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 VEYRON-I2S diff --git a/src/conf/ucm/ROCKCHIP-I2S/Makefile.am b/src/conf/ucm/ROCKCHIP-I2S/Makefile.am deleted file mode 100644 index 485a740..0000000 --- a/src/conf/ucm/ROCKCHIP-I2S/Makefile.am +++ /dev/null @@ -1,4 +0,0 @@ -alsaconfigdir = @ALSA_CONFIG_DIR@ -ucmdir = $(alsaconfigdir)/ucm/ROCKCHIP-I2S -ucm_DATA = ROCKCHIP-I2S.conf HiFi.conf -EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf b/src/conf/ucm/VEYRON-I2S/HiFi.conf similarity index 91% rename from src/conf/ucm/ROCKCHIP-I2S/HiFi.conf rename to src/conf/ucm/VEYRON-I2S/HiFi.conf index 7bfe995..292b8fc 100644 --- a/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf +++ b/src/conf/ucm/VEYRON-I2S/HiFi.conf @@ -5,7 +5,7 @@ SectionVerb { } EnableSequence [ - cdev "hw:ROCKCHIPI2S" + cdev "hw:VEYRONI2S" cset "name='Left Speaker Mixer Left DAC Switch' on" cset "name='Right Speaker Mixer Right DAC Switch' on" @@ -49,19 +49,19 @@ SectionVerb { SectionDevice."Headphone".0 { Value { - JackName "ROCKCHIP-I2S Headset Jack" + JackName "VEYRON-I2S Headset Jack" OutputDspName "" } EnableSequence [ - cdev "hw:ROCKCHIPI2S" + cdev "hw:VEYRONI2S" cset "name='Speaker Switch' off" cset "name='Headphone Left Switch' on" cset "name='Headphone Right Switch' on" ] DisableSequence [ - cdev "hw:ROCKCHIPI2S" + cdev "hw:VEYRONI2S" cset "name='Headphone Left Switch' off" cset "name='Headphone Right Switch' off" @@ -71,11 +71,11 @@ SectionDevice."Headphone".0 { SectionDevice."Mic".0 { Value { - JackName "ROCKCHIP-I2S Headset Jack" + JackName "VEYRON-I2S Headset Jack" } EnableSequence [ - cdev "hw:ROCKCHIPI2S" + cdev "hw:VEYRONI2S" cset "name='Int Mic Switch' off" cset "name='DMIC Mux' ADC" @@ -84,7 +84,7 @@ SectionDevice."Mic".0 { ] DisableSequence [ - cdev "hw:ROCKCHIPI2S" + cdev "hw:VEYRONI2S" cset "name='Headset Mic Switch' off" cset "name='DMIC Mux' DMIC" diff --git a/src/conf/ucm/VEYRON-I2S/Makefile.am b/src/conf/ucm/VEYRON-I2S/Makefile.am new file mode 100644 index 0000000..2bb628e --- /dev/null +++ b/src/conf/ucm/VEYRON-I2S/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/VEYRON-I2S +ucm_DATA = VEYRON-I2S.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf b/src/conf/ucm/VEYRON-I2S/VEYRON-I2S.conf similarity index 67% rename from src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf rename to src/conf/ucm/VEYRON-I2S/VEYRON-I2S.conf index 0c399b0..2a92b6f 100644 --- a/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf +++ b/src/conf/ucm/VEYRON-I2S/VEYRON-I2S.conf @@ -1,4 +1,4 @@ -Comment "Rockchip card" +Comment "Rockchip Veyron card" SectionUseCase."HiFi" { File "HiFi.conf" -- 2.8.0.rc3

3 2

[alsa-devel] [PATCH] ALSA: hda - Add PCI ID for Kabylake
by Vinod Koul 09 Jun '16

09 Jun '16

Kabylake shows up as PCI ID 0xa171. And Kabylake-LP as 0x9d71. Since these are similar to Skylake add these to SKL_PLUS macro Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/hda_intel.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9a0d1445ca5c..94089fc71884 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -365,8 +365,11 @@ enum { #define IS_SKL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa170) #define IS_SKL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d70) +#define IS_KBL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa171) +#define IS_KBL_LP(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9d71) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) -#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) +#define IS_SKL_PLUS(pci) (IS_SKL(pci) || IS_SKL_LP(pci) || IS_BXT(pci)) || \ + IS_KBL(pci) || IS_KBL_LP(pci) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -2181,6 +2184,12 @@ static const struct pci_device_id azx_ids[] = { /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Kabylake */ + { PCI_DEVICE(0x8086, 0xa171), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, + /* Kabylake-LP */ + { PCI_DEVICE(0x8086, 0x9d71), + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE }, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BROXTON }, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: adg :: AUDIO-CLKOUTn can synchronizes with L/R clock.
by Kuninori Morimoto 09 Jun '16

09 Jun '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> AUDIO-CLKOUTn can synchronizes with L/R clock, and Salvator board needs it. Otherwise, specific frequency sound will be noisy. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- .../devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ sound/soc/sh/rcar/adg.c | 18 ++++++++++++++++++ 2 files changed, 20 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index c7b29df..065d9c8 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -373,6 +373,8 @@ Optional properties: - #clock-cells : it must be 0 if your system has audio_clkout it must be 1 if your system has audio_clkout0/1/2/3 - clock-frequency : for all audio_clkout0/1/2/3 +- clkout-lr-synchronous : boolean property. it indicates that audio_clkoutn + synchronizes with lr-clock. SSI subnode properties: - interrupts : Should contain SSI interrupt for PIO transfer diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index c4c51a4..1586271 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -33,11 +33,15 @@ struct rsnd_adg { struct clk *clkout[CLKOUTMAX]; struct clk_onecell_data onecell; struct rsnd_mod mod; + u32 flags; int rbga_rate_for_441khz; /* RBGA */ int rbgb_rate_for_48khz; /* RBGB */ }; +#define LRCLK_SYNC (1 << 0) +#define adg_mode_flags(adg) (adg->flags) + #define for_each_rsnd_clk(pos, adg, i) \ for (i = 0; \ (i < CLKMAX) && \ @@ -355,6 +359,16 @@ found_clock: rsnd_adg_set_ssi_clk(ssi_mod, data); + if (adg_mode_flags(adg) & LRCLK_SYNC) { + struct rsnd_mod *adg_mod = rsnd_mod_get(adg); + u32 ckr = 0; + + if (0 == (rate % 8000)) + ckr = 0x80000000; + + rsnd_mod_bset(adg_mod, SSICKR, 0x80000000, ckr); + } + dev_dbg(dev, "ADG: %s[%d] selects 0x%x for %d\n", rsnd_mod_name(ssi_mod), rsnd_mod_id(ssi_mod), data, rate); @@ -532,6 +546,7 @@ int rsnd_adg_probe(struct rsnd_priv *priv) { struct rsnd_adg *adg; struct device *dev = rsnd_priv_to_dev(priv); + struct device_node *np = dev->of_node; adg = devm_kzalloc(dev, sizeof(*adg), GFP_KERNEL); if (!adg) { @@ -545,6 +560,9 @@ int rsnd_adg_probe(struct rsnd_priv *priv) rsnd_adg_get_clkin(priv, adg); rsnd_adg_get_clkout(priv, adg); + if (of_get_property(np, "clkout-lr-synchronous", NULL)) + adg->flags = LRCLK_SYNC; + priv->adg = adg; return 0; -- 1.9.1

2 4

[alsa-devel] [PATCH v2] ASoC: cs53l30: Correct clock inversion check
by Nicolin Chen 09 Jun '16

09 Jun '16

SND_SOC_DAIFMT_IB_NF = 0x3 (11b) | SND_SOC_DAIFMT_IB_IF = 0x4 (100b) creates a mask 0x7 (111b) which also includes SND_SOC_DAIFMT_NB_IF = 0x2 (10b). So this patch uses the traditional way to check the clock inversion. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog: v1->v2: * Correct one typo in the commit log sound/soc/codecs/cs53l30.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ac90dd7..62daa0c 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -598,8 +598,14 @@ static int cs53l30_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) } /* Check to see if the SCLK is inverted */ - if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + case SND_SOC_DAIFMT_IB_IF: aspcfg ^= CS53L30_ASP_SCLK_INV; + break; + default: + break; + } regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL, CS53L30_ASP_MS | CS53L30_ASP_SCLK_INV, aspcfg); -- 2.1.4

1 0

[alsa-devel] [PATCH 0/4] alsabat: add noise detection
by han.lu＠intel.com 08 Jun '16

08 Jun '16

From: "Lu, Han" <han.lu(a)intel.com> Add noise detection for alsabat. Alsabat reports error when noise above threshold be detected. Use either of the options below to designate the threshold. (e.g. if the ratio of noise to signal is 5%, the snr is about 26dB.) --snr-db <value in dB> --snr-pc <value in %> The noise detection is performed in time domain. On each period of the sine wave being analyzed, alsabat substracts a clean sine wave from the source, calculates the RMS value of the residual, and compares the result with the threshold. At last, alsabat returns the number of periods with noise above threshold. 0 is returned when the source is clean. patch content: 1. align the data type and process on float 2. add a parameter for adjust_waveform() 3. add a single channel sine wave generator 4. add noise detection Lu, Han (4): alsabat: align the data type on float alsabat: add channels parameter for adjust_waveform() alsabat: add a single channel sine wave generator alsabat: add noise detection bat/alsabat-test.sh | 4 + bat/alsabat.1 | 8 ++ bat/analyze.c | 216 +++++++++++++++++++++++++++++++++++++++++++++++----- bat/bat-signal.h | 1 + bat/bat.c | 62 ++++++++++++--- bat/common.h | 36 ++++++++- bat/convert.c | 8 +- bat/convert.h | 8 +- bat/signal.c | 27 ++++++- configure.ac | 2 +- 10 files changed, 327 insertions(+), 45 deletions(-) -- 2.5.0

2 5

[alsa-devel] [PATCH] alsabat: add round trip audio latency test
by vivian.zhang＠intel.com 08 Jun '16

08 Jun '16

From: "vivian,zhang" <vivian.zhang(a)intel.com> Audio latency is the time delay as an audio signal passes through a system. There are many kinds of audio latency metrics. One useful metric is the round trip latency, which is the sum of output latency and input latency. The measurement step works like below: 1. Listen and measure the average loudness of the environment for one second; 2. Create a threshold value 16 decibels higher than the average loudness; 3. Begin playing a ~1000 Hz sine wave and start counting the samples elapsed; 4. Stop counting and playing if the input's loudness is higher than the threshold, as the output wave is probably coming back; 5. Calculate the audio latency value in milliseconds. Signed-off-by: Zhang Vivian <vivian.zhang(a)intel.com> --- bat/Makefile.am | 2 + bat/alsa.c | 144 +++++++++++++++++++++++++++++- bat/alsabat-test.sh | 2 + bat/alsabat.1 | 6 ++ bat/bat.c | 36 ++++++++ bat/common.h | 29 ++++++ bat/latencytest.c | 242 +++++++++++++++++++++++++++++++++++++++++++++++++++ bat/latencytest.h | 17 ++++ bat/tinyalsa.c | 95 +++++++++++++++++++- 9 files changed, 568 insertions(+), 5 deletions(-) create mode 100644 bat/latencytest.c create mode 100644 bat/latencytest.h diff --git a/bat/Makefile.am b/bat/Makefile.am index 24e3eb8..6883826 100644 --- a/bat/Makefile.am +++ b/bat/Makefile.am @@ -7,11 +7,13 @@ alsabat_SOURCES = \ bat.c \ common.c \ signal.c \ + latencytest.c \ convert.c noinst_HEADERS = \ common.h \ bat-signal.h \ + latencytest.h \ convert.h if HAVE_LIBFFTW3 diff --git a/bat/alsa.c b/bat/alsa.c index 0a37714..cef1734 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -27,6 +27,7 @@ #include "common.h" #include "alsa.h" +#include "latencytest.h" struct pcm_container { snd_pcm_t *handle; @@ -148,7 +149,33 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) } if (bat->buffer_size > 0 && bat->period_size == 0) - bat->period_size = bat->buffer_size / DIV_BUFFERTIME; + bat->period_size = bat->buffer_size / DIV_BUFFERSIZE; + + if (bat->roundtriplatency && bat->buffer_size == 0) { + /* Set to minimum buffer size and period size + for latency test */ + if (snd_pcm_hw_params_get_buffer_size_min(params, + &buffer_size) < 0) { + fprintf(bat->err, + _("Get parameter from device error: ")); + fprintf(bat->err, _("buffer size min: %d %s: %s(%d)\n"), + (int) buffer_size, + device_name, snd_strerror(err), err); + return -EINVAL; + } + + if (snd_pcm_hw_params_get_period_size_min(params, + &period_size, 0) < 0) { + fprintf(bat->err, + _("Get parameter from device error: ")); + fprintf(bat->err, _("period size min: %d %s: %s(%d)\n"), + (int) period_size, + device_name, snd_strerror(err), err); + return -EINVAL; + } + bat->buffer_size = (int) buffer_size; + bat->period_size = (int) period_size; + } if (bat->buffer_size > 0) { buffer_size = bat->buffer_size; @@ -289,6 +316,8 @@ static int write_to_pcm(const struct pcm_container *sndpcm, } else if (err == -EPIPE) { fprintf(bat->err, _("Underrun: %s(%d)\n"), snd_strerror(err), err); + if (bat->roundtriplatency) + bat->latency.xrun_error = true; snd_pcm_prepare(sndpcm->handle); } else if (err < 0) { fprintf(bat->err, _("Write PCM device error: %s(%d)\n"), @@ -305,6 +334,43 @@ static int write_to_pcm(const struct pcm_container *sndpcm, return 0; } +/** + * Process output data for latency test + */ +static int latencytest_process_output(struct pcm_container *sndpcm, + struct bat *bat) +{ + int err = 0; + int bytes = sndpcm->period_bytes; /* playback buffer size */ + int frames = sndpcm->period_size; /* frame count */ + + bat->latency.is_playing = true; + + while (1) { + /* generate output data */ + err = handleoutput(bat, sndpcm->buffer, bytes, frames); + if (err != 0) + break; + + err = write_to_pcm(sndpcm, frames, bat); + if (err != 0) + break; + + /* Xrun error, terminate the playback thread*/ + if (bat->latency.xrun_error == true) + break; + + if (bat->latency.state == LATENCY_STATE_COMPLETE_SUCCESS) + break; + + bat->periods_played++; + } + + bat->latency.is_playing = false; + + return err; +} + static int write_to_pcm_loop(struct pcm_container *sndpcm, struct bat *bat) { int err = 0; @@ -414,7 +480,10 @@ void *playback_alsa(struct bat *bat) } } - err = write_to_pcm_loop(&sndpcm, bat); + if (bat->roundtriplatency) + err = latencytest_process_output(&sndpcm, bat); + else + err = write_to_pcm_loop(&sndpcm, bat); if (err < 0) { retval_play = err; goto exit4; @@ -447,6 +516,8 @@ static int read_from_pcm(struct pcm_container *sndpcm, snd_pcm_prepare(sndpcm->handle); fprintf(bat->err, _("Overrun: %s(%d)\n"), snd_strerror(err), err); + if (bat->roundtriplatency) + bat->latency.xrun_error = true; } else if (err < 0) { fprintf(bat->err, _("Read PCM device error: %s(%d)\n"), snd_strerror(err), err); @@ -517,6 +588,70 @@ static int read_from_pcm_loop(struct pcm_container *sndpcm, struct bat *bat) return err; } +/** + * Process input data for latency test + */ +static int latencytest_process_input(struct pcm_container *sndpcm, + struct bat *bat) +{ + int err = 0; + FILE *fp = NULL; + int bytes_read = 0; + int frames = sndpcm->period_size; + int size = sndpcm->period_bytes; + int bytes_count = bat->frames * bat->frame_size; + + remove(bat->capture.file); + fp = fopen(bat->capture.file, "wb"); + err = -errno; + if (fp == NULL) { + fprintf(bat->err, _("Cannot open file: %s %d\n"), + bat->capture.file, err); + return err; + } + /* leave space for file header */ + if (fseek(fp, sizeof(struct wav_container), SEEK_SET) != 0) { + fclose(fp); + return err; + } + + bat->latency.is_capturing = true; + + while (bytes_read < bytes_count) { + /* read a chunk from pcm device */ + err = read_from_pcm(sndpcm, frames, bat); + if (err != 0) + break; + + /* Xrun error, terminate the capture thread*/ + if (bat->latency.xrun_error == true) + break; + + err = handleinput(bat, sndpcm->buffer, frames); + if (err != 0) + break; + + if (bat->latency.is_playing == false) + break; + + /* write the chunk to file */ + if (fwrite(sndpcm->buffer, 1, size, fp) != size) { + err = -EIO; + break; + } + + bytes_read += size; + } + + bat->latency.is_capturing = false; + + update_wav_header(bat, fp, bytes_read); + + fclose(fp); + return err; +} + + static void pcm_cleanup(void *p) { snd_pcm_close(p); @@ -558,7 +693,10 @@ void *record_alsa(struct bat *bat) pthread_cleanup_push(free, sndpcm.buffer); fprintf(bat->log, _("Recording ...\n")); - err = read_from_pcm_loop(&sndpcm, bat); + if (bat->roundtriplatency) + err = latencytest_process_input(&sndpcm, bat); + else + err = read_from_pcm_loop(&sndpcm, bat); if (err != 0) { retval_record = err; goto exit3; diff --git a/bat/alsabat-test.sh b/bat/alsabat-test.sh index 2d31a77..24caa0b 100755 --- a/bat/alsabat-test.sh +++ b/bat/alsabat-test.sh @@ -81,6 +81,8 @@ feature_list_test () { latestfile=`ls -t1 /tmp/bat.wav.* | head -n 1` feature_test "--local -F $maxfreq --file $latestfile" \ "local mode: analyze local file" + feature_test "--roundtriplatency" \ + "round trip latency test" print_result } diff --git a/bat/alsabat.1 b/bat/alsabat.1 index 3f9b767..231b154 100644 --- a/bat/alsabat.1 +++ b/bat/alsabat.1 @@ -132,6 +132,12 @@ just like in normal mode, but will not be analyzed. The ALSABAT being built without libfftw3 support is always in standalone mode. The ALSABAT in normal mode can also bypass data analysis using option "--standalone". +.TP +\fI\-\-roundtriplatency\fP +Round trip latency test. +Audio latency is the time delay as an audio signal passes through a system. +There are many kinds of audio latency metrics. One useful metric is the +round trip latency, which is the sum of output latency and input latency. .SH EXAMPLES diff --git a/bat/bat.c b/bat/bat.c index cd4ea2d..9f4cc59 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -40,6 +40,7 @@ #ifdef HAVE_LIBFFTW3 #include "analyze.h" #endif +#include "latencytest.h" static int get_duration(struct bat *bat) { @@ -299,6 +300,7 @@ _("Usage: alsabat [-options]...\n" " --saveplay=# file that storing playback content, for debug\n" " --local internal loop, set to bypass pcm hardware devices\n" " --standalone standalone mode, to bypass analysis\n" +" --roundtriplatency round trip latency mode\n" )); fprintf(bat->log, _("Recognized sample formats are: ")); fprintf(bat->log, _("U8 S16_LE S24_3LE S32_LE\n")); @@ -328,6 +330,7 @@ static void set_defaults(struct bat *bat) bat->local = false; bat->buffer_size = 0; bat->period_size = 0; + bat->roundtriplatency = false; #ifdef HAVE_LIBTINYALSA bat->channels = 2; bat->playback.fct = &playback_tinyalsa; @@ -355,6 +358,7 @@ static void parse_arguments(struct bat *bat, int argc, char *argv[]) {"saveplay", 1, 0, OPT_SAVEPLAY}, {"local", 0, 0, OPT_LOCAL}, {"standalone", 0, 0, OPT_STANDALONE}, + {"roundtriplatency", 0, 0, OPT_ROUNDTRIPLATENCY}, {0, 0, 0, 0} }; @@ -376,6 +380,9 @@ static void parse_arguments(struct bat *bat, int argc, char *argv[]) case OPT_STANDALONE: bat->standalone = true; break; + case OPT_ROUNDTRIPLATENCY: + bat->roundtriplatency = true; + break; case 'D': if (bat->playback.device == NULL) bat->playback.device = optarg; @@ -616,6 +623,35 @@ int main(int argc, char *argv[]) if (err < 0) goto out; + /* round trip latency test thread */ + if (bat.roundtriplatency) { + while (1) { + fprintf(bat.log, + _("\nStart round trip latency\n")); + roundtrip_latency_init(&bat); + test_loopback(&bat); + + if (bat.latency.xrun_error == false) + break; + else { + /* Xrun error in playback or capture, + increase period size and try again */ + bat.period_size += bat.rate / 1000; + bat.buffer_size = + bat.period_size * DIV_BUFFERSIZE; + + /* terminate the test if period_size is + large enough */ + if (bat.period_size > bat.rate * 0.2) + break; + } + + /* Waiting 500ms and start the next round */ + usleep(CAPTURE_DELAY * 1000); + } + goto out; + } + /* single line playback thread: playback only, no capture */ if (bat.playback.mode == MODE_SINGLE) { test_playback(&bat); diff --git a/bat/common.h b/bat/common.h index ad02a5a..0e8f042 100644 --- a/bat/common.h +++ b/bat/common.h @@ -22,6 +22,7 @@ #define OPT_SAVEPLAY (OPT_BASE + 3) #define OPT_LOCAL (OPT_BASE + 4) #define OPT_STANDALONE (OPT_BASE + 5) +#define OPT_ROUNDTRIPLATENCY (OPT_BASE + 6) #define COMPOSE(a, b, c, d) ((a) | ((b)<<8) | ((c)<<16) | ((d)<<24)) #define WAV_RIFF COMPOSE('R', 'I', 'F', 'F') @@ -53,6 +54,10 @@ /* default period size for tinyalsa */ #define TINYALSA_PERIODSIZE 1024 +#define LATENCY_TEST_NUMBER 5 +#define LATENCY_TEST_TIME_LIMIT 25 +#define DIV_BUFFERSIZE 2 + #define EBATBASE 1000 #define ENOPEAK (EBATBASE + 1) #define EONLYDC (EBATBASE + 2) @@ -130,6 +135,14 @@ enum _bat_op_mode { MODE_LAST }; +enum latency_state { + LATENCY_STATE_COMPLETE_FAILURE = -1, + LATENCY_STATE_COMPLETE_SUCCESS = 0, + LATENCY_STATE_MEASURE_FOR_1_SECOND, + LATENCY_STATE_PLAY_AND_LISTEN, + LATENCY_STATE_WAITING, +}; + struct pcm { unsigned int card_tiny; unsigned int device_tiny; @@ -151,6 +164,20 @@ struct sin_generator { float magnitude; }; +struct roundtrip_latency { + int number; + enum latency_state state; + float result[LATENCY_TEST_NUMBER]; + int final_result; + int samples; + float sum; + int threshold; + int error; + bool is_capturing; + bool is_playing; + bool xrun_error; +}; + struct bat { unsigned int rate; /* sampling rate */ int channels; /* nb of channels */ @@ -169,9 +196,11 @@ struct bat { char *logarg; /* path name of log file */ char *debugplay; /* path name to store playback signal */ bool standalone; /* enable to bypass analysis */ + bool roundtriplatency; /* enable round trip latency */ struct pcm playback; struct pcm capture; + struct roundtrip_latency latency; unsigned int periods_played; unsigned int periods_total; diff --git a/bat/latencytest.c b/bat/latencytest.c new file mode 100644 index 0000000..fae191c --- /dev/null +++ b/bat/latencytest.c @@ -0,0 +1,242 @@ +/* + * Copyright (C) 2013-2015 Intel Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ + +#include <math.h> +#include <stdio.h> +#include <stdlib.h> +#include <string.h> +#include <stdbool.h> + +#include "common.h" +#include "bat-signal.h" +#include "gettext.h" + +/* How one measurement step works: + - Listen and measure the average loudness of the environment for 1 second. + - Create a threshold value 16 decibels higher than the average loudness. + - Begin playing a ~1000 Hz sine wave and start counting the samples elapsed. + - Stop counting and playing if the input's loudness is higher than the + threshold, as the output wave is probably coming back. + - Calculate the round trip audio latency value in milliseconds. */ + +static float sumaudio(struct bat *bat, short int *buffer, int frames) +{ + float sum = 0; + int n = 0; + + while (frames) { + frames--; + + for (n = 0; n < bat->channels; n++) { + sum += abs(buffer[0]); + buffer++; + } + } + + sum = sum / bat->channels; + + return sum; +} + +static void play_and_listen(struct bat *bat, void *buffer, int frames) +{ + int averageinput; + int n = 0; + float sum = 0; + float max = 0; + float min = 100000.0f; + short int *input; + int num = bat->latency.number; + + averageinput = (int) (sumaudio(bat, buffer, frames) / frames); + + /* The signal is above threshold + So our sine wave comes back on the input */ + if (averageinput > bat->latency.threshold) { + input = buffer; + + /* Check the location when it became loud enough */ + while (n < frames) { + if (*input++ > bat->latency.threshold) + break; + *input += bat->channels; + n++; + } + + /* Now we get the total round trip latency*/ + bat->latency.samples += n; + + /* Expect at least 1 buffer of round trip latency. */ + if (bat->latency.samples > frames) { + bat->latency.result[num - 1] = + (float) bat->latency.samples * 1000 / bat->rate; + fprintf(bat->log, + _("Test%d, round trip latency %dms\n"), + num, + (int) bat->latency.result[num - 1]); + + for (n = 0; n < num; n++) { + if (bat->latency.result[n] > max) + max = bat->latency.result[n]; + if (bat->latency.result[n] < min) + min = bat->latency.result[n]; + sum += bat->latency.result[n]; + } + + /* The maximum is higher than the minimum's double */ + if (max / min > 2.0f) { + bat->latency.state = + LATENCY_STATE_COMPLETE_FAILURE; + bat->latency.is_capturing = false; + return; + + /* Final results */ + } else if (num == LATENCY_TEST_NUMBER) { + bat->latency.final_result = + (int) (sum / LATENCY_TEST_NUMBER); + fprintf(bat->log, + _("Final round trip latency: %dms\n"), + bat->latency.final_result); + + bat->latency.state = + LATENCY_STATE_COMPLETE_SUCCESS; + bat->latency.is_capturing = false; + return; + + /* Next step */ + } else + bat->latency.state = LATENCY_STATE_WAITING; + + bat->latency.number++; + + } else + /* Happens when an early noise comes in */ + bat->latency.state = LATENCY_STATE_WAITING; + + } else { + /* Still listening */ + bat->latency.samples += frames; + + /* Do not listen to more than a second + Maybe too much background noise */ + if (bat->latency.samples > bat->rate) { + bat->latency.error++; + + if (bat->latency.error > LATENCY_TEST_NUMBER) { + fprintf(bat->err, + _("Could not detect signal.")); + fprintf(bat->err, + _("Too much background noise?\n")); + bat->latency.state = + LATENCY_STATE_COMPLETE_FAILURE; + bat->latency.is_capturing = false; + return; + } + + /* let's start over */ + bat->latency.state = LATENCY_STATE_WAITING; + } + } + + return; +} + +static void calculate_threshold(struct bat *bat) +{ + float average; + float reference; + + /* Calculate the average loudness of the environment and create + a threshold value 16 decibels higher than the average loudness */ + average = bat->latency.sum / bat->latency.samples / 32767.0f; + reference = 20.0f * log10f(average) + 16.0f; + bat->latency.threshold = (int) (powf(10.0f, reference / 20.0f) + * 32767.0f); +} + +void roundtrip_latency_init(struct bat *bat) +{ + bat->latency.number = 1; + bat->latency.state = LATENCY_STATE_MEASURE_FOR_1_SECOND; + bat->latency.final_result = 0; + bat->latency.samples = 0; + bat->latency.sum = 0; + bat->latency.threshold = 0; + bat->latency.is_capturing = false; + bat->latency.is_playing = false; + bat->latency.error = 0; + bat->latency.xrun_error = false; + bat->format = BAT_PCM_FORMAT_S16_LE; + bat->frames = LATENCY_TEST_TIME_LIMIT * bat->rate; + bat->periods_played = 0; +} + +int handleinput(struct bat *bat, void *buffer, int frames) +{ + switch (bat->latency.state) { + /* Measuring average loudness for 1 second */ + case LATENCY_STATE_MEASURE_FOR_1_SECOND: + bat->latency.sum += sumaudio(bat, buffer, frames); + bat->latency.samples += frames; + + /* 1 second elapsed */ + if (bat->latency.samples >= bat->rate) { + calculate_threshold(bat); + bat->latency.state = LATENCY_STATE_PLAY_AND_LISTEN; + bat->latency.samples = 0; + bat->latency.sum = 0; + } + break; + + /* Playing sine wave and listening if it comes back */ + case LATENCY_STATE_PLAY_AND_LISTEN: + play_and_listen(bat, buffer, frames); + break; + + /* Waiting 1 second */ + case LATENCY_STATE_WAITING: + bat->latency.samples += frames; + + if (bat->latency.samples > bat->rate) { + /* 1 second elapsed, start over */ + bat->latency.samples = 0; + bat->latency.state = LATENCY_STATE_MEASURE_FOR_1_SECOND; + } + break; + + default: + return 0; + } + + return 0; +} + +int handleoutput(struct bat *bat, void *buffer, int bytes, int frames) +{ + int err = 0; + + /* If capture completed, terminate the playback */ + if (bat->periods_played * frames > 2 * bat->rate + && bat->latency.is_capturing == false) + return bat->latency.state; + + if (bat->latency.state == LATENCY_STATE_PLAY_AND_LISTEN) + err = generate_sine_wave(bat, frames, buffer); + else + /* Output silence */ + memset(buffer, 0, bytes); + + return err; +} diff --git a/bat/latencytest.h b/bat/latencytest.h new file mode 100644 index 0000000..be5e661 --- /dev/null +++ b/bat/latencytest.h @@ -0,0 +1,17 @@ +/* + * Copyright (C) 2013-2015 Intel Corporation + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + */ +void roundtrip_latency_init(struct bat *); +int handleinput(struct bat *, void *, int); +int handleoutput(struct bat *, void *, int, int); diff --git a/bat/tinyalsa.c b/bat/tinyalsa.c index a19dd90..f1c20fa 100644 --- a/bat/tinyalsa.c +++ b/bat/tinyalsa.c @@ -27,6 +27,7 @@ #include "common.h" #include "tinyalsa.h" +#include "latencytest.h" struct format_map_table { enum _bat_pcm_format format_bat; @@ -156,6 +157,40 @@ exit: } /** + * Process output data for latency test + */ +static int latencytest_process_output(struct bat *bat, struct pcm *pcm, + void *buffer, int bytes) +{ + int err = 0; + int frames = bytes / bat->frame_size; + + fprintf(bat->log, _("Play sample with %d frames buffer\n"), frames); + + bat->latency.is_playing = true; + + while (1) { + /* generate output data */ + err = handleoutput(bat, buffer, bytes, frames); + if (err != 0) + break; + + err = pcm_write(pcm, buffer, bytes); + if (err != 0) + break; + + if (bat->latency.state == LATENCY_STATE_COMPLETE_SUCCESS) + break; + + bat->periods_played++; + } + + bat->latency.is_playing = false; + + return err; +} + +/** * Play sample */ static int play_sample(struct bat *bat, struct pcm *pcm, @@ -322,7 +357,10 @@ void *playback_tinyalsa(struct bat *bat) } } - err = play_sample(bat, pcm, buffer, bufbytes); + if (bat->roundtriplatency) + err = latencytest_process_output(bat, pcm, buffer, bufbytes); + else + err = play_sample(bat, pcm, buffer, bufbytes); if (err < 0) { retval_play = err; goto exit4; @@ -385,6 +423,56 @@ static int capture_sample(struct bat *bat, struct pcm *pcm, } /** + * Process input data for latency test + */ +static int latencytest_process_input(struct bat *bat, struct pcm *pcm, + void *buffer, unsigned int bytes) +{ + int err = 0; + FILE *fp = NULL; + unsigned int bytes_read = 0; + unsigned int bytes_count = bat->frames * bat->frame_size; + + remove(bat->capture.file); + fp = fopen(bat->capture.file, "wb"); + err = -errno; + if (fp == NULL) { + fprintf(bat->err, _("Cannot open file: %s %d\n"), + bat->capture.file, err); + return err; + } + /* leave space for file header */ + if (fseek(fp, sizeof(struct wav_container), SEEK_SET) != 0) { + err = -errno; + fclose(fp); + return err; + } + + bat->latency.is_capturing = true; + + while (bytes_read < bytes_count && !pcm_read(pcm, buffer, bytes)) { + if (fwrite(buffer, 1, bytes, fp) != bytes) + break; + + err = handleinput(bat, buffer, bytes / bat->frame_size); + if (err != 0) + break; + + if (bat->latency.is_playing == false) + break; + + bytes_read += bytes; + } + + bat->latency.is_capturing = false; + + err = update_wav_header(bat, fp, bytes_read); + + fclose(fp); + return err; +} + +/** * Record */ void *record_tinyalsa(struct bat *bat) @@ -441,7 +529,10 @@ void *record_tinyalsa(struct bat *bat) pthread_cleanup_push(free, buffer); fprintf(bat->log, _("Recording ...\n")); - err = capture_sample(bat, pcm, buffer, bufbytes); + if (bat->roundtriplatency) + err = latencytest_process_input(bat, pcm, buffer, bufbytes); + else + err = capture_sample(bat, pcm, buffer, bufbytes); if (err != 0) { retval_record = err; goto exit3; -- 1.7.9.5

2 1

[alsa-devel] [PATCH v4] ASoC: rockchip: Add machine driver for RK3399 GRU Boards
by Xing Zheng 08 Jun '16

08 Jun '16

Because we need to support the multiple codecs (MAX98357A/RT5514/DA7219) on the RK3399 GRU boards, this patch can help us to support these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v4: - the compatible name needs to be based on a single reference design, and the name of the board is called "GRU" - use one i2sX card "rockchip,cpu = <&i2s0>" instead of array that is based on GRU - rename DOC to rockchip,rk3399-gru-sound.txt - rename compatible to rockchip,rk3399-gru-sound - rename source code to rk3399_gru_sound.c Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../bindings/sound/rockchip,rk3399-gru-sound.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rk3399_gru_sound.c | 339 ++++++++++++++++++++ 4 files changed, 367 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt create mode 100644 sound/soc/rockchip/rk3399_gru_sound.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt new file mode 100644 index 0000000..f19b6c8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-gru-sound.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs on GRU boards + +Required properties: +- compatible: "rockchip,rk3399-gru-sound" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-gru-sound"; + rockchip,cpu = <&i2s0>; + rockchip,codec = <&max98357a &rt5514 &da7219>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..c15dcc8 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_GRU_SOUND + tristate "ASoC support multiple codecs for Rockchip RK3399 GRU boards" + depends on SND_SOC_ROCKCHIP && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support multiple codecs for SoC + audio on Rockchip RK3399 GRU boards. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..84e5c7c 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-gru-sound-objs := rk3399_gru_sound.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_GRU_SOUND) += snd-soc-rk3399-gru-sound.o diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c new file mode 100644 index 0000000..cc9b943 --- /dev/null +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -0,0 +1,339 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-gru-sound" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speakers"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_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_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + snd_soc_dapm_disable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speakers"); + snd_soc_dapm_sync(dapm); + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_rt5514_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 *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + snd_soc_dapm_disable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Int Mic"); + snd_soc_dapm_sync(dapm); + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_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 *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + int mclk, ret; + + snd_soc_dapm_disable_pin(dapm, "Speakers"); + snd_soc_dapm_disable_pin(dapm, "Int Mic"); + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_sync(dapm); + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-gru-sound", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + cpu_node = of_parse_phandle(pdev->dev.of_node, "rockchip,cpu", 0); + if (!cpu_node) { + dev_err(&pdev->dev, "Property 'rockchip,cpu' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < card->num_links; i++) { + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-gru-sound", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

1 0

[alsa-devel] [PATCH v4] ASoC: rt5514: add rt5514 SPI driver
by Oder Chiou 08 Jun '16

08 Jun '16

The device has multiple control interfaces, I2C and SPI. The I2C interface mainly controls the register settings of codec. The SPI interface is in order to provide the high speed transmission of data. For example, high bandwidth memory read/write of DSP. The patch adds the rt5514 SPI driver for loading the firmware of DSP and retrieving the voice data from DSP after the system is waked up by specific voice. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5514-spi.c | 459 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514-spi.h | 38 ++++ sound/soc/codecs/rt5514.c | 136 ++++++++++++- sound/soc/codecs/rt5514.h | 4 + 6 files changed, 640 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/rt5514-spi.c create mode 100644 sound/soc/codecs/rt5514-spi.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..164bfb3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -660,6 +660,9 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate +config SND_SOC_RT5514_SPI + tristate + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..8260bd5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -103,6 +103,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o +snd-soc-rt5514-spi-objs := rt5514-spi.o snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -320,6 +321,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o +obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c new file mode 100644 index 0000000..8a9382e --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.c @@ -0,0 +1,459 @@ +/* + * rt5514-spi.c -- RT5514 SPI driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/kthread.h> +#include <linux/uaccess.h> +#include <linux/miscdevice.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5514-spi.h" + +static struct spi_device *rt5514_spi; + +struct rt5514_dsp { + struct device *dev; + struct delayed_work copy_work; + struct mutex dma_lock; + struct snd_pcm_substream *substream; + unsigned int buf_base, buf_limit, buf_rp; + size_t buf_size; + size_t dma_offset; + size_t dsp_offset; +}; + +static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = 0x20000 / 8, + .periods_min = 8, + .periods_max = 8, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 0x20000, +}; + +static struct snd_soc_dai_driver rt5514_spi_dai = { + .name = "rt5514-dsp-cpu-dai", + .id = 0, + .capture = { + .stream_name = "DSP Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static void rt5514_spi_copy_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, copy_work.work); + struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + size_t period_bytes, truncated_bytes = 0; + + mutex_lock(&rt5514_dsp->dma_lock); + if (!rt5514_dsp->substream) { + dev_err(rt5514_dsp->dev, "No pcm substream\n"); + goto done; + } + + period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + + if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) + period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + + if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + period_bytes); + + if (rt5514_dsp->buf_rp + period_bytes == rt5514_dsp->buf_limit) + rt5514_dsp->buf_rp = rt5514_dsp->buf_base; + else + rt5514_dsp->buf_rp += period_bytes; + } else { + truncated_bytes = rt5514_dsp->buf_limit - rt5514_dsp->buf_rp; + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + truncated_bytes); + + rt5514_spi_burst_read(rt5514_dsp->buf_base, + runtime->dma_area + rt5514_dsp->dma_offset + + truncated_bytes, period_bytes - truncated_bytes); + + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + + period_bytes - truncated_bytes; + } + + rt5514_dsp->dma_offset += period_bytes; + if (rt5514_dsp->dma_offset >= runtime->dma_bytes) + rt5514_dsp->dma_offset = 0; + + rt5514_dsp->dsp_offset += period_bytes; + + snd_pcm_period_elapsed(rt5514_dsp->substream); + + if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 5); +done: + mutex_unlock(&rt5514_dsp->dma_lock); +} + +/* PCM for streaming audio from the DSP buffer */ +static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &rt5514_spi_pcm_hardware); + + return 0; +} + +static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + mutex_lock(&rt5514_dsp->dma_lock); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + rt5514_dsp->substream = substream; + mutex_unlock(&rt5514_dsp->dma_lock); + + return ret; +} + +static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + mutex_lock(&rt5514_dsp->dma_lock); + rt5514_dsp->substream = NULL; + mutex_unlock(&rt5514_dsp->dma_lock); + + cancel_delayed_work_sync(&rt5514_dsp->copy_work); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int rt5514_spi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + u8 buf[8]; + + rt5514_dsp->dma_offset = 0; + rt5514_dsp->dsp_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + return 0; +} + +static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); + } + + return 0; +} + +static snd_pcm_uframes_t rt5514_spi_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + return bytes_to_frames(runtime, rt5514_dsp->dma_offset); +} + +static struct snd_pcm_ops rt5514_spi_pcm_ops = { + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .trigger = rt5514_spi_trigger, + .prepare = rt5514_spi_prepare, + .pointer = rt5514_spi_pcm_pointer, + .mmap = snd_pcm_lib_mmap_vmalloc, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) +{ + struct rt5514_dsp *rt5514_dsp; + + rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), + GFP_KERNEL); + + rt5514_dsp->dev = &rt5514_spi->dev; + mutex_init(&rt5514_dsp->dma_lock); + INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); + snd_soc_platform_set_drvdata(platform, rt5514_dsp); + + return 0; +} + +static struct snd_soc_platform_driver rt5514_spi_platform = { + .probe = rt5514_spi_pcm_probe, + .ops = &rt5514_spi_pcm_ops, +}; + +static const struct snd_soc_component_driver rt5514_spi_dai_component = { + .name = "rt5514-spi-dai", +}; + +/** + * rt5514_spi_burst_read - Read data from SPI by rt5514 address. + * @addr: Start address. + * @rxbuf: Data Buffer for reading. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; + int status; + u8 write_buf[8]; + unsigned int i, end, offset = 0; + + struct spi_message message; + struct spi_transfer x[3]; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + spi_message_init(&message); + memset(x, 0, sizeof(x)); + + x[0].len = 5; + x[0].tx_buf = write_buf; + spi_message_add_tail(&x[0], &message); + + x[1].len = 4; + x[1].tx_buf = write_buf; + spi_message_add_tail(&x[1], &message); + + x[2].len = end; + x[2].rx_buf = rxbuf + offset; + spi_message_add_tail(&x[2], &message); + + status = spi_sync(rt5514_spi, &message); + + if (status) + return false; + + offset += RT5514_SPI_BUF_LEN; + } + + for (i = 0; i < len; i += 8) { + write_buf[0] = rxbuf[i + 0]; + write_buf[1] = rxbuf[i + 1]; + write_buf[2] = rxbuf[i + 2]; + write_buf[3] = rxbuf[i + 3]; + write_buf[4] = rxbuf[i + 4]; + write_buf[5] = rxbuf[i + 5]; + write_buf[6] = rxbuf[i + 6]; + write_buf[7] = rxbuf[i + 7]; + + rxbuf[i + 0] = write_buf[7]; + rxbuf[i + 1] = write_buf[6]; + rxbuf[i + 2] = write_buf[5]; + rxbuf[i + 3] = write_buf[4]; + rxbuf[i + 4] = write_buf[3]; + rxbuf[i + 5] = write_buf[2]; + rxbuf[i + 6] = write_buf[1]; + rxbuf[i + 7] = write_buf[0]; + } + + return true; +} + +/** + * rt5514_spi_burst_write - Write data to SPI by rt5514 address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = txbuf[offset + i + 0]; + write_buf[i + 11] = txbuf[offset + i + 1]; + write_buf[i + 10] = txbuf[offset + i + 2]; + write_buf[i + 9] = txbuf[offset + i + 3]; + write_buf[i + 8] = txbuf[offset + i + 4]; + write_buf[i + 7] = txbuf[offset + i + 5]; + write_buf[i + 6] = txbuf[offset + i + 6]; + write_buf[i + 5] = txbuf[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + spi_write(rt5514_spi, write_buf, end + 6); + + offset += RT5514_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); + +static int rt5514_spi_probe(struct spi_device *spi) +{ + int ret; + + rt5514_spi = spi; + + ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register platform.\n"); + goto err_plat; + } + + ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register component.\n"); + goto err_comp; + } + + return 0; +err_comp: + snd_soc_unregister_platform(&spi->dev); +err_plat: + + return 0; +} + +static int rt5514_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_component(&spi->dev); + snd_soc_unregister_platform(&spi->dev); + + return 0; +} + +static const struct of_device_id rt5514_of_match[] = { + { .compatible = "realtek,rt5514", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5514_of_match); + +static struct spi_driver rt5514_spi_driver = { + .driver = { + .name = "rt5514", + .of_match_table = of_match_ptr(rt5514_of_match), + }, + .probe = rt5514_spi_probe, + .remove = rt5514_spi_remove, +}; +module_spi_driver(rt5514_spi_driver); + +MODULE_DESCRIPTION("RT5514 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h new file mode 100644 index 0000000..f69b1cd --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.h @@ -0,0 +1,38 @@ +/* + * rt5514-spi.h -- RT5514 driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5514_SPI_H__ +#define __RT5514_SPI_H__ + +/** + * RT5514_SPI_BUF_LEN is the buffer size of SPI master controller. +*/ +#define RT5514_SPI_BUF_LEN 240 + +#define RT5514_BUFFER_VOICE_BASE 0x18001034 +#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 +#define RT5514_BUFFER_VOICE_RP 0x1800103c +#define RT5514_BUFFER_VOICE_SIZE 0x18001040 + +/* SPI Command */ +enum { + RT5514_SPI_CMD_16_READ = 0, + RT5514_SPI_CMD_16_WRITE, + RT5514_SPI_CMD_32_READ, + RT5514_SPI_CMD_32_WRITE, + RT5514_SPI_CMD_BURST_READ, + RT5514_SPI_CMD_BURST_WRITE, +}; + +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len); +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5514_SPI_H__ */ diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 879bf60..ecb0989 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -30,6 +30,9 @@ #include "rl6231.h" #include "rt5514.h" +#if defined(CONFIG_SND_SOC_RT5514_SPI) +#include "rt5514-spi.h" +#endif static const struct reg_sequence rt5514_i2c_patch[] = { {0x1800101c, 0x00000000}, @@ -110,6 +113,35 @@ static const struct reg_default rt5514_reg[] = { {RT5514_VENDOR_ID2, 0x10ec5514}, }; +static void rt5514_enable_dsp_prepare(struct rt5514_priv *rt5514) +{ + /* Reset */ + regmap_write(rt5514->i2c_regmap, 0x18002000, 0x000010ec); + /* LDO_I_limit */ + regmap_write(rt5514->i2c_regmap, 0x18002200, 0x00028604); + /* I2C bypass enable */ + regmap_write(rt5514->i2c_regmap, 0xfafafafa, 0x00000001); + /* mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x0005514b); + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* I2C bypass disable */ + regmap_write(rt5514->i2c_regmap, 0xfafafafa, 0x00000000); + /* PIN config */ + regmap_write(rt5514->i2c_regmap, 0x18002070, 0x00000040); + /* PLL3(QN)=RCOSC*(10+2) */ + regmap_write(rt5514->i2c_regmap, 0x18002240, 0x0000000a); + /* PLL3 source=RCOSC, fsi=rt_clk */ + regmap_write(rt5514->i2c_regmap, 0x18002100, 0x0000000b); + /* Power on RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* DSP clk source = pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* Enable DSP clk auto switch */ + regmap_write(rt5514->i2c_regmap, 0x18001114, 0x00000001); + /* Reduce DSP power */ + regmap_write(rt5514->i2c_regmap, 0x18001118, 0x00000001); +} + static bool rt5514_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -248,6 +280,74 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static int rt5514_dsp_voice_wake_up_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5514->dsp_enabled; + + return 0; +} + +static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + const struct firmware *fw = NULL; + + if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) + return 0; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + rt5514->dsp_enabled = ucontrol->value.integer.value[0]; + + if (rt5514->dsp_enabled) { + rt5514_enable_dsp_prepare(rt5514); + + request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff60000, fw->data, + ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, "There is no SPI driver for" + " loading the firmware\n"); +#endif + release_firmware(fw); + fw = NULL; + } + + request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ffc0000, fw->data, + ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, "There is no SPI driver for" + " loading the firmware\n"); +#endif + release_firmware(fw); + fw = NULL; + } + + /* DSP run */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, + 0x00055148); + } else { + regmap_multi_reg_write(rt5514->i2c_regmap, + rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->regmap); + } + } + + return 0; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -257,6 +357,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, adc_vol_tlv), + SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, + rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), }; /* ADC Mixer*/ @@ -365,6 +467,35 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int rt5514_pre_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /** + * If the DSP is enabled in start of recording, the DSP + * should be disabled, and sync back to normal recording + * settings to make sure recording properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regmap_multi_reg_write(rt5514->i2c_regmap, + rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->regmap); + } + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -472,6 +603,8 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PRE("DAPM Pre", rt5514_pre_event), }; static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { @@ -871,7 +1004,6 @@ static const struct regmap_config rt5514_i2c_regmap = { .reg_bits = 32, .val_bits = 32, - .max_register = RT5514_DSP_MAPPING | RT5514_VENDOR_ID2, .readable_reg = rt5514_i2c_readable_register, .cache_type = REGCACHE_NONE, @@ -944,7 +1076,7 @@ static int rt5514_i2c_probe(struct i2c_client *i2c, return -ENODEV; } - ret = regmap_register_patch(rt5514->i2c_regmap, rt5514_i2c_patch, + ret = regmap_multi_reg_write(rt5514->i2c_regmap, rt5514_i2c_patch, ARRAY_SIZE(rt5514_i2c_patch)); if (ret != 0) dev_warn(&i2c->dev, "Failed to apply i2c_regmap patch: %d\n", diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 6ad8a61..6e89e7d 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -225,6 +225,9 @@ #define RT5514_PLL_INP_MAX 40000000 #define RT5514_PLL_INP_MIN 256000 +#define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" +#define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" + /* System Clock Source */ enum { RT5514_SCLK_S_MCLK, @@ -247,6 +250,7 @@ struct rt5514_priv { int pll_src; int pll_in; int pll_out; + int dsp_enabled; }; #endif /* __RT5514_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 00/10] ASoC: Intel: Skylake: Improvements in Skylake driver
by jeeja.kp＠intel.com 08 Jun '16

08 Jun '16

From: Jeeja KP <jeeja.kp(a)intel.com> This series has the below changes: o Modify the sequence when deleting the pipeline o Disable SRAM retention feature o Cleanup of driver resource in suspend o Reset the pipe in case of XRUN o Enable simultaneous HDMI and DP playback o Instead of modifying the runtime->delay in pointer query, just report position in pointer query Dharageswari R (4): ASoC: Intel: Skylake: Don't pause stopped pipeline while deleting ASoC: Intel: Skylake: Disable SRAM Retention before D3 ASoC: Intel: Skylake: Avoid freeing up of unallocated memory/mcps ASoC: Intel: Skylake: Clean up of driver resources in suspend Jeeja KP (6): ASoC: Intel: Skylake: Reset DSP pipe when host/link DMA is reset ASoC: Intel: Skylake: Set the pipe state to paused when paused ASoC: Intel: Skylake: Create Pipe to widget list in soc probe ASoC: Intel: Skylake: Set the DSP pipe type ASoC: Intel: Skylake: Copy the pipe parameter by pipe type ASoC: Intel: Skylake: Report position in pointer query sound/soc/intel/skylake/skl-messages.c | 45 +++++-- sound/soc/intel/skylake/skl-pcm.c | 93 +++++--------- sound/soc/intel/skylake/skl-sst-ipc.h | 1 + sound/soc/intel/skylake/skl-sst.c | 10 ++ sound/soc/intel/skylake/skl-topology.c | 213 +++++++++++++++++++++++---------- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 4 + sound/soc/intel/skylake/skl.h | 4 + 8 files changed, 243 insertions(+), 133 deletions(-) -- 2.5.0

5 24

[alsa-devel] [PATCH 0/2 v8] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 07 Jun '16

07 Jun '16

ARC AXS10x platforms consist of a mainboard with several peripherals. One of those peripherals is an HDMI output port controlled by the ADV7511 transmitter. This patch set adds I2S audio for the AXS10x platform. NOTE: Although the mainline I2S driver uses ALSA DMA engine, this controller can be built without DMA support so it was necessary to add this custom platform driver so that HDMI audio works in AXS boards. Changes v7 -> v8: * Build PIO PCM as module * Always unmask interrupts even when in DMA mode * Fallback to PIO mode only if ALSA DMA engine probe fails Changes v6 -> v7: * Discard the use of memcpy * Report IRQ_HANDLED only when there is an IRQ * Use interrupts to check if PIO mode is in use * Unmask interrupts only when in PIO mode * Remove empty functions Changes v5 -> v6: * Use SNDRV_DMA_TYPE_CONTINUOUS Changes v4 -> v5: * Resolve undefined references when compiling as module * Dropped adv7511 audio patches * Use DMA properties in I2S to check which mode to use: PIO or DMA (as suggested by Lars-Peter Clausen) Changes v3 -> v4: * Reintroduced custom PCM driver (see note below) * Use DT boolean to switch between ALSA DMA engine PCM or custom PCM * Use fifo depth to program I2S FCR * Update I2S documentation Changes v2 -> v3: * Removed pll_config functions (as suggested by Alexey Brodkin) * Removed HDMI start at adv7511_core (as suggested by Archit Taneja) * Use NOP functions for adv7511_audio (as suggested by Archit Taneja) * Added adv7511_audio_exit() function (as suggested by Archit Taneja) * Moved adv7511 to its own folder (as suggested by Archit Taneja) * Separated file rename of adv7511_core (as suggested by Emil Velikov) * Compile adv7511 as module if ALSA SoC is compiled as module * Load adv7511 audio only if declared in device tree (as suggested by Laurent Pinchart) * Dropped custom platform driver, using now ALSA DMA engine * Dropped IRQ handler for I2S Changes v1 -> v2: * DT bindings moved to separate patch (as suggested by Alexey Brodkin) * Removed defconfigs entries (as suggested by Alexey Brodkin) Cc: Carlos Palminha <palminha(a)synopsys.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Rob Herring <robh(a)kernel.org> Cc: Alexey Brodkin <abrodkin(a)synopsys.com> Cc: linux-snps-arc(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Jose Abreu (2): ASoC: dwc: Add PIO PCM extension ASoC: dwc: Add irq parameter to DOCUMENTATION .../devicetree/bindings/sound/designware-i2s.txt | 4 + sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware_i2s.c | 152 ++++++-------- sound/soc/dwc/designware_pcm.c | 222 +++++++++++++++++++++ sound/soc/dwc/local.h | 122 +++++++++++ 6 files changed, 422 insertions(+), 88 deletions(-) create mode 100644 sound/soc/dwc/designware_pcm.c create mode 100644 sound/soc/dwc/local.h -- 1.9.1

2 3

[alsa-devel] [PATCH v4 0/2] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 07 Jun '16

07 Jun '16

This patchset implements audio interface in HDMI drm driver. Implementation is based on ASoC generic hdmi codec driver( https://patchwork.kernel.org/patch/8713141/). It also proposes helper functions to compute N and CTS parameters according to HDMI 1.4b specification. V4: fixes for "video: hdmi: add helper functions for N and CTS" - typo error and additional comments - cts_1_ratio computation - warning reported by kbuild test robot - add rounded value for 297/1.001 MHz V3: - video: hdmi: add helper function for N and CTS Also used on Mediatek platform (https://patchwork.kernel.org/patch/8887341) delta vs V2: - typo fixes - if/else code optimisation - drm: sti: Add ASoC generic hdmi codec support. - typo fixes - add audio registers in debugfs information V2: RFC https://patchwork.kernel.org/patch/8091531/("video: hdmi: add helper function for N and CTS") https://patchwork.kernel.org/patch/8091561/("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") - patch: video: hdmi: add helper function for N and CTS Fixes based on Russel King remarks - Duplicate function to have a separte treatment for coherent and non-coherent clocks - Add ratio field for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") https://patchwork.kernel.org/patch/8062611/ ("video: hdmi: add helper function for N and CTS") Arnaud Pouliquen (2): video: hdmi: add helper functions for N and CTS drm: sti: Add ASoC generic hdmi codec support. drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 248 ++++++++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 13 +++ drivers/video/hdmi.c | 208 ++++++++++++++++++++++++++++++++++ include/linux/hdmi.h | 24 ++++ 5 files changed, 477 insertions(+), 17 deletions(-) -- 1.9.1

2 9

[alsa-devel] [PATCH v3] ASoC: rockchip: Add machine driver for MAX98357A/RT5514/DA7219
by Xing Zheng 07 Jun '16

07 Jun '16

There are multi codec devices on the RK3399 platform, we can use this patch support and control these codecs. Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- Changes in v3: - rename DOC to rockchip,rk3399-max98357a-rt5514-da7219.txt - rename compatible to rockchip,rk3399-max98357a-rt5514-da7219 - rename source code to rk3399_max98357a_rt5514_da7219.c Changes in v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../rockchip,rk3399-max98357a-rt5514-da7219.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + .../soc/rockchip/rk3399_max98357a_rt5514_da7219.c | 333 ++++++++++++++++++++ 4 files changed, 361 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip,rk3399-max98357a-rt5514-da7219.txt create mode 100644 sound/soc/rockchip/rk3399_max98357a_rt5514_da7219.c diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3399-max98357a-rt5514-da7219.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3399-max98357a-rt5514-da7219.txt new file mode 100644 index 0000000..3ae603e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3399-max98357a-rt5514-da7219.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs + +Required properties: +- compatible: "rockchip,rk3399-max98357a-rt5514-da7219" +- rockchip,cpu: The phandle of the Rockchip I2S controller that's + connected to the codecs +- rockchip,codec: The phandle of the MAX98357A/RT5514/DA7219 codecs + +Example: + +sound { + compatible = "rockchip,rk3399-max98357a-rt5514-da7219"; + rockchip,cpu = <&i2s0 &i2s0 &i2s1>; + rockchip,codec = <&max98357a &headsetcodec &codec>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..e5bd1f9 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -41,3 +41,14 @@ config SND_SOC_ROCKCHIP_RT5645 help Say Y or M here if you want to add support for SoC audio on Rockchip boards using the RT5645/RT5650 codec, such as Veyron. + +config SND_SOC_RK3399_MAX98357A_RT5514_DA7219 + tristate "ASoC support for Rockchip RK3399 boards using the MAX98357A/RT5514/DA7219" + depends on SND_SOC_ROCKCHIP && GPIOLIB + select SND_SOC_ROCKCHIP_I2S + select SND_SOC_MAX98357A + select SND_SOC_RT5514 + select SND_SOC_DA7219 + help + Say Y or M here if you want to add support for SoC audio on Rockchip RK3399 + boards using the MAX98357A/RT5514/DA7219. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..0d9ca0a 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rk3399-max98357a-rt5514-da7219-objs := rk3399_max98357a_rt5514_da7219.o obj-$(CONFIG_SND_SOC_ROCKCHIP_MAX98090) += snd-soc-rockchip-max98090.o obj-$(CONFIG_SND_SOC_ROCKCHIP_RT5645) += snd-soc-rockchip-rt5645.o +obj-$(CONFIG_SND_SOC_RK3399_MAX98357A_RT5514_DA7219) += snd-soc-rk3399-max98357a-rt5514-da7219.o diff --git a/sound/soc/rockchip/rk3399_max98357a_rt5514_da7219.c b/sound/soc/rockchip/rk3399_max98357a_rt5514_da7219.c new file mode 100644 index 0000000..ec88600 --- /dev/null +++ b/sound/soc/rockchip/rk3399_max98357a_rt5514_da7219.c @@ -0,0 +1,333 @@ +/* + * Rockchip machine ASoC driver for boards using MAX98357A/RT5514/DA7219 + * + * Copyright (c) 2016, ROCKCHIP CORPORATION. All rights reserved. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/delay.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "rockchip_i2s.h" +#include "../codecs/da7219.h" +#include "../codecs/da7219-aad.h" +#include "../codecs/rt5514.h" + +#define DRV_NAME "rk3399-max98357a-rt5514-da7219" + +#define SOUND_FS 256 + +static struct snd_soc_jack rockchip_sound_jack; + +static const struct snd_soc_dapm_widget rockchip_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route rockchip_dapm_routes[] = { + /* Input Lines */ + {"MIC", NULL, "Headset Mic"}, + {"DMIC1L", NULL, "Int Mic"}, + {"DMIC1R", NULL, "Int Mic"}, + + /* Output Lines */ + {"Headphones", NULL, "HPL"}, + {"Headphones", NULL, "HPR"}, + {"Speakers", NULL, "Speakers"}, +}; + +static const struct snd_kcontrol_new rockchip_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static int rockchip_sound_max98357a_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_dapm_context *dapm = &rtd->card->dapm; + unsigned int mclk; + int ret; + + /* max98357a supports these sample rates */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = params_rate(params) * SOUND_FS; + break; + default: + dev_err(rtd->card->dev, "%s() doesn't support this sample rate: %d\n", + __func__, params_rate(params)); + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(rtd->cpu_dai, 0, mclk, 0); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, mclk, ret); + return ret; + } + + snd_soc_dapm_disable_pin(dapm, "Headphones"); + snd_soc_dapm_enable_pin(dapm, "Speakers"); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int rockchip_sound_rt5514_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 *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + unsigned int mclk; + int ret; + + mclk = params_rate(params) * SOUND_FS; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->card->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5514_SCLK_S_MCLK, + mclk, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(rtd->card->dev, "%s() error setting sysclk to %u: %d\n", + __func__, params_rate(params) * 512, ret); + return ret; + } + + return 0; +} + +static int rockchip_sound_da7219_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 *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dapm_context *dapm = &rtd->card->dapm; + int mclk, ret; + + /* in bypass mode, the mclk has to be one of the frequencies below */ + switch (params_rate(params)) { + case 8000: + case 16000: + case 24000: + case 32000: + case 48000: + case 64000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + case 88200: + mclk = 11289600; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, + SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set codec clock in %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_MCLK, 0, 0); + if (ret < 0) { + dev_err(codec_dai->dev, "Can't set pll sysclk mclk %d\n", ret); + return ret; + } + + snd_soc_dapm_disable_pin(dapm, "Speakers"); + snd_soc_dapm_enable_pin(dapm, "Headphones"); + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec_dais[0]->codec; + int ret; + + /* Enable Headset and 4 Buttons Jack detection */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &rockchip_sound_jack, NULL, 0); + + if (ret) { + dev_err(rtd->card->dev, "New Headset Jack failed! (%d)\n", ret); + return ret; + } + + da7219_aad_jack_det(codec, &rockchip_sound_jack); + + return 0; +} + +static struct snd_soc_ops rockchip_sound_max98357a_ops = { + .hw_params = rockchip_sound_max98357a_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_rt5514_ops = { + .hw_params = rockchip_sound_rt5514_hw_params, +}; + +static struct snd_soc_ops rockchip_sound_da7219_ops = { + .hw_params = rockchip_sound_da7219_hw_params, +}; + +enum { + DAILINK_MAX98357A, + DAILINK_RT5514, + DAILINK_DA7219, +}; + +static struct snd_soc_dai_link rockchip_dailinks[] = { + [DAILINK_MAX98357A] = { + .name = "MAX98357A", + .stream_name = "MAX98357A PCM", + .codec_dai_name = "HiFi", + .ops = &rockchip_sound_max98357a_ops, + /* set max98357a as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_RT5514] = { + .name = "RT5514", + .stream_name = "RT5514 PCM", + .codec_dai_name = "rt5514-aif1", + .ops = &rockchip_sound_rt5514_ops, + /* set rt5514 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, + [DAILINK_DA7219] = { + .name = "DA7219", + .stream_name = "DA7219 PCM", + .codec_dai_name = "da7219-hifi", + .init = rockchip_sound_da7219_init, + .ops = &rockchip_sound_da7219_ops, + /* set da7219 as slave */ + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + }, +}; + +static struct snd_soc_card rockchip_sound_card = { + .name = "rk3399-max98357a-rt5514-da7219", + .owner = THIS_MODULE, + .dai_link = rockchip_dailinks, + .num_links = ARRAY_SIZE(rockchip_dailinks), + .dapm_widgets = rockchip_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rockchip_dapm_widgets), + .dapm_routes = rockchip_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rockchip_dapm_routes), + .controls = rockchip_controls, + .num_controls = ARRAY_SIZE(rockchip_controls), +}; + +static int rockchip_sound_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &rockchip_sound_card; + struct device_node *cpu_node; + int i, ret; + + for (i = 0; i < card->num_links; i++) { + cpu_node = of_parse_phandle(pdev->dev.of_node, + "rockchip,cpu", i); + if (!cpu_node) { + dev_err(&pdev->dev, "Property[%d] 'rockchip,cpu' missing or invalid\n", i); + return -EINVAL; + } + + rockchip_dailinks[i].platform_of_node = cpu_node; + rockchip_dailinks[i].cpu_of_node = cpu_node; + + rockchip_dailinks[i].codec_of_node = + of_parse_phandle(pdev->dev.of_node, "rockchip,codec", i); + if (!rockchip_dailinks[i].codec_of_node) { + dev_err(&pdev->dev, + "Property[%d] 'rockchip,codec' missing or invalid\n", i); + return -EINVAL; + } + } + + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + + ret = devm_snd_soc_register_card(&pdev->dev, card); + if (ret) + dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", + __func__, ret); + + return ret; +} + +static const struct of_device_id rockchip_sound_of_match[] = { + { .compatible = "rockchip,rk3399-max98357a-rt5514-da7219", }, + {}, +}; + +static struct platform_driver rockchip_sound_driver = { + .probe = rockchip_sound_probe, + .driver = { + .name = DRV_NAME, + .of_match_table = rockchip_sound_of_match, +#ifdef CONFIG_PM + .pm = &snd_soc_pm_ops, +#endif + }, +}; + +module_platform_driver(rockchip_sound_driver); + +MODULE_AUTHOR("Xing Zheng <zhengxing(a)rock-chips.com>"); +MODULE_DESCRIPTION("Rockchip ASoC Machine Driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:" DRV_NAME); +MODULE_DEVICE_TABLE(of, rockchip_sound_of_match); -- 1.7.9.5

3 13

[alsa-devel] [PATCH] ASoC: rt5670: fix HP Playback Volume control
by Bard Liao 07 Jun '16

07 Jun '16

The register setting for HP Playback Volume is inverted. So, set the invert flag in SOC_DOUBLE_TLV. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5670.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 49a9e70..0af5ddb 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -619,7 +619,7 @@ static const struct snd_kcontrol_new rt5670_snd_controls[] = { RT5670_L_MUTE_SFT, RT5670_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5670_HP_VOL, RT5670_L_VOL_SFT, RT5670_R_VOL_SFT, - 39, 0, out_vol_tlv), + 39, 1, out_vol_tlv), /* OUTPUT Control */ SOC_DOUBLE("OUT Channel Switch", RT5670_LOUT1, RT5670_VOL_L_SFT, RT5670_VOL_R_SFT, 1, 1), -- 1.8.1.1.439.g50a6b54

2 2

Re: [alsa-devel] [PATCH] hda/realtek: Add T560 docking unit fixup
by Takashi Iwai 07 Jun '16

07 Jun '16

On Tue, 07 Jun 2016 13:14:21 +0200, Torsten Hilbrich wrote: > > Tested with Lenovo Ultradock. Fixes the non-working headphone jack on > the docking unit. > > Signed-off-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> > Tested-by: Torsten Hilbrich <torsten.hilbrich(a)secunet.com> Applied, thanks. Takashi > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index 37e327f..8370127 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -5573,6 +5573,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x17aa, 0x2218, "Thinkpad X1 Carbon 2nd", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2223, "ThinkPad T550", ALC292_FIXUP_TPT440_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2226, "ThinkPad X250", ALC292_FIXUP_TPT440_DOCK), > + SND_PCI_QUIRK(0x17aa, 0x2231, "Thinkpad T560", ALC292_FIXUP_TPT460), > SND_PCI_QUIRK(0x17aa, 0x2233, "Thinkpad", ALC292_FIXUP_TPT460), > SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), > SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), > -- > 2.1.4 > >

1 0

[alsa-devel] [PATCH] ALSA: Add rate defines for 352k8 and 384k
by DigitalDreamtime 07 Jun '16

07 Jun '16

Add SNDRV_PCM_RATE_352800 and SNDRV_PCM_RATE_384000 defines to pcm.h, for 352k8 and 384k sample rates. Add SNDRV_PCM_RATE_8000_384000 define to pcm.h. Update rates list in pcm_native.c. Yes, we can use CONTINUOUS/KNOT and constraints, to support the x8 sample rates, but having the defines requires less code in drivers. Many more DAC chips are now supporting the x8 44k1/48k multiples. >From a high-res viewpoint, the DXD standard is 352k8, 24 bit. That alone should justify the addition of a define for it. Signed-off-by: DigitalDreamtime <clive.messer(a)digitaldreamtime.co.uk> --- include/sound/pcm.h | 5 +++++ sound/core/pcm_native.c | 5 +++-- 2 files changed, 8 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index af1fb37..86e0bbe 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -129,6 +129,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ @@ -141,6 +143,9 @@ struct snd_pcm_ops { SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) #define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\ SNDRV_PCM_RATE_192000) +#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000 |\ + SNDRV_PCM_RATE_352800 |\ + SNDRV_PCM_RATE_384000) #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c61fd50..21cc13d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1981,12 +1981,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_384000 != 1 << 14 #error "Change this table" #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000, + 352800, 384000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), -- 2.5.5

3 2

[alsa-devel] [PATCH] ALSA: Add rate defines for 352k8 and 384k
by DigitalDreamtime 07 Jun '16

07 Jun '16

Add SNDRV_PCM_RATE_352800 and SNDRV_PCM_RATE_384000 defines to pcm.h, for 352k8 and 384k sample rates. Add SNDRV_PCM_RATE_8000_384000 define to pcm.h. Update rates list in pcm_native.c. Yes, we can use CONTINUOUS/KNOT and constraints, to support the x8 sample rates, but having the defines requires less code in drivers. Many more DAC chips are now supporting the x8 44k1/48k multiples. >From a high-res viewpoint, the DXD standard is 352k8, 24 bit. That alone should justify the addition of a define for it. Signed-off-by: DigitalDreamtime <clive.messer(a)digitaldreamtime.co.uk> --- include/sound/pcm.h | 5 +++++ sound/core/pcm_native.c | 5 +++-- 2 files changed, 8 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index af1fb37..86e0bbe 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -129,6 +129,8 @@ struct snd_pcm_ops { #define SNDRV_PCM_RATE_96000 (1<<10) /* 96000Hz */ #define SNDRV_PCM_RATE_176400 (1<<11) /* 176400Hz */ #define SNDRV_PCM_RATE_192000 (1<<12) /* 192000Hz */ +#define SNDRV_PCM_RATE_352800 (1<<13) /* 352800Hz */ +#define SNDRV_PCM_RATE_384000 (1<<14) /* 384000Hz */ #define SNDRV_PCM_RATE_CONTINUOUS (1<<30) /* continuous range */ #define SNDRV_PCM_RATE_KNOT (1<<31) /* supports more non-continuos rates */ @@ -141,6 +143,9 @@ struct snd_pcm_ops { SNDRV_PCM_RATE_88200|SNDRV_PCM_RATE_96000) #define SNDRV_PCM_RATE_8000_192000 (SNDRV_PCM_RATE_8000_96000|SNDRV_PCM_RATE_176400|\ SNDRV_PCM_RATE_192000) +#define SNDRV_PCM_RATE_8000_384000 (SNDRV_PCM_RATE_8000_192000 |\ + SNDRV_PCM_RATE_352800 |\ + SNDRV_PCM_RATE_384000) #define _SNDRV_PCM_FMTBIT(fmt) (1ULL << (__force int)SNDRV_PCM_FORMAT_##fmt) #define SNDRV_PCM_FMTBIT_S8 _SNDRV_PCM_FMTBIT(S8) #define SNDRV_PCM_FMTBIT_U8 _SNDRV_PCM_FMTBIT(U8) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c61fd50..21cc13d 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1981,12 +1981,13 @@ static int snd_pcm_hw_rule_sample_bits(struct snd_pcm_hw_params *params, return snd_interval_refine(hw_param_interval(params, rule->var), &t); } -#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_192000 != 1 << 12 +#if SNDRV_PCM_RATE_5512 != 1 << 0 || SNDRV_PCM_RATE_384000 != 1 << 14 #error "Change this table" #endif static unsigned int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, - 48000, 64000, 88200, 96000, 176400, 192000 }; + 48000, 64000, 88200, 96000, 176400, 192000, + 352800, 384000 }; const struct snd_pcm_hw_constraint_list snd_pcm_known_rates = { .count = ARRAY_SIZE(rates), -- 2.5.5

3 8

[alsa-devel] [PATCH] ALSA: sh: aica: Remove deprecated create_workqueue
by Bhaktipriya Shridhar 07 Jun '16

07 Jun '16

System workqueues have been able to handle high level of concurrency for a long time now and there's no reason to use dedicated workqueues just to gain concurrency. Since aica_queue for AICA sound driver has workitem dreamcastcard->spu_dma_work (maps to run_spu_dma) which is involved in aica dma transfers and is not being used on a memory reclaim path, dedicated aica_queue has been replaced with the use of system_wq. Unlike a dedicated per-cpu workqueue created with create_workqueue(), system_wq allows multiple work items to overlap executions even on the same CPU; however, a per-cpu workqueue doesn't have any CPU locality or global ordering guarantees unless the target CPU is explicitly specified and thus the increase of local concurrency shouldn't make any difference. Since the work items could be pending, flush_work() has been used in snd_aicapcm_pcm_close() to ensure that there is no pending task while disconnecting the driver. Signed-off-by: Bhaktipriya Shridhar <bhaktipriya96(a)gmail.com> --- sound/sh/aica.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/sh/aica.c b/sound/sh/aica.c index ad3d9ae..fbbc252 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -63,9 +63,6 @@ MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); module_param(enable, bool, 0644); MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); -/* Use workqueue */ -static struct workqueue_struct *aica_queue; - /* Simple platform device */ static struct platform_device *pd; static struct resource aica_memory_space[2] = { @@ -327,7 +324,7 @@ static void aica_period_elapsed(unsigned long timer_var) dreamcastcard->current_period = play_period; if (unlikely(dreamcastcard->dma_check == 0)) dreamcastcard->dma_check = 1; - queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); + schedule_work(&(dreamcastcard->spu_dma_work)); } static void spu_begin_dma(struct snd_pcm_substream *substream) @@ -337,7 +334,7 @@ static void spu_begin_dma(struct snd_pcm_substream *substream) runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; /*get the queue to do the work */ - queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); + schedule_work(&(dreamcastcard->spu_dma_work)); /* Timer may already be running */ if (unlikely(dreamcastcard->timer.data)) { mod_timer(&dreamcastcard->timer, jiffies + 4); @@ -381,7 +378,7 @@ static int snd_aicapcm_pcm_close(struct snd_pcm_substream *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; - flush_workqueue(aica_queue); + flush_work(&(dreamcastcard->spu_dma_work)); if (dreamcastcard->timer.data) del_timer(&dreamcastcard->timer); kfree(dreamcastcard->channel); @@ -633,9 +630,6 @@ static int snd_aica_probe(struct platform_device *devptr) if (unlikely(err < 0)) goto freedreamcast; platform_set_drvdata(devptr, dreamcastcard); - aica_queue = create_workqueue(CARD_NAME); - if (unlikely(!aica_queue)) - goto freedreamcast; snd_printk ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); return 0; @@ -671,10 +665,6 @@ static int __init aica_init(void) static void __exit aica_exit(void) { - /* Destroy the aica kernel thread * - * being extra cautious to check if it exists*/ - if (likely(aica_queue)) - destroy_workqueue(aica_queue); platform_device_unregister(pd); platform_driver_unregister(&snd_aica_driver); /* Kill any sound still playing and reset ARM7 to safe state */ -- 2.1.4

2 1

[alsa-devel] ASoC: sgtl5000: SMALL_POP and loud click noise 5 secs after playback stopped
by Clemens Gruber 07 Jun '16

07 Jun '16

Hi, since commit b101acfabc9377469af3abfb7cb63112da367284 I can hear a loud click / pop noise, always occuring 5 seconds after playback stopped. Hardware: i.MX6Q board with SGTL5000 Kernel: 4.7-rc2 master In the commit history of sgtl5000.h I saw that SGTL5000_SMALL_POP was previously 0 and that Fabio, who changed it to 0, had a similar problem back then. Now with SGTL5000_SMALL_POP at 1, the problem reappeared. Changing it back to 0 fixes the problem for me, but as Axel Lin explained in the commit mentioned above, having 0 as mask leads to not writing anything to the CHIP_REF_CTRL register and the SMALL_POP bit stays as it was. So why does this help? Maybe having SMALL_POP active causes that noise in some circumstances? Then we could add a devicetree property to turn it on or off as needed. Or, the documentation of the SMALL_POP bit in the datasheet is not correct / inverted? Then we could remove that write to the CHIP_REF_CTRL register and leave the SMALL_POP bit at its default value? Axel: Did changing SGTL5000_SMALL_POP to 1 remove a click/pop noise on your board or did you just notice the error with the mask argument of snd_soc_update_bits and fixed it? Thanks, Clemens

2 1

[alsa-devel] [PATCH v6 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by Paul.Handrigan＠cirrus.com 07 Jun '16

07 Jun '16

From: Paul Handrigan <Paul.Handrigan(a)cirrus.com> Initial commit of the Cirrus Logic cs35l33 8V boosted class D amplifier. Signed-off-by: Paul Handrigan <Paul.Handrigan(a)cirrus.com> --- include/sound/cs35l33.h | 48 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1300 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1576 insertions(+) create mode 100644 include/sound/cs35l33.h create mode 100644 sound/soc/codecs/cs35l33.c create mode 100644 sound/soc/codecs/cs35l33.h diff --git a/include/sound/cs35l33.h b/include/sound/cs35l33.h new file mode 100644 index 0000000..b6eadce --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,48 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2016 Cirrus Logic Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __CS35L33_H +#define __CS35L33_H + +struct cs35l33_hg { + bool enable_hg_algo; + unsigned int mem_depth; + unsigned int release_rate; + unsigned int hd_rm; + unsigned int ldo_thld; + unsigned int ldo_path_disable; + unsigned int ldo_entry_delay; + bool vp_hg_auto; + unsigned int vp_hg; + unsigned int vp_hg_rate; + unsigned int vp_hg_va; +}; + +struct cs35l33_pdata { + /* Boost Controller Voltage Setting */ + unsigned int boost_ctl; + + /* Boost Controller Peak Current */ + unsigned int boost_ipk; + + /* Amplifier Drive Select */ + unsigned int amp_drv_sel; + + /* soft volume ramp */ + unsigned int ramp_rate; + + /* IMON adc scale */ + unsigned int imon_adc_scale; + + /* H/G algo configuration */ + struct cs35l33_hg hg_config; +}; + +#endif /* __CS35L33_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cb57f8f..627a380 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -47,6 +47,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_BT_SCO select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC select SND_SOC_CS35L32 if I2C + select SND_SOC_CS35L33 if I2C select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C && INPUT select SND_SOC_CS42L56 if I2C && INPUT @@ -386,6 +387,10 @@ config SND_SOC_CS35L32 tristate "Cirrus Logic CS35L32 CODEC" depends on I2C +config SND_SOC_CS35L33 + tristate "Cirrus Logic CS35L33 CODEC" + depends on I2C + config SND_SOC_CS42L51 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 550a174..c279597 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -36,6 +36,7 @@ snd-soc-arizona-objs := arizona.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs35l32-objs := cs35l32.o +snd-soc-cs35l33-objs := cs35l33.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o @@ -254,6 +255,7 @@ obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS35L32) += snd-soc-cs35l32.o +obj-$(CONFIG_SND_SOC_CS35L33) += snd-soc-cs35l33.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/cs35l33.c b/sound/soc/codecs/cs35l33.c new file mode 100644 index 0000000..da0262e --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1300 @@ +/* + * cs35l33.c -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/workqueue.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <sound/cs35l33.h> +#include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> +#include <linux/regulator/machine.h> +#include <linux/of_gpio.h> +#include <linux/of.h> +#include <linux/of_device.h> +#include <linux/of_irq.h> + +#include "cs35l33.h" + +#define CS35L33_BOOT_DELAY 50 +#define CS35L33_INT_ATTEMPTS 5 + +struct cs35l33_private { + struct snd_soc_codec *codec; + struct cs35l33_pdata pdata; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + bool amp_cal; + int mclk_int; + struct regulator_bulk_data core_supplies[2]; + int num_core_supplies; + bool is_tdm_mode; + bool enable_soft_ramp; +}; + +static const struct reg_default cs35l33_reg[] = { + {CS35L33_PWRCTL1, 0x85}, + {CS35L33_PWRCTL2, 0xFE}, + {CS35L33_CLK_CTL, 0x0C}, + {CS35L33_BST_PEAK_CTL, 0x90}, + {CS35L33_PROTECT_CTL, 0x55}, + {CS35L33_BST_CTL1, 0x00}, + {CS35L33_BST_CTL2, 0x01}, + {CS35L33_ADSP_CTL, 0x00}, + {CS35L33_ADC_CTL, 0xC8}, + {CS35L33_DAC_CTL, 0x14}, + {CS35L33_DIG_VOL_CTL, 0x00}, + {CS35L33_CLASSD_CTL, 0x04}, + {CS35L33_AMP_CTL, 0x90}, + {CS35L33_INT_MASK_1, 0xFF}, + {CS35L33_INT_MASK_2, 0xFF}, + {CS35L33_DIAG_LOCK, 0x00}, + {CS35L33_DIAG_CTRL_1, 0x40}, + {CS35L33_DIAG_CTRL_2, 0x00}, + {CS35L33_HG_MEMLDO_CTL, 0x62}, + {CS35L33_HG_REL_RATE, 0x03}, + {CS35L33_LDO_DEL, 0x12}, + {CS35L33_HG_HEAD, 0x0A}, + {CS35L33_HG_EN, 0x05}, + {CS35L33_TX_VMON, 0x00}, + {CS35L33_TX_IMON, 0x03}, + {CS35L33_TX_VPMON, 0x02}, + {CS35L33_TX_VBSTMON, 0x05}, + {CS35L33_TX_FLAG, 0x06}, + {CS35L33_TX_EN1, 0x00}, + {CS35L33_TX_EN2, 0x00}, + {CS35L33_TX_EN3, 0x00}, + {CS35L33_TX_EN4, 0x00}, + {CS35L33_RX_AUD, 0x40}, + {CS35L33_RX_SPLY, 0x03}, + {CS35L33_RX_ALIVE, 0x04}, + {CS35L33_BST_CTL4, 0x63}, +}; + +static const struct reg_sequence cs35l33_patch[] = { + { 0x00, 0x99, 0 }, + { 0x59, 0x02, 0 }, + { 0x52, 0x30, 0 }, + { 0x39, 0x45, 0 }, + { 0x57, 0x30, 0 }, + { 0x2C, 0x68, 0 }, + { 0x00, 0x00, 0 }, +}; + +static bool cs35l33_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return true; + default: + return false; + } +} + +static bool cs35l33_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + /* these are read only registers */ + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_HG_STATUS: + return false; + default: + return true; + } +} + +static bool cs35l33_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS35L33_DEVID_AB: + case CS35L33_DEVID_CD: + case CS35L33_DEVID_E: + case CS35L33_REV_ID: + case CS35L33_PWRCTL1: + case CS35L33_PWRCTL2: + case CS35L33_CLK_CTL: + case CS35L33_BST_PEAK_CTL: + case CS35L33_PROTECT_CTL: + case CS35L33_BST_CTL1: + case CS35L33_BST_CTL2: + case CS35L33_ADSP_CTL: + case CS35L33_ADC_CTL: + case CS35L33_DAC_CTL: + case CS35L33_DIG_VOL_CTL: + case CS35L33_CLASSD_CTL: + case CS35L33_AMP_CTL: + case CS35L33_INT_MASK_1: + case CS35L33_INT_MASK_2: + case CS35L33_INT_STATUS_1: + case CS35L33_INT_STATUS_2: + case CS35L33_DIAG_LOCK: + case CS35L33_DIAG_CTRL_1: + case CS35L33_DIAG_CTRL_2: + case CS35L33_HG_MEMLDO_CTL: + case CS35L33_HG_REL_RATE: + case CS35L33_LDO_DEL: + case CS35L33_HG_HEAD: + case CS35L33_HG_EN: + case CS35L33_TX_VMON: + case CS35L33_TX_IMON: + case CS35L33_TX_VPMON: + case CS35L33_TX_VBSTMON: + case CS35L33_TX_FLAG: + case CS35L33_TX_EN1: + case CS35L33_TX_EN2: + case CS35L33_TX_EN3: + case CS35L33_TX_EN4: + case CS35L33_RX_AUD: + case CS35L33_RX_SPLY: + case CS35L33_RX_ALIVE: + case CS35L33_BST_CTL4: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(classd_ctl_tlv, 900, 100, 0); +static DECLARE_TLV_DB_SCALE(dac_tlv, -10200, 50, 0); + +static const struct snd_kcontrol_new cs35l33_snd_controls[] = { + + SOC_SINGLE_TLV("SPK Amp Volume", CS35L33_AMP_CTL, + 4, 0x09, 0, classd_ctl_tlv), + SOC_SINGLE_SX_TLV("DAC Volume", CS35L33_DIG_VOL_CTL, + 0, 0x34, 0xE4, dac_tlv), +}; + +static int cs35l33_spkrdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!priv->amp_cal) { + usleep_range(8000, 9000); + priv->amp_cal = true; + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, 0); + dev_dbg(codec->dev, "Amp calibration done\n"); + } + dev_dbg(codec->dev, "Amp turned on\n"); + break; + case SND_SOC_DAPM_POST_PMD: + dev_dbg(codec->dev, "Amp turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + break; + } + + return 0; +} + +static int cs35l33_sdin_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, 0); + val = priv->is_tdm_mode ? 0 : CS35L33_PDN_TDM; + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, val); + dev_dbg(codec->dev, "BST turned on\n"); + break; + case SND_SOC_DAPM_POST_PMU: + dev_dbg(codec->dev, "SDIN turned on\n"); + if (!priv->amp_cal) { + regmap_update_bits(priv->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_CAL, CS35L33_AMP_CAL); + dev_dbg(codec->dev, "Amp calibration started\n"); + usleep_range(10000, 11000); + } + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_PDN_TDM, CS35L33_PDN_TDM); + usleep_range(4000, 4100); + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_BST, CS35L33_PDN_BST); + dev_dbg(codec->dev, "BST and SDIN turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + + } + + return 0; +} + +static int cs35l33_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int mask = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + unsigned int mask2 = CS35L33_SDOUT_3ST_TDM; + unsigned int val, val2; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (priv->is_tdm_mode) { + /* set sdout_3st_i2s and reset pdn_tdm */ + val = CS35L33_SDOUT_3ST_I2S; + /* reset sdout_3st_tdm */ + val2 = 0; + } else { + /* reset sdout_3st_i2s and set pdn_tdm */ + val = CS35L33_PDN_TDM; + /* set sdout_3st_tdm */ + val2 = CS35L33_SDOUT_3ST_TDM; + } + dev_dbg(codec->dev, "SDOUT turned on\n"); + break; + case SND_SOC_DAPM_PRE_PMD: + val = CS35L33_SDOUT_3ST_I2S | CS35L33_PDN_TDM; + val2 = CS35L33_SDOUT_3ST_TDM; + dev_dbg(codec->dev, "SDOUT turned off\n"); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return 0; + } + + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + mask, val); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + mask2, val2); + + return 0; +} + +static const struct snd_soc_dapm_widget cs35l33_dapm_widgets[] = { + + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_OUT_DRV_E("SPKDRV", CS35L33_PWRCTL1, 7, 1, NULL, 0, + cs35l33_spkrdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_AIF_IN_E("SDIN", NULL, 0, CS35L33_PWRCTL2, + 2, 1, cs35l33_sdin_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_INPUT("MON"), + + SND_SOC_DAPM_ADC("VMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VMON_SHIFT, 1), + SND_SOC_DAPM_ADC("IMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_IMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VPMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VPMON_SHIFT, 1), + SND_SOC_DAPM_ADC("VBSTMON", NULL, + CS35L33_PWRCTL2, CS35L33_PDN_VBSTMON_SHIFT, 1), + + SND_SOC_DAPM_AIF_OUT_E("SDOUT", NULL, 0, SND_SOC_NOPM, 0, 0, + cs35l33_sdout_event, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_PRE_PMD), +}; + +static const struct snd_soc_dapm_route cs35l33_audio_map[] = { + {"SDIN", NULL, "CS35L33 Playback"}, + {"SPKDRV", NULL, "SDIN"}, + {"SPK", NULL, "SPKDRV"}, + + {"VMON", NULL, "MON"}, + {"IMON", NULL, "MON"}, + + {"SDOUT", NULL, "VMON"}, + {"SDOUT", NULL, "IMON"}, + {"CS35L33 Capture", NULL, "SDOUT"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vphg_auto_route[] = { + {"SPKDRV", NULL, "VPMON"}, + {"VPMON", NULL, "CS35L33 Playback"}, +}; + +static const struct snd_soc_dapm_route cs35l33_vp_vbst_mon_route[] = { + {"SDOUT", NULL, "VPMON"}, + {"VPMON", NULL, "MON"}, + {"SDOUT", NULL, "VBSTMON"}, + {"VBSTMON", NULL, "MON"}, +}; + +static int cs35l33_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int val; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, 0); + break; + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(priv->regmap, CS35L33_PWRCTL1, + CS35L33_PDN_ALL, CS35L33_PDN_ALL); + regmap_read(priv->regmap, CS35L33_INT_STATUS_2, &val); + usleep_range(1000, 1100); + if (val & CS35L33_PDN_DONE) + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS, CS35L33_MCLKDIS); + break; + case SND_SOC_BIAS_OFF: + break; + default: + return -EINVAL; + } + + dapm->bias_level = level; + + return 0; +} + +struct cs35l33_mclk_div { + int mclk; + int srate; + u8 adsp_rate; + u8 int_fs_ratio; +}; + +static const struct cs35l33_mclk_div cs35l33_mclk_coeffs[] = { + /* MCLK, Sample Rate, adsp_rate, int_fs_ratio */ + {5644800, 11025, 0x4, CS35L33_INT_FS_RATE}, + {5644800, 22050, 0x8, CS35L33_INT_FS_RATE}, + {5644800, 44100, 0xC, CS35L33_INT_FS_RATE}, + + {6000000, 8000, 0x1, 0}, + {6000000, 11025, 0x2, 0}, + {6000000, 11029, 0x3, 0}, + {6000000, 12000, 0x4, 0}, + {6000000, 16000, 0x5, 0}, + {6000000, 22050, 0x6, 0}, + {6000000, 22059, 0x7, 0}, + {6000000, 24000, 0x8, 0}, + {6000000, 32000, 0x9, 0}, + {6000000, 44100, 0xA, 0}, + {6000000, 44118, 0xB, 0}, + {6000000, 48000, 0xC, 0}, + + {6144000, 8000, 0x1, CS35L33_INT_FS_RATE}, + {6144000, 12000, 0x4, CS35L33_INT_FS_RATE}, + {6144000, 16000, 0x5, CS35L33_INT_FS_RATE}, + {6144000, 24000, 0x8, CS35L33_INT_FS_RATE}, + {6144000, 32000, 0x9, CS35L33_INT_FS_RATE}, + {6144000, 48000, 0xC, CS35L33_INT_FS_RATE}, +}; + +static int cs35l33_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs35l33_mclk_coeffs); i++) { + if (cs35l33_mclk_coeffs[i].mclk == mclk && + cs35l33_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; +} + +static int cs35l33_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, CS35L33_MS_MASK); + dev_dbg(codec->dev, "Audio port in master mode\n"); + break; + case SND_SOC_DAIFMT_CBS_CFS: + regmap_update_bits(priv->regmap, CS35L33_ADSP_CTL, + CS35L33_MS_MASK, 0); + dev_dbg(codec->dev, "Audio port in slave mode\n"); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + /* + * tdm mode in cs35l33 resembles dsp-a mode very + * closely, it is dsp-a with fsync shifted left by half bclk + */ + priv->is_tdm_mode = true; + dev_dbg(codec->dev, "Audio port in TDM mode\n"); + break; + case SND_SOC_DAIFMT_I2S: + priv->is_tdm_mode = false; + dev_dbg(codec->dev, "Audio port in I2S mode\n"); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int cs35l33_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + int sample_size = params_width(params); + int coeff = cs35l33_get_mclk_coeff(priv->mclk_int, params_rate(params)); + + if (coeff < 0) + return coeff; + + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_ADSP_FS | CS35L33_INT_FS_RATE, + cs35l33_mclk_coeffs[coeff].int_fs_ratio + | cs35l33_mclk_coeffs[coeff].adsp_rate); + + if (priv->is_tdm_mode) { + sample_size = (sample_size / 8) - 1; + if (sample_size > 2) + sample_size = 2; + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_AUDIN_RX_DEPTH, + sample_size << CS35L33_AUDIN_RX_DEPTH_SHIFT); + } + + dev_dbg(codec->dev, "sample rate=%d, bits per sample=%d\n", + params_rate(params), params_width(params)); + + return 0; +} + +static const unsigned int cs35l33_src_rates[] = { + 8000, 11025, 11029, 12000, 16000, 22050, + 22059, 24000, 32000, 44100, 44118, 48000 +}; + +static const struct snd_pcm_hw_constraint_list cs35l33_constraints = { + .count = ARRAY_SIZE(cs35l33_src_rates), + .list = cs35l33_src_rates, +}; + +static int cs35l33_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &cs35l33_constraints); + return 0; +} + +static int cs35l33_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (tristate) { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, CS35L33_SDOUT_3ST_I2S); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, CS35L33_SDOUT_3ST_TDM); + } else { + regmap_update_bits(priv->regmap, CS35L33_PWRCTL2, + CS35L33_SDOUT_3ST_I2S, 0); + regmap_update_bits(priv->regmap, CS35L33_CLK_CTL, + CS35L33_SDOUT_3ST_TDM, 0); + } + + return 0; +} + +static int cs35l33_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg, bit_pos, i; + int slot, slot_num; + + if (slot_width != 8) + return -EINVAL; + + /* scan rx_mask for aud slot */ + slot = ffs(rx_mask) - 1; + if (slot >= 0) { + regmap_update_bits(priv->regmap, CS35L33_RX_AUD, + CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "Audio starts from slots %d", slot); + } + + /* + * scan tx_mask: vmon(2 slots); imon (2 slots); + * vpmon (1 slot) vbstmon (1 slot) + */ + slot = ffs(tx_mask) - 1; + slot_num = 0; + + for (i = 0; i < 2 ; i++) { + /* disable vpmon/vbstmon: enable later if set in tx_mask */ + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON + i, + CS35L33_X_STATE | CS35L33_X_LOC, CS35L33_X_STATE + | CS35L33_X_LOC); + } + + /* disconnect {vp,vbst}_mon routes: eanble later if set in tx_mask*/ + snd_soc_dapm_del_routes(dapm, cs35l33_vp_vbst_mon_route, + ARRAY_SIZE(cs35l33_vp_vbst_mon_route)); + + while (slot >= 0) { + /* configure VMON_TX_LOC */ + if (slot_num == 0) { + regmap_update_bits(priv->regmap, CS35L33_TX_VMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "VMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure IMON_TX_LOC */ + if (slot_num == 3) { + regmap_update_bits(priv->regmap, CS35L33_TX_IMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + dev_dbg(codec->dev, "IMON enabled in slots %d-%d", + slot, slot + 1); + } + + /* configure VPMON_TX_LOC */ + if (slot_num == 4) { + regmap_update_bits(priv->regmap, CS35L33_TX_VPMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[0], 2); + dev_dbg(codec->dev, "VPMON enabled in slots %d", slot); + } + + /* configure VBSTMON_TX_LOC */ + if (slot_num == 5) { + regmap_update_bits(priv->regmap, CS35L33_TX_VBSTMON, + CS35L33_X_STATE | CS35L33_X_LOC, slot); + snd_soc_dapm_add_routes(dapm, + &cs35l33_vp_vbst_mon_route[2], 2); + dev_dbg(codec->dev, + "VBSTMON enabled in slots %d", slot); + } + + /* Enable the relevant tx slot */ + reg = CS35L33_TX_EN4 - (slot/8); + bit_pos = slot - ((slot / 8) * (8)); + regmap_update_bits(priv->regmap, reg, + 1 << bit_pos, 1 << bit_pos); + + tx_mask &= ~(1 << slot); + slot = ffs(tx_mask) - 1; + slot_num++; + } + + return 0; +} + +static int cs35l33_codec_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + switch (freq) { + case CS35L33_MCLK_5644: + case CS35L33_MCLK_6: + case CS35L33_MCLK_6144: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, 0); + cs35l33->mclk_int = freq; + break; + case CS35L33_MCLK_11289: + case CS35L33_MCLK_12: + case CS35L33_MCLK_12288: + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIV2, CS35L33_MCLKDIV2); + cs35l33->mclk_int = freq/2; + break; + default: + cs35l33->mclk_int = 0; + return -EINVAL; + } + + dev_dbg(codec->dev, "external mclk freq=%d, internal mclk freq=%d\n", + freq, cs35l33->mclk_int); + + return 0; +} + +static const struct snd_soc_dai_ops cs35l33_ops = { + .startup = cs35l33_pcm_startup, + .set_tristate = cs35l33_set_tristate, + .set_fmt = cs35l33_set_dai_fmt, + .hw_params = cs35l33_pcm_hw_params, + .set_tdm_slot = cs35l33_set_tdm_slot, +}; + +static struct snd_soc_dai_driver cs35l33_dai = { + .name = "cs35l33-dai", + .id = 0, + .playback = { + .stream_name = "CS35L33 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .capture = { + .stream_name = "CS35L33 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = CS35L33_RATES, + .formats = CS35L33_FORMATS, + }, + .ops = &cs35l33_ops, + .symmetric_rates = 1, +}; + +static int cs35l33_set_hg_data(struct snd_soc_codec *codec, + struct cs35l33_pdata *pdata) +{ + struct cs35l33_hg *hg_config = &pdata->hg_config; + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs35l33_private *priv = snd_soc_codec_get_drvdata(codec); + + if (hg_config->enable_hg_algo) { + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_MEM_DEPTH_MASK, + hg_config->mem_depth << CS35L33_MEM_DEPTH_SHIFT); + regmap_write(priv->regmap, CS35L33_HG_REL_RATE, + hg_config->release_rate); + regmap_update_bits(priv->regmap, CS35L33_HG_HEAD, + CS35L33_HD_RM_MASK, + hg_config->hd_rm << CS35L33_HD_RM_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_THLD_MASK, + hg_config->ldo_thld << CS35L33_LDO_THLD_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_MEMLDO_CTL, + CS35L33_LDO_DISABLE_MASK, + hg_config->ldo_path_disable << + CS35L33_LDO_DISABLE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_LDO_ENTRY_DELAY_MASK, + hg_config->ldo_entry_delay << + CS35L33_LDO_ENTRY_DELAY_SHIFT); + if (hg_config->vp_hg_auto) { + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_AUTO_MASK, + CS35L33_VP_HG_AUTO_MASK); + snd_soc_dapm_add_routes(dapm, cs35l33_vphg_auto_route, + ARRAY_SIZE(cs35l33_vphg_auto_route)); + } + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_VP_HG_MASK, + hg_config->vp_hg << CS35L33_VP_HG_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_RATE_MASK, + hg_config->vp_hg_rate << CS35L33_VP_HG_RATE_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_LDO_DEL, + CS35L33_VP_HG_VA_MASK, + hg_config->vp_hg_va << CS35L33_VP_HG_VA_SHIFT); + regmap_update_bits(priv->regmap, CS35L33_HG_EN, + CS35L33_CLASS_HG_EN_MASK, CS35L33_CLASS_HG_EN_MASK); + } + return 0; +} + +static int cs35l33_probe(struct snd_soc_codec *codec) +{ + struct cs35l33_private *cs35l33 = snd_soc_codec_get_drvdata(codec); + + cs35l33->codec = codec; + pm_runtime_get_sync(codec->dev); + + regmap_update_bits(cs35l33->regmap, CS35L33_PROTECT_CTL, + CS35L33_ALIVE_WD_DIS, 0x8); + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL2, + CS35L33_ALIVE_WD_DIS2, + CS35L33_ALIVE_WD_DIS2); + + /* Set Platform Data */ + regmap_update_bits(cs35l33->regmap, CS35L33_BST_CTL1, + CS35L33_BST_CTL_MASK, cs35l33->pdata.boost_ctl); + regmap_update_bits(cs35l33->regmap, CS35L33_CLASSD_CTL, + CS35L33_AMP_DRV_SEL_MASK, + cs35l33->pdata.amp_drv_sel << CS35L33_AMP_DRV_SEL_SHIFT); + + if (cs35l33->pdata.boost_ipk) + regmap_write(cs35l33->regmap, CS35L33_BST_PEAK_CTL, + cs35l33->pdata.boost_ipk); + + if (cs35l33->enable_soft_ramp) { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, CS35L33_DIGSFT); + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DSR_RATE, cs35l33->pdata.ramp_rate); + } else { + snd_soc_update_bits(codec, CS35L33_DAC_CTL, + CS35L33_DIGSFT, 0); + } + + /* update IMON scaling rate if different from default of 0x8 */ + if (cs35l33->pdata.imon_adc_scale != 0x8) + snd_soc_update_bits(codec, CS35L33_ADC_CTL, + CS35L33_IMON_SCALE, cs35l33->pdata.imon_adc_scale); + + cs35l33_set_hg_data(codec, &(cs35l33->pdata)); + + /* + * unmask important interrupts that causes the chip to enter + * speaker safe mode and hence deserves user attention + */ + regmap_update_bits(cs35l33->regmap, CS35L33_INT_MASK_1, + CS35L33_M_OTE | CS35L33_M_OTW | CS35L33_M_AMP_SHORT | + CS35L33_M_CAL_ERR, 0); + + pm_runtime_put_sync(codec->dev); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .set_bias_level = cs35l33_set_bias_level, + .set_sysclk = cs35l33_codec_set_sysclk, + + .dapm_widgets = cs35l33_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs35l33_dapm_widgets), + .dapm_routes = cs35l33_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs35l33_audio_map), + .controls = cs35l33_snd_controls, + .num_controls = ARRAY_SIZE(cs35l33_snd_controls), + + .idle_bias_off = true, +}; + +static const struct regmap_config cs35l33_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS35L33_MAX_REGISTER, + .reg_defaults = cs35l33_reg, + .num_reg_defaults = ARRAY_SIZE(cs35l33_reg), + .volatile_reg = cs35l33_volatile_register, + .readable_reg = cs35l33_readable_register, + .writeable_reg = cs35l33_writeable_register, + .cache_type = REGCACHE_RBTREE, + .use_single_rw = true, +}; + +static int cs35l33_runtime_resume(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + int ret; + + dev_dbg(dev, "%s\n", __func__); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable core supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, false); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + + ret = regcache_sync(cs35l33->regmap); + if (ret != 0) { + dev_err(dev, "Failed to restore register cache\n"); + goto err; + } + + return 0; + +err: + regcache_cache_only(cs35l33->regmap, true); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return ret; +} + +static int cs35l33_runtime_suspend(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + dev_dbg(dev, "%s\n", __func__); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + regcache_cache_only(cs35l33->regmap, true); + regcache_mark_dirty(cs35l33->regmap); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + + return 0; +} + +static const struct dev_pm_ops cs35l33_pm_ops = { + SET_RUNTIME_PM_OPS(cs35l33_runtime_suspend, + cs35l33_runtime_resume, + NULL) +}; + +static int cs35l33_get_hg_data(const struct device_node *np, + struct cs35l33_pdata *pdata) +{ + struct device_node *hg; + struct cs35l33_hg *hg_config = &pdata->hg_config; + u32 val32; + + hg = of_get_child_by_name(np, "hg-algo"); + hg_config->enable_hg_algo = hg ? true : false; + + if (hg_config->enable_hg_algo) { + if (of_property_read_u32(hg, "mem-depth", &val32) >= 0) + hg_config->mem_depth = val32; + if (of_property_read_u32(hg, "release-rate", &val32) >= 0) + hg_config->release_rate = val32; + if (of_property_read_u32(hg, "hd-rm", &val32) >= 0) + hg_config->hd_rm = val32; + if (of_property_read_u32(hg, "ldo-thld", &val32) >= 0) + hg_config->ldo_thld = val32; + if (of_property_read_u32(hg, "ldo-path-disable", &val32) >= 0) + hg_config->ldo_path_disable = val32; + if (of_property_read_u32(hg, "ldo-entry-delay", &val32) >= 0) + hg_config->ldo_entry_delay = val32; + hg_config->vp_hg_auto = of_property_read_bool(hg, "vp-hg-auto"); + if (of_property_read_u32(hg, "vp-hg", &val32) >= 0) + hg_config->vp_hg = val32; + if (of_property_read_u32(hg, "vp-hg-rate", &val32) >= 0) + hg_config->vp_hg_rate = val32; + if (of_property_read_u32(hg, "vp-hg-va", &val32) >= 0) + hg_config->vp_hg_va = val32; + } + + of_node_put(hg); + + return 0; +} + +static irqreturn_t cs35l33_irq_thread(int irq, void *data) +{ + struct cs35l33_private *cs35l33 = data; + struct snd_soc_codec *codec = cs35l33->codec; + unsigned int sticky_val1, sticky_val2, current_val, mask1, mask2; + int i; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_2, &mask2); + regmap_read(cs35l33->regmap, CS35L33_INT_MASK_1, &mask1); + + /* Check to see if the unmasked bits are active, + * if not then exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + return IRQ_NONE; + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_crit(codec->dev, "Amp short error\n"); + if (!(current_val & CS35L33_AMP_SHORT)) { + dev_dbg(codec->dev, + "Amp short error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, + CS35L33_AMP_SHORT_RLS, + CS35L33_AMP_SHORT_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_AMP_SHORT_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_CAL_ERR) { + dev_err(codec->dev, "Cal error\n"); + + /* redo the calibration in next power up */ + cs35l33->amp_cal = false; + + if (!(current_val & CS35L33_CAL_ERR)) { + dev_dbg(codec->dev, "Cal error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + CS35L33_CAL_ERR_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_CAL_ERR_RLS, + 0); + } + } + + if (sticky_val1 & CS35L33_OTE) { + dev_crit(codec->dev, "Over temperature error\n"); + if (!(current_val & CS35L33_OTE)) { + dev_dbg(codec->dev, + "Over temperature error release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, + CS35L33_OTE_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTE_RLS, 0); + } + } + + if (sticky_val1 & CS35L33_OTW) { + dev_err(codec->dev, "Over temperature warning\n"); + if (!(current_val & CS35L33_OTW)) { + dev_dbg(codec->dev, + "Over temperature warning release\n"); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, + CS35L33_OTW_RLS); + regmap_update_bits(cs35l33->regmap, + CS35L33_AMP_CTL, CS35L33_OTW_RLS, 0); + } + } + if (CS35L33_ALIVE_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: ADSPCLK Interrupt\n"); + + if (CS35L33_MCLK_ERR & sticky_val1) + dev_err(codec->dev, "ERROR: MCLK Interrupt\n"); + + if (CS35L33_VMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VMON Overflow Interrupt\n"); + + if (CS35L33_IMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: IMON Overflow Interrupt\n"); + + if (CS35L33_VPMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + if (CS35L33_VBSTMON_OVFL & sticky_val2) + dev_err(codec->dev, + "ERROR: VPMON Overflow Interrupt\n"); + + /* Make sure that the bits are cleared */ + for (i = 0; i < CS35L33_INT_ATTEMPTS; i++) { + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &sticky_val1); + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_2, + &sticky_val2); + + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + break; + } + + if (i == CS35L33_INT_ATTEMPTS) + dev_warn(codec->dev, "WARN: Interrupts may not be cleared\n"); + + return IRQ_HANDLED; +} + +static const char * const cs35l33_core_supplies[] = { + "VA", + "VP", +}; + +static int cs35l33_of_get_pdata(struct device *dev, + struct cs35l33_private *cs35l33) +{ + struct device_node *np = dev->of_node; + struct cs35l33_pdata *pdata = &cs35l33->pdata; + u32 val32; + + if (!np) + return 0; + + if (of_property_read_u32(np, "boost-ctl", &val32) >= 0) { + pdata->boost_ctl = val32; + pdata->amp_drv_sel = 1; + } + + if (of_property_read_u32(np, "ramp-rate", &val32) >= 0) { + pdata->ramp_rate = val32; + cs35l33->enable_soft_ramp = true; + } + + if (of_property_read_u32(np, "boost-ipk", &val32) >= 0) + pdata->boost_ipk = val32; + + if (of_property_read_u32(np, "imon-adc-scale", &val32) >= 0) { + if ((val32 == 0x0) || (val32 == 0x7) || (val32 == 0x6)) + pdata->imon_adc_scale = val32; + else + /* use default value */ + pdata->imon_adc_scale = 0x8; + } else { + /* use default value */ + pdata->imon_adc_scale = 0x8; + } + + cs35l33_get_hg_data(np, pdata); + + return 0; +} + +static int cs35l33_i2c_probe(struct i2c_client *i2c_client, + const struct i2c_device_id *id) +{ + struct cs35l33_private *cs35l33; + struct cs35l33_pdata *pdata = dev_get_platdata(&i2c_client->dev); + int ret, devid, i; + unsigned int reg; + + cs35l33 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs35l33_private), + GFP_KERNEL); + if (!cs35l33) + return -ENOMEM; + + i2c_set_clientdata(i2c_client, cs35l33); + cs35l33->regmap = devm_regmap_init_i2c(i2c_client, &cs35l33_regmap); + if (IS_ERR(cs35l33->regmap)) { + ret = PTR_ERR(cs35l33->regmap); + dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + + regcache_cache_only(cs35l33->regmap, true); + + for (i = 0; i < ARRAY_SIZE(cs35l33_core_supplies); i++) + cs35l33->core_supplies[i].supply + = cs35l33_core_supplies[i]; + cs35l33->num_core_supplies = ARRAY_SIZE(cs35l33_core_supplies); + + ret = devm_regulator_bulk_get(&i2c_client->dev, + cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to request core supplies: %d\n", + ret); + return ret; + } + + if (pdata) { + cs35l33->pdata = *pdata; + } else { + cs35l33_of_get_pdata(&i2c_client->dev, cs35l33); + pdata = &cs35l33->pdata; + } + + ret = request_threaded_irq(i2c_client->irq, NULL, cs35l33_irq_thread, + IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs35l33", cs35l33); + if (ret != 0) + dev_err(&i2c_client->dev, "Failed to request IRQ: %d\n", ret); + + /* We could issue !RST or skip it based on AMP topology */ + cs35l33->reset_gpio = devm_gpiod_get_optional(&i2c_client->dev, + "reset-gpios", GPIOD_OUT_HIGH); + + if (PTR_ERR(cs35l33->reset_gpio) == -ENOENT) { + dev_warn(&i2c_client->dev, + "%s WARNING: No reset gpio assigned\n", __func__); + } else if (IS_ERR(cs35l33->reset_gpio)) { + dev_err(&i2c_client->dev, "%s ERROR: Can't get reset GPIO\n", + __func__); + return PTR_ERR(cs35l33->reset_gpio); + } + + ret = regulator_bulk_enable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + if (ret != 0) { + dev_err(&i2c_client->dev, + "Failed to enable core supplies: %d\n", + ret); + goto err_irq; + } + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 1); + + msleep(CS35L33_BOOT_DELAY); + regcache_cache_only(cs35l33->regmap, false); + + /* initialize codec */ + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_AB, &reg); + devid = (reg & 0xFF) << 12; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_CD, &reg); + devid |= (reg & 0xFF) << 4; + ret = regmap_read(cs35l33->regmap, CS35L33_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS35L33_CHIP_ID) { + dev_err(&i2c_client->dev, + "CS35L33 Device ID (%X). Expected ID %X\n", + devid, CS35L33_CHIP_ID); + goto err_enable; + } + + ret = regmap_read(cs35l33->regmap, CS35L33_REV_ID, &reg); + if (ret < 0) { + dev_err(&i2c_client->dev, "Get Revision ID failed\n"); + goto err_enable; + } + + dev_info(&i2c_client->dev, + "Cirrus Logic CS35L33, Revision: %02X\n", ret & 0xFF); + + ret = regmap_register_patch(cs35l33->regmap, + cs35l33_patch, ARRAY_SIZE(cs35l33_patch)); + if (ret < 0) { + dev_err(&i2c_client->dev, + "Error in applying regmap patch: %d\n", ret); + goto err_enable; + } + + /* disable mclk and tdm */ + regmap_update_bits(cs35l33->regmap, CS35L33_CLK_CTL, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM, + CS35L33_MCLKDIS | CS35L33_SDOUT_3ST_TDM); + + pm_runtime_set_autosuspend_delay(&i2c_client->dev, 100); + pm_runtime_use_autosuspend(&i2c_client->dev); + pm_runtime_set_active(&i2c_client->dev); + pm_runtime_enable(&i2c_client->dev); + + ret = snd_soc_register_codec(&i2c_client->dev, + &soc_codec_dev_cs35l33, &cs35l33_dai, 1); + if (ret < 0) { + dev_err(&i2c_client->dev, "%s: Register codec failed\n", + __func__); + goto err_irq; + } + + return 0; + +err_enable: + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); +err_irq: + free_irq(i2c_client->irq, cs35l33); + + return ret; +} + +static int cs35l33_i2c_remove(struct i2c_client *client) +{ + struct cs35l33_private *cs35l33 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + if (cs35l33->reset_gpio) + gpiod_set_value_cansleep(cs35l33->reset_gpio, 0); + + pm_runtime_disable(&client->dev); + regulator_bulk_disable(cs35l33->num_core_supplies, + cs35l33->core_supplies); + free_irq(client->irq, cs35l33); + + return 0; +} + +static const struct of_device_id cs35l33_of_match[] = { + { .compatible = "cirrus,cs35l33", }, + {}, +}; +MODULE_DEVICE_TABLE(of, cs35l33_of_match); + +static const struct i2c_device_id cs35l33_id[] = { + {"cs35l33", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs35l33_id); + +static struct i2c_driver cs35l33_i2c_driver = { + .driver = { + .name = "cs35l33", + .owner = THIS_MODULE, + .pm = &cs35l33_pm_ops, + .of_match_table = cs35l33_of_match, + + }, + .id_table = cs35l33_id, + .probe = cs35l33_i2c_probe, + .remove = cs35l33_i2c_remove, + +}; +module_i2c_driver(cs35l33_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS35L33 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <paul.handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs35l33.h b/sound/soc/codecs/cs35l33.h new file mode 100644 index 0000000..c045737 --- /dev/null +++ b/sound/soc/codecs/cs35l33.h @@ -0,0 +1,221 @@ +/* + * cs35l33.h -- CS35L33 ALSA SoC audio driver + * + * Copyright 2016 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <paul.handrigan(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS35L33_H__ +#define __CS35L33_H__ + +#define CS35L33_CHIP_ID 0x00035A33 +#define CS35L33_DEVID_AB 0x01 /* Device ID A & B [RO] */ +#define CS35L33_DEVID_CD 0x02 /* Device ID C & D [RO] */ +#define CS35L33_DEVID_E 0x03 /* Device ID E [RO] */ +#define CS35L33_FAB_ID 0x04 /* Fab ID [RO] */ +#define CS35L33_REV_ID 0x05 /* Revision ID [RO] */ +#define CS35L33_PWRCTL1 0x06 /* Power Ctl 1 */ +#define CS35L33_PWRCTL2 0x07 /* Power Ctl 2 */ +#define CS35L33_CLK_CTL 0x08 /* Clock Ctl */ +#define CS35L33_BST_PEAK_CTL 0x09 /* Max Current for Boost */ +#define CS35L33_PROTECT_CTL 0x0A /* Amp Protection Parameters */ +#define CS35L33_BST_CTL1 0x0B /* Boost Converter CTL1 */ +#define CS35L33_BST_CTL2 0x0C /* Boost Converter CTL2 */ +#define CS35L33_ADSP_CTL 0x0D /* Serial Port Control */ +#define CS35L33_ADC_CTL 0x0E /* ADC Control */ +#define CS35L33_DAC_CTL 0x0F /* DAC Control */ +#define CS35L33_DIG_VOL_CTL 0x10 /* Digital Volume CTL */ +#define CS35L33_CLASSD_CTL 0x11 /* Class D Amp CTL */ +#define CS35L33_AMP_CTL 0x12 /* Amp Gain/Protecton Release CTL */ +#define CS35L33_INT_MASK_1 0x13 /* Interrupt Mask 1 */ +#define CS35L33_INT_MASK_2 0x14 /* Interrupt Mask 2 */ +#define CS35L33_INT_STATUS_1 0x15 /* Interrupt Status 1 [RO] */ +#define CS35L33_INT_STATUS_2 0x16 /* Interrupt Status 2 [RO] */ +#define CS35L33_DIAG_LOCK 0x17 /* Diagnostic Mode Register Lock */ +#define CS35L33_DIAG_CTRL_1 0x18 /* Diagnostic Mode Register Control */ +#define CS35L33_DIAG_CTRL_2 0x19 /* Diagnostic Mode Register Control 2 */ +#define CS35L33_HG_MEMLDO_CTL 0x23 /* H/G Memory/LDO CTL */ +#define CS35L33_HG_REL_RATE 0x24 /* H/G Release Rate */ +#define CS35L33_LDO_DEL 0x25 /* LDO Entry Delay/VPhg Control 1 */ +#define CS35L33_HG_HEAD 0x29 /* H/G Headroom */ +#define CS35L33_HG_EN 0x2A /* H/G Enable/VPhg CNT2 */ +#define CS35L33_TX_VMON 0x2D /* TDM TX Control 1 (VMON) */ +#define CS35L33_TX_IMON 0x2E /* TDM TX Control 2 (IMON) */ +#define CS35L33_TX_VPMON 0x2F /* TDM TX Control 3 (VPMON) */ +#define CS35L33_TX_VBSTMON 0x30 /* TDM TX Control 4 (VBSTMON) */ +#define CS35L33_TX_FLAG 0x31 /* TDM TX Control 5 (FLAG) */ +#define CS35L33_TX_EN1 0x32 /* TDM TX Enable 1 */ +#define CS35L33_TX_EN2 0x33 /* TDM TX Enable 2 */ +#define CS35L33_TX_EN3 0x34 /* TDM TX Enable 3 */ +#define CS35L33_TX_EN4 0x35 /* TDM TX Enable 4 */ +#define CS35L33_RX_AUD 0x36 /* TDM RX Control 1 */ +#define CS35L33_RX_SPLY 0x37 /* TDM RX Control 2 */ +#define CS35L33_RX_ALIVE 0x38 /* TDM RX Control 3 */ +#define CS35L33_BST_CTL4 0x39 /* Boost Converter Control 4 */ +#define CS35L33_HG_STATUS 0x3F /* H/G Status */ +#define CS35L33_MAX_REGISTER 0x59 + +#define CS35L33_MCLK_5644 5644800 +#define CS35L33_MCLK_6144 6144000 +#define CS35L33_MCLK_6 6000000 +#define CS35L33_MCLK_11289 11289600 +#define CS35L33_MCLK_12 12000000 +#define CS35L33_MCLK_12288 12288000 + +/* CS35L33_PWRCTL1 */ +#define CS35L33_PDN_AMP (1 << 7) +#define CS35L33_PDN_BST (1 << 2) +#define CS35L33_PDN_ALL 1 + +/* CS35L33_PWRCTL2 */ +#define CS35L33_PDN_VMON_SHIFT 7 +#define CS35L33_PDN_VMON (1 << CS35L33_PDN_VMON_SHIFT) +#define CS35L33_PDN_IMON_SHIFT 6 +#define CS35L33_PDN_IMON (1 << CS35L33_PDN_IMON_SHIFT) +#define CS35L33_PDN_VPMON_SHIFT 5 +#define CS35L33_PDN_VPMON (1 << CS35L33_PDN_VPMON_SHIFT) +#define CS35L33_PDN_VBSTMON_SHIFT 4 +#define CS35L33_PDN_VBSTMON (1 << CS35L33_PDN_VBSTMON_SHIFT) +#define CS35L33_SDOUT_3ST_I2S_SHIFT 3 +#define CS35L33_SDOUT_3ST_I2S (1 << CS35L33_SDOUT_3ST_I2S_SHIFT) +#define CS35L33_PDN_SDIN_SHIFT 2 +#define CS35L33_PDN_SDIN (1 << CS35L33_PDN_SDIN_SHIFT) +#define CS35L33_PDN_TDM_SHIFT 1 +#define CS35L33_PDN_TDM (1 << CS35L33_PDN_TDM_SHIFT) + +/* CS35L33_CLK_CTL */ +#define CS35L33_MCLKDIS (1 << 7) +#define CS35L33_MCLKDIV2 (1 << 6) +#define CS35L33_SDOUT_3ST_TDM (1 << 5) +#define CS35L33_INT_FS_RATE (1 << 4) +#define CS35L33_ADSP_FS 0xF + +/* CS35L33_PROTECT_CTL */ +#define CS35L33_ALIVE_WD_DIS (3 << 2) + +/* CS35L33_BST_CTL1 */ +#define CS35L33_BST_CTL_SRC (1 << 6) +#define CS35L33_BST_CTL_SHIFT (1 << 5) +#define CS35L33_BST_CTL_MASK 0x3F + +/* CS35L33_BST_CTL2 */ +#define CS35L33_TDM_WD_SEL (1 << 4) +#define CS35L33_ALIVE_WD_DIS2 (1 << 3) +#define CS35L33_VBST_SR_STEP 0x3 + +/* CS35L33_ADSP_CTL */ +#define CS35L33_ADSP_DRIVE (1 << 7) +#define CS35L33_MS_MASK (1 << 6) +#define CS35L33_SDIN_LOC (3 << 4) +#define CS35L33_ALIVE_RATE 0x3 + +/* CS35L33_ADC_CTL */ +#define CS35L33_INV_VMON (1 << 7) +#define CS35L33_INV_IMON (1 << 6) +#define CS35L33_ADC_NOTCH_DIS (1 << 5) +#define CS35L33_IMON_SCALE 0xF + +/* CS35L33_DAC_CTL */ +#define CS35L33_INV_DAC (1 << 7) +#define CS35L33_DAC_NOTCH_DIS (1 << 5) +#define CS35L33_DIGSFT (1 << 4) +#define CS35L33_DSR_RATE 0xF + +/* CS35L33_CLASSD_CTL */ +#define CS35L33_AMP_SD (1 << 6) +#define CS35L33_AMP_DRV_SEL_SRC (1 << 5) +#define CS35L33_AMP_DRV_SEL_MASK 0x10 +#define CS35L33_AMP_DRV_SEL_SHIFT 4 +#define CS35L33_AMP_CAL (1 << 3) +#define CS35L33_GAIN_CHG_ZC_MASK 0x04 +#define CS35L33_GAIN_CHG_ZC_SHIFT 2 +#define CS35L33_CLASS_D_CTL_MASK 0x3F + +/* CS35L33_AMP_CTL */ +#define CS35L33_AMP_GAIN 0xF0 +#define CS35L33_CAL_ERR_RLS (1 << 3) +#define CS35L33_AMP_SHORT_RLS (1 << 2) +#define CS35L33_OTW_RLS (1 << 1) +#define CS35L33_OTE_RLS 1 + +/* CS35L33_INT_MASK_1 */ +#define CS35L33_M_CAL_ERR_SHIFT 6 +#define CS35L33_M_CAL_ERR (1 << CS35L33_M_CAL_ERR_SHIFT) +#define CS35L33_M_ALIVE_ERR_SHIFT 5 +#define CS35L33_M_ALIVE_ERR (1 << CS35L33_M_ALIVE_ERR_SHIFT) +#define CS35L33_M_AMP_SHORT_SHIFT 2 +#define CS35L33_M_AMP_SHORT (1 << CS35L33_M_AMP_SHORT_SHIFT) +#define CS35L33_M_OTW_SHIFT 1 +#define CS35L33_M_OTW (1 << CS35L33_M_OTW_SHIFT) +#define CS35L33_M_OTE_SHIFT 0 +#define CS35L33_M_OTE (1 << CS35L33_M_OTE_SHIFT) + +/* CS35L33_INT_STATUS_1 */ +#define CS35L33_CAL_ERR (1 << 6) +#define CS35L33_ALIVE_ERR (1 << 5) +#define CS35L33_ADSPCLK_ERR (1 << 4) +#define CS35L33_MCLK_ERR (1 << 3) +#define CS35L33_AMP_SHORT (1 << 2) +#define CS35L33_OTW (1 << 1) +#define CS35L33_OTE (1 << 0) + +/* CS35L33_INT_STATUS_2 */ +#define CS35L33_VMON_OVFL (1 << 7) +#define CS35L33_IMON_OVFL (1 << 6) +#define CS35L33_VPMON_OVFL (1 << 5) +#define CS35L33_VBSTMON_OVFL (1 << 4) +#define CS35L33_PDN_DONE 1 + +/* CS35L33_BST_CTL4 */ +#define CS35L33_BST_RGS 0x70 +#define CS35L33_BST_COEFF3 0xF + +/* CS35L33_HG_MEMLDO_CTL */ +#define CS35L33_MEM_DEPTH_SHIFT 5 +#define CS35L33_MEM_DEPTH_MASK (0x3 << CS35L33_MEM_DEPTH_SHIFT) +#define CS35L33_LDO_THLD_SHIFT 1 +#define CS35L33_LDO_THLD_MASK (0xF << CS35L33_LDO_THLD_SHIFT) +#define CS35L33_LDO_DISABLE_SHIFT 0 +#define CS35L33_LDO_DISABLE_MASK (0x1 << CS35L33_LDO_DISABLE_SHIFT) + +/* CS35L33_LDO_DEL */ +#define CS35L33_VP_HG_VA_SHIFT 5 +#define CS35L33_VP_HG_VA_MASK (0x7 << CS35L33_VP_HG_VA_SHIFT) +#define CS35L33_LDO_ENTRY_DELAY_SHIFT 2 +#define CS35L33_LDO_ENTRY_DELAY_MASK (0x7 << CS35L33_LDO_ENTRY_DELAY_SHIFT) +#define CS35L33_VP_HG_RATE_SHIFT 0 +#define CS35L33_VP_HG_RATE_MASK (0x3 << CS35L33_VP_HG_RATE_SHIFT) + +/* CS35L33_HG_HEAD */ +#define CS35L33_HD_RM_SHIFT 0 +#define CS35L33_HD_RM_MASK (0x7F << CS35L33_HD_RM_SHIFT) + +/* CS35L33_HG_EN */ +#define CS35L33_CLASS_HG_ENA_SHIFT 7 +#define CS35L33_CLASS_HG_EN_MASK (0x1 << CS35L33_CLASS_HG_ENA_SHIFT) +#define CS35L33_VP_HG_AUTO_SHIFT 6 +#define CS35L33_VP_HG_AUTO_MASK (0x1 << 6) +#define CS35L33_VP_HG_SHIFT 0 +#define CS35L33_VP_HG_MASK (0x1F << CS35L33_VP_HG_SHIFT) + +#define CS35L33_RATES (SNDRV_PCM_RATE_8000_48000) +#define CS35L33_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +/* CS35L33_{RX,TX}_X */ +#define CS35L33_X_STATE_SHIFT 7 +#define CS35L33_X_STATE (1 << CS35L33_X_STATE_SHIFT) +#define CS35L33_X_LOC_SHIFT 0 +#define CS35L33_X_LOC (0x1F << CS35L33_X_LOC_SHIFT) + +/* CS35L33_RX_AUD */ +#define CS35L33_AUDIN_RX_DEPTH_SHIFT 5 +#define CS35L33_AUDIN_RX_DEPTH (0x7 << CS35L33_AUDIN_RX_DEPTH_SHIFT) + +#endif -- 1.9.1

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[alsa-devel] [PATCH] ASoC: atmel: fix build failure
by Sudip Mukherjee 06 Jun '16

06 Jun '16

m32r allmodconfig build is failing with the error: ERROR: "bad_dma_ops" [sound/soc/atmel/snd-soc-atmel-pcm-pdc.ko] undefined! The code is using DMA but the related dependency is not mentioned in the Kconfig. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- build log is at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/134651165 sound/soc/atmel/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 06e099e..22aec9a 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -10,6 +10,7 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC tristate + depends on HAS_DMA default m if SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=m default y if SND_ATMEL_SOC_SSC_PDC=y || (SND_ATMEL_SOC_SSC_PDC=m && SND_ATMEL_SOC_SSC=y) -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rt298: remove idle_bias_off = true
by Bard Liao 06 Jun '16

06 Jun '16

From: bardliao <bardliao(a)realtek.com> Codec will not go into suspend if there is any widget forced on with idle_bias_off true. We want codec go into suspend when the system is suspend. Also, we don't do anything in bias level OFF case. So it is actually no different in bias level STANDBY or OFF case. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index f80cfe4..d403c20 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1094,7 +1094,6 @@ static struct snd_soc_codec_driver soc_codec_dev_rt298 = { .suspend = rt298_suspend, .resume = rt298_resume, .set_bias_level = rt298_set_bias_level, - .idle_bias_off = true, .controls = rt298_snd_controls, .num_controls = ARRAY_SIZE(rt298_snd_controls), .dapm_widgets = rt298_dapm_widgets, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v5] ASoC: cs53l30: Add support for Cirrus Logic CS53L30
by tim.howe＠cirrus.com 06 Jun '16

06 Jun '16

From: Tim Howe <tim.howe(a)cirrus.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- .../devicetree/bindings/sound/cs53l30.txt | 21 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 943 +++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 282 ++++++ 5 files changed, 1254 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..a2dff95 +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,21 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; +}; + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b282169..2dd97d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -60,6 +60,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271_SPI if SPI_MASTER select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -449,6 +450,11 @@ config SND_SOC_CS4349 tristate "Cirrus Logic CS4349 CODEC" depends on I2C +# Cirrus Logic Quad-Channel ADC +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 81324bc..88dff59 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -52,6 +52,7 @@ snd-soc-cs4271-spi-objs := cs4271-spi.o snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -252,6 +253,7 @@ obj-$(CONFIG_SND_SOC_CS4271_SPI) += snd-soc-cs4271-spi.o obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..2002ecd +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,943 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/gpio/consumer.h> +#include <linux/of_gpio.h> +#include <linux/platform_device.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include "cs53l30.h" + +struct cs53l30_private { + struct regmap *regmap; + struct gpio_desc *reset_gpio; + u8 asp_config_ctl; + u32 mclk; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, CS53L30_THMS_PDN }, + { CS53L30_MCLKCTL, CS53L30_MCLK_DIV_DFLT }, + { CS53L30_INT_SR_CTL, CS53L30_INTRNL_FS_DFLT }, + { CS53L30_MICBIAS_CTL, CS53L30_MIC_BIAS_DFLT }, + { CS53L30_ASPCFG_CTL, CS53L30_ASP_RATE_48K }, + { CS53L30_ASP1_CTL, CS53L30_ASP1_TDM_PDN }, + { CS53L30_ASP1_TDMTX_CTL1, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL2, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL3, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_CTL4, CS53L30_ASP1_CHTX_SLT47 }, + { CS53L30_ASP1_TDMTX_EN1, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN2, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN3, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN4, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN5, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP1_TDMTX_EN6, CS53L30_ASP_TX_DISABLED }, + { CS53L30_ASP2_CTL, CS53L30_ASP2_CTRL_DFLT }, + { CS53L30_SFT_RAMP, CS53L30_SFT_RMP_DFLT }, + { CS53L30_LRCLK_CTL1, CS53L30_LRCK_CTLX_DFLT }, + { CS53L30_LRCLK_CTL2, CS53L30_LRCK_CTLX_DFLT }, + { CS53L30_MUTEP_CTL1, CS53L30_MUTE_CTRL1_DFLT }, + { CS53L30_MUTEP_CTL2, CS53L30_MUTE_PDN_ULP }, + { CS53L30_INBIAS_CTL1, CS53L30_INBIAS_X_DFLT }, + { CS53L30_INBIAS_CTL2, CS53L30_INBIAS_X_DFLT }, + { CS53L30_DMIC1_STR_CTL, CS53L30_DMIC1_ST_DFLT }, + { CS53L30_DMIC2_STR_CTL, CS53L30_DMIC2_ST_DFLT }, + { CS53L30_ADCDMIC1_CTL1, CS53L30_ADC1_ON_AB_IN }, + { CS53L30_ADCDMIC1_CTL2, CS53L30_A1_D1_CTL2_DFLT }, + { CS53L30_ADC1_CTL3, CS53L30_ADC1_HPF_EN }, + { CS53L30_ADC1_NG_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1A_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC1B_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADCDMIC2_CTL1, CS53L30_ADC2_ON_AB_IN }, + { CS53L30_ADCDMIC2_CTL2, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2_CTL3, CS53L30_ADC2_HPF_EN }, + { CS53L30_ADC2_NG_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_AFE_CTL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2A_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_ADC2B_DIG_VOL, CS53L30_ADCX_ZERO_DFLT }, + { CS53L30_INT_MASK, CS53L30_DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP1_CTL: + case CS53L30_ASP1_TDMTX_CTL1: + case CS53L30_ASP1_TDMTX_CTL2: + case CS53L30_ASP1_TDMTX_CTL3: + case CS53L30_ASP1_TDMTX_CTL4: + case CS53L30_ASP1_TDMTX_EN1: + case CS53L30_ASP1_TDMTX_EN2: + case CS53L30_ASP1_TDMTX_EN3: + case CS53L30_ASP1_TDMTX_EN4: + case CS53L30_ASP1_TDMTX_EN5: + case CS53L30_ASP1_TDMTX_EN6: + case CS53L30_ASP2_CTL: + case CS53L30_SFT_RAMP: + case CS53L30_LRCLK_CTL1: + case CS53L30_LRCLK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); + +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); + +static const char * const input1_sel_text[] = { "DMIC1 On AB In", + "DMIC1 On A In", "DMIC1 On B In", "ADC1 On AB In", "ADC1 On A In", + "ADC1 On B In", "DMIC1 Off ADC1 Off", }; + +unsigned int const input1_sel_values[] = { CS53L30_DMIC1_ON_AB_IN, + CS53L30_DMIC1_ON_A_IN, CS53L30_DMIC1_ON_B_IN, CS53L30_ADC1_ON_AB_IN, + CS53L30_ADC1_ON_A_IN, CS53L30_ADC1_ON_B_IN, CS53L30_D1_OFF_A1_OFF, }; + +static const char * const input2_sel_text[] = { "DMIC2 On AB In", + "DMIC2 On A In", "DMIC2 On B In", "ADC2 On AB In", "ADC2 On A In", + "ADC2 On B In", "DMIC2 Off ADC2 Off", }; + +unsigned int const input2_sel_values[] = { CS53L30_DMIC2_ON_AB_IN, + CS53L30_DMIC2_ON_A_IN, CS53L30_DMIC2_ON_B_IN, CS53L30_ADC2_ON_AB_IN, + CS53L30_ADC2_ON_A_IN, CS53L30_ADC2_ON_B_IN, CS53L30_D2_OFF_A2_OFF, }; + +static const char * const input1_route_sel_text[] = { "ADC1_SEL", + "DMIC1_SEL" }; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input1_route_sel_text), input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + CS53L30_A1_D1_PDN_MASK, input1_sel_text, input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { "ADC2_SEL", + "DMIC2_SEL" }; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + CS53L30_A1_D1_PDN_MASK, input2_sel_text, input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms" }; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 0, + ARRAY_SIZE(cs53l30_ng_delay_text), cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB"}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, 2, + ARRAY_SIZE(cs53l30_ng_thres_text), cs53l30_ng_thres_text); + +/* ADC Preamp gain select */ +static const char * const cs53l30_preamp_gain_sel_text[] = { + "0dB", "10dB", "20dB"}; + +static const struct soc_enum adc1a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc1b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2a_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2A_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +static const struct soc_enum adc2b_preamp_gain_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2B_AFE_CTL, 6, + ARRAY_SIZE(cs53l30_preamp_gain_sel_text), + cs53l30_preamp_gain_sel_text); + +/* Set MIC Bias Voltage Control */ +static const char * const cs53l30_micbias_text[] = { + "HiZ", "1.8V", "2.75V"}; + +static const struct soc_enum micbias_enum = + SOC_ENUM_SINGLE(CS53L30_MICBIAS_CTL, 0, + ARRAY_SIZE(cs53l30_micbias_text), + cs53l30_micbias_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz"}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, 1, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, 5, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", + CS53L30_ADC1_CTL3, 0, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", + CS53L30_ADC2_CTL3, 0, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", + CS53L30_ADC1_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 6, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", + CS53L30_ADC2_NG_CTL, 7, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", + CS53L30_ADCDMIC1_CTL2, 7, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", + CS53L30_ADCDMIC2_CTL2, 7, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", + CS53L30_ADCDMIC1_CTL2, 4, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", + CS53L30_ADCDMIC1_CTL2, 5, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", + CS53L30_ADCDMIC2_CTL2, 4, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", + CS53L30_ADCDMIC2_CTL2, 5, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", + CS53L30_ADCDMIC1_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 0, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", + CS53L30_ADCDMIC2_CTL2, 1, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", + CS53L30_ADC1_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", + CS53L30_ADC2_NG_CTL, 5, 1, 0, adc_ng_boost_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + SOC_ENUM("ADC1A Pre Amp Gain", adc1a_preamp_gain_enum), + SOC_ENUM("ADC1B Pre Amp Gain", adc1b_preamp_gain_enum), + SOC_ENUM("ADC2A Pre Amp Gain", adc2a_preamp_gain_enum), + SOC_ENUM("ADC2B Pre Amp Gain", adc2b_preamp_gain_enum), + SOC_ENUM("Mic Bias Voltage Select", micbias_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static int cs53l30_asp_sdout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + CS53L30_ASP1_3ST, 0); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + CS53L30_ASP1_3ST, 1); + break; + default: + dev_err(codec->dev, "Invalid event = 0x%x\n", event); + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, 4, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, 5, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, 6, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, 7, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT1", NULL, 0, CS53L30_ASP1_CTL, + CS53L30_ASP1_SDOUT_PDN, 1, + cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + SND_SOC_DAPM_AIF_OUT_E("ASP_SDOUT2", NULL, 0, CS53L30_ASP2_CTL, + CS53L30_ASP2_SDOUT_PDN, 1, + cs53l30_asp_sdout_event, + (SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD)), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, 7, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, 6, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, 7, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, 2, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, 2, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_audio_map[] = { + + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, + + /* Output Paths */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"ASP1 Capture", NULL, "ASP_SDOUT1"}, + {"ASP2 Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, 1, 1}, + {5644800, 22050, 0x8, 1, 1}, + {5644800, 44100, 0xC, 1, 1}, + + {6000000, 8000, 0x1, 0, 1}, + {6000000, 11025, 0x2, 0, 1}, + {6000000, 12000, 0x4, 0, 1}, + {6000000, 16000, 0x5, 0, 1}, + {6000000, 22050, 0x6, 0, 1}, + {6000000, 24000, 0x8, 0, 1}, + {6000000, 32000, 0x9, 0, 1}, + {6000000, 44100, 0xA, 0, 1}, + {6000000, 48000, 0xC, 0, 1}, + + {6144000, 8000, 0x1, 1, 1}, + {6144000, 11025, 0x2, 1, 1}, + {6144000, 12000, 0x4, 1, 1}, + {6144000, 16000, 0x5, 1, 1}, + {6144000, 22050, 0x6, 1, 1}, + {6144000, 24000, 0x8, 1, 1}, + {6144000, 32000, 0x9, 1, 1}, + {6144000, 44100, 0xA, 1, 1}, + {6144000, 48000, 0xC, 1, 1}, + + {6400000, 8000, 0x1, 1, 1}, + {6400000, 11025, 0x2, 1, 1}, + {6400000, 12000, 0x4, 1, 1}, + {6400000, 16000, 0x5, 1, 1}, + {6400000, 22050, 0x6, 1, 1}, + {6400000, 24000, 0x8, 1, 1}, + {6400000, 32000, 0x9, 1, 1}, + {6400000, 44100, 0xA, 1, 1}, + {6400000, 48000, 0xC, 1, 1}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, 0}, + {6000000, 1, 0}, + {6144000, 1, 0}, + {11289600, 2, 1}, + {12288000, 2, 1}, + {12000000, 2, 1}, + {19200000, 3, 2}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk == mclk && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + return -EINVAL; + +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + int mclkx_coeff; + u32 mclk; + unsigned int mclk_ctl; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + mclk_ctl &= ~CS53L30_MCLK_DIV; + mclk_ctl |= cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv; + + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, CS53L30_MCLK_DIV, + mclk_ctl << CS53L30_MCLK_DIV); + priv->mclk = mclk; + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int asp_config_ctl; + + regmap_read(priv->regmap, CS53L30_ASPCFG_CTL, &asp_config_ctl); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + asp_config_ctl |= CS53L30_ASP_MS; + break; + + case SND_SOC_DAIFMT_CBS_CFS: + asp_config_ctl &= ~CS53L30_ASP_MS; + break; + + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + asp_config_ctl |= CS53L30_ASP_SCLK_INV; + else + asp_config_ctl &= ~CS53L30_ASP_SCLK_INV; + + priv->asp_config_ctl = asp_config_ctl; + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + int mclk_coeff; + int srate = params_rate(params); + unsigned int int_sr_ctl, mclk_ctl; + + /* MCLK -> srate */ + mclk_coeff = cs53l30_get_mclk_coeff(priv->mclk, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_read(priv->regmap, CS53L30_INT_SR_CTL, &int_sr_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio) + int_sr_ctl |= CS53L30_INTRNL_FS_RATIO; + else + int_sr_ctl &= ~CS53L30_INTRNL_FS_RATIO; + regmap_update_bits(priv->regmap, CS53L30_INT_SR_CTL, + CS53L30_INTL_FS_RAT_MSK, int_sr_ctl << + CS53L30_INTL_FS_RAT_SFT); + + regmap_read(priv->regmap, CS53L30_MCLKCTL, &mclk_ctl); + if (cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale) + mclk_ctl |= CS53L30_MCLK_INT_SCALE; + else + mclk_ctl &= ~CS53L30_MCLK_INT_SCALE; + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCK_INT_SCL_MSK, mclk_ctl << + CS53L30_MCK_INT_SCL_SFT); + + priv->asp_config_ctl &= CS53L30_ASP_CNFG_MASK; + priv->asp_config_ctl |= (cs53l30_mclk_coeffs[mclk_coeff].asp_rate + & CS53L30_ASP_RATE_MASK); + regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL, + CS53L30_ASP_RATE_MASK, priv->asp_config_ctl); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_LP, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIS, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_LP, CS53L30_PDN_LP); + } + break; + + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + CS53L30_PDN_DONE, 0); + /* If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & CS53L30_DIGSFT) + inter_max_check = CS53L30_PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP, CS53L30_PDN_ULP); + msleep(20); /* PDN_DONE will take a min of 20ms to be set.*/ + regmap_read(priv->regmap, CS53L30_IS, &reg); /* Clr status */ + for (i = 0; i < inter_max_check; i++) { + if (inter_max_check < 10) { + usleep_range(1000, 1100); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & CS53L30_PDN_DONE) + break; + } else { + usleep_range(10000, 10100); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & CS53L30_PDN_DONE) + break; + } + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + CS53L30_PDN_DONE, CS53L30_PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIS, CS53L30_MCLK_DIS); + break; + } + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_update_bits(priv->regmap, CS53L30_ASP1_CTL, + CS53L30_ASP1_3ST, + (CS53L30_ASP1_3ST_VAL(tristate) & + CS53L30_ASP1_3ST)); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, + 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &src_constraints); + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, +}; + +static struct snd_soc_dai_driver cs53l30_dai[] = { + { + .name = "cs53l30-asp1", + .id = CS53L30_ASP1, + .capture = { + .stream_name = "ASP1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + }, + { + .name = "cs53l30-asp2", + .id = CS53L30_ASP2, + .capture = { + .stream_name = "ASP2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, + } +}; + +static struct snd_soc_codec_driver soc_codec_dev_cs53l30 = { + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_audio_map, + .num_dapm_routes = ARRAY_SIZE(cs53l30_audio_map), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct cs53l30_private *cs53l30; + int ret = 0; + unsigned int devid = 0; + unsigned int reg; + + cs53l30 = devm_kzalloc(&client->dev, + sizeof(struct cs53l30_private), GFP_KERNEL); + if (!cs53l30) { + dev_err(&client->dev, "could not allocate codec\n"); + return -ENOMEM; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(&client->dev, + "reset", GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) + return PTR_ERR(cs53l30->reset_gpio); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(&client->dev, "regmap_init() failed: %d\n", ret); + return ret; + } + /* initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(&client->dev, + "CS53L30 Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + return ret; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(&client->dev, "Get Revision ID failed\n"); + return ret; + } + + dev_info(&client->dev, + "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_cs53l30, cs53l30_dai, + ARRAY_SIZE(cs53l30_dai)); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + {"cs53l30", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, + +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..116e358 +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,282 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP1_CTL 0x0D /* ASP1 Control. */ +#define CS53L30_ASP1_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP1_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP1_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP1_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP1_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP1_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP1_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP1_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP1_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP1_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP2_CTL 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCLK_CTL1 0x1B /* LRCLK Control 1. */ +#define CS53L30_LRCLK_CTL2 0x1C /* LRCLK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define CS53L30_PDN_POLL_MAX 90 + +/* Bitfield Definitions */ + +/* CS53L30_PWRCTL */ +#define CS53L30_PDN_ULP (1 << 7) +#define CS53L30_PDN_LP (1 << 6) +#define CS53L30_DISCHARGE_FILT (1 << 5) +#define CS53L30_THMS_PDN (1 << 4) + +/* CS53L30_MCLKCTL */ +#define CS53L30_MCLK_DIS (1 << 7) +#define CS53L30_MCLK_INT_SCALE (1 << 6) +#define CS53L30_DMIC_DRIVE (1 << 5) +#define CS53L30_MCLK_DIV (3 << 2) +#define CS53L30_MCLK_DIV_DFLT (1 << 2) +#define CS53L30_SYNC_EN (1 << 1) +#define CS53L30_MCK_INT_SCL_MSK 0x40 +#define CS53L30_MCK_INT_SCL_SFT 6 + +/* CS53L30_INT_SR_CTL */ +#define CS53L30_INTRNL_FS_RATIO (1 << 4) +#define CS53L30_INTRNL_FS_DFLT (7 << 2) +#define CS53L30_MCLK_19MHZ_EN (1 << 0) +#define CS53L30_INTL_FS_RAT_MSK 0x10 +#define CS53L30_INTL_FS_RAT_SFT 4 + +/* CS53L30_MICBIAS_CTL */ +#define CS53L30_MIC4_BIAS_PDN (1 << 7) +#define CS53L30_MIC3_BIAS_PDN (1 << 6) +#define CS53L30_MIC2_BIAS_PDN (1 << 5) +#define CS53L30_MIC1_BIAS_PDN (1 << 4) +#define CS53L30_VP_MIN (1 << 2) +#define CS53L30_MIC_BIAS_CTRL (3 << 0) +#define CS53L30_BIAS_ALL_PDN 0xF0 +#define CS53L30_MIC_BIAS_DFLT (CS53L30_BIAS_ALL_PDN | CS53L30_VP_MIN) + +/* CS53L30_ASPCFG_CTL */ +#define CS53L30_ASP_MS (1 << 7) +#define CS53L30_ASP_SCLK_INV (1 << 4) +#define CS53L30_ASP_RATE_48K (3 << 2) +#define CS53L30_ASP_RATE_MASK 0x0F +#define CS53L30_ASP_CNFG_MASK 0xF0 + +/* CS53L30_ASP1_CTL */ +#define CS53L30_ASP1_TDM_PDN (1 << 7) +#define CS53L30_ASP1_SDOUT_PDN (1 << 6) +#define CS53L30_ASP1_3ST (1 << 5) +#define CS53L30_SHIFT_LEFT (1 << 4) +#define CS53L30_ASP1_DRIVE (1 << 0) +#define CS53L30_ASP1_3ST_VAL(x) ((x) << 5) + +/* CS53L30_ASP1_TDMTX_CTL */ +#define CS53L30_ASP1_CHX_TX_ST (1 << 7) +#define CS53L30_ASP1_CHX_TX_LOC 0x3F +#define CS53L30_ASP1_CHTX_SLT47 0x2F +#define CS53L30_ASP_TX_DISABLED 0x00 + +/* CS53L30_ASP2_CTL */ +#define CS53L30_ASP2_SDOUT_PDN (1 << 6) +#define CS53L30_ASP2_DRIVE (1 << 0) +#define CS53L30_ASP2_CTRL_DFLT 0x00 + +/* CS53L30_SFT_RAMP */ +#define CS53L30_DIGSFT (1 << 5) +#define CS53L30_SFT_RMP_DFLT 0x00 + +/* CS53L30_LRCLK_CTL2 */ +#define CS53L30_LRCK_50_NPW (1 << 3) +#define CS53L30_LRCK_TPWH (7 << 0) +#define CS53L30_LRCK_CTLX_DFLT 0x00 + +/* CS53L30_MUTEP_CTL */ +#define CS53L30_MUTE_PDN_ULP (1 << 7) +#define CS53L30_MUTE_PDN_LP (1 << 6) +#define CS53L30_MUTE_M4B_PDN (1 << 4) +#define CS53L30_MUTE_M3B_PDN (1 << 3) +#define CS53L30_MUTE_M2B_PDN (1 << 2) +#define CS53L30_MUTE_M1B_PDN (1 << 1) +#define CS53L30_MUTE_MB_ALL_PDN (1 << 0) +#define CS53L30_MUTE_CTRL1_DFLT 0x00 + +/* CS53L30_MUTEP_CTL2 */ +#define CS53L30_MUTE_PIN_PLRTY (1 << 7) +#define CS53L30_MUTE_ASPTDM_PDN (1 << 6) +#define CS53L30_MTE_ASPSDO2_PDN (1 << 5) +#define CS53L30_MTE_ASPSDO1_PDN (1 << 4) +#define CS53L30_MUTE_ADC2B_PDN (1 << 3) +#define CS53L30_MUTE_ADC2A_PDN (1 << 2) +#define CS53L30_MUTE_ADC1B_PDN (1 << 1) +#define CS53L30_MUTE_ADC1A_PDN (1 << 0) + +/* CS53L30_INBIAS_CTL1 */ +#define CS53L30_IN4M_BIAS (3 << 6) +#define CS53L30_IN4P_BIAS (3 << 4) +#define CS53L30_IN3M_BIAS (3 << 2) +#define CS53L30_IN3P_BIAS (3 << 0) + +/* CS53L30_INBIAS_CTL2 */ +#define CS53L30_IN2M_BIAS (3 << 6) +#define CS53L30_IN2P_BIAS (3 << 4) +#define CS53L30_IN1M_BIAS (3 << 2) +#define CS53L30_IN1P_BIAS (3 << 0) +#define CS53L30_INBIAS_X_DFLT 0xAA + +/* CS53L30_DMIC1_STR_CTL */ +#define CS53L30_DMIC1_STERO_EN (1 << 5) +#define CS53L30_DMIC1_ST_DFLT 0xA8 + +/* CS53L30_DMIC2_STR_CTL */ +#define CS53L30_DMIC2_STERO_EN (1 << 5) +#define CS53L30_DMIC2_ST_DFLT 0xEC + +/* CS53L30_ADCDMIC1_CTL1 */ +#define CS53L30_ADC1B_PDN (1 << 7) +#define CS53L30_ADC1A_PDN (1 << 6) +#define CS53L30_DMIC1_PDN (1 << 2) +#define CS53L30_DMIC1_SCLK_DIV (1 << 1) +#define CS53L30_CH_TYPE (1 << 0) +#define CS53L30_DMIC1_ON_AB_IN (CS53L30_CH_TYPE) +#define CS53L30_DMIC1_ON_A_IN (CS53L30_ADC1B_PDN | CS53L30_CH_TYPE) +#define CS53L30_DMIC1_ON_B_IN (CS53L30_ADC1A_PDN | CS53L30_CH_TYPE) +#define CS53L30_ADC1_ON_AB_IN (CS53L30_DMIC1_PDN) +#define CS53L30_ADC1_ON_A_IN (CS53L30_ADC1B_PDN | CS53L30_DMIC1_PDN) +#define CS53L30_ADC1_ON_B_IN (CS53L30_ADC1A_PDN | CS53L30_DMIC1_PDN) +#define CS53L30_D1_OFF_A1_OFF (CS53L30_ADC1A_PDN | \ + CS53L30_ADC1B_PDN | \ + CS53L30_DMIC1_PDN) +#define CS53L30_A1_D1_PDN_MASK 0xFF + +/* CS53L30_ADCDMIC1_CTL2 */ +#define CS53L30_ADC1_NOTCH_DIS (1 << 7) +#define CS53L30_ADC1B_INV (1 << 5) +#define CS53L30_ADC1A_INV (1 << 4) +#define CS53L30_ADC1B_DIG_BOOST (1 << 1) +#define CS53L30_ADC1A_DIG_BOOST (1 << 0) +#define CS53L30_A1_D1_CTL2_DFLT 0x00 + +/* CS53L30_ADC1_CTL3 */ +#define CS53L30_ADC1_HPF_EN (1 << 3) +#define CS53L30_ADC1_HPF_CF (3 << 1) +#define CS53L30_ADC1_NG_ALL (1 << 0) + +/* CS53L30_ADC1_NG_CTL */ +#define CS53L30_ADC1B_NG (1 << 7) +#define CS53L30_ADC1A_NG (1 << 6) +#define CS53L30_ADC1_NG_BOOST (1 << 5) +#define CS53L30_ADC1_NG_THRESH (7 << 2) +#define CS53L30_ADC1_NG_DELAY (3 << 0) +#define CS53L30_ADCX_ZERO_DFLT 0x00 + +/* CS53L30_ADC1A_AFE_CTL */ +#define CS53L30_ADC1A_PREAMP (3 << 6) +#define CS53L30_ADC1A_PGA_VOL 0x3F + +/* CS53L30_ADC1B_AFE_CTL */ +#define CS53L30_ADC1B_PREAMP (3 << 6) +#define CS53L30_ADC1B_PGA_VOL 0x3F + +/* CS53L30_ADCXX_DIG_VOL */ +#define CS53L30_MUTE_DIG_OUT (1 << 7) + +/* CS53L30_ADCDMIC2_CTL1 */ +#define CS53L30_ADC2B_PDN (1 << 7) +#define CS53L30_ADC2A_PDN (1 << 6) +#define CS53L30_DMIC2_PDN (1 << 2) +#define CS53L30_DMIC2_CLKDIV (1 << 1) +#define CS53L30_DMIC2_ON_AB_IN 0x00 /* CH_TYPE must = 1 */ +#define CS53L30_DMIC2_ON_A_IN (CS53L30_ADC2B_PDN) /* CH_TYPE must = 1 */ +#define CS53L30_DMIC2_ON_B_IN (CS53L30_ADC2A_PDN) /* CH_TYPE must = 1 */ +#define CS53L30_ADC2_ON_AB_IN (CS53L30_DMIC2_PDN) /* CH_TYPE must = 0 */ +#define CS53L30_ADC2_ON_A_IN (CS53L30_ADC2B_PDN | \ + CS53L30_DMIC2_PDN) +#define CS53L30_ADC2_ON_B_IN (CS53L30_ADC2A_PDN | \ + CS53L30_DMIC2_PDN) +#define CS53L30_D2_OFF_A2_OFF (CS53L30_ADC2A_PDN | \ + CS53L30_ADC2B_PDN | \ + CS53L30_DMIC2_PDN) + +/* CS53L30_ADCDMIC2_CTL2 */ +#define CS53L30_ADC2_NOTCH_DIS (1 << 7) +#define CS53L30_ADC2B_INV (1 << 5) +#define CS53L30_ADC2A_INV (1 << 4) +#define CS53L30_ADC2B_DIG_BOOST (1 << 1) +#define CS53L30_ADC2A_DIG_BOOST (1 << 0) + +/* CS53L30_ADC2_CTL3 */ +#define CS53L30_ADC2_HPF_EN (1 << 3) +#define CS53L30_ADC2_HPF_CF (3 << 1) +#define CS53L30_ADC2_NG_ALL (1 << 0) + +/* CS53L30_INT */ +#define CS53L30_PDN_DONE (1 << 7) +#define CS53L30_THMS_TRIP (1 << 6) +#define CS53L30_SYNC_DONE (1 << 5) +#define CS53L30_ADC2B_OVFL (1 << 4) +#define CS53L30_ADC2A_OVFL (1 << 3) +#define CS53L30_ADC1B_OVFL (1 << 2) +#define CS53L30_ADC1A_OVFL (1 << 1) +#define CS53L30_MUTE_PIN (1 << 0) +#define CS53L30_DEVICE_INT_MASK 0xFF + +/* Serial Ports */ +#define CS53L30_ASP1 0 +#define CS53L30_ASP2 1 + +#endif /* __CS53L30_H__ */ -- 2.4.5

5 12

[alsa-devel] [PATCH V2 linux-next] ASoC: cs53l30: include gpio/consumer.h
by Fabian Frederick 06 Jun '16

06 Jun '16

cs53l30 breaks kernel compilation when CONFIG_GPIOLIB is disabled. sound/soc/codecs/cs53l30.c:931:2: error: implicit declaration of function devm_gpiod_get_optional [-Werror=implicit-function-declaration] cs53l30->reset_gpio = devm_gpiod_get_optional(dev, "reset", ^ sound/soc/codecs/cs53l30.c:932:13: error: GPIOD_OUT_LOW undeclared (first use in this function) Including gpio/consumer.h declares minimal functions in that case. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- V2: Add in alphabetical order (suggested by Nicolin Chen) sound/soc/codecs/cs53l30.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ac90dd7..4f6ede9 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -14,6 +14,7 @@ #include <linux/clk.h> #include <linux/delay.h> +#include <linux/gpio/consumer.h> #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_gpio.h> -- 2.1.4

3 2

[alsa-devel] [GIT PULL] Braswell audio firmware update
by Vinod Koul 06 Jun '16

06 Jun '16

Please PULL to receive the updated BSW firmware binary which contains some odd playback noise fixes The following changes since commit f5b546819836f207a6df58d48423ca6112cd2801: linux-firmware: Update firmware patch for Intel Bluetooth 7260 (B5/B6) (2016-06-02 09:51:44 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git BSW for you to fetch changes up to 1d079ff33967673e9de44f1ac33de8b62f021b32: linux-firmware: update audio firmware for Braswell platform (2016-06-06 14:24:20 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: update audio firmware for Braswell platform WHENCE | 2 +- intel/fw_sst_22a8.bin | Bin 701614 -> 701694 bytes 2 files changed, 1 insertion(+), 1 deletion(-) Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: only check VDDD-supply, not revision
by Clemens Gruber 06 Jun '16

06 Jun '16

Instead of checking the SGTL5000 chip revision, we should only check if the VDDD regulator exists and only call sgtl5000_replace_vddd_with_ldo if the regulator is missing. Otherwise, the user reads in the kernel log that the internal LDO is used, even though he did follow the NXP recommendation to use external VDDD and also specified VDDD-supply in the devicetree. Also remove the comment, which incorrectly states that external VDDD is only supported for SGTL5000 chip revisions < 0x11. Official NXP documentation recommends using external VDDD and not the internal LDO due to the SGTL5000 erratum ER1. This also applies to revisions >= 0x11. Tested on an i.MX6Q board with SGTL5000 rev 0x11 and external VDDD. Signed-off-by: Clemens Gruber <clemens.gruber(a)pqgruber.com> --- sound/soc/codecs/sgtl5000.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 08b4046..fbad4fb 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1286,17 +1286,14 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - /* External VDDD only works before revision 0x11 */ - if (sgtl5000->revision < 0x11) { - vddd = regulator_get_optional(codec->dev, "VDDD"); - if (IS_ERR(vddd)) { - /* See if it's just not registered yet */ - if (PTR_ERR(vddd) == -EPROBE_DEFER) - return -EPROBE_DEFER; - } else { - external_vddd = 1; - regulator_put(vddd); - } + vddd = regulator_get_optional(codec->dev, "VDDD"); + if (IS_ERR(vddd)) { + /* See if it's just not registered yet */ + if (PTR_ERR(vddd) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + external_vddd = 1; + regulator_put(vddd); } if (!external_vddd) { -- 2.8.3

4 12

[alsa-devel] [REGRESSION] Headphones no longer working on MacPro6, 1 with 4.4
by Laura Abbott 06 Jun '16

06 Jun '16

Hi, We received a bug report https://bugzilla.redhat.com/show_bug.cgi?id=1316119 that the headphone jack on a MacPro6,1 stopped working on an upgrade from 4.3 to 4.4. The bugzilla has the alsainfo, diffing shows that the Amp-Out vals are different. I tried a revert of 9f660a1c4 (" ALSA: hda/realtek - Fix silent headphone output on MacPro 4,1 (v2)") but that didn't help. Any ideas before asking for a bisect? Does this hardware version need to have the vref fixup as well? Thanks, Laura

2 9

[alsa-devel] sgtl5000: probe failed with error -5
by Catalin Crenguta 06 Jun '16

06 Jun '16

I'm working with a phycore vybrid SoM and I'm having some troubles binding the sgtl5000 audio codec using devicetree. As you can see, the codec driver could not be probed due to error -5. The dts looks like this: sound { compatible = "fsl,imx-audio-sgtl5000"; model = "phycore-vybrid-sgtl5000"; audio-cpu = <&sai2>; audio-codec = <&codec>; audio-routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT"; mux-int-port = <1>; mux-ext-port = <3>; }; /* .... */ &i2c2 { clock-frequency = <400000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; status = "okay"; /* ... */ codec: sgtl5000@0a { compatible = "fsl,sgtl5000"; reg = <0x0a>; pinctrl-names = "default"; clocks = <&clks 150>; micbias-resistor-k-ohms = <2>; micbias-voltage-m-volts = <2250>; VDDA-supply = <&reg_3p3v>; VDDIO-supply = <&reg_3p3v>; }; }; /* ... */ &iomuxc { vf610-phycore { pinctrl_i2c2: i2c2grp { fsl,pins = < VF610_PAD_PTA22__I2C2_SCL 0x34d3 VF610_PAD_PTA23__I2C2_SDA 0x34d3 >; }; }; }; After the system boot, the output of dmesg looks like this: dmesg | grep sgtl5000 [ 1.856390] bus: 'i2c': add driver sgtl5000 [ 1.856441] bus: 'i2c': driver_probe_device: matched device 0-000a with driver sgtl5000 [ 1.856462] bus: 'i2c': really_probe: probing driver sgtl5000 with device 0-000a [ 1.856512] sgtl5000 0-000a: no default pinctrl state [ 1.857164] sgtl5000: probe of 0-000a failed with error -5 dmesg | grep sai [ 1.863020] bus: 'platform': add driver fsl-sai [ 1.863098] bus: 'platform': driver_probe_device: matched device 40031000.sai with driver fsl-sai [ 1.863121] bus: 'platform': really_probe: probing driver fsl-sai with device 40031000.sai [ 1.863276] fsl-sai 40031000.sai: no sleep pinctrl state [ 1.863301] fsl-sai 40031000.sai: no idle pinctrl state [ 1.863695] driver: '40031000.sai': driver_bound: bound to device 'fsl-sai' [ 1.863726] bus: 'platform': really_probe: bound device 40031000.sai to driver fsl-sai

1 0

[alsa-devel] [RFC PATCH] drm: msm: Add ASoC generic hdmi audio codec support.
by Srinivas Kandagatla 06 Jun '16

06 Jun '16

This patch adds support to generic audio codec via ASoC hdmi-codec infrastucture which is merged recently. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/gpu/drm/msm/Kconfig | 1 + drivers/gpu/drm/msm/hdmi/hdmi.c | 120 +++++++++++++++++++++++++++++++++++++++- drivers/gpu/drm/msm/hdmi/hdmi.h | 14 +++++ 3 files changed, 134 insertions(+), 1 deletion(-) diff --git a/drivers/gpu/drm/msm/Kconfig b/drivers/gpu/drm/msm/Kconfig index 167a497..7c7a031 100644 --- a/drivers/gpu/drm/msm/Kconfig +++ b/drivers/gpu/drm/msm/Kconfig @@ -10,6 +10,7 @@ config DRM_MSM select SHMEM select TMPFS select QCOM_SCM + select SND_SOC_HDMI_CODEC if SND_SOC default y help DRM/KMS driver for MSM/snapdragon. diff --git a/drivers/gpu/drm/msm/hdmi/hdmi.c b/drivers/gpu/drm/msm/hdmi/hdmi.c index 51b9ea5..3281496 100644 --- a/drivers/gpu/drm/msm/hdmi/hdmi.c +++ b/drivers/gpu/drm/msm/hdmi/hdmi.c @@ -19,6 +19,7 @@ #include <linux/of_irq.h> #include <linux/of_gpio.h> +#include <sound/hdmi-codec.h> #include "hdmi.h" void msm_hdmi_set_mode(struct hdmi *hdmi, bool power_on) @@ -434,6 +435,114 @@ static int msm_hdmi_get_gpio(struct device_node *of_node, const char *name) return gpio; } +/* + * HDMI audio codec callbacks + */ +static int msm_hdmi_audio_hw_params(struct device *dev, + struct hdmi_codec_daifmt *daifmt, + struct hdmi_codec_params *params) +{ + struct hdmi *hdmi = dev_get_drvdata(dev); + unsigned int chan; + unsigned int channel_allocation = 0; + unsigned int rate; + unsigned int level_shift = 0; /* 0dB */ + bool down_mix = false; + + dev_dbg(dev, "%u Hz, %d bit, %d channels\n", params->sample_rate, + params->sample_width, params->cea.channels); + + switch (params->cea.channels) { + case 2: + /* FR and FL speakers */ + channel_allocation = 0; + chan = MSM_HDMI_AUDIO_CHANNEL_2; + break; + case 4: + /* FC, LFE, FR and FL speakers */ + channel_allocation = 0x3; + chan = MSM_HDMI_AUDIO_CHANNEL_4; + break; + case 6: + /* RR, RL, FC, LFE, FR and FL speakers */ + channel_allocation = 0x0B; + chan = MSM_HDMI_AUDIO_CHANNEL_6; + break; + case 8: + /* FRC, FLC, RR, RL, FC, LFE, FR and FL speakers */ + channel_allocation = 0x1F; + chan = MSM_HDMI_AUDIO_CHANNEL_8; + break; + default: + return -EINVAL; + } + + switch (params->sample_rate) { + case 32000: + rate = HDMI_SAMPLE_RATE_32KHZ; + break; + case 44100: + rate = HDMI_SAMPLE_RATE_48KHZ; + break; + case 48000: + rate = HDMI_SAMPLE_RATE_48KHZ; + break; + case 88200: + rate = HDMI_SAMPLE_RATE_88_2KHZ; + break; + case 96000: + rate = HDMI_SAMPLE_RATE_96KHZ; + break; + case 176400: + rate = HDMI_SAMPLE_RATE_176_4KHZ; + break; + case 192000: + rate = HDMI_SAMPLE_RATE_192KHZ; + break; + default: + dev_err(dev, "rate[%d] not supported!\n", + params->sample_rate); + return -EINVAL; + } + + msm_hdmi_audio_set_sample_rate(hdmi, rate); + msm_hdmi_audio_info_setup(hdmi, 1, chan, channel_allocation, + level_shift, down_mix); + + return 0; +} + +static void msm_hdmi_audio_shutdown(struct device *dev) +{ + struct hdmi *hdmi = dev_get_drvdata(dev); + + msm_hdmi_audio_info_setup(hdmi, 0, 0, 0, 0, 0); +} + +static const struct hdmi_codec_ops msm_hdmi_audio_codec_ops = { + .hw_params = msm_hdmi_audio_hw_params, + .audio_shutdown = msm_hdmi_audio_shutdown, +}; + +static struct hdmi_codec_pdata codec_data = { + .ops = &msm_hdmi_audio_codec_ops, + .max_i2s_channels = 8, + .i2s = 1, +}; + +static int msm_hdmi_register_audio_driver(struct hdmi *hdmi, struct device *dev) +{ + hdmi->audio_pdev = platform_device_register_data(dev, + HDMI_CODEC_DRV_NAME, + PLATFORM_DEVID_AUTO, + &codec_data, + sizeof(codec_data)); + if (IS_ERR(hdmi->audio_pdev)) + return PTR_ERR(hdmi->audio_pdev); + + return 0; +} + static int msm_hdmi_bind(struct device *dev, struct device *master, void *data) { struct drm_device *drm = dev_get_drvdata(master); @@ -441,7 +550,7 @@ static int msm_hdmi_bind(struct device *dev, struct device *master, void *data) static struct hdmi_platform_config *hdmi_cfg; struct hdmi *hdmi; struct device_node *of_node = dev->of_node; - int i; + int i, err; hdmi_cfg = (struct hdmi_platform_config *) of_device_get_match_data(dev); @@ -468,6 +577,12 @@ static int msm_hdmi_bind(struct device *dev, struct device *master, void *data) return PTR_ERR(hdmi); priv->hdmi = hdmi; + err = msm_hdmi_register_audio_driver(hdmi, dev); + if (err) { + DRM_ERROR("Failed to attach an audio codec %d\n", err); + hdmi->audio_pdev = NULL; + } + return 0; } @@ -477,6 +592,9 @@ static void msm_hdmi_unbind(struct device *dev, struct device *master, struct drm_device *drm = dev_get_drvdata(master); struct msm_drm_private *priv = drm->dev_private; if (priv->hdmi) { + if (priv->hdmi->audio_pdev) + platform_device_unregister(priv->hdmi->audio_pdev); + msm_hdmi_destroy(priv->hdmi); priv->hdmi = NULL; } diff --git a/drivers/gpu/drm/msm/hdmi/hdmi.h b/drivers/gpu/drm/msm/hdmi/hdmi.h index bc7ba0b..accc9a6 100644 --- a/drivers/gpu/drm/msm/hdmi/hdmi.h +++ b/drivers/gpu/drm/msm/hdmi/hdmi.h @@ -50,6 +50,7 @@ struct hdmi_hdcp_ctrl; struct hdmi { struct drm_device *dev; struct platform_device *pdev; + struct platform_device *audio_pdev; const struct hdmi_platform_config *config; @@ -210,6 +211,19 @@ static inline int msm_hdmi_pll_8996_init(struct platform_device *pdev) /* * audio: */ +/* Supported HDMI Audio channels and rates */ +#define MSM_HDMI_AUDIO_CHANNEL_2 0 +#define MSM_HDMI_AUDIO_CHANNEL_4 1 +#define MSM_HDMI_AUDIO_CHANNEL_6 2 +#define MSM_HDMI_AUDIO_CHANNEL_8 3 + +#define HDMI_SAMPLE_RATE_32KHZ 0 +#define HDMI_SAMPLE_RATE_44_1KHZ 1 +#define HDMI_SAMPLE_RATE_48KHZ 2 +#define HDMI_SAMPLE_RATE_88_2KHZ 3 +#define HDMI_SAMPLE_RATE_96KHZ 4 +#define HDMI_SAMPLE_RATE_176_4KHZ 5 +#define HDMI_SAMPLE_RATE_192KHZ 6 int msm_hdmi_audio_update(struct hdmi *hdmi); int msm_hdmi_audio_info_setup(struct hdmi *hdmi, bool enabled, -- 2.8.2

2 2

[alsa-devel] [PATCH 1/1 linux-next] ASoC: cs53l30: include gpio/consumer.h
by Fabian Frederick 06 Jun '16

06 Jun '16

cs53l30 breaks kernel compilation when CONFIG_GPIOLIB is disabled. sound/soc/codecs/cs53l30.c:931:2: error: implicit declaration of function devm_gpiod_get_optional [-Werror=implicit-function-declaration] cs53l30->reset_gpio = devm_gpiod_get_optional(dev, "reset", ^ sound/soc/codecs/cs53l30.c:932:13: error: GPIOD_OUT_LOW undeclared (first use in this function) Including gpio/consumer.h declares minimal functions in that case. Signed-off-by: Fabian Frederick <fabf(a)skynet.be> --- sound/soc/codecs/cs53l30.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index ac90dd7..4818324 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -15,6 +15,7 @@ #include <linux/clk.h> #include <linux/delay.h> #include <linux/i2c.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of_gpio.h> #include <linux/regulator/consumer.h> -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: use 19.2 MCLK instead of the bit clock for bytcr_rt5651.c machine driver
by Pietro Di Costanzo 06 Jun '16

06 Jun '16

Use the MCLK to solve audio quality issues found on some Cherryview-based platforms Signed-off-by: Pietro Di Costanzo <pxpert(a)gmail.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 84 +++++++++++++++++++++++++++++++---- 1 file changed, 75 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 35f591e..e51ea5b 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -31,11 +31,76 @@ #include "../../codecs/rt5651.h" #include "../atom/sst-atom-controls.h" +#define CHT_PLAT_CLK_3_HZ 19200000 +#define CHT_CODEC_DAI_5651 "rt5651-aif1" + +struct cht_acpi_card { + char *codec_id; + int codec_type; + struct snd_soc_card *soc_card; +}; + +struct cht_mc_private { + struct snd_soc_jack jack; + struct cht_acpi_card *acpi_card; +}; + + +static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd; + + list_for_each_entry(rtd, &card->rtd_list, list) { + if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI_5651, + strlen(CHT_CODEC_DAI_5651))) + return rtd->codec_dai; + } + return NULL; +} + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret; + + codec_dai = cht_get_codec_dai(card); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set platform clock\n"); + return -EIO; + } + + if (!SND_SOC_DAPM_EVENT_OFF(event)) + return 0; + + /* Set codec sysclk source to its internal clock because codec PLL will + * be off when idle and MCLK will also be off by ACPI when codec is + * runtime suspended. Codec needs clock for jack detection and button + * press. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5651_SCLK_S_RCCLK, + 0, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(card->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + + + + static const struct snd_soc_dapm_widget byt_rt5651_widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("Internal Mic", NULL), SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_POST_PMD), + }; static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { @@ -52,6 +117,9 @@ static const struct snd_soc_dapm_route byt_rt5651_audio_map[] = { {"Headphone", NULL, "HPOR"}, {"Speaker", NULL, "LOUTL"}, {"Speaker", NULL, "LOUTR"}, + {"Headphone", NULL, "Platform Clock"}, + {"Speaker", NULL, "Platform Clock"}, + }; static const struct snd_soc_dapm_route byt_rt5651_intmic_dmic1_map[] = { @@ -93,7 +161,13 @@ static int byt_rt5651_aif1_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dai; int ret; - snd_soc_dai_set_bclk_ratio(codec_dai, 50); + /* set codec PLL source to the 19.2MHz platform clock (MCLK) */ + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5651_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } ret = snd_soc_dai_set_sysclk(codec_dai, RT5651_SCLK_S_PLL1, params_rate(params) * 512, @@ -103,14 +177,6 @@ static int byt_rt5651_aif1_hw_params(struct snd_pcm_substream *substream, return ret; } - ret = snd_soc_dai_set_pll(codec_dai, 0, RT5651_PLL1_S_BCLK1, - params_rate(params) * 50, - params_rate(params) * 512); - if (ret < 0) { - dev_err(rtd->dev, "can't set codec pll: %d\n", ret); - return ret; - } - return 0; } -- 2.7.3

3 6

[alsa-devel] usb-audio: four new quirks
by Damien Zammit 04 Jun '16

04 Jun '16

Hi folks, I wrote these quirks, but they need to be tested on hardware, as I only had the lsusb output from someone who had the devices. - POD HD Pro quirk: http://pastebin.com/raw/TrPcYBGT - POD X3 Pro quirk: http://pastebin.com/raw/DAEPuqvU - Denon DN-X 1600 quirk: http://pastebin.com/raw/mpLUK01g - Yamaha 01v96i quirk: http://pastebin.com/raw/Qb0er0vc Regards, Damien

1 0

[alsa-devel] mcasp: stream has more channels (2) than are enabled in mcasp (0)
by Rick Mann 03 Jun '16

03 Jun '16

I've got a Beaglebone Green with a custom cape (CODEC and amp and some other stuff). Kernel 4.4.6-bone-rt-r6. When I load the overlay that should set up my CODEC in the device tree, I get the following in the log. [ 367.172337] davinci-mcasp 48038000.mcasp: stream has more channels (2) than are enabled in mcasp (0) [ 367.172374] davinci-mcasp 48038000.mcasp: ASoC: can't set 48038000.mcasp hw params: -22 You can see a little more log context, along with the DTS, here: http://pastebin.com/UPB1xjPF Can anyone tell me what this means? Thanks! -- Rick Mann rmann(a)latencyzero.com

3 14

[alsa-devel] [PATCH v3] ASoC: rt5514: add rt5514 SPI driver
by Oder Chiou 03 Jun '16

03 Jun '16

The device has multiple control interfaces, I2C and SPI. The I2C interface mainly controls the register settings of codec. The SPI interface is in order to provide the high speed transmission of data. For example, high bandwidth memory read/write of DSP. The patch adds the rt5514 SPI driver for loading the firmware of DSP and retrieving the voice data from DSP after the system is waked up by specific voice. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5514-spi.c | 457 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5514-spi.h | 38 ++++ sound/soc/codecs/rt5514.c | 134 ++++++++++++- sound/soc/codecs/rt5514.h | 4 + 6 files changed, 637 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/rt5514-spi.c create mode 100644 sound/soc/codecs/rt5514-spi.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c6f86ef..90749d2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -639,6 +639,9 @@ config SND_SOC_RT298 config SND_SOC_RT5514 tristate +config SND_SOC_RT5514_SPI + tristate + config SND_SOC_RT5616 tristate "Realtek RT5616 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 94873ee..9ab7016 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -99,6 +99,7 @@ snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt286-objs := rt286.o snd-soc-rt298-objs := rt298.o snd-soc-rt5514-objs := rt5514.o +snd-soc-rt5514-spi-objs := rt5514-spi.o snd-soc-rt5616-objs := rt5616.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o @@ -312,6 +313,7 @@ obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_RT298) += snd-soc-rt298.o obj-$(CONFIG_SND_SOC_RT5514) += snd-soc-rt5514.o +obj-$(CONFIG_SND_SOC_RT5514_SPI) += snd-soc-rt5514-spi.o obj-$(CONFIG_SND_SOC_RT5616) += snd-soc-rt5616.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o diff --git a/sound/soc/codecs/rt5514-spi.c b/sound/soc/codecs/rt5514-spi.c new file mode 100644 index 0000000..32b5fb6 --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.c @@ -0,0 +1,457 @@ +/* + * rt5514-spi.c -- RT5514 SPI driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/input.h> +#include <linux/spi/spi.h> +#include <linux/device.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/irq.h> +#include <linux/slab.h> +#include <linux/gpio.h> +#include <linux/sched.h> +#include <linux/kthread.h> +#include <linux/uaccess.h> +#include <linux/miscdevice.h> +#include <linux/regulator/consumer.h> +#include <linux/pm_qos.h> +#include <linux/sysfs.h> +#include <linux/clk.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5514-spi.h" + +static struct spi_device *rt5514_spi; + +struct rt5514_dsp { + struct device *dev; + struct delayed_work copy_work; + struct mutex dma_lock; + struct snd_pcm_substream *substream; + unsigned int buf_base, buf_limit, buf_rp; + size_t buf_size; + size_t dma_offset; + size_t dsp_offset; +}; + +static const struct snd_pcm_hardware rt5514_spi_pcm_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = PAGE_SIZE, + .period_bytes_max = 0x20000 / 8, + .periods_min = 8, + .periods_max = 8, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 0x20000, +}; + +static struct snd_soc_dai_driver rt5514_spi_dai = { + .name = "rt5514-dsp-cpu-dai", + .id = 0, + .capture = { + .stream_name = "DSP Capture", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, +}; + +static void rt5514_spi_copy_work(struct work_struct *work) +{ + struct rt5514_dsp *rt5514_dsp = + container_of(work, struct rt5514_dsp, copy_work.work); + struct snd_pcm_runtime *runtime = rt5514_dsp->substream->runtime; + size_t period_bytes, truncated_bytes = 0; + + mutex_lock(&rt5514_dsp->dma_lock); + if (!rt5514_dsp->substream) { + dev_err(rt5514_dsp->dev, "No pcm substream\n"); + goto done; + } + + period_bytes = snd_pcm_lib_period_bytes(rt5514_dsp->substream); + + if (rt5514_dsp->buf_size - rt5514_dsp->dsp_offset < period_bytes) + period_bytes = rt5514_dsp->buf_size - rt5514_dsp->dsp_offset; + + if (rt5514_dsp->buf_rp + period_bytes <= rt5514_dsp->buf_limit) { + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + period_bytes); + + if (rt5514_dsp->buf_rp + period_bytes == rt5514_dsp->buf_limit) + rt5514_dsp->buf_rp = rt5514_dsp->buf_base; + else + rt5514_dsp->buf_rp += period_bytes; + } else { + truncated_bytes = rt5514_dsp->buf_limit - rt5514_dsp->buf_rp; + rt5514_spi_burst_read(rt5514_dsp->buf_rp, + runtime->dma_area + rt5514_dsp->dma_offset, + truncated_bytes); + + rt5514_spi_burst_read(rt5514_dsp->buf_base, + runtime->dma_area + rt5514_dsp->dma_offset + + truncated_bytes, period_bytes - truncated_bytes); + + rt5514_dsp->buf_rp = rt5514_dsp->buf_base + + period_bytes - truncated_bytes; + } + + rt5514_dsp->dma_offset += period_bytes; + if (rt5514_dsp->dma_offset >= runtime->dma_bytes) + rt5514_dsp->dma_offset = 0; + + rt5514_dsp->dsp_offset += period_bytes; + + snd_pcm_period_elapsed(rt5514_dsp->substream); + + if (rt5514_dsp->dsp_offset < rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 5); +done: + mutex_unlock(&rt5514_dsp->dma_lock); +} + +/* PCM for streaming audio from the DSP buffer */ +static int rt5514_spi_pcm_open(struct snd_pcm_substream *substream) +{ + snd_soc_set_runtime_hwparams(substream, &rt5514_spi_pcm_hardware); + + return 0; +} + +static int rt5514_spi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + int ret; + + mutex_lock(&rt5514_dsp->dma_lock); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + rt5514_dsp->substream = substream; + mutex_unlock(&rt5514_dsp->dma_lock); + + return ret; +} + +static int rt5514_spi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + mutex_lock(&rt5514_dsp->dma_lock); + rt5514_dsp->substream = NULL; + mutex_unlock(&rt5514_dsp->dma_lock); + + return snd_pcm_lib_free_vmalloc_buffer(substream); +} + +static int rt5514_spi_prepare(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + u8 buf[8]; + + rt5514_dsp->dma_offset = 0; + rt5514_dsp->dsp_offset = 0; + + /** + * The address area x1800XXXX is the register address, and it cannot + * support spi burst read perfectly. So we use the spi burst read + * individually to make sure the data correctly. + */ + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_BASE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_base = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_LIMIT, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_limit = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_RP, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_rp = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + rt5514_spi_burst_read(RT5514_BUFFER_VOICE_SIZE, (u8 *)&buf, + sizeof(buf)); + rt5514_dsp->buf_size = buf[0] | buf[1] << 8 | buf[2] << 16 | + buf[3] << 24; + + return 0; +} + +static int rt5514_spi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + if (rt5514_dsp->buf_base && rt5514_dsp->buf_limit && + rt5514_dsp->buf_rp && rt5514_dsp->buf_size) + schedule_delayed_work(&rt5514_dsp->copy_work, 0); + } + + return 0; +} + +static snd_pcm_uframes_t rt5514_spi_pcm_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct rt5514_dsp *rt5514_dsp = + snd_soc_platform_get_drvdata(rtd->platform); + + return bytes_to_frames(runtime, rt5514_dsp->dma_offset); +} + +static struct snd_pcm_ops rt5514_spi_pcm_ops = { + .open = rt5514_spi_pcm_open, + .hw_params = rt5514_spi_hw_params, + .hw_free = rt5514_spi_hw_free, + .trigger = rt5514_spi_trigger, + .prepare = rt5514_spi_prepare, + .pointer = rt5514_spi_pcm_pointer, + .mmap = snd_pcm_lib_mmap_vmalloc, + .page = snd_pcm_lib_get_vmalloc_page, +}; + +static int rt5514_spi_pcm_probe(struct snd_soc_platform *platform) +{ + struct rt5514_dsp *rt5514_dsp; + + rt5514_dsp = devm_kzalloc(platform->dev, sizeof(*rt5514_dsp), + GFP_KERNEL); + + rt5514_dsp->dev = &rt5514_spi->dev; + mutex_init(&rt5514_dsp->dma_lock); + INIT_DELAYED_WORK(&rt5514_dsp->copy_work, rt5514_spi_copy_work); + snd_soc_platform_set_drvdata(platform, rt5514_dsp); + + return 0; +} + +static struct snd_soc_platform_driver rt5514_spi_platform = { + .probe = rt5514_spi_pcm_probe, + .ops = &rt5514_spi_pcm_ops, +}; + +static const struct snd_soc_component_driver rt5514_spi_dai_component = { + .name = "rt5514-spi-dai", +}; + +/** + * rt5514_spi_burst_read - Read data from SPI by rt5514 address. + * @addr: Start address. + * @rxbuf: Data Buffer for reading. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_READ; + int status; + u8 write_buf[8]; + unsigned int i, end, offset = 0; + + struct spi_message message; + struct spi_transfer x[3]; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + spi_message_init(&message); + memset(x, 0, sizeof(x)); + + x[0].len = 5; + x[0].tx_buf = write_buf; + spi_message_add_tail(&x[0], &message); + + x[1].len = 4; + x[1].tx_buf = write_buf; + spi_message_add_tail(&x[1], &message); + + x[2].len = end; + x[2].rx_buf = rxbuf + offset; + spi_message_add_tail(&x[2], &message); + + status = spi_sync(rt5514_spi, &message); + + if (status) + return false; + + offset += RT5514_SPI_BUF_LEN; + } + + for (i = 0; i < len; i += 8) { + write_buf[0] = rxbuf[i + 0]; + write_buf[1] = rxbuf[i + 1]; + write_buf[2] = rxbuf[i + 2]; + write_buf[3] = rxbuf[i + 3]; + write_buf[4] = rxbuf[i + 4]; + write_buf[5] = rxbuf[i + 5]; + write_buf[6] = rxbuf[i + 6]; + write_buf[7] = rxbuf[i + 7]; + + rxbuf[i + 0] = write_buf[7]; + rxbuf[i + 1] = write_buf[6]; + rxbuf[i + 2] = write_buf[5]; + rxbuf[i + 3] = write_buf[4]; + rxbuf[i + 4] = write_buf[3]; + rxbuf[i + 5] = write_buf[2]; + rxbuf[i + 6] = write_buf[1]; + rxbuf[i + 7] = write_buf[0]; + } + + return true; +} + +/** + * rt5514_spi_burst_write - Write data to SPI by rt5514 address. + * @addr: Start address. + * @txbuf: Data Buffer for writng. + * @len: Data length, it must be a multiple of 8. + * + * + * Returns true for success. + */ +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len) +{ + u8 spi_cmd = RT5514_SPI_CMD_BURST_WRITE; + u8 *write_buf; + unsigned int i, end, offset = 0; + + write_buf = kmalloc(RT5514_SPI_BUF_LEN + 6, GFP_KERNEL); + + if (write_buf == NULL) + return -ENOMEM; + + while (offset < len) { + if (offset + RT5514_SPI_BUF_LEN <= len) + end = RT5514_SPI_BUF_LEN; + else + end = len % RT5514_SPI_BUF_LEN; + + write_buf[0] = spi_cmd; + write_buf[1] = ((addr + offset) & 0xff000000) >> 24; + write_buf[2] = ((addr + offset) & 0x00ff0000) >> 16; + write_buf[3] = ((addr + offset) & 0x0000ff00) >> 8; + write_buf[4] = ((addr + offset) & 0x000000ff) >> 0; + + for (i = 0; i < end; i += 8) { + write_buf[i + 12] = txbuf[offset + i + 0]; + write_buf[i + 11] = txbuf[offset + i + 1]; + write_buf[i + 10] = txbuf[offset + i + 2]; + write_buf[i + 9] = txbuf[offset + i + 3]; + write_buf[i + 8] = txbuf[offset + i + 4]; + write_buf[i + 7] = txbuf[offset + i + 5]; + write_buf[i + 6] = txbuf[offset + i + 6]; + write_buf[i + 5] = txbuf[offset + i + 7]; + } + + write_buf[end + 5] = spi_cmd; + + spi_write(rt5514_spi, write_buf, end + 6); + + offset += RT5514_SPI_BUF_LEN; + } + + kfree(write_buf); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5514_spi_burst_write); + +static int rt5514_spi_probe(struct spi_device *spi) +{ + int ret; + + rt5514_spi = spi; + + ret = snd_soc_register_platform(&spi->dev, &rt5514_spi_platform); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register platform.\n"); + goto err_plat; + } + + ret = snd_soc_register_component(&spi->dev, &rt5514_spi_dai_component, + &rt5514_spi_dai, 1); + if (ret < 0) { + dev_err(&spi->dev, "Failed to register component.\n"); + goto err_comp; + } + + return 0; +err_comp: + snd_soc_unregister_platform(&spi->dev); +err_plat: + + return 0; +} + +static int rt5514_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_component(&spi->dev); + snd_soc_unregister_platform(&spi->dev); + + return 0; +} + +static const struct of_device_id rt5514_of_match[] = { + { .compatible = "realtek,rt5514", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5514_of_match); + +static struct spi_driver rt5514_spi_driver = { + .driver = { + .name = "rt5514", + .of_match_table = of_match_ptr(rt5514_of_match), + }, + .probe = rt5514_spi_probe, + .remove = rt5514_spi_remove, +}; +module_spi_driver(rt5514_spi_driver); + +MODULE_DESCRIPTION("RT5514 SPI driver"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5514-spi.h b/sound/soc/codecs/rt5514-spi.h new file mode 100644 index 0000000..f69b1cd --- /dev/null +++ b/sound/soc/codecs/rt5514-spi.h @@ -0,0 +1,38 @@ +/* + * rt5514-spi.h -- RT5514 driver + * + * Copyright 2015 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5514_SPI_H__ +#define __RT5514_SPI_H__ + +/** + * RT5514_SPI_BUF_LEN is the buffer size of SPI master controller. +*/ +#define RT5514_SPI_BUF_LEN 240 + +#define RT5514_BUFFER_VOICE_BASE 0x18001034 +#define RT5514_BUFFER_VOICE_LIMIT 0x18001038 +#define RT5514_BUFFER_VOICE_RP 0x1800103c +#define RT5514_BUFFER_VOICE_SIZE 0x18001040 + +/* SPI Command */ +enum { + RT5514_SPI_CMD_16_READ = 0, + RT5514_SPI_CMD_16_WRITE, + RT5514_SPI_CMD_32_READ, + RT5514_SPI_CMD_32_WRITE, + RT5514_SPI_CMD_BURST_READ, + RT5514_SPI_CMD_BURST_WRITE, +}; + +int rt5514_spi_burst_read(unsigned int addr, u8 *rxbuf, size_t len); +int rt5514_spi_burst_write(u32 addr, const u8 *txbuf, size_t len); + +#endif /* __RT5514_SPI_H__ */ diff --git a/sound/soc/codecs/rt5514.c b/sound/soc/codecs/rt5514.c index 879bf60..475b932 100644 --- a/sound/soc/codecs/rt5514.c +++ b/sound/soc/codecs/rt5514.c @@ -30,6 +30,9 @@ #include "rl6231.h" #include "rt5514.h" +#if defined(CONFIG_SND_SOC_RT5514_SPI) +#include "rt5514-spi.h" +#endif static const struct reg_sequence rt5514_i2c_patch[] = { {0x1800101c, 0x00000000}, @@ -110,6 +113,35 @@ static const struct reg_default rt5514_reg[] = { {RT5514_VENDOR_ID2, 0x10ec5514}, }; +static void rt5514_enable_dsp_prepare(struct rt5514_priv *rt5514) +{ + /* Reset */ + regmap_write(rt5514->i2c_regmap, 0x18002000, 0x000010ec); + /* LDO_I_limit */ + regmap_write(rt5514->i2c_regmap, 0x18002200, 0x00028604); + /* I2C bypass enable */ + regmap_write(rt5514->i2c_regmap, 0xfafafafa, 0x00000001); + /* mini-core reset */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x0005514b); + regmap_write(rt5514->i2c_regmap, 0x18002f00, 0x00055149); + /* I2C bypass disable */ + regmap_write(rt5514->i2c_regmap, 0xfafafafa, 0x00000000); + /* PIN config */ + regmap_write(rt5514->i2c_regmap, 0x18002070, 0x00000040); + /* PLL3(QN)=RCOSC*(10+2) */ + regmap_write(rt5514->i2c_regmap, 0x18002240, 0x0000000a); + /* PLL3 source=RCOSC, fsi=rt_clk */ + regmap_write(rt5514->i2c_regmap, 0x18002100, 0x0000000b); + /* Power on RCOSC, pll3 */ + regmap_write(rt5514->i2c_regmap, 0x18002004, 0x00808b81); + /* DSP clk source = pll3, ENABLE DSP clk */ + regmap_write(rt5514->i2c_regmap, 0x18002f08, 0x00000005); + /* Enable DSP clk auto switch */ + regmap_write(rt5514->i2c_regmap, 0x18001114, 0x00000001); + /* Reduce DSP power */ + regmap_write(rt5514->i2c_regmap, 0x18001118, 0x00000001); +} + static bool rt5514_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { @@ -248,6 +280,74 @@ static const DECLARE_TLV_DB_RANGE(bst_tlv, static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static int rt5514_dsp_voice_wake_up_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt5514->dsp_enabled; + + return 0; +} + +static int rt5514_dsp_voice_wake_up_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt5514_priv *rt5514 = snd_soc_component_get_drvdata(component); + struct snd_soc_codec *codec = rt5514->codec; + const struct firmware *fw = NULL; + + if (ucontrol->value.integer.value[0] == rt5514->dsp_enabled) + return 0; + + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { + rt5514->dsp_enabled = ucontrol->value.integer.value[0]; + + if (rt5514->dsp_enabled) { + rt5514_enable_dsp_prepare(rt5514); + + request_firmware(&fw, RT5514_FIRMWARE1, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ff60000, fw->data, + ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, "There is no SPI driver for" + " loading the firmware\n"); +#endif + release_firmware(fw); + fw = NULL; + } + + request_firmware(&fw, RT5514_FIRMWARE2, codec->dev); + if (fw) { +#if defined(CONFIG_SND_SOC_RT5514_SPI) + rt5514_spi_burst_write(0x4ffc0000, fw->data, + ((fw->size/8)+1)*8); +#else + dev_err(codec->dev, "There is no SPI driver for" + " loading the firmware\n"); +#endif + release_firmware(fw); + fw = NULL; + } + + /* DSP run */ + regmap_write(rt5514->i2c_regmap, 0x18002f00, + 0x00055148); + } else { + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + } + + return 0; +} + static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_TLV("MIC Boost Volume", RT5514_ANA_CTRL_MICBST, RT5514_SEL_BSTL_SFT, RT5514_SEL_BSTR_SFT, 8, 0, bst_tlv), @@ -257,6 +357,8 @@ static const struct snd_kcontrol_new rt5514_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC2 Capture Volume", RT5514_DOWNFILTER1_CTRL1, RT5514_DOWNFILTER1_CTRL2, RT5514_AD_GAIN_SFT, 127, 0, adc_vol_tlv), + SOC_SINGLE_EXT("DSP Voice Wake Up", SND_SOC_NOPM, 0, 1, 0, + rt5514_dsp_voice_wake_up_get, rt5514_dsp_voice_wake_up_put), }; /* ADC Mixer*/ @@ -365,6 +467,35 @@ static int rt5514_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int rt5514_pre_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5514_priv *rt5514 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + /** + * If the DSP is enabled in start of recording, the DSP + * should be disabled, and sync back to normal recording + * settings to make sure recording properly. + */ + if (rt5514->dsp_enabled) { + rt5514->dsp_enabled = 0; + regcache_mark_dirty(rt5514->i2c_regmap); + regcache_mark_dirty(rt5514->regmap); + regcache_sync(rt5514->i2c_regmap); + regcache_sync(rt5514->regmap); + } + break; + + default: + return 0; + } + + return 0; +} + static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Input Lines */ SND_SOC_DAPM_INPUT("DMIC1L"), @@ -472,6 +603,8 @@ static const struct snd_soc_dapm_widget rt5514_dapm_widgets[] = { /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_PRE("DAPM Pre", rt5514_pre_event), }; static const struct snd_soc_dapm_route rt5514_dapm_routes[] = { @@ -871,7 +1004,6 @@ static const struct regmap_config rt5514_i2c_regmap = { .reg_bits = 32, .val_bits = 32, - .max_register = RT5514_DSP_MAPPING | RT5514_VENDOR_ID2, .readable_reg = rt5514_i2c_readable_register, .cache_type = REGCACHE_NONE, diff --git a/sound/soc/codecs/rt5514.h b/sound/soc/codecs/rt5514.h index 6ad8a61..6e89e7d 100644 --- a/sound/soc/codecs/rt5514.h +++ b/sound/soc/codecs/rt5514.h @@ -225,6 +225,9 @@ #define RT5514_PLL_INP_MAX 40000000 #define RT5514_PLL_INP_MIN 256000 +#define RT5514_FIRMWARE1 "rt5514_dsp_fw1.bin" +#define RT5514_FIRMWARE2 "rt5514_dsp_fw2.bin" + /* System Clock Source */ enum { RT5514_SCLK_S_MCLK, @@ -247,6 +250,7 @@ struct rt5514_priv { int pll_src; int pll_in; int pll_out; + int dsp_enabled; }; #endif /* __RT5514_H__ */ -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH] ASoC: nau8825: correct typo in biquad filter coefficients
by John Hsu 03 Jun '16

03 Jun '16

There is typo in the name of biquad filter coefficients control. The patch is to fix the typo. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index dbb91aa..43cb677 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -386,7 +386,7 @@ static const struct snd_kcontrol_new nau8825_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), /* programmable biquad filter */ SOC_ENUM("BIQ Path Select", nau8825_biq_path_enum), - SND_SOC_BYTES_EXT("BIQ Coefficeints", 20, + SND_SOC_BYTES_EXT("BIQ Coefficients", 20, nau8825_biq_coeff_get, nau8825_biq_coeff_put), }; -- 2.6.4

2 1

[alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Kai Chieh Chuang 03 Jun '16

03 Jun '16

Hi, Let me illustrate the system first. FE1 -----+----> BE | FE2 -----+ When there is multi-FE stream to a single BE, the BE will receive SND_SOC_DAPM_STREAM_STOP if one of the FE shutdown. However, the BE is stilled used by another, this will let BE cpu_dai and codec_dai think they are inactive. The later power check will go wrong. I have check the latest linux version, and didn't find any related change. The following is my change to address this, and will re-send a formal patch if it's ok. diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 70e8088..9c2d159 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -163,6 +163,13 @@ int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, dev_dbg(be->dev, "ASoC: BE %s event %d dir %d\n", be->dai_link->name, event, dir); + /* don't sent stop event if this BE is used by other FE */ + if (event == SND_SOC_DAPM_STREAM_STOP && + be->dpcm[dir].users >= 1) { + pr_warn("kc, %s(), be->dai_link->name %s skip stop event\n", __func__, be->dai_link->name); + continue; + } + snd_soc_dapm_stream_event(be, dir, event); } Sincerely, Kai Chieh

2 1

[alsa-devel] [PATCH] ASoC: fsl: fix build failure
by Sudip Mukherjee 03 Jun '16

03 Jun '16

m32r allmodconfig build is failing with the error: ERROR: "bad_dma_ops" [sound/soc/fsl/snd-soc-fsl-asrc.ko] undefined! The code is using DMA but the related dependency is not mentioned in the Kconfig. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- build log is at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/134651165 sound/soc/fsl/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 35aabf9..19bdcac 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -4,6 +4,7 @@ comment "Common SoC Audio options for Freescale CPUs:" config SND_SOC_FSL_ASRC tristate "Asynchronous Sample Rate Converter (ASRC) module support" + depends on HAS_DMA select REGMAP_MMIO select SND_SOC_GENERIC_DMAENGINE_PCM help -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: mediatek: add MCLK source selection
by PC Liao 02 Jun '16

02 Jun '16

The new machine's MCLK source is from mt8173 which is dynamic from sampling rate*256. This patch provides the selection for device tree. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +++ sound/soc/mediatek/mt8173-rt5650.c | 45 +++++++++++++++++++- 2 files changed, 48 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt index 5bfa6b6..f250fc7 100644 --- a/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt +++ b/Documentation/devicetree/bindings/sound/mt8173-rt5650.txt @@ -12,12 +12,17 @@ Required codec-capture subnode properties: <&rt5650 0> : Default setting. Connect rt5650 I2S1 for capture. (dai_name = rt5645-aif1) <&rt5650 1> : Connect rt5650 I2S2 for capture. (dai_name = rt5645-aif2) +- mediatek,mclk: the MCLK source + 0 : external oscillator, MCLK = 12.288M + 1 : internal source from mt8173, MCLK = sampling rate*256 + Example: sound { compatible = "mediatek,mt8173-rt5650"; mediatek,audio-codec = <&rt5650>; mediatek,platform = <&afe>; + mediatek,mclk = <0>; codec-capture { sound-dai = <&rt5650 1>; }; diff --git a/sound/soc/mediatek/mt8173-rt5650.c b/sound/soc/mediatek/mt8173-rt5650.c index a27a667..072934b 100644 --- a/sound/soc/mediatek/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173-rt5650.c @@ -23,6 +23,20 @@ #define MCLK_FOR_CODECS 12288000 +enum mt8173_rt5650_mclk { + MT8173_RT5650_MCLK_EXTERNAL = 0, + MT8173_RT5650_MCLK_INTERNAL, +}; + +struct mt8173_rt5650_platform_data { + enum mt8173_rt5650_mclk pll_from; + /* 0 = external oscillator; 1 = internal source from mt8173 */ +}; + +static struct mt8173_rt5650_platform_data mt8173_rt5650_priv = { + .pll_from = MT8173_RT5650_MCLK_EXTERNAL, +}; + static const struct snd_soc_dapm_widget mt8173_rt5650_widgets[] = { SND_SOC_DAPM_SPK("Speaker", NULL), SND_SOC_DAPM_MIC("Int Mic", NULL), @@ -54,13 +68,29 @@ static int mt8173_rt5650_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + unsigned int mclk_clock; int i, ret; + switch (mt8173_rt5650_priv.pll_from) { + case MT8173_RT5650_MCLK_EXTERNAL: + /* mclk = 12.288M */ + mclk_clock = MCLK_FOR_CODECS; + break; + case MT8173_RT5650_MCLK_INTERNAL: + /* mclk = sampling rate*256 */ + mclk_clock = params_rate(params) * 256; + break; + default: + /* mclk = 12.288M */ + mclk_clock = MCLK_FOR_CODECS; + break; + } + for (i = 0; i < rtd->num_codecs; i++) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - /* pll from mclk 12.288M */ - ret = snd_soc_dai_set_pll(codec_dai, 0, 0, MCLK_FOR_CODECS, + /* pll from mclk */ + ret = snd_soc_dai_set_pll(codec_dai, 0, 0, mclk_clock, params_rate(params) * 512); if (ret) return ret; @@ -243,6 +273,17 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) mt8173_rt5650_codecs[1].dai_name = codec_capture_dai; } + if (device_property_present(&pdev->dev, "mediatek,mclk")) { + ret = device_property_read_u32(&pdev->dev, + "mediatek,mclk", + &mt8173_rt5650_priv.pll_from); + if (ret) { + dev_err(&pdev->dev, + "%s snd_soc_register_card fail %d\n", + __func__, ret); + } + } + card->dev = &pdev->dev; platform_set_drvdata(pdev, card); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: hdmi-codec: select CONFIG_HDMI
by Arnd Bergmann 02 Jun '16

02 Jun '16

SND_SOC_HDMI_CODEC can be enabled without HDMI support, leading to a link error: In function `hdmi_codec_hw_params': sound/soc/codecs/hdmi-codec.c:188: undefined reference to `hdmi_audio_infoframe_init' sound/built-in.o:(.debug_addr+0x1a5c0): undefined reference to `hdmi_audio_infoframe_init' This changes the Kconfig file to select HDMI, as the other codec using hdmi_audio_infoframe_init already does. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> Fixes: 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") --- sound/soc/codecs/Kconfig | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 4d82a58ff6b0..f3fb98f0a995 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -483,9 +483,10 @@ config SND_SOC_DMIC tristate config SND_SOC_HDMI_CODEC - tristate - select SND_PCM_ELD - select SND_PCM_IEC958 + tristate + select SND_PCM_ELD + select SND_PCM_IEC958 + select HDMI config SND_SOC_ES8328 tristate "Everest Semi ES8328 CODEC" -- 2.7.0

3 2

[alsa-devel] [PATCH 1/2] ASoC: Add ADAU7002 Stereo PDM-to-I2S/TDM Converter DT bindings
by Lars-Peter Clausen 02 Jun '16

02 Jun '16

The ADAU7002 takes a stereo PDM signal (e.g. from two digital microphones) and converts it into a I2S/TDM PCM stream. The chip does not have a control interface and has a single power supply that is modeled in the devicetree. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- .../devicetree/bindings/sound/adi,adau7002.txt | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/adi,adau7002.txt diff --git a/Documentation/devicetree/bindings/sound/adi,adau7002.txt b/Documentation/devicetree/bindings/sound/adi,adau7002.txt new file mode 100644 index 0000000..f144ee1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/adi,adau7002.txt @@ -0,0 +1,19 @@ +Analog Devices ADAU7002 Stereo PDM-to-I2S/TDM Converter + +Required properties: + + - compatible: Must be "adi,adau7002" + +Optional properties: + + - IOVDD-supply: Phandle and specifier for the power supply providing the IOVDD + supply as covered in Documentation/devicetree/bindings/regulator/regulator.txt + + If this property is not present it is assumed that the supply pin is + hardwired to always on. + +Example: + adau7002: pdm-to-i2s { + compatible = "adi,adau7002"; + IOVDD-supply = <&supply>; + }; -- 2.1.4

2 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix dra7 DMA offset when using CFG port
by Peter Ujfalusi 02 Jun '16

02 Jun '16

The TX and RX offset is different for each serializers when using the CFG port for DMA access. When using the CFG port only one serializer can be used per direction so print error message and only configure the first serializer's offset. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Suggested-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++++++++++++++++--- sound/soc/davinci/davinci-mcasp.h | 4 +-- 2 files changed, 54 insertions(+), 6 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e060798d94cb..f8cb3c7d3c45 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1534,8 +1534,9 @@ static struct davinci_mcasp_pdata am33xx_mcasp_pdata = { }; static struct davinci_mcasp_pdata dra7_mcasp_pdata = { - .tx_dma_offset = 0x200, - .rx_dma_offset = 0x284, + /* The CFG port offset will be calculated if it is needed */ + .tx_dma_offset = 0, + .rx_dma_offset = 0, .version = MCASP_VERSION_4, }; @@ -1755,6 +1756,52 @@ static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp) return PCM_EDMA; } +static u32 davinci_mcasp_txdma_offset(struct davinci_mcasp_pdata *pdata) +{ + int i; + u32 offset = 0; + + if (pdata->version != MCASP_VERSION_4) + return pdata->tx_dma_offset; + + for (i = 0; i < pdata->num_serializer; i++) { + if (pdata->serial_dir[i] == TX_MODE) { + if (!offset) { + offset = DAVINCI_MCASP_TXBUF_REG(i); + } else { + pr_err("%s: Only one serializer allowed!\n", + __func__); + break; + } + } + } + + return offset; +} + +static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata) +{ + int i; + u32 offset = 0; + + if (pdata->version != MCASP_VERSION_4) + return pdata->rx_dma_offset; + + for (i = 0; i < pdata->num_serializer; i++) { + if (pdata->serial_dir[i] == RX_MODE) { + if (!offset) { + offset = DAVINCI_MCASP_RXBUF_REG(i); + } else { + pr_err("%s: Only one serializer allowed!\n", + __func__); + break; + } + } + } + + return offset; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1883,7 +1930,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) dma_data->addr = dat->start; else - dma_data->addr = mem->start + pdata->tx_dma_offset; + dma_data->addr = mem->start + davinci_mcasp_txdma_offset(pdata); dma = &mcasp->dma_request[SNDRV_PCM_STREAM_PLAYBACK]; res = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -1904,7 +1951,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) if (dat) dma_data->addr = dat->start; else - dma_data->addr = mem->start + pdata->rx_dma_offset; + dma_data->addr = + mem->start + davinci_mcasp_rxdma_offset(pdata); dma = &mcasp->dma_request[SNDRV_PCM_STREAM_CAPTURE]; res = platform_get_resource(pdev, IORESOURCE_DMA, 1); diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index a3be108a8c17..10b52d6bdeca 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -85,9 +85,9 @@ (n << 2)) /* Transmit Buffer for Serializer n */ -#define DAVINCI_MCASP_TXBUF_REG 0x200 +#define DAVINCI_MCASP_TXBUF_REG(n) (0x200 + (n << 2)) /* Receive Buffer for Serializer n */ -#define DAVINCI_MCASP_RXBUF_REG 0x280 +#define DAVINCI_MCASP_RXBUF_REG(n) (0x280 + (n << 2)) /* McASP FIFO Registers */ #define DAVINCI_MCASP_V2_AFIFO_BASE (0x1010) -- 2.8.3

2 2

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Use a copy of pdata per instance during DT boot
by Peter Ujfalusi 02 Jun '16

02 Jun '16

Instead of modifying the static pdata struct per McASP instance we need to allocate pdata for each McASP. This way we can avoid configuration leakage from prior McASP to McASP drivers probed at later time. Reported-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f8cb3c7d3c45..e10cfda21d89 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1620,7 +1620,14 @@ static struct davinci_mcasp_pdata *davinci_mcasp_set_pdata_from_of( pdata = pdev->dev.platform_data; return pdata; } else if (match) { - pdata = (struct davinci_mcasp_pdata*) match->data; + pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata), + GFP_KERNEL); + if (!pdata) { + dev_err(&pdev->dev, + "Failed to allocate memory for pdata\n"); + ret = -ENOMEM; + return pdata; + } } else { /* control shouldn't reach here. something is wrong */ ret = -EINVAL; -- 2.8.3

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: Place optional properties in the correct section
by Fabio Estevam 02 Jun '16

02 Jun '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> 'micbias-resistor-k-ohms' and 'micbias-voltage-m-volts' are optional properties, so move them below the 'Optional properties' line. While at it, fix a typo in 'mentioned'. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 0e5e4eb..5666da7 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -7,6 +7,14 @@ Required properties: - clocks : the clock provider of SYS_MCLK +- VDDA-supply : the regulator provider of VDDA + +- VDDIO-supply: the regulator provider of VDDIO + +Optional properties: + +- VDDD-supply : the regulator provider of VDDD + - micbias-resistor-k-ohms : the bias resistor to be used in kOmhs The resistor can take values of 2k, 4k or 8k. If set to 0 it will be off. @@ -15,17 +23,9 @@ Required properties: - micbias-voltage-m-volts : the bias voltage to be used in mVolts The voltage can take values from 1.25V to 3V by 250mV steps - If this node is not mentionned or the value is unknown, then + If this node is not mentioned or the value is unknown, then the value is set to 1.25V. -- VDDA-supply : the regulator provider of VDDA - -- VDDIO-supply: the regulator provider of VDDIO - -Optional properties: - -- VDDD-supply : the regulator provider of VDDD - Example: codec: sgtl5000@0a { -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs53l30: Rename the volume controls for preamplifier
by Nicolin Chen 02 Jun '16

02 Jun '16

Volume controls should end with 'Volume', so this patch renames them for ADC preamplifier. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/codecs/cs53l30.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c index 714e579..9aff449 100644 --- a/sound/soc/codecs/cs53l30.c +++ b/sound/soc/codecs/cs53l30.c @@ -331,10 +331,10 @@ static const struct snd_kcontrol_new cs53l30_snd_controls[] = { SOC_SINGLE_TLV("ADC2 NG Boost Volume", CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_BOOST_SHIFT, 1, 0, adc_ng_boost_tlv), - SOC_DOUBLE_R_TLV("ADC1 Pre Amp Gain", CS53L30_ADC1A_AFE_CTL, + SOC_DOUBLE_R_TLV("ADC1 Preamplifier Volume", CS53L30_ADC1A_AFE_CTL, CS53L30_ADC1B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT, 2, 0, pga_preamp_tlv), - SOC_DOUBLE_R_TLV("ADC2 Pre Amp Gain", CS53L30_ADC2A_AFE_CTL, + SOC_DOUBLE_R_TLV("ADC2 Preamplifier Volume", CS53L30_ADC2A_AFE_CTL, CS53L30_ADC2B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT, 2, 0, pga_preamp_tlv), -- 2.1.4

3 5

[alsa-devel] hda-intel: ALC1150 (alc882 patch) produces constant noise on line-out port
by Maximilian.ammann 02 Jun '16

02 Jun '16

Hello, I experienced weird behaviour on my "ASUS MAXIMUS VII RANGER" which has a realtek soundcard with the ALC1150 codec. The problem is that if the patch_882 is enabled I get a constant noise on the line-out port. I started experienced and noticed that when using the generic model the problem disappeared. Therefore the problem also disappeared when removing the folowing line in sound/pci/hda/patch_realtek.c: HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882), I really would like to know whether this behaviour is expected or how I (myself) could fix it by creating a patch. alsa-info.sh output: Without alc882 patch: http://www.alsa-project.org/db/?f=3fdf926c64a02273e0e68035eeb94282ca44a 7df With alc882 patch: http://www.alsa-project.org/db/?f=336755aab10811bb0af5440dbda7b9cda3d9b 116 Thansk in advance, if possible it would be awesome if you could provide me some tips since I really would like to fix it myself as my very first patch. Max

2 2

Re: [alsa-devel] Applied "ASoC: Intel: add bytct-rt5651 machine driver" to the asoc tree
by Pietro 02 Jun '16

02 Jun '16

Hi, I'm an owner of a 2 in 1 Netbook with an Atom X5 Z8300 SoC with rt5651 codec (Cube iWork 11). To get audio working I Added these lines of code to the ./soc/intel/atom/sst/sst_acpi.c file: --- ./soc/intel/atom/sst/sst_acpi.c.old 2016-05-31 09:05:40.045682194 +0200 +++ ./soc/intel/atom/sst/sst_acpi.c 2016-05-31 09:07:26.829324148 +0200 @@ -342,9 +342,12 @@ &chv_platform_data }, {"193C9890", "cht-bsw-max98090", "intel/fw_sst_22a8.bin", "cht-bsw", NULL, &chv_platform_data }, + {"10EC5651", "bytcr_rt5651", "intel/fw_sst_0f28.bin", "bytcr_rt5651", NULL, + &byt_rvp_platform_data }, {}, }; + static const struct acpi_device_id sst_acpi_ids[] = { { "80860F28", (unsigned long)&sst_acpi_bytcr}, { "808622A8", (unsigned long) &sst_acpi_chv}, Then i added the UCM files linked in these archive: https://github.com/plbossart/UCM/tree/master/bytcr-rt5651 With this configuration, audio seems to work, but is 'slow': Playback speed is at slow motion, and seconds move slowly (Tried with aplay and audacious with many sound files, using directly Alsa and Pulseaudio with no differencies). This seems to be a DSP Clock problem. Have you got any suggestions to solve this problem? Thanks, Pietro

2 10

[alsa-devel] [PATCH] ASoC: tlv320aix31xx: Add ACPI match for Lenovo 100S
by Bastien Nocera 01 Jun '16

01 Jun '16

The Lenovo 100S netbook has a codec controller for which there is a driver, but doesn't know how to access the device. This adds the necessary ACPI table for the driver to find the device. Device (TTLV) { Name (_ADR, Zero) // _ADR: Address Name (_HID, "10TI3100") // _HID: Hardware ID Name (_CID, "10TI3100") // _CID: Compatible ID Name (_DDN, "TI TLV320AIC3100 Codec Controller ") // _DDN: DOS Device Name Name (_UID, One) // _UID: Unique ID Signed-off-by: Bastien Nocera <hadess(a)hadess.net> Tested-by: Jan Schmidt <jan(a)centricular.com> --- sound/soc/codecs/tlv320aic31xx.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index ee4def4..3c5e1df 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -28,6 +28,7 @@ #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/acpi.h> #include <linux/of.h> #include <linux/of_gpio.h> #include <linux/slab.h> @@ -1280,10 +1281,19 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +#ifdef CONFIG_ACPI +static const struct acpi_device_id aic31xx_acpi_match[] = { + { "10TI3100", 0 }, + { } +}; +MODULE_DEVICE_TABLE(acpi, aic31xx_acpi_match); +#endif + static struct i2c_driver aic31xx_i2c_driver = { .driver = { .name = "tlv320aic31xx-codec", .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + .acpi_match_table = ACPI_PTR(aic31xx_acpi_match), }, .probe = aic31xx_i2c_probe, .remove = aic31xx_i2c_remove, -- 2.7.3

3 7

[alsa-devel] [PATCH v2] ASoC: tas571x: add input channel mixer for TAS5717/19
by Petr Kulhavy 01 Jun '16

01 Jun '16

Add channel 1 and 2 input mixer registers and the related ALSA mixer controls for TAS5717/19 chips. The mixer control coefficients on the chip are linear in the range -3.99999 to +3.99999, encoded in 3.23 number format. In this patch the mixer controls are limited to 128 values from 0.0 to 1.0 in 1/64 steps. Signed-off-by: Petr Kulhavy <brain(a)jikos.cz> --- v1: initial v2: re-apply on topics/tas571x sound/soc/codecs/tas571x.c | 18 ++++++++++++++++++ sound/soc/codecs/tas571x.h | 5 +++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index bc1fbaf..d8baca3 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -64,6 +64,10 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_INPUT_MUX_REG: case TAS571X_CH4_SRC_SELECT_REG: case TAS571X_PWM_MUX_REG: + case TAS5717_CH1_RIGHT_CH_MIX_REG: + case TAS5717_CH1_LEFT_CH_MIX_REG: + case TAS5717_CH2_LEFT_CH_MIX_REG: + case TAS5717_CH2_RIGHT_CH_MIX_REG: return 4; default: return 1; @@ -397,6 +401,16 @@ static const struct snd_kcontrol_new tas5711_controls[] = { TAS571X_SOFT_MUTE_REG, TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, 1, 1), + + SOC_DOUBLE_R_RANGE("CH1 Mixer Volume", + TAS5717_CH1_LEFT_CH_MIX_REG, + TAS5717_CH1_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), + + SOC_DOUBLE_R_RANGE("CH2 Mixer Volume", + TAS5717_CH2_LEFT_CH_MIX_REG, + TAS5717_CH2_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), }; static const struct regmap_range tas571x_readonly_regs_range[] = { @@ -520,6 +534,10 @@ static const struct reg_default tas5717_reg_defaults[] = { { 0x08, 0x00c0 }, { 0x09, 0x00c0 }, { 0x1b, 0x82 }, + { TAS5717_CH1_RIGHT_CH_MIX_REG, 0x0 }, + { TAS5717_CH1_LEFT_CH_MIX_REG, 0x800000}, + { TAS5717_CH2_LEFT_CH_MIX_REG, 0x0 }, + { TAS5717_CH2_RIGHT_CH_MIX_REG, 0x800000}, }; static const struct regmap_config tas5717_regmap_config = { diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index bf4d436..c45677b 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -87,4 +87,9 @@ #define TAS5717_CH3_BQ0_REG 0x5e #define TAS5717_CH3_BQ1_REG 0x5f +#define TAS5717_CH1_RIGHT_CH_MIX_REG 0x72 +#define TAS5717_CH1_LEFT_CH_MIX_REG 0x73 +#define TAS5717_CH2_LEFT_CH_MIX_REG 0x76 +#define TAS5717_CH2_RIGHT_CH_MIX_REG 0x77 + #endif /* _TAS571X_H */ -- 2.5.0

2 1

[alsa-devel] [REPOST PATCH 0/2]ASoC: Intel: Add Broxton-P Dialog Maxim machine driver
by sathyanarayana.nujella＠intel.com 01 Jun '16

01 Jun '16

From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> Reposting this patchset as some recipents didn't get and also alsa-devel rejected. This patch set adds the Intel Broxton-P machine driver and updates machine table. Sathyanarayana Nujella (2): ASoC: Intel: Add Broxton-P Dialog Maxim machine driver ASoC: Intel: Add Broxton-P Dialog+Maxim machine driver entry sound/soc/intel/Kconfig | 16 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bxt_da7219_max98357a.c | 460 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 1 + 4 files changed, 479 insertions(+) create mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c -- 2.8.3

2 4

[alsa-devel] [RESEND PATCH 0/6] Rockchip Type-C and DispplayPort driver
by Chris Zhong 01 Jun '16

01 Jun '16

Sorry, forgot a head file in patch 1, so resend. Hi all This series patch is for rockchip Type-C phy and DisplayPort controller driver. The USB Type-C PHY is designed to support the USB3 and DP applications. The PHY basically has two main components: USB3 and DisplyPort. USB3 operates in SuperSpeed mode and the DP can operate at RBR, HBR and HBR2 data rates. The Type-C cable orientation detection and Power Delivery (PD) is accomplished using a PD PHY or a exernal PD chip. The DP controller is compliant with DisplayPort Specification, Version 1.3, This IP is compatible with the rockchip type-c PHY IP. There is a uCPU in DP controller, it need a firmware to work, please put the firmware file to /lib/firmware/cdn/dptx.bin. The uCPU in charge of aux communication and link training, the host use mailbox to communicate with the ucpu. The PHY driver has register a notification, to get the alt mode from PD, the PD driver need call the tcphy_notifier_call_chain to notify PHY and DP controller. This series is based on Mark Yao's branch: https://github.com/markyzq/kernel-drm-rockchip/tree/drm-rockchip-next-2016-… I test this patches on the rk3399-evb board, with a fusb302 driver, this branch has no rk3399.dtsi, so the patch about dts is not included in this series. Chris Zhong (6): phy: Add USB Type-C PHY driver for rk3399 Documentation: bindings: add dt doc for Rockchip USB Type-C PHY drm/rockchip: vop: add cdn DP support for rk3399 Documentation: bindings: add dt documentation for cdn DP controller ASoC: cdn-dp: Add cdn DP codec driver ASoC: rockchip: Add machine driver for cdn dp codec .../bindings/display/rockchip/cdn-dp-rockchip.txt | 57 ++ .../devicetree/bindings/phy/phy-rockchip-typec.txt | 55 ++ .../bindings/sound/rockchip-cdn-dp-audio.txt | 12 + drivers/gpu/drm/rockchip/Kconfig | 9 + drivers/gpu/drm/rockchip/Makefile | 1 + drivers/gpu/drm/rockchip/cdn-dp-core.c | 620 ++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-core.h | 95 +++ drivers/gpu/drm/rockchip/cdn-dp-reg.c | 730 ++++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-reg.h | 404 ++++++++++ drivers/gpu/drm/rockchip/rockchip_drm_vop.c | 9 +- drivers/gpu/drm/rockchip/rockchip_drm_vop.h | 2 + drivers/gpu/drm/rockchip/rockchip_vop_reg.c | 2 + drivers/phy/Kconfig | 7 + drivers/phy/Makefile | 1 + drivers/phy/phy-rockchip-typec.c | 823 +++++++++++++++++++++ include/linux/phy/phy-rockchip-typec.h | 72 ++ include/sound/cdn-dp-audio.h | 51 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cdn-dp-audio.c | 246 ++++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip-cdn-dp-audio.c | 167 +++++ 23 files changed, 3378 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/display/rockchip/cdn-dp-rockchip.txt create mode 100644 Documentation/devicetree/bindings/phy/phy-rockchip-typec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-cdn-dp-audio.txt create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-core.c create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-core.h create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-reg.c create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-reg.h create mode 100644 drivers/phy/phy-rockchip-typec.c create mode 100644 include/linux/phy/phy-rockchip-typec.h create mode 100644 include/sound/cdn-dp-audio.h create mode 100644 sound/soc/codecs/cdn-dp-audio.c create mode 100644 sound/soc/rockchip/rockchip-cdn-dp-audio.c -- 2.6.3

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[alsa-devel] [PATCH v7] ASoC: cs53l30: Add codec driver support for Cirrus CS53L30
by Nicolin Chen 01 Jun '16

01 Jun '16

CS53L30 is a Quad-Channel ADC from Cirrus Logic with an I2S/TDM DAI. So this patch adds a codec driver for CS53L30 that includes 4-channel 24-bit recording and TDM mode supports. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog: v6->v7 * Added vendor prefixes to properties in DT binding doc * Updated the property names in the driver accordingly * Added two required power supplies in the example * Refined two dev_err messages * Re-ordered the header files * Revised the commit log v5->v6 * Added clock controls * Reconstructed bitfield macros and improved coding style * Fixed some misuses of regmap_update_bits() * Added TLV volume controls to replace enums for preamplifier * Added micbias level in DT instead of exposing to user-space * Simplified SDOUTx controls by removing cs53l30_asp_sdout_event() (Still be able to use set_tristate() instead) * Removed useless header files and sort the rest alphabetically * Registered runtime pm functions to the driver * Added regulator controls (VA and VP) * Added writeable register function for regmap * Merged two ASP dais * Added TDM support * Added a simple introduction in commit log .../devicetree/bindings/sound/cs53l30.txt | 40 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs53l30.c | 1097 ++++++++++++++++++++ sound/soc/codecs/cs53l30.h | 458 ++++++++ 5 files changed, 1603 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs53l30.txt create mode 100644 sound/soc/codecs/cs53l30.c create mode 100644 sound/soc/codecs/cs53l30.h diff --git a/Documentation/devicetree/bindings/sound/cs53l30.txt b/Documentation/devicetree/bindings/sound/cs53l30.txt new file mode 100644 index 0000000..18d6b99 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,40 @@ +CS53L30 audio CODEC + +Required properties: + + - compatible : "cirrus,cs53l30" + + - reg : the I2C address of the device + + - VA-supply, VP-supply : power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt. + +Optional properties: + + - reset-gpios : a GPIO spec for the reset pin. + + - cirrus,micbias-lvl : Set the output voltage level on the MICBIAS Pin. + 0 = Hi-Z + 1 = 1.80 V + 2 = 2.75 V + + - cirrus,use-sdout2 : This is a boolean property. If present, it indicates + the hardware design connects both SDOUT1 and SDOUT2 + pins to output data. Otherwise, it indicates that + only SDOUT1 is connected for data output. + * CS53l30 supports 4-channel data output in the same + * frame using two different ways: + * 1) Normal I2S mode on two data pins -- each SDOUT + * carries 2-channel data in the same time. + * 2) TDM mode on one signle data pin -- SDOUT1 carries + * 4-channel data per frame. + +Example: + +codec: cs53l30@48 { + compatible = "cirrus,cs53l30"; + reg = <0x48>; + reset-gpios = <&gpio 54 0>; + VA-supply = <&cs53l30_va>; + VP-supply = <&cs53l30_vp>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3afae9..970ca04 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -56,6 +56,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CS4349 if I2C select SND_SOC_CS47L24 if MFD_CS47L24 + select SND_SOC_CS53L30 if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI select SND_SOC_DA7213 if I2C @@ -444,6 +445,11 @@ config SND_SOC_CS4349 config SND_SOC_CS47L24 tristate +# Cirrus Logic Quad-Channel ADC +config SND_SOC_CS53L30 + tristate "Cirrus Logic CS53L30 CODEC" + depends on I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b7b9941..f38428c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-cs42xx8-objs := cs42xx8.o snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cs4349-objs := cs4349.o snd-soc-cs47l24-objs := cs47l24.o +snd-soc-cs53l30-objs := cs53l30.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -259,6 +260,7 @@ obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o obj-$(CONFIG_SND_SOC_CS47L24) += snd-soc-cs47l24.o +obj-$(CONFIG_SND_SOC_CS53L30) += snd-soc-cs53l30.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs53l30.c b/sound/soc/codecs/cs53l30.c new file mode 100644 index 0000000..714e579 --- /dev/null +++ b/sound/soc/codecs/cs53l30.c @@ -0,0 +1,1097 @@ +/* + * cs53l30.c -- CS53l30 ALSA Soc Audio driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Authors: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/i2c.h> +#include <linux/module.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include "cs53l30.h" + +#define CS53L30_NUM_SUPPLIES 2 +static const char *const cs53l30_supply_names[CS53L30_NUM_SUPPLIES] = { + "VA", + "VP", +}; + +struct cs53l30_private { + struct regulator_bulk_data supplies[CS53L30_NUM_SUPPLIES]; + struct regmap *regmap; + struct gpio_desc *reset_gpio; + struct clk *mclk; + bool use_sdout2; + u32 mclk_rate; +}; + +static const struct reg_default cs53l30_reg_defaults[] = { + { CS53L30_PWRCTL, CS53L30_PWRCTL_DEFAULT }, + { CS53L30_MCLKCTL, CS53L30_MCLKCTL_DEFAULT }, + { CS53L30_INT_SR_CTL, CS53L30_INT_SR_CTL_DEFAULT }, + { CS53L30_MICBIAS_CTL, CS53L30_MICBIAS_CTL_DEFAULT }, + { CS53L30_ASPCFG_CTL, CS53L30_ASPCFG_CTL_DEFAULT }, + { CS53L30_ASP_CTL1, CS53L30_ASP_CTL1_DEFAULT }, + { CS53L30_ASP_TDMTX_CTL1, CS53L30_ASP_TDMTX_CTLx_DEFAULT }, + { CS53L30_ASP_TDMTX_CTL2, CS53L30_ASP_TDMTX_CTLx_DEFAULT }, + { CS53L30_ASP_TDMTX_CTL3, CS53L30_ASP_TDMTX_CTLx_DEFAULT }, + { CS53L30_ASP_TDMTX_CTL4, CS53L30_ASP_TDMTX_CTLx_DEFAULT }, + { CS53L30_ASP_TDMTX_EN1, CS53L30_ASP_TDMTX_ENx_DEFAULT }, + { CS53L30_ASP_TDMTX_EN2, CS53L30_ASP_TDMTX_ENx_DEFAULT }, + { CS53L30_ASP_TDMTX_EN3, CS53L30_ASP_TDMTX_ENx_DEFAULT }, + { CS53L30_ASP_TDMTX_EN4, CS53L30_ASP_TDMTX_ENx_DEFAULT }, + { CS53L30_ASP_TDMTX_EN5, CS53L30_ASP_TDMTX_ENx_DEFAULT }, + { CS53L30_ASP_TDMTX_EN6, CS53L30_ASP_TDMTX_ENx_DEFAULT }, + { CS53L30_ASP_CTL2, CS53L30_ASP_CTL2_DEFAULT }, + { CS53L30_SFT_RAMP, CS53L30_SFT_RMP_DEFAULT }, + { CS53L30_LRCK_CTL1, CS53L30_LRCK_CTLx_DEFAULT }, + { CS53L30_LRCK_CTL2, CS53L30_LRCK_CTLx_DEFAULT }, + { CS53L30_MUTEP_CTL1, CS53L30_MUTEP_CTL1_DEFAULT }, + { CS53L30_MUTEP_CTL2, CS53L30_MUTEP_CTL2_DEFAULT }, + { CS53L30_INBIAS_CTL1, CS53L30_INBIAS_CTL1_DEFAULT }, + { CS53L30_INBIAS_CTL2, CS53L30_INBIAS_CTL2_DEFAULT }, + { CS53L30_DMIC1_STR_CTL, CS53L30_DMIC1_STR_CTL_DEFAULT }, + { CS53L30_DMIC2_STR_CTL, CS53L30_DMIC2_STR_CTL_DEFAULT }, + { CS53L30_ADCDMIC1_CTL1, CS53L30_ADCDMICx_CTL1_DEFAULT }, + { CS53L30_ADCDMIC1_CTL2, CS53L30_ADCDMIC1_CTL2_DEFAULT }, + { CS53L30_ADC1_CTL3, CS53L30_ADCx_CTL3_DEFAULT }, + { CS53L30_ADC1_NG_CTL, CS53L30_ADCx_NG_CTL_DEFAULT }, + { CS53L30_ADC1A_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT }, + { CS53L30_ADC1B_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT }, + { CS53L30_ADC1A_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT }, + { CS53L30_ADC1B_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT }, + { CS53L30_ADCDMIC2_CTL1, CS53L30_ADCDMICx_CTL1_DEFAULT }, + { CS53L30_ADCDMIC2_CTL2, CS53L30_ADCDMIC1_CTL2_DEFAULT }, + { CS53L30_ADC2_CTL3, CS53L30_ADCx_CTL3_DEFAULT }, + { CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_CTL_DEFAULT }, + { CS53L30_ADC2A_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT }, + { CS53L30_ADC2B_AFE_CTL, CS53L30_ADCxy_AFE_CTL_DEFAULT }, + { CS53L30_ADC2A_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT }, + { CS53L30_ADC2B_DIG_VOL, CS53L30_ADCxy_DIG_VOL_DEFAULT }, + { CS53L30_INT_MASK, CS53L30_DEVICE_INT_MASK }, +}; + +static bool cs53l30_volatile_register(struct device *dev, unsigned int reg) +{ + if (reg == CS53L30_IS) + return true; + else + return false; +} + +static bool cs53l30_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_IS: + return false; + default: + return true; + } +} + +static bool cs53l30_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS53L30_DEVID_AB: + case CS53L30_DEVID_CD: + case CS53L30_DEVID_E: + case CS53L30_REVID: + case CS53L30_PWRCTL: + case CS53L30_MCLKCTL: + case CS53L30_INT_SR_CTL: + case CS53L30_MICBIAS_CTL: + case CS53L30_ASPCFG_CTL: + case CS53L30_ASP_CTL1: + case CS53L30_ASP_TDMTX_CTL1: + case CS53L30_ASP_TDMTX_CTL2: + case CS53L30_ASP_TDMTX_CTL3: + case CS53L30_ASP_TDMTX_CTL4: + case CS53L30_ASP_TDMTX_EN1: + case CS53L30_ASP_TDMTX_EN2: + case CS53L30_ASP_TDMTX_EN3: + case CS53L30_ASP_TDMTX_EN4: + case CS53L30_ASP_TDMTX_EN5: + case CS53L30_ASP_TDMTX_EN6: + case CS53L30_ASP_CTL2: + case CS53L30_SFT_RAMP: + case CS53L30_LRCK_CTL1: + case CS53L30_LRCK_CTL2: + case CS53L30_MUTEP_CTL1: + case CS53L30_MUTEP_CTL2: + case CS53L30_INBIAS_CTL1: + case CS53L30_INBIAS_CTL2: + case CS53L30_DMIC1_STR_CTL: + case CS53L30_DMIC2_STR_CTL: + case CS53L30_ADCDMIC1_CTL1: + case CS53L30_ADCDMIC1_CTL2: + case CS53L30_ADC1_CTL3: + case CS53L30_ADC1_NG_CTL: + case CS53L30_ADC1A_AFE_CTL: + case CS53L30_ADC1B_AFE_CTL: + case CS53L30_ADC1A_DIG_VOL: + case CS53L30_ADC1B_DIG_VOL: + case CS53L30_ADCDMIC2_CTL1: + case CS53L30_ADCDMIC2_CTL2: + case CS53L30_ADC2_CTL3: + case CS53L30_ADC2_NG_CTL: + case CS53L30_ADC2A_AFE_CTL: + case CS53L30_ADC2B_AFE_CTL: + case CS53L30_ADC2A_DIG_VOL: + case CS53L30_ADC2B_DIG_VOL: + case CS53L30_INT_MASK: + return true; + default: + return false; + } +} + +static DECLARE_TLV_DB_SCALE(adc_boost_tlv, 0, 2000, 0); +static DECLARE_TLV_DB_SCALE(adc_ng_boost_tlv, 0, 3000, 0); +static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); +static DECLARE_TLV_DB_SCALE(dig_tlv, -9600, 100, 1); +static DECLARE_TLV_DB_SCALE(pga_preamp_tlv, 0, 10000, 0); + +static const char * const input1_sel_text[] = { + "DMIC1 On AB In", + "DMIC1 On A In", + "DMIC1 On B In", + "ADC1 On AB In", + "ADC1 On A In", + "ADC1 On B In", + "DMIC1 Off ADC1 Off", +}; + +unsigned int const input1_sel_values[] = { + CS53L30_CH_TYPE, + CS53L30_ADCxB_PDN | CS53L30_CH_TYPE, + CS53L30_ADCxA_PDN | CS53L30_CH_TYPE, + CS53L30_DMICx_PDN, + CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN, + CS53L30_ADCxA_PDN | CS53L30_DMICx_PDN, + CS53L30_ADCxA_PDN | CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN, +}; + +static const char * const input2_sel_text[] = { + "DMIC2 On AB In", + "DMIC2 On A In", + "DMIC2 On B In", + "ADC2 On AB In", + "ADC2 On A In", + "ADC2 On B In", + "DMIC2 Off ADC2 Off", +}; + +unsigned int const input2_sel_values[] = { + 0x0, + CS53L30_ADCxB_PDN, + CS53L30_ADCxA_PDN, + CS53L30_DMICx_PDN, + CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN, + CS53L30_ADCxA_PDN | CS53L30_DMICx_PDN, + CS53L30_ADCxA_PDN | CS53L30_ADCxB_PDN | CS53L30_DMICx_PDN, +}; + +static const char * const input1_route_sel_text[] = { + "ADC1_SEL", "DMIC1_SEL", +}; + +static const struct soc_enum input1_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, CS53L30_CH_TYPE_SHIFT, + ARRAY_SIZE(input1_route_sel_text), + input1_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input1_sel_enum, CS53L30_ADCDMIC1_CTL1, 0, + CS53L30_ADCDMICx_PDN_MASK, input1_sel_text, + input1_sel_values); + +static const struct snd_kcontrol_new input1_route_sel_mux = + SOC_DAPM_ENUM("Input 1 Route", input1_route_sel_enum); + +static const char * const input2_route_sel_text[] = { + "ADC2_SEL", "DMIC2_SEL", +}; + +/* Note: CS53L30_ADCDMIC1_CTL1 CH_TYPE controls inputs 1 and 2 */ +static const struct soc_enum input2_route_sel_enum = + SOC_ENUM_SINGLE(CS53L30_ADCDMIC1_CTL1, 0, + ARRAY_SIZE(input2_route_sel_text), + input2_route_sel_text); + +static SOC_VALUE_ENUM_SINGLE_DECL(input2_sel_enum, CS53L30_ADCDMIC2_CTL1, 0, + CS53L30_ADCDMICx_PDN_MASK, input2_sel_text, + input2_sel_values); + +static const struct snd_kcontrol_new input2_route_sel_mux = + SOC_DAPM_ENUM("Input 2 Route", input2_route_sel_enum); + +/* + * TB = 6144*(MCLK(int) scaling factor)/MCLK(internal) + * TB - Time base + * NOTE: If MCLK_INT_SCALE = 0, then TB=1 + */ +static const char * const cs53l30_ng_delay_text[] = { + "TB*50ms", "TB*100ms", "TB*150ms", "TB*200ms", +}; + +static const struct soc_enum adc1_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, CS53L30_ADCx_NG_DELAY_SHIFT, + ARRAY_SIZE(cs53l30_ng_delay_text), + cs53l30_ng_delay_text); + +static const struct soc_enum adc2_ng_delay_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_DELAY_SHIFT, + ARRAY_SIZE(cs53l30_ng_delay_text), + cs53l30_ng_delay_text); + +/* The noise gate threshold selected will depend on NG Boost */ +static const char * const cs53l30_ng_thres_text[] = { + "-64dB/-34dB", "-66dB/-36dB", "-70dB/-40dB", "-73dB/-43dB", + "-76dB/-46dB", "-82dB/-52dB", "-58dB", "-64dB", +}; + +static const struct soc_enum adc1_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_NG_CTL, CS53L30_ADCx_NG_THRESH_SHIFT, + ARRAY_SIZE(cs53l30_ng_thres_text), + cs53l30_ng_thres_text); + +static const struct soc_enum adc2_ng_thres_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_NG_CTL, CS53L30_ADCx_NG_THRESH_SHIFT, + ARRAY_SIZE(cs53l30_ng_thres_text), + cs53l30_ng_thres_text); + +/* Corner frequencies are with an Fs of 48kHz. */ +static const char * const hpf_corner_freq_text[] = { + "1.86Hz", "120Hz", "235Hz", "466Hz", +}; + +static const struct soc_enum adc1_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC1_CTL3, CS53L30_ADCx_HPF_CF_SHIFT, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct soc_enum adc2_hpf_enum = + SOC_ENUM_SINGLE(CS53L30_ADC2_CTL3, CS53L30_ADCx_HPF_CF_SHIFT, + ARRAY_SIZE(hpf_corner_freq_text), hpf_corner_freq_text); + +static const struct snd_kcontrol_new cs53l30_snd_controls[] = { + SOC_SINGLE("Digital Soft-Ramp Switch", CS53L30_SFT_RAMP, + CS53L30_DIGSFT_SHIFT, 1, 0), + SOC_SINGLE("ADC1 Noise Gate Ganging Switch", CS53L30_ADC1_CTL3, + CS53L30_ADCx_NG_ALL_SHIFT, 1, 0), + SOC_SINGLE("ADC2 Noise Gate Ganging Switch", CS53L30_ADC2_CTL3, + CS53L30_ADCx_NG_ALL_SHIFT, 1, 0), + SOC_SINGLE("ADC1A Noise Gate Enable Switch", CS53L30_ADC1_NG_CTL, + CS53L30_ADCxA_NG_SHIFT, 1, 0), + SOC_SINGLE("ADC1B Noise Gate Enable Switch", CS53L30_ADC1_NG_CTL, + CS53L30_ADCxB_NG_SHIFT, 1, 0), + SOC_SINGLE("ADC2A Noise Gate Enable Switch", CS53L30_ADC2_NG_CTL, + CS53L30_ADCxA_NG_SHIFT, 1, 0), + SOC_SINGLE("ADC2B Noise Gate Enable Switch", CS53L30_ADC2_NG_CTL, + CS53L30_ADCxB_NG_SHIFT, 1, 0), + SOC_SINGLE("ADC1 Notch Filter Switch", CS53L30_ADCDMIC1_CTL2, + CS53L30_ADCx_NOTCH_DIS_SHIFT, 1, 1), + SOC_SINGLE("ADC2 Notch Filter Switch", CS53L30_ADCDMIC2_CTL2, + CS53L30_ADCx_NOTCH_DIS_SHIFT, 1, 1), + SOC_SINGLE("ADC1A Invert Switch", CS53L30_ADCDMIC1_CTL2, + CS53L30_ADCxA_INV_SHIFT, 1, 0), + SOC_SINGLE("ADC1B Invert Switch", CS53L30_ADCDMIC1_CTL2, + CS53L30_ADCxB_INV_SHIFT, 1, 0), + SOC_SINGLE("ADC2A Invert Switch", CS53L30_ADCDMIC2_CTL2, + CS53L30_ADCxA_INV_SHIFT, 1, 0), + SOC_SINGLE("ADC2B Invert Switch", CS53L30_ADCDMIC2_CTL2, + CS53L30_ADCxB_INV_SHIFT, 1, 0), + + SOC_SINGLE_TLV("ADC1A Digital Boost Volume", CS53L30_ADCDMIC1_CTL2, + CS53L30_ADCxA_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1B Digital Boost Volume", CS53L30_ADCDMIC1_CTL2, + CS53L30_ADCxB_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2A Digital Boost Volume", CS53L30_ADCDMIC2_CTL2, + CS53L30_ADCxA_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC2B Digital Boost Volume", CS53L30_ADCDMIC2_CTL2, + CS53L30_ADCxB_DIG_BOOST_SHIFT, 1, 0, adc_boost_tlv), + SOC_SINGLE_TLV("ADC1 NG Boost Volume", CS53L30_ADC1_NG_CTL, + CS53L30_ADCx_NG_BOOST_SHIFT, 1, 0, adc_ng_boost_tlv), + SOC_SINGLE_TLV("ADC2 NG Boost Volume", CS53L30_ADC2_NG_CTL, + CS53L30_ADCx_NG_BOOST_SHIFT, 1, 0, adc_ng_boost_tlv), + + SOC_DOUBLE_R_TLV("ADC1 Pre Amp Gain", CS53L30_ADC1A_AFE_CTL, + CS53L30_ADC1B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT, + 2, 0, pga_preamp_tlv), + SOC_DOUBLE_R_TLV("ADC2 Pre Amp Gain", CS53L30_ADC2A_AFE_CTL, + CS53L30_ADC2B_AFE_CTL, CS53L30_ADCxy_PREAMP_SHIFT, + 2, 0, pga_preamp_tlv), + + SOC_ENUM("Input 1 Channel Select", input1_sel_enum), + SOC_ENUM("Input 2 Channel Select", input2_sel_enum), + + SOC_ENUM("ADC1 HPF Select", adc1_hpf_enum), + SOC_ENUM("ADC2 HPF Select", adc2_hpf_enum), + SOC_ENUM("ADC1 NG Threshold", adc1_ng_thres_enum), + SOC_ENUM("ADC2 NG Threshold", adc2_ng_thres_enum), + SOC_ENUM("ADC1 NG Delay", adc1_ng_delay_enum), + SOC_ENUM("ADC2 NG Delay", adc2_ng_delay_enum), + + SOC_SINGLE_SX_TLV("ADC1A PGA Volume", + CS53L30_ADC1A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC1B PGA Volume", + CS53L30_ADC1B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2A PGA Volume", + CS53L30_ADC2A_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + SOC_SINGLE_SX_TLV("ADC2B PGA Volume", + CS53L30_ADC2B_AFE_CTL, 0, 0x34, 0x18, pga_tlv), + + SOC_SINGLE_SX_TLV("ADC1A Digital Volume", + CS53L30_ADC1A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC1B Digital Volume", + CS53L30_ADC1B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2A Digital Volume", + CS53L30_ADC2A_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), + SOC_SINGLE_SX_TLV("ADC2B Digital Volume", + CS53L30_ADC2B_DIG_VOL, 0, 0xA0, 0x0C, dig_tlv), +}; + +static const struct snd_soc_dapm_widget cs53l30_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("IN1_DMIC1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3_DMIC2"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_SUPPLY("MIC1 Bias", CS53L30_MICBIAS_CTL, + CS53L30_MIC1_BIAS_PDN_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC2 Bias", CS53L30_MICBIAS_CTL, + CS53L30_MIC2_BIAS_PDN_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC3 Bias", CS53L30_MICBIAS_CTL, + CS53L30_MIC3_BIAS_PDN_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("MIC4 Bias", CS53L30_MICBIAS_CTL, + CS53L30_MIC4_BIAS_PDN_SHIFT, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("ASP_SDOUT1", NULL, 0, CS53L30_ASP_CTL1, + CS53L30_ASP_SDOUTx_PDN_SHIFT, 1), + SND_SOC_DAPM_AIF_OUT("ASP_SDOUT2", NULL, 0, CS53L30_ASP_CTL2, + CS53L30_ASP_SDOUTx_PDN_SHIFT, 1), + + SND_SOC_DAPM_MUX("Input Mux 1", SND_SOC_NOPM, 0, 0, + &input1_route_sel_mux), + SND_SOC_DAPM_MUX("Input Mux 2", SND_SOC_NOPM, 0, 0, + &input2_route_sel_mux), + + SND_SOC_DAPM_ADC("ADC1A", NULL, CS53L30_ADCDMIC1_CTL1, + CS53L30_ADCxA_PDN_SHIFT, 1), + SND_SOC_DAPM_ADC("ADC1B", NULL, CS53L30_ADCDMIC1_CTL1, + CS53L30_ADCxB_PDN_SHIFT, 1), + SND_SOC_DAPM_ADC("ADC2A", NULL, CS53L30_ADCDMIC2_CTL1, + CS53L30_ADCxA_PDN_SHIFT, 1), + SND_SOC_DAPM_ADC("ADC2B", NULL, CS53L30_ADCDMIC2_CTL1, + CS53L30_ADCxB_PDN_SHIFT, 1), + SND_SOC_DAPM_ADC("DMIC1", NULL, CS53L30_ADCDMIC1_CTL1, + CS53L30_DMICx_PDN_SHIFT, 1), + SND_SOC_DAPM_ADC("DMIC2", NULL, CS53L30_ADCDMIC2_CTL1, + CS53L30_DMICx_PDN_SHIFT, 1), +}; + +static const struct snd_soc_dapm_route cs53l30_dapm_routes[] = { + /* ADC Input Paths */ + {"ADC1A", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "ADC1_SEL", "ADC1A"}, + {"ADC1B", NULL, "IN2"}, + + {"ADC2A", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "ADC2_SEL", "ADC2A"}, + {"ADC2B", NULL, "IN4"}, + + /* MIC Bias Paths */ + {"ADC1A", NULL, "MIC1 Bias"}, + {"ADC1B", NULL, "MIC2 Bias"}, + {"ADC2A", NULL, "MIC3 Bias"}, + {"ADC2B", NULL, "MIC4 Bias"}, + + /* DMIC Paths */ + {"DMIC1", NULL, "IN1_DMIC1"}, + {"Input Mux 1", "DMIC1_SEL", "DMIC1"}, + + {"DMIC2", NULL, "IN3_DMIC2"}, + {"Input Mux 2", "DMIC2_SEL", "DMIC2"}, +}; + +static const struct snd_soc_dapm_route cs53l30_dapm_routes_sdout1[] = { + /* Output Paths when using SDOUT1 only */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT1", NULL, "ADC2A"}, + {"ASP_SDOUT1", NULL, "Input Mux 2"}, + {"ASP_SDOUT1", NULL, "ADC2B"}, + + {"Capture", NULL, "ASP_SDOUT1"}, +}; + +static const struct snd_soc_dapm_route cs53l30_dapm_routes_sdout2[] = { + /* Output Paths when using both SDOUT1 and SDOUT2 */ + {"ASP_SDOUT1", NULL, "ADC1A" }, + {"ASP_SDOUT1", NULL, "Input Mux 1"}, + {"ASP_SDOUT1", NULL, "ADC1B"}, + + {"ASP_SDOUT2", NULL, "ADC2A"}, + {"ASP_SDOUT2", NULL, "Input Mux 2"}, + {"ASP_SDOUT2", NULL, "ADC2B"}, + + {"Capture", NULL, "ASP_SDOUT1"}, + {"Capture", NULL, "ASP_SDOUT2"}, +}; + +struct cs53l30_mclk_div { + u32 mclk_rate; + u32 srate; + u8 asp_rate; + u8 internal_fs_ratio; + u8 mclk_int_scale; +}; + +static struct cs53l30_mclk_div cs53l30_mclk_coeffs[] = { + /* NOTE: Enable MCLK_INT_SCALE to save power. */ + + /* MCLK, Sample Rate, asp_rate, internal_fs_ratio, mclk_int_scale */ + {5644800, 11025, 0x4, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {5644800, 22050, 0x8, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {5644800, 44100, 0xC, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + + {6000000, 8000, 0x1, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 11025, 0x2, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 12000, 0x4, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 16000, 0x5, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 22050, 0x6, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 24000, 0x8, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 32000, 0x9, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 44100, 0xA, 0, CS53L30_MCLK_INT_SCALE}, + {6000000, 48000, 0xC, 0, CS53L30_MCLK_INT_SCALE}, + + {6144000, 8000, 0x1, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 11025, 0x2, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 12000, 0x4, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 16000, 0x5, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 22050, 0x6, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 24000, 0x8, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 32000, 0x9, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 44100, 0xA, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6144000, 48000, 0xC, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + + {6400000, 8000, 0x1, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 11025, 0x2, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 12000, 0x4, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 16000, 0x5, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 22050, 0x6, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 24000, 0x8, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 32000, 0x9, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 44100, 0xA, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, + {6400000, 48000, 0xC, CS53L30_INTRNL_FS_RATIO, CS53L30_MCLK_INT_SCALE}, +}; + +struct cs53l30_mclkx_div { + u32 mclkx; + u8 ratio; + u8 mclkdiv; +}; + +static struct cs53l30_mclkx_div cs53l30_mclkx_coeffs[] = { + {5644800, 1, CS53L30_MCLK_DIV_BY_1}, + {6000000, 1, CS53L30_MCLK_DIV_BY_1}, + {6144000, 1, CS53L30_MCLK_DIV_BY_1}, + {11289600, 2, CS53L30_MCLK_DIV_BY_2}, + {12288000, 2, CS53L30_MCLK_DIV_BY_2}, + {12000000, 2, CS53L30_MCLK_DIV_BY_2}, + {19200000, 3, CS53L30_MCLK_DIV_BY_3}, +}; + +static int cs53l30_get_mclkx_coeff(int mclkx) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclkx_coeffs); i++) { + if (cs53l30_mclkx_coeffs[i].mclkx == mclkx) + return i; + } + + return -EINVAL; +} + +static int cs53l30_get_mclk_coeff(int mclk_rate, int srate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(cs53l30_mclk_coeffs); i++) { + if (cs53l30_mclk_coeffs[i].mclk_rate == mclk_rate && + cs53l30_mclk_coeffs[i].srate == srate) + return i; + } + + return -EINVAL; +} + +static int cs53l30_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + int mclkx_coeff; + u32 mclk_rate; + + /* MCLKX -> MCLK */ + mclkx_coeff = cs53l30_get_mclkx_coeff(freq); + if (mclkx_coeff < 0) + return mclkx_coeff; + + mclk_rate = cs53l30_mclkx_coeffs[mclkx_coeff].mclkx / + cs53l30_mclkx_coeffs[mclkx_coeff].ratio; + + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIV_MASK, + cs53l30_mclkx_coeffs[mclkx_coeff].mclkdiv); + + priv->mclk_rate = mclk_rate; + + return 0; +} + +static int cs53l30_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + u8 aspcfg = 0, aspctl1 = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + aspcfg |= CS53L30_ASP_MS; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* Set TDM_PDN to turn off TDM mode -- Reset default */ + aspctl1 |= CS53L30_ASP_TDM_PDN; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Clear TDM_PDN and SHIFT_LEFT, invert SCLK */ + aspcfg |= CS53L30_ASP_SCLK_INV; + break; + default: + return -EINVAL; + } + + /* Check to see if the SCLK is inverted */ + if (fmt & (SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_IB_IF)) + aspcfg ^= CS53L30_ASP_SCLK_INV; + + regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL, + CS53L30_ASP_MS | CS53L30_ASP_SCLK_INV, aspcfg); + + regmap_update_bits(priv->regmap, CS53L30_ASP_CTL1, + CS53L30_ASP_TDM_PDN | CS53L30_SHIFT_LEFT, aspctl1); + + return 0; +} + +static int cs53l30_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + int srate = params_rate(params); + int mclk_coeff; + + /* MCLK -> srate */ + mclk_coeff = cs53l30_get_mclk_coeff(priv->mclk_rate, srate); + if (mclk_coeff < 0) + return -EINVAL; + + regmap_update_bits(priv->regmap, CS53L30_INT_SR_CTL, + CS53L30_INTRNL_FS_RATIO_MASK, + cs53l30_mclk_coeffs[mclk_coeff].internal_fs_ratio); + + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_INT_SCALE_MASK, + cs53l30_mclk_coeffs[mclk_coeff].mclk_int_scale); + + regmap_update_bits(priv->regmap, CS53L30_ASPCFG_CTL, + CS53L30_ASP_RATE_MASK, + cs53l30_mclk_coeffs[mclk_coeff].asp_rate); + + return 0; +} + +static int cs53l30_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + int i, inter_max_check, ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_LP_MASK, 0); + break; + case SND_SOC_BIAS_STANDBY: + if (dapm->bias_level == SND_SOC_BIAS_OFF) { + ret = clk_prepare_enable(priv->mclk); + if (ret) { + dev_err(codec->dev, + "failed to enable MCLK: %d\n", ret); + return ret; + } + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIS_MASK, 0); + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP_MASK, 0); + msleep(50); + } else { + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP_MASK, + CS53L30_PDN_ULP); + } + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + CS53L30_PDN_DONE, 0); + /* + * If digital softramp is set, the amount of time required + * for power down increases and depends on the digital + * volume setting. + */ + + /* Set the max possible time if digsft is set */ + regmap_read(priv->regmap, CS53L30_SFT_RAMP, &reg); + if (reg & CS53L30_DIGSFT_MASK) + inter_max_check = CS53L30_PDN_POLL_MAX; + else + inter_max_check = 10; + + regmap_update_bits(priv->regmap, CS53L30_PWRCTL, + CS53L30_PDN_ULP_MASK, + CS53L30_PDN_ULP); + /* PDN_DONE will take a min of 20ms to be set.*/ + msleep(20); + /* Clr status */ + regmap_read(priv->regmap, CS53L30_IS, &reg); + for (i = 0; i < inter_max_check; i++) { + if (inter_max_check < 10) { + usleep_range(1000, 1100); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & CS53L30_PDN_DONE) + break; + } else { + usleep_range(10000, 10100); + regmap_read(priv->regmap, CS53L30_IS, &reg); + if (reg & CS53L30_PDN_DONE) + break; + } + } + /* PDN_DONE is set. We now can disable the MCLK */ + regmap_update_bits(priv->regmap, CS53L30_INT_MASK, + CS53L30_PDN_DONE, CS53L30_PDN_DONE); + regmap_update_bits(priv->regmap, CS53L30_MCLKCTL, + CS53L30_MCLK_DIS_MASK, + CS53L30_MCLK_DIS); + clk_disable_unprepare(priv->mclk); + break; + } + + return 0; +} + +static int cs53l30_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + u8 val = tristate ? CS53L30_ASP_3ST : 0; + + return regmap_update_bits(priv->regmap, CS53L30_ASP_CTL1, + CS53L30_ASP_3ST_MASK, val); +} + +unsigned int const cs53l30_src_rates[] = { + 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 +}; + +static struct snd_pcm_hw_constraint_list src_constraints = { + .count = ARRAY_SIZE(cs53l30_src_rates), + .list = cs53l30_src_rates, +}; + +static int cs53l30_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &src_constraints); + + return 0; +} + +/* + * Note: CS53L30 counts the slot number per byte while ASoC counts the slot + * number per slot_width. So there is a difference between the slots of ASoC + * and the slots of CS53L30. + */ +static int cs53l30_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(dai->codec); + unsigned int loc[CS53L30_TDM_SLOT_MAX] = {48, 48, 48, 48}; + unsigned int slot_next, slot_step; + u64 tx_enable = 0; + int i; + + if (!rx_mask) { + dev_err(dai->dev, "rx masks must not be 0\n"); + return -EINVAL; + } + + /* Assuming slot_width is not supposed to be greater than 64 */ + if (slots <= 0 || slot_width <= 0 || slot_width > 64) { + dev_err(dai->dev, "invalid slot number or slot width\n"); + return -EINVAL; + } + + if (slot_width & 0x7) { + dev_err(dai->dev, "slot width must count in byte\n"); + return -EINVAL; + } + + /* How many bytes in each ASoC slot */ + slot_step = slot_width >> 3; + + for (i = 0; rx_mask && i < CS53L30_TDM_SLOT_MAX; i++) { + /* Find the first slot from LSB */ + slot_next = __ffs(rx_mask); + /* Save the slot location by converting to CS53L30 slot */ + loc[i] = slot_next * slot_step; + /* Create the mask of CS53L30 slot */ + tx_enable |= (u64)((u64)(1 << slot_step) - 1) << (u64)loc[i]; + /* Clear this slot from rx_mask */ + rx_mask &= ~(1 << slot_next); + } + + /* Error out to avoid slot shift */ + if (rx_mask && i == CS53L30_TDM_SLOT_MAX) { + dev_err(dai->dev, "rx_mask exceeds max slot number: %d\n", + CS53L30_TDM_SLOT_MAX); + return -EINVAL; + } + + /* Validate the last CS53L30 slot */ + slot_next = loc[CS53L30_TDM_SLOT_MAX - 1] + slot_step - 1; + if (slot_next > 47) { + dev_err(dai->dev, "slot selection out of bounds: %u\n", + slot_next); + return -EINVAL; + } + + for (i = 0; i < CS53L30_TDM_SLOT_MAX && loc[i] != 48; i++) { + regmap_update_bits(priv->regmap, CS53L30_ASP_TDMTX_CTL(i), + CS53L30_ASP_CHx_TX_LOC_MASK, loc[i]); + dev_dbg(dai->dev, "loc[%d]=%x\n", i, loc[i]); + } + + for (i = 0; i < CS53L30_ASP_TDMTX_ENx_MAX && tx_enable; i++) { + regmap_write(priv->regmap, CS53L30_ASP_TDMTX_ENx(i), + tx_enable & 0xff); + tx_enable >>= 8; + dev_dbg(dai->dev, "en_reg=%x, tx_enable=%llx\n", + CS53L30_ASP_TDMTX_ENx(i), tx_enable & 0xff); + } + + return 0; +} + +/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ +#define CS53L30_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) + +#define CS53L30_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops cs53l30_ops = { + .startup = cs53l30_pcm_startup, + .hw_params = cs53l30_pcm_hw_params, + .set_fmt = cs53l30_set_dai_fmt, + .set_sysclk = cs53l30_set_sysclk, + .set_tristate = cs53l30_set_tristate, + .set_tdm_slot = cs53l30_set_dai_tdm_slot, +}; + +static struct snd_soc_dai_driver cs53l30_dai = { + .name = "cs53l30", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = CS53L30_RATES, + .formats = CS53L30_FORMATS, + }, + .ops = &cs53l30_ops, + .symmetric_rates = 1, +}; + +static int cs53l30_codec_probe(struct snd_soc_codec *codec) +{ + struct cs53l30_private *priv = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + if (priv->use_sdout2) + snd_soc_dapm_add_routes(dapm, cs53l30_dapm_routes_sdout2, + ARRAY_SIZE(cs53l30_dapm_routes_sdout2)); + else + snd_soc_dapm_add_routes(dapm, cs53l30_dapm_routes_sdout1, + ARRAY_SIZE(cs53l30_dapm_routes_sdout1)); + + return 0; +} + +static struct snd_soc_codec_driver cs53l30_driver = { + .probe = cs53l30_codec_probe, + .set_bias_level = cs53l30_set_bias_level, + + .dapm_widgets = cs53l30_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs53l30_dapm_widgets), + .dapm_routes = cs53l30_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs53l30_dapm_routes), + + .controls = cs53l30_snd_controls, + .num_controls = ARRAY_SIZE(cs53l30_snd_controls), +}; + +static struct regmap_config cs53l30_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS53L30_MAX_REGISTER, + .reg_defaults = cs53l30_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs53l30_reg_defaults), + .volatile_reg = cs53l30_volatile_register, + .writeable_reg = cs53l30_writeable_register, + .readable_reg = cs53l30_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int cs53l30_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + const struct device_node *np = client->dev.of_node; + struct device *dev = &client->dev; + struct cs53l30_private *cs53l30; + unsigned int devid = 0; + unsigned int reg; + int ret = 0, i; + u8 val; + + cs53l30 = devm_kzalloc(dev, sizeof(*cs53l30), GFP_KERNEL); + if (!cs53l30) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(cs53l30->supplies); i++) + cs53l30->supplies[i].supply = cs53l30_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(cs53l30->supplies), + cs53l30->supplies); + if (ret) { + dev_err(dev, "failed to get supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs53l30->supplies), + cs53l30->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the Device */ + cs53l30->reset_gpio = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_LOW); + if (IS_ERR(cs53l30->reset_gpio)) { + ret = PTR_ERR(cs53l30->reset_gpio); + goto error; + } + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + i2c_set_clientdata(client, cs53l30); + + cs53l30->mclk_rate = 0; + + cs53l30->regmap = devm_regmap_init_i2c(client, &cs53l30_regmap); + if (IS_ERR(cs53l30->regmap)) { + ret = PTR_ERR(cs53l30->regmap); + dev_err(dev, "regmap_init() failed: %d\n", ret); + goto error; + } + + /* Initialize codec */ + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_AB, &reg); + devid = reg << 12; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_CD, &reg); + devid |= reg << 4; + + ret = regmap_read(cs53l30->regmap, CS53L30_DEVID_E, &reg); + devid |= (reg & 0xF0) >> 4; + + if (devid != CS53L30_DEVID) { + ret = -ENODEV; + dev_err(dev, "Device ID (%X). Expected %X\n", + devid, CS53L30_DEVID); + goto error; + } + + ret = regmap_read(cs53l30->regmap, CS53L30_REVID, &reg); + if (ret < 0) { + dev_err(dev, "failed to get Revision ID: %d\n", ret); + goto error; + } + + /* Check if MCLK provided */ + cs53l30->mclk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs53l30->mclk)) { + if (PTR_ERR(cs53l30->mclk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto error; + } + /* Otherwise mark the mclk pointer to NULL */ + cs53l30->mclk = NULL; + } + + if (!of_property_read_u8(np, "cirrus,micbias-lvl", &val)) + regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL, + CS53L30_MIC_BIAS_CTRL_MASK, val); + + if (of_property_read_bool(np, "cirrus,use-sdout2")) + cs53l30->use_sdout2 = true; + + dev_info(dev, "Cirrus Logic CS53L30, Revision: %02X\n", reg & 0xFF); + + ret = snd_soc_register_codec(dev, &cs53l30_driver, &cs53l30_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec: %d\n", ret); + goto error; + } + + return 0; + +error: + regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies), + cs53l30->supplies); + return ret; +} + +static int cs53l30_i2c_remove(struct i2c_client *client) +{ + struct cs53l30_private *cs53l30 = i2c_get_clientdata(client); + + snd_soc_unregister_codec(&client->dev); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies), + cs53l30->supplies); + + return 0; +} + +#ifdef CONFIG_PM +static int cs53l30_runtime_suspend(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + + regcache_cache_only(cs53l30->regmap, true); + + /* Hold down reset */ + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 0); + + regulator_bulk_disable(ARRAY_SIZE(cs53l30->supplies), + cs53l30->supplies); + + return 0; +} + +static int cs53l30_runtime_resume(struct device *dev) +{ + struct cs53l30_private *cs53l30 = dev_get_drvdata(dev); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(cs53l30->supplies), + cs53l30->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + if (cs53l30->reset_gpio) + gpiod_set_value_cansleep(cs53l30->reset_gpio, 1); + + regcache_cache_only(cs53l30->regmap, false); + regcache_sync(cs53l30->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops cs53l30_runtime_pm = { + SET_RUNTIME_PM_OPS(cs53l30_runtime_suspend, cs53l30_runtime_resume, + NULL) +}; + +static const struct of_device_id cs53l30_of_match[] = { + { .compatible = "cirrus,cs53l30", }, + {}, +}; + +MODULE_DEVICE_TABLE(of, cs53l30_of_match); + +static const struct i2c_device_id cs53l30_id[] = { + { "cs53l30", 0 }, + {} +}; + +MODULE_DEVICE_TABLE(i2c, cs53l30_id); + +static struct i2c_driver cs53l30_i2c_driver = { + .driver = { + .name = "cs53l30", + .pm = &cs53l30_runtime_pm, + }, + .id_table = cs53l30_id, + .probe = cs53l30_i2c_probe, + .remove = cs53l30_i2c_remove, +}; + +module_i2c_driver(cs53l30_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS53L30 driver"); +MODULE_AUTHOR("Paul Handrigan, Cirrus Logic Inc, <Paul.Handrigan(a)cirrus.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs53l30.h b/sound/soc/codecs/cs53l30.h new file mode 100644 index 0000000..0dd4afb --- /dev/null +++ b/sound/soc/codecs/cs53l30.h @@ -0,0 +1,458 @@ +/* + * ALSA SoC CS53L30 codec driver + * + * Copyright 2015 Cirrus Logic, Inc. + * + * Author: Paul Handrigan <Paul.Handrigan(a)cirrus.com>, + * Tim Howe <Tim.Howe(a)cirrus.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef __CS53L30_H__ +#define __CS53L30_H__ + +/* I2C Registers */ +#define CS53L30_DEVID_AB 0x01 /* Device ID A & B [RO]. */ +#define CS53L30_DEVID_CD 0x02 /* Device ID C & D [RO]. */ +#define CS53L30_DEVID_E 0x03 /* Device ID E [RO]. */ +#define CS53L30_REVID 0x05 /* Revision ID [RO]. */ +#define CS53L30_PWRCTL 0x06 /* Power Control. */ +#define CS53L30_MCLKCTL 0x07 /* MCLK Control. */ +#define CS53L30_INT_SR_CTL 0x08 /* Internal Sample Rate Control. */ +#define CS53L30_MICBIAS_CTL 0x0A /* Mic Bias Control. */ +#define CS53L30_ASPCFG_CTL 0x0C /* ASP Config Control. */ +#define CS53L30_ASP_CTL1 0x0D /* ASP1 Control. */ +#define CS53L30_ASP_TDMTX_CTL1 0x0E /* ASP1 TDM TX Control 1 */ +#define CS53L30_ASP_TDMTX_CTL2 0x0F /* ASP1 TDM TX Control 2 */ +#define CS53L30_ASP_TDMTX_CTL3 0x10 /* ASP1 TDM TX Control 3 */ +#define CS53L30_ASP_TDMTX_CTL4 0x11 /* ASP1 TDM TX Control 4 */ +#define CS53L30_ASP_TDMTX_EN1 0x12 /* ASP1 TDM TX Enable 1 */ +#define CS53L30_ASP_TDMTX_EN2 0x13 /* ASP1 TDM TX Enable 2 */ +#define CS53L30_ASP_TDMTX_EN3 0x14 /* ASP1 TDM TX Enable 3 */ +#define CS53L30_ASP_TDMTX_EN4 0x15 /* ASP1 TDM TX Enable 4 */ +#define CS53L30_ASP_TDMTX_EN5 0x16 /* ASP1 TDM TX Enable 5 */ +#define CS53L30_ASP_TDMTX_EN6 0x17 /* ASP1 TDM TX Enable 6 */ +#define CS53L30_ASP_CTL2 0x18 /* ASP2 Control. */ +#define CS53L30_SFT_RAMP 0x1A /* Soft Ramp Control. */ +#define CS53L30_LRCK_CTL1 0x1B /* LRCK Control 1. */ +#define CS53L30_LRCK_CTL2 0x1C /* LRCK Control 2. */ +#define CS53L30_MUTEP_CTL1 0x1F /* Mute Pin Control 1. */ +#define CS53L30_MUTEP_CTL2 0x20 /* Mute Pin Control 2. */ +#define CS53L30_INBIAS_CTL1 0x21 /* Input Bias Control 1. */ +#define CS53L30_INBIAS_CTL2 0x22 /* Input Bias Control 2. */ +#define CS53L30_DMIC1_STR_CTL 0x23 /* DMIC1 Stereo Control. */ +#define CS53L30_DMIC2_STR_CTL 0x24 /* DMIC2 Stereo Control. */ +#define CS53L30_ADCDMIC1_CTL1 0x25 /* ADC1/DMIC1 Control 1. */ +#define CS53L30_ADCDMIC1_CTL2 0x26 /* ADC1/DMIC1 Control 2. */ +#define CS53L30_ADC1_CTL3 0x27 /* ADC1 Control 3. */ +#define CS53L30_ADC1_NG_CTL 0x28 /* ADC1 Noise Gate Control. */ +#define CS53L30_ADC1A_AFE_CTL 0x29 /* ADC1A AFE Control. */ +#define CS53L30_ADC1B_AFE_CTL 0x2A /* ADC1B AFE Control. */ +#define CS53L30_ADC1A_DIG_VOL 0x2B /* ADC1A Digital Volume. */ +#define CS53L30_ADC1B_DIG_VOL 0x2C /* ADC1B Digital Volume. */ +#define CS53L30_ADCDMIC2_CTL1 0x2D /* ADC2/DMIC2 Control 1. */ +#define CS53L30_ADCDMIC2_CTL2 0x2E /* ADC2/DMIC2 Control 2. */ +#define CS53L30_ADC2_CTL3 0x2F /* ADC2 Control 3. */ +#define CS53L30_ADC2_NG_CTL 0x30 /* ADC2 Noise Gate Control. */ +#define CS53L30_ADC2A_AFE_CTL 0x31 /* ADC2A AFE Control. */ +#define CS53L30_ADC2B_AFE_CTL 0x32 /* ADC2B AFE Control. */ +#define CS53L30_ADC2A_DIG_VOL 0x33 /* ADC2A Digital Volume. */ +#define CS53L30_ADC2B_DIG_VOL 0x34 /* ADC2B Digital Volume. */ +#define CS53L30_INT_MASK 0x35 /* Interrupt Mask. */ +#define CS53L30_IS 0x36 /* Interrupt Status. */ +#define CS53L30_MAX_REGISTER 0x36 + +#define CS53L30_TDM_SLOT_MAX 4 +#define CS53L30_ASP_TDMTX_CTL(x) (CS53L30_ASP_TDMTX_CTL1 + (x)) +/* x : index for registers; n : index for slot; 8 slots per register */ +#define CS53L30_ASP_TDMTX_ENx(x) (CS53L30_ASP_TDMTX_EN6 - (x)) +#define CS53L30_ASP_TDMTX_ENn(n) CS53L30_ASP_TDMTX_ENx((n) >> 3) +#define CS53L30_ASP_TDMTX_ENx_MAX 6 + +/* Device ID */ +#define CS53L30_DEVID 0x53A30 + +/* PDN_DONE Poll Maximum + * If soft ramp is set it will take much longer to power down + * the system. + */ +#define CS53L30_PDN_POLL_MAX 90 + +/* Bitfield Definitions */ + +/* R6 (0x06) CS53L30_PWRCTL - Power Control */ +#define CS53L30_PDN_ULP_SHIFT 7 +#define CS53L30_PDN_ULP_MASK (1 << CS53L30_PDN_ULP_SHIFT) +#define CS53L30_PDN_ULP (1 << CS53L30_PDN_ULP_SHIFT) +#define CS53L30_PDN_LP_SHIFT 6 +#define CS53L30_PDN_LP_MASK (1 << CS53L30_PDN_LP_SHIFT) +#define CS53L30_PDN_LP (1 << CS53L30_PDN_LP_SHIFT) +#define CS53L30_DISCHARGE_FILT_SHIFT 5 +#define CS53L30_DISCHARGE_FILT_MASK (1 << CS53L30_DISCHARGE_FILT_SHIFT) +#define CS53L30_DISCHARGE_FILT (1 << CS53L30_DISCHARGE_FILT_SHIFT) +#define CS53L30_THMS_PDN_SHIFT 4 +#define CS53L30_THMS_PDN_MASK (1 << CS53L30_THMS_PDN_SHIFT) +#define CS53L30_THMS_PDN (1 << CS53L30_THMS_PDN_SHIFT) + +#define CS53L30_PWRCTL_DEFAULT (CS53L30_THMS_PDN) + +/* R7 (0x07) CS53L30_MCLKCTL - MCLK Control */ +#define CS53L30_MCLK_DIS_SHIFT 7 +#define CS53L30_MCLK_DIS_MASK (1 << CS53L30_MCLK_DIS_SHIFT) +#define CS53L30_MCLK_DIS (1 << CS53L30_MCLK_DIS_SHIFT) +#define CS53L30_MCLK_INT_SCALE_SHIFT 6 +#define CS53L30_MCLK_INT_SCALE_MASK (1 << CS53L30_MCLK_INT_SCALE_SHIFT) +#define CS53L30_MCLK_INT_SCALE (1 << CS53L30_MCLK_INT_SCALE_SHIFT) +#define CS53L30_DMIC_DRIVE_SHIFT 5 +#define CS53L30_DMIC_DRIVE_MASK (1 << CS53L30_DMIC_DRIVE_SHIFT) +#define CS53L30_DMIC_DRIVE (1 << CS53L30_DMIC_DRIVE_SHIFT) +#define CS53L30_MCLK_DIV_SHIFT 2 +#define CS53L30_MCLK_DIV_WIDTH 2 +#define CS53L30_MCLK_DIV_MASK (((1 << CS53L30_MCLK_DIV_WIDTH) - 1) << CS53L30_MCLK_DIV_SHIFT) +#define CS53L30_MCLK_DIV_BY_1 (0x0 << CS53L30_MCLK_DIV_SHIFT) +#define CS53L30_MCLK_DIV_BY_2 (0x1 << CS53L30_MCLK_DIV_SHIFT) +#define CS53L30_MCLK_DIV_BY_3 (0x2 << CS53L30_MCLK_DIV_SHIFT) +#define CS53L30_SYNC_EN_SHIFT 1 +#define CS53L30_SYNC_EN_MASK (1 << CS53L30_SYNC_EN_SHIFT) +#define CS53L30_SYNC_EN (1 << CS53L30_SYNC_EN_SHIFT) + +#define CS53L30_MCLKCTL_DEFAULT (CS53L30_MCLK_DIV_BY_2) + +/* R8 (0x08) CS53L30_INT_SR_CTL - Internal Sample Rate Control */ +#define CS53L30_INTRNL_FS_RATIO_SHIFT 4 +#define CS53L30_INTRNL_FS_RATIO_MASK (1 << CS53L30_INTRNL_FS_RATIO_SHIFT) +#define CS53L30_INTRNL_FS_RATIO (1 << CS53L30_INTRNL_FS_RATIO_SHIFT) +#define CS53L30_MCLK_19MHZ_EN_SHIFT 0 +#define CS53L30_MCLK_19MHZ_EN_MASK (1 << CS53L30_MCLK_19MHZ_EN_SHIFT) +#define CS53L30_MCLK_19MHZ_EN (1 << CS53L30_MCLK_19MHZ_EN_SHIFT) + +/* 0x6 << 1 is reserved bits */ +#define CS53L30_INT_SR_CTL_DEFAULT (CS53L30_INTRNL_FS_RATIO | 0x6 << 1) + +/* R10 (0x0A) CS53L30_MICBIAS_CTL - Mic Bias Control */ +#define CS53L30_MIC4_BIAS_PDN_SHIFT 7 +#define CS53L30_MIC4_BIAS_PDN_MASK (1 << CS53L30_MIC4_BIAS_PDN_SHIFT) +#define CS53L30_MIC4_BIAS_PDN (1 << CS53L30_MIC4_BIAS_PDN_SHIFT) +#define CS53L30_MIC3_BIAS_PDN_SHIFT 6 +#define CS53L30_MIC3_BIAS_PDN_MASK (1 << CS53L30_MIC3_BIAS_PDN_SHIFT) +#define CS53L30_MIC3_BIAS_PDN (1 << CS53L30_MIC3_BIAS_PDN_SHIFT) +#define CS53L30_MIC2_BIAS_PDN_SHIFT 5 +#define CS53L30_MIC2_BIAS_PDN_MASK (1 << CS53L30_MIC2_BIAS_PDN_SHIFT) +#define CS53L30_MIC2_BIAS_PDN (1 << CS53L30_MIC2_BIAS_PDN_SHIFT) +#define CS53L30_MIC1_BIAS_PDN_SHIFT 4 +#define CS53L30_MIC1_BIAS_PDN_MASK (1 << CS53L30_MIC1_BIAS_PDN_SHIFT) +#define CS53L30_MIC1_BIAS_PDN (1 << CS53L30_MIC1_BIAS_PDN_SHIFT) +#define CS53L30_MICx_BIAS_PDN (0xf << CS53L30_MIC1_BIAS_PDN_SHIFT) +#define CS53L30_VP_MIN_SHIFT 2 +#define CS53L30_VP_MIN_MASK (1 << CS53L30_VP_MIN_SHIFT) +#define CS53L30_VP_MIN (1 << CS53L30_VP_MIN_SHIFT) +#define CS53L30_MIC_BIAS_CTRL_SHIFT 0 +#define CS53L30_MIC_BIAS_CTRL_WIDTH 2 +#define CS53L30_MIC_BIAS_CTRL_MASK (((1 << CS53L30_MIC_BIAS_CTRL_WIDTH) - 1) << CS53L30_MIC_BIAS_CTRL_SHIFT) +#define CS53L30_MIC_BIAS_CTRL_HIZ (0 << CS53L30_MIC_BIAS_CTRL_SHIFT) +#define CS53L30_MIC_BIAS_CTRL_1V8 (1 << CS53L30_MIC_BIAS_CTRL_SHIFT) +#define CS53L30_MIC_BIAS_CTRL_2V75 (2 << CS53L30_MIC_BIAS_CTRL_SHIFT) + +#define CS53L30_MICBIAS_CTL_DEFAULT (CS53L30_MICx_BIAS_PDN | CS53L30_VP_MIN) + +/* R12 (0x0C) CS53L30_ASPCFG_CTL - ASP Configuration Control */ +#define CS53L30_ASP_MS_SHIFT 7 +#define CS53L30_ASP_MS_MASK (1 << CS53L30_ASP_MS_SHIFT) +#define CS53L30_ASP_MS (1 << CS53L30_ASP_MS_SHIFT) +#define CS53L30_ASP_SCLK_INV_SHIFT 4 +#define CS53L30_ASP_SCLK_INV_MASK (1 << CS53L30_ASP_SCLK_INV_SHIFT) +#define CS53L30_ASP_SCLK_INV (1 << CS53L30_ASP_SCLK_INV_SHIFT) +#define CS53L30_ASP_RATE_SHIFT 0 +#define CS53L30_ASP_RATE_WIDTH 4 +#define CS53L30_ASP_RATE_MASK (((1 << CS53L30_ASP_RATE_WIDTH) - 1) << CS53L30_ASP_RATE_SHIFT) +#define CS53L30_ASP_RATE_48K (0xc << CS53L30_ASP_RATE_SHIFT) + +#define CS53L30_ASPCFG_CTL_DEFAULT (CS53L30_ASP_RATE_48K) + +/* R13/R24 (0x0D/0x18) CS53L30_ASP_CTL1 & CS53L30_ASP_CTL2 - ASP Control 1~2 */ +#define CS53L30_ASP_TDM_PDN_SHIFT 7 +#define CS53L30_ASP_TDM_PDN_MASK (1 << CS53L30_ASP_TDM_PDN_SHIFT) +#define CS53L30_ASP_TDM_PDN (1 << CS53L30_ASP_TDM_PDN_SHIFT) +#define CS53L30_ASP_SDOUTx_PDN_SHIFT 6 +#define CS53L30_ASP_SDOUTx_PDN_MASK (1 << CS53L30_ASP_SDOUTx_PDN_SHIFT) +#define CS53L30_ASP_SDOUTx_PDN (1 << CS53L30_ASP_SDOUTx_PDN_SHIFT) +#define CS53L30_ASP_3ST_SHIFT 5 +#define CS53L30_ASP_3ST_MASK (1 << CS53L30_ASP_3ST_SHIFT) +#define CS53L30_ASP_3ST (1 << CS53L30_ASP_3ST_SHIFT) +#define CS53L30_SHIFT_LEFT_SHIFT 4 +#define CS53L30_SHIFT_LEFT_MASK (1 << CS53L30_SHIFT_LEFT_SHIFT) +#define CS53L30_SHIFT_LEFT (1 << CS53L30_SHIFT_LEFT_SHIFT) +#define CS53L30_ASP_SDOUTx_DRIVE_SHIFT 0 +#define CS53L30_ASP_SDOUTx_DRIVE_MASK (1 << CS53L30_ASP_SDOUTx_DRIVE_SHIFT) +#define CS53L30_ASP_SDOUTx_DRIVE (1 << CS53L30_ASP_SDOUTx_DRIVE_SHIFT) + +#define CS53L30_ASP_CTL1_DEFAULT (CS53L30_ASP_TDM_PDN) +#define CS53L30_ASP_CTL2_DEFAULT (0) + +/* R14 (0x0E) ~ R17 (0x11) CS53L30_ASP_TDMTX_CTLx - ASP TDM TX Control 1~4 */ +#define CS53L30_ASP_CHx_TX_STATE_SHIFT 7 +#define CS53L30_ASP_CHx_TX_STATE_MASK (1 << CS53L30_ASP_CHx_TX_STATE_SHIFT) +#define CS53L30_ASP_CHx_TX_STATE (1 << CS53L30_ASP_CHx_TX_STATE_SHIFT) +#define CS53L30_ASP_CHx_TX_LOC_SHIFT 0 +#define CS53L30_ASP_CHx_TX_LOC_WIDTH 6 +#define CS53L30_ASP_CHx_TX_LOC_MASK (((1 << CS53L30_ASP_CHx_TX_LOC_WIDTH) - 1) << CS53L30_ASP_CHx_TX_LOC_SHIFT) +#define CS53L30_ASP_CHx_TX_LOC_MAX (47 << CS53L30_ASP_CHx_TX_LOC_SHIFT) +#define CS53L30_ASP_CHx_TX_LOC(x) ((x) << CS53L30_ASP_CHx_TX_LOC_SHIFT) + +#define CS53L30_ASP_TDMTX_CTLx_DEFAULT (CS53L30_ASP_CHx_TX_LOC_MAX) + +/* R18 (0x12) ~ R23 (0x17) CS53L30_ASP_TDMTX_ENx - ASP TDM TX Enable 1~6 */ +#define CS53L30_ASP_TDMTX_ENx_DEFAULT (0) + +/* R26 (0x1A) CS53L30_SFT_RAMP - Soft Ramp Control */ +#define CS53L30_DIGSFT_SHIFT 5 +#define CS53L30_DIGSFT_MASK (1 << CS53L30_DIGSFT_SHIFT) +#define CS53L30_DIGSFT (1 << CS53L30_DIGSFT_SHIFT) + +#define CS53L30_SFT_RMP_DEFAULT (0) + +/* R28 (0x1C) CS53L30_LRCK_CTL2 - LRCK Control 2 */ +#define CS53L30_LRCK_50_NPW_SHIFT 3 +#define CS53L30_LRCK_50_NPW_MASK (1 << CS53L30_LRCK_50_NPW_SHIFT) +#define CS53L30_LRCK_50_NPW (1 << CS53L30_LRCK_50_NPW_SHIFT) +#define CS53L30_LRCK_TPWH_SHIFT 0 +#define CS53L30_LRCK_TPWH_WIDTH 3 +#define CS53L30_LRCK_TPWH_MASK (((1 << CS53L30_LRCK_TPWH_WIDTH) - 1) << CS53L30_LRCK_TPWH_SHIFT) +#define CS53L30_LRCK_TPWH(x) (((x) << CS53L30_LRCK_TPWH_SHIFT) & CS53L30_LRCK_TPWH_MASK) + +#define CS53L30_LRCK_CTLx_DEFAULT (0) + +/* R31 (0x1F) CS53L30_MUTEP_CTL1 - MUTE Pin Control 1 */ +#define CS53L30_MUTE_PDN_ULP_SHIFT 7 +#define CS53L30_MUTE_PDN_ULP_MASK (1 << CS53L30_MUTE_PDN_ULP_SHIFT) +#define CS53L30_MUTE_PDN_ULP (1 << CS53L30_MUTE_PDN_ULP_SHIFT) +#define CS53L30_MUTE_PDN_LP_SHIFT 6 +#define CS53L30_MUTE_PDN_LP_MASK (1 << CS53L30_MUTE_PDN_LP_SHIFT) +#define CS53L30_MUTE_PDN_LP (1 << CS53L30_MUTE_PDN_LP_SHIFT) +#define CS53L30_MUTE_M4B_PDN_SHIFT 4 +#define CS53L30_MUTE_M4B_PDN_MASK (1 << CS53L30_MUTE_M4B_PDN_SHIFT) +#define CS53L30_MUTE_M4B_PDN (1 << CS53L30_MUTE_M4B_PDN_SHIFT) +#define CS53L30_MUTE_M3B_PDN_SHIFT 3 +#define CS53L30_MUTE_M3B_PDN_MASK (1 << CS53L30_MUTE_M3B_PDN_SHIFT) +#define CS53L30_MUTE_M3B_PDN (1 << CS53L30_MUTE_M3B_PDN_SHIFT) +#define CS53L30_MUTE_M2B_PDN_SHIFT 2 +#define CS53L30_MUTE_M2B_PDN_MASK (1 << CS53L30_MUTE_M2B_PDN_SHIFT) +#define CS53L30_MUTE_M2B_PDN (1 << CS53L30_MUTE_M2B_PDN_SHIFT) +#define CS53L30_MUTE_M1B_PDN_SHIFT 1 +#define CS53L30_MUTE_M1B_PDN_MASK (1 << CS53L30_MUTE_M1B_PDN_SHIFT) +#define CS53L30_MUTE_M1B_PDN (1 << CS53L30_MUTE_M1B_PDN_SHIFT) +/* Note: be careful - x starts from 0 */ +#define CS53L30_MUTE_MxB_PDN_SHIFT(x) (CS53L30_MUTE_M1B_PDN_SHIFT + (x)) +#define CS53L30_MUTE_MxB_PDN_MASK(x) (1 << CS53L30_MUTE_MxB_PDN_SHIFT(x)) +#define CS53L30_MUTE_MxB_PDN(x) (1 << CS53L30_MUTE_MxB_PDN_SHIFT(x)) +#define CS53L30_MUTE_MB_ALL_PDN_SHIFT 0 +#define CS53L30_MUTE_MB_ALL_PDN_MASK (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) +#define CS53L30_MUTE_MB_ALL_PDN (1 << CS53L30_MUTE_MB_ALL_PDN_SHIFT) + +#define CS53L30_MUTEP_CTL1_DEFAULT (0) + +/* R32 (0x20) CS53L30_MUTEP_CTL2 - MUTE Pin Control 2 */ +#define CS53L30_MUTE_PIN_POLARITY_SHIFT 7 +#define CS53L30_MUTE_PIN_POLARITY_MASK (1 << CS53L30_MUTE_PIN_POLARITY_SHIFT) +#define CS53L30_MUTE_PIN_POLARITY (1 << CS53L30_MUTE_PIN_POLARITY_SHIFT) +#define CS53L30_MUTE_ASP_TDM_PDN_SHIFT 6 +#define CS53L30_MUTE_ASP_TDM_PDN_MASK (1 << CS53L30_MUTE_ASP_TDM_PDN_SHIFT) +#define CS53L30_MUTE_ASP_TDM_PDN (1 << CS53L30_MUTE_ASP_TDM_PDN_SHIFT) +#define CS53L30_MUTE_ASP_SDOUT2_PDN_SHIFT 5 +#define CS53L30_MUTE_ASP_SDOUT2_PDN_MASK (1 << CS53L30_MUTE_ASP_SDOUT2_PDN_SHIFT) +#define CS53L30_MUTE_ASP_SDOUT2_PDN (1 << CS53L30_MUTE_ASP_SDOUT2_PDN_SHIFT) +#define CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT 4 +#define CS53L30_MUTE_ASP_SDOUT1_PDN_MASK (1 << CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT) +#define CS53L30_MUTE_ASP_SDOUT1_PDN (1 << CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT) +/* Note: be careful - x starts from 0 */ +#define CS53L30_MUTE_ASP_SDOUTx_PDN_SHIFT(x) ((x) + CS53L30_MUTE_ASP_SDOUT1_PDN_SHIFT) +#define CS53L30_MUTE_ASP_SDOUTx_PDN_MASK(x) (1 << CS53L30_MUTE_ASP_SDOUTx_PDN_SHIFT(x)) +#define CS53L30_MUTE_ASP_SDOUTx_PDN (1 << CS53L30_MUTE_ASP_SDOUTx_PDN_SHIFT(x)) +#define CS53L30_MUTE_ADC2B_PDN_SHIFT 3 +#define CS53L30_MUTE_ADC2B_PDN_MASK (1 << CS53L30_MUTE_ADC2B_PDN_SHIFT) +#define CS53L30_MUTE_ADC2B_PDN (1 << CS53L30_MUTE_ADC2B_PDN_SHIFT) +#define CS53L30_MUTE_ADC2A_PDN_SHIFT 2 +#define CS53L30_MUTE_ADC2A_PDN_MASK (1 << CS53L30_MUTE_ADC2A_PDN_SHIFT) +#define CS53L30_MUTE_ADC2A_PDN (1 << CS53L30_MUTE_ADC2A_PDN_SHIFT) +#define CS53L30_MUTE_ADC1B_PDN_SHIFT 1 +#define CS53L30_MUTE_ADC1B_PDN_MASK (1 << CS53L30_MUTE_ADC1B_PDN_SHIFT) +#define CS53L30_MUTE_ADC1B_PDN (1 << CS53L30_MUTE_ADC1B_PDN_SHIFT) +#define CS53L30_MUTE_ADC1A_PDN_SHIFT 0 +#define CS53L30_MUTE_ADC1A_PDN_MASK (1 << CS53L30_MUTE_ADC1A_PDN_SHIFT) +#define CS53L30_MUTE_ADC1A_PDN (1 << CS53L30_MUTE_ADC1A_PDN_SHIFT) + +#define CS53L30_MUTEP_CTL2_DEFAULT (CS53L30_MUTE_PIN_POLARITY) + +/* R33 (0x21) CS53L30_INBIAS_CTL1 - Input Bias Control 1 */ +#define CS53L30_IN4M_BIAS_SHIFT 6 +#define CS53L30_IN4M_BIAS_WIDTH 2 +#define CS53L30_IN4M_BIAS_MASK (((1 << CS53L30_IN4M_BIAS_WIDTH) - 1) << CS53L30_IN4M_BIAS_SHIFT) +#define CS53L30_IN4M_BIAS_OPEN (0 << CS53L30_IN4M_BIAS_SHIFT) +#define CS53L30_IN4M_BIAS_PULL_DOWN (1 << CS53L30_IN4M_BIAS_SHIFT) +#define CS53L30_IN4M_BIAS_VCM (2 << CS53L30_IN4M_BIAS_SHIFT) +#define CS53L30_IN4P_BIAS_SHIFT 4 +#define CS53L30_IN4P_BIAS_WIDTH 2 +#define CS53L30_IN4P_BIAS_MASK (((1 << CS53L30_IN4P_BIAS_WIDTH) - 1) << CS53L30_IN4P_BIAS_SHIFT) +#define CS53L30_IN4P_BIAS_OPEN (0 << CS53L30_IN4P_BIAS_SHIFT) +#define CS53L30_IN4P_BIAS_PULL_DOWN (1 << CS53L30_IN4P_BIAS_SHIFT) +#define CS53L30_IN4P_BIAS_VCM (2 << CS53L30_IN4P_BIAS_SHIFT) +#define CS53L30_IN3M_BIAS_SHIFT 2 +#define CS53L30_IN3M_BIAS_WIDTH 2 +#define CS53L30_IN3M_BIAS_MASK (((1 << CS53L30_IN3M_BIAS_WIDTH) - 1) << CS53L30_IN4M_BIAS_SHIFT) +#define CS53L30_IN3M_BIAS_OPEN (0 << CS53L30_IN3M_BIAS_SHIFT) +#define CS53L30_IN3M_BIAS_PULL_DOWN (1 << CS53L30_IN3M_BIAS_SHIFT) +#define CS53L30_IN3M_BIAS_VCM (2 << CS53L30_IN3M_BIAS_SHIFT) +#define CS53L30_IN3P_BIAS_SHIFT 0 +#define CS53L30_IN3P_BIAS_WIDTH 2 +#define CS53L30_IN3P_BIAS_MASK (((1 << CS53L30_IN3P_BIAS_WIDTH) - 1) << CS53L30_IN3P_BIAS_SHIFT) +#define CS53L30_IN3P_BIAS_OPEN (0 << CS53L30_IN3P_BIAS_SHIFT) +#define CS53L30_IN3P_BIAS_PULL_DOWN (1 << CS53L30_IN3P_BIAS_SHIFT) +#define CS53L30_IN3P_BIAS_VCM (2 << CS53L30_IN3P_BIAS_SHIFT) + +#define CS53L30_INBIAS_CTL1_DEFAULT (CS53L30_IN4M_BIAS_VCM | CS53L30_IN4P_BIAS_VCM |\ + CS53L30_IN3M_BIAS_VCM | CS53L30_IN3P_BIAS_VCM) + +/* R34 (0x22) CS53L30_INBIAS_CTL2 - Input Bias Control 2 */ +#define CS53L30_IN2M_BIAS_SHIFT 6 +#define CS53L30_IN2M_BIAS_WIDTH 2 +#define CS53L30_IN2M_BIAS_MASK (((1 << CS53L30_IN2M_BIAS_WIDTH) - 1) << CS53L30_IN2M_BIAS_SHIFT) +#define CS53L30_IN2M_BIAS_OPEN (0 << CS53L30_IN2M_BIAS_SHIFT) +#define CS53L30_IN2M_BIAS_PULL_DOWN (1 << CS53L30_IN2M_BIAS_SHIFT) +#define CS53L30_IN2M_BIAS_VCM (2 << CS53L30_IN2M_BIAS_SHIFT) +#define CS53L30_IN2P_BIAS_SHIFT 4 +#define CS53L30_IN2P_BIAS_WIDTH 2 +#define CS53L30_IN2P_BIAS_MASK (((1 << CS53L30_IN2P_BIAS_WIDTH) - 1) << CS53L30_IN2P_BIAS_SHIFT) +#define CS53L30_IN2P_BIAS_OPEN (0 << CS53L30_IN2P_BIAS_SHIFT) +#define CS53L30_IN2P_BIAS_PULL_DOWN (1 << CS53L30_IN2P_BIAS_SHIFT) +#define CS53L30_IN2P_BIAS_VCM (2 << CS53L30_IN2P_BIAS_SHIFT) +#define CS53L30_IN1M_BIAS_SHIFT 2 +#define CS53L30_IN1M_BIAS_WIDTH 2 +#define CS53L30_IN1M_BIAS_MASK (((1 << CS53L30_IN1M_BIAS_WIDTH) - 1) << CS53L30_IN1M_BIAS_SHIFT) +#define CS53L30_IN1M_BIAS_OPEN (0 << CS53L30_IN1M_BIAS_SHIFT) +#define CS53L30_IN1M_BIAS_PULL_DOWN (1 << CS53L30_IN1M_BIAS_SHIFT) +#define CS53L30_IN1M_BIAS_VCM (2 << CS53L30_IN1M_BIAS_SHIFT) +#define CS53L30_IN1P_BIAS_SHIFT 0 +#define CS53L30_IN1P_BIAS_WIDTH 2 +#define CS53L30_IN1P_BIAS_MASK (((1 << CS53L30_IN1P_BIAS_WIDTH) - 1) << CS53L30_IN1P_BIAS_SHIFT) +#define CS53L30_IN1P_BIAS_OPEN (0 << CS53L30_IN1P_BIAS_SHIFT) +#define CS53L30_IN1P_BIAS_PULL_DOWN (1 << CS53L30_IN1P_BIAS_SHIFT) +#define CS53L30_IN1P_BIAS_VCM (2 << CS53L30_IN1P_BIAS_SHIFT) + +#define CS53L30_INBIAS_CTL2_DEFAULT (CS53L30_IN2M_BIAS_VCM | CS53L30_IN2P_BIAS_VCM |\ + CS53L30_IN1M_BIAS_VCM | CS53L30_IN1P_BIAS_VCM) + +/* R35 (0x23) & R36 (0x24) CS53L30_DMICx_STR_CTL - DMIC1 & DMIC2 Stereo Control */ +#define CS53L30_DMICx_STEREO_ENB_SHIFT 5 +#define CS53L30_DMICx_STEREO_ENB_MASK (1 << CS53L30_DMICx_STEREO_ENB_SHIFT) +#define CS53L30_DMICx_STEREO_ENB (1 << CS53L30_DMICx_STEREO_ENB_SHIFT) + +/* 0x88 and 0xCC are reserved bits */ +#define CS53L30_DMIC1_STR_CTL_DEFAULT (CS53L30_DMICx_STEREO_ENB | 0x88) +#define CS53L30_DMIC2_STR_CTL_DEFAULT (CS53L30_DMICx_STEREO_ENB | 0xCC) + +/* R37/R45 (0x25/0x2D) CS53L30_ADCDMICx_CTL1 - ADC1/DMIC1 & ADC2/DMIC2 Control 1 */ +#define CS53L30_ADCxB_PDN_SHIFT 7 +#define CS53L30_ADCxB_PDN_MASK (1 << CS53L30_ADCxB_PDN_SHIFT) +#define CS53L30_ADCxB_PDN (1 << CS53L30_ADCxB_PDN_SHIFT) +#define CS53L30_ADCxA_PDN_SHIFT 6 +#define CS53L30_ADCxA_PDN_MASK (1 << CS53L30_ADCxA_PDN_SHIFT) +#define CS53L30_ADCxA_PDN (1 << CS53L30_ADCxA_PDN_SHIFT) +#define CS53L30_DMICx_PDN_SHIFT 2 +#define CS53L30_DMICx_PDN_MASK (1 << CS53L30_DMICx_PDN_SHIFT) +#define CS53L30_DMICx_PDN (1 << CS53L30_DMICx_PDN_SHIFT) +#define CS53L30_DMICx_SCLK_DIV_SHIFT 1 +#define CS53L30_DMICx_SCLK_DIV_MASK (1 << CS53L30_DMICx_SCLK_DIV_SHIFT) +#define CS53L30_DMICx_SCLK_DIV (1 << CS53L30_DMICx_SCLK_DIV_SHIFT) +#define CS53L30_CH_TYPE_SHIFT 0 +#define CS53L30_CH_TYPE_MASK (1 << CS53L30_CH_TYPE_SHIFT) +#define CS53L30_CH_TYPE (1 << CS53L30_CH_TYPE_SHIFT) + +#define CS53L30_ADCDMICx_PDN_MASK 0xFF +#define CS53L30_ADCDMICx_CTL1_DEFAULT (CS53L30_DMICx_PDN) + +/* R38/R46 (0x26/0x2E) CS53L30_ADCDMICx_CTL2 - ADC1/DMIC1 & ADC2/DMIC2 Control 2 */ +#define CS53L30_ADCx_NOTCH_DIS_SHIFT 7 +#define CS53L30_ADCx_NOTCH_DIS_MASK (1 << CS53L30_ADCx_NOTCH_DIS_SHIFT) +#define CS53L30_ADCx_NOTCH_DIS (1 << CS53L30_ADCx_NOTCH_DIS_SHIFT) +#define CS53L30_ADCxB_INV_SHIFT 5 +#define CS53L30_ADCxB_INV_MASK (1 << CS53L30_ADCxB_INV_SHIFT) +#define CS53L30_ADCxB_INV (1 << CS53L30_ADCxB_INV_SHIFT) +#define CS53L30_ADCxA_INV_SHIFT 4 +#define CS53L30_ADCxA_INV_MASK (1 << CS53L30_ADCxA_INV_SHIFT) +#define CS53L30_ADCxA_INV (1 << CS53L30_ADCxA_INV_SHIFT) +#define CS53L30_ADCxB_DIG_BOOST_SHIFT 1 +#define CS53L30_ADCxB_DIG_BOOST_MASK (1 << CS53L30_ADCxB_DIG_BOOST_SHIFT) +#define CS53L30_ADCxB_DIG_BOOST (1 << CS53L30_ADCxB_DIG_BOOST_SHIFT) +#define CS53L30_ADCxA_DIG_BOOST_SHIFT 0 +#define CS53L30_ADCxA_DIG_BOOST_MASK (1 << CS53L30_ADCxA_DIG_BOOST_SHIFT) +#define CS53L30_ADCxA_DIG_BOOST (1 << CS53L30_ADCxA_DIG_BOOST_SHIFT) + +#define CS53L30_ADCDMIC1_CTL2_DEFAULT (0) + +/* R39/R47 (0x27/0x2F) CS53L30_ADCx_CTL3 - ADC1/ADC2 Control 3 */ +#define CS53L30_ADCx_HPF_EN_SHIFT 3 +#define CS53L30_ADCx_HPF_EN_MASK (1 << CS53L30_ADCx_HPF_EN_SHIFT) +#define CS53L30_ADCx_HPF_EN (1 << CS53L30_ADCx_HPF_EN_SHIFT) +#define CS53L30_ADCx_HPF_CF_SHIFT 1 +#define CS53L30_ADCx_HPF_CF_WIDTH 2 +#define CS53L30_ADCx_HPF_CF_MASK (((1 << CS53L30_ADCx_HPF_CF_WIDTH) - 1) << CS53L30_ADCx_HPF_CF_SHIFT) +#define CS53L30_ADCx_HPF_CF_1HZ86 (0 << CS53L30_ADCx_HPF_CF_SHIFT) +#define CS53L30_ADCx_HPF_CF_120HZ (1 << CS53L30_ADCx_HPF_CF_SHIFT) +#define CS53L30_ADCx_HPF_CF_235HZ (2 << CS53L30_ADCx_HPF_CF_SHIFT) +#define CS53L30_ADCx_HPF_CF_466HZ (3 << CS53L30_ADCx_HPF_CF_SHIFT) +#define CS53L30_ADCx_NG_ALL_SHIFT 0 +#define CS53L30_ADCx_NG_ALL_MASK (1 << CS53L30_ADCx_NG_ALL_SHIFT) +#define CS53L30_ADCx_NG_ALL (1 << CS53L30_ADCx_NG_ALL_SHIFT) + +#define CS53L30_ADCx_CTL3_DEFAULT (CS53L30_ADCx_HPF_EN) + +/* R40/R48 (0x28/0x30) CS53L30_ADCx_NG_CTL - ADC1/ADC2 Noise Gate Control */ +#define CS53L30_ADCxB_NG_SHIFT 7 +#define CS53L30_ADCxB_NG_MASK (1 << CS53L30_ADCxB_NG_SHIFT) +#define CS53L30_ADCxB_NG (1 << CS53L30_ADCxB_NG_SHIFT) +#define CS53L30_ADCxA_NG_SHIFT 6 +#define CS53L30_ADCxA_NG_MASK (1 << CS53L30_ADCxA_NG_SHIFT) +#define CS53L30_ADCxA_NG (1 << CS53L30_ADCxA_NG_SHIFT) +#define CS53L30_ADCx_NG_BOOST_SHIFT 5 +#define CS53L30_ADCx_NG_BOOST_MASK (1 << CS53L30_ADCx_NG_BOOST_SHIFT) +#define CS53L30_ADCx_NG_BOOST (1 << CS53L30_ADCx_NG_BOOST_SHIFT) +#define CS53L30_ADCx_NG_THRESH_SHIFT 2 +#define CS53L30_ADCx_NG_THRESH_WIDTH 3 +#define CS53L30_ADCx_NG_THRESH_MASK (((1 << CS53L30_ADCx_NG_THRESH_WIDTH) - 1) << CS53L30_ADCx_NG_THRESH_SHIFT) +#define CS53L30_ADCx_NG_DELAY_SHIFT 0 +#define CS53L30_ADCx_NG_DELAY_WIDTH 2 +#define CS53L30_ADCx_NG_DELAY_MASK (((1 << CS53L30_ADCx_NG_DELAY_WIDTH) - 1) << CS53L30_ADCx_NG_DELAY_SHIFT) + +#define CS53L30_ADCx_NG_CTL_DEFAULT (0) + +/* R41/R42/R49/R50 (0x29/0x2A/0x31/0x32) CS53L30_ADCxy_AFE_CTL - ADC1A/1B/2A/2B AFE Control */ +#define CS53L30_ADCxy_PREAMP_SHIFT 6 +#define CS53L30_ADCxy_PREAMP_WIDTH 2 +#define CS53L30_ADCxy_PREAMP_MASK (((1 << CS53L30_ADCxy_PREAMP_WIDTH) - 1) << CS53L30_ADCxy_PREAMP_SHIFT) +#define CS53L30_ADCxy_PGA_VOL_SHIFT 0 +#define CS53L30_ADCxy_PGA_VOL_WIDTH 6 +#define CS53L30_ADCxy_PGA_VOL_MASK (((1 << CS53L30_ADCxy_PGA_VOL_WIDTH) - 1) << CS53L30_ADCxy_PGA_VOL_SHIFT) + +#define CS53L30_ADCxy_AFE_CTL_DEFAULT (0) + +/* R43/R44/R51/R52 (0x2B/0x2C/0x33/0x34) CS53L30_ADCxy_DIG_VOL - ADC1A/1B/2A/2B Digital Volume */ +#define CS53L30_ADCxy_VOL_MUTE (0x80) + +#define CS53L30_ADCxy_DIG_VOL_DEFAULT (0x0) + +/* CS53L30_INT */ +#define CS53L30_PDN_DONE (1 << 7) +#define CS53L30_THMS_TRIP (1 << 6) +#define CS53L30_SYNC_DONE (1 << 5) +#define CS53L30_ADC2B_OVFL (1 << 4) +#define CS53L30_ADC2A_OVFL (1 << 3) +#define CS53L30_ADC1B_OVFL (1 << 2) +#define CS53L30_ADC1A_OVFL (1 << 1) +#define CS53L30_MUTE_PIN (1 << 0) +#define CS53L30_DEVICE_INT_MASK 0xFF + +#endif /* __CS53L30_H__ */ -- 2.1.4

3 2

[alsa-devel] [PATCH v5 0/1] sti: add audio interface to the hdmi driver
by Arnaud Pouliquen 01 Jun '16

01 Jun '16

This patchset implements audio interface in HDMI drm driver. Implementation is based on ASoC generic hdmi codec driver( https://patchwork.kernel.org/patch/8713141/). V5: Helper functions are suppressed, following dicussion on IRC with Daniel Vetter. request is that other drivers that can use helper are also updated helper function patch will be proposed in a next, separed patchset. for this track, hdmi_audio_get_non_coherent_n is integrated in sti driver. V4: fixes for "video: hdmi: add helper functions for N and CTS" - typo error and additional comments - cts_1_ratio computation - warning reported by kbuild test robot - add rounded value for 297/1.001 MHz V3: - video: hdmi: add helper function for N and CTS Also used on Mediatek platform (https://patchwork.kernel.org/patch/8887341) delta vs V2: - typo fixes - if/else code optimisation - drm: sti: Add ASoC generic hdmi codec support. - typo fixes - add audio registers in debugfs information V2: RFC https://patchwork.kernel.org/patch/8091531/("video: hdmi: add helper function for N and CTS") https://patchwork.kernel.org/patch/8091561/("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") - patch: video: hdmi: add helper function for N and CTS Fixes based on Russel King remarks - Duplicate function to have a separte treatment for coherent and non-coherent clocks - Add ratio field for alternate CTS value - Clock frequency in Hz for TMDS and audio clocks - Add information concerning clocks and CTS calculation. V1: This RFC is the implementation of audio HDMI on sti platform based on generic hdmi-codec driver: https://patchwork.kernel.org/patch/7215271/ ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") https://patchwork.kernel.org/patch/8062611/ ("video: hdmi: add helper function for N and CTS") Arnaud Pouliquen (1): drm: sti: Add ASoC generic hdmi codec support. drivers/gpu/drm/sti/Kconfig | 1 + drivers/gpu/drm/sti/sti_hdmi.c | 294 ++++++++++++++++++++++++++++++++++++++--- drivers/gpu/drm/sti/sti_hdmi.h | 13 ++ 3 files changed, 291 insertions(+), 17 deletions(-) -- 1.9.1

3 3

[alsa-devel] [PATCH 1/1] alsabat: fix a possible memory leak
by han.lu＠intel.com 01 Jun '16

01 Jun '16

From: "Lu, Han" <han.lu(a)intel.com> Fix a possible memory leak in generate_sine_wave(). Memory free was ignored when the function return an error. Signed-off-by: Lu, Han <han.lu(a)intel.com> diff --git a/bat/signal.c b/bat/signal.c index a47ba97..61d2824 100644 --- a/bat/signal.c +++ b/bat/signal.c @@ -168,16 +168,17 @@ int generate_sine_wave(struct bat *bat, int frames, void *buf) /* reorder samples to interleaved mode */ err = reorder(bat, sinus_f, frames); if (err != 0) - return err; + goto exit; /* adjust amplitude and offset of waveform */ err = adjust_waveform(bat, sinus_f, frames); if (err != 0) - return err; + goto exit; bat->convert_float_to_sample(sinus_f, buf, frames, bat->channels); +exit: free(sinus_f); - return 0; + return err; } -- 2.5.0

2 1

[alsa-devel] [PATCH alsa-lib] async: Handle previously installed signal handler
by Takashi Iwai 01 Jun '16

01 Jun '16

From: Eliot Miranda <eliot.miranda(a)gmail.com> The issue is with the signal handler installed and deinstalled in alsa-lib async handler. This code makes no attempt to remember any previously installed signal handlers for SIGIO, if SIGIO is used. Consequently it does not call any previous handlers from its own handler once installed, and does not reinstall any previous handler when deinstalling its handler. Consequently, use of also-lib within applications that depend on SIGIO will break those applications, rendering them inoperative once alsa-lib is running because their signal handlers are no longer called. This patch does remember and restore any previous handler, and chains calls to the handler if it exists. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/async.c | 20 +++++++++++++------- 1 file changed, 13 insertions(+), 7 deletions(-) diff --git a/src/async.c b/src/async.c index 98aec78e00a8..0e133c3a34cd 100644 --- a/src/async.c +++ b/src/async.c @@ -28,6 +28,9 @@ #include "control/control_local.h" #include <signal.h> +static struct sigaction previous_action; +#define MAX_SIG_FUNCTION_CODE 10 /* i.e. SIG_DFL SIG_IGN SIG_HOLD et al */ + #ifdef SND_ASYNC_RT_SIGNAL /** async signal number */ static int snd_async_signo; @@ -54,6 +57,9 @@ static void snd_async_handler(int signo ATTRIBUTE_UNUSED, siginfo_t *siginfo, vo int fd; struct list_head *i; //assert(siginfo->si_code == SI_SIGIO); + if (signo == SIGIO + && (unsigned long)(previous_action.sa_sigaction) > MAX_SIG_FUNCTION_CODE) + previous_action.sa_sigaction(signo, siginfo, context); fd = siginfo->si_fd; list_for_each(i, &snd_async_handlers) { snd_async_handler_t *h = list_entry(i, snd_async_handler_t, glist); @@ -114,7 +120,8 @@ int snd_async_add_handler(snd_async_handler_t **handler, int fd, act.sa_flags = SA_RESTART | SA_SIGINFO; act.sa_sigaction = snd_async_handler; sigemptyset(&act.sa_mask); - err = sigaction(snd_async_signo, &act, NULL); + assert(!previous_action.sa_sigaction); + err = sigaction(snd_async_signo, &act, &previous_action); if (err < 0) { SYSERR("sigaction"); return -errno; @@ -131,18 +138,17 @@ int snd_async_add_handler(snd_async_handler_t **handler, int fd, int snd_async_del_handler(snd_async_handler_t *handler) { int err = 0; + int was_empty = list_empty(&snd_async_handlers); assert(handler); list_del(&handler->glist); - if (list_empty(&snd_async_handlers)) { - struct sigaction act; - memset(&act, 0, sizeof(act)); - act.sa_flags = 0; - act.sa_handler = SIG_DFL; - err = sigaction(snd_async_signo, &act, NULL); + if (!was_empty + && list_empty(&snd_async_handlers)) { + err = sigaction(snd_async_signo, &previous_action, NULL); if (err < 0) { SYSERR("sigaction"); return -errno; } + memset(&previous_action, 0, sizeof(previous_action)); } if (handler->type == SND_ASYNC_HANDLER_GENERIC) goto _end; -- 2.8.3

1 0

[alsa-devel] [PATCH 00/11] ALSA: Utilize the module_isa_driver macro
by William Breathitt Gray 01 Jun '16

01 Jun '16

The module_isa_driver macro is a helper macro for ISA drivers which do not do anything special in module init/exit. This patchset eliminates a lot of ISA driver registration boilerplate code by utilizing module_isa_driver, which replaces module_init and module_exit. William Breathitt Gray (11): ALSA: sb8: Utilize the module_isa_driver macro ALSA: jazz16: Utilize the module_isa_driver macro ALSA: ad1848: Utilize the module_isa_driver macro ALSA: cmi8328: Utilize the module_isa_driver macro ALSA: cs4231: Utilize the module_isa_driver macro ALSA: gusmax: Utilize the module_isa_driver macro ALSA: gusextreme: Utilize the module_isa_driver macro ALSA: gusclassic: Utilize the module_isa_driver macro ALSA: sc6000: Utilize the module_isa_driver macro ALSA: galaxy: Utilize the module_isa_driver macro ALSA: adlib: Utilize the module_isa_driver macro sound/isa/ad1848/ad1848.c | 13 +------------ sound/isa/adlib.c | 13 +------------ sound/isa/cmi8328.c | 13 +------------ sound/isa/cs423x/cs4231.c | 13 +------------ sound/isa/galaxy/galaxy.c | 13 +------------ sound/isa/gus/gusclassic.c | 13 +------------ sound/isa/gus/gusextreme.c | 13 +------------ sound/isa/gus/gusmax.c | 13 +------------ sound/isa/sb/jazz16.c | 13 +------------ sound/isa/sb/sb8.c | 13 +------------ sound/isa/sc6000.c | 13 +------------ 11 files changed, 11 insertions(+), 132 deletions(-) -- 2.7.3

2 12

[alsa-devel] [PATCH] pcm: Don't store the state for SND_PCM_STATE_SUSPENDED
by Shengjiu Wang 01 Jun '16

01 Jun '16

The resume function don't update the dmix->state, if store SUSPENDED state in snd_pcm_dmix_state, the write function after resume will return error -ESTRPIPE, because the snd_pcm_write_areas() will check the state of the pcm device. This patch remove the store SND_PCM_STATE_SUSPENDED state operation for dmix,dshare,dsnoop. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)freescale.com> --- src/pcm/pcm_dmix.c | 2 +- src/pcm/pcm_dshare.c | 2 +- src/pcm/pcm_dsnoop.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 007d356..66bb288 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -451,9 +451,9 @@ static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) state = snd_pcm_state(dmix->spcm); switch (state) { case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; + case SND_PCM_STATE_SUSPENDED: return state; default: break; diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index adb3587..a133c72 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -241,9 +241,9 @@ static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) state = snd_pcm_state(dshare->spcm); switch (state) { case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; + case SND_PCM_STATE_SUSPENDED: return state; default: break; diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 8ff0ba5..d56dd97 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -205,9 +205,9 @@ static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) state = snd_pcm_state(dsnoop->spcm); switch (state) { case SND_PCM_STATE_XRUN: - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; + case SND_PCM_STATE_SUSPENDED: return state; default: break; -- 1.9.1

3 21

[alsa-devel] [GIT PULL]: Add audio firmware for Broxton
by Vinod Koul 01 Jun '16

01 Jun '16

Hello Linux firmware maintainers, Please PULL to receive the Intel audio firmware for audio DSP in Broxton platforms. This firmware comes with same license as audio firmware for Skylake and is fully redistributable. The following changes since commit 7db6ab934fc0196c41f40ea07218de2cab27abb5: qed: Add FW 8.10.5.0 (2016-05-29 10:16:35 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/firmware.git apl for you to fetch changes up to f5598347032b1e4dd49e15401a1a9b867062625c: linux-firmware: intel: Add Broxton audio firmware (2016-05-31 09:20:44 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Add Broxton audio firmware WHENCE | 6 ++++++ intel/dsp_fw_bxtn.bin | 1 + intel/dsp_fw_bxtn_v430.bin | Bin 0 -> 505608 bytes 3 files changed, 7 insertions(+) create mode 120000 intel/dsp_fw_bxtn.bin create mode 100644 intel/dsp_fw_bxtn_v430.bin Thanks -- ~Vinod

2 1

[alsa-devel] [PATCH] ASoC: sgtl5000: only check VDDD-supply, not revision
by Clemens Gruber 31 May '16

31 May '16

Instead of checking the SGTL5000 chip revision, we should only check if the VDDD regulator exists and only call sgtl5000_replace_vddd_with_ldo if the regulator is missing. Otherwise, the user reads in the kernel log that the internal LDO is used, even though he did follow the NXP recommendation to use external VDDD and also specified VDDD-supply in the devicetree. Also remove the comment, which incorrectly states that external VDDD is only supported for SGTL5000 chip revisions < 0x11. Official NXP documentation recommends using external VDDD and not the internal LDO due to the SGTL5000 erratum ER1. This also applies to revisions >= 0x11. Tested on an i.MX6Q board with SGTL5000 rev 0x11 and external VDDD. Signed-off-by: Clemens Gruber <clemens.gruber(a)pqgruber.com> --- sound/soc/codecs/sgtl5000.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 08b4046..fbad4fb 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1286,17 +1286,14 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - /* External VDDD only works before revision 0x11 */ - if (sgtl5000->revision < 0x11) { - vddd = regulator_get_optional(codec->dev, "VDDD"); - if (IS_ERR(vddd)) { - /* See if it's just not registered yet */ - if (PTR_ERR(vddd) == -EPROBE_DEFER) - return -EPROBE_DEFER; - } else { - external_vddd = 1; - regulator_put(vddd); - } + vddd = regulator_get_optional(codec->dev, "VDDD"); + if (IS_ERR(vddd)) { + /* See if it's just not registered yet */ + if (PTR_ERR(vddd) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + external_vddd = 1; + regulator_put(vddd); } if (!external_vddd) { -- 2.8.3

1 0

[alsa-devel] [RFC] sgtl5000: check VDDD-supply instead of revision
by Clemens Gruber 31 May '16

31 May '16

Hi, I am using a SGTL5000 rev 0x11 with external VDDD, as recommended by NXP documentation and errata sheet: http://cache.nxp.com/files/analog/doc/errata/SGTL5000ER.pdf The linux sgtl5000 codec driver contains a comment, warning that external VDDD can't be used on revisions >= 0x11, which is probably a leftover from the first implementation. This is very confusing because it contradicts the NXP documentation. Also, on every boot, the user gets a warning that the SGTL5000 is using the internal LDO (even though external VDDD is present). Maybe instead of checking the revision, we should only check if the VDDD-supply regulator exists and only if it does not exist, call sgtl5000_replace_vddd_with_ldo? I'd like to hear your comments if this approach is valid or if I missed something in my assumptions. In my DT, the VDDD-supply regulator is just a dummy / regulator-fixed with regulator-always-on; set, because the 1.8V external VDDD supply on my board is always powered on. I could not test if it works with regulator-fixed with gpio nodes, which actually switch the supply line on or off. If everything is OK, I'd send a formal patch to the mailing list. See below for the current state of my patch. Tested on an i.MX6Q board with SGTL5000 rev 0x11 and external VDDD. Thanks, Clemens --- sound/soc/codecs/sgtl5000.c | 19 ++++++++----------- 1 file changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 08b4046..fbad4fb 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1286,17 +1286,14 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(sgtl5000->supplies); i++) sgtl5000->supplies[i].supply = supply_names[i]; - /* External VDDD only works before revision 0x11 */ - if (sgtl5000->revision < 0x11) { - vddd = regulator_get_optional(codec->dev, "VDDD"); - if (IS_ERR(vddd)) { - /* See if it's just not registered yet */ - if (PTR_ERR(vddd) == -EPROBE_DEFER) - return -EPROBE_DEFER; - } else { - external_vddd = 1; - regulator_put(vddd); - } + vddd = regulator_get_optional(codec->dev, "VDDD"); + if (IS_ERR(vddd)) { + /* See if it's just not registered yet */ + if (PTR_ERR(vddd) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } else { + external_vddd = 1; + regulator_put(vddd); } if (!external_vddd) { -- 2.8.3

2 1

[alsa-devel] [PATCH 1/2 v2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 31 May '16

31 May '16

The driver changes jack type detection interruption to non-clock archi- tecture for less 1mW power saving. The architecture is called manual mode jack detection. It has no hardware debounce, no jack type detection, but only detecting jack insertion. After jack insertion, the driver will switch to auto mode jack detection with internal clock which can detect microphone, jack type and do hardware debounce. The manual architecture has these main changes including codec initiation, interruption, clock control, and power management. When codec initiation or system resume, the clock is closed as jack insertion detection at man- ual mode, and bypass debounce circuit. These configurations move to resume setup function when setup bias level after resume. When jack insertion detection happens, the manual mode turns off and make configuration about jack type detection interruption at auto mode in auto irq setup function which can detect microphone and jack type. The inter- ruption will switch to manual mode again with clock free until jack ejec- tion happens. The system clock configuration adds clock disable option which can disable internal VCO clock. Before the system clock change, there is an restric- tion added to make sure clock disabled and not config any clock when no headset connected. In power management, we involve the solution about races and jack detec- tion in resume from Ben Zhang in the following patch and list his comment. [PATCH] ASoC: nau8825: Fix jack detection across suspend "Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power." "Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly." Change SAR widget to supply type which can prevent the codec keeping at SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke normally. Before the system suspends, the driver turns off all interruptions. Keep the interruption quiet before resume setup completes. The ADC channel will be disabled which is needed for interruptions at audo mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 249 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 188 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 81fc97b..abaa5f4 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -30,10 +30,16 @@ #include "nau8825.h" + +#define NUVOTON_CODEC_DAI "nau8825-hifi" + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +static int nau8825_configure_sysclk(struct nau8825 *nau8825, + int clk_id, unsigned int freq); + struct nau8825_fll { int mclk_src; int ratio; @@ -617,9 +623,6 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); - return 0; } EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); @@ -635,16 +638,6 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) static void nau8825_restart_jack_detection(struct regmap *regmap) { - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - /* this will restart the entire jack detection process including MIC/GND * switching and create interrupts. We have to go from 0 to 1 and back * to 0 to restart. @@ -655,6 +648,22 @@ static void nau8825_restart_jack_detection(struct regmap *regmap) NAU8825_JACK_DET_RESTART, 0); } +static void nau8825_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8825_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + } +} + static void nau8825_eject_jack(struct nau8825 *nau8825) { struct snd_soc_dapm_context *dapm = nau8825->dapm; @@ -669,6 +678,69 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable the insertion interruption, disable the ejection inter- + * ruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_EJECT_DIS | NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_EJECT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); + + /* Close clock for jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); +} + +/* Enable audo mode interruptions with internal clock. */ +static void nau8825_setup_auto_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable headset jack type detection complete interruption and + * jack ejection interruption. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interruptions */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, 0); + + /* Unmask all interruptions */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + + /* Restart the jack detection process at auto mode */ + nau8825_restart_jack_detection(regmap); } static int nau8825_button_decode(int value) @@ -805,6 +877,26 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == + NAU8825_JACK_INSERTION_DETECTED) { + /* One more step to check GPIO status directly. Thus, the + * driver can confirm the real insertion interruption because + * the intrruption at manual mode has bypassed debounce + * circuit which can get rid of unstable status. + */ + if (nau8825_is_jack_inserted(regmap)) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_INSERT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_INSERT_EN, NAU8825_IRQ_INSERT_EN); + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8825_setup_auto_irq(nau8825); + } } if (!clear_irq) @@ -954,8 +1046,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) } static const struct regmap_config nau8825_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = NAU8825_REG_DATA_LEN, + .reg_bits = NAU8825_REG_ADDR_LEN, .max_register = NAU8825_REG_MAX, .readable_reg = nau8825_readable_reg, @@ -974,12 +1066,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* Unmask interruptions. Handler uses dapm object so we can enable - * interruptions only after dapm is fully initialized. - */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); - return 0; } @@ -1144,6 +1230,14 @@ static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) return 0; } +static void nau8825_configure_mclk_as_sysclk(struct regmap *regmap) +{ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, 0); +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -1151,10 +1245,17 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, int ret; switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + nau8825_configure_mclk_as_sysclk(regmap); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; case NAU8825_CLK_MCLK: - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); - regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + nau8825_configure_mclk_as_sysclk(regmap); /* MCLK not changed by clock tree */ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0); @@ -1164,17 +1265,25 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_INTERNAL: - regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, - NAU8825_DCO_EN); - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); - /* Decrease the VCO frequency for power saving */ - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, 0xf); - regmap_update_bits(regmap, NAU8825_REG_FLL1, - NAU8825_FLL_RATIO_MASK, 0x10); - regmap_update_bits(regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN, NAU8825_SDM_EN); + if (nau8825_is_jack_inserted(nau8825->regmap)) { + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + /* Decrease the VCO frequency for power saving */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + /* The clock turns off intentionally for power saving + * when no headset connected. + */ + nau8825_configure_mclk_as_sysclk(regmap); + dev_warn(nau8825->dev, "Disable clock for power saving when no headset connected\n"); + } if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1225,6 +1334,31 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable both insertion and ejection interruptions, and then + * bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_HEADSET_COMPLETE_EN | + NAU8825_IRQ_EJECT_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS | NAU8825_IRQ_EJECT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1247,10 +1381,20 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return ret; } } + /* Setup codec configuration after resume */ + nau8825_resume_setup(nau8825); } break; case SND_SOC_BIAS_OFF: + /* Turn off all interruptions before system shutdown. Keep the + * interruption quiet before resume setup completes. + */ + regmap_write(nau8825->regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff); + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); break; @@ -1264,6 +1408,7 @@ static int nau8825_suspend(struct snd_soc_codec *codec) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); @@ -1274,32 +1419,10 @@ static int nau8825_resume(struct snd_soc_codec *codec) { struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); - /* The chip may lose power and reset in S3. regcache_sync restores - * register values including configurations for sysclk, irq, and - * jack/button detection. - */ regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); - - /* Check the jack plug status directly. If the headset is unplugged - * during S3 when the chip has no power, there will be no jack - * detection irq even after the nau8825_restart_jack_detection below, - * because the chip just thinks no headset has ever been plugged in. - */ - if (!nau8825_is_jack_inserted(nau8825->regmap)) { - nau8825_eject_jack(nau8825); - snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); - } - enable_irq(nau8825->irq); - /* Run jack detection to check the type (OMTP or CTIA) of the headset - * if there is one. This handles the case where a different type of - * headset is plugged in during S3. This triggers an IRQ iff a headset - * is already plugged in. - */ - nau8825_restart_jack_detection(nau8825->regmap); - return 0; } #else @@ -1408,20 +1531,8 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1100385 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -93,6 +93,9 @@ #define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 #define NAU8825_REG_GENERAL_STATUS 0x82 #define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8825_REG_ADDR_LEN 16 +#define NAU8825_REG_DATA_LEN 16 /* ENA_CTRL (0x1) */ #define NAU8825_ENABLE_DACR_SFT 10 @@ -145,6 +148,7 @@ /* JACK_DET_CTRL (0xd) */ #define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_DET_DB_BYPASS (1 << 8) #define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 #define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) #define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 @@ -157,6 +161,7 @@ #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) +#define NAU8825_IRQ_INSERT_EN (1 << 0) /* IRQ_STATUS (0x10) */ #define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) @@ -177,6 +182,7 @@ #define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) #define NAU8825_IRQ_EJECT_DIS (1 << 2) +#define NAU8825_IRQ_INSERT_DIS (1 << 0) /* SAR_CTRL (0x13) */ #define NAU8825_SAR_ADC_EN_SFT 12 @@ -333,7 +339,8 @@ /* System Clock Source */ enum { - NAU8825_CLK_MCLK = 0, + NAU8825_CLK_DIS = 0, + NAU8825_CLK_MCLK, NAU8825_CLK_INTERNAL, NAU8825_CLK_FLL_MCLK, NAU8825_CLK_FLL_BLK, -- 2.6.4

2 1

[alsa-devel] [PATCH v2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 31 May '16

31 May '16

To resolve the conflict of the first verion. The information is as the following. > > ASoC: nau8825: non-clock jack detection for power saving at standby > > has been applied to the asoc tree at > > git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git > This seems to conflict with some power management fixes in Linus' tree > so I've dropped it. Please check and resend. The driver changes jack type detection interruption to non-clock archi- tecture for less 1mW power saving. The architecture is called manual mode jack detection. It has no hardware debounce, no jack type detection, but only detecting jack insertion. After jack insertion, the driver will switch to auto mode jack detection with internal clock which can detect microphone, jack type and do hardware debounce. The manual architecture has these main changes including codec initiation, interruption, clock control, and power management. When codec initiation or system resume, the clock is closed as jack insertion detection at man- ual mode, and bypass debounce circuit. These configurations move to resume setup function when setup bias level after resume. When jack insertion detection happens, the manual mode turns off and make configuration about jack type detection interruption at auto mode in auto irq setup function which can detect microphone and jack type. The inter- ruption will switch to manual mode again with clock free until jack ejec- tion happens. The system clock configuration adds clock disable option which can disable internal VCO clock. Before the system clock change, there is an restric- tion added to make sure clock disabled and not config any clock when no headset connected. In power management, we involve the solution about races and jack detec- tion in resume from Ben Zhang in the following patch and list his comment. [PATCH] ASoC: nau8825: Fix jack detection across suspend "Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power." "Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly." Change SAR widget to supply type which can prevent the codec keeping at SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke normally. Before the system suspends, the driver turns off all interruptions. Keep the interruption quiet before resume setup completes. The ADC channel will be disabled which is needed for interruptions at audo mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 249 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 188 insertions(+), 70 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 81fc97b..abaa5f4 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -30,10 +30,16 @@ #include "nau8825.h" + +#define NUVOTON_CODEC_DAI "nau8825-hifi" + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +static int nau8825_configure_sysclk(struct nau8825 *nau8825, + int clk_id, unsigned int freq); + struct nau8825_fll { int mclk_src; int ratio; @@ -617,9 +623,6 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); - return 0; } EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); @@ -635,16 +638,6 @@ static bool nau8825_is_jack_inserted(struct regmap *regmap) static void nau8825_restart_jack_detection(struct regmap *regmap) { - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - /* this will restart the entire jack detection process including MIC/GND * switching and create interrupts. We have to go from 0 to 1 and back * to 0 to restart. @@ -655,6 +648,22 @@ static void nau8825_restart_jack_detection(struct regmap *regmap) NAU8825_JACK_DET_RESTART, 0); } +static void nau8825_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8825_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + } +} + static void nau8825_eject_jack(struct nau8825 *nau8825) { struct snd_soc_dapm_context *dapm = nau8825->dapm; @@ -669,6 +678,69 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable the insertion interruption, disable the ejection inter- + * ruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_EJECT_DIS | NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_EJECT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); + + /* Close clock for jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); +} + +/* Enable audo mode interruptions with internal clock. */ +static void nau8825_setup_auto_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable headset jack type detection complete interruption and + * jack ejection interruption. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interruptions */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, 0); + + /* Unmask all interruptions */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + + /* Restart the jack detection process at auto mode */ + nau8825_restart_jack_detection(regmap); } static int nau8825_button_decode(int value) @@ -805,6 +877,26 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == + NAU8825_JACK_INSERTION_DETECTED) { + /* One more step to check GPIO status directly. Thus, the + * driver can confirm the real insertion interruption because + * the intrruption at manual mode has bypassed debounce + * circuit which can get rid of unstable status. + */ + if (nau8825_is_jack_inserted(regmap)) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_INSERT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_INSERT_EN, NAU8825_IRQ_INSERT_EN); + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8825_setup_auto_irq(nau8825); + } } if (!clear_irq) @@ -954,8 +1046,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) } static const struct regmap_config nau8825_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = NAU8825_REG_DATA_LEN, + .reg_bits = NAU8825_REG_ADDR_LEN, .max_register = NAU8825_REG_MAX, .readable_reg = nau8825_readable_reg, @@ -974,12 +1066,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* Unmask interruptions. Handler uses dapm object so we can enable - * interruptions only after dapm is fully initialized. - */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); - return 0; } @@ -1144,6 +1230,14 @@ static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) return 0; } +static void nau8825_configure_mclk_as_sysclk(struct regmap *regmap) +{ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, 0); +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -1151,10 +1245,17 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, int ret; switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + nau8825_configure_mclk_as_sysclk(regmap); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; case NAU8825_CLK_MCLK: - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); - regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + nau8825_configure_mclk_as_sysclk(regmap); /* MCLK not changed by clock tree */ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0); @@ -1164,17 +1265,25 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_INTERNAL: - regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, - NAU8825_DCO_EN); - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); - /* Decrease the VCO frequency for power saving */ - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, 0xf); - regmap_update_bits(regmap, NAU8825_REG_FLL1, - NAU8825_FLL_RATIO_MASK, 0x10); - regmap_update_bits(regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN, NAU8825_SDM_EN); + if (nau8825_is_jack_inserted(nau8825->regmap)) { + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + /* Decrease the VCO frequency for power saving */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + /* The clock turns off intentionally for power saving + * when no headset connected. + */ + nau8825_configure_mclk_as_sysclk(regmap); + dev_warn(nau8825->dev, "Disable clock for power saving when no headset connected\n"); + } if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1225,6 +1334,31 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable both insertion and ejection interruptions, and then + * bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_HEADSET_COMPLETE_EN | + NAU8825_IRQ_EJECT_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS | NAU8825_IRQ_EJECT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1247,10 +1381,20 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return ret; } } + /* Setup codec configuration after resume */ + nau8825_resume_setup(nau8825); } break; case SND_SOC_BIAS_OFF: + /* Turn off all interruptions before system shutdown. Keep the + * interruption quiet before resume setup completes. + */ + regmap_write(nau8825->regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff); + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); break; @@ -1264,6 +1408,7 @@ static int nau8825_suspend(struct snd_soc_codec *codec) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(nau8825->regmap, true); regcache_mark_dirty(nau8825->regmap); @@ -1274,32 +1419,10 @@ static int nau8825_resume(struct snd_soc_codec *codec) { struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); - /* The chip may lose power and reset in S3. regcache_sync restores - * register values including configurations for sysclk, irq, and - * jack/button detection. - */ regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); - - /* Check the jack plug status directly. If the headset is unplugged - * during S3 when the chip has no power, there will be no jack - * detection irq even after the nau8825_restart_jack_detection below, - * because the chip just thinks no headset has ever been plugged in. - */ - if (!nau8825_is_jack_inserted(nau8825->regmap)) { - nau8825_eject_jack(nau8825); - snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); - } - enable_irq(nau8825->irq); - /* Run jack detection to check the type (OMTP or CTIA) of the headset - * if there is one. This handles the case where a different type of - * headset is plugged in during S3. This triggers an IRQ iff a headset - * is already plugged in. - */ - nau8825_restart_jack_detection(nau8825->regmap); - return 0; } #else @@ -1408,20 +1531,8 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1100385 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -93,6 +93,9 @@ #define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 #define NAU8825_REG_GENERAL_STATUS 0x82 #define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8825_REG_ADDR_LEN 16 +#define NAU8825_REG_DATA_LEN 16 /* ENA_CTRL (0x1) */ #define NAU8825_ENABLE_DACR_SFT 10 @@ -145,6 +148,7 @@ /* JACK_DET_CTRL (0xd) */ #define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_DET_DB_BYPASS (1 << 8) #define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 #define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) #define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 @@ -157,6 +161,7 @@ #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) +#define NAU8825_IRQ_INSERT_EN (1 << 0) /* IRQ_STATUS (0x10) */ #define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) @@ -177,6 +182,7 @@ #define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) #define NAU8825_IRQ_EJECT_DIS (1 << 2) +#define NAU8825_IRQ_INSERT_DIS (1 << 0) /* SAR_CTRL (0x13) */ #define NAU8825_SAR_ADC_EN_SFT 12 @@ -333,7 +339,8 @@ /* System Clock Source */ enum { - NAU8825_CLK_MCLK = 0, + NAU8825_CLK_DIS = 0, + NAU8825_CLK_MCLK, NAU8825_CLK_INTERNAL, NAU8825_CLK_FLL_MCLK, NAU8825_CLK_FLL_BLK, -- 2.6.4

2 1

[alsa-devel] [PATCH] ASoC: arizona: Add data structure for voice trigger notifier
by Charles Keepax 31 May '16

31 May '16

64-bit builds would generate a warning when we passed the core number as a pointer through the notifier data: sound/soc/codecs/cs47l24.c:1091:13: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] (void *)i); Rather than just fix this up with more casting add a data structure that holds information for the notifier chain. This will make it easier to add additional information in the future as well. Fixes: 7baa7e2490e1 ("ASoC: arizona: Add event notification on voice trigger events") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.h | 4 ++++ sound/soc/codecs/cs47l24.c | 7 +++++-- sound/soc/codecs/wm5110.c | 7 +++++-- 3 files changed, 14 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index ef38413..2caf4d0 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -98,6 +98,10 @@ struct arizona_priv { bool dvfs_cached; }; +struct arizona_voice_trigger_info { + int core; +}; + #define ARIZONA_NUM_MIXER_INPUTS 104 extern const unsigned int arizona_mixer_tlv[]; diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index b4d05cf..954a4f5 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1078,6 +1078,7 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data) { struct cs47l24_priv *priv = data; struct arizona *arizona = priv->core.arizona; + struct arizona_voice_trigger_info info; int serviced = 0; int i, ret; @@ -1085,10 +1086,12 @@ static irqreturn_t cs47l24_adsp2_irq(int irq, void *data) ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]); if (ret != -ENODEV) serviced++; - if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) + if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) { + info.core = i; arizona_call_notifiers(arizona, ARIZONA_NOTIFY_VOICE_TRIGGER, - (void *)i); + &info); + } } if (!serviced) { diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index f1b3294..3b8db83 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2233,6 +2233,7 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data) { struct wm5110_priv *priv = data; struct arizona *arizona = priv->core.arizona; + struct arizona_voice_trigger_info info; int serviced = 0; int i, ret; @@ -2240,10 +2241,12 @@ static irqreturn_t wm5110_adsp2_irq(int irq, void *data) ret = wm_adsp_compr_handle_irq(&priv->core.adsp[i]); if (ret != -ENODEV) serviced++; - if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) + if (ret == WM_ADSP_COMPR_VOICE_TRIGGER) { + info.core = i; arizona_call_notifiers(arizona, ARIZONA_NOTIFY_VOICE_TRIGGER, - (void *)i); + &info); + } } if (!serviced) { -- 2.1.4

3 2

[alsa-devel] [PATCH v2] ASoC: ak4642: Implement suspend callback
by Peter Ujfalusi 31 May '16

31 May '16

Add the suspend callback to accompany the existing resume operation. With the suspend/resume callbacks the regmap (regcache) state handling can follow the recommended sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Changes since v1: - take into use the suspend callback (set the suspend in soc_codec_dev_ak4642) Regards, Peter sound/soc/codecs/ak4642.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4d8b9e49e8d6..cc941d66ec3d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -523,15 +523,23 @@ static struct snd_soc_dai_driver ak4642_dai = { .symmetric_rates = 1, }; -static int ak4642_resume(struct snd_soc_codec *codec) +static int ak4642_suspend(struct snd_soc_codec *codec) { struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + regcache_cache_only(regmap, true); regcache_mark_dirty(regmap); - regcache_sync(regmap); return 0; } +static int ak4642_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_cache_only(regmap, false); + regcache_sync(regmap); + return 0; +} static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec); @@ -544,6 +552,7 @@ static int ak4642_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4642 = { .probe = ak4642_probe, + .suspend = ak4642_suspend, .resume = ak4642_resume, .set_bias_level = ak4642_set_bias_level, .controls = ak4642_snd_controls, -- 2.8.3

2 1

[alsa-devel] [PATCH resend v5 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 31 May '16

31 May '16

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is rebased on top of v4.6-rc1 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v5 Changes from v4: - Fix power suspend function and add power resume function - Remove clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Removed surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fixed print format of type size_t and pointer. Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1513 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

5 18

[alsa-devel] [PATCH v6 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 31 May '16

31 May '16

Hi, This patchset contains audio support for Broadcom's Cygnus SoC. It contains DT bindings and core audio driver. The audio driver supports both capture and playback of Audio PCM samples over I2S/TDM interface and provides playback support over SPDIF interface. This patchset is derived from a previously submitted patchset: http://lkml.iu.edu/hypermail/linux/kernel/1503.3/05434.html This patchset has been tested on Cygnus wireless audio bcm958305K board. It is based on v4.6-rc1 and is available from github: repo: https://github.com/Broadcom/cygnus-linux/tree/cygnus-sound-v6 Changes from v5: - Fix code style, e.g. change "if" statements to "switch" statements - Fix SPDIF output enable register field - Set BUFFER_PAIR_ENABLE for both mono, stereo and TDM modes - Reflect PCM bit formats for SPDIF, SSP and TDM as supported by hardware Changes from v4: - Fix power suspend function and add power resume function - Move clock initialization code from audio driver to clock framework Changes from v3: - Fix the subject lines to match the style for the subsystem Changes from v2: - Split patchset 2/2 from v2 into patchsets 2/3 and 3/3. - Remove SND_SOC_CYGNUS_DIAG. Diagnostics can be performed using standard kernel trace infrastructure. - Fix interrupt handler. Acknowledge only those interrupts that are handledby ISR. - Modify configure_vco() and the pll_macro_entry() struct to make it better readable. The functionality did not change. - Remove casts on macros - Remove surround sound channel grouping from the driver. Changes from v1: - Address code review comments. Fix print format of type size_t and pointer. Simran Rai (3): ASoC: cygnus: Add DT bindings for Broadcom Cygnus audio ASoC: cygnus: Add Cygnus audio DAI driver ASoC: cygnus: Add Cygnus audio DMA driver .../bindings/sound/brcm,cygnus-audio.txt | 67 + sound/soc/bcm/Kconfig | 9 + sound/soc/bcm/Makefile | 5 + sound/soc/bcm/cygnus-pcm.c | 861 +++++++++++ sound/soc/bcm/cygnus-ssp.c | 1529 ++++++++++++++++++++ sound/soc/bcm/cygnus-ssp.h | 139 ++ 6 files changed, 2610 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt create mode 100644 sound/soc/bcm/cygnus-pcm.c create mode 100644 sound/soc/bcm/cygnus-ssp.c create mode 100644 sound/soc/bcm/cygnus-ssp.h -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ASoC: add api to retrieve dmic array info from coreboot nhlt
by Yong Zhi 31 May '16

31 May '16

skylake can be configured with either both 2 and 4 channel DMIC array, or 2 channel DMIC array only, this patch provides an API to retrieve the DMIC info from nhlt. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/skylake/skl-nhlt.c | 37 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-nhlt.h | 22 ++++++++++++++++++++++ sound/soc/intel/skylake/skl.c | 12 ++++++++++-- sound/soc/intel/skylake/skl.h | 6 ++++++ 4 files changed, 75 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 7d73648..be4c4a3 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -18,6 +18,7 @@ * */ #include "skl.h" +#include <linux/pci.h> /* Unique identification for getting NHLT blobs */ static u8 OSC_UUID[16] = {0x6E, 0x88, 0x9F, 0xA6, 0xEB, 0x6C, 0x94, 0x45, @@ -149,6 +150,42 @@ struct nhlt_specific_cfg return NULL; } +int skl_get_dmic_geo(struct skl *skl) +{ + struct nhlt_acpi_table *nhlt = (struct nhlt_acpi_table *)skl->nhlt; + struct nhlt_endpoint *epnt; + struct nhlt_dmic_array_config *cfg; + struct device *dev = &(skl->pci->dev); + unsigned int dmic_geo = 0; + u8 j; + + epnt = (struct nhlt_endpoint *)nhlt->desc; + + for (j = 0; j < nhlt->endpoint_count; j++) { + if (epnt->linktype == NHLT_LINK_DMIC) { + cfg = (struct nhlt_dmic_array_config *) + (epnt->config.caps); + switch (cfg->array_type) { + case NHLT_MIC_ARRAY_2CH_SMALL: + case NHLT_MIC_ARRAY_2CH_BIG: + dmic_geo |= MIC_ARRAY_2CH; + break; + case NHLT_MIC_ARRAY_4CH_1ST_GEOM: + case NHLT_MIC_ARRAY_4CH_L_SHAPED: + case NHLT_MIC_ARRAY_4CH_2ND_GEOM: + dmic_geo |= MIC_ARRAY_4CH; + break; + default: + dev_warn(dev, "undefined DMIC array_type 0x%0x\n", + cfg->array_type); + } + } + epnt = (struct nhlt_endpoint *)((u8 *)epnt + epnt->length); + } + + return dmic_geo; +} + static void skl_nhlt_trim_space(struct skl *skl) { char *s = skl->tplg_name; diff --git a/sound/soc/intel/skylake/skl-nhlt.h b/sound/soc/intel/skylake/skl-nhlt.h index 3769f9f..116534e7 100644 --- a/sound/soc/intel/skylake/skl-nhlt.h +++ b/sound/soc/intel/skylake/skl-nhlt.h @@ -103,4 +103,26 @@ struct nhlt_resource_desc { u64 length; } __packed; +#define MIC_ARRAY_2CH 2 +#define MIC_ARRAY_4CH 4 + +struct nhlt_tdm_config { + u8 virtual_slot; + u8 config_type; +} __packed; + +struct nhlt_dmic_array_config { + struct nhlt_tdm_config tdm_config; + u8 array_type; +} __packed; + +enum { + NHLT_MIC_ARRAY_2CH_SMALL = 0xa, + NHLT_MIC_ARRAY_2CH_BIG = 0xb, + NHLT_MIC_ARRAY_4CH_1ST_GEOM = 0xc, + NHLT_MIC_ARRAY_4CH_L_SHAPED = 0xd, + NHLT_MIC_ARRAY_4CH_2ND_GEOM = 0xe, + NHLT_MIC_ARRAY_VENDOR_DEFINED = 0xf, +}; + #endif diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index 06d8c26..b0f7226 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -35,6 +35,8 @@ #include "skl-sst-dsp.h" #include "skl-sst-ipc.h" +static struct skl_machine_pdata skl_dmic_data; + /* * initialize the PCI registers */ @@ -397,6 +399,10 @@ static int skl_machine_device_register(struct skl *skl, void *driver_data) platform_device_put(pdev); return -EIO; } + + if (mach->pdata) + dev_set_drvdata(&pdev->dev, mach->pdata); + skl->i2s_dev = pdev; return 0; @@ -666,6 +672,8 @@ static int skl_probe(struct pci_dev *pci, pci_set_drvdata(skl->pci, ebus); + skl_dmic_data.dmic_num = skl_get_dmic_geo(skl); + /* check if dsp is there */ if (ebus->ppcap) { err = skl_machine_device_register(skl, @@ -787,9 +795,9 @@ static void skl_remove(struct pci_dev *pci) static struct sst_acpi_mach sst_skl_devdata[] = { { "INT343A", "skl_alc286s_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, { "INT343B", "skl_nau88l25_ssm4567_i2s", "intel/dsp_fw_release.bin", - NULL, NULL, NULL }, + NULL, NULL, &skl_dmic_data }, { "MX98357A", "skl_nau88l25_max98357a_i2s", "intel/dsp_fw_release.bin", - NULL, NULL, NULL }, + NULL, NULL, &skl_dmic_data }, {} }; diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 4b4b387..f66be17 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -90,6 +90,11 @@ struct skl_dma_params { u8 stream_tag; }; +/* to pass dmic data */ +struct skl_machine_pdata { + u32 dmic_num; +}; + struct skl_dsp_ops { int id; struct skl_dsp_loader_ops (*loader_ops)(void); @@ -108,6 +113,7 @@ void skl_nhlt_free(struct nhlt_acpi_table *addr); struct nhlt_specific_cfg *skl_get_ep_blob(struct skl *skl, u32 instance, u8 link_type, u8 s_fmt, u8 no_ch, u32 s_rate, u8 dirn); +int skl_get_dmic_geo(struct skl *skl); int skl_nhlt_update_topology_bin(struct skl *skl); int skl_init_dsp(struct skl *skl); int skl_free_dsp(struct skl *skl); -- 1.9.1

4 22

[alsa-devel] [PATCH] ASoC: nau8825: add programmable biquad filter control
by John Hsu 31 May '16

31 May '16

Add programmable biquad filter configuration control for user space. The filter is configurable for low pass filters, high pass filters, Notch filter, etc in the ADC and DAC path. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 53 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 8 +++++++ 2 files changed, 61 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index e988f89..88e01f9 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -217,6 +217,7 @@ static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) case NAU8825_REG_SARDOUT_RAM_STATUS: case NAU8825_REG_CHARGE_PUMP_INPUT_READ: case NAU8825_REG_GENERAL_STATUS: + case NAU8825_REG_BIQ_CTRL ... NAU8825_REG_BIQ_COF10: return true; default: return false; @@ -293,6 +294,54 @@ static int nau8825_output_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int nau8825_biq_coeff_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + + if (!component->regmap) + return -EINVAL; + + regmap_raw_read(component->regmap, NAU8825_REG_BIQ_COF1, + ucontrol->value.bytes.data, params->max); + return 0; +} + +static int nau8825_biq_coeff_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + void *data; + + if (!component->regmap) + return -EINVAL; + + data = kmemdup(ucontrol->value.bytes.data, + params->max, GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; + + regmap_update_bits(component->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, 0); + regmap_raw_write(component->regmap, NAU8825_REG_BIQ_COF1, + data, params->max); + regmap_update_bits(component->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, NAU8825_BIQ_WRT_EN); + + kfree(data); + return 0; +} + +static const char * const nau8825_biq_path[] = { + "ADC", "DAC" +}; + +static const struct soc_enum nau8825_biq_path_enum = + SOC_ENUM_SINGLE(NAU8825_REG_BIQ_CTRL, NAU8825_BIQ_PATH_SFT, + ARRAY_SIZE(nau8825_biq_path), nau8825_biq_path); + static const char * const nau8825_adc_decimation[] = { "32", "64", "128", "256" }; @@ -329,6 +378,10 @@ static const struct snd_kcontrol_new nau8825_controls[] = { SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), + /* programmable biquad filter */ + SOC_ENUM("BIQ Path Select", nau8825_biq_path_enum), + SND_SOC_BYTES_EXT("BIQ Coefficeints", 20, + nau8825_biq_coeff_get, nau8825_biq_coeff_put), }; /* DAC Mux 0x33[9] and 0x34[9] */ diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1293d1b 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -231,6 +231,14 @@ #define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) #define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) +/* BIQ_CTRL (0x20) */ +#define NAU8825_BIQ_WRT_SFT 4 +#define NAU8825_BIQ_WRT_EN (1 << NAU8825_BIQ_WRT_SFT) +#define NAU8825_BIQ_PATH_SFT 0 +#define NAU8825_BIQ_PATH_MASK (1 << NAU8825_BIQ_PATH_SFT) +#define NAU8825_BIQ_PATH_ADC (0 << NAU8825_BIQ_PATH_SFT) +#define NAU8825_BIQ_PATH_DAC (1 << NAU8825_BIQ_PATH_SFT) + /* ADC_RATE (0x2b) */ #define NAU8825_ADC_SYNC_DOWN_SFT 0 #define NAU8825_ADC_SYNC_DOWN_MASK 0x3 -- 2.6.4

2 2

[alsa-devel] [PATCH] ASoC: tas571x: add input channel mixer for TAS5717/19
by Petr Kulhavy 31 May '16

31 May '16

Add channel 1 and 2 input mixer registers and the related ALSA mixer controls for TAS5717/19 chips. The mixer control coefficients on the chip are linear in the range -3.99999 to +3.99999, encoded in 3.23 number format. In this patch the mixer controls are limited to 128 values from 0.0 to 1.0 in 1/64 steps. Signed-off-by: Petr Kulhavy <brain(a)jikos.cz> --- sound/soc/codecs/tas571x.c | 19 +++++++++++++++++++ sound/soc/codecs/tas571x.h | 5 +++++ 2 files changed, 24 insertions(+) diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c index 39307ad..bca0e84 100644 --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -57,6 +57,11 @@ static int tas571x_register_size(struct tas571x_private *priv, unsigned int reg) case TAS571X_CH1_VOL_REG: case TAS571X_CH2_VOL_REG: return priv->chip->vol_reg_size; + case TAS5717_CH1_RIGHT_CH_MIX_REG: + case TAS5717_CH1_LEFT_CH_MIX_REG: + case TAS5717_CH2_LEFT_CH_MIX_REG: + case TAS5717_CH2_RIGHT_CH_MIX_REG: + return 4; default: return 1; } @@ -293,6 +298,16 @@ static const struct snd_kcontrol_new tas5717_controls[] = { TAS571X_SOFT_MUTE_REG, TAS571X_SOFT_MUTE_CH1_SHIFT, TAS571X_SOFT_MUTE_CH2_SHIFT, 1, 1), + + SOC_DOUBLE_R_RANGE("CH1 Mixer Volume", + TAS5717_CH1_LEFT_CH_MIX_REG, + TAS5717_CH1_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), + + SOC_DOUBLE_R_RANGE("CH2 Mixer Volume", + TAS5717_CH2_LEFT_CH_MIX_REG, + TAS5717_CH2_RIGHT_CH_MIX_REG, + 16, 0, 0x80, 0), }; static const struct reg_default tas5717_reg_defaults[] = { @@ -303,6 +318,10 @@ static const struct reg_default tas5717_reg_defaults[] = { { 0x08, 0x00c0 }, { 0x09, 0x00c0 }, { 0x1b, 0x82 }, + { TAS5717_CH1_RIGHT_CH_MIX_REG, 0x0 }, + { TAS5717_CH1_LEFT_CH_MIX_REG, 0x800000}, + { TAS5717_CH2_LEFT_CH_MIX_REG, 0x0 }, + { TAS5717_CH2_RIGHT_CH_MIX_REG, 0x800000}, }; static const struct regmap_config tas5717_regmap_config = { diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index 0aee471..b4175d6 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -30,4 +30,9 @@ #define TAS571X_OSC_TRIM_REG 0x1b +#define TAS5717_CH1_RIGHT_CH_MIX_REG 0x72 +#define TAS5717_CH1_LEFT_CH_MIX_REG 0x73 +#define TAS5717_CH2_LEFT_CH_MIX_REG 0x76 +#define TAS5717_CH2_RIGHT_CH_MIX_REG 0x77 + #endif /* _TAS571X_H */ -- 2.5.0

2 1

[alsa-devel] [PATCH] UAPI: Add three missing header files to Kbuild file
by Robert P. J. Day 31 May '16

31 May '16

include/uapi/sound/Kbuild was missing the inclusion of three header files in that directory. Signed-off-by: Robert P. J. Day <rpjday(a)crashcourse.ca> --- diff --git a/include/uapi/sound/Kbuild b/include/uapi/sound/Kbuild index a7f2770..691984c 100644 --- a/include/uapi/sound/Kbuild +++ b/include/uapi/sound/Kbuild @@ -1,5 +1,6 @@ # UAPI Header export list header-y += asequencer.h +header-y += asoc.h header-y += asound.h header-y += asound_fm.h header-y += compress_offload.h @@ -10,3 +11,5 @@ header-y += hdsp.h header-y += hdspm.h header-y += sb16_csp.h header-y += sfnt_info.h +header-y += tlv.h +header-y += usb_stream.h -- ======================================================================== Robert P. J. Day Ottawa, Ontario, CANADA http://crashcourse.ca Twitter: http://twitter.com/rpjday LinkedIn: http://ca.linkedin.com/in/rpjday ========================================================================

2 1

[alsa-devel] any reason for include/uapi/sound header files not listed in Kbuild?
by Robert P. J. Day 31 May '16

31 May '16

i just noticed that, in include/uapi/sound/, there are three header files: * asoc.h * tlv.h * usb_stream.h that are not referenced by the corresponding Kbuild file. is there any reason for this? or, in general, is there *any* reason for there to be header files under include/uapi/ that are not listed in their Kbuild files? thanks. rday -- ======================================================================== Robert P. J. Day Ottawa, Ontario, CANADA http://crashcourse.ca Twitter: http://twitter.com/rpjday LinkedIn: http://ca.linkedin.com/in/rpjday ========================================================================

2 3

[alsa-devel] Baytrail audio port problem
by alan Paris 31 May '16

31 May '16

Hi everybody, well I send this message because I've got a problem with Baytrail audio port (rt5640). I followed the discussion about writing an alsa driver (see it : http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/094114.html) and I tried many solutions (as like that : https://groups.google.com/forum/#!topic/android-x86/GKBFxCZyxQg ). Now the system see the Baytrail card audio for speakers but as Michele Curti when I start the system I've got a minute freeze and I haven't got still sound. When I launch dmesg command, I see only those messages : Baytrail Audio Port: sst: error code = -22 bytt100_rt5640 bytt100_rt5640: BUG: , pos = -22, buffer size = 3840, period size = 480 Do you know how I can solve this problem and it is possible to operate baytrail audio device with alsa or pulseaudio ? Thanks. Regards. Alan.

2 1

[alsa-devel] ALSA timestamping for kernel SoC drivers
by Alan Young 31 May '16

31 May '16

Hello, I need to get accurate timestamping of the kind currently available for HDA into some Freescale (iMX) SoC drivers. Is there anyone already working in this area or is there anyone who would be prepared to discuss the design and implementation details? Alan.

1 0

[alsa-devel] ALSA timestamping for kernel SoC drivers
by Alan Young 31 May '16

31 May '16

Hello, I need to get accurate timestamping of the kind currently available for HDA into some Freescale (iMX) SoC drivers. Is there anyone already working in this area or is there anyone who would be prepared to discuss the design and implementation details? Alan.

1 0

Re: [alsa-devel] next-20160531 build: 0 failures 3 warnings (next-20160531)
by Arnd Bergmann 31 May '16

31 May '16

On Tuesday, May 31, 2016 7:24:43 AM CEST Build bot for Mark Brown wrote: > ------------------------------------------------------------------------------- > arm64-allmodconfig : PASS, 0 errors, 3 warnings, 0 section mismatches > > Warnings: > ../sound/soc/codecs/cs47l24.c:1091:13: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] > ../sound/soc/codecs/wm5110.c:2246:13: warning: cast to pointer from integer of different size [-Wint-to-pointer-cast] These got introduced by 7baa7e2490e1 ("ASoC: arizona: Add event notification on voice trigger events"). Arnd

1 0

[alsa-devel] [PATCH] ASoC: ak4642: Implement suspend callback
by Peter Ujfalusi 31 May '16

31 May '16

Add the suspend callback to accompany the existing resume operation. With the suspend/resume callbacks the regmap (regcache) state handling can follow the recommended sequence. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/ak4642.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4d8b9e49e8d6..ca9e82568b3c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -523,15 +523,23 @@ static struct snd_soc_dai_driver ak4642_dai = { .symmetric_rates = 1, }; -static int ak4642_resume(struct snd_soc_codec *codec) +static int ak4642_suspend(struct snd_soc_codec *codec) { struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + regcache_cache_only(regmap, true); regcache_mark_dirty(regmap); - regcache_sync(regmap); return 0; } +static int ak4642_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_cache_only(regmap, false); + regcache_sync(regmap); + return 0; +} static int ak4642_probe(struct snd_soc_codec *codec) { struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec); -- 2.8.3

1 1

[alsa-devel] [RFC v2 0/3] ASoC: Add support to Qualcomm msm8916-wcd codec
by Srinivas Kandagatla 31 May '16

31 May '16

This patchset aims at adding msm8916-wcd codec support. msm8916-wcd codec is found in Qualcomm msm8916 and apq8016 processors. This codec IP is split in to two parts(Digital & Analog), Analog part is integrated in to PMIC PM8916 and the digital part is integrated into Application processor. Register access to the analog part is done via SPMI interface to PMIC, and registers on the Application processor are memory mapped. Data transfer between Analog and Digital Die is done via a internal bus called PDM. This series also add a new macro SOC_SINGLE_S8_TLV() which is used in the codec driver to control gain on a 8 bit signed gain control register. Most of this driver is ported from downstream Andriod v3.10 kernel. Jack support and Headset button detection code is not added in to this series. This codec support: - 3 Microphones: Primary Mic(Handset mic), Headset Mic and Secondary Mic. - 2 Digital Microphones. - 2 Mic Bias Circuits. - Earpiece - Headset - Loud Speaker. - Jack Detect. - Headset Button detection circuit. Tested this code on v4.6 with lpass driver Headset (Mic and speakers), Secondary Mic and Loudspeaker on DB410C. DMIC should work too. Thanks, srini Changes since RFC v1 (https://lkml.org/lkml/2016/2/16/710) - Lot of code cleanup as suggest by Mark Brown. - Fixed bindings as suggested by Mark Rutland. - merged all the patches to one as spliting the patches did not help review. Srinivas Kandagatla (3): ASoC: msm8916: Add codec Device Tree bindings. ASoC: Introduce SOC_SINGLE_S8_TLV() macro ASoC: msm8916: Add msm8916-wcd codec driver .../devicetree/bindings/sound/qcom,msm8916-wcd.txt | 101 ++ include/dt-bindings/sound/msm8916-wcd.h | 7 + include/sound/soc.h | 11 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 3 +- sound/soc/codecs/msm8916-wcd-registers.h | 710 +++++++++ sound/soc/codecs/msm8916-wcd.c | 1582 ++++++++++++++++++++ sound/soc/codecs/msm8916-wcd.h | 308 ++++ 8 files changed, 2725 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd.txt create mode 100644 include/dt-bindings/sound/msm8916-wcd.h create mode 100644 sound/soc/codecs/msm8916-wcd-registers.h create mode 100644 sound/soc/codecs/msm8916-wcd.c create mode 100644 sound/soc/codecs/msm8916-wcd.h -- 2.8.2

3 8

Re: [alsa-devel] [PATCH 1/2] ASoC: Intel: Add Broxton-P Dialog Maxim machine driver
by Vinod Koul 31 May '16

31 May '16

On Mon, May 30, 2016 at 10:10:38PM -0700, sathyanarayana.nujella(a)intel.com wrote: > From: Sathyanarayana Nujella <sathyanarayana.nujella(a)intel.com> > > This patch adds Broxton-P I2S machine driver which uses > DA7219 and MAX98357A codecs. Both, Acked-by: Vinod Koul <vinod.koul(a)intel.com> Thanks -- ~Vinod

1 0

[alsa-devel] [PATCH v2] alsabat: add buffer size and period size settings
by vivian.zhang＠intel.com 31 May '16

31 May '16

From: "vivian,zhang" <vivian.zhang(a)intel.com> Add buffer size and period size settings in alsabat. With -E and -B options, alsabat performs the test with specified buffer size and period size Signed-off-by: Zhang Vivian <vivian.zhang(a)intel.com> --- Changes in v2: - Use a micro for min and max values of buffer size and period size - Use a micro for default period size in tinyalsa.c bat/alsa.c | 95 +++++++++++++++++++++++++++++++++++++++----------------- bat/bat.c | 18 +++++++++-- bat/common.h | 8 +++++ bat/tinyalsa.c | 5 ++- 4 files changed, 94 insertions(+), 32 deletions(-) diff --git a/bat/alsa.c b/bat/alsa.c index 75158cb..0a37714 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -74,6 +74,8 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) snd_pcm_format_t format; unsigned int buffer_time = 0; unsigned int period_time = 0; + snd_pcm_uframes_t buffer_size = 0; + snd_pcm_uframes_t period_size = 0; unsigned int rate; int err; const char *device_name = snd_pcm_name(sndpcm->handle); @@ -145,39 +147,71 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) return -EINVAL; } - if (snd_pcm_hw_params_get_buffer_time_max(params, - &buffer_time, 0) < 0) { - fprintf(bat->err, _("Get parameter from device error: ")); - fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), - buffer_time, - device_name, snd_strerror(err), err); - return -EINVAL; - } + if (bat->buffer_size > 0 && bat->period_size == 0) + bat->period_size = bat->buffer_size / DIV_BUFFERTIME; - if (buffer_time > MAX_BUFFERTIME) - buffer_time = MAX_BUFFERTIME; + if (bat->buffer_size > 0) { + buffer_size = bat->buffer_size; + period_size = bat->period_size; - period_time = buffer_time / DIV_BUFFERTIME; + fprintf(bat->log, _("Set period size: %d buffer size: %d\n"), + (int) period_size, (int) buffer_size); - /* Set buffer time and period time */ - err = snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, params, - &buffer_time, 0); - if (err < 0) { - fprintf(bat->err, _("Set parameter to device error: ")); - fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), - buffer_time, - device_name, snd_strerror(err), err); - return err; - } + err = snd_pcm_hw_params_set_buffer_size_near(sndpcm->handle, + params, &buffer_size); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("buffer size: %d %s: %s(%d)\n"), + (int) buffer_size, + device_name, snd_strerror(err), err); + return err; + } - err = snd_pcm_hw_params_set_period_time_near(sndpcm->handle, params, - &period_time, 0); - if (err < 0) { - fprintf(bat->err, _("Set parameter to device error: ")); - fprintf(bat->err, _("period time: %d %s: %s(%d)\n"), - period_time, - device_name, snd_strerror(err), err); - return err; + err = snd_pcm_hw_params_set_period_size_near(sndpcm->handle, + params, &period_size, 0); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("period size: %d %s: %s(%d)\n"), + (int) period_size, + device_name, snd_strerror(err), err); + return err; + } + } else { + if (snd_pcm_hw_params_get_buffer_time_max(params, + &buffer_time, 0) < 0) { + fprintf(bat->err, + _("Get parameter from device error: ")); + fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), + buffer_time, + device_name, snd_strerror(err), err); + return -EINVAL; + } + + if (buffer_time > MAX_BUFFERTIME) + buffer_time = MAX_BUFFERTIME; + + period_time = buffer_time / DIV_BUFFERTIME; + + /* Set buffer time and period time */ + err = snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, + params, &buffer_time, 0); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), + buffer_time, + device_name, snd_strerror(err), err); + return err; + } + + err = snd_pcm_hw_params_set_period_time_near(sndpcm->handle, + params, &period_time, 0); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("period time: %d %s: %s(%d)\n"), + period_time, + device_name, snd_strerror(err), err); + return err; + } } /* Write the parameters to the driver */ @@ -214,6 +248,9 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) return -EINVAL; } + fprintf(bat->log, _("Get period size: %d buffer size: %d\n"), + (int) sndpcm->period_size, (int) sndpcm->buffer_size); + err = snd_pcm_format_physical_width(format); if (err < 0) { fprintf(bat->err, _("Invalid parameters: ")); diff --git a/bat/bat.c b/bat/bat.c index 1afdcb4..cd4ea2d 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -292,6 +292,8 @@ _("Usage: alsabat [-options]...\n" " -k parameter for frequency detecting threshold\n" " -F target frequency\n" " -p total number of periods to play/capture\n" +" -B buffer size in frames\n" +" -E period size in frames\n" " --log=# file that both stdout and strerr redirecting to\n" " --file=# file for playback\n" " --saveplay=# file that storing playback content, for debug\n" @@ -324,6 +326,8 @@ static void set_defaults(struct bat *bat) bat->capture.device = NULL; bat->buf = NULL; bat->local = false; + bat->buffer_size = 0; + bat->period_size = 0; #ifdef HAVE_LIBTINYALSA bat->channels = 2; bat->playback.fct = &playback_tinyalsa; @@ -342,8 +346,8 @@ static void set_defaults(struct bat *bat) static void parse_arguments(struct bat *bat, int argc, char *argv[]) { - int c, option_index; - static const char short_options[] = "D:P:C:f:n:F:c:r:s:k:p:lth"; + int c, option_index, err; + static const char short_options[] = "D:P:C:f:n:F:c:r:s:k:p:B:E:lth"; static const struct option long_options[] = { {"help", 0, 0, 'h'}, {"log", 1, 0, OPT_LOG}, @@ -414,6 +418,16 @@ static void parse_arguments(struct bat *bat, int argc, char *argv[]) bat->periods_total = atoi(optarg); bat->period_is_limited = true; break; + case 'B': + err = atoi(optarg); + bat->buffer_size = err >= MIN_BUFFERSIZE + && err < MAX_BUFFERSIZE ? err : 0; + break; + case 'E': + err = atoi(optarg); + bat->period_size = err >= MIN_PERIODSIZE + && err < MAX_PERIODSIZE ? err : 0; + break; case 'h': default: usage(bat); diff --git a/bat/common.h b/bat/common.h index b789af5..ad02a5a 100644 --- a/bat/common.h +++ b/bat/common.h @@ -46,6 +46,12 @@ #define DIV_BUFFERTIME 8 /* margin to avoid sign inversion when generate sine wav */ #define RANGE_FACTOR 0.95 +#define MAX_BUFFERSIZE 200000 +#define MIN_BUFFERSIZE 32 +#define MAX_PERIODSIZE 200000 +#define MIN_PERIODSIZE 32 +/* default period size for tinyalsa */ +#define TINYALSA_PERIODSIZE 1024 #define EBATBASE 1000 #define ENOPEAK (EBATBASE + 1) @@ -152,6 +158,8 @@ struct bat { int frame_size; /* size of frame */ int sample_size; /* size of sample */ enum _bat_pcm_format format; /* PCM format */ + int buffer_size; /* buffer size in frames */ + int period_size; /* period size in frames */ float sigma_k; /* threshold for peak detection */ float target_freq[MAX_CHANNELS]; diff --git a/bat/tinyalsa.c b/bat/tinyalsa.c index ea5f848..a19dd90 100644 --- a/bat/tinyalsa.c +++ b/bat/tinyalsa.c @@ -57,7 +57,10 @@ static int init_config(struct bat *bat, struct pcm_config *config) { config->channels = bat->channels; config->rate = bat->rate; - config->period_size = 1024; + if (bat->period_size > 0) + config->period_size = bat->period_size; + else + config->period_size = TINYALSA_PERIODSIZE; config->period_count = 4; config->start_threshold = 0; config->stop_threshold = 0; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: wm8731: Add 32 bit functionality
by Matt Flax 31 May '16

31 May '16

This patch adds 32 bit functionality to the wm8731 codec driver. Enabling this functionality for the Audio Injector sound card. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/wm8731.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 4bcf5f8..a9fc3bc 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -5,6 +5,7 @@ * Copyright 2006-12 Wolfson Microelectronics, plc * * Author: Richard Purdie <richard(a)openedhand.com> + * Author: Matt Flax <flatmax(a)flatmax.com> added 32bit functionality. * * Based on wm8753.c by Liam Girdwood * @@ -358,6 +359,9 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, case 24: iface |= 0x0008; break; + case 32: + iface |= 0x000c; + break; } wm8731_set_deemph(codec); @@ -541,7 +545,7 @@ static int wm8731_startup(struct snd_pcm_substream *substream, #define WM8731_RATES SNDRV_PCM_RATE_8000_96000 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ - SNDRV_PCM_FMTBIT_S24_LE) + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops wm8731_dai_ops = { .startup = wm8731_startup, -- 2.5.0

1 0

[alsa-devel] ALC256 speaker noise issue
by Kailang 31 May '16

31 May '16

Hi Takashi, ALC256 had speaker noise issue. Attach patch will solve it. This issue is cause from LDO output voltage control. This patch is updated the right LDO register value. Many Thanks. Kailang

2 5

[alsa-devel] [PATCH 0/7] ASoC: Intel: Skylake: Manifest parsing support
by Vinod Koul 30 May '16

30 May '16

The firmware file contains the extended manifest which needs to be stripped before downloading of firmware. The firmware manifest also contains manifest which has module information, Driver needs to parse and create module information and use it for sending IPCs. So this series adds common API for stripping manifest and then strips it for SKL and BXT. Also it adds core support to parse and add module information and its use in SKL and BXT. Changes in this series: - modified manifest parse to be generic utility code, added calling patches from SKL n BXTN - fix issues in last rvw - add extended manifest parsing in this series - make manifest series common and use list instead of table and bxtn modules will have manifest parsing and addition later Ramesh Babu (1): ASoC: Intel: Skylake: Add strip extended manifest utility Shreyas NC (1): ASoC: Intel: Skylake: Add DSP firmware manifest parsing Vinod Koul (5): ASoC: Intel: Skylake: Don't use local pointer for firmware ASoC: Intel: Skylake: Strip manifest for Skylake platform ASoC: Intel: Skylake: Strip manifest for Broxton platform ASoC: Intel: Skylake: Find uuids for Skylake ASoC: Intel: Skylake: Find uuids for Broxton sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/bxt-sst.c | 26 +++- sound/soc/intel/skylake/skl-sst-dsp.h | 8 + sound/soc/intel/skylake/skl-sst-ipc.h | 3 + sound/soc/intel/skylake/skl-sst-utils.c | 256 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst.c | 23 ++- sound/soc/intel/skylake/skl-topology.c | 4 + 7 files changed, 315 insertions(+), 7 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-sst-utils.c -- 1.9.1

2 14

[alsa-devel] [PATCH 1/2 v2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 30 May '16

30 May '16

The driver changes jack type detection interruption to non-clock archi- tecture for less 1mW power saving. The architecture is called manual mode jack detection. It has no hardware debounce, no jack type detection, but only detecting jack insertion. After jack insertion, the driver will switch to auto mode jack detection with internal clock which can detect microphone, jack type and do hardware debounce. The manual architecture has these main changes including codec initiation, interruption, clock control, and power management. When codec initiation or system resume, the clock is closed as jack insertion detection at man- ual mode, and bypass debounce circuit. These configurations move to resume setup function when setup bias level after resume. When jack insertion detection happens, the manual mode turns off and make configuration about jack type detection interruption at auto mode in auto irq setup function which can detect microphone and jack type. The inter- ruption will switch to manual mode again with clock free until jack ejec- tion happens. The system clock configuration adds clock disable option which can disable internal VCO clock. Before the system clock change, there is an restric- tion added to make sure clock disabled and not config any clock when no headset connected. In power management, we involve the solution about races and jack detec- tion in resume from Ben Zhang in the following patch and list his comment. [PATCH] ASoC: nau8825: Fix jack detection across suspend "Jack plug status is rechecked at resume to handle plug/unplug in S3 when the chip has no power." "Suspend/resume callbacks are moved from the i2c dev_pm_ops to snd_soc_codec_driver. soc_resume_deferred is a delayed work which may trigger nau8825_set_bias_level. The bias change races against dev_pm_ops, causing jack detection issues. soc_resume_deferred ensures bias change and snd_soc_codec_driver suspend/resume are sequenced correctly." Change SAR widget to supply type which can prevent the codec keeping at SND_SOC_BIAS_ON during suspend. The codec suspend function can just invoke normally. Before the system suspends, the driver turns off all interruptions. Keep the interruption quiet before resume setup completes. The ADC channel will be disabled which is needed for interruptions at audo mode. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> Signed-off-by: Ben Zhang <benzh(a)chromium.org> --- sound/soc/codecs/nau8825.c | 295 ++++++++++++++++++++++++++++++++------------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 222 insertions(+), 82 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..e738234 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -30,10 +30,16 @@ #include "nau8825.h" + +#define NUVOTON_CODEC_DAI "nau8825-hifi" + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 124000000 #define NAU_FVCO_MIN 90000000 +static int nau8825_configure_sysclk(struct nau8825 *nau8825, + int clk_id, unsigned int freq); + struct nau8825_fll { int mclk_src; int ratio; @@ -368,9 +374,12 @@ static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, 0), - /* ADC for button press detection */ - SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, - NAU8825_SAR_ADC_EN_SFT, 0), + /* ADC for button press detection. A dapm supply widget is used to + * prevent dapm_power_widgets keeping the codec at SND_SOC_BIAS_ON + * during suspend. + */ + SND_SOC_DAPM_SUPPLY("SAR", NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACL", 2, NAU8825_REG_RDAC, 12, 0, NULL, 0), SND_SOC_DAPM_PGA_S("ADACR", 2, NAU8825_REG_RDAC, 13, 0, NULL, 0), @@ -614,9 +623,6 @@ int nau8825_enable_jack_detect(struct snd_soc_codec *codec, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); - return 0; } EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); @@ -642,6 +648,22 @@ static void nau8825_restart_jack_detection(struct regmap *regmap) NAU8825_JACK_DET_RESTART, 0); } +static void nau8825_int_status_clear_all(struct regmap *regmap) +{ + int active_irq, clear_irq, i; + + /* Reset the intrruption status from rightmost bit if the corres- + * ponding irq event occurs. + */ + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + for (i = 0; i < NAU8825_REG_DATA_LEN; i++) { + clear_irq = (0x1 << i); + if (active_irq & clear_irq) + regmap_write(regmap, + NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + } +} + static void nau8825_eject_jack(struct nau8825 *nau8825) { struct snd_soc_dapm_context *dapm = nau8825->dapm; @@ -656,6 +678,69 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); snd_soc_dapm_sync(dapm); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable the insertion interruption, disable the ejection inter- + * ruption, and then bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_EJECT_DIS | NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_EJECT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_EJECT_EN | + NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); + + /* Close clock for jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); +} + +/* Enable audo mode interruptions with internal clock. */ +static void nau8825_setup_auto_irq(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Enable headset jack type detection complete interruption and + * jack ejection interruption. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable ADC needed for interruptions */ + regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); + + /* Chip needs one FSCLK cycle in order to generate interruptions, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + /* Not bypass de-bounce circuit */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, 0); + + /* Unmask all interruptions */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + + /* Restart the jack detection process at auto mode */ + nau8825_restart_jack_detection(regmap); } static int nau8825_button_decode(int value) @@ -789,6 +874,26 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) event_mask |= SND_JACK_HEADSET; clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } else if ((active_irq & NAU8825_JACK_INSERTION_IRQ_MASK) == + NAU8825_JACK_INSERTION_DETECTED) { + /* One more step to check GPIO status directly. Thus, the + * driver can confirm the real insertion interruption because + * the intrruption at manual mode has bypassed debounce + * circuit which can get rid of unstable status. + */ + if (nau8825_is_jack_inserted(regmap)) { + /* Turn off insertion interruption at manual mode */ + regmap_update_bits(regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, + NAU8825_IRQ_INSERT_DIS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_INSERT_EN, NAU8825_IRQ_INSERT_EN); + /* Enable interruption for jack type detection at audo + * mode which can detect microphone and jack type. + */ + nau8825_setup_auto_irq(nau8825); + } } if (!clear_irq) @@ -938,8 +1043,8 @@ static void nau8825_init_regs(struct nau8825 *nau8825) } static const struct regmap_config nau8825_regmap_config = { - .val_bits = 16, - .reg_bits = 16, + .val_bits = NAU8825_REG_DATA_LEN, + .reg_bits = NAU8825_REG_ADDR_LEN, .max_register = NAU8825_REG_MAX, .readable_reg = nau8825_readable_reg, @@ -958,12 +1063,6 @@ static int nau8825_codec_probe(struct snd_soc_codec *codec) nau8825->dapm = dapm; - /* Unmask interruptions. Handler uses dapm object so we can enable - * interruptions only after dapm is fully initialized. - */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); - return 0; } @@ -1128,6 +1227,14 @@ static int nau8825_mclk_prepare(struct nau8825 *nau8825, unsigned int freq) return 0; } +static void nau8825_configure_mclk_as_sysclk(struct regmap *regmap) +{ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, 0); +} + static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, unsigned int freq) { @@ -1135,10 +1242,17 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, int ret; switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + nau8825_configure_mclk_as_sysclk(regmap); + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; case NAU8825_CLK_MCLK: - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); - regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + nau8825_configure_mclk_as_sysclk(regmap); /* MCLK not changed by clock tree */ regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_MCLK_SRC_MASK, 0); @@ -1148,17 +1262,25 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, break; case NAU8825_CLK_INTERNAL: - regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, - NAU8825_DCO_EN); - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); - /* Decrease the VCO frequency for power saving */ - regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, - NAU8825_CLK_MCLK_SRC_MASK, 0xf); - regmap_update_bits(regmap, NAU8825_REG_FLL1, - NAU8825_FLL_RATIO_MASK, 0x10); - regmap_update_bits(regmap, NAU8825_REG_FLL6, - NAU8825_SDM_EN, NAU8825_SDM_EN); + if (nau8825_is_jack_inserted(nau8825->regmap)) { + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_DCO_EN, NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + /* Decrease the VCO frequency for power saving */ + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_MCLK_SRC_MASK, 0xf); + regmap_update_bits(regmap, NAU8825_REG_FLL1, + NAU8825_FLL_RATIO_MASK, 0x10); + regmap_update_bits(regmap, NAU8825_REG_FLL6, + NAU8825_SDM_EN, NAU8825_SDM_EN); + } else { + /* The clock turns off intentionally for power saving + * when no headset connected. + */ + nau8825_configure_mclk_as_sysclk(regmap); + dev_warn(nau8825->dev, "Disable clock for power saving when no headset connected\n"); + } if (nau8825->mclk_freq) { clk_disable_unprepare(nau8825->mclk); nau8825->mclk_freq = 0; @@ -1209,6 +1331,31 @@ static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, return nau8825_configure_sysclk(nau8825, clk_id, freq); } +static int nau8825_resume_setup(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Close clock when jack type detection at manual mode */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_DIS, 0); + + /* Clear all interruption status */ + nau8825_int_status_clear_all(regmap); + + /* Enable both insertion and ejection interruptions, and then + * bypass de-bounce circuit. + */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_HEADSET_COMPLETE_EN | + NAU8825_IRQ_EJECT_EN | NAU8825_IRQ_INSERT_EN, + NAU8825_IRQ_OUTPUT_EN | NAU8825_IRQ_HEADSET_COMPLETE_EN); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_DB_BYPASS, NAU8825_JACK_DET_DB_BYPASS); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS | NAU8825_IRQ_EJECT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1238,11 +1385,22 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, "Failed to sync cache: %d\n", ret); return ret; } + + /* Setup codec configuration after resume */ + nau8825_resume_setup(nau8825); } break; case SND_SOC_BIAS_OFF: + /* Turn off all interruptions before system shutdown. Keep the + * interruption quiet before resume setup completes. + */ + regmap_write(nau8825->regmap, + NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff); + /* Disable ADC needed for interruptions at audo mode */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_ADC, 0); if (nau8825->mclk_freq) clk_disable_unprepare(nau8825->mclk); @@ -1252,12 +1410,40 @@ static int nau8825_set_bias_level(struct snd_soc_codec *codec, return 0; } +#ifdef CONFIG_PM_SLEEP +static int nau8825_codec_suspend(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + disable_irq(nau8825->irq); + snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(nau8825->regmap, true); + + return 0; +} + +static int nau8825_codec_resume(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(nau8825->regmap, false); + enable_irq(nau8825->irq); + + return 0; +} +#else +#define nau8825_codec_suspend NULL +#define nau8825_codec_resume NULL +#endif + static struct snd_soc_codec_driver nau8825_codec_driver = { .probe = nau8825_codec_probe, .set_sysclk = nau8825_set_sysclk, .set_pll = nau8825_set_pll, .set_bias_level = nau8825_set_bias_level, .suspend_bias_off = true, + .suspend = nau8825_codec_suspend, + .resume = nau8825_codec_resume, .controls = nau8825_controls, .num_controls = ARRAY_SIZE(nau8825_controls), @@ -1351,30 +1537,8 @@ static int nau8825_read_device_properties(struct device *dev, static int nau8825_setup_irq(struct nau8825 *nau8825) { - struct regmap *regmap = nau8825->regmap; int ret; - /* IRQ Output Enable */ - regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, - NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); - - /* Enable internal VCO needed for interruptions */ - nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); - - /* Enable ADC needed for interrupts */ - regmap_update_bits(regmap, NAU8825_REG_ENA_CTRL, - NAU8825_ENABLE_ADC, NAU8825_ENABLE_ADC); - - /* Chip needs one FSCLK cycle in order to generate interrupts, - * as we cannot guarantee one will be provided by the system. Turning - * master mode on then off enables us to generate that FSCLK cycle - * with a minimum of contention on the clock bus. - */ - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); - regmap_update_bits(regmap, NAU8825_REG_I2S_PCM_CTRL2, - NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); - ret = devm_request_threaded_irq(nau8825->dev, nau8825->irq, NULL, nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, "nau8825", nau8825); @@ -1442,36 +1606,6 @@ static int nau8825_i2c_remove(struct i2c_client *client) return 0; } -#ifdef CONFIG_PM_SLEEP -static int nau8825_suspend(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - disable_irq(client->irq); - regcache_cache_only(nau8825->regmap, true); - regcache_mark_dirty(nau8825->regmap); - - return 0; -} - -static int nau8825_resume(struct device *dev) -{ - struct i2c_client *client = to_i2c_client(dev); - struct nau8825 *nau8825 = dev_get_drvdata(dev); - - regcache_cache_only(nau8825->regmap, false); - regcache_sync(nau8825->regmap); - enable_irq(client->irq); - - return 0; -} -#endif - -static const struct dev_pm_ops nau8825_pm = { - SET_SYSTEM_SLEEP_PM_OPS(nau8825_suspend, nau8825_resume) -}; - static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } @@ -1498,7 +1632,6 @@ static struct i2c_driver nau8825_driver = { .name = "nau8825", .of_match_table = of_match_ptr(nau8825_of_ids), .acpi_match_table = ACPI_PTR(nau8825_acpi_match), - .pm = &nau8825_pm, }, .probe = nau8825_i2c_probe, .remove = nau8825_i2c_remove, diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..1100385 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -93,6 +93,9 @@ #define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 #define NAU8825_REG_GENERAL_STATUS 0x82 #define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS +/* 16-bit control register address, and 16-bits control register data */ +#define NAU8825_REG_ADDR_LEN 16 +#define NAU8825_REG_DATA_LEN 16 /* ENA_CTRL (0x1) */ #define NAU8825_ENABLE_DACR_SFT 10 @@ -145,6 +148,7 @@ /* JACK_DET_CTRL (0xd) */ #define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_DET_DB_BYPASS (1 << 8) #define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 #define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) #define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 @@ -157,6 +161,7 @@ #define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) #define NAU8825_IRQ_EJECT_EN (1 << 2) +#define NAU8825_IRQ_INSERT_EN (1 << 0) /* IRQ_STATUS (0x10) */ #define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) @@ -177,6 +182,7 @@ #define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) #define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) #define NAU8825_IRQ_EJECT_DIS (1 << 2) +#define NAU8825_IRQ_INSERT_DIS (1 << 0) /* SAR_CTRL (0x13) */ #define NAU8825_SAR_ADC_EN_SFT 12 @@ -333,7 +339,8 @@ /* System Clock Source */ enum { - NAU8825_CLK_MCLK = 0, + NAU8825_CLK_DIS = 0, + NAU8825_CLK_MCLK, NAU8825_CLK_INTERNAL, NAU8825_CLK_FLL_MCLK, NAU8825_CLK_FLL_BLK, -- 2.6.4

2 13

[alsa-devel] [PATCH] alsabat: add buffer size and period size setup support
by vivian.zhang＠intel.com 30 May '16

30 May '16

From: "vivian,zhang" <vivian.zhang(a)intel.com> Signed-off-by: Zhang Vivian <vivian.zhang(a)intel.com> --- bat/alsa.c | 95 +++++++++++++++++++++++++++++++++++++++----------------- bat/bat.c | 16 ++++++++-- bat/common.h | 2 ++ bat/tinyalsa.c | 5 ++- 4 files changed, 86 insertions(+), 32 deletions(-) diff --git a/bat/alsa.c b/bat/alsa.c index 75158cb..167dbad 100644 --- a/bat/alsa.c +++ b/bat/alsa.c @@ -74,6 +74,8 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) snd_pcm_format_t format; unsigned int buffer_time = 0; unsigned int period_time = 0; + snd_pcm_uframes_t buffer_size = 0; + snd_pcm_uframes_t period_size = 0; unsigned int rate; int err; const char *device_name = snd_pcm_name(sndpcm->handle); @@ -145,39 +147,71 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) return -EINVAL; } - if (snd_pcm_hw_params_get_buffer_time_max(params, - &buffer_time, 0) < 0) { - fprintf(bat->err, _("Get parameter from device error: ")); - fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), - buffer_time, - device_name, snd_strerror(err), err); - return -EINVAL; - } + if (bat->buffer_size > 0) { + if (bat->period_size == 0) + bat->period_size = bat->buffer_size / DIV_BUFFERTIME; - if (buffer_time > MAX_BUFFERTIME) - buffer_time = MAX_BUFFERTIME; + buffer_size = bat->buffer_size; + period_size = bat->period_size; - period_time = buffer_time / DIV_BUFFERTIME; + fprintf(bat->log, _("Set period size: %d buffer size: %d\n"), + (int) period_size, (int) buffer_size); - /* Set buffer time and period time */ - err = snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, params, - &buffer_time, 0); - if (err < 0) { - fprintf(bat->err, _("Set parameter to device error: ")); - fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), - buffer_time, - device_name, snd_strerror(err), err); - return err; - } + err = snd_pcm_hw_params_set_buffer_size_near(sndpcm->handle, + params, &buffer_size); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("buffer size: %d %s: %s(%d)\n"), + (int) buffer_size, + device_name, snd_strerror(err), err); + return err; + } - err = snd_pcm_hw_params_set_period_time_near(sndpcm->handle, params, - &period_time, 0); - if (err < 0) { - fprintf(bat->err, _("Set parameter to device error: ")); - fprintf(bat->err, _("period time: %d %s: %s(%d)\n"), - period_time, - device_name, snd_strerror(err), err); - return err; + err = snd_pcm_hw_params_set_period_size_near(sndpcm->handle, + params, &period_size, 0); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("period size: %d %s: %s(%d)\n"), + (int) period_size, + device_name, snd_strerror(err), err); + return err; + } + } else { + if (snd_pcm_hw_params_get_buffer_time_max(params, + &buffer_time, 0) < 0) { + fprintf(bat->err, + _("Get parameter from device error: ")); + fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), + buffer_time, + device_name, snd_strerror(err), err); + return -EINVAL; + } + + if (buffer_time > MAX_BUFFERTIME) + buffer_time = MAX_BUFFERTIME; + + period_time = buffer_time / DIV_BUFFERTIME; + + /* Set buffer time and period time */ + err = snd_pcm_hw_params_set_buffer_time_near(sndpcm->handle, + params, &buffer_time, 0); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("buffer time: %d %s: %s(%d)\n"), + buffer_time, + device_name, snd_strerror(err), err); + return err; + } + + err = snd_pcm_hw_params_set_period_time_near(sndpcm->handle, + params, &period_time, 0); + if (err < 0) { + fprintf(bat->err, _("Set parameter to device error: ")); + fprintf(bat->err, _("period time: %d %s: %s(%d)\n"), + period_time, + device_name, snd_strerror(err), err); + return err; + } } /* Write the parameters to the driver */ @@ -214,6 +248,9 @@ static int set_snd_pcm_params(struct bat *bat, struct pcm_container *sndpcm) return -EINVAL; } + fprintf(bat->log, _("Get period size: %d buffer size: %d\n"), + (int) sndpcm->period_size, (int) sndpcm->buffer_size); + err = snd_pcm_format_physical_width(format); if (err < 0) { fprintf(bat->err, _("Invalid parameters: ")); diff --git a/bat/bat.c b/bat/bat.c index 1afdcb4..0d5aaa5 100644 --- a/bat/bat.c +++ b/bat/bat.c @@ -292,6 +292,8 @@ _("Usage: alsabat [-options]...\n" " -k parameter for frequency detecting threshold\n" " -F target frequency\n" " -p total number of periods to play/capture\n" +" -B buffer size in frames\n" +" -E period size in frames\n" " --log=# file that both stdout and strerr redirecting to\n" " --file=# file for playback\n" " --saveplay=# file that storing playback content, for debug\n" @@ -324,6 +326,8 @@ static void set_defaults(struct bat *bat) bat->capture.device = NULL; bat->buf = NULL; bat->local = false; + bat->buffer_size = 0; + bat->period_size = 0; #ifdef HAVE_LIBTINYALSA bat->channels = 2; bat->playback.fct = &playback_tinyalsa; @@ -342,8 +346,8 @@ static void set_defaults(struct bat *bat) static void parse_arguments(struct bat *bat, int argc, char *argv[]) { - int c, option_index; - static const char short_options[] = "D:P:C:f:n:F:c:r:s:k:p:lth"; + int c, option_index, err; + static const char short_options[] = "D:P:C:f:n:F:c:r:s:k:p:B:E:lth"; static const struct option long_options[] = { {"help", 0, 0, 'h'}, {"log", 1, 0, OPT_LOG}, @@ -414,6 +418,14 @@ static void parse_arguments(struct bat *bat, int argc, char *argv[]) bat->periods_total = atoi(optarg); bat->period_is_limited = true; break; + case 'B': + err = atoi(optarg); + bat->buffer_size = err >= 32 && err < 200000 ? err : 0; + break; + case 'E': + err = atoi(optarg); + bat->period_size = err >= 32 && err < 200000 ? err : 0; + break; case 'h': default: usage(bat); diff --git a/bat/common.h b/bat/common.h index b789af5..52c8cb1 100644 --- a/bat/common.h +++ b/bat/common.h @@ -152,6 +152,8 @@ struct bat { int frame_size; /* size of frame */ int sample_size; /* size of sample */ enum _bat_pcm_format format; /* PCM format */ + int buffer_size; /* buffer size in frames */ + int period_size; /* period size in frames */ float sigma_k; /* threshold for peak detection */ float target_freq[MAX_CHANNELS]; diff --git a/bat/tinyalsa.c b/bat/tinyalsa.c index ea5f848..faa1ad0 100644 --- a/bat/tinyalsa.c +++ b/bat/tinyalsa.c @@ -57,7 +57,10 @@ static int init_config(struct bat *bat, struct pcm_config *config) { config->channels = bat->channels; config->rate = bat->rate; - config->period_size = 1024; + if (bat->period_size > 0) + config->period_size = bat->period_size; + else + config->period_size = 1024; config->period_count = 4; config->start_threshold = 0; config->stop_threshold = 0; -- 1.7.9.5

2 1

[alsa-devel] Custom card returns "arecord: main:722: audio open error: Invalid argument"
by Rob Nertney 29 May '16

29 May '16

I've build a card that is registered correctly as far as I can tell. My hardware is: static struct snd_pcm_hardware my_pcm_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_U8, .rates = SNDRV_PCM_RATE_8000, .rate_min = 8000, .rate_max = 8000, .channels_min = 1, .channels_max = 1, .buffer_bytes_max = DMA_LENGTH, .period_bytes_min = 48, .period_bytes_max = 48, .periods_min = 1, .periods_max = 16, }; When I run "arecord -d 2 foo.wav", I get the error message in the subject. Printks show that I get to my_pcm_open, then immediately jump to my_hw_free, my_pcm_close. I'm in an FPGA embedded system. there are no other cards but mine via arecord -L. Here's my _init function for the module: struct snd_card *card; struct aaw_device *my_dev; struct snd_pcm *pcm; struct device *dev; int ret; printk(KERN_ERR "INIT start\n"); snd_card_new(dev,SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, THIS_MODULE, sizeof(struct my_device), &card); my_dev = card->private_data; my_dev->card = card; sprintf(card->driver, "my_driver"); sprintf(card->shortname, "My First Audio Driver"); snd_card_set_dev(card,dev); ret = snd_device_new(card, SNDRV_DEV_LOWLEVEL, my_dev, &dev_ops); if(ret < 0) printk(KERN_ERR "new device err\n"); ret = snd_pcm_new(card, card->driver, 0, 0, 1, &pcm); if(ret < 0) printk(KERN_ERR "new pcm err\n"); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &my_pcm_ops); pcm->private_data = my_dev; pcm->info_flags = 0; strcpy(pcm->name, card->shortname); ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), DMA_LENGTH, DMA_LENGTH); if(ret < 0) printk(KERN_ERR "preallocate page err\n"); ret = snd_card_register(card); if (ret < 0) printk(KERN_ERR "error in register snd\n"); Here's my _open function: static int my_pcm_open(struct snd_pcm_substream *ss) { struct my_device *my_dev = ss->private_data; printk(KERN_ERR "in pcm_open\n"); ss->runtime->hw = my_pcm_hw; my_dev->substream = ss; ss->runtime->private_data = my_dev; return 0; } Any help would be most appreciated. Thanks, Rob

2 1

[alsa-devel] [PATCH] pcm: Fix secondary retry in dsnoop and dshare
by Takashi Iwai 28 May '16

28 May '16

The commit [fdba9e1bad8f: pcm: Fallback open as the first instance for dmix & co] introduced a mechanism to retry the open of slave PCM for the secondary streams, but this also introduced a regression in dsnoop and dshare plugins: since the retry goto-tag was placed at a wrong position, it retries to re-fetch the shm unnecessarily and eventually leads to the fatal error. The bug can be easily reproduced by starting arecord and killing it via SIGKILL, then starting arecord again. The second arecord fails. The fix is obviously to move the wrong retry goto-tags to the right positions. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_dshare.c | 2 +- src/pcm/pcm_dsnoop.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index adb3587a2869..05854dedf259 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -690,7 +690,6 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, break; } - retry: first_instance = ret = snd_pcm_direct_shm_create_or_connect(dshare); if (ret < 0) { SNDERR("unable to create IPC shm instance"); @@ -705,6 +704,7 @@ int snd_pcm_dshare_open(snd_pcm_t **pcmp, const char *name, dshare->max_periods = opts->max_periods; dshare->sync_ptr = snd_pcm_dshare_sync_ptr; + retry: if (first_instance) { /* recursion is already checked in snd_pcm_direct_get_slave_ipc_offset() */ diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 8ff0ba57cb14..2d45171dda01 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -583,7 +583,6 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, break; } - retry: first_instance = ret = snd_pcm_direct_shm_create_or_connect(dsnoop); if (ret < 0) { SNDERR("unable to create IPC shm instance"); @@ -598,6 +597,7 @@ int snd_pcm_dsnoop_open(snd_pcm_t **pcmp, const char *name, dsnoop->max_periods = opts->max_periods; dsnoop->sync_ptr = snd_pcm_dsnoop_sync_ptr; + retry: if (first_instance) { /* recursion is already checked in snd_pcm_direct_get_slave_ipc_offset() */ -- 2.8.3

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[alsa-devel] [PATCH 0/5] Add support for trigger only voice recognition
by Charles Keepax 27 May '16

27 May '16

In some systems our voice trigger firmwares will be used only to provide an event that indicates the trigger phrase was spoken, this patch chain adds support for this use-case. There are two parts to adding this support firstly, a mechanism is required to get the DSP to power on without the compressed stream being opened, this is done by means of a virtual output for the voice trigger in the DAPM graph. Secondly, the trigger event needs to be handled. Typically this will be used to generate a key press event, but there are numerous options what could be done here, and vendors often have their own rather strong opinions. Rather than dictate the usage at the driver level we we simply provide a notifier that can be caught in the machine driver and appropriate action can be taken there. Thanks, Charles Charles Keepax (5): ASoC: arizona: Tie SYSCLK to DRC signal activity widgets ASoC: arizona: Add voice trigger output widget ASoC: wm_adsp: Specifically propagate voice trigger event to caller ASoC: arizona: Add a notifier chain for CODEC events ASoC: arizona: Add event notification on voice trigger events include/linux/mfd/arizona/core.h | 10 ++++++++++ sound/soc/codecs/arizona.c | 43 ++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 15 ++++++++++++++ sound/soc/codecs/cs47l24.c | 16 +++++++++++++++ sound/soc/codecs/wm5102.c | 1 + sound/soc/codecs/wm5110.c | 16 +++++++++++++++ sound/soc/codecs/wm8998.c | 1 + sound/soc/codecs/wm_adsp.c | 5 +++++ sound/soc/codecs/wm_adsp.h | 4 ++++ 9 files changed, 111 insertions(+) -- 2.1.4

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[alsa-devel] [GIT PULL] ASoC updates for v4.7
by Mark Brown 27 May '16

27 May '16

The following changes since commit 515511a7920c69aebf7f5fef0cb8e1df6767f34c: Merge remote-tracking branch 'asoc/topic/hdmi' into asoc-next (2016-05-13 14:27:16 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.7-2 for you to fetch changes up to bf65921380cd50b87618df550398e06581f4a361: Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next (2016-05-27 13:46:02 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.7 part 2 Really sorry about this late pull request. It looks like at the time I sent my pull request for v4.7 there was some conflict or other issue which caused my script to stop merging the ASoC branches at some point after the HDMI changes. It's all specific driver updates, including: - New drivers for MAX98371 and TAS5720. - SPI support for TLV320AIC32x4. - TDM support for STI Uniperf IPs. This code should all have been in -next prior to the merge window apart from some fixes, it dropped out on the 18th. ---------------------------------------------------------------- Akinobu Mita (1): spi: omap2-mcspi: Undo broken fix for dma transfer of vmalloced buffer Al Viro (6): do_splice_to(): cap the size before passing to ->splice_read() fix the copy vs. map logics in blk_rq_map_user_iov() atomic_open(): fix the handling of create_error ecryptfs: fix handling of directory opening get_rock_ridge_filename(): handle malformed NM entries Merge branch 'ovl-fixes' into for-linus Alex Deucher (2): drm/radeon: fix DP mode validation drm/amdgpu: fix DP mode validation Alexander Shishkin (2): perf/x86/intel/pt: Generate PMI in the STOP region as well perf/core: Disable the event on a truncated AUX record Andrea Adami (1): ASoC: pxa: Fix module autoload for platform drivers Andrea Arcangeli (1): mm: thp: calculate the mapcount correctly for THP pages during WP faults Andreas Dannenberg (2): ASoC: add TA5720 digital amplifier DT bindings ASoC: add support for TAS5720 digital amplifier Andrey Ryabinin (1): perf/x86: Fix undefined shift on 32-bit kernels Andrey Utkin (1): kvmconfig: add more virtio drivers Andrey Vostrikov (1): spi: spi-fsl-dspi: Fix cs_change handling in message transfer Andy Lutomirski (1): perf/core: Change the default paranoia level to 2 Arindam Nath (1): drm/radeon: fix DP link training issue with second 4K monitor Arnaldo Carvalho de Melo (9): perf tools: Use readdir() instead of deprecated readdir_r() perf script: Use readdir() instead of deprecated readdir_r() perf thread_map: Use readdir() instead of deprecated readdir_r() perf tools: Use readdir() instead of deprecated readdir_r() perf dwarf: Guard !x86_64 definitions under #ifdef else clause perf probe: Check if dwarf_getlocations() is available perf evsel: Improve EPERM error handling in open_strerror() perf evsel: Handle EACCESS + perf_event_paranoid=2 in fallback() perf stat: Fallback to user only counters when perf_event_paranoid > 1 Arnd Bergmann (6): regmap: fix documentation to match code Revert "net/mlx5: Kconfig: Fix MLX5_EN/VXLAN build issue" net/mlx5e: make VXLAN support conditional Merge tag 'at91-fixes' of git://git.kernel.org/.../nferre/linux-at91 into fixes Merge tag 'at91-fixes2' of git://git.kernel.org/.../nferre/linux-at91 into fixes net: mvneta: bm: fix dependencies again Axel Lin (2): ASoC: rt5677: Avoid duplicate the same test in each switch case ASoC: max98371 Remove duplicate entry in max98371_reg Bard Liao (2): ASoC: rt298: reset AD dilter is there is no MCLK ASoC: rt298: fix capture doesn't work at some cases Bastien Nocera (1): ASoC: tlv320aix31xx: Add ACPI match for Lenovo 100S Ben Hutchings (2): spi: spi-ti-qspi: Fix FLEN and WLEN settings if bits_per_word is overridden spi: spi-ti-qspi: Handle truncated frames properly Boris Brezillon (1): ARM: dts: at91: sam9x5: Fix the memory range assigned to the PMC Chris Diamand (1): Input: byd - update copyright header Colin Ian King (1): tools: bpf_jit_disasm: check for klogctl failure Dan Carpenter (6): scsi_dh_alua: uninitialized variable in alua_rtpg() netxen: fix error handling in netxen_get_flash_block() netxen: reversed condition in netxen_nic_set_link_parameters() netxen: netxen_rom_fast_read() doesn't return -1 qede: uninitialized variable in qede_start_xmit() qlcnic: potential NULL dereference in qlcnic_83xx_get_minidump_template() Dan Williams (1): libnvdimm, pfn: fix ARCH=alpha allmodconfig build failure Daniel Jurgens (1): net/mlx4_en: Fix endianness bug in IPV6 csum calculation Daniel Vetter (1): drm/i915: Bail out of pipe config compute loop on LPT Dave Airlie (3): Merge tag 'drm-intel-fixes-2016-05-11' of git://anongit.freedesktop.org/drm-intel into drm-fixes Merge branch 'drm-fixes-4.6' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge branch 'drm-fixes-4.6' of git://people.freedesktop.org/~agd5f/linux into drm-fixes David Ahern (1): net: ipv6: tcp reset, icmp need to consider L3 domain David Howells (1): KEYS: Fix ASN.1 indefinite length object parsing David S. Miller (9): Merge branch 'master' of git://git.kernel.org/.../klassert/ipsec Merge branch 'bnxt_en-fixes' Merge branch 'mlx5-build-fix' Merge git://git.kernel.org/.../pablo/nf Merge tag 'wireless-drivers-for-davem-2016-05-09' of git://git.kernel.org/.../kvalo/wireless-drivers Merge branch 'nps_enet-fixes' Merge branch 'net-sched-fixes' Merge branch 'bnxt_en-fixes' Merge branch 'xgene-fixes' Dmitry V. Levin (1): x86: Use compat version for preadv2 and pwritev2 Elad Kanfi (2): net: nps_enet: Tx handler synchronization net: nps_enet: bug fix - handle lost tx interrupts Emmanuel Grumbach (1): iwlwifi: mvm: don't override the rate with the AMSDU len Eric Dumazet (2): macvtap: segmented packet is consumed tcp: refresh skb timestamp at retransmit time Fabio Estevam (6): ASoC: wm8960: Provide a menu selection text ASoC: wm8962: Disable clock if wm8962_runtime_resume() fails ASoC: wm8962: Fit error message into a single line ASoC: wm8962: Adjust clk definitions so that simple card can work ASoC: hdac_hdmi: Remove the unused 'timeout' variable spi: spi-ep93xx: Fix the PTR_ERR() argument Felipe Balbi (1): cgroup: fix compile warning Florian Westphal (1): netfilter: conntrack: init all_locks to avoid debug warning Geert Uytterhoeven (2): ravb: Add missing free_irq() call to ravb_close() regulator: da9063: Correct module alias prefix to fix module autoloading Guneshwor Singh (1): ASoC: topology: Set CPU DAI name and enable DPCM by default for FE link H. Nikolaus Schaller (1): Input: twl6040-vibra - fix DT node memory management Hans de Goede (2): regulator: axp20x: Fix axp22x ldo_io voltage ranges regulator: axp20x: Fix axp22x ldo_io registration error on cold boot Herbert Xu (2): crypto: hash - Fix page length clamping in hash walk crypto: testmgr - Use kmalloc memory for RSA input Ian Campbell (1): VSOCK: do not disconnect socket when peer has shutdown SEND only Ido Schimmel (2): mlxsw: spectrum: Fix rollback order in LAG join failure mlxsw: spectrum: Add missing rollback in flood configuration Imre Deak (1): drm/i915/bdw: Add missing delay during L3 SQC credit programming Ingo Molnar (3): MAINTAINERS: Add mmiotrace entry Revert "sched/fair: Fix fairness issue on migration" Merge tag 'perf-urgent-for-mingo-20160512' of git://git.kernel.org/.../acme/linux into perf/urgent Iyappan Subramanian (5): drivers: net: xgene: fix IPv4 forward crash drivers: net: xgene: fix sharing of irqs drivers: net: xgene: fix ununiform latency across queues drivers: net: xgene: fix statistics counters race condition drivers: net: xgene: fix register offset Jack Pham (1): regmap: spmi: Fix regmap_spmi_ext_read in multi-byte case Jamal Hadi Salim (7): export tc ife uapi header net sched: vlan action fix late binding net sched: ipt action fix late binding net sched: mirred action fix late binding net sched: simple action fix late binding net sched: skbedit action fix late binding net sched: ife action fix late binding Jani Nikula (1): drm/i915/lvds: separate border enable readout from panel fitter Jarkko Nikula (1): spi: pxa2xx: Do not detect number of enabled chip selects on Intel SPT Jarno Rajahalme (2): udp_tunnel: Remove redundant udp_tunnel_gro_complete(). udp_offload: Set encapsulation before inner completes. Jeeja KP (1): ASoC: topology: Fix memory leak in widget creation Jeremy McDermond (7): ASoC: tlv320aic32x4: Change name of probe function ASoC: tlv320aic32x4: Break out I2C support into separate module ASoC: tlv320aic32x4: Add SPI support ASoC: tlv320aic32x4: Add 96k sample rate ASoC: tlv320aic32x4: Implement resistors on input pins ASoC: tlv320aic32x4: Add additional input pins ASoC: tlv320aic32x4: Properly implement the positive and negative pins into the mixers Jim Lodes (1): ASoC: omap-pcm: Initialize DMA configuration Jiri Benc (1): gre: do not keep the GRE header around in collect medata mode Joe Stringer (1): openvswitch: Fix cached ct with helper. Johannes Thumshirn (1): qla1280: Don't allocate 512kb of host tags Josh Poimboeuf (2): compiler-gcc: require gcc 4.8 for powerpc __builtin_bswap16() x86/kvm: Add stack frame dependency to fastop() inline asm Junxiao Bi (2): ocfs2: revert using ocfs2_acl_chmod to avoid inode cluster lock hang ocfs2: fix posix_acl_create deadlock Kaho Ng (1): ALSA: hda - Fix white noise on Asus UX501VW headset Kalle Valo (1): Merge tag 'iwlwifi-for-kalle-2016-05-04' of https://git.kernel.org/.../iwlwifi/iwlwifi-fixes Kangjie Lu (3): net: fix infoleak in llc net: fix infoleak in rtnetlink net: fix a kernel infoleak in x25 module Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver Krzysztof Kozlowski (1): regulator: s2mps11: Fix invalid selector mask and voltages for buck9 Kuninori Morimoto (3): ASoC: tidyup alphabetical order for SND_SOC_Bxx ASoC: rsnd: count .probe/.remove for rsnd_mod_call() ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS Linus Lüssing (1): bridge: fix igmp / mld query parsing Linus Torvalds (31): Merge git://git.kernel.org/.../davem/net Merge branch 'linus' of git://git.kernel.org/.../herbert/crypto-2.6 Merge branch 'akpm' (patches from Andrew) Merge branch 'libnvdimm-fixes' of git://git.kernel.org/.../nvdimm/nvdimm Merge branch 'sched-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge tag 'pci-v4.6-fixes-3' of git://git.kernel.org/.../helgaas/pci Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge tag 'spi-fix-v4.6-rc7' of git://git.kernel.org/.../broonie/spi Merge git://git.kernel.org/.../davem/net Merge tag 'scsi-fixes' of git://git.kernel.org/.../jejb/scsi Merge tag 'pinctrl-v4.6-4' of git://git.kernel.org/.../linusw/linux-pinctrl Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge tag 'sound-4.6' of git://git.kernel.org/.../tiwai/sound Merge tag 'keys-fixes-20160512' of git://git.kernel.org/.../dhowells/linux-fs Merge branch 'akpm' (patches from Andrew) Merge branch 'linus' of git://git.kernel.org/.../herbert/crypto-2.6 Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'media/v4.6-6' of git://git.kernel.org/.../mchehab/linux-media Merge tag 'regmap-fix-v4.6-rc7' of git://git.kernel.org/.../broonie/regmap Merge tag 'regulator-fix-v4.6-rc7' of git://git.kernel.org/.../broonie/regulator Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'sched-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'for-4.6-fixes' of git://git.kernel.org/.../tj/wq Merge branch 'for-4.6-fixes' of git://git.kernel.org/.../tj/cgroup Merge branch 'for-linus' of git://git.kernel.org/.../viro/vfs nf_conntrack: avoid kernel pointer value leak in slab name Merge git://git.kernel.org/.../davem/net Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Linux 4.6 Liping Zhang (1): netfilter: IDLETIMER: fix race condition when destroy the target Lucas Stach (1): drm/radeon: fix PLL sharing on DCE6.1 (v2) Ludovic Desroches (1): pinctrl: at91-pio4: fix pull-up/down logic Lukas Wunner (2): PCI: Fix BUG on device attach failure PCI: Do not treat EPROBE_DEFER as device attach failure Lyude (1): Revert "drm/i915: start adding dp mst audio" Marek Szyprowski (1): Input: max8997-haptic - fix NULL pointer dereference Mark Brown (31): regmap: mmio: Fix value endianness selection regmap: Fix implicit inclusion of device.h regmap: mmio: Parse endianness definitions from DT ASoC: wm8960: Depends on I2C regmap: mmio: Explicitly say little endian is the defualt in the bus config Merge branches 'topic/dai-link' and 'topic/find-dai' of git://git.kernel.org/.../broonie/sound into asoc-topology Merge remote-tracking branches 'spi/fix/fsl-dspi', 'spi/fix/omap2-mcspi', 'spi/fix/pxa2xx' and 'spi/fix/ti-qspi' into spi-linus Merge remote-tracking branch 'regmap/fix/mmio' into regmap-linus Merge remote-tracking branches 'regmap/fix/be', 'regmap/fix/doc' and 'regmap/fix/spmi' into regmap-linus Merge remote-tracking branches 'regulator/fix/axp20x', 'regulator/fix/da9063', 'regulator/fix/gpio' and 'regulator/fix/s2mps11' into regulator-linus ASoC: ak4642: Enable cache usage to fix crashes on resume Merge tag 'asoc-v4.7' into asoc-linus Merge remote-tracking branches 'asoc/fix/ak4642', 'asoc/fix/ep93xx', 'asoc/fix/kirkwood' and 'asoc/fix/twl6040' into asoc-linus Merge remote-tracking branch 'asoc/topic/tlv320aic32x4' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/max98371' into asoc-next Merge remote-tracking branch 'asoc/topic/twl6040' into asoc-next Merge remote-tracking branch 'asoc/topic/simple' into asoc-next Merge remote-tracking branch 'asoc/topic/pxa' into asoc-next Merge remote-tracking branch 'asoc/topic/topology' into asoc-next Merge remote-tracking branch 'asoc/topic/tas5270' into asoc-next Merge remote-tracking branch 'asoc/topic/omap' into asoc-next Merge remote-tracking branch 'asoc/topic/mtk' into asoc-next Merge remote-tracking branch 'asoc/topic/rt298' into asoc-next Merge remote-tracking branch 'asoc/topic/kconfig' into asoc-next Merge remote-tracking branch 'asoc/topic/tlv320aic31xx' into asoc-next Merge remote-tracking branch 'asoc/topic/sti' into asoc-next Merge remote-tracking branch 'asoc/topic/qcom' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8960' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5677' into asoc-next Merge remote-tracking branch 'asoc/topic/wm8962' into asoc-next Mathias Krause (1): x86/extable: ensure entries are swapped completely when sorting Matthias Brugger (1): drivers: net: xgene: Fix error handling Mauro Carvalho Chehab (1): Revert "[media] videobuf2-v4l2: Verify planes array in buffer dequeueing" Maxime Ripard (1): regulator: axp20x: Fix LDO4 linear voltage range Mengdong Lin (4): ASoC: topology: ABI - Define types for vendor tuples ASoC: topology: Set the link ID when creating a FE DAI link ASoC: topology: Check failure to create a widget ASoC: topology: Check size mismatch of ABI objects before parsing Michael Chan (4): bnxt_en: Need memory barrier when processing the completion ring. bnxt_en: Setup multicast properly after resetting device. bnxt_en: Add workaround to detect bad opaque in rx completion (part 1) bnxt_en: Add workaround to detect bad opaque in rx completion (part 2) Mihai Mihalache (1): regulator: gpio: check return value of of_get_named_gpio Mikko Rapeli (1): uapi glibc compat: fix compile errors when glibc net/if.h included before linux/if.h Miklos Szeredi (4): vfs: add vfs_select_inode() helper vfs: rename: check backing inode being equal vfs: add lookup_hash() helper ovl: ignore permissions on underlying lookup Moise Gergaud (10): ASoC: sti: correct typo errors ASoC: sti: macro for uniperif tdm regs access ASoC: sti: rename unip player type into common player & reader type ASoC: sti: define tdm type & default tdm hw config ASoC: sti: helper functions for unip tdm slots configuration ASoC: sti: helper functions to fix tdm runtime params ASoC: sti: unip player tdm mode ASoC: sti: unip reader tdm mode ASoC: sti: select player for I2S/TDM TX bus ASoC: sti-asoc-card: update tdm mode Namhyung Kim (1): perf diff: Fix duplicated output column Nikolay Aleksandrov (1): net: bridge: fix old ioctl unlocked net device walk PC Liao (2): ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver ASoC: mediatek: HDMI audio LR channel swapped Paolo Abeni (1): net/route: enforce hoplimit max value Paul Durrant (1): xen-netback: fix extra_info handling in xenvif_tx_err() Peter Ujfalusi (4): ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3 ASoC: simple-card: Add pm callbacks to platform driver ASoC: twl6040: Select LPPLL during standby ASoC: twl6040: Disconnect AUX output pads on digital mute Peter Zijlstra (1): perf/x86/msr: Fix SMI overflow Petr Kulhavy (5): ASoC: tas571x: implemented digital mute ASoC: tas571x: chip type detection via I2C name ASoC: tas571x: added missing register literals ASoC: tas571x: added support for TAS5721 ASoC: tas571x: new chip added into TAS571x binding Phil Turnbull (1): netfilter: nfnetlink_acct: validate NFACCT_QUOTA parameter Rafael J. Wysocki (1): sched/fair: Fix !CONFIG_SMP kernel cpufreq governor breakage Robin Humble (1): Revert "proc/base: make prompt shell start from new line after executing "cat /proc/$pid/wchan"" Sabrina Dubroca (1): macsec: key identifier is 128 bits, not 64 Serge E. Hallyn (2): cgroup, kernfs: make mountinfo show properly scoped path for cgroup namespaces kernfs: kernfs_sop_show_path: don't return 0 after seq_dentry call Serge Hallyn (1): kernfs_path_from_node_locked: don't overwrite nlen Sergei Shtylyov (1): rcar: src: skip disabled-SRC nodes Sergey Senozhatsky (1): zsmalloc: fix zs_can_compact() integer overflow Shaohui Xie (1): net: phylib: fix interrupts re-enablement in phy_start Shmulik Ladkani (1): Documentation/networking: more accurate LCO explanation Srinivas Kandagatla (2): ASoC: qcom: Fix uninitialized symbol warning. ASoC: qcom: remove IS_ERR_VALUE usage on int. Steffen Klassert (3): flowcache: Avoid OOM condition under preasure xfrm: Reset encapsulation field of the skb before transformation vti: Add pmtu handling to vti_xmit. Stephen Boyd (1): ASoC: rsnd: Remove CLK_IS_ROOT Steven Rostedt (1): tools lib traceevent: Do not reassign parg after collapse_tree() Sudarsana Reddy Kalluru (1): qede: prevent chip hang when increasing channels Sudip Mukherjee (1): ASoC: kirkwood: fix build failure Suzuki K Poulose (1): arm64/sunxi: 4.6-rc1: Add dependency on generic irq chip Tadeusz Struk (3): crypto: qat - fix invalid pf2vf_resp_wq logic crypto: qat - fix adf_ctl_drv.c:undefined reference to adf_init_pf_wq crypto: rsa - select crypto mgr dependency Takashi Iwai (4): ALSA: usb-audio: Quirk for yet another Phoenix Audio devices (v2) ALSA: hda - Fix broken reconfig ALSA: hda - Fix regression on ATI HDMI audio ALSA: usb-audio: Yet another Phoneix Audio device quirk Thomas Gleixner (2): x86/topology: Handle CPUID bogosity gracefully x86/topology: Set x86_max_cores to 1 for CONFIG_SMP=n Uwe Kleine-König (1): net: fec: only clear a queue's work bit if the queue was emptied Ville Syrjälä (1): drm/i915: Update CDCLK_FREQ register on BDW after changing cdclk frequency Wanpeng Li (1): workqueue: fix rebind bound workers warning Wenyou Yang (1): ARM: dts: at91: sama5d2: use "atmel,sama5d3-nfc" compatible for nfc Xunlei Pang (1): sched/rt, sched/dl: Don't push if task's scheduling class was changed Yura Pakhuchiy (1): ALSA: hda - Fix subwoofer pin on ASUS N751 and N551 Zhou Chengming (1): ksm: fix conflict between mmput and scan_get_next_rmap_item Zi Shen Lim (1): arm64: bpf: jit JMP_JSET_{X,K} anish kumar (1): ASoC: Add max98371 codec driver hchrzani (1): perf/x86/intel/uncore: Fix CHA registers configuration procedure for Knights Landing platform xypron.glpk(a)gmx.de (1): net: thunderx: avoid exposing kernel stack .../devicetree/bindings/regmap/regmap.txt | 59 +- .../devicetree/bindings/sound/max98371.txt | 17 + .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 10 + .../devicetree/bindings/sound/st,sti-asoc-card.txt | 48 +- .../devicetree/bindings/sound/tas571x.txt | 10 +- .../devicetree/bindings/sound/tas5720.txt | 25 + Documentation/networking/checksum-offloads.txt | 14 +- Documentation/sysctl/kernel.txt | 2 +- MAINTAINERS | 14 + Makefile | 2 +- arch/arm/boot/dts/at91sam9x5.dtsi | 2 +- arch/arm/boot/dts/sama5d2.dtsi | 2 +- arch/arm64/Kconfig.platforms | 1 + arch/arm64/net/bpf_jit_comp.c | 1 + arch/x86/configs/kvm_guest.config | 3 + arch/x86/entry/syscalls/syscall_32.tbl | 4 +- arch/x86/events/intel/core.c | 2 +- arch/x86/events/intel/pt.c | 2 + arch/x86/events/intel/uncore_snbep.c | 7 + arch/x86/events/msr.c | 2 +- arch/x86/include/asm/uaccess.h | 8 + arch/x86/kernel/cpu/intel.c | 2 +- arch/x86/kernel/smpboot.c | 5 + arch/x86/kvm/emulate.c | 6 +- block/blk-map.c | 47 +- crypto/Kconfig | 1 + crypto/ahash.c | 3 +- crypto/testmgr.c | 27 +- drivers/base/regmap/internal.h | 1 + drivers/base/regmap/regmap-mmio.c | 5 +- drivers/base/regmap/regmap-spmi.c | 2 +- drivers/crypto/qat/qat_common/adf_common_drv.h | 11 + drivers/crypto/qat/qat_common/adf_ctl_drv.c | 6 + drivers/crypto/qat/qat_common/adf_sriov.c | 26 +- drivers/gpu/drm/amd/amdgpu/atombios_dp.c | 4 +- drivers/gpu/drm/i915/i915_debugfs.c | 16 - drivers/gpu/drm/i915/i915_reg.h | 2 + drivers/gpu/drm/i915/intel_audio.c | 9 +- drivers/gpu/drm/i915/intel_crt.c | 8 +- drivers/gpu/drm/i915/intel_ddi.c | 24 +- drivers/gpu/drm/i915/intel_display.c | 5 +- drivers/gpu/drm/i915/intel_dp_mst.c | 22 - drivers/gpu/drm/i915/intel_drv.h | 2 - drivers/gpu/drm/i915/intel_lvds.c | 4 + drivers/gpu/drm/i915/intel_pm.c | 6 + drivers/gpu/drm/radeon/atombios_crtc.c | 10 + drivers/gpu/drm/radeon/atombios_dp.c | 4 +- drivers/gpu/drm/radeon/radeon_dp_auxch.c | 2 +- drivers/input/misc/max8997_haptic.c | 6 +- drivers/input/misc/twl6040-vibra.c | 1 + drivers/input/mouse/byd.c | 4 + drivers/media/v4l2-core/videobuf2-v4l2.c | 6 - drivers/net/ethernet/apm/xgene/xgene_enet_cle.c | 11 +- drivers/net/ethernet/apm/xgene/xgene_enet_cle.h | 2 + drivers/net/ethernet/apm/xgene/xgene_enet_hw.c | 19 +- drivers/net/ethernet/apm/xgene/xgene_enet_hw.h | 8 +- drivers/net/ethernet/apm/xgene/xgene_enet_main.c | 82 ++- drivers/net/ethernet/apm/xgene/xgene_enet_main.h | 18 +- drivers/net/ethernet/apm/xgene/xgene_enet_sgmac.h | 2 +- drivers/net/ethernet/broadcom/bnxt/bnxt.c | 86 ++- drivers/net/ethernet/broadcom/bnxt/bnxt.h | 2 + drivers/net/ethernet/cavium/thunder/nicvf_queues.c | 4 + drivers/net/ethernet/ezchip/nps_enet.c | 30 +- drivers/net/ethernet/ezchip/nps_enet.h | 2 - drivers/net/ethernet/freescale/fec_main.c | 10 +- drivers/net/ethernet/marvell/Kconfig | 2 +- drivers/net/ethernet/mellanox/mlx4/en_rx.c | 2 +- drivers/net/ethernet/mellanox/mlx5/core/Kconfig | 8 +- drivers/net/ethernet/mellanox/mlx5/core/Makefile | 3 +- drivers/net/ethernet/mellanox/mlx5/core/en.h | 2 + drivers/net/ethernet/mellanox/mlx5/core/en_main.c | 4 + drivers/net/ethernet/mellanox/mlx5/core/vxlan.h | 11 +- drivers/net/ethernet/mellanox/mlxsw/spectrum.c | 4 +- .../ethernet/mellanox/mlxsw/spectrum_switchdev.c | 8 + drivers/net/ethernet/qlogic/netxen/netxen_nic_hw.c | 14 +- .../net/ethernet/qlogic/netxen/netxen_nic_main.c | 3 +- drivers/net/ethernet/qlogic/qede/qede_main.c | 8 +- .../net/ethernet/qlogic/qlcnic/qlcnic_minidump.c | 8 +- drivers/net/ethernet/renesas/ravb_main.c | 2 + drivers/net/geneve.c | 5 +- drivers/net/macsec.c | 19 +- drivers/net/macvtap.c | 2 +- drivers/net/phy/phy.c | 8 +- drivers/net/vxlan.c | 5 +- drivers/net/wireless/intel/iwlwifi/mvm/tx.c | 83 +-- drivers/net/xen-netback/netback.c | 1 + drivers/nvdimm/pmem.c | 4 +- drivers/pci/bus.c | 6 +- drivers/pinctrl/pinctrl-at91-pio4.c | 2 + drivers/regulator/axp20x-regulator.c | 12 +- drivers/regulator/da9063-regulator.c | 2 +- drivers/regulator/gpio-regulator.c | 2 + drivers/regulator/s2mps11.c | 28 +- drivers/scsi/device_handler/scsi_dh_alua.c | 1 + drivers/scsi/qla1280.c | 2 +- drivers/spi/spi-ep93xx.c | 2 +- drivers/spi/spi-fsl-dspi.c | 4 +- drivers/spi/spi-omap2-mcspi.c | 62 ++- drivers/spi/spi-pxa2xx.c | 2 +- drivers/spi/spi-ti-qspi.c | 45 +- fs/ecryptfs/file.c | 71 ++- fs/isofs/rock.c | 13 +- fs/kernfs/dir.c | 6 +- fs/kernfs/mount.c | 15 + fs/namei.c | 59 +- fs/ocfs2/acl.c | 87 +++ fs/ocfs2/acl.h | 5 + fs/ocfs2/file.c | 4 +- fs/ocfs2/namei.c | 23 +- fs/ocfs2/refcounttree.c | 17 +- fs/ocfs2/xattr.c | 14 +- fs/ocfs2/xattr.h | 4 +- fs/open.c | 12 +- fs/overlayfs/super.c | 4 +- fs/proc/base.c | 2 +- fs/splice.c | 3 + include/linux/compiler-gcc.h | 2 +- include/linux/dcache.h | 12 + include/linux/kernfs.h | 2 + include/linux/mfd/samsung/s2mps11.h | 2 + include/linux/mfd/twl6040.h | 1 + include/linux/mm.h | 9 + include/linux/namei.h | 2 + include/linux/netdevice.h | 3 + include/linux/swap.h | 6 +- include/linux/uio.h | 1 + include/net/netns/xfrm.h | 1 + include/net/udp_tunnel.h | 9 - include/uapi/linux/if.h | 28 + include/uapi/linux/if_macsec.h | 4 +- include/uapi/linux/libc-compat.h | 44 ++ include/uapi/linux/tc_act/Kbuild | 1 + include/uapi/sound/asoc.h | 44 +- kernel/cgroup.c | 63 +++ kernel/events/core.c | 2 +- kernel/events/ring_buffer.c | 10 +- kernel/sched/deadline.c | 1 + kernel/sched/fair.c | 29 +- kernel/sched/rt.c | 1 + kernel/workqueue.c | 11 + lib/asn1_decoder.c | 16 +- lib/iov_iter.c | 19 + mm/huge_memory.c | 71 ++- mm/ksm.c | 15 +- mm/memory.c | 22 +- mm/swapfile.c | 13 +- mm/zsmalloc.c | 7 +- net/bridge/br_ioctl.c | 5 +- net/bridge/br_multicast.c | 12 +- net/core/flow.c | 14 +- net/core/rtnetlink.c | 18 +- net/ipv4/fib_semantics.c | 2 + net/ipv4/fou.c | 6 +- net/ipv4/ip_gre.c | 7 +- net/ipv4/ip_vti.c | 18 + net/ipv4/tcp_output.c | 6 +- net/ipv4/udp_offload.c | 8 +- net/ipv6/icmp.c | 5 +- net/ipv6/route.c | 2 + net/ipv6/tcp_ipv6.c | 7 +- net/llc/af_llc.c | 1 + net/netfilter/nf_conntrack_core.c | 6 +- net/netfilter/nfnetlink_acct.c | 2 + net/netfilter/xt_IDLETIMER.c | 1 + net/openvswitch/conntrack.c | 13 + net/sched/act_ife.c | 14 +- net/sched/act_ipt.c | 19 +- net/sched/act_mirred.c | 19 +- net/sched/act_simple.c | 18 +- net/sched/act_skbedit.c | 18 +- net/sched/act_vlan.c | 22 +- net/vmw_vsock/af_vsock.c | 21 +- net/x25/x25_facilities.c | 1 + net/xfrm/xfrm_output.c | 3 + sound/pci/hda/hda_sysfs.c | 8 - sound/pci/hda/patch_hdmi.c | 7 +- sound/pci/hda/patch_realtek.c | 13 + sound/soc/codecs/Kconfig | 38 +- sound/soc/codecs/Makefile | 11 +- sound/soc/codecs/ak4642.c | 3 + sound/soc/codecs/hdac_hdmi.c | 1 - sound/soc/codecs/max98371.c | 441 +++++++++++++++ sound/soc/codecs/max98371.h | 67 +++ sound/soc/codecs/rt298.c | 51 +- sound/soc/codecs/rt298.h | 2 + sound/soc/codecs/rt5677.c | 24 +- sound/soc/codecs/tas571x.c | 141 ++++- sound/soc/codecs/tas571x.h | 22 + sound/soc/codecs/tas5720.c | 620 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ sound/soc/codecs/tlv320aic31xx.c | 10 + sound/soc/codecs/tlv320aic32x4-i2c.c | 74 +++ sound/soc/codecs/tlv320aic32x4-spi.c | 76 +++ sound/soc/codecs/tlv320aic32x4.c | 279 +++++++--- sound/soc/codecs/tlv320aic32x4.h | 7 + sound/soc/codecs/twl6040.c | 16 +- sound/soc/codecs/wm8962.c | 9 +- sound/soc/codecs/wm8962.h | 6 +- sound/soc/generic/simple-card.c | 1 + sound/soc/kirkwood/Kconfig | 1 + sound/soc/mediatek/Kconfig | 1 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 27 + sound/soc/mediatek/mt8173-rt5650.c | 50 +- sound/soc/mediatek/mtk-afe-pcm.c | 2 + sound/soc/omap/mcbsp.c | 8 + sound/soc/omap/omap-pcm.c | 2 + sound/soc/pxa/brownstone.c | 1 + sound/soc/pxa/mioa701_wm9713.c | 1 + sound/soc/pxa/mmp-pcm.c | 1 + sound/soc/pxa/mmp-sspa.c | 1 + sound/soc/pxa/palm27x.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + sound/soc/qcom/lpass-platform.c | 8 +- sound/soc/sh/rcar/adg.c | 8 +- sound/soc/sh/rcar/dma.c | 12 +- sound/soc/sh/rcar/rsnd.h | 13 +- sound/soc/sh/rcar/src.c | 4 + sound/soc/soc-topology.c | 48 +- sound/soc/sti/sti_uniperif.c | 144 ++++- sound/soc/sti/uniperif.h | 220 +++++++- sound/soc/sti/uniperif_player.c | 182 ++++-- sound/soc/sti/uniperif_reader.c | 229 ++++++-- sound/usb/quirks.c | 3 + tools/build/Makefile.feature | 2 + tools/build/feature/Makefile | 4 + tools/build/feature/test-all.c | 5 + tools/build/feature/test-dwarf_getlocations.c | 12 + tools/lib/traceevent/parse-filter.c | 4 +- tools/net/bpf_jit_disasm.c | 3 + tools/perf/arch/x86/util/dwarf-regs.c | 8 +- tools/perf/builtin-script.c | 70 ++- tools/perf/builtin-stat.c | 7 +- tools/perf/config/Makefile | 6 + tools/perf/util/dwarf-aux.c | 9 + tools/perf/util/event.c | 12 +- tools/perf/util/evsel.c | 23 +- tools/perf/util/parse-events.c | 60 +- tools/perf/util/sort.c | 3 + tools/perf/util/thread_map.c | 8 +- 242 files changed, 4385 insertions(+), 1065 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h create mode 100644 sound/soc/codecs/tlv320aic32x4-i2c.c create mode 100644 sound/soc/codecs/tlv320aic32x4-spi.c create mode 100644 tools/build/feature/test-dwarf_getlocations.c

2 2

[alsa-devel] [PATCH 00/14] clk/mfd/ASoC/ARM: OMAP4/5: McPDM/twl6040 pdmclk support
by Peter Ujfalusi 27 May '16

27 May '16

Hi, this is something I wanted to do for a long time... First of all: sorry for the cross domain series. I did tested alone all the domain patches and they are not causing any regression. When they come together we will have ability to control the pdmclk needed by McPDM. CLK patches: The driver was w/o DT support and needed some cleanup so we can use the driver. I have also renamed it to twl6040-pdmclk from twl6040-clk to be precise. MFD patches: The regmap_sync() was not working since the twl6040 does not support bulk access and now we are going to create the needed child for the pdmclk. ARM patches: Updatding the board DTS files and add the needed lines for the pdmclk. ASoC patches: The machine driver needs to select the twl6040-clk driver as it will be used by the omap-mcpdm. I needed to do some cleanup and add also suspend/resume support before adding the code for pdmclk handling since I don't want to keep the twl6040 powered on when the board is suspended. At the moment it is not possible to do true dynamic twl6040 power up/down due to pop noises, but I will be looking at that later. The driver in the future will fail if it is not able to get the pdmclk to avoid kernel crash (McPDM registers are not accessible when pdmclk is not available). I have tested the patches by domain and also in all (I hope) permutation they could be. I have not experienced any regression. Regards, Peter --- Peter Ujfalusi (14): clk: twl6040: Correct clk_ops clk: twl6040: Register the clock as of_clk_provider clk: twl6040: Rename the driver and use consistent names in the code mfd: twl6040: The chip does not support bulk access mfd: twl6040: Register child device for twl6040-pdmclk ARM: dts: omap5-board-common: Add pdmclk binding for audio ARM: dts: omap4-panda-common: Add pdmclk binding for audio ARM: dts: omap4-sdp: Add pdmclk binding for audio ARM: dts: omap4-var-som-om44: Add pdmclk binding for audio ARM: dts: omap4-duovero: Add pdmclk binding for audio ASoC: omap: Kconfig: SND_OMAP_SOC_OMAP_ABE_TWL6040 to select CLK_TWL6040 ASoC: omap-mcpdm: Move the WD enable write inside omap_mcpdm_open_streams() ASoC: omap-mcpdm: Support for suspend resume ASoC: omap-mcpdm: Add support for pdmclk clock handling Documentation/devicetree/bindings/mfd/twl6040.txt | 1 + .../devicetree/bindings/sound/omap-mcpdm.txt | 10 +++ arch/arm/boot/dts/omap4-duovero.dtsi | 5 ++ arch/arm/boot/dts/omap4-panda-common.dtsi | 5 ++ arch/arm/boot/dts/omap4-sdp.dts | 5 ++ arch/arm/boot/dts/omap4-var-som-om44.dtsi | 5 ++ arch/arm/boot/dts/omap5-board-common.dtsi | 5 ++ drivers/clk/clk-twl6040.c | 80 +++++++++++++--------- drivers/mfd/twl6040.c | 6 ++ include/linux/mfd/twl6040.h | 2 +- sound/soc/omap/Kconfig | 1 + sound/soc/omap/omap-mcpdm.c | 74 ++++++++++++++++++-- 12 files changed, 160 insertions(+), 39 deletions(-) -- 2.8.2

3 17

[alsa-devel] [PATCH v4 0/7] ARM/ASoC: OMAP3: Fix McBSP2/3 sidetone support
by Peter Ujfalusi 27 May '16

27 May '16

Hi, Changes since v3: - Rebased on next-20160525 [1] to apply cleanly - Fixed subject line for patch5 - Added Mark's acked-by to the ASoC patches [1] "ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3" is gone missing from linux-next - it was in next-20160517 and I'm not sure when it got dropped. Tony: since Paul did not replied to the series, I'm sending the v4 now, I hope he will have time to look at this. Commit message from v3: Based on the ongoing discussion on v2 (ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone) I have dropped the removal of the sidetone hwmod and only corrected it. The series now includes all related changes needed to have correct sidetone support whenever we boot in legacy or in DT mode. This time the ASoC part have dependency on earlier patches so they can not be applied separately. Regards, Peter --- Peter Ujfalusi (7): ARM: OMAP3: hwmod data: Fix McBSP2/3 sidetone data ARM: dts: omap3: Add clocks to McBSP nodes ARM: OMAP3: McBSP: New callback for McBSP2/3 ICLK idle configuration ARM: OMAP3: pdata-quirks: Add support for McBSP2/3 sidetone handling ASoC: omap-mcbsp: Rename omap_mcbsp_sysfs_remove() to omap_mcbsp_cleanup() ASoC: omap-mcbsp: sidetone: Use the new callback for iclk handling ARM: OMAP2+: McBSP: Remove the old iclk allow/deny idle code arch/arm/boot/dts/omap3.dtsi | 10 ++++++++++ arch/arm/mach-omap2/mcbsp.c | 31 ++++++++++++++++------------- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 24 ++++------------------ arch/arm/mach-omap2/pdata-quirks.c | 18 +++++++++++++++++ include/linux/platform_data/asoc-ti-mcbsp.h | 4 +++- sound/soc/omap/mcbsp.c | 21 +++++++++++++------ sound/soc/omap/mcbsp.h | 3 ++- sound/soc/omap/omap-mcbsp.c | 5 ++++- 8 files changed, 73 insertions(+), 43 deletions(-) -- 2.8.2

2 8

[alsa-devel] [PATCH v4 0/7] ARM/ASoC: OMAP3: Fix McBSP2/3 sidetone support
by Peter Ujfalusi 27 May '16

27 May '16

Hi, Changes since v3: - Rebased on next-20160525 [1] to apply cleanly - Fixed subject line for patch5 - Added Mark's acked-by to the ASoC patches [1] "ASoC: omap-mcbsp: Enable/disable sidetone block auto clock gating for omap3" is gone missing from linux-next - it was in next-20160517 and I'm not sure when it got dropped. Tony: since Paul did not replied to the series, I'm sending the v4 now, I hope he will have time to look at this. Commit message from v3: Based on the ongoing discussion on v2 (ARM: OMAP3: Fix McBSP2/3 hwmod setup for sidetone) I have dropped the removal of the sidetone hwmod and only corrected it. The series now includes all related changes needed to have correct sidetone support whenever we boot in legacy or in DT mode. This time the ASoC part have dependency on earlier patches so they can not be applied separately. Regards, Peter --- Peter Ujfalusi (7): ARM: OMAP3: hwmod data: Fix McBSP2/3 sidetone data ARM: dts: omap3: Add clocks to McBSP nodes ARM: OMAP3: McBSP: New callback for McBSP2/3 ICLK idle configuration ARM: OMAP3: pdata-quirks: Add support for McBSP2/3 sidetone handling ASoC: omap-mcbsp: Rename omap_mcbsp_sysfs_remove() to omap_mcbsp_cleanup() ASoC: omap-mcbsp: sidetone: Use the new callback for iclk handling ARM: OMAP2+: McBSP: Remove the old iclk allow/deny idle code arch/arm/boot/dts/omap3.dtsi | 10 ++++++++++ arch/arm/mach-omap2/mcbsp.c | 31 ++++++++++++++++------------- arch/arm/mach-omap2/omap_hwmod_3xxx_data.c | 24 ++++------------------ arch/arm/mach-omap2/pdata-quirks.c | 18 +++++++++++++++++ include/linux/platform_data/asoc-ti-mcbsp.h | 4 +++- sound/soc/omap/mcbsp.c | 21 +++++++++++++------ sound/soc/omap/mcbsp.h | 3 ++- sound/soc/omap/omap-mcbsp.c | 5 ++++- 8 files changed, 73 insertions(+), 43 deletions(-) -- 2.8.2

2 5

[alsa-devel] [PATCH v2] ASoC: mediatek: Change the order of MCLK clock configuration
by PC Liao 27 May '16

27 May '16

Because MCLK turns on later and turns off earlier than codec, this patch changes the order of MCLK clock configuration. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- Changes since v1: update commit message --- sound/soc/mediatek/mtk-afe-pcm.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index f1c58a2..440ae06 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -360,8 +360,6 @@ static int mtk_afe_i2s_startup(struct snd_pcm_substream *substream, if (dai->active) return 0; - mtk_afe_dais_enable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); - mtk_afe_dais_enable_clks(afe, afe->clocks[MTK_CLK_I2S2_M], NULL); regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, 0); return 0; @@ -380,8 +378,6 @@ static void mtk_afe_i2s_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M, AUD_TCON0_PDN_22M | AUD_TCON0_PDN_24M); - mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S1_M], NULL); - mtk_afe_dais_disable_clks(afe, afe->clocks[MTK_CLK_I2S2_M], NULL); } static int mtk_afe_i2s_prepare(struct snd_pcm_substream *substream, @@ -1132,6 +1128,8 @@ static int mtk_afe_runtime_suspend(struct device *dev) regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_AFE, AUD_TCON0_PDN_AFE); + clk_disable_unprepare(afe->clocks[MTK_CLK_I2S1_M]); + clk_disable_unprepare(afe->clocks[MTK_CLK_I2S2_M]); clk_disable_unprepare(afe->clocks[MTK_CLK_BCK0]); clk_disable_unprepare(afe->clocks[MTK_CLK_BCK1]); clk_disable_unprepare(afe->clocks[MTK_CLK_TOP_PDN_AUD]); @@ -1164,6 +1162,12 @@ static int mtk_afe_runtime_resume(struct device *dev) ret = clk_prepare_enable(afe->clocks[MTK_CLK_BCK1]); if (ret) goto err_bck0; + ret = clk_prepare_enable(afe->clocks[MTK_CLK_I2S1_M]); + if (ret) + goto err_i2s1_m; + ret = clk_prepare_enable(afe->clocks[MTK_CLK_I2S2_M]); + if (ret) + goto err_i2s2_m; /* enable AFE clk */ regmap_update_bits(afe->regmap, AUDIO_TOP_CON0, AUD_TCON0_PDN_AFE, 0); @@ -1179,6 +1183,10 @@ static int mtk_afe_runtime_resume(struct device *dev) regmap_update_bits(afe->regmap, AFE_DAC_CON0, 0x1, 0x1); return 0; +err_i2s1_m: + clk_disable_unprepare(afe->clocks[MTK_CLK_I2S1_M]); +err_i2s2_m: + clk_disable_unprepare(afe->clocks[MTK_CLK_I2S2_M]); err_bck0: clk_disable_unprepare(afe->clocks[MTK_CLK_BCK0]); err_top_aud: -- 1.7.9.5

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[alsa-devel] [PATCH 0/6] Rockchip Type-C and DispplayPort driver
by Chris Zhong 27 May '16

27 May '16

Hi all This series patch is for rockchip Type-C phy and DisplayPort controller driver. The USB Type-C PHY is designed to support the USB3 and DP applications. The PHY basically has two main components: USB3 and DisplyPort. USB3 operates in SuperSpeed mode and the DP can operate at RBR, HBR and HBR2 data rates. The Type-C cable orientation detection and Power Delivery (PD) is accomplished using a PD PHY or a exernal PD chip. The DP controller is compliant with DisplayPort Specification, Version 1.3, This IP is compatible with the rockchip type-c PHY IP. There is a uCPU in DP controller, it need a firmware to work, please put the firmware file to /lib/firmware/cdn/dptx.bin. The uCPU in charge of aux communication and link training, the host use mailbox to communicate with the ucpu. The PHY driver has register a notification, to get the alt mode from PD, the PD driver need call the tcphy_notifier_call_chain to notify PHY and DP controller. This series is based on Mark Yao's branch: https://github.com/markyzq/kernel-drm-rockchip/tree/drm-rockchip-next-2016-… I test this patches on the rk3399-evb board, with a fusb302 driver, this branch has no rk3399.dtsi, so the patch about dts is not included in this series. Chris Zhong (6): phy: Add USB Type-C PHY driver for rk3399 Documentation: bindings: add dt doc for Rockchip USB Type-C PHY drm/rockchip: vop: add cdn DP support for rk3399 Documentation: bindings: add dt documentation for cdn DP controller ASoC: cdn-dp: Add cdn DP codec driver ASoC: rockchip: Add machine driver for cdn dp codec .../bindings/display/rockchip/cdn-dp-rockchip.txt | 57 ++ .../devicetree/bindings/phy/phy-rockchip-typec.txt | 55 ++ .../bindings/sound/rockchip-cdn-dp-audio.txt | 12 + drivers/gpu/drm/rockchip/Kconfig | 9 + drivers/gpu/drm/rockchip/Makefile | 1 + drivers/gpu/drm/rockchip/cdn-dp-core.c | 620 ++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-core.h | 95 +++ drivers/gpu/drm/rockchip/cdn-dp-reg.c | 730 ++++++++++++++++++ drivers/gpu/drm/rockchip/cdn-dp-reg.h | 404 ++++++++++ drivers/gpu/drm/rockchip/rockchip_drm_vop.c | 9 +- drivers/gpu/drm/rockchip/rockchip_drm_vop.h | 2 + drivers/gpu/drm/rockchip/rockchip_vop_reg.c | 2 + drivers/phy/Kconfig | 7 + drivers/phy/Makefile | 1 + drivers/phy/phy-rockchip-typec.c | 823 +++++++++++++++++++++ include/sound/cdn-dp-audio.h | 51 ++ sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cdn-dp-audio.c | 246 ++++++ sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip-cdn-dp-audio.c | 167 +++++ 22 files changed, 3306 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/display/rockchip/cdn-dp-rockchip.txt create mode 100644 Documentation/devicetree/bindings/phy/phy-rockchip-typec.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-cdn-dp-audio.txt create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-core.c create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-core.h create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-reg.c create mode 100644 drivers/gpu/drm/rockchip/cdn-dp-reg.h create mode 100644 drivers/phy/phy-rockchip-typec.c create mode 100644 include/sound/cdn-dp-audio.h create mode 100644 sound/soc/codecs/cdn-dp-audio.c create mode 100644 sound/soc/rockchip/rockchip-cdn-dp-audio.c -- 2.6.3

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