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[alsa-devel] [PATCH 0/2][RFC] ASoC: add simple-graph-card support
by Kuninori Morimoto 27 May '16

27 May '16

Hi Mark Cc Laurent This is graph base DT support with simple-card. Because it is still [RFC], I don't post full-patch-set yet, but now we can use simple-card with graph style :) (If my previous simple-card cleanup patches are accepted :) Because it is based on simple-card, simple-graph-card can use similar feature. But unfortunately not full feature are tested well, because I don't have some of them. Thesea are not well tested feature. Maybe no big problem, but I don't know simple-audio-card,widgets simple-audio-card,routing simple-audio-card,mclk-fs simple-audio-card,hp-det-gpio simple-audio-card,mic-det-gpio simple-audio-card,dai-tdm-slot-num simple-audio-card,dai-tdm-slot-width Big difference between simple-card <-> simple-graph-card is that simple-graph-card doesn't support "platform" endpoint. I don't know how to handle it under graph. To support graph base DT, CPU/Codec driver might need to adjust. And, as Jean-Francois mentioned before, CPU/Platform driver need to register simple-graph-card card. 1 big concern is that it doesn't care about video port at this point. So, if device has both video/sound ports, simple-graph-card can't probe card correctly. We need to have .type or similar property to know it was "video port" or "sound port" ? (ASoC need to know dai name, and it is based on total sound port number) Below are simple-card <-> simple-graph-card DT style ===================== simple-card ===================== sound_card { compatible = "simple-audio-card"; simple-audio-card,format = "left_j"; simple-audio-card,bitclock-master = <&sndcodec>; simple-audio-card,frame-master = <&sndcodec>; sndcpu: simple-audio-card,cpu { sound-dai = <&soc_port>; }; sndcodec: simple-audio-card,codec { sound-dai = <&codec_port>; clocks = <&xxx>; }; }; codec_port: codec { #sound-dai-cells = <0>; ... }; soc_port: SoC { #sound-dai-cells = <0>; ... soc-port-specific-property }; ===================== simple-graph-card ===================== codec { ... port { codec_port: endpoint { remote-endpoint = <&soc_port>; clocks = <&xxx>; }; }; }; SoC { ... port { simple-audio-card,format = "left_j"; simple-audio-card,bitclock-master = <&codec_port>; simple-audio-card,frame-master = <&codec_port>; soc_port: endpoint { remote-endpoint = <&codec_port>; soc-port-specific-property }; }; }; Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH v3 1/2] ASoC: cs35l33: Initial commit of the cs35l33 CODEC driver.
by Paul.Handrigan＠cirrus.com 27 May '16

27 May '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 | 54 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs35l33.c | 1304 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l33.h | 221 ++++++++ 5 files changed, 1586 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..d18181c --- /dev/null +++ b/include/sound/cs35l33.h @@ -0,0 +1,54 @@ +/* + * linux/sound/cs35l33.h -- Platform data for CS35l33 + * + * Copyright (c) 2015 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 { + /* GPIO for IRQ */ + int gpio_irq; + + /* irq line */ + int irq; + + /* 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 c6f86ef..6ba6f8f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -46,6 +46,7 @@ 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 @@ -379,6 +380,10 @@ 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 94873ee..22e7474 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -35,6 +35,7 @@ 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 @@ -249,6 +250,7 @@ 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..7713627 --- /dev/null +++ b/sound/soc/codecs/cs35l33.c @@ -0,0 +1,1304 @@ +/* + * 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 20 + +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 regmap *cs35l33_get_regmap(struct device *dev) +{ + struct cs35l33_private *cs35l33 = dev_get_drvdata(dev); + + return cs35l33->regmap; +} + +static struct snd_soc_codec_driver soc_codec_dev_cs35l33 = { + .probe = cs35l33_probe, + + .get_regmap = cs35l33_get_regmap, + .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; + + for (i = 0; i < CS35L33_INT_ATTEMPTS; i++) { + /* ack the irq by reading both status registers */ + 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 at the exit. + */ + if (!(sticky_val1 & ~mask1) && !(sticky_val2 & ~mask2)) + break; + + + regmap_read(cs35l33->regmap, CS35L33_INT_STATUS_1, + &current_val); + + /* handle the interrupts */ + + if (sticky_val1 & CS35L33_AMP_SHORT) { + dev_err(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_err(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"); + } + + if (i == CS35L33_INT_ATTEMPTS) { + dev_err(codec->dev, + "ERROR: Interrupt handler exceeded max attpmpts\n"); + return -EINVAL; + } + + 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

3 5

[alsa-devel] [PATCH v5] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 26 May '16

26 May '16

From: Petr Kulhavy <petr(a)barix.com> The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch adds the WM8758 support into the WM8985 driver. The chip selection is done by the I2C name. The SPI probe supports only the WM8985. Signed-off-by: Petr Kulhavy <petr(a)barix.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- v1: initial v2: use chip type enum instead of chip structure do not copy the complete controls, widget, routes but only the differences and add widgets dynamically using add_widgets() v3: merge the wm8985_xxx_snd_controls into wm8985_common_snd_controls and wm8985_specific_snd_controls in order to reduce the number of structures use the static initialization in snd_soc_codec_driver for the common part v4: avoid duplicating left/right_out_mixer and left/right_boost_mixer by moving the aux controls at the end and providing shorter size for wm8758 do not move the high pass filter controls v5: reapplied to git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git branch topics/wm8985 sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8985.c | 143 ++++++++++++++++++++++++++++++++++------------ 2 files changed, 109 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b3afae9..5c635f7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -944,7 +944,7 @@ config SND_SOC_WM8983 tristate config SND_SOC_WM8985 - tristate + tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 6ac76fe..7347abf 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1,10 +1,13 @@ /* - * wm8985.c -- WM8985 ALSA SoC Audio driver + * wm8985.c -- WM8985 / WM8758 ALSA SoC Audio driver * * Copyright 2010 Wolfson Microelectronics plc - * * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> * + * WM8758 support: + * Copyright: 2016 Barix AG + * Author: Petr Kulhavy <petr(a)barix.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. @@ -40,6 +43,11 @@ static const char *wm8985_supply_names[WM8985_NUM_SUPPLIES] = { "AVDD2" }; +enum wm8985_type { + WM8985, + WM8758, +}; + static const struct reg_default wm8985_reg_defaults[] = { { 1, 0x0000 }, /* R1 - Power management 1 */ { 2, 0x0000 }, /* R2 - Power management 2 */ @@ -181,6 +189,7 @@ static const int volume_update_regs[] = { struct wm8985_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8985_NUM_SUPPLIES]; + enum wm8985_type dev_type; unsigned int sysclk; unsigned int bclk; }; @@ -289,7 +298,7 @@ static const char *depth_3d_text[] = { }; static SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, depth_3d_text); -static const struct snd_kcontrol_new wm8985_snd_controls[] = { +static const struct snd_kcontrol_new wm8985_common_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8985_COMPANDING_CONTROL, 0, 1, 0), @@ -355,10 +364,6 @@ static const struct snd_kcontrol_new wm8985_snd_controls[] = { SOC_ENUM("High Pass Filter Mode", filter_mode), SOC_SINGLE("High Pass Filter Cutoff", WM8985_ADC_CONTROL, 4, 7, 0), - SOC_DOUBLE_R_TLV("Aux Bypass Volume", - WM8985_LEFT_MIXER_CTRL, WM8985_RIGHT_MIXER_CTRL, 6, 7, 0, - aux_tlv), - SOC_DOUBLE_R_TLV("Input PGA Bypass Volume", WM8985_LEFT_MIXER_CTRL, WM8985_RIGHT_MIXER_CTRL, 2, 7, 0, bypass_tlv), @@ -379,20 +384,30 @@ static const struct snd_kcontrol_new wm8985_snd_controls[] = { SOC_SINGLE_TLV("EQ5 Volume", WM8985_EQ5_HIGH_SHELF, 0, 24, 1, eq_tlv), SOC_ENUM("3D Depth", depth_3d), +}; + +static const struct snd_kcontrol_new wm8985_specific_snd_controls[] = { + SOC_DOUBLE_R_TLV("Aux Bypass Volume", + WM8985_LEFT_MIXER_CTRL, WM8985_RIGHT_MIXER_CTRL, 6, 7, 0, + aux_tlv), SOC_ENUM("Speaker Mode", speaker_mode) }; static const struct snd_kcontrol_new left_out_mixer[] = { SOC_DAPM_SINGLE("Line Switch", WM8985_LEFT_MIXER_CTRL, 1, 1, 0), - SOC_DAPM_SINGLE("Aux Switch", WM8985_LEFT_MIXER_CTRL, 5, 1, 0), SOC_DAPM_SINGLE("PCM Switch", WM8985_LEFT_MIXER_CTRL, 0, 1, 0), + + /* --- WM8985 only --- */ + SOC_DAPM_SINGLE("Aux Switch", WM8985_LEFT_MIXER_CTRL, 5, 1, 0), }; static const struct snd_kcontrol_new right_out_mixer[] = { SOC_DAPM_SINGLE("Line Switch", WM8985_RIGHT_MIXER_CTRL, 1, 1, 0), - SOC_DAPM_SINGLE("Aux Switch", WM8985_RIGHT_MIXER_CTRL, 5, 1, 0), SOC_DAPM_SINGLE("PCM Switch", WM8985_RIGHT_MIXER_CTRL, 0, 1, 0), + + /* --- WM8985 only --- */ + SOC_DAPM_SINGLE("Aux Switch", WM8985_RIGHT_MIXER_CTRL, 5, 1, 0), }; static const struct snd_kcontrol_new left_input_mixer[] = { @@ -410,6 +425,8 @@ static const struct snd_kcontrol_new right_input_mixer[] = { static const struct snd_kcontrol_new left_boost_mixer[] = { SOC_DAPM_SINGLE_TLV("L2 Volume", WM8985_LEFT_ADC_BOOST_CTRL, 4, 7, 0, boost_tlv), + + /* --- WM8985 only --- */ SOC_DAPM_SINGLE_TLV("AUXL Volume", WM8985_LEFT_ADC_BOOST_CTRL, 0, 7, 0, boost_tlv) }; @@ -417,11 +434,13 @@ static const struct snd_kcontrol_new left_boost_mixer[] = { static const struct snd_kcontrol_new right_boost_mixer[] = { SOC_DAPM_SINGLE_TLV("R2 Volume", WM8985_RIGHT_ADC_BOOST_CTRL, 4, 7, 0, boost_tlv), + + /* --- WM8985 only --- */ SOC_DAPM_SINGLE_TLV("AUXR Volume", WM8985_RIGHT_ADC_BOOST_CTRL, 0, 7, 0, boost_tlv) }; -static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { +static const struct snd_soc_dapm_widget wm8985_common_dapm_widgets[] = { SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8985_POWER_MANAGEMENT_3, 0, 0), SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8985_POWER_MANAGEMENT_3, @@ -431,21 +450,11 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8985_POWER_MANAGEMENT_2, 1, 0), - SND_SOC_DAPM_MIXER("Left Output Mixer", WM8985_POWER_MANAGEMENT_3, - 2, 0, left_out_mixer, ARRAY_SIZE(left_out_mixer)), - SND_SOC_DAPM_MIXER("Right Output Mixer", WM8985_POWER_MANAGEMENT_3, - 3, 0, right_out_mixer, ARRAY_SIZE(right_out_mixer)), - SND_SOC_DAPM_MIXER("Left Input Mixer", WM8985_POWER_MANAGEMENT_2, 2, 0, left_input_mixer, ARRAY_SIZE(left_input_mixer)), SND_SOC_DAPM_MIXER("Right Input Mixer", WM8985_POWER_MANAGEMENT_2, 3, 0, right_input_mixer, ARRAY_SIZE(right_input_mixer)), - SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8985_POWER_MANAGEMENT_2, - 4, 0, left_boost_mixer, ARRAY_SIZE(left_boost_mixer)), - SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8985_POWER_MANAGEMENT_2, - 5, 0, right_boost_mixer, ARRAY_SIZE(right_boost_mixer)), - SND_SOC_DAPM_PGA("Left Capture PGA", WM8985_LEFT_INP_PGA_GAIN_CTRL, 6, 1, NULL, 0), SND_SOC_DAPM_PGA("Right Capture PGA", WM8985_RIGHT_INP_PGA_GAIN_CTRL, @@ -468,8 +477,6 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LIP"), SND_SOC_DAPM_INPUT("RIN"), SND_SOC_DAPM_INPUT("RIP"), - SND_SOC_DAPM_INPUT("AUXL"), - SND_SOC_DAPM_INPUT("AUXR"), SND_SOC_DAPM_INPUT("L2"), SND_SOC_DAPM_INPUT("R2"), SND_SOC_DAPM_OUTPUT("HPL"), @@ -478,13 +485,42 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR") }; -static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { +static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Left Output Mixer", WM8985_POWER_MANAGEMENT_3, + 2, 0, left_out_mixer, ARRAY_SIZE(left_out_mixer)), + SND_SOC_DAPM_MIXER("Right Output Mixer", WM8985_POWER_MANAGEMENT_3, + 3, 0, right_out_mixer, ARRAY_SIZE(right_out_mixer)), + + SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 4, 0, left_boost_mixer, ARRAY_SIZE(left_boost_mixer)), + SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 5, 0, right_boost_mixer, ARRAY_SIZE(right_boost_mixer)), + + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), +}; + +static const struct snd_soc_dapm_widget wm8758_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Left Output Mixer", WM8985_POWER_MANAGEMENT_3, + 2, 0, left_out_mixer, + ARRAY_SIZE(left_out_mixer) - 1), + SND_SOC_DAPM_MIXER("Right Output Mixer", WM8985_POWER_MANAGEMENT_3, + 3, 0, right_out_mixer, + ARRAY_SIZE(right_out_mixer) - 1), + + SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 4, 0, left_boost_mixer, + ARRAY_SIZE(left_boost_mixer) - 1), + SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 5, 0, right_boost_mixer, + ARRAY_SIZE(right_boost_mixer) - 1), +}; + +static const struct snd_soc_dapm_route wm8985_common_dapm_routes[] = { { "Right Output Mixer", "PCM Switch", "Right DAC" }, - { "Right Output Mixer", "Aux Switch", "AUXR" }, { "Right Output Mixer", "Line Switch", "Right Boost Mixer" }, { "Left Output Mixer", "PCM Switch", "Left DAC" }, - { "Left Output Mixer", "Aux Switch", "AUXL" }, { "Left Output Mixer", "Line Switch", "Left Boost Mixer" }, { "Right Headphone Out", NULL, "Right Output Mixer" }, @@ -501,13 +537,11 @@ static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { { "Right ADC", NULL, "Right Boost Mixer" }, - { "Right Boost Mixer", "AUXR Volume", "AUXR" }, { "Right Boost Mixer", NULL, "Right Capture PGA" }, { "Right Boost Mixer", "R2 Volume", "R2" }, { "Left ADC", NULL, "Left Boost Mixer" }, - { "Left Boost Mixer", "AUXL Volume", "AUXL" }, { "Left Boost Mixer", NULL, "Left Capture PGA" }, { "Left Boost Mixer", "L2 Volume", "L2" }, @@ -522,6 +556,38 @@ static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { { "Left Input Mixer", "MicN Switch", "LIN" }, { "Left Input Mixer", "MicP Switch", "LIP" }, }; +static const struct snd_soc_dapm_route wm8985_aux_dapm_routes[] = { + { "Right Output Mixer", "Aux Switch", "AUXR" }, + { "Left Output Mixer", "Aux Switch", "AUXL" }, + + { "Right Boost Mixer", "AUXR Volume", "AUXR" }, + { "Left Boost Mixer", "AUXL Volume", "AUXL" }, +}; + +static int wm8985_add_widgets(struct snd_soc_codec *codec) +{ + struct wm8985_priv *wm8985 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + switch (wm8985->dev_type) { + case WM8758: + snd_soc_dapm_new_controls(dapm, wm8758_dapm_widgets, + ARRAY_SIZE(wm8758_dapm_widgets)); + break; + + case WM8985: + snd_soc_add_codec_controls(codec, wm8985_specific_snd_controls, + ARRAY_SIZE(wm8985_specific_snd_controls)); + + snd_soc_dapm_new_controls(dapm, wm8985_dapm_widgets, + ARRAY_SIZE(wm8985_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, wm8985_aux_dapm_routes, + ARRAY_SIZE(wm8985_aux_dapm_routes)); + break; + } + + return 0; +} static int eqmode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -999,6 +1065,8 @@ static int wm8985_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8985_BIAS_CTRL, WM8985_BIASCUT, WM8985_BIASCUT); + wm8985_add_widgets(codec); + return 0; err_reg_enable: @@ -1042,12 +1110,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8985 = { .set_bias_level = wm8985_set_bias_level, .suspend_bias_off = true, - .controls = wm8985_snd_controls, - .num_controls = ARRAY_SIZE(wm8985_snd_controls), - .dapm_widgets = wm8985_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8985_dapm_widgets), - .dapm_routes = wm8985_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(wm8985_dapm_routes), + .controls = wm8985_common_snd_controls, + .num_controls = ARRAY_SIZE(wm8985_common_snd_controls), + .dapm_widgets = wm8985_common_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8985_common_dapm_widgets), + .dapm_routes = wm8985_common_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8985_common_dapm_routes), }; static const struct regmap_config wm8985_regmap = { @@ -1074,6 +1142,8 @@ static int wm8985_spi_probe(struct spi_device *spi) spi_set_drvdata(spi, wm8985); + wm8985->dev_type = WM8985; + wm8985->regmap = devm_regmap_init_spi(spi, &wm8985_regmap); if (IS_ERR(wm8985->regmap)) { ret = PTR_ERR(wm8985->regmap); @@ -1115,6 +1185,8 @@ static int wm8985_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8985); + wm8985->dev_type = id->driver_data; + wm8985->regmap = devm_regmap_init_i2c(i2c, &wm8985_regmap); if (IS_ERR(wm8985->regmap)) { ret = PTR_ERR(wm8985->regmap); @@ -1135,7 +1207,8 @@ static int wm8985_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8985_i2c_id[] = { - { "wm8985", 0 }, + { "wm8985", WM8985 }, + { "wm8758", WM8758 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8985_i2c_id); @@ -1183,6 +1256,6 @@ static void __exit wm8985_exit(void) } module_exit(wm8985_exit); -MODULE_DESCRIPTION("ASoC WM8985 driver"); +MODULE_DESCRIPTION("ASoC WM8985 / WM8758 driver"); MODULE_AUTHOR("Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); -- 2.5.0

1 0

[alsa-devel] [PATCH 1/2] ASoC: tas571x: add biquad registers for TAS5717/19
by Petr Kulhavy 26 May '16

26 May '16

Add biquad register definitions for TAS5717/19. Signed-off-by: Petr Kulhavy <brain(a)jikos.cz> --- sound/soc/codecs/tas571x.h | 35 +++++++++++++++++++++++++++++++++++ 1 file changed, 35 insertions(+) diff --git a/sound/soc/codecs/tas571x.h b/sound/soc/codecs/tas571x.h index cf800c3..bf4d436 100644 --- a/sound/soc/codecs/tas571x.h +++ b/sound/soc/codecs/tas571x.h @@ -52,4 +52,39 @@ #define TAS571X_CH4_SRC_SELECT_REG 0x21 #define TAS571X_PWM_MUX_REG 0x25 +/* 20-byte biquad registers */ +#define TAS5717_CH1_BQ0_REG 0x26 +#define TAS5717_CH1_BQ1_REG 0x27 +#define TAS5717_CH1_BQ2_REG 0x28 +#define TAS5717_CH1_BQ3_REG 0x29 +#define TAS5717_CH1_BQ4_REG 0x2a +#define TAS5717_CH1_BQ5_REG 0x2b +#define TAS5717_CH1_BQ6_REG 0x2c +#define TAS5717_CH1_BQ7_REG 0x2d +#define TAS5717_CH1_BQ8_REG 0x2e +#define TAS5717_CH1_BQ9_REG 0x2f + +#define TAS5717_CH2_BQ0_REG 0x30 +#define TAS5717_CH2_BQ1_REG 0x31 +#define TAS5717_CH2_BQ2_REG 0x32 +#define TAS5717_CH2_BQ3_REG 0x33 +#define TAS5717_CH2_BQ4_REG 0x34 +#define TAS5717_CH2_BQ5_REG 0x35 +#define TAS5717_CH2_BQ6_REG 0x36 +#define TAS5717_CH2_BQ7_REG 0x37 +#define TAS5717_CH2_BQ8_REG 0x38 +#define TAS5717_CH2_BQ9_REG 0x39 + +#define TAS5717_CH1_BQ10_REG 0x58 +#define TAS5717_CH1_BQ11_REG 0x59 + +#define TAS5717_CH4_BQ0_REG 0x5a +#define TAS5717_CH4_BQ1_REG 0x5b + +#define TAS5717_CH2_BQ10_REG 0x5c +#define TAS5717_CH2_BQ11_REG 0x5d + +#define TAS5717_CH3_BQ0_REG 0x5e +#define TAS5717_CH3_BQ1_REG 0x5f + #endif /* _TAS571X_H */ -- 2.5.0

1 1

[alsa-devel] [PATCH] ASoC: mediatek: Change the order of MCLK clock configuration
by PC Liao 26 May '16

26 May '16

Because MCLK opens later and closes earlier than codec, this patch changes the order of MCLK clock configuration. Signed-off-by: PC Liao <pc.liao(a)mediatek.com> --- 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

1 0

[alsa-devel] [PATCH v4] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 26 May '16

26 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch adds the WM8758 support into the WM8985 driver. The chip selection is done by the I2C name. The SPI probe supports only the WM8985. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- v1: initial v2: use chip type enum instead of chip structure do not copy the complete controls, widget, routes but only the differences and add widgets dynamically using add_widgets() v3: merge the wm8985_xxx_snd_controls into wm8985_common_snd_controls and wm8985_specific_snd_controls in order to reduce the number of structures use the static initialization in snd_soc_codec_driver for the common part v4: avoid duplicating left/right_out_mixer and left/right_boost_mixer by moving the aux controls at the end and providing shorter size for wm8758 do not move the high pass filter controls sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/wm8985.c | 143 ++++++++++++++++++++++++++++++++++------------ 2 files changed, 109 insertions(+), 36 deletions(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 649e92a252ae..5947e0c94d02 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -933,7 +933,7 @@ config SND_SOC_WM8983 tristate config SND_SOC_WM8985 - tristate + tristate "Wolfson Microelectronics WM8985 and WM8758 codec driver" config SND_SOC_WM8988 tristate diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 42696e709580..c5b79e29624f 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1,10 +1,13 @@ /* - * wm8985.c -- WM8985 ALSA SoC Audio driver + * wm8985.c -- WM8985 / WM8758 ALSA SoC Audio driver * * Copyright 2010 Wolfson Microelectronics plc - * * Author: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> * + * WM8758 support: + * Copyright: 2016 Barix AG + * Author: Petr Kulhavy <petr(a)barix.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. @@ -40,6 +43,11 @@ static const char *wm8985_supply_names[WM8985_NUM_SUPPLIES] = { "AVDD2" }; +enum wm8985_type { + WM8985, + WM8758, +}; + static const struct reg_default wm8985_reg_defaults[] = { { 1, 0x0000 }, /* R1 - Power management 1 */ { 2, 0x0000 }, /* R2 - Power management 2 */ @@ -181,6 +189,7 @@ static const int volume_update_regs[] = { struct wm8985_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8985_NUM_SUPPLIES]; + enum wm8985_type dev_type; unsigned int pllout; /* input rate to the MCLKDIV divider */ }; @@ -283,7 +292,7 @@ static const char *depth_3d_text[] = { static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, depth_3d_text); -static const struct snd_kcontrol_new wm8985_snd_controls[] = { +static const struct snd_kcontrol_new wm8985_common_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8985_COMPANDING_CONTROL, 0, 1, 0), @@ -349,10 +358,6 @@ static const struct snd_kcontrol_new wm8985_snd_controls[] = { SOC_ENUM("High Pass Filter Mode", filter_mode), SOC_SINGLE("High Pass Filter Cutoff", WM8985_ADC_CONTROL, 4, 7, 0), - SOC_DOUBLE_R_TLV("Aux Bypass Volume", - WM8985_LEFT_MIXER_CTRL, WM8985_RIGHT_MIXER_CTRL, 6, 7, 0, - aux_tlv), - SOC_DOUBLE_R_TLV("Input PGA Bypass Volume", WM8985_LEFT_MIXER_CTRL, WM8985_RIGHT_MIXER_CTRL, 2, 7, 0, bypass_tlv), @@ -373,20 +378,30 @@ static const struct snd_kcontrol_new wm8985_snd_controls[] = { SOC_SINGLE_TLV("EQ5 Volume", WM8985_EQ5_HIGH_SHELF, 0, 24, 1, eq_tlv), SOC_ENUM("3D Depth", depth_3d), +}; + +static const struct snd_kcontrol_new wm8985_specific_snd_controls[] = { + SOC_DOUBLE_R_TLV("Aux Bypass Volume", + WM8985_LEFT_MIXER_CTRL, WM8985_RIGHT_MIXER_CTRL, 6, 7, 0, + aux_tlv), SOC_ENUM("Speaker Mode", speaker_mode) }; static const struct snd_kcontrol_new left_out_mixer[] = { SOC_DAPM_SINGLE("Line Switch", WM8985_LEFT_MIXER_CTRL, 1, 1, 0), - SOC_DAPM_SINGLE("Aux Switch", WM8985_LEFT_MIXER_CTRL, 5, 1, 0), SOC_DAPM_SINGLE("PCM Switch", WM8985_LEFT_MIXER_CTRL, 0, 1, 0), + + /* --- WM8985 only --- */ + SOC_DAPM_SINGLE("Aux Switch", WM8985_LEFT_MIXER_CTRL, 5, 1, 0), }; static const struct snd_kcontrol_new right_out_mixer[] = { SOC_DAPM_SINGLE("Line Switch", WM8985_RIGHT_MIXER_CTRL, 1, 1, 0), - SOC_DAPM_SINGLE("Aux Switch", WM8985_RIGHT_MIXER_CTRL, 5, 1, 0), SOC_DAPM_SINGLE("PCM Switch", WM8985_RIGHT_MIXER_CTRL, 0, 1, 0), + + /* --- WM8985 only --- */ + SOC_DAPM_SINGLE("Aux Switch", WM8985_RIGHT_MIXER_CTRL, 5, 1, 0), }; static const struct snd_kcontrol_new left_input_mixer[] = { @@ -404,6 +419,8 @@ static const struct snd_kcontrol_new right_input_mixer[] = { static const struct snd_kcontrol_new left_boost_mixer[] = { SOC_DAPM_SINGLE_TLV("L2 Volume", WM8985_LEFT_ADC_BOOST_CTRL, 4, 7, 0, boost_tlv), + + /* --- WM8985 only --- */ SOC_DAPM_SINGLE_TLV("AUXL Volume", WM8985_LEFT_ADC_BOOST_CTRL, 0, 7, 0, boost_tlv) }; @@ -411,11 +428,13 @@ static const struct snd_kcontrol_new left_boost_mixer[] = { static const struct snd_kcontrol_new right_boost_mixer[] = { SOC_DAPM_SINGLE_TLV("R2 Volume", WM8985_RIGHT_ADC_BOOST_CTRL, 4, 7, 0, boost_tlv), + + /* --- WM8985 only --- */ SOC_DAPM_SINGLE_TLV("AUXR Volume", WM8985_RIGHT_ADC_BOOST_CTRL, 0, 7, 0, boost_tlv) }; -static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { +static const struct snd_soc_dapm_widget wm8985_common_dapm_widgets[] = { SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8985_POWER_MANAGEMENT_3, 0, 0), SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8985_POWER_MANAGEMENT_3, @@ -425,21 +444,11 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8985_POWER_MANAGEMENT_2, 1, 0), - SND_SOC_DAPM_MIXER("Left Output Mixer", WM8985_POWER_MANAGEMENT_3, - 2, 0, left_out_mixer, ARRAY_SIZE(left_out_mixer)), - SND_SOC_DAPM_MIXER("Right Output Mixer", WM8985_POWER_MANAGEMENT_3, - 3, 0, right_out_mixer, ARRAY_SIZE(right_out_mixer)), - SND_SOC_DAPM_MIXER("Left Input Mixer", WM8985_POWER_MANAGEMENT_2, 2, 0, left_input_mixer, ARRAY_SIZE(left_input_mixer)), SND_SOC_DAPM_MIXER("Right Input Mixer", WM8985_POWER_MANAGEMENT_2, 3, 0, right_input_mixer, ARRAY_SIZE(right_input_mixer)), - SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8985_POWER_MANAGEMENT_2, - 4, 0, left_boost_mixer, ARRAY_SIZE(left_boost_mixer)), - SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8985_POWER_MANAGEMENT_2, - 5, 0, right_boost_mixer, ARRAY_SIZE(right_boost_mixer)), - SND_SOC_DAPM_PGA("Left Capture PGA", WM8985_LEFT_INP_PGA_GAIN_CTRL, 6, 1, NULL, 0), SND_SOC_DAPM_PGA("Right Capture PGA", WM8985_RIGHT_INP_PGA_GAIN_CTRL, @@ -462,8 +471,6 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_INPUT("LIP"), SND_SOC_DAPM_INPUT("RIN"), SND_SOC_DAPM_INPUT("RIP"), - SND_SOC_DAPM_INPUT("AUXL"), - SND_SOC_DAPM_INPUT("AUXR"), SND_SOC_DAPM_INPUT("L2"), SND_SOC_DAPM_INPUT("R2"), SND_SOC_DAPM_OUTPUT("HPL"), @@ -472,13 +479,42 @@ static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("SPKR") }; -static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { +static const struct snd_soc_dapm_widget wm8985_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Left Output Mixer", WM8985_POWER_MANAGEMENT_3, + 2, 0, left_out_mixer, ARRAY_SIZE(left_out_mixer)), + SND_SOC_DAPM_MIXER("Right Output Mixer", WM8985_POWER_MANAGEMENT_3, + 3, 0, right_out_mixer, ARRAY_SIZE(right_out_mixer)), + + SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 4, 0, left_boost_mixer, ARRAY_SIZE(left_boost_mixer)), + SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 5, 0, right_boost_mixer, ARRAY_SIZE(right_boost_mixer)), + + SND_SOC_DAPM_INPUT("AUXL"), + SND_SOC_DAPM_INPUT("AUXR"), +}; + +static const struct snd_soc_dapm_widget wm8758_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Left Output Mixer", WM8985_POWER_MANAGEMENT_3, + 2, 0, left_out_mixer, + ARRAY_SIZE(left_out_mixer) - 1), + SND_SOC_DAPM_MIXER("Right Output Mixer", WM8985_POWER_MANAGEMENT_3, + 3, 0, right_out_mixer, + ARRAY_SIZE(right_out_mixer) - 1), + + SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 4, 0, left_boost_mixer, + ARRAY_SIZE(left_boost_mixer) - 1), + SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8985_POWER_MANAGEMENT_2, + 5, 0, right_boost_mixer, + ARRAY_SIZE(right_boost_mixer) - 1), +}; + +static const struct snd_soc_dapm_route wm8985_common_dapm_routes[] = { { "Right Output Mixer", "PCM Switch", "Right DAC" }, - { "Right Output Mixer", "Aux Switch", "AUXR" }, { "Right Output Mixer", "Line Switch", "Right Boost Mixer" }, { "Left Output Mixer", "PCM Switch", "Left DAC" }, - { "Left Output Mixer", "Aux Switch", "AUXL" }, { "Left Output Mixer", "Line Switch", "Left Boost Mixer" }, { "Right Headphone Out", NULL, "Right Output Mixer" }, @@ -495,13 +531,11 @@ static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { { "Right ADC", NULL, "Right Boost Mixer" }, - { "Right Boost Mixer", "AUXR Volume", "AUXR" }, { "Right Boost Mixer", NULL, "Right Capture PGA" }, { "Right Boost Mixer", "R2 Volume", "R2" }, { "Left ADC", NULL, "Left Boost Mixer" }, - { "Left Boost Mixer", "AUXL Volume", "AUXL" }, { "Left Boost Mixer", NULL, "Left Capture PGA" }, { "Left Boost Mixer", "L2 Volume", "L2" }, @@ -516,6 +550,38 @@ static const struct snd_soc_dapm_route wm8985_dapm_routes[] = { { "Left Input Mixer", "MicN Switch", "LIN" }, { "Left Input Mixer", "MicP Switch", "LIP" }, }; +static const struct snd_soc_dapm_route wm8985_aux_dapm_routes[] = { + { "Right Output Mixer", "Aux Switch", "AUXR" }, + { "Left Output Mixer", "Aux Switch", "AUXL" }, + + { "Right Boost Mixer", "AUXR Volume", "AUXR" }, + { "Left Boost Mixer", "AUXL Volume", "AUXL" }, +}; + +static int wm8985_add_widgets(struct snd_soc_codec *codec) +{ + struct wm8985_priv *wm8985 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + + switch (wm8985->dev_type) { + case WM8758: + snd_soc_dapm_new_controls(dapm, wm8758_dapm_widgets, + ARRAY_SIZE(wm8758_dapm_widgets)); + break; + + case WM8985: + snd_soc_add_codec_controls(codec, wm8985_specific_snd_controls, + ARRAY_SIZE(wm8985_specific_snd_controls)); + + snd_soc_dapm_new_controls(dapm, wm8985_dapm_widgets, + ARRAY_SIZE(wm8985_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, wm8985_aux_dapm_routes, + ARRAY_SIZE(wm8985_aux_dapm_routes)); + break; + } + + return 0; +} static int eqmode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1005,6 +1071,8 @@ static int wm8985_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8985_BIAS_CTRL, WM8985_BIASCUT, WM8985_BIASCUT); + wm8985_add_widgets(codec); + return 0; err_reg_enable: @@ -1048,12 +1116,12 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8985 = { .set_bias_level = wm8985_set_bias_level, .suspend_bias_off = true, - .controls = wm8985_snd_controls, - .num_controls = ARRAY_SIZE(wm8985_snd_controls), - .dapm_widgets = wm8985_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(wm8985_dapm_widgets), - .dapm_routes = wm8985_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(wm8985_dapm_routes), + .controls = wm8985_common_snd_controls, + .num_controls = ARRAY_SIZE(wm8985_common_snd_controls), + .dapm_widgets = wm8985_common_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm8985_common_dapm_widgets), + .dapm_routes = wm8985_common_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm8985_common_dapm_routes), }; static const struct regmap_config wm8985_regmap = { @@ -1080,6 +1148,8 @@ static int wm8985_spi_probe(struct spi_device *spi) spi_set_drvdata(spi, wm8985); + wm8985->dev_type = WM8985; + wm8985->regmap = devm_regmap_init_spi(spi, &wm8985_regmap); if (IS_ERR(wm8985->regmap)) { ret = PTR_ERR(wm8985->regmap); @@ -1121,6 +1191,8 @@ static int wm8985_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8985); + wm8985->dev_type = id->driver_data; + wm8985->regmap = devm_regmap_init_i2c(i2c, &wm8985_regmap); if (IS_ERR(wm8985->regmap)) { ret = PTR_ERR(wm8985->regmap); @@ -1141,7 +1213,8 @@ static int wm8985_i2c_remove(struct i2c_client *i2c) } static const struct i2c_device_id wm8985_i2c_id[] = { - { "wm8985", 0 }, + { "wm8985", WM8985 }, + { "wm8758", WM8758 }, { } }; MODULE_DEVICE_TABLE(i2c, wm8985_i2c_id); @@ -1189,6 +1262,6 @@ static void __exit wm8985_exit(void) } module_exit(wm8985_exit); -MODULE_DESCRIPTION("ASoC WM8985 driver"); +MODULE_DESCRIPTION("ASoC WM8985 / WM8758 driver"); MODULE_AUTHOR("Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); -- 1.9.1

3 4

[alsa-devel] [RESEND PATCH v2] ASoC: rockchip: Add machine driver for MAX98357A/RT5514/DA7219
by Xing Zheng 26 May '16

26 May '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 v2: - use the FS 256 to set mclks of the max98357a and rt5514 danamically - add more sample rate for da7219 .../sound/rockchip-max98357a-rt5514-da7219.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + .../rockchip/rockchip_max98357a_rt5514_da7219.c | 333 ++++++++++++++++++++ 4 files changed, 361 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt create mode 100644 sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt new file mode 100644 index 0000000..5928384 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs + +Required properties: +- compatible: "rockchip,rockchip-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,rockchip-sound"; + rockchip,cpu = <&i2s0 &i2s0 &i2s1>; + rockchip,codec = <&max98357a &headsetcodec &codec>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..d02aa94 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_ROCKCHIP_MAX98357A_RT5514_DA7219 + tristate "ASoC support for Rockchip 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 + boards using the MAX98357A/RT5514/DA7219. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..ca97489 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rockchip-max98357a-rt5514-da7219-objs := rockchip_max98357a_rt5514_da7219.o obj-$(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_ROCKCHIP_MAX98357A_RT5514_DA7219) += snd-soc-rockchip-max98357a-rt5514-da7219.o diff --git a/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c new file mode 100644 index 0000000..818c47e --- /dev/null +++ b/sound/soc/rockchip/rockchip_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 "rockchip-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; + + /* 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 = "rockchip-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; + + 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,rockchip-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

2 2

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: Make the register values human readable
by Peter Ujfalusi 26 May '16

26 May '16

The datasheet uses decimal numbers for the register addresses, convert the register values from hexadecimal to decimal and introduce macro for the register definitions. This way it is easier to look up registers in the documentation. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic31xx.h | 134 ++++++++++++++++++++------------------- 1 file changed, 68 insertions(+), 66 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index fe16c34607bb..ac9b146526eb 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -38,141 +38,143 @@ struct aic31xx_pdata { int micbias_vg; }; +#define AIC31XX_REG(page, reg) ((page * 128) + reg) + /* Page Control Register */ -#define AIC31XX_PAGECTL 0x00 +#define AIC31XX_PAGECTL AIC31XX_REG(0, 0) /* Page 0 Registers */ /* Software reset register */ -#define AIC31XX_RESET 0x01 +#define AIC31XX_RESET AIC31XX_REG(0, 1) /* OT FLAG register */ -#define AIC31XX_OT_FLAG 0x03 +#define AIC31XX_OT_FLAG AIC31XX_REG(0, 3) /* Clock clock Gen muxing, Multiplexers*/ -#define AIC31XX_CLKMUX 0x04 +#define AIC31XX_CLKMUX AIC31XX_REG(0, 4) /* PLL P and R-VAL register */ -#define AIC31XX_PLLPR 0x05 +#define AIC31XX_PLLPR AIC31XX_REG(0, 5) /* PLL J-VAL register */ -#define AIC31XX_PLLJ 0x06 +#define AIC31XX_PLLJ AIC31XX_REG(0, 6) /* PLL D-VAL MSB register */ -#define AIC31XX_PLLDMSB 0x07 +#define AIC31XX_PLLDMSB AIC31XX_REG(0, 7) /* PLL D-VAL LSB register */ -#define AIC31XX_PLLDLSB 0x08 +#define AIC31XX_PLLDLSB AIC31XX_REG(0, 8) /* DAC NDAC_VAL register*/ -#define AIC31XX_NDAC 0x0B +#define AIC31XX_NDAC AIC31XX_REG(0, 11) /* DAC MDAC_VAL register */ -#define AIC31XX_MDAC 0x0C +#define AIC31XX_MDAC AIC31XX_REG(0, 12) /* DAC OSR setting register 1, MSB value */ -#define AIC31XX_DOSRMSB 0x0D +#define AIC31XX_DOSRMSB AIC31XX_REG(0, 13) /* DAC OSR setting register 2, LSB value */ -#define AIC31XX_DOSRLSB 0x0E -#define AIC31XX_MINI_DSP_INPOL 0x10 +#define AIC31XX_DOSRLSB AIC31XX_REG(0, 14) +#define AIC31XX_MINI_DSP_INPOL AIC31XX_REG(0, 16) /* Clock setting register 8, PLL */ -#define AIC31XX_NADC 0x12 +#define AIC31XX_NADC AIC31XX_REG(0, 18) /* Clock setting register 9, PLL */ -#define AIC31XX_MADC 0x13 +#define AIC31XX_MADC AIC31XX_REG(0, 19) /* ADC Oversampling (AOSR) Register */ -#define AIC31XX_AOSR 0x14 +#define AIC31XX_AOSR AIC31XX_REG(0, 20) /* Clock setting register 9, Multiplexers */ -#define AIC31XX_CLKOUTMUX 0x19 +#define AIC31XX_CLKOUTMUX AIC31XX_REG(0, 25) /* Clock setting register 10, CLOCKOUT M divider value */ -#define AIC31XX_CLKOUTMVAL 0x1A +#define AIC31XX_CLKOUTMVAL AIC31XX_REG(0, 26) /* Audio Interface Setting Register 1 */ -#define AIC31XX_IFACE1 0x1B +#define AIC31XX_IFACE1 AIC31XX_REG(0, 27) /* Audio Data Slot Offset Programming */ -#define AIC31XX_DATA_OFFSET 0x1C +#define AIC31XX_DATA_OFFSET AIC31XX_REG(0, 28) /* Audio Interface Setting Register 2 */ -#define AIC31XX_IFACE2 0x1D +#define AIC31XX_IFACE2 AIC31XX_REG(0, 29) /* Clock setting register 11, BCLK N Divider */ -#define AIC31XX_BCLKN 0x1E +#define AIC31XX_BCLKN AIC31XX_REG(0, 30) /* Audio Interface Setting Register 3, Secondary Audio Interface */ -#define AIC31XX_IFACESEC1 0x1F +#define AIC31XX_IFACESEC1 AIC31XX_REG(0, 31) /* Audio Interface Setting Register 4 */ -#define AIC31XX_IFACESEC2 0x20 +#define AIC31XX_IFACESEC2 AIC31XX_REG(0, 32) /* Audio Interface Setting Register 5 */ -#define AIC31XX_IFACESEC3 0x21 +#define AIC31XX_IFACESEC3 AIC31XX_REG(0, 33) /* I2C Bus Condition */ -#define AIC31XX_I2C 0x22 +#define AIC31XX_I2C AIC31XX_REG(0, 34) /* ADC FLAG */ -#define AIC31XX_ADCFLAG 0x24 +#define AIC31XX_ADCFLAG AIC31XX_REG(0, 36) /* DAC Flag Registers */ -#define AIC31XX_DACFLAG1 0x25 -#define AIC31XX_DACFLAG2 0x26 +#define AIC31XX_DACFLAG1 AIC31XX_REG(0, 37) +#define AIC31XX_DACFLAG2 AIC31XX_REG(0, 38) /* Sticky Interrupt flag (overflow) */ -#define AIC31XX_OFFLAG 0x27 +#define AIC31XX_OFFLAG AIC31XX_REG(0, 39) /* Sticy DAC Interrupt flags */ -#define AIC31XX_INTRDACFLAG 0x2C +#define AIC31XX_INTRDACFLAG AIC31XX_REG(0, 44) /* Sticy ADC Interrupt flags */ -#define AIC31XX_INTRADCFLAG 0x2D +#define AIC31XX_INTRADCFLAG AIC31XX_REG(0, 45) /* DAC Interrupt flags 2 */ -#define AIC31XX_INTRDACFLAG2 0x2E +#define AIC31XX_INTRDACFLAG2 AIC31XX_REG(0, 46) /* ADC Interrupt flags 2 */ -#define AIC31XX_INTRADCFLAG2 0x2F +#define AIC31XX_INTRADCFLAG2 AIC31XX_REG(0, 47) /* INT1 interrupt control */ -#define AIC31XX_INT1CTRL 0x30 +#define AIC31XX_INT1CTRL AIC31XX_REG(0, 48) /* INT2 interrupt control */ -#define AIC31XX_INT2CTRL 0x31 +#define AIC31XX_INT2CTRL AIC31XX_REG(0, 49) /* GPIO1 control */ -#define AIC31XX_GPIO1 0x33 +#define AIC31XX_GPIO1 AIC31XX_REG(0, 50) -#define AIC31XX_DACPRB 0x3C +#define AIC31XX_DACPRB AIC31XX_REG(0, 60) /* ADC Instruction Set Register */ -#define AIC31XX_ADCPRB 0x3D +#define AIC31XX_ADCPRB AIC31XX_REG(0, 61) /* DAC channel setup register */ -#define AIC31XX_DACSETUP 0x3F +#define AIC31XX_DACSETUP AIC31XX_REG(0, 63) /* DAC Mute and volume control register */ -#define AIC31XX_DACMUTE 0x40 +#define AIC31XX_DACMUTE AIC31XX_REG(0, 64) /* Left DAC channel digital volume control */ -#define AIC31XX_LDACVOL 0x41 +#define AIC31XX_LDACVOL AIC31XX_REG(0, 65) /* Right DAC channel digital volume control */ -#define AIC31XX_RDACVOL 0x42 +#define AIC31XX_RDACVOL AIC31XX_REG(0, 66) /* Headset detection */ -#define AIC31XX_HSDETECT 0x43 +#define AIC31XX_HSDETECT AIC31XX_REG(0, 67) /* ADC Digital Mic */ -#define AIC31XX_ADCSETUP 0x51 +#define AIC31XX_ADCSETUP AIC31XX_REG(0, 81) /* ADC Digital Volume Control Fine Adjust */ -#define AIC31XX_ADCFGA 0x52 +#define AIC31XX_ADCFGA AIC31XX_REG(0, 82) /* ADC Digital Volume Control Coarse Adjust */ -#define AIC31XX_ADCVOL 0x53 +#define AIC31XX_ADCVOL AIC31XX_REG(0, 83) /* Page 1 Registers */ /* Headphone drivers */ -#define AIC31XX_HPDRIVER 0x9F +#define AIC31XX_HPDRIVER AIC31XX_REG(1, 31) /* Class-D Speakear Amplifier */ -#define AIC31XX_SPKAMP 0xA0 +#define AIC31XX_SPKAMP AIC31XX_REG(1, 32) /* HP Output Drivers POP Removal Settings */ -#define AIC31XX_HPPOP 0xA1 +#define AIC31XX_HPPOP AIC31XX_REG(1, 33) /* Output Driver PGA Ramp-Down Period Control */ -#define AIC31XX_SPPGARAMP 0xA2 +#define AIC31XX_SPPGARAMP AIC31XX_REG(1, 34) /* DAC_L and DAC_R Output Mixer Routing */ -#define AIC31XX_DACMIXERROUTE 0xA3 +#define AIC31XX_DACMIXERROUTE AIC31XX_REG(1, 35) /* Left Analog Vol to HPL */ -#define AIC31XX_LANALOGHPL 0xA4 +#define AIC31XX_LANALOGHPL AIC31XX_REG(1, 36) /* Right Analog Vol to HPR */ -#define AIC31XX_RANALOGHPR 0xA5 +#define AIC31XX_RANALOGHPR AIC31XX_REG(1, 37) /* Left Analog Vol to SPL */ -#define AIC31XX_LANALOGSPL 0xA6 +#define AIC31XX_LANALOGSPL AIC31XX_REG(1, 38) /* Right Analog Vol to SPR */ -#define AIC31XX_RANALOGSPR 0xA7 +#define AIC31XX_RANALOGSPR AIC31XX_REG(1, 39) /* HPL Driver */ -#define AIC31XX_HPLGAIN 0xA8 +#define AIC31XX_HPLGAIN AIC31XX_REG(1, 40) /* HPR Driver */ -#define AIC31XX_HPRGAIN 0xA9 +#define AIC31XX_HPRGAIN AIC31XX_REG(1, 41) /* SPL Driver */ -#define AIC31XX_SPLGAIN 0xAA +#define AIC31XX_SPLGAIN AIC31XX_REG(1, 42) /* SPR Driver */ -#define AIC31XX_SPRGAIN 0xAB +#define AIC31XX_SPRGAIN AIC31XX_REG(1, 43) /* HP Driver Control */ -#define AIC31XX_HPCONTROL 0xAC +#define AIC31XX_HPCONTROL AIC31XX_REG(1, 44) /* MIC Bias Control */ -#define AIC31XX_MICBIAS 0xAE +#define AIC31XX_MICBIAS AIC31XX_REG(1, 46) /* MIC PGA*/ -#define AIC31XX_MICPGA 0xAF +#define AIC31XX_MICPGA AIC31XX_REG(1, 47) /* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ -#define AIC31XX_MICPGAPI 0xB0 +#define AIC31XX_MICPGAPI AIC31XX_REG(1, 48) /* ADC Input Selection for M-Terminal */ -#define AIC31XX_MICPGAMI 0xB1 +#define AIC31XX_MICPGAMI AIC31XX_REG(1, 49) /* Input CM Settings */ -#define AIC31XX_MICPGACM 0xB2 +#define AIC31XX_MICPGACM AIC31XX_REG(1, 50) /* Bits, masks and shifts */ -- 2.8.3

2 1

[alsa-devel] [PATCH v6] ASoC: cs53l30: Add codec driver support for Cirrus CS53L30
by Nicolin Chen 25 May '16

25 May '16

CS53L30 is a Quad-Channel ADC from Cirrus Logic with an I2S/TDM DAI. So this patch adds support for CS53L30 that supports 24-bit recording feature. Signed-off-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Changelog: 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 | 38 + 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, 1601 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..ace7ffe --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs53l30.txt @@ -0,0 +1,38 @@ +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. + + - micbias-lvl : Set the output voltage level on the MICBIAS Pin. + 0 = Hi-Z + 1 = 1.80 V + 2 = 2.75 V + + - 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>; +}; 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..ae71859 --- /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 <sound/pcm_params.h> +#include <linux/regulator/consumer.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 request 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, "micbias-lvl", &val)) + regmap_update_bits(cs53l30->regmap, CS53L30_MICBIAS_CTL, + CS53L30_MIC_BIAS_CTRL_MASK, val); + + if (of_property_read_bool(np, "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 4

[alsa-devel] [RFC-PATCH 0/2] ASoC: simple_card: support for hw-params rules
by kernel＠martin.sperl.org 25 May '16

25 May '16

From: Martin Sperl <kernel(a)martin.sperl.org> Simple_card does require under some circumstances the ability to configure certain hw_parameters based on clocks, bits, channels. This patchset adds a generic way to configure these kind of things via the device tree easily. This patchset implements this for simple_card, but other drivers can just as easily make use of this. For now we have the following matchers and actions: * matchers: * match_sample_bits * match_rate * match_channels * actions: * set_fixed_bclk_size As a note: the available matching rules and action rules right now are hard-coded, but this could in principle get extended to be more dynamic via kallsyms_lookup_name that would lookup the requested symbol and assume it is a struct asoc_generic_hw_params_method, on which it could apply several sanity-checks before using the pointers for real. This would allow other kinds of action methods to get exposed in codec-drivers, that can then get used to handle some settings specific to the codec. I leave this is an exercise for later... Martin Sperl (2): ASoC: hw-params-rules: add generic hw_params-rules ASoC: simple_card: add support for hw_params_rules .../devicetree/bindings/sound/hw-params-rules.txt | 86 +++++ .../devicetree/bindings/sound/simple-card.txt | 1 + sound/soc/generic/Kconfig | 6 + sound/soc/generic/Makefile | 2 + sound/soc/generic/hw-params-rules.c | 402 +++++++++++++++++++++ sound/soc/generic/hw-params-rules.h | 49 +++ sound/soc/generic/simple-card.c | 11 +- 7 files changed, 554 insertions(+), 3 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/hw-params-rules.txt create mode 100644 sound/soc/generic/hw-params-rules.c create mode 100644 sound/soc/generic/hw-params-rules.h -- 2.1.4

6 11

[alsa-devel] [PATCH 1/2] ASoC: pcm5102a: Remove owner assignment from platform_driver
by Fabio Estevam 25 May '16

25 May '16

This platform_driver does not need to set an owner as it will be populated by the driver core. Generated by scripts/coccinelle/api/platform_no_drv_owner.cocci. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/pcm5102a.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/pcm5102a.c b/sound/soc/codecs/pcm5102a.c index ed51567..8ba322a 100644 --- a/sound/soc/codecs/pcm5102a.c +++ b/sound/soc/codecs/pcm5102a.c @@ -57,7 +57,6 @@ static struct platform_driver pcm5102a_codec_driver = { .remove = pcm5102a_remove, .driver = { .name = "pcm5102a-codec", - .owner = THIS_MODULE, .of_match_table = pcm5102a_of_match, }, }; -- 1.9.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Fix headset mic detection problem for one Dell machine
by Hui Wang 25 May '16

25 May '16

Add the pin configuration value of this machine into the pin_quirk table to make DELL1_MIC_NO_PRESENCE apply to this machine. Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a7f7df1..d53c25e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5720,6 +5720,9 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x14, 0x90170110}, {0x21, 0x02211020}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, + {0x14, 0x90170130}, + {0x21, 0x02211040}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, {0x12, 0x90a60140}, {0x14, 0x90170110}, {0x21, 0x02211020}), -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: bcm2835: add support for hifiberry-dac
by kernel＠martin.sperl.org 25 May '16

25 May '16

From: Martin Sperl <kernel(a)martin.sperl.org> Add support for the hifiberry-dac hat that can get attached to a raspberry pi. This hat also the ti-pcm5102a DAC, so support for this DAC has been added to codecs. This is a backport of the drivers in the rpi-downstream kernel. Changelog: * Drivers written by: Florian Meier <florian.meier(a)koalo.de> * ref-count patch by Matthias Reichl <hias(a)horus.com> * rebased, checkpath fixed and dt-binding documentation added by Martin Sperl <kernel(a)martin.sperl.org> Florian Meier (2): ASoC: pcm5102a: Add support for PCM5102A codec ASoC: Add support for HifiBerry DAC .../bindings/sound/hifiberry,hifiberry-dac.txt | 12 ++ .../devicetree/bindings/sound/pcm5102a.txt | 13 +++ sound/soc/bcm/Kconfig | 7 ++ sound/soc/bcm/Makefile | 3 +- sound/soc/bcm/hifiberry_dac.c | 126 +++++++++++++++++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm5102a.c | 69 +++++++++++ 8 files changed, 235 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/hifiberry,hifiberry-dac.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt create mode 100644 sound/soc/bcm/hifiberry_dac.c create mode 100644 sound/soc/codecs/pcm5102a.c -- 2.1.4

6 14

[alsa-devel] [PATCH] ASoC: rockchip: i2s: rename I2S_CKR_TRCM_TX/RXSHARE to I2S_CKR_TRCM_TX/RXONLY
by Xing Zheng 24 May '16

24 May '16

From: Sugar Zhang <sugar.zhang(a)rock-chips.com> this patch make it more reasonable and readable, because when we chose I2S_CKR_TRCM_TXONLY, we only output clk_lrck_tx, and hardware need to confirm this signal is wired to external codec lrck_tx/rx at the same time. for convenience, we just handle lrck_txonly if we enable symmetric_rates in driver and dai_link. otherwise, we use the separate lrck_tx/rx. Signed-off-by: Sugar Zhang <sugar.zhang(a)rock-chips.com> Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> --- sound/soc/rockchip/rockchip_i2s.c | 4 ++-- sound/soc/rockchip/rockchip_i2s.h | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 574c6af..92a405f 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -306,8 +306,8 @@ static int rockchip_i2s_hw_params(struct snd_pcm_substream *substream, I2S_DMACR_RDL(16)); val = I2S_CKR_TRCM_TXRX; - if (dai->driver->symmetric_rates || rtd->dai_link->symmetric_rates) - val = I2S_CKR_TRCM_TXSHARE; + if (dai->driver->symmetric_rates && rtd->dai_link->symmetric_rates) + val = I2S_CKR_TRCM_TXONLY; regmap_update_bits(i2s->regmap, I2S_CKR, I2S_CKR_TRCM_MASK, diff --git a/sound/soc/rockchip/rockchip_i2s.h b/sound/soc/rockchip/rockchip_i2s.h index dc6e2c7..13b1d44 100644 --- a/sound/soc/rockchip/rockchip_i2s.h +++ b/sound/soc/rockchip/rockchip_i2s.h @@ -81,8 +81,8 @@ #define I2S_CKR_TRCM_SHIFT 28 #define I2S_CKR_TRCM(x) (x << I2S_CKR_TRCM_SHIFT) #define I2S_CKR_TRCM_TXRX (0 << I2S_CKR_TRCM_SHIFT) -#define I2S_CKR_TRCM_TXSHARE (1 << I2S_CKR_TRCM_SHIFT) -#define I2S_CKR_TRCM_RXSHARE (2 << I2S_CKR_TRCM_SHIFT) +#define I2S_CKR_TRCM_TXONLY (1 << I2S_CKR_TRCM_SHIFT) +#define I2S_CKR_TRCM_RXONLY (2 << I2S_CKR_TRCM_SHIFT) #define I2S_CKR_TRCM_MASK (3 << I2S_CKR_TRCM_SHIFT) #define I2S_CKR_MSS_SHIFT 27 #define I2S_CKR_MSS_MASTER (0 << I2S_CKR_MSS_SHIFT) -- 1.7.9.5

2 1

[alsa-devel] Applied "MAINTAINERS: ASoC: samsung: Add Sylwester Nawrocki and Krzysztof Kozlowski" to the asoc tree
by Mark Brown 24 May '16

24 May '16

The patch MAINTAINERS: ASoC: samsung: Add Sylwester Nawrocki and Krzysztof Kozlowski has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From eea5495bab0d365f4dca04869b21e557a9df48d5 Mon Sep 17 00:00:00 2001 From: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> Date: Tue, 24 May 2016 15:35:42 +0200 Subject: [PATCH] MAINTAINERS: ASoC: samsung: Add Sylwester Nawrocki and Krzysztof Kozlowski Extend maintainer entry for Samsung SoC sound drivers with Krzysztof Kozlowski and Sylwester Nawrocki. The file pattern is duplicated in main Exynos ARM section so remove it from there. Cc: Kukjin Kim <kgene(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-samsung-soc(a)vger.kernel.org Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 03e00c7c88eb..45dd1db87cff 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1542,7 +1542,6 @@ F: drivers/*/*s3c2410* F: drivers/*/*/*s3c2410* F: drivers/soc/samsung/* F: drivers/spi/spi-s3c* -F: sound/soc/samsung/* F: Documentation/arm/Samsung/ F: Documentation/devicetree/bindings/arm/samsung/ F: Documentation/devicetree/bindings/sram/samsung-sram.txt @@ -9601,7 +9600,9 @@ S: Maintained F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS +M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> M: Sangbeom Kim <sbkim73(a)samsung.com> +M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung/ -- 2.8.1

1 0

[alsa-devel] [PATCH]: ASoC: intel: make function stub static
by Randy Dunlap 24 May '16

24 May '16

From: Randy Dunlap <rdunlap(a)infradead.org> This function stub should have been 'static' in the original patch so that multiple uses of the header file (in different drivers) will not cause multiple function definitions. sound/soc/intel/boards/built-in.o: In function `sst_acpi_find_name_from_hid': (.text+0x560): multiple definition of `sst_acpi_find_name_from_hid' sound/soc/intel/atom/built-in.o:(.text+0x10610): first defined here ../scripts/Makefile.build:369: recipe for target 'sound/soc/intel/built-in.o' failed Fixes: f17131a93f43: add function stub when ACPI is not enabled Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Borislav Petkov <bp(a)alien8.de> Cc: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/common/sst-acpi.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20160520.orig/sound/soc/intel/common/sst-acpi.h +++ linux-next-20160520/sound/soc/intel/common/sst-acpi.h @@ -20,7 +20,7 @@ #if IS_ENABLED(CONFIG_ACPI) const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]); #else -inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) +static inline const char *sst_acpi_find_name_from_hid(const u8 hid[ACPI_ID_LEN]) { return NULL; }

2 1

Re: [alsa-devel] Add support for ALC295
by Takashi Iwai 24 May '16

24 May '16

On Tue, 24 May 2016 10:53:37 +0200, Kailang wrote: > > Hi Takashi, > > We have a new codec ALC295 that is brother and sister with ALC225. > Attach is the support patch. Applied now. Thanks. Takashi

1 0

[alsa-devel] [PATCH] MAINTAINERS: ASoC: samsung: Add Sylwester Nawrocki and Krzysztof Kozlowski
by Krzysztof Kozlowski 24 May '16

24 May '16

Extend maintainer entry for Samsung SoC sound drivers with Krzysztof Kozlowski and Sylwester Nawrocki. The file pattern is duplicated in main Exynos ARM section so remove it from there. Cc: Kukjin Kim <kgene(a)kernel.org> Cc: Sangbeom Kim <sbkim73(a)samsung.com> Cc: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org Cc: linux-samsung-soc(a)vger.kernel.org Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- MAINTAINERS | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 243c0811a4d8..f0f4ca58d264 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1603,7 +1603,6 @@ F: drivers/*/*/*s3c2410* F: drivers/memory/samsung/* F: drivers/soc/samsung/* F: drivers/spi/spi-s3c* -F: sound/soc/samsung/* F: Documentation/arm/Samsung/ F: Documentation/devicetree/bindings/arm/samsung/ F: Documentation/devicetree/bindings/sram/samsung-sram.txt @@ -9857,7 +9856,9 @@ S: Maintained F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS +M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> M: Sangbeom Kim <sbkim73(a)samsung.com> +M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung/ -- 1.9.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: da7219: Convert driver to use generic FW functions
by Adam Thomson 24 May '16

24 May '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 v4.6 kernel. 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

3 5

[alsa-devel] [PATCH v2 1/2] ASoC: wm8985: add device tree binding for WM8985
by Petr Kulhavy 24 May '16

24 May '16

Add device tree binding for the WM8985 codec driver. Signed-off-by: Petr Kulhavy <petr(a)barix.com> Acked-by: Rob Herring <robh(a)kernel.org> --- v1: initial v2: no change Documentation/devicetree/bindings/sound/wm8985.txt | 36 ++++++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8985.txt diff --git a/Documentation/devicetree/bindings/sound/wm8985.txt b/Documentation/devicetree/bindings/sound/wm8985.txt new file mode 100644 index 000000000000..788d64766257 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8985.txt @@ -0,0 +1,36 @@ +Wolfson Microelectronics WM8985 and WM8758 audio codecs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8985" or "wlf,wm8758" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Pins on the device for linking into audio routes: + + * LIN : left microphone pre-amp negative input + * LIP : left microphone pre-amp positive input + * RIN : right microphone pre-amp negative input + * RIP : right microphone pre-amp positive input + * L2 : left line input / secondary pre-amp positive input + * R2 : right line input / secondary pre-amp positive input + * AUXL : left auxiliary input (WM8985 only) + * AUXR : right auxiliary input (WM8985 only) + + * HPL : left headphone / line output (the LOUT1 pin) + * HPR : right headphone / line output (the ROUT1 pin) + * SPKL : left headphone / line output (the LOUT2 pin) + * SPKR : right headphone / line output (the ROUT2 pin) + * Mic Bias : microphone bias + + +Example: + +wm8758: audio-codec@1a { + compatible = "wlf,wm8758"; + reg = <0x1a>; +}; -- 1.9.1

2 7

[alsa-devel] [PATCH v3 0/2] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 24 May '16

24 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch set adds the WM8758 support into the WM8985 driver. Changes in v3: * merge the wm8985_xxx_snd_controls into wm8985_common_snd_controls and wm8985_specific_snd_controls in order to reduce the number of structures use the static initialization in snd_soc_codec_driver for the common part

2 4

[alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by Chuang Eric 24 May '16

24 May '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

1 0

[alsa-devel] DPCM: skip DAPM_STREAM_STOP event to BE, if still used by other FE
by srv_kaichieh.chaung 24 May '16

24 May '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

1 0

[alsa-devel] [PATCH v2] ASoC: rockchip: Add machine driver for MAX98357A/RT5514/DA7219
by Xing Zheng 24 May '16

24 May '16

There are multi codec devices on the RK3399 platform, we can use this patch support and control these codecs. --- Changes in v2: Signed-off-by: Xing Zheng <zhengxing(a)rock-chips.com> .../sound/rockchip-max98357a-rt5514-da7219.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + .../rockchip/rockchip_max98357a_rt5514_da7219.c | 333 ++++++++++++++++++++ 4 files changed, 361 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt create mode 100644 sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt new file mode 100644 index 0000000..5928384 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs + +Required properties: +- compatible: "rockchip,rockchip-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,rockchip-sound"; + rockchip,cpu = <&i2s0 &i2s0 &i2s1>; + rockchip,codec = <&max98357a &headsetcodec &codec>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..d02aa94 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_ROCKCHIP_MAX98357A_RT5514_DA7219 + tristate "ASoC support for Rockchip 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 + boards using the MAX98357A/RT5514/DA7219. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..ca97489 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rockchip-max98357a-rt5514-da7219-objs := rockchip_max98357a_rt5514_da7219.o obj-$(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_ROCKCHIP_MAX98357A_RT5514_DA7219) += snd-soc-rockchip-max98357a-rt5514-da7219.o diff --git a/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c new file mode 100644 index 0000000..818c47e --- /dev/null +++ b/sound/soc/rockchip/rockchip_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 "rockchip-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; + + /* 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 = "rockchip-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; + + 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,rockchip-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

2 2

[alsa-devel] [PATCH 43/54] MAINTAINERS: Add file patterns for sound device tree bindings
by Geert Uytterhoeven 23 May '16

23 May '16

Submitters of device tree binding documentation may forget to CC the subsystem maintainer if this is missing. Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: alsa-devel(a)alsa-project.org --- Please apply this patch directly if you want to be involved in device tree binding documentation for your subsystem. --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 139824579fa97c2c..9d81eff6577ff886 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -10728,6 +10728,7 @@ T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported +F: Documentation/devicetree/bindings/sound/ F: Documentation/sound/alsa/soc/ F: sound/soc/ F: include/sound/soc* -- 1.9.1

2 2

[alsa-devel] [PATCH 0056/1529] Fix typo
by Andrea Gelmini 23 May '16

23 May '16

Signed-off-by: Andrea Gelmini <andrea.gelmini(a)gelma.net> --- Documentation/sound/alsa/soc/machine.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 74056db..6bf2d20 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -3,7 +3,7 @@ ASoC Machine Driver The ASoC machine (or board) driver is the code that glues together all the component drivers (e.g. codecs, platforms and DAIs). It also describes the -relationships between each componnent which include audio paths, GPIOs, +relationships between each component which include audio paths, GPIOs, interrupts, clocking, jacks and voltage regulators. The machine driver can contain codec and platform specific code. It registers -- 2.8.2.534.g1f66975

2 1

[alsa-devel] [PATCH] ASoC: kirkwood: fix build failure
by Sudip Mukherjee 23 May '16

23 May '16

From: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> While building m32r allmodconfig the build failed with: ERROR: "bad_dma_ops" [sound/soc/kirkwood/snd-soc-kirkwood.ko] undefined! To satisfy the dependency CONFIG_SND_KIRKWOOD_SOC should depend on HAS_DMA. Signed-off-by: Sudip Mukherjee <sudip.mukherjee(a)codethink.co.uk> --- build log at: https://travis-ci.org/sudipm-mukherjee/parport/jobs/132180911 sound/soc/kirkwood/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig index 132bb83..bc3c7b5 100644 --- a/sound/soc/kirkwood/Kconfig +++ b/sound/soc/kirkwood/Kconfig @@ -1,6 +1,7 @@ config SND_KIRKWOOD_SOC tristate "SoC Audio for the Marvell Kirkwood and Dove chips" depends on ARCH_DOVE || ARCH_MVEBU || COMPILE_TEST + depends on HAS_DMA help Say Y or M if you want to add support for codecs attached to the Kirkwood I2S interface. You will also need to select the -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 23 May '16

23 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch set adds the WM8758 support into the WM8985 driver.

2 4

[alsa-devel] [PATCH v2 0/2] ASoC: wm8985: add support for WM8758
by Petr Kulhavy 23 May '16

23 May '16

The WM8758 chip is almost identical to WM8985 with the difference that it doesn't feature the AUX input. This patch set adds the WM8758 support into the WM8985 driver. Changes in v2: * use chip type enum instead of chip structure * do not copy the complete controls, widget, routes but only the differences and add widgets dynamically using add_widgets()

2 7

[alsa-devel] [PATCH] ASoC: rockchip: Add machine driver for MAX98357A/RT5514/DA7219
by Xing Zheng 23 May '16

23 May '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> --- .../sound/rockchip-max98357a-rt5514-da7219.txt | 15 + sound/soc/rockchip/Kconfig | 11 + sound/soc/rockchip/Makefile | 2 + .../rockchip/rockchip_max98357a_rt5514_da7219.c | 339 ++++++++++++++++++++ 4 files changed, 367 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt create mode 100644 sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c diff --git a/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt new file mode 100644 index 0000000..5928384 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rockchip-max98357a-rt5514-da7219.txt @@ -0,0 +1,15 @@ +ROCKCHIP with MAX98357A/RT5514/DA7219 codecs + +Required properties: +- compatible: "rockchip,rockchip-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,rockchip-sound"; + rockchip,cpu = <&i2s0 &i2s0 &i2s1>; + rockchip,codec = <&max98357a &headsetcodec &codec>; +}; diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index f1e0c70..d02aa94 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_ROCKCHIP_MAX98357A_RT5514_DA7219 + tristate "ASoC support for Rockchip 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 + boards using the MAX98357A/RT5514/DA7219. diff --git a/sound/soc/rockchip/Makefile b/sound/soc/rockchip/Makefile index c0bf560..ca97489 100644 --- a/sound/soc/rockchip/Makefile +++ b/sound/soc/rockchip/Makefile @@ -7,6 +7,8 @@ obj-$(CONFIG_SND_SOC_ROCKCHIP_SPDIF) += snd-soc-rockchip-spdif.o snd-soc-rockchip-max98090-objs := rockchip_max98090.o snd-soc-rockchip-rt5645-objs := rockchip_rt5645.o +snd-soc-rockchip-max98357a-rt5514-da7219-objs := rockchip_max98357a_rt5514_da7219.o obj-$(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_ROCKCHIP_MAX98357A_RT5514_DA7219) += snd-soc-rockchip-max98357a-rt5514-da7219.o diff --git a/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.c new file mode 100644 index 0000000..50c0d35 --- /dev/null +++ b/sound/soc/rockchip/rockchip_max98357a_rt5514_da7219.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 "rockchip-sound" + +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; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 48000: + case 96000: + mclk = 12288000; + break; + default: + 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; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + 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(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; + + switch (params_rate(params)) { + case 8000: + case 16000: + case 32000: + case 48000: + case 96000: + mclk = 12288000; + break; + case 11025: + case 22050: + case 44100: + 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 = "rockchip-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; + + 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,rockchip-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

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[alsa-devel] journalctl suggests I report Bug to ALSA developers
by P.W. Pulto 21 May '16

21 May '16

Dear ALSA developers, Having a problem with my Sennheiser DW Pro 2 Dect Headset, I looked in the journalctl. It suggested I report a bug to the ALSA developers. I have no idea if this would be a "kernel driver bug" or an "other ALSA bug". Hopefully I'm at the right address. The problem occurs when I take the headset off the base station and a connection between base station and headset is established. My ALSA-info: ------------------- http://www.alsa-project.org/db/?f=2eaeea040fef7cc2476c239e1fec90dd2bf4eea8 The journalctl output: ----------------------------- journalctl --no-pager --since "25 min ago" May 20 09:05:34 studiopulto-ubuntu-1 pulseaudio[2098]: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! May 20 09:05:34 studiopulto-ubuntu-1 pulseaudio[2098]: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. May 20 09:05:34 studiopulto-ubuntu-1 pulseaudio[2098]: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Hopefully this is sufficient information. Obviously you welcome to ask for missing info! Kind regards, P.W. Pulto -- Like us on Facebook <https://www.facebook.com/studiopulto> dhr. P.W. Pulto Studio Pulto vzw Jef Vandermeulenstraat 37 9050 Gentbrugge Bélgica tel: 0032 9 220 73 40 e-mail: pwpulto(a)studiopulto.org http://www.studiopulto.org IBAN.: BE03 4487 6576 5184 BIC: KREDBEBB ondern nr.: BE 0470 427 828

1 0

[alsa-devel] [PATCH 1/1] ASoC: pxa: Fix module autoload for platform drivers
by Andrea Adami 21 May '16

21 May '16

These platform drivers are lacking MODULE_ALIAS so module autoloading doesn't work. Tested on corgi and poodle with kernel 4.4. Signed-off-by: Andrea Adami <andrea.adami(a)gmail.com> --- 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 + 8 files changed, 8 insertions(+) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index ec522e9..b6cb995 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -133,3 +133,4 @@ module_platform_driver(mmp_driver); MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); MODULE_DESCRIPTION("ALSA SoC Brownstone"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:brownstone-audio"); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index 5c8f9db..d1661fa 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -207,3 +207,4 @@ module_platform_driver(mioa701_wm9713_driver); MODULE_AUTHOR("Robert Jarzmik (rjarzmik(a)free.fr)"); MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mioa701-wm9713"); diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c index 51e790d..96df9b2 100644 --- a/sound/soc/pxa/mmp-pcm.c +++ b/sound/soc/pxa/mmp-pcm.c @@ -248,3 +248,4 @@ module_platform_driver(mmp_pcm_driver); MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); MODULE_DESCRIPTION("MMP Soc Audio DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-pcm-audio"); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index eca60c2..ca8b23f 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -482,3 +482,4 @@ module_platform_driver(asoc_mmp_sspa_driver); MODULE_AUTHOR("Leo Yan <leoy(a)marvell.com>"); MODULE_DESCRIPTION("MMP SSPA SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:mmp-sspa-dai"); diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index 4e74d95..bcc81e9 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -161,3 +161,4 @@ module_platform_driver(palm27x_wm9712_driver); MODULE_AUTHOR("Marek Vasut <marek.vasut(a)gmail.com>"); MODULE_DESCRIPTION("ALSA SoC Palm T|X, T5 and LifeDrive"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:palm27x-asoc"); diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index da03fad..3cad990 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -833,3 +833,4 @@ module_platform_driver(asoc_ssp_driver); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_DESCRIPTION("PXA SSP/PCM SoC Interface"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-ssp-dai"); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f3de615..9615e6d 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -287,3 +287,4 @@ module_platform_driver(pxa2xx_ac97_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("AC97 driver for the Intel PXA2xx chip"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa2xx-ac97"); diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 9f39039..410d48b 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -117,3 +117,4 @@ module_platform_driver(pxa_pcm_driver); MODULE_AUTHOR("Nicolas Pitre"); MODULE_DESCRIPTION("Intel PXA2xx PCM DMA module"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:pxa-pcm-audio"); -- 1.9.1

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Re: [alsa-devel] [PATCH v7 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Scott Branden 20 May '16

20 May '16

Looks good Simran. Mark, anything else for us to do before this driver can be accepted upstream? On 16-05-17 12:46 PM, Simran Rai wrote: > From: Simran <ssimran(a)lbrmn-ssimran-lnxub64.ric.broadcom.com> > > 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-v7 > > Changes from v6: > - DT bindings acknowledged by Rob Herring > 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 >

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headphone noise on Dell XPS 13 9360
by Kai-Heng Feng 20 May '16

20 May '16

The headphone has noise when playing sound or switching microphone sources. It uses the same codec on XPS 13 9350, but with different subsystem ID. Applying the fixup can solve the issue. Also, changing the model name to better differentiate models. v2: Reorder by device ID. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 002f153..10ca900 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5466,8 +5466,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.8.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix headphone noise on Dell XPS 13 9360
by Kai-Heng Feng 20 May '16

20 May '16

The headphone has noise when playing sound or switching microphone sources. It uses the same codec on XPS 13 9350, but with different subsystem ID. Applying the fixup can solve the issue. Also, changing the model name to better differentiate models. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 002f153..e6af865 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5466,7 +5466,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC293_FIXUP_DISABLE_AAMIX_MULTIJACK), - SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x0704, "Dell XPS 13 9350", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), + SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), -- 2.8.1

2 1

[alsa-devel] [PATCH v4 0/7] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 19 May '16

19 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 4 of the patch: * minor correction in the binding example ("interrupt names") * re-added index note to the "mpu" property.

4 18

[alsa-devel] [PATCH 1/1] ASoC: core: Do not print an actual error when deferring probe
by Alexander Stein 19 May '16

19 May '16

In case there is no DAI (yet), do not print an error, this might happen a lot of times. Print a notice instead. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> --- sound/soc/soc-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..352f7c6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1001,7 +1001,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, cpu_dai_component.dai_name = dai_link->cpu_dai_name; rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { - dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); goto _err_defer; } @@ -1013,7 +1013,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); goto _err_defer; } @@ -1042,7 +1042,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->platform = platform; } if (!rtd->platform) { - dev_err(card->dev, "ASoC: platform %s not registered\n", + dev_notice(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } -- 2.7.3

4 12

[alsa-devel] [PATCH] ASoC: ak4642: Enable cache usage to fix crashes on resume
by Mark Brown 19 May '16

19 May '16

The ak4642 driver is using a regmap cache sync to restore the configuration of the chip on resume but (as Peter observed) does not actually define a register cache which means that the resume is never going to work and we trigger asserts in regmap. Fix this by enabling caching. Reported-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reported-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/codecs/ak4642.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1ee8506c06c7..4d8b9e49e8d6 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -560,6 +560,7 @@ static const struct regmap_config ak4642_regmap = { .max_register = FIL1_3, .reg_defaults = ak4642_reg, .num_reg_defaults = NUM_AK4642_REG_DEFAULTS, + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4643_regmap = { @@ -568,6 +569,7 @@ static const struct regmap_config ak4643_regmap = { .max_register = SPK_MS, .reg_defaults = ak4643_reg, .num_reg_defaults = ARRAY_SIZE(ak4643_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct regmap_config ak4648_regmap = { @@ -576,6 +578,7 @@ static const struct regmap_config ak4648_regmap = { .max_register = EQ_FBEQE, .reg_defaults = ak4648_reg, .num_reg_defaults = ARRAY_SIZE(ak4648_reg), + .cache_type = REGCACHE_RBTREE, }; static const struct ak4642_drvdata ak4642_drvdata = { -- 2.8.1

3 4

[alsa-devel] [PATCH 1/2] ASoC: Intel: Skylake: Add channel constraints for refcap
by Yong Zhi 19 May '16

19 May '16

Add constraint for ref DMIC to match with the topology firmware config. Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> --- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 7cc7251..463f0e5 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -339,8 +339,22 @@ static struct snd_pcm_hw_constraint_list constraints_16000 = { .list = rates_16000, }; +static const unsigned int ch_mono[] = { + 1, +}; + +static const struct snd_pcm_hw_constraint_list constraints_refcap = { + .count = ARRAY_SIZE(ch_mono), + .list = ch_mono, +}; + static int skylake_refcap_startup(struct snd_pcm_substream *substream) { + substream->runtime->hw.channels_max = 1; + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_refcap); + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_16000); -- 1.9.1

2 2

[alsa-devel] [PATCH V2 1/2] dt: binding: ASOC: add dt-binding doc for pcm5102a
by kernel＠martin.sperl.org 18 May '16

18 May '16

From: Martin Sperl <kernel(a)martin.sperl.org> Add dt-binding documentation for the TI pcm5102a DAC. Signed-off-by: Martin Sperl <kernel(a)martin.sperl.org> --- Documentation/devicetree/bindings/sound/pcm5102a.txt | 13 +++++++++++++ 1 file changed, 13 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt diff --git a/Documentation/devicetree/bindings/sound/pcm5102a.txt b/Documentation/devicetree/bindings/sound/pcm5102a.txt new file mode 100644 index 0000000..87d72be --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm5102a.txt @@ -0,0 +1,13 @@ +PCM5102a audio CODEC + +This device is a i2s only device. + +Required properties: + +- compatible : set as "ti,pcm5102a" + +Examples: + + pcm5102a: pcm5102a { + compatible = "ti,pcm5102a"; + }; -- 2.1.4

2 2

[alsa-devel] [PATCH v2 1/1] ASoC: core: Do not print an actual error when deferring probe
by Alexander Stein 18 May '16

18 May '16

In case there is no DAI (yet), do not print an error, this might happen a lot of times. Print a notice instead. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> --- Changes in v2: * Add addional change in soc_bind_aux_dev() sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d2e62b15..ce5f835 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1001,7 +1001,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, cpu_dai_component.dai_name = dai_link->cpu_dai_name; rtd->cpu_dai = snd_soc_find_dai(&cpu_dai_component); if (!rtd->cpu_dai) { - dev_err(card->dev, "ASoC: CPU DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CPU DAI %s not registered\n", dai_link->cpu_dai_name); goto _err_defer; } @@ -1013,7 +1013,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, for (i = 0; i < rtd->num_codecs; i++) { codec_dais[i] = snd_soc_find_dai(&codecs[i]); if (!codec_dais[i]) { - dev_err(card->dev, "ASoC: CODEC DAI %s not registered\n", + dev_notice(card->dev, "ASoC: CODEC DAI %s not registered\n", codecs[i].dai_name); goto _err_defer; } @@ -1042,7 +1042,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, rtd->platform = platform; } if (!rtd->platform) { - dev_err(card->dev, "ASoC: platform %s not registered\n", + dev_notice(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1698,7 +1698,7 @@ static int soc_bind_aux_dev(struct snd_soc_card *card, int num) return 0; err_defer: - dev_err(card->dev, "ASoC: %s not registered\n", name); + dev_notice(card->dev, "ASoC: %s not registered\n", name); return -EPROBE_DEFER; } -- 2.7.3

3 4

[alsa-devel] [PATCH] ASoC: add api to retrieve dmic array info from coreboot nhlt
by Yong Zhi 18 May '16

18 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 | 10 +++++++++- sound/soc/intel/skylake/skl.h | 6 ++++++ 4 files changed, 74 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-nhlt.c b/sound/soc/intel/skylake/skl-nhlt.c index 7d73648..5448a48 100644 --- a/sound/soc/intel/skylake/skl-nhlt.c +++ b/sound/soc/intel/skylake/skl-nhlt.c @@ -149,6 +149,43 @@ struct nhlt_specific_cfg return NULL; } +int skl_get_ep_dmic(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 hdac_bus *bus = ebus_to_hbus(&skl->ebus); + struct device *dev = bus->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..8da502b 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 { + uint8_t virtual_slot; + uint8_t config_type; +} __packed; + +struct nhlt_dmic_array_config { + struct nhlt_tdm_config tdm_config; + uint8_t 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..a75b05f 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_dmic_info 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_ep_dmic(skl); + /* check if dsp is there */ if (ebus->ppcap) { err = skl_machine_device_register(skl, @@ -789,7 +797,7 @@ static struct sst_acpi_mach sst_skl_devdata[] = { { "INT343B", "skl_nau88l25_ssm4567_i2s", "intel/dsp_fw_release.bin", NULL, NULL, NULL }, { "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..889ec30 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_dmic_info { + 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_ep_dmic(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

1 0

[alsa-devel] [PATCH] ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type
by Takashi Sakamoto 18 May '16

18 May '16

Commit a9c4284bf5a9 ("ALSA: firewire-lib: add context information to tracepoints") adds new members to tracepoint events of this module, to represent context information. One of the members is bool type and this causes sparse warnings. 16:1: warning: expression using sizeof bool 60:1: warning: expression using sizeof bool 16:1: warning: odd constant _Bool cast (ffffffffffffffff becomes 1) 60:1: warning: odd constant _Bool cast (ffffffffffffffff becomes 1) This commit suppresses the warnings, by changing type of the member to 'unsigned int'. Additionally, this commit applies '!!' idiom to get 0/1 from 'in_interrupt()'. Fixes: a9c4284bf5a9 ("ALSA: firewire-lib: add context information to tracepoints") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 1622579..9c04faf 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -26,7 +26,7 @@ TRACE_EVENT(in_packet, __field(u32, cip_header1) __field(unsigned int, payload_quadlets) __field(unsigned int, packet_index) - __field(bool, irq) + __field(unsigned int, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -39,7 +39,7 @@ TRACE_EVENT(in_packet, __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; __entry->packet_index = s->packet_index; - __entry->irq = in_interrupt(); + __entry->irq = !!in_interrupt(); __entry->index = index; ), TP_printk( @@ -70,7 +70,7 @@ TRACE_EVENT(out_packet, __field(u32, cip_header1) __field(unsigned int, payload_quadlets) __field(unsigned int, packet_index) - __field(bool, irq) + __field(unsigned int, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -83,7 +83,7 @@ TRACE_EVENT(out_packet, __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; __entry->packet_index = s->packet_index; - __entry->irq = in_interrupt(); + __entry->irq = !!in_interrupt(); __entry->index = index; ), TP_printk( -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: twl6040: Disconnect AUX output pads on digital mute
by Peter Ujfalusi 18 May '16

18 May '16

Disconnect also the path to AUXL from the HF path during digital_mute to avoid pop noise leakage to Line-out pads. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- include/linux/mfd/twl6040.h | 1 + sound/soc/codecs/twl6040.c | 4 ++-- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/twl6040.h b/include/linux/mfd/twl6040.h index a7c50e54f3e0..8015db974157 100644 --- a/include/linux/mfd/twl6040.h +++ b/include/linux/mfd/twl6040.h @@ -134,6 +134,7 @@ #define TWL6040_HFDACENA (1 << 0) #define TWL6040_HFPGAENA (1 << 1) #define TWL6040_HFDRVENA (1 << 4) +#define TWL6040_HFSWENA (1 << 6) /* VIBCTLL/R (0x18/0x1A) fields */ diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 7a744b581cc1..1f7081043566 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -987,9 +987,9 @@ static void twl6040_mute_path(struct snd_soc_codec *codec, enum twl6040_dai_id i if (mute) { /* Power down drivers and DACs */ hflctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); hfrctl &= ~(TWL6040_HFDACENA | TWL6040_HFPGAENA | - TWL6040_HFDRVENA); + TWL6040_HFDRVENA | TWL6040_HFSWENA); } twl6040_reg_write(twl6040, TWL6040_REG_HFLCTL, hflctl); -- 2.8.2

1 0

[alsa-devel] [PATCH v7 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Simran Rai 18 May '16

18 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-v7 Changes from v6: - DT bindings acknowledged by Rob Herring 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 -- 1.9.1

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Re: [alsa-devel] [PATCH v7 0/3] ASoC: cygnus: Add audio support for Broadcom Cygnus SoC
by Florian Fainelli 17 May '16

17 May '16

On 05/17/2016 12:46 PM, Simran Rai wrote: > From: Simran <ssimran(a)lbrmn-ssimran-lnxub64.ric.broadcom.com> Please fix your git email address and resubmit. -- Florian

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[alsa-devel] Imx6 i2s master driver without i2c
by nick83ola 17 May '16

17 May '16

Hello Alsa dev! Anyone have an example of an i2s only driver for imx6 solo? I need a driver without i2c....

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: remove ununsed variable 'timeout'
by Colin King 17 May '16

17 May '16

From: Colin Ian King <colin.king(a)canonical.com> Fix build warning by removing unused variable timeout Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/hdac_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3733297..181cd3b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1718,7 +1718,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; - unsigned long timeout; struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; -- 2.8.1

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[alsa-devel] [PATCH] sound: oss: Use setup_timer and mod_timer.
by Muhammad Falak R Wani 17 May '16

17 May '16

The function setup_timer combines the initialization of a timer with the initialization of the timer's function and data fields. The mulitiline code for timer initialization is now replaced with function setup_timer. Also, quoting the mod_timer() function comment: -> mod_timer() is a more efficient way to update the expire field of an active timer (if the timer is inactive it will be activated). Use setup_timer() and mod_timer() to setup and arm a timer, making the code compact and aid readablity. Signed-off-by: Muhammad Falak R Wani <falakreyaz(a)gmail.com> --- sound/oss/waveartist.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index b36ea47..0b8d0de 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -1414,11 +1414,9 @@ attach_waveartist(struct address_info *hw, const struct waveartist_mixer_info *m else { #ifdef CONFIG_ARCH_NETWINDER if (machine_is_netwinder()) { - init_timer(&vnc_timer); - vnc_timer.function = vnc_slider_tick; - vnc_timer.expires = jiffies; - vnc_timer.data = nr_waveartist_devs; - add_timer(&vnc_timer); + setup_timer(&vnc_timer, vnc_slider_tick, + nr_waveartist_devs); + mod_timer(&vnc_timer, jiffies); vnc_configure_mixer(devc, 0); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: nau8825: Export I2C module alias information
by Javier Martinez Canillas 17 May '16

17 May '16

The I2C driver has an i2c_device_id array but that information isn't exported to the module using the MODULE_DEVICE_TABLE() macro. So the module autoloading won't work if the I2C device is registered using OF or legacy board files due missing alias information in the module. The issue was found using Kieran Bingham's coccinelle semantic patch: https://lkml.org/lkml/2016/5/10/520 Signed-off-by: Javier Martinez Canillas <javier(a)osg.samsung.com> --- sound/soc/codecs/nau8825.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 81fc97b07751..e988f89ef715 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -1493,6 +1493,7 @@ static const struct i2c_device_id nau8825_i2c_ids[] = { { "nau8825", 0 }, { } }; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); #ifdef CONFIG_OF static const struct of_device_id nau8825_of_ids[] = { -- 2.5.5

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[alsa-devel] [PATCH] alsa-lib:Make snd_atomic_write_* truly atomic
by Andreas x Larsson 17 May '16

17 May '16

Under some usecases a race condition appears inside the snd_atomic_write_* functions. The 'begin' and 'end' variables are updated with the ++ operator which is not atomic but needs to be. This can be achieved with the gcc atomic_* built-ins. Combined with __ATOMIC_SEQ_CST as the memory model, memory barriers are introduced so those can also be removed from the code. Signed-off-by: Andreas x Larsson <andrelan(a)axis.com> Signed-off-by: Arvid Nihlgård Lindell <arvidnl(a)axis.com> --- alsa-lib/include/iatomic.h | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/alsa-lib/include/iatomic.h b/alsa-lib/include/iatomic.h index acdd3e2..7fafe20 100644 --- a/alsa-lib/include/iatomic.h +++ b/alsa-lib/include/iatomic.h @@ -140,14 +140,12 @@ static __inline__ void snd_atomic_write_init(snd_atomic_write_t *w) static __inline__ void snd_atomic_write_begin(snd_atomic_write_t *w) { - w->begin++; - wmb(); + __atomic_add_fetch(&w->begin, 1,__ATOMIC_SEQ_CST); } static __inline__ void snd_atomic_write_end(snd_atomic_write_t *w) { - wmb(); - w->end++; + __atomic_add_fetch(&w->end, 1, __ATOMIC_SEQ_CST); } static __inline__ void snd_atomic_read_init(snd_atomic_read_t *r, snd_atomic_write_t *w) -- 2.1.4

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[alsa-devel] [PATCH v2 0/5] ASoC: rockchip: Fix audio on Veyron.
by Enric Balletbo i Serra 17 May '16

17 May '16

Hi all, This patch series adds support and fixes the audio support for Veyron devices. With these patches I'm able to playback from speakers and headphones, and record from the microphone on a Veyron Jerry Chromebook. Patches from 1/5 to 3/5 should go through the Mark Brown ASoC tree. Patches 4/5 and 5/5 should go through the Heiko Stuebner rockchip tree. Changes since v1: - Improve log messages trying to explain better the issues (suggested by Mark Brown) - Remove patches from John Keeping as are already cherry-picked. - https://git.kernel.org/cgit/linux/kernel/git/broonie/sound.git/commit/?h=fo… - https://git.kernel.org/cgit/linux/kernel/git/broonie/sound.git/commit/?h=fo… Enric Balletbo i Serra (5): ASoC: rockchip-max98090: Fix NULL pointer dereference while accessing to jack. ASoC: rockchip-max98090: Fix the Headset Mic route. ASoC: rockchip-max98090: Fix jack detection and event reporting. ARM: dts: rockchip: Add shared file for audio related nodes for veyron boards ARM: dts: rockchip: veyron: Add analog audio codecs. arch/arm/boot/dts/rk3288-veyron-analog-audio.dtsi | 102 ++++++++++++++++++++++ arch/arm/boot/dts/rk3288-veyron-chromebook.dtsi | 1 + sound/soc/rockchip/rockchip_max98090.c | 65 ++++++++------ 3 files changed, 140 insertions(+), 28 deletions(-) create mode 100644 arch/arm/boot/dts/rk3288-veyron-analog-audio.dtsi -- 2.1.0

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[alsa-devel] BUG since "ASoC: simple-card: Add pm callbacks to platform driver"
by Geert Uytterhoeven 17 May '16

17 May '16

Hi Peter, Mark, Liam, With today's sound-asoc/for-next, I'm seeing on r8a7791/koelsch and sh73a0/kzm9g: kernel BUG at drivers/base/regmap/regcache.c:347! Internal error: Oops - BUG: 0 [#1] SMP ARM Modules linked in: CPU: 0 PID: 20 Comm: kworker/0:1 Not tainted 4.6.0-koelsch-07135-g1481dd77fa7d01cb #2621 Hardware name: Generic R8A7791 (Flattened Device Tree) Workqueue: events soc_resume_deferred task: ef512100 ti: ef5a6000 task.ti: ef5a6000 PC is at regcache_sync+0xc/0x118 LR is at ak4642_resume+0x20/0x28 pc : [<c04932a4>] lr : [<c05bd34c>] psr: 60000153 sp : ef5a7ee0 ip : ee7ea62c fp : 00000008 r10: 00000000 r9 : ee7ebd0c r8 : ee7ebd18 r7 : ee7ebc10 r6 : ee7ebcb8 r5 : ee7ebd08 r4 : eefc5800 r3 : 00000000 r2 : eefc5814 r1 : 001c001b r0 : eefc5800 Flags: nZCv IRQs on FIQs off Mode SVC_32 ISA ARM Segment kernel Control: 30c5387d Table: 6eaee540 DAC: fffffffd Process kworker/0:1 (pid: 20, stack limit = 0xef5a6210) Stack: (0xef5a7ee0 to 0xef5a8000) 7ee0: eefc5600 c05ae878 ef588100 ee7ebd08 ef9bdf00 00000000 ef9c4400 c023341c 7f00: ef588100 ee7ebd08 ef588100 ef9bdf00 ef9bdf00 ef5a6000 ef9bdf24 c0e02100 7f20: ef588118 c0233ae0 ef512100 ef59ae00 00000000 ef588100 c0233838 00000000 7f40: 00000000 00000000 00000000 c0237b58 00000001 00000000 ef43b040 ef588100 7f60: 00000000 00000000 dead4ead ffffffff ffffffff ef5a7f74 ef5a7f74 00000000 7f80: 00000000 dead4ead ffffffff ffffffff ef5a7f90 ef5a7f90 ef5a7fac ef59ae00 7fa0: c0237a80 00000000 00000000 c0206b68 00000000 00000000 00000000 00000000 7fc0: 00000000 00000000 00000000 00000000 00000000 00000000 00000000 00000000 7fe0: 00000000 00000000 00000000 00000000 00000013 00000000 ef5a7ff4 00000000 [<c04932a4>] (regcache_sync) from [<ee7ebd08>] (0xee7ebd08) Code: c0978f0f e590312c e3530000 1a000000 (e7f001f2) I've bisected this to commit 7c3767115a04bc7aa87bdbd3352d1801d4bbeea4 Author: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Date: Mon May 9 13:38:10 2016 +0300 ASoC: simple-card: Add pm callbacks to platform driver Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Reverting it fixes the problem for me. 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

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[alsa-devel] [PATCH v2] ASoC: rsnd: open 31bit of SSICKR mask
by Kuninori Morimoto 17 May '16

17 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSICKR (Gen2) / BRGCKR (Gen3) 31bit mask should be opened, because BRGB (= for 48kHz) might select it. Special thanks Yokoyama-san Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - removed #define DEBUG (^o^) sound/soc/sh/rcar/adg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399d..0891014 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -522,7 +522,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, } } - rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr); + rsnd_mod_bset(adg_mod, SSICKR, 0x80FF0000, ckr); rsnd_mod_write(adg_mod, BRRA, rbga); rsnd_mod_write(adg_mod, BRRB, rbgb); -- 1.9.1

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[alsa-devel] [PATCH] ASoC: rsnd: open 31bit of SSICKR mask
by Kuninori Morimoto 17 May '16

17 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> SSICKR (Gen2) / BRGCKR (Gen3) 31bit mask should be opened, because BRGB (= for 48kHz) might select it. Special thanks Yokoyama-san Reported-by: Hiroyuki Yokoyama <hiroyuki.yokoyama.vx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 606399d..061deb7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -7,6 +7,7 @@ * License. See the file "COPYING" in the main directory of this archive * for more details. */ +#define DEBUG #include <linux/clk-provider.h> #include "rsnd.h" @@ -522,7 +523,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, } } - rsnd_mod_bset(adg_mod, SSICKR, 0x00FF0000, ckr); + rsnd_mod_bset(adg_mod, SSICKR, 0x80FF0000, ckr); rsnd_mod_write(adg_mod, BRRA, rbga); rsnd_mod_write(adg_mod, BRRB, rbgb); -- 1.9.1

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[alsa-devel] Bug report: Sound skipping/stuttering every 5-20 minutes
by darknessdescends＠tutanota.com 16 May '16

16 May '16

My sound skips/stutters for about half a second (certainly no more than 1 second) on average every 5-20 minutes. I initially had no sound on this computer with Ubuntu 14.04, which is probably because Ubuntu 14.04 has an outdated version of Alsa. I then tested with a live disc of Ubuntu 16.04 (before official release date), which did give me sound. At this point, I installed the the latest version of Alsa onto Ubuntu 14.04 from here (because 16.04 wasn't officially released at the time and I didn't want to take chances with an unstable system): https://wiki.ubuntu.com/Audio/UpgradingAlsa/DKMS Sound immediately worked with the latest version of Alsa's driver (March 24, 2016 at the time) on Ubuntu 14.04. However, it came with a cost: this stuttering problem. I then retested with the live disc of Ubuntu 16.04 (this time after the official release date), which also produced sound but with the exact same stuttering problem. I found that on Ubuntu 14.04, the stuttering problem remains unfixed since the March 24, 2016 release of the Alsa daily driver until at least the version I have installed in this bug report. These facts strongly suggest an Alsa problem: 1. The problem does not occur at all on Windows, thus ruling out the possibility of a hardware problem. 2. I had no sound at all before installing the Alsa daily drivers, but did have stuttering sound after installing them. 3. The problem persists on Ubuntu 16.04, which suggests the problem is specific to the driver rather than the operating system. 4. The problem occurs whether playing a locally stored media file with a media player installed on my computer or watching an online video, thus ruling out the possibility that the problem is due to a single browser or plugin issue. 5. Is my hardware too new? Probably not. Here are the results of running "grep "Codec:" /proc/asound/card*/codec*": /proc/asound/card0/codec#0:Codec: Realtek ALC891 /proc/asound/card1/codec#0:Codec: Nvidia Generic The fact that my card is identified by vendor and chip name rather than ID as specified at https://wiki.ubuntu.com/Audio/HDAGeneric suggests that my hardware is supported. By the way, I'm using the Realtek, not the Nvidia HDMI. I've tried disabling all unnecessary channels under alsamixer, but that didn't fix the problem. I tried all 3 potential fixes at https://wiki.ubuntu.com/Audio/PositionReporting, but none worked either. I commented my /etc/modprobe.d/alsa-base.conf and /etc/pulse/default.pa files appropriately to show what I tried (I know at least alsa-base.conf is uploaded). I really appreciate the effort you developers put in to supporting sound on Linux. I hope my bug report will help fix this problem for a lot of users on Skylake systems. More information and attached files can be found here (I filed a bug report with Ubuntu before I found out about Alsa's developer mailing list): https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1580744

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[alsa-devel] [GIT PULL]: Update sst audio firmware for Skylake
by Vinod Koul 16 May '16

16 May '16

Hello Linux firmware maintainers, Please PULL to receive the updated version of Skylake audio firmware version 927. This contains off fixes for system wake and wov module handler. The following changes since commit 8afadbe553017bec1e442b5a9fba859b54fd33fb: ath10k: QCA9377 hw1.0: add firmware and board files (2016-05-03 13:15:57 -0700) 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 40d7ffa911fe52d63a12a4cc51231580f81a5fcf: linux-firmware: intel: Update Skylake audio firmware (2016-05-05 12:06:56 +0530) ---------------------------------------------------------------- Vinod Koul (1): linux-firmware: intel: Update Skylake audio firmware WHENCE | 4 ++-- intel/dsp_fw_release.bin | 2 +- intel/dsp_fw_release_v927.bin | Bin 0 -> 233472 bytes 3 files changed, 3 insertions(+), 3 deletions(-) create mode 100644 intel/dsp_fw_release_v927.bin -- ~Vinod

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[alsa-devel] Porting WM8904 codec to new board - trying to get ALSA aplay going
by Nounou Dadoun 16 May '16

16 May '16

Not sure whether this should go to the dev or users list (or somewhere in between) so I'll start off trying both. I'm trying to port a wm8904 audio driver/codec to a new board and the new board uses a device tree configuration which is new to both me and our project. The target arm board uses a linux 3-10 kernel, I have the wm8904 on the i2c bus and configured the Sound in the device tree to use an i2s controller. The codec and platform driver are modprobed on startup. I've made a lot of progress, I (finally!) have the codec loading/modprobed properly and the platform driver probed and successful (up to a point), so I have in the serial log: [ 6.766524] snd_soc_card_avigilon sound.3: wm8904-hifi <-> 6000a000.i2s mapping ok And at the console: # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AVIGILONWM8904 [AVIGILON-WM8904], device 0: WM8904-STREAM wm8904-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 # ls /proc/asound/ AVIGILONWM8904 card0 cards devices pcm timers version # cat /proc/asound/cards 0 [AVIGILONWM8904 ]: AVIGILON-WM8904 - AVIGILON-WM8904 AVIGILON-WM8904 But when I try to play a wave file, I get (no sound and): # aplay -v audiocheck.net_sin_220Hz_0dBFS_10s.wav Playing WAVE 'audiocheck.net_sin_220Hz_0dBFS_10s.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono aplay: set_params:1297: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 44100 PERIOD_TIME: (46439 46440) PERIOD_SIZE: 2048 PERIOD_BYTES: 4096 PERIODS: 11 BUFFER_TIME: (510839 510840) BUFFER_SIZE: 22528 BUFFER_BYTES: 45056 TICK_TIME: 0 # But (encouragingly) when I try this, I get my trace statement from the platform driver: [ 3794.140675] Calling avigilon_board_startup in the serial log. >From the canonical/generic platform driver file, the soc_snd_probe probe routine seems to be doing a bunch of dai link configuration and other stuff (including reading values out of the device tree) which I'm not currently doing in my equivalent probe routine. (And which I haven't been able to find any documentation for.) So a few questions: Am I correct in assuming that the " Unable to install hw params " error indicates that something is missing at the board probe level? Any suggestions about what's necessary to get it going? Is there any documentation about what the board probe should be doing to set up the hardware for playback properly? Is the board startup routine required to do anything? (It didn't in the non-device tree implementation which I'm taking it from.) Any other useful hints or pointers? Is there a question which I should be asking which I'm not? Thanks in advance for any assistance which you might be able to provide, my progress to date has involved a lot of trial and error and has been much slower than I would like! ... N Nou Dadoun Senior Firmware Developer, Security Specialist Office: 604.629.5182 ext 2632 Support: 888.281.5182 | avigilon.com Follow Twitter | Follow LinkedIn This email, including any files attached hereto (the "email"), contains privileged and confidential information and is only for the intended addressee(s). If this email has been sent to you in error, such sending does not constitute waiver of privilege and we request that you kindly delete the email and notify the sender. Any unauthorized use or disclosure of this email is prohibited. Avigilon and certain other trade names used herein are the registered and/or unregistered trademarks of Avigilon Corporation and/or its affiliates in Canada and other jurisdictions worldwide.

1 0

[alsa-devel] ALSA Device Driver Programming
by Fredrick Cantabery 16 May '16

16 May '16

First off, let me state that I hope that I am not inappropriately infringing upon anyone's time and hope this post is proper for the audience. Language Acoustics is a startup loudspeaker company who is trying to trying to develop an ALSA compliant sound card driver to support various hardware devices in Linux on an x86 based platform. The sound devices will be multi-channel with both analog and digital inputs and outputs. This has proven to be a difficult hurdle for us and we are reaching out to you to see if you might know someone who might be able to assist in the development effort. We view this as a freelance opportunity for someone but could be interested in other alternative partnerships if someone has interest. Experience in sample rate conversion and the JACK audio connection kit would be highly desirable as well. Thank you for time and consideration. Rick Cantabery Owner, Language Acoustics

1 0

[alsa-devel] ADAU1761 default register value problem
by Ricard Wanderlof 16 May '16

16 May '16

Hi Lars, A while ago I posted a question to the list regarding the default values set up for the ADAU1761 (and similar CODECs), but I didn't get a response. I really need to try and figure out a way forward here though so I'll bring this up again. The problem I have is that the driver assumes that the CODEC has been initialized with the default register settings when the system boots. However, if the system has not been power cycled but just rebooted, this is not necessarily the case, as the CODEC has no external reset pin but derives its reset signal internally from the power level. Furthermore, if the system upon reboot tries to set a register value to one of the default settings (for example as the result of a hw_params() call), then the regcache mechanism will disregard the write as it assumes the default value is already set, which is not necessarily the case. As I see it the solution to this problem is simply not to assume anything about the register settings, so that the first write to a given register always takes effect. I can work on a patch to accomplish this, however, since so much work apparently has been laid down in getting the default values into the driver in the first place, I'm assuming there's something more to the story that I'm missing, so I'd like to get a better grip on this before I dive in. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v3] ASoC: MAX9860: new driver
by Peter Rosin 16 May '16

16 May '16

This is a driver for the MAX9860 Mono Audio Voice Codec. https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf This driver does not support sidetone since the DVST register field is backwards with the mute near the maximum level instead of the minimum. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- .../devicetree/bindings/sound/max9860.txt | 28 + MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 761 +++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 +++++ 6 files changed, 966 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h New in v3: - The updated bindings file went missing in v2. Sorry for the confusion. New in v2: - Add comment about fall through when Integer Clock Mode is not possible. - Drop export of max9860_probe. - Ignore clk docs and read the mclk rate w/o enabling the clock. - Manage the DVDDIO supply. Cheers, Peter diff --git a/Documentation/devicetree/bindings/sound/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt new file mode 100644 index 000000000000..e0d4e95e31b3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9860.txt @@ -0,0 +1,28 @@ +MAX9860 Mono Audio Voice Codec + +Required properties: + + - compatible : "maxim,max9860" + + - reg : the I2C address of the device + + - AVDD-supply, DVDD-supply and DVDDIO-supply : power supplies for + the device, as covered in bindings/regulator/regulator.txt + + - clock-names : Required element: "mclk". + + - clocks : A clock specifier for the clock connected as MCLK. + +Examples: + + max9860: max9860@10 { + compatible = "maxim,max9860"; + reg = <0x10>; + + AVDD-supply = <&reg_1v8>; + DVDD-supply = <&reg_1v8>; + DVDDIO-supply = <&reg_3v0>; + + clock-names = "mclk"; + clocks = <&pck2>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index a727d9959ecd..0803396b10f4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7000,6 +7000,13 @@ F: Documentation/devicetree/bindings/i2c/max6697.txt F: drivers/hwmon/max6697.c F: include/linux/platform_data/max6697.h +MAX9860 MONO AUDIO VOICE CODEC DRIVER +M: Peter Rosin <peda(a)axentia.se> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/max9860.txt +F: sound/soc/codecs/max9860* + MAXIM MUIC CHARGER DRIVERS FOR EXYNOS BASED BOARDS M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> L: linux-pm(a)vger.kernel.org diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c16478..241a23c9aa9f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C + select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -534,6 +535,11 @@ config SND_SOC_MAX98926 config SND_SOC_MAX9850 tristate +config SND_SOC_MAX9860 + tristate "Maxim MAX9860 Mono Audio Voice Codec" + depends on I2C + select REGMAP_I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..e435f4df1787 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,6 +78,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o +snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o @@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c new file mode 100644 index 000000000000..ee30832b0afb --- /dev/null +++ b/sound/soc/codecs/max9860.c @@ -0,0 +1,761 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf + * + * The driver does not support sidetone since the DVST register field is + * backwards with the mute near the maximum level instead of the minimum. + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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. + * + * 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 <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "max9860.h" + +struct max9860_priv { + struct regmap *regmap; + struct regulator *dvddio; + struct notifier_block dvddio_nb; + u8 psclk; + unsigned long pclk_rate; + int fmt; +}; + +static int max9860_dvddio_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct max9860_priv *max9860 = container_of(nb, struct max9860_priv, + dvddio_nb); + if (event & REGULATOR_EVENT_DISABLE) { + regcache_mark_dirty(max9860->regmap); + regcache_cache_only(max9860->regmap, true); + } + + return 0; +} + +static const struct reg_default max9860_reg_defaults[] = { + { MAX9860_PWRMAN, 0x00 }, + { MAX9860_INTEN, 0x00 }, + { MAX9860_SYSCLK, 0x00 }, + { MAX9860_AUDIOCLKHIGH, 0x00 }, + { MAX9860_AUDIOCLKLOW, 0x00 }, + { MAX9860_IFC1A, 0x00 }, + { MAX9860_IFC1B, 0x00 }, + { MAX9860_VOICEFLTR, 0x00 }, + { MAX9860_DACATTN, 0x00 }, + { MAX9860_ADCLEVEL, 0x00 }, + { MAX9860_DACGAIN, 0x00 }, + { MAX9860_MICGAIN, 0x00 }, + { MAX9860_MICADC, 0x00 }, + { MAX9860_NOISEGATE, 0x00 }, +}; + +static bool max9860_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + case MAX9860_REVISION: + return true; + } + + return false; +} + +static bool max9860_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTEN ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + return true; + } + + return false; +} + +static bool max9860_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + case MAX9860_MICREADBACK: + return true; + } + + return false; +} + +static bool max9860_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + return true; + } + + return false; +} + +const struct regmap_config max9860_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max9860_readable, + .writeable_reg = max9860_writeable, + .volatile_reg = max9860_volatile, + .precious_reg = max9860_precious, + + .max_register = MAX9860_MAX_REGISTER, + .reg_defaults = max9860_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_RANGE(pam_tlv, + 0, MAX9860_PAM_MAX - 1, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + MAX9860_PAM_MAX, MAX9860_PAM_MAX, TLV_DB_SCALE_ITEM(3000, 0, 0)); +static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1); +static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0); + +static const char * const agchld_text[] = { + "AGC Disabled", "50ms", "100ms", "400ms" +}; + +static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC, + MAX9860_AGCHLD_SHIFT, agchld_text); + +static const char * const agcsrc_text[] = { + "Left ADC", "Left/Right ADC" +}; + +static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC, + MAX9860_AGCSRC_SHIFT, agcsrc_text); + +static const char * const agcatk_text[] = { + "3ms", "12ms", "50ms", "200ms" +}; + +static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC, + MAX9860_AGCATK_SHIFT, agcatk_text); + +static const char * const agcrls_text[] = { + "78ms", "156ms", "312ms", "625ms", + "1.25s", "2.5s", "5s", "10s" +}; + +static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC, + MAX9860_AGCRLS_SHIFT, agcrls_text); + +static const char * const filter_text[] = { + "Disabled", + "Elliptical HP 217Hz notch (16kHz)", + "Butterworth HP 500Hz (16kHz)", + "Elliptical HP 217Hz notch (8kHz)", + "Butterworth HP 500Hz (8kHz)", + "Butterworth HP 200Hz (48kHz)" +}; + +static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR, + MAX9860_AVFLT_SHIFT, filter_text); + +static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR, + MAX9860_DVFLT_SHIFT, filter_text); + +static const struct snd_kcontrol_new max9860_controls[] = { +SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN, + MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv), +SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN, + MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv), +SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL, + MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1, + adc_tlv), + +SOC_ENUM("AGC Hold Time", agchld_enum), +SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum), +SOC_ENUM("AGC Attack Time", agcatk_enum), +SOC_ENUM("AGC Release Time", agcrls_enum), + +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE, + MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv), +SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE, + MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv), + +SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN, + MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv), +SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN, + MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv), + +SOC_ENUM("ADC Filter", avflt_enum), +SOC_ENUM("DAC Filter", dvflt_enum), +}; + +static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICL"), +SND_SOC_DAPM_INPUT("MICR"), + +SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("OUT"), + +SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0), +SND_SOC_DAPM_CLOCK_SUPPLY("mclk"), +}; + +static const struct snd_soc_dapm_route max9860_dapm_routes[] = { + { "ADCL", NULL, "MICL" }, + { "ADCR", NULL, "MICR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, + + { "DAC", NULL, "AIFINL" }, + { "DAC", NULL, "AIFINR" }, + { "OUT", NULL, "DAC" }, + + { "Supply", NULL, "AVDD" }, + { "Supply", NULL, "DVDD" }, + { "Supply", NULL, "mclk" }, + + { "DAC", NULL, "Supply" }, + { "ADCL", NULL, "Supply" }, + { "ADCR", NULL, "Supply" }, +}; + +static int max9860_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + return 0; + + default: + return -EINVAL; + } +} + +static int max9860_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 max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + u8 master; + u8 ifc1a = 0; + u8 ifc1b = 0; + u8 sysclk = 0; + unsigned long n; + int ret; + + dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n", + params_rate(params), + params_channels(params)); + + if (params_channels(params) == 2) + ifc1b |= MAX9860_ST; + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + master = MAX9860_MASTER; + break; + default: + return -EINVAL; + } + ifc1a |= master; + + if (master) { + if (params_width(params) * params_channels(params) > 48) + ifc1b |= MAX9860_BSEL_64X; + else + ifc1b |= MAX9860_BSEL_48X; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifc1a |= MAX9860_DDLY; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifc1a |= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_DSP_A: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_A works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_DSP_B: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_B works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + break; + default: + return -EINVAL; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_IB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + /* fall through */ + case SND_SOC_DAIFMT_IB_NF: + ifc1a ^= MAX9860_DBCI; + ifc1b ^= MAX9860_ABCI; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "IFC1A %02x\n", ifc1a); + ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "IFC1B %02x\n", ifc1b); + ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret); + return ret; + } + + /* + * Check if Integer Clock Mode is possible, but avoid it in slave mode + * since we then do not know if lrclk is derived from pclk and the + * datasheet mentions that the frequencies have to match exactly in + * order for this to work. + */ + if (params_rate(params) == 8000 || params_rate(params) == 16000) { + if (master) { + switch (max9860->pclk_rate) { + case 12000000: + sysclk = MAX9860_FREQ_12MHZ; + break; + case 13000000: + sysclk = MAX9860_FREQ_13MHZ; + break; + case 19200000: + sysclk = MAX9860_FREQ_19_2MHZ; + break; + default: + /* + * Integer Clock Mode not possible. Leave + * sysclk at zero and fall through to the + * code below for PLL mode. + */ + break; + } + + if (sysclk && params_rate(params) == 16000) + sysclk |= MAX9860_16KHZ; + } + } + + /* + * Largest possible n: + * 65536 * 96 * 48kHz / 10MHz -> 30199 + * Smallest possible n: + * 65536 * 96 * 8kHz / 20MHz -> 2517 + * Both fit nicely in the available 15 bits, no need to apply any mask. + */ + n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params), + max9860->pclk_rate); + + if (!sysclk) { + /* PLL mode */ + if (params_rate(params) > 24000) + sysclk |= MAX9860_16KHZ; + + if (!master) + n |= 1; /* trigger rapid pll lock mode */ + } + + sysclk |= max9860->psclk; + dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk); + ret = regmap_write(max9860->regmap, + MAX9860_SYSCLK, sysclk); + if (ret) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "N %lu\n", n); + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKHIGH, n >> 8); + if (ret) { + dev_err(codec->dev, "Failed to set NHI: %d\n", ret); + return ret; + } + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKLOW, n & 0xff); + if (ret) { + dev_err(codec->dev, "Failed to set NLO: %d\n", ret); + return ret; + } + + if (!master) { + dev_dbg(codec->dev, "Enable PLL\n"); + ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH, + MAX9860_PLL, MAX9860_PLL); + if (ret) { + dev_err(codec->dev, "Failed to enable PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + max9860->fmt = fmt; + return 0; +} + +static const struct snd_soc_dai_ops max9860_dai_ops = { + .startup = max9860_dai_startup, + .hw_params = max9860_hw_params, + .set_fmt = max9860_set_fmt, +}; + +static struct snd_soc_dai_driver max9860_dai = { + .name = "max9860-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &max9860_dai_ops, + .symmetric_rates = 1, +}; + +static int max9860_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max9860_priv *max9860 = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, MAX9860_SHDN); + if (ret) { + dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, 0); + if (ret) { + dev_err(codec->dev, "Failed to request SHDN: %d\n", + ret); + return ret; + } + break; + } + + return 0; +} + +static struct snd_soc_codec_driver max9860_codec_driver = { + .set_bias_level = max9860_set_bias_level, + .idle_bias_off = true, + + .controls = max9860_controls, + .num_controls = ARRAY_SIZE(max9860_controls), + .dapm_widgets = max9860_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets), + .dapm_routes = max9860_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes), +}; + +#ifdef CONFIG_PM +static int max9860_suspend(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, MAX9860_PSCLK_OFF); + if (ret) { + dev_err(dev, "Failed to disable clock: %d\n", ret); + return ret; + } + + regulator_disable(max9860->dvddio); + + return 0; +} + +static int max9860_resume(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regulator_enable(max9860->dvddio); + if (ret) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + regcache_cache_only(max9860->regmap, false); + ret = regcache_sync(max9860->regmap); + if (ret) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, max9860->psclk); + if (ret) { + dev_err(dev, "Failed to enable clock: %d\n", ret); + return ret; + } + + return 0; +} +#endif + +const struct dev_pm_ops max9860_pm_ops = { + SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) +}; + +static int max9860_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860; + int ret; + struct clk *mclk; + unsigned long mclk_rate; + int i; + int intr; + + max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL); + if (!max9860) + return -ENOMEM; + + max9860->dvddio = devm_regulator_get(dev, "DVDDIO"); + if (IS_ERR(max9860->dvddio)) { + ret = PTR_ERR(max9860->dvddio); + dev_err(dev, "Failed to get DVDDIO supply: %d\n", ret); + return ret; + } + + max9860->dvddio_nb.notifier_call = max9860_dvddio_event; + + ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb); + if (ret) + dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret); + + ret = regulator_enable(max9860->dvddio); + if (ret != 0) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap); + if (IS_ERR(max9860->regmap)) { + ret = PTR_ERR(max9860->regmap); + goto err_regulator; + } + + dev_set_drvdata(dev, max9860); + + /* + * mclk has to be in the 10MHz to 60MHz range. + * psclk is used to scale mclk into pclk so that + * pclk is in the 10MHz to 20MHz range. + */ + mclk = clk_get(dev, "mclk"); + + if (IS_ERR(mclk)) { + ret = PTR_ERR(mclk); + dev_err(dev, "Failed to get MCLK: %d\n", ret); + goto err_regulator; + } + + mclk_rate = clk_get_rate(mclk); + clk_put(mclk); + + if (mclk_rate > 60000000 || mclk_rate < 10000000) { + dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n", + mclk_rate); + ret = -EINVAL; + goto err_regulator; + } + if (mclk_rate >= 40000000) + max9860->psclk = 3; + else if (mclk_rate >= 20000000) + max9860->psclk = 2; + else + max9860->psclk = 1; + max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1); + max9860->psclk <<= MAX9860_PSCLK_SHIFT; + dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate); + + regcache_cache_bypass(max9860->regmap, true); + for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) { + ret = regmap_write(max9860->regmap, + max9860_regmap.reg_defaults[i].reg, + max9860_regmap.reg_defaults[i].def); + if (ret) { + dev_err(dev, "Failed to initialize register %u: %d\n", + max9860_regmap.reg_defaults[i].reg, ret); + goto err_regulator; + } + } + regcache_cache_bypass(max9860->regmap, false); + + ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr); + if (ret) { + dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret); + goto err_regulator; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &max9860_codec_driver, + &max9860_dai, 1); + if (ret) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_regulator: + regulator_disable(max9860->dvddio); + return ret; +} + +static int max9860_remove(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860 = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + regulator_disable(max9860->dvddio); + return 0; +} + +static const struct i2c_device_id max9860_i2c_id[] = { + { "max9860", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9860_i2c_id); + +static const struct of_device_id max9860_of_match[] = { + { .compatible = "maxim,max9860", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9860_of_match); + +static struct i2c_driver max9860_i2c_driver = { + .probe = max9860_probe, + .remove = max9860_remove, + .id_table = max9860_i2c_id, + .driver = { + .name = "max9860", + .of_match_table = max9860_of_match, + .pm = &max9860_pm_ops, + }, +}; + +module_i2c_driver(max9860_i2c_driver); + +MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver"); +MODULE_AUTHOR("Peter Rosin <peda(a)axentia.se>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h new file mode 100644 index 000000000000..22041bd67a7d --- /dev/null +++ b/sound/soc/codecs/max9860.h @@ -0,0 +1,162 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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. + * + * 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. + */ + +#ifndef _SND_SOC_MAX9860 +#define _SND_SOC_MAX9860 + +#define MAX9860_INTRSTATUS 0x00 +#define MAX9860_MICREADBACK 0x01 +#define MAX9860_INTEN 0x02 +#define MAX9860_SYSCLK 0x03 +#define MAX9860_AUDIOCLKHIGH 0x04 +#define MAX9860_AUDIOCLKLOW 0x05 +#define MAX9860_IFC1A 0x06 +#define MAX9860_IFC1B 0x07 +#define MAX9860_VOICEFLTR 0x08 +#define MAX9860_DACATTN 0x09 +#define MAX9860_ADCLEVEL 0x0a +#define MAX9860_DACGAIN 0x0b +#define MAX9860_MICGAIN 0x0c +#define MAX9860_RESERVED 0x0d +#define MAX9860_MICADC 0x0e +#define MAX9860_NOISEGATE 0x0f +#define MAX9860_PWRMAN 0x10 +#define MAX9860_REVISION 0xff + +#define MAX9860_MAX_REGISTER 0xff + +/* INTRSTATUS */ +#define MAX9860_CLD 0x80 +#define MAX9860_SLD 0x40 +#define MAX9860_ULK 0x20 + +/* MICREADBACK */ +#define MAX9860_NG 0xe0 +#define MAX9860_AGC 0x1f + +/* INTEN */ +#define MAX9860_ICLD 0x80 +#define MAX9860_ISLD 0x40 +#define MAX9860_IULK 0x20 + +/* SYSCLK */ +#define MAX9860_PSCLK 0x30 +#define MAX9860_PSCLK_OFF 0x00 +#define MAX9860_PSCLK_SHIFT 4 +#define MAX9860_FREQ 0x06 +#define MAX9860_FREQ_NORMAL 0x00 +#define MAX9860_FREQ_12MHZ 0x02 +#define MAX9860_FREQ_13MHZ 0x04 +#define MAX9860_FREQ_19_2MHZ 0x06 +#define MAX9860_16KHZ 0x01 + +/* AUDIOCLKHIGH */ +#define MAX9860_PLL 0x80 +#define MAX9860_NHI 0x7f + +/* AUDIOCLKLOW */ +#define MAX9860_NLO 0xff + +/* IFC1A */ +#define MAX9860_MASTER 0x80 +#define MAX9860_WCI 0x40 +#define MAX9860_DBCI 0x20 +#define MAX9860_DDLY 0x10 +#define MAX9860_HIZ 0x08 +#define MAX9860_TDM 0x04 + +/* IFC1B */ +#define MAX9860_ABCI 0x20 +#define MAX9860_ADLY 0x10 +#define MAX9860_ST 0x08 +#define MAX9860_BSEL 0x07 +#define MAX9860_BSEL_OFF 0x00 +#define MAX9860_BSEL_64X 0x01 +#define MAX9860_BSEL_48X 0x02 +#define MAX9860_BSEL_PCLK_2 0x04 +#define MAX9860_BSEL_PCLK_4 0x05 +#define MAX9860_BSEL_PCLK_8 0x06 +#define MAX9860_BSEL_PCLK_16 0x07 + +/* VOICEFLTR */ +#define MAX9860_AVFLT 0xf0 +#define MAX9860_AVFLT_SHIFT 4 +#define MAX9860_AVFLT_COUNT 6 +#define MAX9860_DVFLT 0x0f +#define MAX9860_DVFLT_SHIFT 0 +#define MAX9860_DVFLT_COUNT 6 + +/* DACATTN */ +#define MAX9860_DVA 0xfe +#define MAX9860_DVA_SHIFT 1 +#define MAX9860_DVA_MUTE 0x5e + +/* ADCLEVEL */ +#define MAX9860_ADCRL 0xf0 +#define MAX9860_ADCRL_SHIFT 4 +#define MAX9860_ADCLL 0x0f +#define MAX9860_ADCLL_SHIFT 0 +#define MAX9860_ADCxL_MIN 15 + +/* DACGAIN */ +#define MAX9860_DVG 0x60 +#define MAX9860_DVG_SHIFT 5 +#define MAX9860_DVG_MAX 3 +#define MAX9860_DVST 0x1f +#define MAX9860_DVST_SHIFT 0 +#define MAX9860_DVST_MIN 31 + +/* MICGAIN */ +#define MAX9860_PAM 0x60 +#define MAX9860_PAM_SHIFT 5 +#define MAX9860_PAM_MAX 3 +#define MAX9860_PGAM 0x1f +#define MAX9860_PGAM_SHIFT 0 +#define MAX9860_PGAM_MIN 20 + +/* MICADC */ +#define MAX9860_AGCSRC 0x80 +#define MAX9860_AGCSRC_SHIFT 7 +#define MAX9860_AGCSRC_COUNT 2 +#define MAX9860_AGCRLS 0x70 +#define MAX9860_AGCRLS_SHIFT 4 +#define MAX9860_AGCRLS_COUNT 8 +#define MAX9860_AGCATK 0x0c +#define MAX9860_AGCATK_SHIFT 2 +#define MAX9860_AGCATK_COUNT 4 +#define MAX9860_AGCHLD 0x03 +#define MAX9860_AGCHLD_OFF 0x00 +#define MAX9860_AGCHLD_SHIFT 0 +#define MAX9860_AGCHLD_COUNT 4 + +/* NOISEGATE */ +#define MAX9860_ANTH 0xf0 +#define MAX9860_ANTH_SHIFT 4 +#define MAX9860_ANTH_MAX 15 +#define MAX9860_AGCTH 0x0f +#define MAX9860_AGCTH_SHIFT 0 +#define MAX9860_AGCTH_MIN 15 + +/* PWRMAN */ +#define MAX9860_SHDN 0x80 +#define MAX9860_DACEN 0x08 +#define MAX9860_DACEN_SHIFT 3 +#define MAX9860_ADCLEN 0x02 +#define MAX9860_ADCLEN_SHIFT 1 +#define MAX9860_ADCREN 0x01 +#define MAX9860_ADCREN_SHIFT 0 + +#endif /* _SND_SOC_MAX9860 */ -- 2.1.4

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[alsa-devel] [PATCH 1/2] ASoC: wm8985: add device tree binding for WM8985
by Petr Kulhavy 16 May '16

16 May '16

Add device tree binding for the WM8985 codec driver. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- Documentation/devicetree/bindings/sound/wm8985.txt | 36 ++++++++++++++++++++++ 1 file changed, 36 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/wm8985.txt diff --git a/Documentation/devicetree/bindings/sound/wm8985.txt b/Documentation/devicetree/bindings/sound/wm8985.txt new file mode 100644 index 000000000000..788d64766257 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wm8985.txt @@ -0,0 +1,36 @@ +Wolfson Microelectronics WM8985 and WM8758 audio codecs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : "wlf,wm8985" or "wlf,wm8758" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + +Pins on the device for linking into audio routes: + + * LIN : left microphone pre-amp negative input + * LIP : left microphone pre-amp positive input + * RIN : right microphone pre-amp negative input + * RIP : right microphone pre-amp positive input + * L2 : left line input / secondary pre-amp positive input + * R2 : right line input / secondary pre-amp positive input + * AUXL : left auxiliary input (WM8985 only) + * AUXR : right auxiliary input (WM8985 only) + + * HPL : left headphone / line output (the LOUT1 pin) + * HPR : right headphone / line output (the ROUT1 pin) + * SPKL : left headphone / line output (the LOUT2 pin) + * SPKR : right headphone / line output (the ROUT2 pin) + * Mic Bias : microphone bias + + +Example: + +wm8758: audio-codec@1a { + compatible = "wlf,wm8758"; + reg = <0x1a>; +}; -- 1.9.1

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[alsa-devel] ERROR messages when PROBE_DEFER ?
by Ricard Wanderlof 16 May '16

16 May '16

In sound/soc/soc-core.c:snd_bind_dai_link() (and a couple of other places), there are a number of dev_err printouts followed by return -EPROBE_DEFER. Should these really be ERROR messages, as a probe deferral is not an error? Admittedly, if there really is a problem, it's nice to get some information about it, so perhaps the error level should be relaxed somewhat? A case in point would be a SOC machine driver which needs a GPIO which is not loaded until after the kernel has booted. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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

16 May '16

The following changes since commit af9cc93c0dee5fc1f9fa32cd9d79a456738a21be: Merge tag 'asoc-fix-v4.6-rc5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus (2016-04-27 17:30:49 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.7 for you to fetch changes up to 515511a7920c69aebf7f5fef0cb8e1df6767f34c: Merge remote-tracking branch 'asoc/topic/hdmi' into asoc-next (2016-05-13 14:27:16 +0100) The diffstat looks terrible because it picks up a couple of -rcs from Linus as well as the actual new changes. ---------------------------------------------------------------- ASoC: Updates for v4.7 The updates this time around are almost all driver code: - Further slow progress on the topology code. - Substantial updates and improvements for the da7219, es8328, fsl-ssi Intel and rcar drivers. ---------------------------------------------------------------- Adam Borowski (1): perf/x86/amd: Set the size of event map array to PERF_COUNT_HW_MAX Adam Thomson (9): ASoC: da7219: Update PLL ranges and dividers to improve locking ASoC: da7219: Disallow unsupported 32KHz clock setting in set_dai_sysclk() ASoC: da7218: Update PLL ranges and dividers to improve locking ASoC: da7219: Add initial ACPI id for device ASoC: da7213: Add DAI DAPM event to control DAI clocks ASoC: da7213: Add checking of SRM lock status before enabling DAI ASoC: da7213: Default PC counter to free-running when DAI disabled ASoC: da7213: Update PLL ranges to improve locking at frequency boundary ASoC: da7213: Allow PLL disable/bypass when using 32KHz sysclk Adrian Hunter (1): mmc: sdhci-acpi: Reduce Baytrail eMMC/SD/SDIO hangs Alex Deucher (4): Revert "drm/amdgpu: disable runtime pm on PX laptops without dGPU power control" drm/amdgpu: print a message if ATPX dGPU power control is missing drm/radeon: make sure vertical front porch is at least 1 drm/amdgpu: make sure vertical front porch is at least 1 Alex Thorlton (1): x86/platform/UV: Bring back the call to map_low_mmrs in uv_system_init Alexander Duyck (2): net: Disable segmentation if checksumming is not supported vxlan: Add checksum check to the features check function Alexander Kurz (1): ARM: mach-imx: sdhci-esdhc-imx: initialize DMA mask Alexander Potapenko (1): lib/stackdepot.c: allow the stack trace hash to be zero Alexander Shishkin (1): perf/x86/intel/pt: Don't die on VMXON Alexandre Bounine (1): rapidio/mport_cdev: fix uapi type definitions Alexei Starovoitov (3): bpf: fix refcnt overflow bpf: fix check_map_func_compatibility logic samples/bpf: fix trace_output example Alexey Brodkin (2): ARC: support generic per-device coherent dma mem ARC: add support for reserved memory defined by device tree Ananth N Mavinakayanahalli (1): Ananth has moved Andi Kleen (2): perf/x86/intel: Add model number for Skylake Server to perf perf/x86: Add model numbers for Kabylake CPUs Andrea Arcangeli (1): mm: thp: kvm: fix memory corruption in KVM with THP enabled Andreas Larsson (1): MAINTAINERS: net: Change maintainer for GRETH 10/100/1G Ethernet MAC device driver Andrew Gabbasov (1): udf: Fix conversion of 'dstring' fields to UTF8 Andrey Ryabinin (1): kcov: don't profile branches in kcov Andy Gross (2): Revert "dts: msm8974: Add blsp2_bam dma node" Revert "dts: msm8974: Add dma channels for blsp2_i2c1 node" Andy Lutomirski (1): Revert "x86/mm/32: Set NX in __supported_pte_mask before enabling paging" Anna-Maria Gleixner (1): net: mvneta: Remove superfluous SMP function call Anton Blanchard (1): powerpc: Fix bad inline asm constraint in create_zero_mask() Antonio Quartulli (2): batman-adv: fix DAT candidate selection (must use vid) batman-adv: B.A.T.M.A.N V - make sure iface is reactivated upon NETDEV_UP event Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. Arnd Bergmann (7): Merge tag 'qcom-fixes-for-4.6-rc2' of git://git.kernel.org/.../agross/linux into fixes Merge tag 'samsung-fixes-4.6' of git://git.kernel.org/.../krzk/linux into fixes Merge tag 'omap-for-v4.6/fixes-rc5-signed' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'renesas-fixes2-for-v4.6' of git://git.kernel.org/.../horms/renesas into fixes ARM: davinci: only use NVMEM when available net/smscx5xx: use the device tree for mac address byteswap: try to avoid __builtin_constant_p gcc bug Ashish Samant (1): fuse: Fix return value from fuse_get_user_pages() Ashutosh Dixit (1): misc: mic: Fix for double fetch security bug in VOP driver Atsushi Kumagai (2): kexec: update VMCOREINFO for compound_order/dtor kexec: export OFFSET(page.compound_head) to find out compound tail page Ayala Beker (1): iwlwifi: mvm: avoid to WARN about gscan capabilities Azael Avalos (1): toshiba_acpi: Fix regression caused by hotkey enabling value Babu Moger (1): sparc/PCI: Fix for panic while enabling SR-IOV Bard Liao (1): ASoC: rt5645: polling jd status in all conditions Bert Kenward (1): MAINTAINERS: net: update sfc maintainers Bin Liu (1): Revert "usb: musb: musb_host: Enable HCD_BH flag to handle urb return in bottom half" Boqun Feng (1): locking/lockdep: Fix ->irq_context calculation Borislav Petkov (1): sparc/defconfigs: Remove CONFIG_IPV6_PRIVACY Brijesh Singh (1): ata: add AMD Seattle platform driver Caleb Crome (1): ASoC: fsl_ssi: add CCSR_SSI_SOR to volatile register list Charles Keepax (9): ASoC: wm_adsp: Fix some subtle races on compressed stream ASoC: wm_adsp: Show avail in bytes to match other messages ASoC: cs47l24: Fix a couple of small whitespace errors ASoC: cs47l24: Add support for audio trace firmware ASoC: wm_adsp: Factor out fetching of stream errors from the DSP ASoC: wm_adsp: Improve DSP error handling ASoC: wm_adsp: Add support for TLV based binary controls ASoC: wm_adsp: Move compr_attach/attached functions ASoC: wm_adsp: Detach compressed stream on free Charmaine Lee (2): drm/vmwgfx: Enable SVGA_3D_CMD_DX_SET_PREDICATION drm/vmwgfx: use vmw_cmd_dx_cid_check for query commands. Chen Yu (1): x86/tsc: Read all ratio bits from MSR_PLATFORM_INFO Chen-Yu Tsai (1): mmc: sunxi: Disable eMMC HS-DDR (MMC_CAP_1_8V_DDR) for Allwinner A80 Chris Wilson (1): drm/i915: Avoid stalling on pending flips for legacy cursor updates Christoph Hellwig (1): IB/iser: Fix max_sectors calculation Christophe Jaillet (1): ps3_gelic: fix memcpy parameter Chunyu Hu (1): tracing: Don't display trigger file for events that can't be enabled Colin Ian King (1): net: dsa: mv88e6xxx: fix uninitialized error return Craig Gallek (1): soreuseport: Fix TCP listener hash collision Dan Carpenter (4): md/raid0: fix uninitialized variable bug firmware: qemu_fw_cfg.c: potential unintialized variable virtio: Silence uninitialized variable warning ASoC: hdac_hdmi: Potential NULL deref in hdac_hdmi_get_spk_alloc() Dan Streetman (1): mm/zswap: provide unique zpool name Dan Williams (2): libnvdimm, pfn: fix memmap reservation sizing nfit: fix translation of command status results Daniel Baluta (1): iio: imu: mpu6050: Fix name/chip_id when using ACPI Dave Airlie (6): Merge branch 'drm-etnaviv-fixes' of git://git.pengutronix.de:/git/lst/linux into drm-fixes Merge branch 'drm-vmwgfx-fixes' of git://people.freedesktop.org/~syeh/repos_linux into drm-fixes Merge branch 'drm-fixes-4.6' of git://people.freedesktop.org/~agd5f/linux into drm-fixes drm/amdgpu: set metadata pointer to NULL after freeing. Merge branch 'drm-fixes-4.6' of git://people.freedesktop.org/~agd5f/linux into drm-fixes Merge tag 'drm-intel-fixes-2016-05-02' of git://anongit.freedesktop.org/drm-intel into drm-fixes Dave Gerlach (1): ARM: OMAP3: Fix external abort on 36xx waking from off mode idle David Ahern (1): net: ipv6: Delete host routes on an ifdown David Rivshin (5): drivers: net: cpsw: fix parsing of phy-handle DT property in dual_emac config drivers: net: cpsw: fix segfault in case of bad phy-handle drivers: net: cpsw: don't ignore phy-mode if phy-handle is used dt: cpsw: phy-handle, phy_id, and fixed-link are mutually exclusive drivers: net: cpsw: use of_phy_connect() in fixed-link case David S. Miller (19): sparc: Write up preadv2/pwritev2 syscalls. Merge branch 'bridge-mdb-fixes' Merge branch 'macsec-fixes' Merge branch 'mlx5-fixes' ipv6: Revert optional address flusing on ifdown. Revert "ipv6: Revert optional address flusing on ifdown." Merge branch 'bnxt_en-fixes' sparc64: Fix bootup regressions on some Kconfig combinations. Merge tag 'wireless-drivers-for-davem-2016-04-25' of git://git.kernel.org/.../kvalo/wireless-drivers Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge tag 'mac80211-for-davem-2016-04-27' of git://git.kernel.org/.../jberg/mac80211 Merge branch 'gre-lwt-fixes' Merge branch 'pegasus-sizes' Merge branch 'cpsw-phy-handle-fixes' Merge branch 'bpf-fixes' Merge tag 'batman-adv-fix-for-davem' of git://git.open-mesh.org/linux-merge Merge branch 'mlx5-fixes' Merge branch 'tunnel-csum-and-sg-offloads' Merge branch 'rds-fixes' Dean Luick (1): IB/hfi1: Use kernel default llseek for ui device Dharageswari.R (2): ASoC: Intel: Skylake: Fix the NULL pointer exception in dsp_clean up ASoC: Intel: Skylake: Prevent sending Set DMA Control IPC if the widget is "On" Dmitry V. Levin (1): parisc: fix a bug when syscall number of tracee is __NR_Linux_syscalls Doug Ledford (1): IB/core: Fix oops in ib_cache_gid_set_default_gid Elad Raz (3): switchdev: Adding complete operation to deferred switchdev ops bridge: mdb: Common function for mdb entry translation bridge: mdb: Marking port-group as offloaded Eli Cohen (1): net/mlx5_core: Remove static from local variable Eric Dumazet (1): net/mlx4_en: fix spurious timestamping callbacks Eric Engestrom (3): Documentation: dt: arc: fix spelling mistakes Documentation: networking: fix spelling mistakes MAINTAINERS: fix Rajendra Nayak's address Eric W. Biederman (1): propogate_mnt: Handle the first propogated copy being a slave Evgeny Voevodin (1): ARCv2: Enable LOCKDEP Fabio Estevam (3): ASoC: fsl_sai: Introduce a compatible string for MX6UL ASoC: fsl_sai: Allow setting the SAI MCLK direction MAINTAINERS: Add myself as reviewer of FSL/NXP SoC sound drivers Flora Cui (2): drm/ttm: fix kref count mess in ttm_bo_move_to_lru_tail drm/amdgpu: disable vm interrupts with vm_fault_stop=2 Florian Fainelli (1): net: macb: Probe MDIO bus before registering netdev Florian Meier (1): ASoC: pcm5102a: Add support for PCM5102A codec Florian Westphal (1): RDMA/nes: don't leak skb if carrier down Frank Rowand (1): .mailmap: add Frank Rowand Franklin S Cooper Jr (2): ARM: dts: am33xx: Fix GPMC dma properties ARM: dts: am437x: Fix GPMC dma properties Gal Pressman (1): net/mlx5: Unmap only the relevant IO memory mapping Geert Uytterhoeven (2): ARM: shmobile: timer: Fix preset_lpj leading to too short delays arm64: dts: r8a7795: Don't disable referenced optional scif clock Gerald Schaefer (2): s390/mm: fix asce_bits handling with dynamic pagetable levels numa: fix /proc/<pid>/numa_maps for THP Greg Kroah-Hartman (3): Merge tag 'iio-fixes-for-4.6c' of git://git.kernel.org/.../jic23/iio into staging-linus Merge tag 'usb-serial-4.6-rc6' of git://git.kernel.org/.../johan/usb-serial into usb-linus Merge tag 'iio-fixes-for-4.6d' of git://git.kernel.org/.../jic23/iio into staging-linus Grygorii Strashko (1): MAINTAINERS: net: add entry for TI Ethernet Switch drivers Gustavo Padovan (1): drm/virtio: send vblank event after crtc updates H. Nikolaus Schaller (2): Input: twl6040-vibra - fix atomic schedule panic ARM: dts: omap5: fix range of permitted wakeup pinmux registers Hamish Martin (1): tipc: only process unicast on intended node Hans Verkuil (2): [media] davinci_vpfe: Revert "staging: media: davinci_vpfe: remove,unnecessary ret variable" [media] v4l2-dv-timings.h: fix polarity for 4k formats Hans de Goede (1): ARM: dts: sun8i-q8-common: Do not set constraints on dc1sw regulator Hariprasad S (1): RDMA/iw_cxgb4: Fix bar2 virt addr calculation for T4 chips Heikki Krogerus (1): device property: Avoid potential dereferences of invalid pointers Heinrich Schuchardt (2): ASoC: fsl: imx-pcm-fiq: use correct format specifier ASoC: au1x: use correct format specifier Horia Geant? (1): crypto: talitos - fix AEAD tcrypt tests Howard Cochran (1): writeback: Fix performance regression in wb_over_bg_thresh() Hugh Dickins (2): huge pagecache: mmap_sem is unlocked when truncation splits pmd mm, cma: prevent nr_isolated_* counters from going negative Ilya Dryomov (3): libceph: make authorizer destruction independent of ceph_auth_client rbd: fix rbd map vs notify races rbd: report unsupported features to syslog Imre Deak (2): drm/i915: Fix system resume if PCI device remained enabled drm/i915/ddi: Fix eDP VDD handling during booting and suspend/resume Ingo Molnar (1): Merge tag 'efi-urgent' of git://git.kernel.org/.../mfleming/efi into efi/urgent Ivan Babrou (1): net: dummy: remove note about being Y by default Ivaylo Dimitrov (2): ARM: dts: omap3: Fix ISP syscon register offset ARM: dts: omap3-n900: Specify peripherals LDO regulators initial mode James Liao (1): Revert "soc: mediatek: SCPSYS: Fix double enabling of regulators" James Morse (2): ARM: cpuidle: Pass on arm_cpuidle_suspend()'s return value kcov: don't trace the code coverage code Jan Beulich (1): xen/evtchn: fix ring resize when binding new events Jann Horn (1): bpf: fix double-fdput in replace_map_fd_with_map_ptr() Jasem Mutlaq (1): USB: serial: cp210x: add Straizona Focusers device ids Jason Baron (1): mm: update min_free_kbytes from khugepaged after core initialization Jason Gunthorpe (1): IB/security: Restrict use of the write() interface Javier Martinez Canillas (2): i2c: exynos5: Fix possible ABBA deadlock by keeping I2C clock prepared ASoC: cs42l56: Use IS_ENABLED() instead of checking for built-in or module Jean-Philippe Brucker (2): ARM: 8571/1: nommu: fix PMSAv7 setup ARM: 8572/1: nommu: change memory reserve for the vectors Jeeja KP (4): ASoC: Intel: Bxtn: Add Broxton DSP support ASoC: Intel: Skylake: Suspend PCMs when marked as active suspend ASoC: Intel: Boards: remove ignore_suspend for WoV streams ASoC: Intel: Skylake: Fix memory leak in nhlt init Jim Lodes (1): ASoC: davinci-mcasp: Fix overwriting of ahclkx Jiri Benc (5): cxgbi: fix uninitialized flowi6 gre: do not assign header_ops in collect metadata mode gre: build header correctly for collect metadata tunnels gre: reject GUE and FOU in collect metadata mode gre: do not pull header in ICMP error processing Johan Hovold (1): Revert "USB / PM: Allow USB devices to remain runtime-suspended when sleeping" Johannes Berg (1): mac80211: fix statistics leak if dev_alloc_name() fails Johannes Weiner (1): mm: memcontrol: let v2 cgroups follow changes in system swappiness John Keeping (11): ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables" ASoC: rockchip: Revert "ASoC: rockchip: i2s: separate capture and playback" ASoC: es8328: Move clock setup to hw_params ASoC: es8328: Fix ADC format setup ASoC: es8328: Fix mask for VMIDSEL ASoC: es8328: Use single R/W for regmap ASoC: es8328: Move sample size setup to hw_params ASoC: es8328: Support more sample formats ASoC: es8328: Support more sample rates ASoC: es8328: Set symmetric rates ASoC: dapm: deprecate MICBIAS widget type John Lin (1): ASoC: rt5645: Add dmi_system_id "Google Setzer" John Paul Adrian Glaubitz (2): sparc: Implement and wire up modalias_show for vio. sparc: Implement and wire up vio_hotplug for vio. Jon Cooper (1): sfc: disable RSS when unsupported Jonas Eymann (1): crypto: talitos - fix crash in talitos_cra_init() Joonas Lahtinen (1): ASoC: Intel: Fix printk formatting Joonsoo Kim (1): lib/stackdepot: avoid to return 0 handle Jose Abreu (2): ASoC: dwc: Unmask I2S interrupts only for enabled channels ASoC: dwc: Use fifo depth to program FCR Josh Boyer (1): x86/efi-bgrt: Switch all pr_err() to pr_notice() for invalid BGRT Jubin John (1): IB/rdmavt: Fix send scheduling Jyri Sarha (3): ALSA: pcm: add IEC958 channel status helper for hw_params ALSA: pcm: Allow 32 bit sample format in IEC958 channel status helper ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders K. Y. Srinivasan (1): Drivers: hv: vmbus: Fix signaling logic in hv_need_to_signal_on_read() Kalle Valo (1): Merge tag 'iwlwifi-for-kalle-2016-04-12_2' of https://git.kernel.org/.../iwlwifi/iwlwifi-fixes Kan Liang (1): perf/x86/intel: Fix incorrect lbr_sel_mask value Keerthy (2): ARM: dts: dra7: Correct clock tree for sys_32k_ck ARM: DRA7: clockdomain: Implement timer workaround for errata i874 Keith Busch (1): x86/apic: Handle zero vector gracefully in clear_vector_irq() Kevin Hilman (7): Merge tag 'v4.6-soc-fixes' of https://github.com/mbgg/linux-mediatek into fixes Merge tag 'omap-for-v4.6/fixes-rc2-v2-signed' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'renesas-fixes-for-v4.6' of git://git.kernel.org/.../horms/renesas into fixes Merge tag 'omap-for-v4.6/fixes-rc3' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'imx-fixes-4.6' of git://git.kernel.org/.../shawnguo/linux into fixes Merge tag 'omap-for-v4.6/dt-ti81xx-signed' of git://git.kernel.org/.../tmlind/linux-omap into fixes Merge tag 'sunxi-fixes-for-4.6' of https://git.kernel.org/.../mripard/linux into fixes Khalid Aziz (1): sparc64: recognize and support Sonoma CPU type Kirill A. Shutemov (1): thp: keep huge zero page pinned until tlb flush Knut Wohlrab (1): Input: zforce_ts - fix dual touch recognition Konstantin Khlebnikov (2): mm/huge_memory: replace VM_NO_THP VM_BUG_ON with actual VMA check mm/memory-failure: fix race with compound page split/merge Krzysztof Kozlowski (3): iio: ak8975: Fix NULL pointer exception on early interrupt ARM: EXYNOS: Properly skip unitialized parent clock in power domain on mailmap: fix Krzysztof Kozlowski's misspelled name Laszlo Ersek (1): efi: Fix out-of-bounds read in variable_matches() Leo Yan (2): thermal: hisilicon: increase temperature resolution thermal: use %d to print S32 parameters Linus Lüssing (1): batman-adv: Fix broadcast/ogm queue limit on a removed interface Linus Torvalds (66): Merge branch 'linus' of git://git.kernel.org/.../herbert/crypto-2.6 devpts: more pty driver interface cleanups Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge git://git.kernel.org/.../davem/net Merge tag 'platform-drivers-x86-v4.6-3' of git://git.infradead.org/users/dvhart/linux-platform-drivers-x86 Merge tag 'nios2-v4.6-fix' of git://git.kernel.org/.../lftan/nios2 Merge tag 'arc-4.6-rc6-fixes' of git://git.kernel.org/.../vgupta/arc Merge branch 'i2c/for-current' of git://git.kernel.org/.../wsa/linux Merge branch 'for-4.6-fixes' of git://git.kernel.org/.../tj/cgroup Merge branch 'for-4.6-fixes' of git://git.kernel.org/.../tj/wq Merge branch 'for-linus' of git://git.kernel.org/.../s390/linux Merge branch 'for-linus' of git://git.kernel.org/.../sage/ceph-client Merge tag 'sound-4.6-rc6' of git://git.kernel.org/.../tiwai/sound Merge tag 'media/v4.6-4' of git://git.kernel.org/.../mchehab/linux-media Merge branch 'efi-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'locking-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'akpm' (patches from Andrew) Merge tag 'for-linus' of git://git.kernel.org/.../dledford/rdma Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'mmc-v4.6-rc4' of git://git.linaro.org/people/ulf.hansson/mmc Merge tag 'pm+acpi-4.6-rc6' of git://git.kernel.org/.../rafael/linux-pm Merge tag 'edac_fix_for_4.6' of git://git.kernel.org/.../bp/bp Merge tag 'powerpc-4.6-4' of git://git.kernel.org/.../powerpc/linux Merge branch 'fixes' of git://git.kernel.org/.../evalenti/linux-soc-thermal Linux 4.6-rc6 Merge git://git.kernel.org/.../davem/sparc Merge git://git.kernel.org/.../davem/net Merge tag 'gpio-v4.6-4' of git://git.kernel.org/.../linusw/linux-gpio Merge branch 'for_linus' of git://git.kernel.org/.../jack/linux-fs Merge tag 'md/4.6-rc6-fix' of git://git.kernel.org/.../shli/md Minimal fix-up of bad hashing behavior of hash_64() Merge tag 'clk-fixes-for-linus' of git://git.kernel.org/.../clk/linux Merge branch 'for-linus' of git://git.kernel.org/.../jikos/hid Merge branch 'for-linus' of git://git.kernel.org/.../mszeredi/fuse Merge git://git.kernel.org/.../davem/net Merge tag 'trace-fixes-v4.6-rc6' of git://git.kernel.org/.../rostedt/linux-trace Merge tag 'for-linus-4.6-rc6-tag' of git://git.kernel.org/.../xen/tip Merge branch 'for-linus' of git://git.kernel.org/.../jmorris/linux-security Merge branch 'for-linus' of git://git.kernel.org/.../dtor/input Merge tag 'for_linus' of git://git.kernel.org/.../mst/vhost Merge branch 'for-linus' of git://git.kernel.org/.../ebiederm/user-namespace Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge tag 'asm-generic-4.6' of git://git.kernel.org/.../arnd/asm-generic Merge branch 'libnvdimm-fixes' of git://git.kernel.org/.../nvdimm/nvdimm mailmap: add John Paul Adrian Glaubitz Merge branch 'akpm' (patches from Andrew) Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'powerpc-4.6-5' of git://git.kernel.org/.../powerpc/linux Merge tag 'arc-4.6-rc7-fixes' of git://git.kernel.org/.../vgupta/arc Merge branch 'parisc-4.6-5' of git://git.kernel.org/.../deller/parisc-linux Merge branch 'efi-urgent-for-linus' of git://git.kernel.org/.../tip/tip 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 tag 'pm+acpi-4.6-rc7' of git://git.kernel.org/.../rafael/linux-pm Merge branch 'x86-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'for-linus' of git://git.kernel.dk/linux-block Merge tag 'for-linus' of git://git.kernel.org/.../dledford/rdma Merge branch 'for-4.6-fixes' of git://git.kernel.org/.../tj/libata Merge tag 'media/v4.6-5' of git://git.kernel.org/.../mchehab/linux-media Merge branch 'fixes' of git://git.armlinux.org.uk/~rmk/linux-arm Merge tag 'usb-4.6-rc7' of git://git.kernel.org/.../gregkh/usb Merge tag 'staging-4.6-rc7' of git://git.kernel.org/.../gregkh/staging Merge tag 'char-misc-4.6-rc7' of git://git.kernel.org/.../gregkh/char-misc Linux 4.6-rc7 Linus Walleij (2): Revert "gpio: rcar: Add Runtime PM handling for interrupts" Revert "gpio: rcar: Fine-grained Runtime PM support" Lothar Waßmann (1): clk: imx6q: fix typo in CAN clock definition Lucas Stach (1): drm/etnaviv: don't move linear memory window on 3D cores without MC2.0 Lyude (1): drm/dp/mst: Restore primary hub guid on resume Majd Dibbiny (2): net/mlx5_core: Add ConnectX-5 to list of supported devices net/mlx5: Add pci shutdown callback Manish Chopra (1): qlcnic: Update version to 5.3.64 Maor Gottlieb (1): net/mlx5_core: Fix soft lockup in steering error flow Marcelo Ricardo Leitner (1): net: fix net_gso_ok for new GSO types. Marek Lindner (1): batman-adv: init neigh node last seen field Marek Szyprowski (2): [media] media: exynos4-is: fix deadlock on driver probe [media] media: s3c-camif: fix deadlock on driver probe() Marek Vasut (2): iio:adc:at91-sama5d2: Repair crash on module removal net: stmmac: socfpga: Remove re-registration of reset controller Mark Brown (18): Merge branch 'topic/dmaengine' of git://git.kernel.org/.../broonie/sound into asoc-bcm2835 Merge branch 'topic/hda' of git://git.kernel.org/.../broonie/sound into asoc-hdmi Merge tag 'asoc-fix-v4.6-rc5' into asoc-intel Merge branch 'topic/arizona' of git://git.kernel.org/.../broonie/sound into asoc-adsp ASoC: da7129: Add missing include of acpi.h Merge branch 'topic/dai-link' of git://git.kernel.org/.../broonie/sound into asoc-intel Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/fsl-ssi', 'asoc/fix/rockchip' and 'asoc/fix/rt286' into asoc-linus Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/dmaengine' into asoc-next Merge remote-tracking branch 'asoc/topic/imx' into asoc-next Merge remote-tracking branch 'asoc/topic/intel' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm5102' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/ak4624', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/bcm2835', 'asoc/topic/cs42l56', 'asoc/topic/da7213', 'asoc/topic/da7218' and 'asoc/topic/da7219' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci' and 'asoc/topic/dwc' into asoc-next Merge remote-tracking branches 'asoc/topic/es8328', 'asoc/topic/find-dai', 'asoc/topic/fsl', 'asoc/topic/fsl-sai' and 'asoc/topic/fsl-ssi' into asoc-next Merge remote-tracking branch 'asoc/topic/hdmi' into asoc-next Martin Schwidefsky (1): s390/sclp_ctl: fix potential information leak with /dev/sclp Masahiro Yamada (1): arm64: dts: uniphier: fix I2C nodes of PH1-LD20 Mathias Krause (1): proc: prevent accessing /proc/<PID>/environ until it's ready Matt Fleming (1): MAINTAINERS: Remove asterisk from EFI directory names Matt Ranostay (1): iio: imu: mpu6050: fix possible NULL dereferences Matthew Finlay (3): net/mlx5: Kconfig: Fix MLX5_EN/VXLAN build issue net/mlx5e: Implement a mlx5e workqueue net/mlx5e: Use workqueue for vxlan ops Matthias Reichl (4): ASoC: bcm2835: add 24bit support ASoC: bcm2835: setup clock only if CPU is clock master ASoC: dmaengine_pcm: Add support for packed transfers ASoC: bcm2835: Add S16_LE support via packed DMA transfers Matti Gottlieb (1): iwlwifi: mvm: fix accessing Null pointer during fw dump collection Mauro Carvalho Chehab (1): [media] media-device: fix builds when USB or PCI is compiled as module Mengdong Lin (3): ASoC: Change DAI link's be_id to a generic id ASoC: Export snd_soc_find_dai() ASoC: Add kerneldoc comments for snd_soc_find_dai Michael Chan (3): bnxt_en: Don't fallback to INTA on VF. bnxt_en: Limit RX BD pages to be no bigger than 32K. bnxt_en: Divide a page into 32K buffers for the aggregation ring if necessary. Michael Ellerman (1): i2c: cpm: Fix build break due to incompatible pointer types Michael Neuling (2): cxl: Keep IRQ mappings on context teardown cxl: Poll for outstanding IRQs when detaching a context Michal Schmidt (1): cxgb3: fix out of bounds read Michał Pecio (1): md/raid0: remove empty line printk from dump_zones Mika Kahola (1): drm/i915: Fix eDP low vswing for Broadwell Mika Westerberg (1): i2c: ismt: Add Intel DNV PCI ID Mike Manning (1): USB: serial: cp210x: add ID for Link ECU Mike Marciniszyn (1): IB/hfi1: Fix missing lock/unlock in verbs drain callback Miklos Szeredi (1): fuse: update mailing list in MAINTAINERS Mimi Zohar (1): ima: fix the string representation of the LSM/IMA hook enumeration ordering Minchan Kim (3): mm: vmscan: reclaim highmem zone if buffer_heads is over limit mm: call swap_slot_free_notify() with page lock held mm/hwpoison: fix wrong num_poisoned_pages accounting Mitko Haralanov (4): IB/hfi1: Prevent NULL pointer deferences in caching code IB/hfi1: Fix deadlock caused by locking with wrong scope IB/hfi1: Prevent unpinning of wrong pages IB/hfi1: Don't attempt to free resources if initialization failed Mousumi Jana (1): ASoC: Intel: Skylake: Fix memory leak during init instance Nazar Mokrynskyi (1): HID: Fix boot delay for Creative SB Omni Surround 5.1 with quirk Neil Armstrong (1): net: ethernet: davinci_emac: Fix devioctl while in fixed link Neil Horman (1): netem: Segment GSO packets on enqueue Nick Dyer (1): Input: atmel_mxt_ts - use mxt_acquire_irq in mxt_soft_reset Nicolas Dichtel (1): ipv6/ila: fix nlsize calculation for lwtunnel Nishanth Menon (3): ARM: OMAP: DRA7: wakeupgen: Skip SAR save for wakeupgen ARM: OMAP: DRA7: Provide proper class to omap2_set_globals_tap ARM: OMAP: Catch callers of revision information prior to it being populated Oleksij Rempel (1): ath9k: ar5008_hw_cmn_spur_mitigate: add missing mask_m & mask_p initialisation Oren Givon (1): iwlwifi: add device IDs for the 8265 device Paolo Abeni (2): ipv4/fib: don't warn when primary address is missing if in_dev is dead ip_tunnel: fix preempt warning in ip tunnel creation/updating Pardha Saradhi K (1): ASoC: Intel: Skylake: Add more SSP DAIs Peter Rosin (1): ASoC: atmel_ssc_dai: read DSP mode A data on rising edges of bclk Peter Ujfalusi (5): ASoC: davinci-mcasp: Do not allow multiple streams in one direction ASoC: davinci-mcasp: Use defines for clkdiv IDs ASoC: davinci-mcasp: Change __davinci_mcasp_set_clkdiv() first parameter ASoC: davinci-mcasp: Restructure the davinci_mcasp_calc_clk_div() ASoC: davinci-mcasp: Calculate AUXCLK divider when setting up master clocks Peter Zijlstra (5): perf/core: Make sysctl_perf_cpu_time_max_percent conform to documentation lockdep: Fix lock_chain::base size nohz/full, sched/rt: Fix missed tick-reenabling bug in sched_can_stop_tick() perf/core: Fix perf_event_open() vs. execve() race perf/x86/amd/iommu: Do not register a task ctx for uncore like PMUs Petko Manolov (2): pegasus: fixes URB buffer allocation size; pegasus: fixes reported packet length Petr Kulhavy (2): ASoC: davinci-mcbsp: add binding for McBSP ASoC: Davinci: McBSP: add device tree support for McBSP Philipp Zabel (4): ASoC: hdmi-codec: Add ELD control ASoC: hdmi-codec: Add ELD control gpu: ipu-v3: Fix imx-ipuv3-crtc module autoloading modpost: fix module autoloading for OF devices with generic compatible property Ping Cheng (2): HID: wacom: Add support for DTK-1651 HID: wacom: add missed stylus_in_proximity line back Prarit Bhargava (1): ACPICA: Dispatcher: Update thread ID for recursive method calls Rafael J. Wysocki (6): Revert "cpufreq: governor: Fix negative idle_time when configured with CONFIG_HZ_PERIODIC" Merge branch 'pm-cpufreq-fixes' cpufreq: intel_pstate: Fix HWP on boot CPU after system resume intel_pstate: Fix intel_pstate_get() Merge branches 'acpica-fixes' and 'device-properties-fixes' Merge branches 'pm-opp-fixes', 'pm-cpufreq-fixes' and 'pm-cpuidle-fixes' Ramesh Babu (1): ASoC: Intel: Add Broxton-P machine driver Rana Shahout (1): net/mlx5e: Fix MLX5E_100BASE_T define Randy Dunlap (1): ASoC: intel: add function stub when ACPI is not enabled Ricardo Ribalda Delgado (2): [media] media: vb2: Fix regression on poll() for RW mode [media] vb2-memops: Fix over allocation of frame vectors Richard Fitzgerald (6): ASoC: arizona: Do not create OUT4R widget for CS47L24/WM1831 ASoC: arizona: Prefer lower FRATIO in pseudo-fractional mode ASoC: arizona: No need to update_bits when writing AEC clock control ASoC: wm_adsp: free memory when unloaded or closed ASoC: arizona: call wm_adsp2_remove when codec driver is removed ASoC: wm_adsp: factor out freeing of alg regions Richard Leitner (1): iio: ak8975: fix maybe-uninitialized warning Roger Quadros (1): ARM: dts: am57xx-beagle-x15: remove extcon_usb1 Romain Perier (1): nios2: memset: use the right constraint modifier for the %4 output operand Roman Pen (1): workqueue: fix ghost PENDING flag while doing MQ IO Ross Lagerwall (3): xen: Fix page <-> pfn conversion on 32 bit systems xen/balloon: Fix crash when ballooning on x86 32 bit PAE xen/qspinlock: Don't kick CPU if IRQ is not initialized Rui Salvaterra (1): powerpc: wire up preadv2 and pwritev2 syscalls Russell King (1): maintainers: update rmk's email address(es) Sabrina Dubroca (9): macsec: add missing NULL check after kmalloc macsec: take rtnl lock before for_each_netdev macsec: don't put a NULL rxsa macsec: fix rx_sa refcounting with decrypt callback macsec: add consistency check to netlink dumps macsec: fix memory leaks around rx_handler (un)registration macsec: fix SA leak if initialization fails macsec: add missing macsec prefix in uapi macsec: fix netlink attribute validation Saeed Mahameed (3): net/mlx5e: Device's mtu field is u16 and not int net/mlx5e: Fix minimum MTU net/mlx5e: Use vport MTU rather than physical port MTU Sagi Grimberg (3): IB/core: Don't drain non-existent rq queue-pair IB/mlx5: Expose correct max_sge_rd limit MAINTAINERS: Update iser/isert maintainer contact info Sakari Ailus (2): [media] videobuf2-core: Check user space planes array in dqbuf [media] videobuf2-v4l2: Verify planes array in buffer dequeueing Sara Sharon (1): iwlwifi: 8000: fix MODULE_FIRMWARE input Sascha Hauer (1): ARM: SoCFPGA: Fix secondary CPU startup in thumb2 kernel Sebastian Ott (1): s390/pci: fix use after free in dma_init Sergei Shtylyov (2): pxa168_eth: fix mdiobus_scan() error check macb: fix mdiobus_scan() error check Shaohua Li (2): MD: make bio mergeable raid5: delete unnecessary warnning Shashank Sharma (1): drm/i915: Fake HDMI live status Shreyas NC (1): ASoC: Intel: Skylake: Use UUID in binary format Sinclair Yeh (1): drm/vmwgfx: Fix order of operation Sjoerd Simons (2): ARM: dts: r8a7791: Don't disable referenced optional clocks Revert "ARM: dts: porter: Enable SCIF_CLK frequency and pins" Sowmini Varadhan (3): sparc/pci: Refactor dev_archdata initialization into pci_init_dev_archdata RDS:TCP: Synchronize rds_tcp_accept_one with rds_send_xmit when resetting t_sock RDS: TCP: Synchronize accept() and connect() paths on t_conn_lock. Srinivas Kandagatla (3): libahci: save port map for forced port map ata: ahci-platform: Add ports-implemented DT bindings. ARM: dts: apq8064: add ahci ports-implemented mask Srinivas Pandruvada (2): perf/x86/intel/rapl: Add missing Haswell model cpufreq: intel_pstate: Fix processing for turbo activation ratio Stanislav Meduna (1): nvmem: mxs-ocotp: fix buffer overflow in read Stanislaw Gruszka (1): myri10ge: fix sleeping with bh disabled Stephen Boyd (2): Merge tag 'imx-clk-fixes-4.6' of git://git.kernel.org/.../shawnguo/linux into clk-fixes ASoC: ak4642: Remove CLK_IS_ROOT Stephen Rothwell (1): ASoC: Intel: fix up for DAI link's be_id change Steve Capper (1): mm: exclude HugeTLB pages from THP page_mapped() logic Steve Wise (3): iw_cxgb4: initialize ibdev.iwcm->ifname for port mapping iw_cxgb3: initialize ibdev.iwcm->ifname for port mapping iw_cxgb4: handle draining an idle qp Subhransu S. Prusty (9): ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: parse eld for channel map capability ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params Sudeep Holla (2): drivers: firmware: psci: unify enable-method binding on ARM {64,32}-bit systems cpufreq: st: enable selective initialization based on the platform Sven Eckelmann (5): batman-adv: Check skb size before using encapsulated ETH+VLAN header batman-adv: Deactivate TO_BE_ACTIVATED hardif on shutdown batman-adv: Reduce refcnt of removed router when updating route batman-adv: Fix reference counting of vlan object for tt_local_entry batman-adv: Fix reference counting of hardif_neigh_node object for neigh_node Tal Shorer (1): usb: musb: gadget: nuke endpoint before setting its descriptor to NULL Tanmay Jagdale (1): i2c: xlp9xx: add support for Broadcom Vulcan Tejun Heo (2): cgroup, cpuset: replace cpuset_post_attach_flush() with cgroup_subsys->post_attach callback memcg: relocate charge moving from ->attach to ->post_attach Tim Bingham (1): net: Implement net_dbg_ratelimited() for CONFIG_DYNAMIC_DEBUG case Timur Tabi (1): net: phy: at803x: only the AT8030 needs a hardware reset on link change Tomi Valkeinen (2): ARM: dts: omap5-board-common: fix ldo1_reg and ldo4_reg ranges ARM: dts: omap5-cm-t54: fix ldo1_reg and ldo4_reg ranges Tony Lindgren (4): ARM: dts: Add clocks for dm814x ADPLL Merge branch 'fixes-rc2' into omap-for-v4.6/fixes Merge tag 'for-v4.6-rc/omap-fixes-b' of git://git.kernel.org/.../pjw/omap-pending into omap-for-v4.6/fixes Revert "ARM: OMAP: Catch callers of revision information prior to it being populated" Tony Luck (1): EDAC: i7core, sb_edac: Don't return NOTIFY_BAD from mce_decoder callback Ville Syrjälä (2): drm/i915: Make RPS EI/thresholds multiple of 25 on SNB-BDW gpiolib-acpi: Duplicate con_id string when adding it to the crs lookup list Vineet Gupta (3): ARC: Add missing io barriers to io{read,write}{16,32}be() ARC: Fix PAE40 boot failures due to PTE truncation ARC: support HIGHMEM even without PAE40 Vinod Koul (6): ASoC: Intel: Atom: fix boot warning ASoC: rt298: fix null deref on acpi driver data ASoC: rt298: Add DMI match for Broxton-P reference platform ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link management ASoC: hdac_hdmi: add link management Viresh Kumar (1): PM / OPP: Remove useless check Vitaly Prosyak (1): drm/radeon: fix vertical bars appear on monitor (v2) Vladimir Murzin (1): ARM: 8573/1: domain: move {set,get}_domain under config guard Vladimir Zapolskiy (2): usb: musb: jz4740: fix error check of usb_get_phy() rapidio: fix potential NULL pointer dereference Vladis Dronov (1): [media] usbvision: revert commit 588afcc1 Vlastimil Babka (2): mm: wake kcompactd before kswapd's short sleep mm: fix kcompactd hang during memory offlining Wang Shanker (1): net: l2tp: fix reversed udp6 checksum flags Wang YanQing (1): x86/sysfb_efi: Fix valid BAR address range check Woojung Huh (2): lan78xx: fix statistics counter error lan78xx: workaround of forced 100 Full/Half duplex mode error Xishi Qiu (1): Documentation/sysctl/vm.txt: update numa_zonelist_order description Yakir Yang (1): i2c: rk3x: add support for rk3228 Yang Shi (1): mm: thp: correct split_huge_pages file permission Yury Norov (1): asm-generic: use compat version for preadv2 and pwritev2 cpaul(a)redhat.com (1): drm/dp/mst: Get validated port ref in drm_dp_update_payload_part1() xuejiufei (1): ocfs2/dlm: return zero if deref_done message is successfully handled .mailmap | 5 + .../devicetree/bindings/arc/archs-pct.txt | 2 +- Documentation/devicetree/bindings/arc/pct.txt | 2 +- Documentation/devicetree/bindings/arm/cpus.txt | 1 - .../devicetree/bindings/ata/ahci-platform.txt | 4 + Documentation/devicetree/bindings/i2c/i2c-rk3x.txt | 4 +- Documentation/devicetree/bindings/net/cpsw.txt | 6 +- .../devicetree/bindings/sound/davinci-mcbsp.txt | 51 +++ .../devicetree/bindings/sound/fsl-sai.txt | 9 +- .../devicetree/bindings/sound/pcm5102a.txt | 13 + Documentation/networking/altera_tse.txt | 6 +- Documentation/networking/ipvlan.txt | 6 +- Documentation/networking/pktgen.txt | 6 +- Documentation/networking/vrf.txt | 2 +- Documentation/networking/xfrm_sync.txt | 6 +- Documentation/sysctl/vm.txt | 19 +- MAINTAINERS | 77 ++-- Makefile | 4 +- arch/arc/Kconfig | 15 + arch/arc/include/asm/io.h | 27 +- arch/arc/include/asm/irqflags-arcv2.h | 36 +- arch/arc/include/asm/mmzone.h | 43 ++ arch/arc/include/asm/page.h | 15 +- arch/arc/include/asm/pgtable.h | 13 +- arch/arc/kernel/entry-arcv2.S | 10 +- arch/arc/kernel/entry-compact.S | 3 + arch/arc/mm/init.c | 58 ++- arch/arm/boot/dts/am33xx.dtsi | 2 +- arch/arm/boot/dts/am4372.dtsi | 2 +- arch/arm/boot/dts/am57xx-beagle-x15.dts | 17 - arch/arm/boot/dts/dm814x-clocks.dtsi | 243 ++++++++++-- arch/arm/boot/dts/dra62x-clocks.dtsi | 26 ++ arch/arm/boot/dts/dra7xx-clocks.dtsi | 18 +- arch/arm/boot/dts/omap3-n900.dts | 9 + arch/arm/boot/dts/omap34xx.dtsi | 2 +- arch/arm/boot/dts/omap5-board-common.dtsi | 4 +- arch/arm/boot/dts/omap5-cm-t54.dts | 4 +- arch/arm/boot/dts/omap5.dtsi | 2 +- arch/arm/boot/dts/qcom-apq8064.dtsi | 3 +- arch/arm/boot/dts/qcom-msm8974.dtsi | 14 +- arch/arm/boot/dts/r8a7791-koelsch.dts | 1 + arch/arm/boot/dts/r8a7791-porter.dts | 14 +- arch/arm/boot/dts/r8a7791.dtsi | 5 +- arch/arm/boot/dts/sun8i-q8-common.dtsi | 2 - arch/arm/include/asm/domain.h | 11 + arch/arm/kernel/head-nommu.S | 2 +- arch/arm/kvm/mmu.c | 2 +- arch/arm/mach-davinci/board-mityomapl138.c | 5 + arch/arm/mach-davinci/common.c | 5 + arch/arm/mach-exynos/pm_domains.c | 2 +- .../mach-imx/devices/platform-sdhci-esdhc-imx.c | 5 +- arch/arm/mach-omap2/clockdomains7xx_data.c | 2 +- arch/arm/mach-omap2/io.c | 3 +- arch/arm/mach-omap2/omap-wakeupgen.c | 7 + arch/arm/mach-omap2/pm34xx.c | 23 +- arch/arm/mach-shmobile/timer.c | 28 +- arch/arm/mach-socfpga/headsmp.S | 1 + arch/arm/mm/nommu.c | 15 +- arch/arm64/boot/dts/renesas/r8a7795.dtsi | 1 - .../boot/dts/socionext/uniphier-ph1-ld20-ref.dts | 1 - .../boot/dts/socionext/uniphier-ph1-ld20.dtsi | 20 +- arch/nios2/lib/memset.c | 2 +- arch/parisc/kernel/syscall.S | 2 +- arch/powerpc/include/asm/systbl.h | 2 + arch/powerpc/include/asm/unistd.h | 2 +- arch/powerpc/include/asm/word-at-a-time.h | 2 +- arch/powerpc/include/uapi/asm/unistd.h | 2 + arch/s390/include/asm/mmu.h | 2 +- arch/s390/include/asm/mmu_context.h | 28 +- arch/s390/include/asm/pgalloc.h | 4 +- arch/s390/include/asm/processor.h | 2 +- arch/s390/include/asm/tlbflush.h | 9 +- arch/s390/mm/init.c | 3 +- arch/s390/mm/mmap.c | 6 +- arch/s390/mm/pgalloc.c | 85 ++-- arch/s390/pci/pci_dma.c | 16 +- arch/sparc/configs/sparc32_defconfig | 1 - arch/sparc/configs/sparc64_defconfig | 1 - arch/sparc/include/asm/spitfire.h | 1 + arch/sparc/include/uapi/asm/unistd.h | 4 +- arch/sparc/kernel/cherrs.S | 14 +- arch/sparc/kernel/cpu.c | 6 + arch/sparc/kernel/cpumap.c | 1 + arch/sparc/kernel/fpu_traps.S | 11 +- arch/sparc/kernel/head_64.S | 32 +- arch/sparc/kernel/misctrap.S | 12 +- arch/sparc/kernel/pci.c | 42 +- arch/sparc/kernel/setup_64.c | 7 +- arch/sparc/kernel/spiterrs.S | 18 +- arch/sparc/kernel/systbls_32.S | 2 +- arch/sparc/kernel/systbls_64.S | 4 +- arch/sparc/kernel/utrap.S | 3 +- arch/sparc/kernel/vio.c | 18 + arch/sparc/kernel/vmlinux.lds.S | 4 + arch/sparc/kernel/winfixup.S | 3 +- arch/sparc/mm/init_64.c | 3 + arch/x86/events/amd/core.c | 2 +- arch/x86/events/amd/iommu.c | 1 + arch/x86/events/intel/core.c | 3 + arch/x86/events/intel/lbr.c | 6 +- arch/x86/events/intel/pt.c | 75 +++- arch/x86/events/intel/pt.h | 3 + arch/x86/events/intel/rapl.c | 1 + arch/x86/include/asm/perf_event.h | 4 + arch/x86/kernel/apic/vector.c | 3 +- arch/x86/kernel/apic/x2apic_uv_x.c | 4 +- arch/x86/kernel/head_32.S | 6 - arch/x86/kernel/sysfb_efi.c | 14 +- arch/x86/kernel/tsc_msr.c | 2 +- arch/x86/kvm/mmu.c | 4 +- arch/x86/kvm/vmx.c | 4 + arch/x86/mm/setup_nx.c | 5 +- arch/x86/platform/efi/efi-bgrt.c | 18 +- arch/x86/xen/spinlock.c | 6 + drivers/acpi/acpica/dsmethod.c | 3 + drivers/acpi/nfit.c | 5 +- drivers/ata/Kconfig | 8 + drivers/ata/Makefile | 1 + drivers/ata/ahci_platform.c | 3 + drivers/ata/ahci_seattle.c | 210 ++++++++++ drivers/ata/libahci.c | 1 + drivers/base/power/opp/core.c | 3 - drivers/base/property.c | 2 +- drivers/block/rbd.c | 52 ++- drivers/clk/imx/clk-imx6q.c | 2 +- drivers/cpufreq/cpufreq.c | 26 +- drivers/cpufreq/cpufreq_governor.c | 8 +- drivers/cpufreq/intel_pstate.c | 31 +- drivers/cpufreq/sti-cpufreq.c | 4 + drivers/cpuidle/cpuidle-arm.c | 2 +- drivers/crypto/talitos.c | 87 +++-- drivers/edac/i7core_edac.c | 2 +- drivers/edac/sb_edac.c | 2 +- drivers/firmware/efi/vars.c | 37 +- drivers/firmware/psci.c | 2 +- drivers/firmware/qemu_fw_cfg.c | 2 +- drivers/gpio/gpio-rcar.c | 65 +--- drivers/gpio/gpiolib-acpi.c | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu_atpx_handler.c | 11 +- drivers/gpu/drm/amd/amdgpu/amdgpu_device.c | 8 +- drivers/gpu/drm/amd/amdgpu/amdgpu_object.c | 1 + drivers/gpu/drm/amd/amdgpu/atombios_encoders.c | 4 + drivers/gpu/drm/amd/amdgpu/gmc_v7_0.c | 5 +- drivers/gpu/drm/amd/amdgpu/gmc_v8_0.c | 5 +- drivers/gpu/drm/drm_dp_mst_topology.c | 20 + drivers/gpu/drm/etnaviv/etnaviv_gpu.c | 31 +- drivers/gpu/drm/i915/i915_drv.c | 32 +- drivers/gpu/drm/i915/i915_reg.h | 9 +- drivers/gpu/drm/i915/intel_ddi.c | 22 +- drivers/gpu/drm/i915/intel_display.c | 3 + drivers/gpu/drm/i915/intel_dp.c | 4 +- drivers/gpu/drm/i915/intel_drv.h | 2 + drivers/gpu/drm/i915/intel_hdmi.c | 12 +- drivers/gpu/drm/radeon/atombios_encoders.c | 4 + drivers/gpu/drm/radeon/evergreen.c | 154 +++++++- drivers/gpu/drm/radeon/evergreen_reg.h | 46 +++ drivers/gpu/drm/ttm/ttm_bo.c | 17 +- drivers/gpu/drm/virtio/virtgpu_display.c | 12 + drivers/gpu/drm/vmwgfx/vmwgfx_execbuf.c | 10 +- drivers/gpu/drm/vmwgfx/vmwgfx_fb.c | 6 +- drivers/gpu/ipu-v3/ipu-common.c | 7 +- drivers/hid/hid-ids.h | 1 + drivers/hid/usbhid/hid-quirks.c | 1 + drivers/hid/wacom_wac.c | 6 + drivers/hv/ring_buffer.c | 26 +- drivers/i2c/busses/Kconfig | 4 +- drivers/i2c/busses/i2c-cpm.c | 4 +- drivers/i2c/busses/i2c-exynos5.c | 24 +- drivers/i2c/busses/i2c-ismt.c | 2 + drivers/i2c/busses/i2c-rk3x.c | 1 + drivers/iio/adc/at91-sama5d2_adc.c | 2 + drivers/iio/imu/inv_mpu6050/inv_mpu_i2c.c | 30 +- drivers/iio/imu/inv_mpu6050/inv_mpu_spi.c | 3 +- drivers/iio/magnetometer/ak8975.c | 6 +- drivers/infiniband/core/cache.c | 3 +- drivers/infiniband/core/ucm.c | 4 + drivers/infiniband/core/ucma.c | 3 + drivers/infiniband/core/uverbs_main.c | 5 + drivers/infiniband/core/verbs.c | 3 +- drivers/infiniband/hw/cxgb3/iwch_provider.c | 2 + drivers/infiniband/hw/cxgb4/cq.c | 2 +- drivers/infiniband/hw/cxgb4/provider.c | 2 + drivers/infiniband/hw/cxgb4/qp.c | 24 +- drivers/infiniband/hw/mlx5/main.c | 6 +- drivers/infiniband/hw/nes/nes_nic.c | 3 - drivers/infiniband/hw/qib/qib_file_ops.c | 5 + drivers/infiniband/sw/rdmavt/qp.c | 4 +- drivers/infiniband/ulp/iser/iscsi_iser.c | 14 +- drivers/input/misc/twl6040-vibra.c | 16 +- drivers/input/touchscreen/atmel_mxt_ts.c | 28 +- drivers/input/touchscreen/zforce_ts.c | 4 +- drivers/md/md.c | 2 + drivers/md/raid0.c | 2 +- drivers/md/raid5.c | 2 - drivers/media/media-device.c | 8 +- drivers/media/platform/exynos4-is/media-dev.c | 13 +- drivers/media/platform/s3c-camif/camif-core.c | 12 +- drivers/media/usb/usbvision/usbvision-video.c | 7 - drivers/media/v4l2-core/videobuf2-core.c | 20 +- drivers/media/v4l2-core/videobuf2-memops.c | 2 +- drivers/media/v4l2-core/videobuf2-v4l2.c | 20 +- drivers/misc/cxl/context.c | 7 + drivers/misc/cxl/cxl.h | 2 + drivers/misc/cxl/irq.c | 1 - drivers/misc/cxl/native.c | 31 ++ drivers/misc/mic/vop/vop_vringh.c | 5 + drivers/mmc/host/Kconfig | 1 + drivers/mmc/host/sdhci-acpi.c | 81 ++++ drivers/mmc/host/sunxi-mmc.c | 5 + drivers/net/Kconfig | 5 +- drivers/net/dsa/mv88e6xxx.c | 2 +- drivers/net/ethernet/broadcom/bnxt/bnxt.c | 53 ++- drivers/net/ethernet/broadcom/bnxt/bnxt.h | 13 + drivers/net/ethernet/cadence/macb.c | 34 +- drivers/net/ethernet/chelsio/cxgb3/cxgb3_main.c | 3 +- drivers/net/ethernet/marvell/mvneta.c | 6 +- drivers/net/ethernet/marvell/pxa168_eth.c | 2 + drivers/net/ethernet/mellanox/mlx4/en_tx.c | 6 +- drivers/net/ethernet/mellanox/mlx5/core/Kconfig | 1 + drivers/net/ethernet/mellanox/mlx5/core/en.h | 3 +- .../net/ethernet/mellanox/mlx5/core/en_ethtool.c | 8 +- drivers/net/ethernet/mellanox/mlx5/core/en_main.c | 106 +++-- drivers/net/ethernet/mellanox/mlx5/core/fs_core.c | 48 +-- drivers/net/ethernet/mellanox/mlx5/core/main.c | 25 +- drivers/net/ethernet/mellanox/mlx5/core/port.c | 10 +- drivers/net/ethernet/mellanox/mlx5/core/uar.c | 6 +- drivers/net/ethernet/mellanox/mlx5/core/vport.c | 40 ++ drivers/net/ethernet/mellanox/mlx5/core/vxlan.c | 50 ++- drivers/net/ethernet/mellanox/mlx5/core/vxlan.h | 11 +- drivers/net/ethernet/myricom/myri10ge/myri10ge.c | 4 +- drivers/net/ethernet/qlogic/qlcnic/qlcnic.h | 4 +- drivers/net/ethernet/sfc/ef10.c | 15 +- .../net/ethernet/stmicro/stmmac/dwmac-socfpga.c | 50 +-- drivers/net/ethernet/ti/cpsw.c | 67 ++-- drivers/net/ethernet/ti/cpsw.h | 1 + drivers/net/ethernet/ti/davinci_emac.c | 5 +- drivers/net/ethernet/toshiba/ps3_gelic_wireless.c | 2 +- drivers/net/macsec.c | 65 ++-- drivers/net/phy/at803x.c | 40 +- drivers/net/usb/lan78xx.c | 44 ++- drivers/net/usb/pegasus.c | 10 +- drivers/net/usb/smsc75xx.c | 12 +- drivers/net/usb/smsc95xx.c | 12 +- drivers/net/wireless/ath/ath9k/ar5008_phy.c | 8 +- drivers/net/wireless/ath/ath9k/ar9002_phy.c | 5 - drivers/net/wireless/intel/iwlwifi/iwl-8000.c | 2 +- drivers/net/wireless/intel/iwlwifi/iwl-drv.c | 26 +- drivers/net/wireless/intel/iwlwifi/mvm/fw-dbg.c | 6 +- drivers/net/wireless/intel/iwlwifi/mvm/fw.c | 2 + drivers/net/wireless/intel/iwlwifi/pcie/drv.c | 10 + drivers/nvdimm/pmem.c | 13 +- drivers/nvmem/mxs-ocotp.c | 4 +- drivers/platform/x86/toshiba_acpi.c | 2 +- drivers/rapidio/devices/rio_mport_cdev.c | 119 +++--- drivers/s390/char/sclp_ctl.c | 12 +- drivers/scsi/cxgbi/libcxgbi.c | 1 + drivers/soc/mediatek/mtk-scpsys.c | 11 +- drivers/staging/media/davinci_vpfe/vpfe_video.c | 54 ++- drivers/staging/rdma/hfi1/TODO | 2 +- drivers/staging/rdma/hfi1/file_ops.c | 91 ++--- drivers/staging/rdma/hfi1/mmu_rb.c | 40 +- drivers/staging/rdma/hfi1/mmu_rb.h | 3 +- drivers/staging/rdma/hfi1/qp.c | 2 + drivers/staging/rdma/hfi1/user_exp_rcv.c | 11 +- drivers/staging/rdma/hfi1/user_sdma.c | 33 +- drivers/thermal/hisi_thermal.c | 4 +- drivers/thermal/thermal_core.c | 2 +- drivers/tty/pty.c | 18 +- drivers/tty/tty_io.c | 6 +- drivers/usb/core/port.c | 6 - drivers/usb/core/usb.c | 8 +- drivers/usb/musb/jz4740.c | 4 +- drivers/usb/musb/musb_gadget.c | 6 +- drivers/usb/musb/musb_host.c | 2 +- drivers/usb/serial/cp210x.c | 4 + drivers/virtio/virtio_ring.c | 2 +- drivers/xen/balloon.c | 16 + drivers/xen/evtchn.c | 20 +- fs/ceph/mds_client.c | 6 +- fs/devpts/inode.c | 53 +-- fs/fuse/file.c | 2 +- fs/ocfs2/dlm/dlmmaster.c | 2 + fs/pnode.c | 25 +- fs/proc/base.c | 3 +- fs/proc/task_mmu.c | 33 +- fs/udf/super.c | 4 +- fs/udf/udfdecl.h | 2 +- fs/udf/unicode.c | 16 +- include/acpi/acpi_bus.h | 4 +- include/linux/bpf.h | 3 +- include/linux/ceph/auth.h | 10 +- include/linux/ceph/osd_client.h | 1 - include/linux/cgroup-defs.h | 1 + include/linux/cpuset.h | 6 - include/linux/devpts_fs.h | 6 +- include/linux/hash.h | 20 +- include/linux/huge_mm.h | 5 + include/linux/if_ether.h | 5 + include/linux/lockdep.h | 8 +- include/linux/mfd/syscon/imx6q-iomuxc-gpr.h | 6 + include/linux/mlx5/device.h | 11 + include/linux/mlx5/driver.h | 7 +- include/linux/mlx5/port.h | 6 +- include/linux/mlx5/vport.h | 2 + include/linux/mm.h | 4 + include/linux/net.h | 10 +- include/linux/netdevice.h | 2 +- include/linux/of.h | 2 +- include/linux/page-flags.h | 22 ++ include/linux/swap.h | 4 + include/linux/tty_driver.h | 4 +- include/media/videobuf2-core.h | 8 + include/net/switchdev.h | 4 + include/net/vxlan.h | 4 +- include/rdma/ib.h | 16 + include/sound/dmaengine_pcm.h | 12 + include/sound/hdaudio_ext.h | 13 + include/sound/hdmi-codec.h | 100 +++++ include/sound/pcm_iec958.h | 2 + include/sound/soc-dapm.h | 3 +- include/sound/soc.h | 5 +- include/uapi/asm-generic/unistd.h | 4 +- include/uapi/linux/if_macsec.h | 4 +- include/{ => uapi}/linux/rio_mport_cdev.h | 144 +++---- include/uapi/linux/swab.h | 24 +- include/uapi/linux/v4l2-dv-timings.h | 30 +- include/xen/page.h | 4 +- kernel/bpf/inode.c | 7 +- kernel/bpf/syscall.c | 24 +- kernel/bpf/verifier.c | 77 ++-- kernel/cgroup.c | 7 +- kernel/cpuset.c | 4 +- kernel/events/core.c | 55 ++- kernel/kcov.c | 3 +- kernel/kexec_core.c | 7 +- kernel/locking/lockdep.c | 37 +- kernel/locking/lockdep_proc.c | 2 + kernel/sched/core.c | 29 +- kernel/trace/trace_events.c | 9 +- kernel/workqueue.c | 29 ++ lib/stackdepot.c | 10 +- mm/compaction.c | 14 +- mm/huge_memory.c | 16 +- mm/memcontrol.c | 37 +- mm/memory-failure.c | 10 +- mm/memory.c | 51 ++- mm/migrate.c | 8 +- mm/page-writeback.c | 6 +- mm/page_alloc.c | 2 +- mm/page_io.c | 6 +- mm/swap.c | 5 + mm/vmscan.c | 30 +- mm/zswap.c | 8 +- net/batman-adv/bat_v.c | 12 + net/batman-adv/distributed-arp-table.c | 17 +- net/batman-adv/hard-interface.c | 6 +- net/batman-adv/originator.c | 17 +- net/batman-adv/routing.c | 9 + net/batman-adv/send.c | 6 + net/batman-adv/soft-interface.c | 8 +- net/batman-adv/translation-table.c | 42 +- net/batman-adv/types.h | 7 + net/bridge/br_mdb.c | 124 +++--- net/bridge/br_multicast.c | 8 +- net/bridge/br_private.h | 4 +- net/ceph/auth.c | 8 +- net/ceph/auth_none.c | 71 ++-- net/ceph/auth_none.h | 3 +- net/ceph/auth_x.c | 21 +- net/ceph/auth_x.h | 1 + net/ceph/osd_client.c | 6 +- net/core/dev.c | 2 +- net/ipv4/fib_frontend.c | 6 +- net/ipv4/inet_hashtables.c | 2 + net/ipv4/ip_gre.c | 30 +- net/ipv4/ip_tunnel.c | 4 +- net/ipv6/addrconf.c | 48 +-- net/ipv6/ila/ila_lwt.c | 3 +- net/l2tp/l2tp_core.c | 4 +- net/mac80211/iface.c | 4 +- net/rds/tcp.c | 3 +- net/rds/tcp.h | 4 + net/rds/tcp_connect.c | 8 + net/rds/tcp_listen.c | 54 ++- net/sched/sch_netem.c | 61 ++- net/switchdev/switchdev.c | 6 + net/tipc/node.c | 5 + samples/bpf/trace_output_kern.c | 1 - scripts/mod/file2alias.c | 69 ++-- security/integrity/ima/ima_policy.c | 4 +- sound/core/pcm_dmaengine.c | 11 +- sound/core/pcm_iec958.c | 65 +++- sound/hda/ext/hdac_ext_bus.c | 3 + sound/hda/ext/hdac_ext_controller.c | 66 ++++ sound/hda/local.h | 10 + sound/soc/atmel/atmel_ssc_dai.c | 4 +- sound/soc/au1x/dbdma2.c | 4 +- sound/soc/bcm/bcm2835-i2s.c | 27 +- sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/ak4642.c | 4 +- sound/soc/codecs/arizona.c | 32 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs47l24.c | 46 ++- sound/soc/codecs/da7213.c | 99 ++++- sound/soc/codecs/da7213.h | 35 +- sound/soc/codecs/da7218.c | 32 +- sound/soc/codecs/da7218.h | 21 +- sound/soc/codecs/da7219.c | 38 +- sound/soc/codecs/da7219.h | 20 +- sound/soc/codecs/es8328.c | 194 ++++++--- sound/soc/codecs/es8328.h | 23 +- sound/soc/codecs/hdac_hdmi.c | 195 +++++++++- sound/soc/codecs/hdmi-codec.c | 432 +++++++++++++++++++++ sound/soc/codecs/pcm5102a.c | 69 ++++ sound/soc/codecs/rt298.c | 19 +- sound/soc/codecs/rt5645.c | 16 +- sound/soc/codecs/wm5102.c | 4 + sound/soc/codecs/wm5110.c | 6 + sound/soc/codecs/wm_adsp.c | 292 ++++++++++---- sound/soc/codecs/wm_adsp.h | 1 + sound/soc/davinci/Kconfig | 6 +- sound/soc/davinci/davinci-i2s.c | 80 ++-- sound/soc/davinci/davinci-mcasp.c | 92 +++-- sound/soc/davinci/davinci-mcasp.h | 5 + sound/soc/dwc/designware_i2s.c | 14 +- sound/soc/fsl/fsl_sai.c | 24 +- sound/soc/fsl/fsl_ssi.c | 81 +++- sound/soc/fsl/imx-pcm-fiq.c | 2 +- sound/soc/intel/Kconfig | 16 + sound/soc/intel/atom/sst-atom-controls.c | 2 +- sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 353 +++++++++++++++++ sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 87 ++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 86 +++- sound/soc/intel/boards/skl_rt286.c | 59 ++- sound/soc/intel/common/sst-acpi.h | 9 + sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/bxt-sst.c | 328 ++++++++++++++++ sound/soc/intel/skylake/skl-messages.c | 122 +++++- sound/soc/intel/skylake/skl-nhlt.c | 15 +- sound/soc/intel/skylake/skl-pcm.c | 119 +++++- sound/soc/intel/skylake/skl-sst-dsp.c | 2 - sound/soc/intel/skylake/skl-sst-dsp.h | 15 +- sound/soc/intel/skylake/skl-sst.c | 13 +- sound/soc/intel/skylake/skl-topology.c | 27 +- sound/soc/intel/skylake/skl-topology.h | 2 +- sound/soc/intel/skylake/skl-tplg-interface.h | 2 +- sound/soc/intel/skylake/skl.c | 39 +- sound/soc/intel/skylake/skl.h | 6 +- sound/soc/rockchip/rockchip_i2s.c | 87 +++-- sound/soc/soc-core.c | 14 +- sound/soc/soc-generic-dmaengine-pcm.c | 57 +-- 461 files changed, 7248 insertions(+), 2394 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm5102a.txt create mode 100644 arch/arc/include/asm/mmzone.h create mode 100644 drivers/ata/ahci_seattle.c create mode 100644 include/sound/hdmi-codec.h rename include/{ => uapi}/linux/rio_mport_cdev.h (74%) create mode 100644 sound/soc/codecs/hdmi-codec.c create mode 100644 sound/soc/codecs/pcm5102a.c create mode 100644 sound/soc/intel/boards/bxt_rt298.c create mode 100644 sound/soc/intel/skylake/bxt-sst.c

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[alsa-devel] [PATCH] ASoC: hdac_hdmi: Remove the unused 'timeout' variable
by Fabio Estevam 16 May '16

16 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> Commit b2047e996cd88d3 ("ASoC: hdac_hdmi: add link management") introuduced the following build warning: sound/soc/codecs/hdac_hdmi.c:1721:16: warning: unused variable 'timeout' [-Wunused-variable] Remove the unused 'timeout' variable. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/hdac_hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3733297..181cd3b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1718,7 +1718,6 @@ static int hdac_hdmi_runtime_suspend(struct device *dev) struct hdac_ext_device *edev = to_hda_ext_device(dev); struct hdac_device *hdac = &edev->hdac; struct hdac_bus *bus = hdac->bus; - unsigned long timeout; struct hdac_ext_bus *ebus = hbus_to_ebus(bus); struct hdac_ext_link *hlink = NULL; int err; -- 1.9.1

1 0

[alsa-devel] [PATCH v4] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 16 May '16

16 May '16

From: Shreyas NC <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can be changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Changes in v4: rebase on asoc-next, resolve rebase conflict Changes in v3: add bound check for firmware file sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 118 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 ++ sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 5 ++ 6 files changed, 245 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..a3cd5d7ab246 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,118 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * 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 <linux/module.h> +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + uuid_le *uuid_mod; + + tbl = ctx->tbl; + num = ctx->num_modules; + uuid_mod = (uuid_le *)uuid; + + for (i = 0; i < num; i++) { + if (uuid_le_cmp(*uuid_mod, tbl[i].uuid) == 0) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int snd_skl_parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + uuid_le *uuid_bin; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Read the UUID(GUID) from FW Manifest. + * The 16 byte UUID is of the format: + * XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX + * Populate the UUID table to store module_id + * and loadable flags for the module. + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + + /* + * we check if current pointer is larger than file size from + * base value to check excceding the file while parsing + */ + if ((const char *)mod_entry >= buf + ctx->fw->size) { + dev_err(ctx->dev, + "Exceeds file bound: Entry %d Ptr %p\n", + i, mod_entry); + + return -EIO; + } + + uuid_bin = (uuid_le *)mod_entry->uuid.id; + memcpy(&tbl[i].uuid, uuid_bin, sizeof(tbl[i].uuid)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_skl_parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..cfdb24c45035 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,107 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * 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 <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u8 id[16]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + uuid_le uuid; + int module_id; + int is_loadable; +}; + +int snd_skl_parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index 13ec8d53b526..db3a45ee102b 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -25,6 +25,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -70,7 +71,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -84,6 +85,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = snd_skl_parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a1f4478fabcb..f690b016203c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1566,6 +1567,10 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, w->priv = mconfig; memcpy(&mconfig->guid, &dfw_config->uuid, 16); + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; -- 1.9.1

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[alsa-devel] [PATCH] ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity
by Takashi Sakamoto 16 May '16

16 May '16

These three commits were merged to improve PCM pointer granularity. commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call") commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position") commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing") The point of them is to handle queued packets not only in software IRQ context of IR/IT contexts, but also in process context. As a result of handling packets, period tasklet is scheduled when acrossing PCM period boundary. This is to prevent recursive call of 'struct snd_pcm_ops.pointer()' in the same context. When the pointer callback is executed in the process context, it's better to avoid the second callback in the software IRQ context. The software IRQ context runs immediately after scheduled in the process context because few packets are queued yet. For the aim, 'pointer_flush' is used, however it causes a race condition between the process context and software IRQ context of IR/IT contexts. Practically, this race is not so critical because it influences process context to skip flushing queued packet and to get worse granularity of PCM pointer. The race condition is quite rare but it should be improved for stable service. The similar effect can be achieved by using 'in_interrupt()' macro. This commit obsoletes 'pointer_flush' with it. Acked-by: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 23 +++++++++++++++++------ sound/firewire/amdtp-stream.h | 1 - 2 files changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 830a95c..024ab7f 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -251,7 +251,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -356,7 +355,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -803,11 +801,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets will be already + * queued in the buffer. These packets should be handled immediately + * to keep better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 349c405..c1bc7fa 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -126,7 +126,6 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; -- 2.7.4

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Re: [alsa-devel] Why is the bufs argument for snd_pcm_writen not const?
by Trent Reed 16 May '16

16 May '16

Absolutely! Sorry about that, I'm used to requesting changes and not being able to make them myself. I'll ping the thread with the patch later this week. :) Thanks, - Trent Reed On May 16, 2016 12:12 AM, "Takashi Iwai" <tiwai(a)suse.de> wrote: On Mon, 16 May 2016 09:06:06 +0200, Trent Reed wrote: > > Thanks Takashi, > > I wasn't able to find anything about this in the documentation. Perhaps if > it isn't changed in the API, it could be added as a @note for Doxygen in > the functions? A patch is always welcome :) Takashi > > - Trent Reed > On May 15, 2016 11:58 PM, "Takashi Iwai" <tiwai(a)suse.de> wrote: > > > On Sun, 15 May 2016 05:05:49 +0200, > > Trent Reed wrote: > > > > > > Hey All, > > > > > > Just wondering what the reasoning is behind snd_pcm_writen not having > > > a constant variable for bufs. To me, this doesn't make sense. Does > > > snd_pcm_writen make any alterations to bufs? I would expect one of the > > > following declaration: > > > > > > // Better, but doesn't promise that the API won't change the channel > > pointers. > > > snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t* pcm, const void** bufs, > > > snd_pcm_uframes_t size); > > > // Best, promises ALSA won't change channel pointers, wont change > > > underlying data. > > > snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t* pcm, const void*const* > > > bufs, snd_pcm_uframes_t size); > > > > > > For now, is it safe to assume that snd_pcm_writen() doesn't alter any > > > of the data in bufs passed into it? Any chance on seeing this patched > > > or is there some reason for this? > > > > We never thought of modifying the data inside the writen API > > function. The lack of const is merely the laziness. > > > > > > Takashi > > > [2 <text/html; UTF-8 (quoted-printable)>] >

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[alsa-devel] Why is the bufs argument for snd_pcm_writen not const?
by Trent Reed 16 May '16

16 May '16

Hey All, Just wondering what the reasoning is behind snd_pcm_writen not having a constant variable for bufs. To me, this doesn't make sense. Does snd_pcm_writen make any alterations to bufs? I would expect one of the following declaration: // Better, but doesn't promise that the API won't change the channel pointers. snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t* pcm, const void** bufs, snd_pcm_uframes_t size); // Best, promises ALSA won't change channel pointers, wont change underlying data. snd_pcm_sframes_t snd_pcm_writen(snd_pcm_t* pcm, const void*const* bufs, snd_pcm_uframes_t size); For now, is it safe to assume that snd_pcm_writen() doesn't alter any of the data in bufs passed into it? Any chance on seeing this patched or is there some reason for this? Thanks, - Trent Reed

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[alsa-devel] [PATCH] conf: don't check if config files are up to date
by Alexander E. Patrakov 16 May '16

16 May '16

Even if no files have been changed, the config may become outdated due to hot-plugged devices. Note: this patch has a huge potential to crash badly-written applications that keep config pointers and strings for too long. But I see no other way to support hot-pluggable devices. Fixes: https://bugs.freedesktop.org/show_bug.cgi?id=54029 Signed-off-by: Alexander E. Patrakov <patrakov(a)gmail.com> --- src/conf.c | 63 ++++++++++++++++++++++++---------------------------------- src/confmisc.c | 4 +++- 2 files changed, 29 insertions(+), 38 deletions(-) diff --git a/src/conf.c b/src/conf.c index f8b7a66..34576aa 100644 --- a/src/conf.c +++ b/src/conf.c @@ -3661,14 +3661,15 @@ SND_DLSYM_BUILD_VERSION(snd_config_hook_load_for_all_cards, SND_CONFIG_DLSYM_VER #endif /** - * \brief Updates a configuration tree by rereading the configuration files (if needed). + * \brief Updates a configuration tree by rereading the configuration files. * \param[in,out] _top Address of the handle to the top-level node. * \param[in,out] _update Address of a pointer to private update information. * \param[in] cfgs A list of configuration file names, delimited with ':'. * If \p cfgs is \c NULL, the default global * configuration file is used. - * \return 0 if \a _top was up to date, 1 if the configuration files - * have been reread, otherwise a negative error code. + * \return 0 if \a _top was up to date (this can happen only in previous + * ALSA-lib versions), 1 if the configuration files have been + * reread successfully, otherwise a negative error code. * * The variables pointed to by \a _top and \a _update can be initialized * to \c NULL before the first call to this function. The private @@ -3679,7 +3680,8 @@ SND_DLSYM_BUILD_VERSION(snd_config_hook_load_for_all_cards, SND_CONFIG_DLSYM_VER * The global configuration files are specified in the environment variable * \c ALSA_CONFIG_PATH. * - * \warning If the configuration tree is reread, all string pointers and + * \warning In order to deal with hot-pluggable devices, the configuration + * tree is always reread, even if nothing changed. All string pointers and * configuration node handles previously obtained from this tree become * invalid. * @@ -3754,35 +3756,6 @@ int snd_config_update_r(snd_config_t **_top, snd_config_update_t **_update, cons local->count--; } } - if (!update) - goto _reread; - if (local->count != update->count) - goto _reread; - for (k = 0; k < local->count; ++k) { - struct finfo *lf = &local->finfo[k]; - struct finfo *uf = &update->finfo[k]; - if (strcmp(lf->name, uf->name) != 0 || - lf->dev != uf->dev || - lf->ino != uf->ino || - lf->mtime != uf->mtime) - goto _reread; - } - err = 0; - - _end: - if (err < 0) { - if (top) { - snd_config_delete(top); - *_top = NULL; - } - if (update) { - snd_config_update_free(update); - *_update = NULL; - } - } - if (local) - snd_config_update_free(local); - return err; _reread: *_top = NULL; @@ -3823,15 +3796,31 @@ int snd_config_update_r(snd_config_t **_top, snd_config_update_t **_update, cons *_top = top; *_update = local; return 1; + +_end: + if (err < 0) { + if (top) { + snd_config_delete(top); + *_top = NULL; + } + if (update) { + snd_config_update_free(update); + *_update = NULL; + } + } + if (local) + snd_config_update_free(local); + return err; } /** - * \brief Updates #snd_config by rereading the global configuration files (if needed). - * \return 0 if #snd_config was up to date, 1 if #snd_config was - * updated, otherwise a negative error code. + * \brief Updates #snd_config by rereading the global configuration files. + * \return 0 if #snd_config was up to date (this can happen only in previous + * ALSA-lib versions), 1 if #snd_config was updated successfully, + * otherwise a negative error code. * * \warning Whenever #snd_config is updated, all string pointers and - * configuration node handles previously obtained from it may become + * configuration node handles previously obtained from it become * invalid. * * \par Errors: diff --git a/src/confmisc.c b/src/confmisc.c index ae0275f..8f97b72 100644 --- a/src/confmisc.c +++ b/src/confmisc.c @@ -599,7 +599,9 @@ static int open_ctl(long card, snd_ctl_t **ctl) char name[16]; snprintf(name, sizeof(name), "hw:%li", card); name[sizeof(name)-1] = '\0'; - return snd_ctl_open(ctl, name, 0); + + /* Take care not to update the config - we may be running hooks on pcms right now */ + return snd_ctl_open_lconf(ctl, name, 0, snd_config); } #if 0 -- 2.8.0

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[alsa-devel] emagic mt4 problems
by Kazuhiro Ito 15 May '16

15 May '16

When I use emagic mt4 (USB-MIDI interface) on Linux, there are some minor problems. 1. mt4 never switches to computer mode until PC receives mt4's response to initialization command. Mt4 has two mode, patch mode and computer mode. When connected to PC, it is in patch mode. To use as MIDI interface, mt4 must switch to computer mode. Linux drvier sends commnad "F0 00 20 31 64 0B 00 00 F7" to make mt4 switch to computer mode. But mt4 is still in patch mode after having received the command. It switches to computer mode when PC receives the response from mt4. I must run the below command to make mt4 really ready for use as MIDI interface. $ amidi -l Dir Device Name IO hw:2,0,0 MT4 MIDI 1 IO hw:2,0,1 MT4 MIDI 2 IO hw:2,0,2 MT4 MIDI 3 O hw:2,0,3 MT4 MIDI 4 O hw:2,0,4 MT4 Broadcast $ amidi -p hw:2,0,0 -r /dev/null -t 1 According to Kurt Arnlund's web page (*1), emagic devices accept "Set Computer Mode" command ("F0 00 20 31 64 0F 00 7F F7"), and actually mt4 switches to computer mode with this command immidiately and receiving mt4's response is not needed. (*1) http://www.potm.org/software/Unitor/mididata/ 2. Outputs to MIDI 1 are broadcasted. After switching to computer mode by the above procedure, output to MIDI 1 is broadcasted as if outputted to broadcast port. The current output port in driver's internal state is initialized to port 0 (MIDI 1). But actual device's current port is initialized to port 4 (broadcast port). So output to MIDI 1 doesn't make driver send port switch command until output to other port occurs. I think driver's current port state should be initialized to 4 (for mt4) or larger number. I worte the patch for the problems (to linux kernel 3.16), but I don't know it works with other emagic devices such as amt4, unitor8. If there is anyone has emagic device, please test the below command on unpatched kernel to check whether the device accepts "Set Computer Mode" command. $ amidi -p hw:2,0,0 -r /dev/null -t 1 $ amidi -p hw:2,0,0 -S 'F0 00 20 31 64 10 00 7F 00 F7' $ amidi -p hw:2,0,0 -S 'F0 00 20 31 64 0F 00 7F F7' My mt4 swithes to computer mode with the first amidi command, and returns to patch mode with second command, and finally re-enters in computer mode with the last command. -- Kazuhiro Ito

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Re: [alsa-devel] WM5102 - Help to make baytrail machine driver work
by Charles Keepax 15 May '16

15 May '16

On Wed, May 04, 2016 at 10:21:21PM -0300, Paulo Sergio wrote: > Hi list, > > I've been trying to make Lenovo Yoga 2 (Model 1051F) sound work on > Android-x86/ Linux. [1] > > Based on already discussed info from Christian Hartmann with members (see > [2] and [3] ) of this list and some efforts (basically patch pxa2xx with > Mika Westerberg changes, set the correct gpio int values {246 for reset / > 405 for ldoena and 342 for irq_gpio} and set LDO1 vcc suppliers) we could > have wm5102 initialized through spi / regmap (arizona code on drivers/mfd) > and register the DAI's. > > I also ported a machine driver code from a kernel Lenovo made available for > a similar hardware (Lenovo yoga 2 1050 - uses the same baytrail Z3745 and > wolfson wm5102 audio) [4] > > Although machine driver is registered, we still don't have sound. Tried > some mixer settings / routes used on raspberry pi, but none help. > Application (in this case alsa / Android players) returns a "Device or > resource busy" when trying to ioctl pcm device: A quick look through your machine driver looks like the parts relating the CODEC look ok. I am not so sure about the AP side routing but for the CODEC, 'HPOUT1L Input 1' AIF1RX1 'HPOUT1R Input 1' AIF1RX2 'HPOUT1 Digital Switch' 1 Should be enough to route some audio out the headphones, assuming AIF1 is connected to the AP and HPOUT1L is the headphones (it is very rare for people to not wire things up that way so its quite a good bet, and your machine driver suggests this is how it is connected). If you provide some kernel logs with the LOG_DEVICE in drivers/base/regmap/regmap.c set to the CODEC spi device, I am happy to sanity check what is going to the CODEC. Although at the moment I think your issue looks mostly like getting through the AP side. Thanks, Charles

2 7

[alsa-devel] i.MX6 SGTL5000 hangs with EIO error
by Ross Wille 14 May '16

14 May '16

Hello all, I am doing an ALSA capture on a Wandboard Quad (i.MX6 quad core with SGTL5000 codec chip) from the LINEIN1 jack. I'm using a 4.6.0-rc3-armv7-x0 kernel. Sometimes when the audio glitches (for example, when I plug/unplug the audio cable or adjust my signal generator) the snd_pcm_readi() function will start returning -5 (EIO). Once this happens, the only way I've been able to recover is to reboot the computer. Calling snd_pcm_close(), snd_pcm_prepare(), snd_pcm_start(), etc. doesn't help. When in this state, running arecord returns IO errors as well. It's interesting that, on rare occasions, I must do a power cycle in order to recover. When a reboot is not effective I've noticed that the capture device doesn't appear in /proc/asound/devices. I don't believe my specific ALSA settings are important, but I'm calling snd_pcm_readi() with ALSA set to a sample rate of 48000, format of S16_LE, channels=1, frames per period of 960 (20 mS periods), and 4 periods per buffer. This same problem happens on two different Wandboards, so I don't think it's a defective board or chip. It has happened on older kernels as well. Any ideas? Thank you! Ross Wille

2 6

[alsa-devel] iMX ASRC
by Petr Kulhavy 14 May '16

14 May '16

Hi, now after the SAI MCLK has been enabled on the iMX6UL in the mainstream kernel I'm having difficulties getting the ASRC work. The default pcm device is the SAI-codec link. That if used does not activate the ASRC. If I use the second pcm device the error message "fsl-sai 202c000.sai: isr: Transmit underrun detected" repeats forever. In /proc/asound only the HiFi-ASRC-FE is present and not the HiFi-ASRC-BE. Any idea how to get it working and/or how to debug this? Thanks! Petr

1 0

[alsa-devel] [PATCH] ASoC: MAX9860: new driver
by Peter Rosin 14 May '16

14 May '16

This is a driver for the MAX9860 Mono Audio Voice Codec. https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf This driver does not support sidetone since the DVST register field is backwards with the mute near the maximum level instead of the minimum. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- .../devicetree/bindings/sound/max9860.txt | 27 + MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 722 +++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 +++++ 6 files changed, 926 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max9860.txt create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h diff --git a/Documentation/devicetree/bindings/sound/max9860.txt b/Documentation/devicetree/bindings/sound/max9860.txt new file mode 100644 index 000000000000..e98b59a0b5aa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max9860.txt @@ -0,0 +1,27 @@ +MAX9860 Mono Audio Voice Codec + +Required properties: + + - compatible : "maxim,max9860" + + - reg : the I2C address of the device + + - AVDD-supply and DVDD-supply : power supplies for the + device, as covered in bindings/regulator/regulator.txt + + - clock-names : Required element: "mclk". + + - clocks : A clock specifier for the clock connected as MCLK. + +Examples: + + max9860: max9860@10 { + compatible = "maxim,max9860"; + reg = <0x10>; + + AVDD-supply = <&reg_1v8>; + DVDD-supply = <&reg_1v8>; + + clock-names = "mclk"; + clocks = <&pck2>; + }; diff --git a/MAINTAINERS b/MAINTAINERS index a727d9959ecd..0803396b10f4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7000,6 +7000,13 @@ F: Documentation/devicetree/bindings/i2c/max6697.txt F: drivers/hwmon/max6697.c F: include/linux/platform_data/max6697.h +MAX9860 MONO AUDIO VOICE CODEC DRIVER +M: Peter Rosin <peda(a)axentia.se> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/max9860.txt +F: sound/soc/codecs/max9860* + MAXIM MUIC CHARGER DRIVERS FOR EXYNOS BASED BOARDS M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> L: linux-pm(a)vger.kernel.org diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c16478..241a23c9aa9f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C + select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -534,6 +535,11 @@ config SND_SOC_MAX98926 config SND_SOC_MAX9850 tristate +config SND_SOC_MAX9860 + tristate "Maxim MAX9860 Mono Audio Voice Codec" + depends on I2C + select REGMAP_I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..e435f4df1787 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,6 +78,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o +snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o @@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c new file mode 100644 index 000000000000..d04630042de7 --- /dev/null +++ b/sound/soc/codecs/max9860.c @@ -0,0 +1,722 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf + * + * The driver does not support sidetone since the DVST register field is + * backwards with the mute near the maximum level instead of the minimum. + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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. + * + * 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 <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "max9860.h" + +struct max9860_priv { + struct regmap *regmap; + u8 psclk; + unsigned long pclk_rate; + int fmt; +}; + +static const struct reg_default max9860_reg_defaults[] = { + { MAX9860_PWRMAN, 0x00 }, + { MAX9860_INTEN, 0x00 }, + { MAX9860_SYSCLK, 0x00 }, + { MAX9860_AUDIOCLKHIGH, 0x00 }, + { MAX9860_AUDIOCLKLOW, 0x00 }, + { MAX9860_IFC1A, 0x00 }, + { MAX9860_IFC1B, 0x00 }, + { MAX9860_VOICEFLTR, 0x00 }, + { MAX9860_DACATTN, 0x00 }, + { MAX9860_ADCLEVEL, 0x00 }, + { MAX9860_DACGAIN, 0x00 }, + { MAX9860_MICGAIN, 0x00 }, + { MAX9860_MICADC, 0x00 }, + { MAX9860_NOISEGATE, 0x00 }, +}; + +static bool max9860_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + case MAX9860_REVISION: + return true; + } + + return false; +} + +static bool max9860_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTEN ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + return true; + } + + return false; +} + +static bool max9860_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + case MAX9860_MICREADBACK: + return true; + } + + return false; +} + +static bool max9860_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + return true; + } + + return false; +} + +const struct regmap_config max9860_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max9860_readable, + .writeable_reg = max9860_writeable, + .volatile_reg = max9860_volatile, + .precious_reg = max9860_precious, + + .max_register = MAX9860_MAX_REGISTER, + .reg_defaults = max9860_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_RANGE(pam_tlv, + 0, MAX9860_PAM_MAX - 1, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + MAX9860_PAM_MAX, MAX9860_PAM_MAX, TLV_DB_SCALE_ITEM(3000, 0, 0)); +static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1); +static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0); + +static const char * const agchld_text[] = { + "AGC Disabled", "50ms", "100ms", "400ms" +}; + +static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC, + MAX9860_AGCHLD_SHIFT, agchld_text); + +static const char * const agcsrc_text[] = { + "Left ADC", "Left/Right ADC" +}; + +static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC, + MAX9860_AGCSRC_SHIFT, agcsrc_text); + +static const char * const agcatk_text[] = { + "3ms", "12ms", "50ms", "200ms" +}; + +static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC, + MAX9860_AGCATK_SHIFT, agcatk_text); + +static const char * const agcrls_text[] = { + "78ms", "156ms", "312ms", "625ms", + "1.25s", "2.5s", "5s", "10s" +}; + +static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC, + MAX9860_AGCRLS_SHIFT, agcrls_text); + +static const char * const filter_text[] = { + "Disabled", + "Elliptical HP 217Hz notch (16kHz)", + "Butterworth HP 500Hz (16kHz)", + "Elliptical HP 217Hz notch (8kHz)", + "Butterworth HP 500Hz (8kHz)", + "Butterworth HP 200Hz (48kHz)" +}; + +static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR, + MAX9860_AVFLT_SHIFT, filter_text); + +static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR, + MAX9860_DVFLT_SHIFT, filter_text); + +static const struct snd_kcontrol_new max9860_controls[] = { +SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN, + MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv), +SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN, + MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv), +SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL, + MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1, + adc_tlv), + +SOC_ENUM("AGC Hold Time", agchld_enum), +SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum), +SOC_ENUM("AGC Attack Time", agcatk_enum), +SOC_ENUM("AGC Release Time", agcrls_enum), + +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE, + MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv), +SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE, + MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv), + +SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN, + MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv), +SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN, + MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv), + +SOC_ENUM("ADC Filter", avflt_enum), +SOC_ENUM("DAC Filter", dvflt_enum), +}; + +static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICL"), +SND_SOC_DAPM_INPUT("MICR"), + +SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("OUT"), + +SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0), +SND_SOC_DAPM_CLOCK_SUPPLY("mclk"), +}; + +static const struct snd_soc_dapm_route max9860_dapm_routes[] = { + { "ADCL", NULL, "MICL" }, + { "ADCR", NULL, "MICR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, + + { "DAC", NULL, "AIFINL" }, + { "DAC", NULL, "AIFINR" }, + { "OUT", NULL, "DAC" }, + + { "Supply", NULL, "AVDD" }, + { "Supply", NULL, "DVDD" }, + { "Supply", NULL, "mclk" }, + + { "DAC", NULL, "Supply" }, + { "ADCL", NULL, "Supply" }, + { "ADCR", NULL, "Supply" }, +}; + +static int max9860_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + return 0; + + default: + return -EINVAL; + } +} + +static int max9860_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 max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + u8 master; + u8 ifc1a = 0; + u8 ifc1b = 0; + u8 sysclk = 0; + unsigned long n; + int ret; + + dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n", + params_rate(params), + params_channels(params)); + + if (params_channels(params) == 2) + ifc1b |= MAX9860_ST; + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + master = MAX9860_MASTER; + break; + default: + return -EINVAL; + } + ifc1a |= master; + + if (master) { + if (params_width(params) * params_channels(params) > 48) + ifc1b |= MAX9860_BSEL_64X; + else + ifc1b |= MAX9860_BSEL_48X; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifc1a |= MAX9860_DDLY; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifc1a |= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_DSP_A: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_A works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_DSP_B: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_B works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + break; + default: + return -EINVAL; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_IB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + /* fall through */ + case SND_SOC_DAIFMT_IB_NF: + ifc1a ^= MAX9860_DBCI; + ifc1b ^= MAX9860_ABCI; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "IFC1A %02x\n", ifc1a); + ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "IFC1B %02x\n", ifc1b); + ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret); + return ret; + } + + /* + * Check if Integer Clock Mode is possible, but avoid it in slave mode + * since we then do not know if lrclk is derived from pclk and the + * datasheet mentions that the frequencies have to match exactly in + * order for this to work. + */ + if (params_rate(params) == 8000 || params_rate(params) == 16000) { + if (master) { + switch (max9860->pclk_rate) { + case 12000000: + sysclk = MAX9860_FREQ_12MHZ; + break; + case 13000000: + sysclk = MAX9860_FREQ_13MHZ; + break; + case 19200000: + sysclk = MAX9860_FREQ_19_2MHZ; + break; + } + + if (sysclk && params_rate(params) == 16000) + sysclk |= MAX9860_16KHZ; + } + } + + /* + * Largest possible n: + * 65536 * 96 * 48kHz / 10MHz -> 30199 + * Smallest possible n: + * 65536 * 96 * 8kHz / 20MHz -> 2517 + * Both fit nicely in the available 15 bits, no need to apply any mask. + */ + n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params), + max9860->pclk_rate); + + if (!sysclk) { + if (params_rate(params) > 24000) + sysclk |= MAX9860_16KHZ; + + if (!master) + n |= 1; /* trigger rapid pll lock mode */ + } + + sysclk |= max9860->psclk; + dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk); + ret = regmap_write(max9860->regmap, + MAX9860_SYSCLK, sysclk); + if (ret) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "N %lu\n", n); + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKHIGH, n >> 8); + if (ret) { + dev_err(codec->dev, "Failed to set NHI: %d\n", ret); + return ret; + } + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKLOW, n & 0xff); + if (ret) { + dev_err(codec->dev, "Failed to set NLO: %d\n", ret); + return ret; + } + + if (!master) { + dev_dbg(codec->dev, "Enable PLL\n"); + ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH, + MAX9860_PLL, MAX9860_PLL); + if (ret) { + dev_err(codec->dev, "Failed to enable PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + max9860->fmt = fmt; + return 0; +} + +static const struct snd_soc_dai_ops max9860_dai_ops = { + .startup = max9860_dai_startup, + .hw_params = max9860_hw_params, + .set_fmt = max9860_set_fmt, +}; + +static struct snd_soc_dai_driver max9860_dai = { + .name = "max9860-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &max9860_dai_ops, + .symmetric_rates = 1, +}; + +static int max9860_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max9860_priv *max9860 = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, MAX9860_SHDN); + if (ret) { + dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, 0); + if (ret) { + dev_err(codec->dev, "Failed to request SHDN: %d\n", + ret); + return ret; + } + break; + } + + return 0; +} + +static struct snd_soc_codec_driver max9860_codec_driver = { + .set_bias_level = max9860_set_bias_level, + .idle_bias_off = true, + + .controls = max9860_controls, + .num_controls = ARRAY_SIZE(max9860_controls), + .dapm_widgets = max9860_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets), + .dapm_routes = max9860_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes), +}; + +#ifdef CONFIG_PM +static int max9860_suspend(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, MAX9860_PSCLK_OFF); + if (ret) { + dev_err(dev, "Failed to disable clock: %d\n", ret); + return ret; + } + + return 0; +} + +static int max9860_resume(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(max9860->regmap, false); + ret = regcache_sync(max9860->regmap); + if (ret) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, max9860->psclk); + if (ret) { + dev_err(dev, "Failed to enable clock: %d\n", ret); + return ret; + } + + return 0; +} +#endif + +const struct dev_pm_ops max9860_pm_ops = { + SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) +}; + +static int max9860_mclk_rate(struct device *dev, unsigned long *mclk_rate) +{ + struct clk *mclk = clk_get(dev, "mclk"); + int ret; + + if (IS_ERR(mclk)) { + ret = PTR_ERR(mclk); + dev_err(dev, "Failed to get MCLK: %d\n", ret); + return ret; + } + + ret = clk_prepare_enable(mclk); + if (ret) { + dev_err(dev, "Failed to enable MCLK: %d\n", ret); + clk_put(mclk); + return ret; + } + + *mclk_rate = clk_get_rate(mclk); + + clk_disable_unprepare(mclk); + clk_put(mclk); + return 0; +} + +static int max9860_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860; + int ret; + unsigned long mclk_rate; + int i; + int intr; + + max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL); + if (!max9860) + return -ENOMEM; + + max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap); + if (IS_ERR(max9860->regmap)) + return PTR_ERR(max9860->regmap); + + dev_set_drvdata(dev, max9860); + + /* + * mclk has to be in the 10MHz to 60MHz range. + * psclk is used to scale mclk into pclk so that + * pclk is in the 10MHz to 20MHz range. + */ + ret = max9860_mclk_rate(dev, &mclk_rate); + if (ret) + return ret; + + if (mclk_rate > 60000000 || mclk_rate < 10000000) { + dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n", + mclk_rate); + return -EINVAL; + } + if (mclk_rate >= 40000000) + max9860->psclk = 3; + else if (mclk_rate >= 20000000) + max9860->psclk = 2; + else + max9860->psclk = 1; + max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1); + max9860->psclk <<= MAX9860_PSCLK_SHIFT; + dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate); + + regcache_cache_bypass(max9860->regmap, true); + for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) { + ret = regmap_write(max9860->regmap, + max9860_regmap.reg_defaults[i].reg, + max9860_regmap.reg_defaults[i].def); + if (ret) { + dev_err(dev, "Failed to initialize register %u: %d\n", + max9860_regmap.reg_defaults[i].reg, ret); + return ret; + } + } + regcache_cache_bypass(max9860->regmap, false); + + ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr); + if (ret) { + dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret); + return ret; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &max9860_codec_driver, + &max9860_dai, 1); + if (ret) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); + return ret; +} +EXPORT_SYMBOL_GPL(max9860_probe); + +static int max9860_remove(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + return 0; +} + +static const struct i2c_device_id max9860_i2c_id[] = { + { "max9860", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9860_i2c_id); + +static const struct of_device_id max9860_of_match[] = { + { .compatible = "maxim,max9860", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9860_of_match); + +static struct i2c_driver max9860_i2c_driver = { + .probe = max9860_probe, + .remove = max9860_remove, + .id_table = max9860_i2c_id, + .driver = { + .name = "max9860", + .of_match_table = max9860_of_match, + .pm = &max9860_pm_ops, + }, +}; + +module_i2c_driver(max9860_i2c_driver); + +MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver"); +MODULE_AUTHOR("Peter Rosin <peda(a)axentia.se>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h new file mode 100644 index 000000000000..22041bd67a7d --- /dev/null +++ b/sound/soc/codecs/max9860.h @@ -0,0 +1,162 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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. + * + * 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. + */ + +#ifndef _SND_SOC_MAX9860 +#define _SND_SOC_MAX9860 + +#define MAX9860_INTRSTATUS 0x00 +#define MAX9860_MICREADBACK 0x01 +#define MAX9860_INTEN 0x02 +#define MAX9860_SYSCLK 0x03 +#define MAX9860_AUDIOCLKHIGH 0x04 +#define MAX9860_AUDIOCLKLOW 0x05 +#define MAX9860_IFC1A 0x06 +#define MAX9860_IFC1B 0x07 +#define MAX9860_VOICEFLTR 0x08 +#define MAX9860_DACATTN 0x09 +#define MAX9860_ADCLEVEL 0x0a +#define MAX9860_DACGAIN 0x0b +#define MAX9860_MICGAIN 0x0c +#define MAX9860_RESERVED 0x0d +#define MAX9860_MICADC 0x0e +#define MAX9860_NOISEGATE 0x0f +#define MAX9860_PWRMAN 0x10 +#define MAX9860_REVISION 0xff + +#define MAX9860_MAX_REGISTER 0xff + +/* INTRSTATUS */ +#define MAX9860_CLD 0x80 +#define MAX9860_SLD 0x40 +#define MAX9860_ULK 0x20 + +/* MICREADBACK */ +#define MAX9860_NG 0xe0 +#define MAX9860_AGC 0x1f + +/* INTEN */ +#define MAX9860_ICLD 0x80 +#define MAX9860_ISLD 0x40 +#define MAX9860_IULK 0x20 + +/* SYSCLK */ +#define MAX9860_PSCLK 0x30 +#define MAX9860_PSCLK_OFF 0x00 +#define MAX9860_PSCLK_SHIFT 4 +#define MAX9860_FREQ 0x06 +#define MAX9860_FREQ_NORMAL 0x00 +#define MAX9860_FREQ_12MHZ 0x02 +#define MAX9860_FREQ_13MHZ 0x04 +#define MAX9860_FREQ_19_2MHZ 0x06 +#define MAX9860_16KHZ 0x01 + +/* AUDIOCLKHIGH */ +#define MAX9860_PLL 0x80 +#define MAX9860_NHI 0x7f + +/* AUDIOCLKLOW */ +#define MAX9860_NLO 0xff + +/* IFC1A */ +#define MAX9860_MASTER 0x80 +#define MAX9860_WCI 0x40 +#define MAX9860_DBCI 0x20 +#define MAX9860_DDLY 0x10 +#define MAX9860_HIZ 0x08 +#define MAX9860_TDM 0x04 + +/* IFC1B */ +#define MAX9860_ABCI 0x20 +#define MAX9860_ADLY 0x10 +#define MAX9860_ST 0x08 +#define MAX9860_BSEL 0x07 +#define MAX9860_BSEL_OFF 0x00 +#define MAX9860_BSEL_64X 0x01 +#define MAX9860_BSEL_48X 0x02 +#define MAX9860_BSEL_PCLK_2 0x04 +#define MAX9860_BSEL_PCLK_4 0x05 +#define MAX9860_BSEL_PCLK_8 0x06 +#define MAX9860_BSEL_PCLK_16 0x07 + +/* VOICEFLTR */ +#define MAX9860_AVFLT 0xf0 +#define MAX9860_AVFLT_SHIFT 4 +#define MAX9860_AVFLT_COUNT 6 +#define MAX9860_DVFLT 0x0f +#define MAX9860_DVFLT_SHIFT 0 +#define MAX9860_DVFLT_COUNT 6 + +/* DACATTN */ +#define MAX9860_DVA 0xfe +#define MAX9860_DVA_SHIFT 1 +#define MAX9860_DVA_MUTE 0x5e + +/* ADCLEVEL */ +#define MAX9860_ADCRL 0xf0 +#define MAX9860_ADCRL_SHIFT 4 +#define MAX9860_ADCLL 0x0f +#define MAX9860_ADCLL_SHIFT 0 +#define MAX9860_ADCxL_MIN 15 + +/* DACGAIN */ +#define MAX9860_DVG 0x60 +#define MAX9860_DVG_SHIFT 5 +#define MAX9860_DVG_MAX 3 +#define MAX9860_DVST 0x1f +#define MAX9860_DVST_SHIFT 0 +#define MAX9860_DVST_MIN 31 + +/* MICGAIN */ +#define MAX9860_PAM 0x60 +#define MAX9860_PAM_SHIFT 5 +#define MAX9860_PAM_MAX 3 +#define MAX9860_PGAM 0x1f +#define MAX9860_PGAM_SHIFT 0 +#define MAX9860_PGAM_MIN 20 + +/* MICADC */ +#define MAX9860_AGCSRC 0x80 +#define MAX9860_AGCSRC_SHIFT 7 +#define MAX9860_AGCSRC_COUNT 2 +#define MAX9860_AGCRLS 0x70 +#define MAX9860_AGCRLS_SHIFT 4 +#define MAX9860_AGCRLS_COUNT 8 +#define MAX9860_AGCATK 0x0c +#define MAX9860_AGCATK_SHIFT 2 +#define MAX9860_AGCATK_COUNT 4 +#define MAX9860_AGCHLD 0x03 +#define MAX9860_AGCHLD_OFF 0x00 +#define MAX9860_AGCHLD_SHIFT 0 +#define MAX9860_AGCHLD_COUNT 4 + +/* NOISEGATE */ +#define MAX9860_ANTH 0xf0 +#define MAX9860_ANTH_SHIFT 4 +#define MAX9860_ANTH_MAX 15 +#define MAX9860_AGCTH 0x0f +#define MAX9860_AGCTH_SHIFT 0 +#define MAX9860_AGCTH_MIN 15 + +/* PWRMAN */ +#define MAX9860_SHDN 0x80 +#define MAX9860_DACEN 0x08 +#define MAX9860_DACEN_SHIFT 3 +#define MAX9860_ADCLEN 0x02 +#define MAX9860_ADCLEN_SHIFT 1 +#define MAX9860_ADCREN 0x01 +#define MAX9860_ADCREN_SHIFT 0 + +#endif /* _SND_SOC_MAX9860 */ -- 2.1.4

3 11

[alsa-devel] TAS571x input mixer
by Petr Kulhavy 14 May '16

14 May '16

Hi, I'm implementing input mixer controls for the TAS571x amplifier driver. The mixer register values are signed 3.23 fixed-point linear coefficients (i.e. numbers between -8.0 and +7.999), where the default value is 1.0 I would like to represent them as ALSA controls and have a few questions: * The mixing coefficients are signed, i.e. using negative values can invert the signal polarity. Does it make sense to allow negative values at all? * Does it make sense to allow values >1.0 i.e. allow digital amplification of the signal? * What is the usual way of representing linear mixer coefficients in ALSA (most of the codecs use dB though)? Shall I try to convert them to dB? Are there such functions in ALSA? Thanks Petr

1 0

[alsa-devel] Errors iterating over PCM stream hints and opening based on NAME/IOID
by Trent Reed 14 May '16

14 May '16

Hey All, (Tried to send an email earlier, it didn't seem to send?) I'm trying to understand ALSA, having a little trouble with some errors that I can't ration over. Gist Repro: https://gist.github.com/TReed0803/353ce27d37bc1ebe965854ea88c8b97b Whenever I run the above simple code (iterate over PCM stream hints, open for Playback/Capture depending on IOID; output also provided at link above) I notice that I sometimes hit errors. I'm having trouble understanding these errors, because as I understand I'm using the ALSA library correctly. For example, dmix is telling me that it "Only supports playback streams", well then, why does the hint IOID return that it supports capture as well? (same for dsnoop). More confusing are the "ALSA lib pcm_route.c:867:(find_matching_chmap) Found no matching channel map" errors. Can someone help me understand these issues, and perhaps what I'm doing wrong here? Would this point to something wrong with an installed package which perhaps configures these hints incorrectly? (Assuming that is probably the case, but I'd like to know for sure). I'm on a fairly new install of Ubuntu 16.04 LTS. Thanks, - Trent Reed

2 3

[alsa-devel] [PATCH v3 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 13 May '16

13 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 58 +++++++++++++++++++++++++++++++++++----------- sound/soc/soc-pcm.c | 3 ++- 3 files changed, 51 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..8eed5c4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..a6976d5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,9 @@ 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 ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1113,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1126,37 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1164,24 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, it should return true if the walk should be + * stopped and false otherwise. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + bool (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1201,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1300,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1960,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

1 1

[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 13 May '16

13 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ 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) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

2 5

[alsa-devel] [PATCH v3 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 13 May '16

13 May '16

Updated driver for TI's TAS5720L/M digital audio amplifiers incorporating previous feedback (thanks Mark Brown). Biggest change is that the driver is no longer using interrupts for codec fault reporting, but instead uses a polling-based approach to check for (and clear) codec fault conditions every 200ms. This is much more graceful and actually lower overhead than trying to keep the actual TAS5720 fault interrupts (that come in every 300us!) under control which is what v2 of the driver was doing. Because of the removal of interrupts the DT doc had to be updated as well, so I did not carry forward Rob Herring's earlier Acked-by because of said change. -- Andreas Dannenberg Texas Instruments Inc Changes since v2: - Switched fault handling from using interrupts to polling mode - Remove interrupt related description from DT bindings doc - Remove unlikely() to simplify an expression - Remove unnesseary tas5720_set_dai_sysclk() function stub Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 26 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 638 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 766 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

3 14

[alsa-devel] [PATCH] ASoC: rt5668: add rt5668 codec driver
by John Lin 13 May '16

13 May '16

This is the initial codec driver for rt5668. Signed-off-by: John Lin <john.lin(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5668.txt | 28 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5668.c | 3280 ++++++++++++++++++++ sound/soc/codecs/rt5668.h | 781 +++++ 5 files changed, 4097 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5668.txt create mode 100644 sound/soc/codecs/rt5668.c create mode 100644 sound/soc/codecs/rt5668.h diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt new file mode 100644 index 0000000..50d7759 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -0,0 +1,28 @@ +RT5668/RT5663 audio CODECs + +These devices support I2C only. + +Required properties: + +- compatible : "realtek,rt5668" or "realtek,rt5663" + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Pins on the device (for linking into audio routes) for RT5668/RT5663: + + * IN1P + * IN1N + * IN2P + * IN2N + * HPOL + * HPOR + +Example: + +codec: rt5668@1a { + compatible = "realtek,rt5668"; + reg = <0x1a>; + interrupts = <347>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index ac7317e..283c4bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -107,6 +107,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5645 if I2C select SND_SOC_RT5651 if I2C select SND_SOC_RT5659 if I2C + select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -610,6 +611,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5645=y default y if SND_SOC_RT5651=y default y if SND_SOC_RT5659=y + default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -618,6 +620,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5645=m default m if SND_SOC_RT5651=m default m if SND_SOC_RT5659=m + default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -660,6 +663,9 @@ config SND_SOC_RT5651 config SND_SOC_RT5659 tristate +config SND_SOC_RT5668 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 94873ee..88b16d9 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -105,6 +105,7 @@ snd-soc-rt5640-objs := rt5640.o snd-soc-rt5645-objs := rt5645.o snd-soc-rt5651-objs := rt5651.o snd-soc-rt5659-objs := rt5659.o +snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -318,6 +319,7 @@ obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o +obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c new file mode 100644 index 0000000..271fd01 --- /dev/null +++ b/sound/soc/codecs/rt5668.c @@ -0,0 +1,3280 @@ +/* + * rt5668.c -- RT5668/RT5663 ALSA SoC audio codec driver + * + * Copyright 2016 Realtek Semiconductor Corp. + * Author: John Lin <john.lin(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/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5668.h" +#include "rl6231.h" + +#define RT5668_DEVICE_ID 0x6451 +#define RT5663_DEVICE_ID 0x6406 + +enum { + CODEC_TYPE_RT5668, + CODEC_TYPE_RT5663, +}; + +struct rt5668_priv { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct delayed_work jack_detect_work; + struct snd_soc_jack *hs_jack; + struct timer_list btn_check_timer; + + int codec_type; + int sysclk; + int sysclk_src; + int lrck; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_default rt5668_reg[] = { + { 0x0000, 0x0000 }, + { 0x0001, 0xc8c8 }, + { 0x0002, 0x8080 }, + { 0x0003, 0x8000 }, + { 0x0004, 0xc80a }, + { 0x0005, 0x0000 }, + { 0x0006, 0x0000 }, + { 0x0007, 0x0000 }, + { 0x000a, 0x0000 }, + { 0x000b, 0x0000 }, + { 0x000c, 0x0000 }, + { 0x000d, 0x0000 }, + { 0x000f, 0x0808 }, + { 0x0010, 0x4000 }, + { 0x0011, 0x0000 }, + { 0x0012, 0x1404 }, + { 0x0013, 0x1000 }, + { 0x0014, 0xa00a }, + { 0x0015, 0x0404 }, + { 0x0016, 0x0404 }, + { 0x0017, 0x0011 }, + { 0x0018, 0xafaf }, + { 0x0019, 0xafaf }, + { 0x001a, 0xafaf }, + { 0x001b, 0x0011 }, + { 0x001c, 0x2f2f }, + { 0x001d, 0x2f2f }, + { 0x001e, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0020, 0x0000 }, + { 0x0021, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0024, 0x000b }, + { 0x0026, 0xc0c0 }, + { 0x0027, 0xc0c0 }, + { 0x0028, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xaaaa }, + { 0x002b, 0xaaaa }, + { 0x002c, 0xaba8 }, + { 0x002d, 0x0000 }, + { 0x002e, 0x0000 }, + { 0x002f, 0x0000 }, + { 0x0030, 0x0000 }, + { 0x0031, 0x5000 }, + { 0x0032, 0x0000 }, + { 0x0033, 0x0000 }, + { 0x0034, 0x0000 }, + { 0x0035, 0x0000 }, + { 0x003a, 0x0000 }, + { 0x003b, 0x0000 }, + { 0x003c, 0x00ff }, + { 0x003d, 0x0000 }, + { 0x003e, 0x00ff }, + { 0x003f, 0x0000 }, + { 0x0040, 0x0000 }, + { 0x0041, 0x00ff }, + { 0x0042, 0x0000 }, + { 0x0043, 0x00ff }, + { 0x0044, 0x0c0c }, + { 0x0049, 0xc00b }, + { 0x004a, 0x0000 }, + { 0x004b, 0x031f }, + { 0x004d, 0x0000 }, + { 0x004e, 0x001f }, + { 0x004f, 0x0000 }, + { 0x0050, 0x001f }, + { 0x0052, 0xf000 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x003f }, + { 0x0067, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006d, 0xff00 }, + { 0x006e, 0x2808 }, + { 0x006f, 0x000a }, + { 0x0070, 0x8000 }, + { 0x0071, 0x8000 }, + { 0x0072, 0x8000 }, + { 0x0073, 0x7000 }, + { 0x0074, 0x7770 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0078, 0x00f0 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0000 }, + { 0x007b, 0x0000 }, + { 0x007c, 0x0000 }, + { 0x007d, 0x0123 }, + { 0x007e, 0x4500 }, + { 0x007f, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0085, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x0088, 0x0000 }, + { 0x0089, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0060 }, + { 0x008f, 0x1000 }, + { 0x0091, 0x0c26 }, + { 0x0092, 0x0073 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0098, 0x0000 }, + { 0x0099, 0x0000 }, + { 0x009a, 0x0007 }, + { 0x009f, 0x0000 }, + { 0x00a0, 0x0000 }, + { 0x00a1, 0x0002 }, + { 0x00a2, 0x0001 }, + { 0x00a3, 0x0002 }, + { 0x00a4, 0x0001 }, + { 0x00ae, 0x2040 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00b9, 0x0000 }, + { 0x00ba, 0x0002 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0aaa }, + { 0x00c2, 0xaa80 }, + { 0x00c3, 0x0003 }, + { 0x00c4, 0x0000 }, + { 0x00d0, 0x0000 }, + { 0x00d1, 0x2244 }, + { 0x00d2, 0x0000 }, + { 0x00d3, 0x3300 }, + { 0x00d4, 0x2200 }, + { 0x00d9, 0x0809 }, + { 0x00da, 0x0000 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x600c }, + { 0x00ea, 0xb320 }, + { 0x00eb, 0x0000 }, + { 0x00ec, 0xb300 }, + { 0x00ed, 0x0000 }, + { 0x00ee, 0xb320 }, + { 0x00ef, 0x0000 }, + { 0x00f0, 0x0201 }, + { 0x00f1, 0x0ddd }, + { 0x00f2, 0x0ddd }, + { 0x00f6, 0x0000 }, + { 0x00f7, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x0000 }, + { 0x00fb, 0x0000 }, + { 0x00fc, 0x0000 }, + { 0x00fd, 0x0000 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6451 }, + { 0x0100, 0xaaaa }, + { 0x0101, 0x000a }, + { 0x010a, 0xaaaa }, + { 0x010b, 0xa0a0 }, + { 0x010c, 0xaeae }, + { 0x010d, 0xaaaa }, + { 0x010e, 0xaaaa }, + { 0x010f, 0xaaaa }, + { 0x0110, 0xe002 }, + { 0x0111, 0xa602 }, + { 0x0112, 0xaaaa }, + { 0x0113, 0x2000 }, + { 0x0117, 0x0f00 }, + { 0x0125, 0x0420 }, + { 0x0132, 0x0000 }, + { 0x0133, 0x0000 }, + { 0x0136, 0x5555 }, + { 0x0137, 0x5540 }, + { 0x0138, 0x3700 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x2020 }, + { 0x013b, 0x2020 }, + { 0x013c, 0x2005 }, + { 0x013f, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0147, 0x0000 }, + { 0x0148, 0x0000 }, + { 0x0160, 0x4ec0 }, + { 0x0161, 0x0080 }, + { 0x0162, 0x0200 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x000f }, + { 0x0168, 0x000f }, + { 0x0170, 0x4e80 }, + { 0x0171, 0x0080 }, + { 0x0172, 0x0200 }, + { 0x0173, 0x0800 }, + { 0x0174, 0x00ff }, + { 0x0175, 0x0000 }, + { 0x0190, 0x4131 }, + { 0x0191, 0x4131 }, + { 0x0192, 0x4131 }, + { 0x0193, 0x4131 }, + { 0x0194, 0x0000 }, + { 0x0195, 0x0000 }, + { 0x0196, 0x0000 }, + { 0x0197, 0x0000 }, + { 0x0198, 0x0000 }, + { 0x0199, 0x0000 }, + { 0x01a0, 0x1e64 }, + { 0x01a1, 0x06a3 }, + { 0x01a2, 0x0000 }, + { 0x01a3, 0x0000 }, + { 0x01a4, 0x0000 }, + { 0x01a5, 0x0000 }, + { 0x01a6, 0x0000 }, + { 0x01a7, 0x0000 }, + { 0x01a8, 0x0000 }, + { 0x01a9, 0x0000 }, + { 0x01aa, 0x0000 }, + { 0x01ab, 0x0000 }, + { 0x01b5, 0x0000 }, + { 0x01b6, 0x01c3 }, + { 0x01b7, 0x02a0 }, + { 0x01b8, 0x03e9 }, + { 0x01b9, 0x1389 }, + { 0x01ba, 0xc351 }, + { 0x01bb, 0x0009 }, + { 0x01bc, 0x0018 }, + { 0x01bd, 0x002a }, + { 0x01be, 0x004c }, + { 0x01bf, 0x0097 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0000 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0000 }, + { 0x01c6, 0x0000 }, + { 0x01c7, 0x0000 }, + { 0x01c8, 0x40af }, + { 0x01c9, 0x0702 }, + { 0x01ca, 0x0000 }, + { 0x01cb, 0x0000 }, + { 0x01cc, 0x5757 }, + { 0x01cd, 0x5757 }, + { 0x01ce, 0x5757 }, + { 0x01cf, 0x5757 }, + { 0x01d0, 0x5757 }, + { 0x01d1, 0x5757 }, + { 0x01d2, 0x5757 }, + { 0x01d3, 0x5757 }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x003c }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dc, 0x0000 }, + { 0x01de, 0x7c00 }, + { 0x01df, 0x0320 }, + { 0x01e0, 0x06a1 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0000 }, + { 0x01e6, 0x0001 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x01f3, 0x0000 }, + { 0x01f4, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x0000 }, + { 0x0202, 0x0000 }, + { 0x0203, 0x0000 }, + { 0x0204, 0x0000 }, + { 0x0205, 0x0000 }, + { 0x0206, 0x0000 }, + { 0x0207, 0x0000 }, + { 0x0208, 0x0000 }, + { 0x0210, 0x60b1 }, + { 0x0211, 0xa000 }, + { 0x0212, 0x024c }, + { 0x0213, 0xf7ff }, + { 0x0214, 0x024c }, + { 0x0215, 0x0102 }, + { 0x0216, 0x00a3 }, + { 0x0217, 0x0048 }, + { 0x0218, 0x92c0 }, + { 0x0219, 0x0000 }, + { 0x021a, 0x00c8 }, + { 0x021b, 0x0020 }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x02ff, 0x0110 }, + { 0x0300, 0x001f }, + { 0x0301, 0x032c }, + { 0x0302, 0x5f21 }, + { 0x0303, 0x4000 }, + { 0x0304, 0x4000 }, + { 0x0305, 0x06d5 }, + { 0x0306, 0x8000 }, + { 0x0307, 0x0700 }, + { 0x0310, 0x4560 }, + { 0x0311, 0xa4a8 }, + { 0x0312, 0x7418 }, + { 0x0313, 0x0000 }, + { 0x0314, 0x0006 }, + { 0x0315, 0xffff }, + { 0x0316, 0xc400 }, + { 0x0317, 0x0000 }, + { 0x0330, 0x00a6 }, + { 0x0331, 0x04c3 }, + { 0x0332, 0x27c8 }, + { 0x0333, 0xbf50 }, + { 0x0334, 0x0045 }, + { 0x0335, 0x0007 }, + { 0x0336, 0x7418 }, + { 0x0337, 0x0501 }, + { 0x0338, 0x0000 }, + { 0x0339, 0x0010 }, + { 0x033a, 0x1010 }, + { 0x03c0, 0x7e00 }, + { 0x03c1, 0x8000 }, + { 0x03c2, 0x8000 }, + { 0x03c3, 0x8000 }, + { 0x03c4, 0x8000 }, + { 0x03c5, 0x8000 }, + { 0x03c6, 0x8000 }, + { 0x03c7, 0x8000 }, + { 0x03c8, 0x8000 }, + { 0x03c9, 0x8000 }, + { 0x03ca, 0x8000 }, + { 0x03cb, 0x8000 }, + { 0x03cc, 0x8000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, + { 0x03fe, 0x0000 }, + { 0x03ff, 0x0000 }, + { 0x07f0, 0x0000 }, + { 0x07fa, 0x0000 }, +}; + +static const struct reg_default rt5663_reg[] = { + { 0x0000, 0x0000 }, + { 0x0002, 0x0008 }, + { 0x0005, 0x1000 }, + { 0x0006, 0x1000 }, + { 0x000a, 0x0000 }, + { 0x0010, 0x000f }, + { 0x0015, 0x42c1 }, + { 0x0016, 0x0000 }, + { 0x0018, 0x000b }, + { 0x0019, 0xafaf }, + { 0x001c, 0x2f2f }, + { 0x001f, 0x0000 }, + { 0x0022, 0x5757 }, + { 0x0023, 0x0039 }, + { 0x0026, 0xc0c0 }, + { 0x0029, 0x8080 }, + { 0x002a, 0xa0a0 }, + { 0x002c, 0x000c }, + { 0x002d, 0x0000 }, + { 0x0040, 0x0808 }, + { 0x0061, 0x0000 }, + { 0x0062, 0x0000 }, + { 0x0063, 0x003e }, + { 0x0064, 0x0000 }, + { 0x0065, 0x0000 }, + { 0x0066, 0x0000 }, + { 0x006b, 0x0000 }, + { 0x006e, 0x0000 }, + { 0x006f, 0x0000 }, + { 0x0070, 0x8020 }, + { 0x0073, 0x1000 }, + { 0x0074, 0xe400 }, + { 0x0075, 0x0002 }, + { 0x0076, 0x0001 }, + { 0x0077, 0x00f0 }, + { 0x0078, 0x0000 }, + { 0x0079, 0x0000 }, + { 0x007a, 0x0123 }, + { 0x007b, 0x8003 }, + { 0x0080, 0x0000 }, + { 0x0081, 0x0000 }, + { 0x0082, 0x0000 }, + { 0x0083, 0x0000 }, + { 0x0084, 0x0000 }, + { 0x0086, 0x0008 }, + { 0x0087, 0x0000 }, + { 0x008a, 0x0000 }, + { 0x008b, 0x0000 }, + { 0x008c, 0x0003 }, + { 0x008e, 0x0004 }, + { 0x008f, 0x1000 }, + { 0x0090, 0x0646 }, + { 0x0091, 0x0e3e }, + { 0x0092, 0x1071 }, + { 0x0093, 0x0000 }, + { 0x0094, 0x0080 }, + { 0x0097, 0x0000 }, + { 0x0098, 0x0000 }, + { 0x009a, 0x0000 }, + { 0x009f, 0x0000 }, + { 0x00ae, 0x2000 }, + { 0x00af, 0x0000 }, + { 0x00b6, 0x0000 }, + { 0x00b7, 0x0000 }, + { 0x00b8, 0x0000 }, + { 0x00ba, 0x0000 }, + { 0x00bb, 0x0000 }, + { 0x00be, 0x0000 }, + { 0x00bf, 0x0000 }, + { 0x00c0, 0x0000 }, + { 0x00c1, 0x0000 }, + { 0x00c5, 0x0000 }, + { 0x00cb, 0xa02f }, + { 0x00cc, 0x0000 }, + { 0x00cd, 0x0e02 }, + { 0x00d9, 0x08f9 }, + { 0x00db, 0x0008 }, + { 0x00dc, 0x00c0 }, + { 0x00dd, 0x6724 }, + { 0x00de, 0x3131 }, + { 0x00df, 0x0008 }, + { 0x00e0, 0x4000 }, + { 0x00e1, 0x3131 }, + { 0x00e2, 0x0043 }, + { 0x00e4, 0x400b }, + { 0x00e5, 0x8031 }, + { 0x00e6, 0x3080 }, + { 0x00e7, 0x4100 }, + { 0x00e8, 0x1400 }, + { 0x00e9, 0xe00a }, + { 0x00ea, 0x0404 }, + { 0x00eb, 0x0404 }, + { 0x00ec, 0xb320 }, + { 0x00ed, 0x0000 }, + { 0x00f4, 0x0000 }, + { 0x00f6, 0x0000 }, + { 0x00f8, 0x0000 }, + { 0x00fa, 0x8000 }, + { 0x00fd, 0x0001 }, + { 0x00fe, 0x10ec }, + { 0x00ff, 0x6406 }, + { 0x0100, 0xa0a0 }, + { 0x0108, 0x4444 }, + { 0x0109, 0x4444 }, + { 0x010a, 0xaaaa }, + { 0x010b, 0x00a0 }, + { 0x010c, 0x8aaa }, + { 0x010d, 0xaaaa }, + { 0x010e, 0x2aaa }, + { 0x010f, 0x002a }, + { 0x0110, 0xa0a4 }, + { 0x0111, 0x4602 }, + { 0x0112, 0x0101 }, + { 0x0113, 0x2000 }, + { 0x0114, 0x0000 }, + { 0x0116, 0x0000 }, + { 0x0117, 0x0f00 }, + { 0x0118, 0x0006 }, + { 0x0125, 0x2224 }, + { 0x0126, 0x5550 }, + { 0x0127, 0x0400 }, + { 0x0128, 0x7711 }, + { 0x0132, 0x0004 }, + { 0x0137, 0x5441 }, + { 0x0139, 0x79a1 }, + { 0x013a, 0x30c0 }, + { 0x013b, 0x2000 }, + { 0x013c, 0x2005 }, + { 0x013d, 0x30c0 }, + { 0x013e, 0x0000 }, + { 0x0140, 0x3700 }, + { 0x0141, 0x1f00 }, + { 0x0144, 0x0000 }, + { 0x0145, 0x0002 }, + { 0x0146, 0x0000 }, + { 0x0160, 0x0e80 }, + { 0x0161, 0x0020 }, + { 0x0162, 0x0080 }, + { 0x0163, 0x0800 }, + { 0x0164, 0x0000 }, + { 0x0165, 0x0000 }, + { 0x0166, 0x0000 }, + { 0x0167, 0x1417 }, + { 0x0168, 0x0017 }, + { 0x0169, 0x0017 }, + { 0x0180, 0x2000 }, + { 0x0181, 0x0000 }, + { 0x0182, 0x0000 }, + { 0x0183, 0x2000 }, + { 0x0184, 0x0000 }, + { 0x0185, 0x0000 }, + { 0x01b0, 0x4b30 }, + { 0x01b1, 0x0000 }, + { 0x01b2, 0xd870 }, + { 0x01b3, 0x0000 }, + { 0x01b4, 0x0030 }, + { 0x01b5, 0x5757 }, + { 0x01b6, 0x5757 }, + { 0x01b7, 0x5757 }, + { 0x01b8, 0x5757 }, + { 0x01c0, 0x433d }, + { 0x01c1, 0x0540 }, + { 0x01c2, 0x0000 }, + { 0x01c3, 0x0000 }, + { 0x01c4, 0x0000 }, + { 0x01c5, 0x0009 }, + { 0x01c6, 0x0018 }, + { 0x01c7, 0x002a }, + { 0x01c8, 0x004c }, + { 0x01c9, 0x0097 }, + { 0x01ca, 0x01c3 }, + { 0x01cb, 0x03e9 }, + { 0x01cc, 0x1389 }, + { 0x01cd, 0xc351 }, + { 0x01ce, 0x0000 }, + { 0x01cf, 0x0000 }, + { 0x01d0, 0x0000 }, + { 0x01d1, 0x0000 }, + { 0x01d2, 0x0000 }, + { 0x01d3, 0x003c }, + { 0x01d4, 0x5757 }, + { 0x01d5, 0x5757 }, + { 0x01d6, 0x5757 }, + { 0x01d7, 0x5757 }, + { 0x01d8, 0x5757 }, + { 0x01d9, 0x5757 }, + { 0x01da, 0x0000 }, + { 0x01db, 0x0000 }, + { 0x01dd, 0x0009 }, + { 0x01de, 0x7f00 }, + { 0x01df, 0x00c8 }, + { 0x01e0, 0x0691 }, + { 0x01e1, 0x0000 }, + { 0x01e2, 0x0000 }, + { 0x01e3, 0x0000 }, + { 0x01e4, 0x0000 }, + { 0x01e5, 0x0040 }, + { 0x01e6, 0x0000 }, + { 0x01e7, 0x0000 }, + { 0x01e8, 0x0000 }, + { 0x01ea, 0x0000 }, + { 0x01eb, 0x0000 }, + { 0x01ec, 0x0000 }, + { 0x01ed, 0x0000 }, + { 0x01ee, 0x0000 }, + { 0x01ef, 0x0000 }, + { 0x01f0, 0x0000 }, + { 0x01f1, 0x0000 }, + { 0x01f2, 0x0000 }, + { 0x0200, 0x0000 }, + { 0x0201, 0x2244 }, + { 0x0202, 0xaaaa }, + { 0x0250, 0x8010 }, + { 0x0251, 0x0000 }, + { 0x0252, 0x028a }, + { 0x02fa, 0x0000 }, + { 0x02fb, 0x0000 }, + { 0x02fc, 0x0000 }, + { 0x0300, 0x0000 }, + { 0x03d0, 0x0000 }, + { 0x03d1, 0x0000 }, + { 0x03d2, 0x0000 }, + { 0x03d3, 0x0000 }, + { 0x03d4, 0x2000 }, + { 0x03d5, 0x2000 }, + { 0x03d6, 0x0000 }, + { 0x03d7, 0x0000 }, + { 0x03d8, 0x2000 }, + { 0x03d9, 0x2000 }, + { 0x03da, 0x2000 }, + { 0x03db, 0x2000 }, + { 0x03dc, 0x0000 }, + { 0x03dd, 0x0000 }, + { 0x03de, 0x0000 }, + { 0x03df, 0x2000 }, + { 0x03e0, 0x0000 }, + { 0x03e1, 0x0000 }, + { 0x03e2, 0x0000 }, + { 0x03e3, 0x0000 }, + { 0x03e4, 0x0000 }, + { 0x03e5, 0x0000 }, + { 0x03e6, 0x0000 }, + { 0x03e7, 0x0000 }, + { 0x03e8, 0x0000 }, + { 0x03e9, 0x0000 }, + { 0x03ea, 0x0000 }, + { 0x03eb, 0x0000 }, + { 0x03ec, 0x0000 }, + { 0x03ed, 0x0000 }, + { 0x03ee, 0x0000 }, + { 0x03ef, 0x0000 }, + { 0x03f0, 0x0800 }, + { 0x03f1, 0x0800 }, + { 0x03f2, 0x0800 }, + { 0x03f3, 0x0800 }, +}; + +static bool rt5668_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0011: + case 0x0082: + case 0x00ae: + case 0x00be: + case 0x00db: + case 0x00df: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x01de: + case 0x01e7: + return true; + default: + return false; + } +} + +static bool rt5668_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0001: + case 0x0002: + case 0x0003: + case 0x0004: + case 0x0005: + case 0x0006: + case 0x0007: + case 0x000a: + case 0x000b: + case 0x000c: + case 0x000d: + case 0x000f: + case 0x0010: + case 0x0011: + case 0x0012: + case 0x0013: + case 0x0014: + case 0x0015: + case 0x0016: + case 0x0017: + case 0x0018: + case 0x0019: + case 0x001a: + case 0x001b: + case 0x001c: + case 0x001d: + case 0x001e: + case 0x001f: + case 0x0020: + case 0x0021: + case 0x0022: + case 0x0023: + case 0x0024: + case 0x0026: + case 0x0027: + case 0x0028: + case 0x0029: + case 0x002a: + case 0x002b: + case 0x002c: + case 0x002d: + case 0x002e: + case 0x002f: + case 0x0030: + case 0x0031: + case 0x0032: + case 0x0033: + case 0x0034: + case 0x0035: + case 0x003a: + case 0x003b: + case 0x003c: + case 0x003d: + case 0x003e: + case 0x003f: + case 0x0040: + case 0x0041: + case 0x0042: + case 0x0043: + case 0x0044: + case 0x0049: + case 0x004a: + case 0x004b: + case 0x004d: + case 0x004e: + case 0x004f: + case 0x0050: + case 0x0052: + case 0x0061: + case 0x0062: + case 0x0063: + case 0x0064: + case 0x0065: + case 0x0066: + case 0x0067: + case 0x006b: + case 0x006d: + case 0x006e: + case 0x006f: + case 0x0070: + case 0x0071: + case 0x0072: + case 0x0073: + case 0x0074: + case 0x0075: + case 0x0076: + case 0x0078: + case 0x0079: + case 0x007a: + case 0x007b: + case 0x007c: + case 0x007d: + case 0x007e: + case 0x007f: + case 0x0080: + case 0x0081: + case 0x0082: + case 0x0083: + case 0x0084: + case 0x0085: + case 0x0086: + case 0x0087: + case 0x0088: + case 0x0089: + case 0x008a: + case 0x008b: + case 0x008c: + case 0x008e: + case 0x008f: + case 0x0091: + case 0x0092: + case 0x0093: + case 0x0094: + case 0x0098: + case 0x0099: + case 0x009a: + case 0x009f: + case 0x00a0: + case 0x00a1: + case 0x00a2: + case 0x00a3: + case 0x00a4: + case 0x00ae: + case 0x00af: + case 0x00b6: + case 0x00b7: + case 0x00b8: + case 0x00b9: + case 0x00ba: + case 0x00bb: + case 0x00be: + case 0x00c0: + case 0x00c1: + case 0x00c2: + case 0x00c3: + case 0x00c4: + case 0x00d0: + case 0x00d1: + case 0x00d2: + case 0x00d3: + case 0x00d4: + case 0x00d9: + case 0x00da: + case 0x00db: + case 0x00dc: + case 0x00dd: + case 0x00de: + case 0x00df: + case 0x00e0: + case 0x00e1: + case 0x00e2: + case 0x00ea: + case 0x00eb: + case 0x00ec: + case 0x00ed: + case 0x00ee: + case 0x00ef: + case 0x00f0: + case 0x00f1: + case 0x00f2: + case 0x00f6: + case 0x00f7: + case 0x00f8: + case 0x00fa: + case 0x00fb: + case 0x00fc: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x0100: + case 0x0101: + case 0x010a: + case 0x010b: + case 0x010c: + case 0x010d: + case 0x010e: + case 0x010f: + case 0x0110: + case 0x0111: + case 0x0112: + case 0x0113: + case 0x0117: + case 0x0125: + case 0x0132: + case 0x0133: + case 0x0136: + case 0x0137: + case 0x0138: + case 0x0139: + case 0x013a: + case 0x013b: + case 0x013c: + case 0x013f: + case 0x0145: + case 0x0146: + case 0x0147: + case 0x0148: + case 0x0160: + case 0x0161: + case 0x0162: + case 0x0163: + case 0x0164: + case 0x0165: + case 0x0166: + case 0x0167: + case 0x0168: + case 0x0170: + case 0x0171: + case 0x0172: + case 0x0173: + case 0x0174: + case 0x0175: + case 0x0190: + case 0x0191: + case 0x0192: + case 0x0193: + case 0x0194: + case 0x0195: + case 0x0196: + case 0x0197: + case 0x0198: + case 0x0199: + case 0x01a0: + case 0x01a1: + case 0x01a2: + case 0x01a3: + case 0x01a4: + case 0x01a5: + case 0x01a6: + case 0x01a7: + case 0x01a8: + case 0x01a9: + case 0x01aa: + case 0x01ab: + case 0x01b5: + case 0x01b6: + case 0x01b7: + case 0x01b8: + case 0x01b9: + case 0x01ba: + case 0x01bb: + case 0x01bc: + case 0x01bd: + case 0x01be: + case 0x01bf: + case 0x01c0: + case 0x01c1: + case 0x01c2: + case 0x01c3: + case 0x01c4: + case 0x01c5: + case 0x01c6: + case 0x01c7: + case 0x01c8: + case 0x01c9: + case 0x01ca: + case 0x01cb: + case 0x01cc: + case 0x01cd: + case 0x01ce: + case 0x01cf: + case 0x01d0: + case 0x01d1: + case 0x01d2: + case 0x01d3: + case 0x01d4: + case 0x01d5: + case 0x01d6: + case 0x01da: + case 0x01db: + case 0x01dc: + case 0x01de: + case 0x01df: + case 0x01e0: + case 0x01e1: + case 0x01e2: + case 0x01e3: + case 0x01e4: + case 0x01e5: + case 0x01e6: + case 0x01e7: + case 0x01e8: + case 0x01ea: + case 0x01eb: + case 0x01ec: + case 0x01ed: + case 0x01ee: + case 0x01ef: + case 0x01f0: + case 0x01f1: + case 0x01f2: + case 0x01f3: + case 0x01f4: + case 0x0200: + case 0x0201: + case 0x0202: + case 0x0203: + case 0x0204: + case 0x0205: + case 0x0206: + case 0x0207: + case 0x0208: + case 0x0210: + case 0x0211: + case 0x0212: + case 0x0213: + case 0x0214: + case 0x0215: + case 0x0216: + case 0x0217: + case 0x0218: + case 0x0219: + case 0x021a: + case 0x021b: + case 0x02fa: + case 0x02fb: + case 0x02fc: + case 0x02ff: + case 0x0300: + case 0x0301: + case 0x0302: + case 0x0303: + case 0x0304: + case 0x0305: + case 0x0306: + case 0x0307: + case 0x0310: + case 0x0311: + case 0x0312: + case 0x0313: + case 0x0314: + case 0x0315: + case 0x0316: + case 0x0317: + case 0x0330: + case 0x0331: + case 0x0332: + case 0x0333: + case 0x0334: + case 0x0335: + case 0x0336: + case 0x0337: + case 0x0338: + case 0x0339: + case 0x033a: + case 0x03c0: + case 0x03c1: + case 0x03c2: + case 0x03c3: + case 0x03c4: + case 0x03c5: + case 0x03c6: + case 0x03c7: + case 0x03c8: + case 0x03c9: + case 0x03ca: + case 0x03cb: + case 0x03cc: + case 0x03d0: + case 0x03d1: + case 0x03d2: + case 0x03d3: + case 0x03d4: + case 0x03d5: + case 0x03d6: + case 0x03d7: + case 0x03d8: + case 0x03d9: + case 0x03da: + case 0x03db: + case 0x03dc: + case 0x03dd: + case 0x03de: + case 0x03df: + case 0x03e0: + case 0x03e1: + case 0x03e2: + case 0x03e3: + case 0x03e4: + case 0x03e5: + case 0x03e6: + case 0x03e7: + case 0x03e8: + case 0x03e9: + case 0x03ea: + case 0x03eb: + case 0x03ec: + case 0x03ed: + case 0x03ee: + case 0x03ef: + case 0x03f0: + case 0x03f1: + case 0x03f2: + case 0x03f3: + case 0x03fe: + case 0x03ff: + case 0x07f0: + case 0x07fa: + return true; + default: + return false; + } +} + +static bool rt5663_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0015: + case 0x0023: + case 0x0029: + case 0x0076: + case 0x0093: + case 0x0097: + case 0x00ae: + case 0x00be: + case 0x00bf: + case 0x00c5: + case 0x00cb: + case 0x00db: + case 0x00df: + case 0x00e4: + case 0x00e6: + case 0x00e7: + case 0x00e8: + case 0x00f8: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x0139: + case 0x0141: + case 0x0160: + case 0x0164: + case 0x0165: + case 0x0166: + case 0x0180: + case 0x0183: + case 0x01c0: + case 0x01c2: + case 0x01c3: + case 0x01c4: + case 0x01de: + case 0x01e6: + case 0x01ea: + case 0x01eb: + case 0x01ec: + case 0x01ed: + case 0x01ee: + case 0x01ef: + case 0x01f0: + case 0x01f1: + case 0x01f2: + case 0x0300: + return true; + default: + return false; + } +} + +static bool rt5663_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0000: + case 0x0002: + case 0x0005: + case 0x0006: + case 0x000a: + case 0x0010: + case 0x0015: + case 0x0016: + case 0x0018: + case 0x0019: + case 0x001c: + case 0x001f: + case 0x0022: + case 0x0023: + case 0x0026: + case 0x0029: + case 0x002a: + case 0x002c: + case 0x002d: + case 0x0040: + case 0x0061: + case 0x0062: + case 0x0063: + case 0x0064: + case 0x0065: + case 0x0066: + case 0x006b: + case 0x006e: + case 0x006f: + case 0x0070: + case 0x0073: + case 0x0074: + case 0x0075: + case 0x0076: + case 0x0077: + case 0x0078: + case 0x0079: + case 0x007a: + case 0x007b: + case 0x0080: + case 0x0081: + case 0x0082: + case 0x0083: + case 0x0084: + case 0x0086: + case 0x0087: + case 0x008a: + case 0x008b: + case 0x008c: + case 0x008e: + case 0x008f: + case 0x0090: + case 0x0091: + case 0x0092: + case 0x0093: + case 0x0094: + case 0x0097: + case 0x0098: + case 0x009a: + case 0x009f: + case 0x00ae: + case 0x00af: + case 0x00b6: + case 0x00b7: + case 0x00b8: + case 0x00ba: + case 0x00bb: + case 0x00be: + case 0x00bf: + case 0x00c0: + case 0x00c1: + case 0x00c5: + case 0x00cb: + case 0x00cc: + case 0x00cd: + case 0x00d9: + case 0x00db: + case 0x00dc: + case 0x00dd: + case 0x00de: + case 0x00df: + case 0x00e0: + case 0x00e1: + case 0x00e2: + case 0x00e4: + case 0x00e5: + case 0x00e6: + case 0x00e7: + case 0x00e8: + case 0x00e9: + case 0x00ea: + case 0x00eb: + case 0x00ec: + case 0x00ed: + case 0x00f4: + case 0x00f6: + case 0x00f8: + case 0x00fa: + case 0x00fd: + case 0x00fe: + case 0x00ff: + case 0x0100: + case 0x0108: + case 0x0109: + case 0x010a: + case 0x010b: + case 0x010c: + case 0x010d: + case 0x010e: + case 0x010f: + case 0x0110: + case 0x0111: + case 0x0112: + case 0x0113: + case 0x0114: + case 0x0116: + case 0x0117: + case 0x0118: + case 0x0125: + case 0x0126: + case 0x0127: + case 0x0128: + case 0x0132: + case 0x0137: + case 0x0139: + case 0x013a: + case 0x013b: + case 0x013c: + case 0x013d: + case 0x013e: + case 0x0140: + case 0x0141: + case 0x0144: + case 0x0145: + case 0x0146: + case 0x0160: + case 0x0161: + case 0x0162: + case 0x0163: + case 0x0164: + case 0x0165: + case 0x0166: + case 0x0167: + case 0x0168: + case 0x0169: + case 0x0180: + case 0x0181: + case 0x0182: + case 0x0183: + case 0x0184: + case 0x0185: + case 0x01b0: + case 0x01b1: + case 0x01b2: + case 0x01b3: + case 0x01b4: + case 0x01b5: + case 0x01b6: + case 0x01b7: + case 0x01b8: + case 0x01c0: + case 0x01c1: + case 0x01c2: + case 0x01c3: + case 0x01c4: + case 0x01c5: + case 0x01c6: + case 0x01c7: + case 0x01c8: + case 0x01c9: + case 0x01ca: + case 0x01cb: + case 0x01cc: + case 0x01cd: + case 0x01ce: + case 0x01cf: + case 0x01d0: + case 0x01d1: + case 0x01d2: + case 0x01d3: + case 0x01d4: + case 0x01d5: + case 0x01d6: + case 0x01d7: + case 0x01d8: + case 0x01d9: + case 0x01da: + case 0x01db: + case 0x01dd: + case 0x01de: + case 0x01df: + case 0x01e0: + case 0x01e1: + case 0x01e2: + case 0x01e3: + case 0x01e4: + case 0x01e5: + case 0x01e6: + case 0x01e7: + case 0x01e8: + case 0x01ea: + case 0x01eb: + case 0x01ec: + case 0x01ed: + case 0x01ee: + case 0x01ef: + case 0x01f0: + case 0x01f1: + case 0x01f2: + case 0x0200: + case 0x0201: + case 0x0202: + case 0x0250: + case 0x0251: + case 0x0252: + case 0x02fa: + case 0x02fb: + case 0x02fc: + case 0x0300: + case 0x03d0: + case 0x03d1: + case 0x03d2: + case 0x03d3: + case 0x03d4: + case 0x03d5: + case 0x03d6: + case 0x03d7: + case 0x03d8: + case 0x03d9: + case 0x03da: + case 0x03db: + case 0x03dc: + case 0x03dd: + case 0x03de: + case 0x03df: + case 0x03e0: + case 0x03e1: + case 0x03e2: + case 0x03e3: + case 0x03e4: + case 0x03e5: + case 0x03e6: + case 0x03e7: + case 0x03e8: + case 0x03e9: + case 0x03ea: + case 0x03eb: + case 0x03ec: + case 0x03ed: + case 0x03ee: + case 0x03ef: + case 0x03f0: + case 0x03f1: + case 0x03f2: + case 0x03f3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(rt5663_hp_vol_tlv, -2400, 150, 0); +static const DECLARE_TLV_DB_SCALE(rt5668_hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -1725, 75, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(in_bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5663_if1_adc_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static const SOC_ENUM_SINGLE_DECL(rt5663_if1_adc_enum, RT5668_TDM_1, + RT5663_DATA_SWAP_ADCDAT1_SHIFT, rt5663_if1_adc_data_select); + +static void rt5668_enable_push_button_irq(struct snd_soc_codec *codec, + bool enable) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + if (enable) { + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_EN); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + if (rt5668->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5668_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_NOR); + else + snd_soc_update_bits(codec, RT5668_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_NOR); + } else { + if (rt5668->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5668_IRQ_3, + RT5668_EN_IRQ_INLINE_MASK, + RT5668_EN_IRQ_INLINE_BYP); + else + snd_soc_update_bits(codec, RT5668_IRQ_2, + RT5663_EN_IRQ_INLINE_MASK, + RT5663_EN_IRQ_INLINE_BYP); + + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_EN_4BTN_INL_MASK, RT5668_EN_4BTN_INL_DIS); + /* reset in-line command */ + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_RESET); + snd_soc_update_bits(codec, RT5668_4BTN_IL_CMD_2, + RT5668_RESET_4BTN_INL_MASK, + RT5668_RESET_4BTN_INL_NOR); + } +} + +/** + * rt5668_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5668_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + if (jack_insert) { + snd_soc_write(codec, RT5668_CBJ_TYPE_2, 0x8040); + snd_soc_write(codec, RT5668_CBJ_TYPE_3, 0x1484); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_force_enable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_update_bits(codec, RT5668_MICBIAS_2, + RT5668_DIG_1M_CLK_MASK, RT5668_DIG_1M_CLK_EN); + snd_soc_update_bits(codec, RT5668_CBJ_TYPE_1, 0x8, 0x8); + + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5668_CBJ_TYPE_2) & 0x0003; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-0011 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + i++; + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5668_enable_push_button_irq(codec, true); + break; + default: + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + rt5668->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + snd_soc_update_bits(codec, RT5668_CBJ_TYPE_1, 0x8, 0x0); + + if (rt5668->jack_type == SND_JACK_HEADSET) { + rt5668_enable_push_button_irq(codec, false); + snd_soc_dapm_disable_pin(dapm, "MICBIAS1"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS2"); + snd_soc_dapm_disable_pin(dapm, "Mic Det Power"); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + } + rt5668->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +/** + * rt5663_jack_detect - Detect headset. + * @codec: SoC audio codec device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ + +static int rt5663_jack_detect(struct snd_soc_codec *codec, int jack_insert) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val, i = 0, sleep_time[5] = {300, 150, 100, 50, 30}; + + dev_dbg(codec->dev, "%s jack_insert:%d\n", __func__, jack_insert); + + if (jack_insert) { + while (i < 5) { + msleep(sleep_time[i]); + val = snd_soc_read(codec, RT5663_EM_JACK_TYPE_2) & + 0x0003; + i++; + if (val == 0x1 || val == 0x2 || val == 0x3) + break; + dev_dbg(codec->dev, "%s: MX-00e7 val=%x sleep %d\n", + __func__, val, sleep_time[i]); + } + dev_dbg(codec->dev, "%s val = %d\n", __func__, val); + switch (val) { + case 1: + case 2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5668_enable_push_button_irq(codec, true); + break; + default: + rt5668->jack_type = SND_JACK_HEADPHONE; + break; + } + } else { + if (rt5668->jack_type == SND_JACK_HEADSET) + rt5668_enable_push_button_irq(codec, false); + rt5668->jack_type = 0; + } + + dev_dbg(codec->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +int rt5668_button_detect(struct snd_soc_codec *codec) +{ + int btn_type, val; + + val = snd_soc_read(codec, RT5668_4BTN_IL_CMD_1); + dev_dbg(codec->dev, "%s: val=0x%x\n", __func__, val); + btn_type = val & 0xfff0; + snd_soc_write(codec, RT5668_4BTN_IL_CMD_1, val); + + return btn_type; +} +EXPORT_SYMBOL(rt5668_button_detect); + +static irqreturn_t rt5668_irq(int irq, void *data) +{ + struct rt5668_priv *rt5668 = data; + + dev_dbg(rt5668->codec->dev, "%s IRQ queue work\n", __func__); + + queue_delayed_work(system_wq, &rt5668->jack_detect_work, + msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5668_btn_check_callback(unsigned long data) +{ + struct rt5668_priv *rt5668 = (struct rt5668_priv *)data; + + queue_delayed_work(system_wq, &rt5668->jack_detect_work, + msecs_to_jiffies(5)); +} + +int rt5668_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + rt5668->hs_jack = hs_jack; + + rt5668_irq(0, rt5668); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5668_set_jack_detect); + +static int rt5668_check_jd_status(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int val; + + /* JD1 */ + if (rt5668->codec_type == CODEC_TYPE_RT5668) + val = snd_soc_read(codec, RT5668_INT_ST_1) & 0x2000; + else + val = snd_soc_read(codec, RT5668_INT_ST_1) & 0x1000; + + if (!val) { /* jack insert */ + dev_dbg(codec->dev, "%s-Jack In\n", __func__); + return 1; + } + + /* jack out */ + dev_dbg(codec->dev, "%s-Jack Out\n", __func__); + return 0; +} + +static void rt5668_jack_detect_work(struct work_struct *work) +{ + struct rt5668_priv *rt5668 = + container_of(work, struct rt5668_priv, jack_detect_work.work); + struct snd_soc_codec *codec = rt5668->codec; + int btn_type, report = 0; + + if (!codec) + return; + + if (rt5668_check_jd_status(codec)) { + /* jack in */ + if (rt5668->jack_type == 0) { + /* jack was out, report jack type */ + if (rt5668->codec_type == CODEC_TYPE_RT5668) + report = rt5668_jack_detect(rt5668->codec, 1); + else + report = rt5663_jack_detect(rt5668->codec, 1); + } else { + /* jack is already in, report button event */ + report = SND_JACK_HEADSET; + btn_type = rt5668_button_detect(rt5668->codec); + /** + * rt5668 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + report |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + report |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + report |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + report |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5668->codec->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + + /* button release or spurious interrput*/ + if (btn_type == 0) + report = rt5668->jack_type; + else + mod_timer(&rt5668->btn_check_timer, + msecs_to_jiffies(100)); + } + } else { + /* jack out */ + if (rt5668->codec_type == CODEC_TYPE_RT5668) + report = rt5668_jack_detect(rt5668->codec, 0); + else + report = rt5663_jack_detect(rt5668->codec, 0); + } + dev_dbg(codec->dev, "%s jack report: 0x%04x\n", __func__, report); + snd_soc_jack_report(rt5668->hs_jack, report, SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); +} + +static const struct snd_kcontrol_new rt5668_snd_controls[] = { + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC Playback Volume", RT5668_DAC1_DIG_VOL, + RT5668_DAC_L1_VOL_SHIFT + 1, RT5668_DAC_R1_VOL_SHIFT + 1, + 87, 0, dac_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5668_STO1_ADC_DIG_VOL, + RT5668_ADC_L_MUTE_SHIFT, RT5668_ADC_R_MUTE_SHIFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5668_STO1_ADC_DIG_VOL, + RT5668_ADC_L_VOL_SHIFT + 1, RT5668_ADC_R_VOL_SHIFT + 1, + 63, 0, adc_vol_tlv), +}; + +static const struct snd_kcontrol_new rt5668_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5668_HPL_GAIN, + RT5668_HPR_GAIN, RT5668_GAIN_HP_SHIFT, 15, 1, + rt5668_hp_vol_tlv), + /* Mic Boost Volume */ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5668_AEC_BST, + RT5668_GAIN_CBJ_SHIFT, 8, 0, in_bst_tlv), +}; + +static const struct snd_kcontrol_new rt5663_specific_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5663_STO_DRE_9, + RT5663_STO_DRE_10, RT5663_DRE_GAIN_HP_SHIFT, 23, 1, + rt5663_hp_vol_tlv), + /* Mic Boost Volume*/ + SOC_SINGLE_TLV("IN1 Capture Volume", RT5663_CBJ_2, + RT5663_GAIN_BST1_SHIFT, 8, 0, in_bst_tlv), + /* Data Swap for Slot0/1 in ADCDAT1 */ + SOC_ENUM("IF1 ADC Data Swap", rt5663_if1_adc_enum), +}; + +static int rt5668_is_sys_clk_from_pll(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + val = snd_soc_read(codec, RT5668_GLB_CLK); + val &= RT5668_SCLK_SRC_MASK; + if (val == RT5668_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int rt5668_is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + switch (w->shift) { + case RT5668_ADC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_3; + shift = RT5668_AD_STO1_TRACK_SHIFT; + break; + case RT5668_DAC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_2; + shift = RT5668_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } else { + switch (w->shift) { + case RT5663_ADC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_2; + shift = RT5663_AD_STO1_TRACK_SHIFT; + break; + case RT5663_DAC_STO1_ASRC_SHIFT: + reg = RT5668_ASRC_2; + shift = RT5668_DA_STO1_TRACK_SHIFT; + break; + default: + return 0; + } + } + + val = (snd_soc_read(codec, reg) >> shift) & 0x7; + + if (val) + return 1; + + return 0; +} + +static int rt5668_i2s_use_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(source->dapm); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + int da_asrc_en, ad_asrc_en; + + da_asrc_en = (snd_soc_read(codec, RT5668_ASRC_2) & + RT5668_DA_STO1_TRACK_MASK) ? 1 : 0; + if (rt5668->codec_type == CODEC_TYPE_RT5668) + ad_asrc_en = (snd_soc_read(codec, RT5668_ASRC_3) & + RT5668_AD_STO1_TRACK_MASK) ? 1 : 0; + else + ad_asrc_en = (snd_soc_read(codec, RT5668_ASRC_2) & + RT5663_AD_STO1_TRACK_MASK) ? 1 : 0; + + + if (da_asrc_en || ad_asrc_en) + if (rt5668->sysclk > rt5668->lrck * 384) + return 1; + + dev_err(codec->dev, "sysclk < 384 x fs, disable i2s asrc\n"); + + return 0; +} + +/** + * rt5668_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @codec: SoC audio codec device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5668 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the codec driver will turn on ASRC + * for these filters if ASRC is selected as their clock source. + */ +int rt5668_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int asrc2_mask = 0; + unsigned int asrc2_value = 0; + unsigned int asrc3_mask = 0; + unsigned int asrc3_value = 0; + + switch (clk_src) { + case RT5668_CLK_SEL_SYS: + case RT5668_CLK_SEL_I2S1_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5668_DA_STEREO_FILTER) { + asrc2_mask |= RT5668_DA_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5668_DA_STO1_TRACK_SHIFT; + } + + if (filter_mask & RT5668_AD_STEREO_FILTER) { + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + asrc3_mask |= RT5668_AD_STO1_TRACK_MASK; + asrc3_value |= clk_src << RT5668_AD_STO1_TRACK_SHIFT; + } else { + asrc2_mask |= RT5663_AD_STO1_TRACK_MASK; + asrc2_value |= clk_src << RT5663_AD_STO1_TRACK_SHIFT; + } + } + + if (asrc2_mask) + snd_soc_update_bits(codec, RT5668_ASRC_2, asrc2_mask, + asrc2_value); + + if (asrc3_mask) + snd_soc_update_bits(codec, RT5668_ASRC_3, asrc3_mask, + asrc3_value); + + return 0; +} +EXPORT_SYMBOL_GPL(rt5668_sel_asrc_clk_src); + +/* Analog Mixer */ +static const struct snd_kcontrol_new rt5668_recmix1l[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST2_SHIFT, 1, 1), + SOC_DAPM_SINGLE("BST1 CBJ Switch", RT5668_RECMIX1L, + RT5668_RECMIX1L_BST1_CBJ_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_recmix1r[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5668_RECMIX1R, + RT5668_RECMIX1R_BST2_SHIFT, 1, 1), +}; + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5668_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R1_SHIFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R2_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_adda_l_mix[] = { + SOC_DAPM_SINGLE("ADC L Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC L Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_adda_r_mix[] = { + SOC_DAPM_SINGLE("ADC R Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_R_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_L_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_L_SHIFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_L1_STO_R_SHIFT, 1, 1), + SOC_DAPM_SINGLE("DAC R Switch", RT5668_STO_DAC_MIXER, + RT5668_M_DAC_R1_STO_R_SHIFT, 1, 1), +}; + +/* Out Switch */ +static const struct snd_kcontrol_new rt5668_hpo_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_AMP_2, + RT5668_EN_DAC_HPO_SHIFT, 1, 0); + +/* Stereo ADC source */ +static const char * const rt5668_sto1_adc_src[] = { + "ADC L", "ADC R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcl_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_L_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcl_mux = + SOC_DAPM_ENUM("STO1 ADC L Mux", rt5668_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5668_sto1_adcr_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_R_SRC_SHIFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcr_mux = + SOC_DAPM_ENUM("STO1 ADC R Mux", rt5668_sto1_adcr_enum); + +/* RT5663: Analog DACL1 input source */ +static const char * const rt5663_alg_dacl_src[] = { + "DAC L", "STO DAC MIXL" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacl_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACL1_SRC_SHIFT, rt5663_alg_dacl_src); + +static const struct snd_kcontrol_new rt5668_alg_dacl_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5663_alg_dacl_enum); + +/* RT5663: Analog DACR1 input source */ +static const char * const rt5663_alg_dacr_src[] = { + "DAC R", "STO DAC MIXR" +}; + +static SOC_ENUM_SINGLE_DECL(rt5663_alg_dacr_enum, RT5663_BYPASS_STO_DAC, + RT5663_DACR1_SRC_SHIFT, rt5663_alg_dacr_src); + +static const struct snd_kcontrol_new rt5668_alg_dacr_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5663_alg_dacr_enum); + +static int rt5668_hp_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 rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_HP_CHARGE_PUMP_1, + RT5668_SEL_PM_HP_SHIFT, RT5668_SEL_PM_HP_HIGH); + snd_soc_update_bits(codec, RT5668_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_SILENCE); + } else { + snd_soc_write(codec, RT5668_DEPOP_2, 0x3003); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x000b, + 0x000b); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x0030, + 0x0030); + snd_soc_update_bits(codec, RT5668_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_DIS); + snd_soc_write(codec, RT5668_HP_CHARGE_PUMP_2, 0x1371); + snd_soc_write(codec, RT5663_HP_BIAS, 0xabba); + snd_soc_write(codec, RT5663_CHARGE_PUMP_1, 0x2224); + snd_soc_write(codec, RT5663_ANA_BIAS_CUR_1, 0x7766); + snd_soc_write(codec, RT5663_HP_BIAS, 0xafaa); + snd_soc_write(codec, RT5663_CHARGE_PUMP_2, 0x7777); + snd_soc_write(codec, RT5663_VREF_RECMIX, 0x0007); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x3000, + 0x3000); + } + break; + + case SND_SOC_DAPM_PRE_PMD: + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, + RT5668_HP_SIG_SRC1_REG); + } else { + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x3000, 0x0); + snd_soc_update_bits(codec, RT5668_HP_CHARGE_PUMP_1, + RT5668_OVCD_HP_MASK, RT5668_OVCD_HP_EN); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x0030, 0x0); + snd_soc_update_bits(codec, RT5668_DEPOP_1, 0x000b, + 0x000b); + } + break; + + default: + return 0; + } + + return 0; +} + +static int rt5668_bst2_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, + RT5668_PWR_BST2 | RT5668_PWR_BST2_OP); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_2, + RT5668_PWR_BST2_MASK | RT5668_PWR_BST2_OP_MASK, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5663_pre_div_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0xff00); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0xfffc); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_write(codec, RT5663_PRE_DIV_GATING_1, 0x0000); + snd_soc_write(codec, RT5663_PRE_DIV_GATING_2, 0x0000); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5668_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PLL", RT5668_PWR_ANLG_3, RT5668_PWR_PLL_SHIFT, 0, + NULL, 0), + + /* micbias */ + SND_SOC_DAPM_MICBIAS("MICBIAS1", RT5668_PWR_ANLG_2, + RT5668_PWR_MB1_SHIFT, 0), + SND_SOC_DAPM_MICBIAS("MICBIAS2", RT5668_PWR_ANLG_2, + RT5668_PWR_MB2_SHIFT, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + + /* REC Mixer Power */ + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5668_PWR_ANLG_2, + RT5668_PWR_RECMIX1_SHIFT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC L Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", RT5668_CHOP_ADC, + RT5668_CKGEN_ADCC_SHIFT, 0, NULL, 0), + + /* ADC Mixer */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXL", SND_SOC_NOPM, + 0, 0, rt5668_sto1_adc_l_mix, + ARRAY_SIZE(rt5668_sto1_adc_l_mix)), + + /* ADC Filter Power */ + SND_SOC_DAPM_SUPPLY("STO1 ADC Filter", RT5668_PWR_DIG_2, + RT5668_PWR_ADC_S1F_SHIFT, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S", RT5668_PWR_DIG_1, RT5668_PWR_I2S1_SHIFT, 0, + NULL, 0), + SND_SOC_DAPM_PGA("IF DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIFTX", "AIF Capture", 0, SND_SOC_NOPM, 0, 0), + + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("ADDA MIXL", SND_SOC_NOPM, 0, 0, rt5668_adda_l_mix, + ARRAY_SIZE(rt5668_adda_l_mix)), + SND_SOC_DAPM_MIXER("ADDA MIXR", SND_SOC_NOPM, 0, 0, rt5668_adda_r_mix, + ARRAY_SIZE(rt5668_adda_r_mix)), + SND_SOC_DAPM_PGA("DAC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("STO1 DAC Filter", RT5668_PWR_DIG_2, + RT5668_PWR_DAC_S1F_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("STO1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_l_mix, ARRAY_SIZE(rt5668_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("STO1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_r_mix, ARRAY_SIZE(rt5668_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_SUPPLY("STO1 DAC L Power", RT5668_PWR_DIG_1, + RT5668_PWR_DAC_L1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("STO1 DAC R Power", RT5668_PWR_DIG_1, + RT5668_PWR_DAC_R1_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_DAC("DAC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R", NULL, SND_SOC_NOPM, 0, 0), + + /* Headphone*/ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5668_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_widget rt5668_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5668_PWR_ANLG_3, + RT5668_PWR_LDO2_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5668_PWR_VOL, + RT5668_PWR_MIC_DET_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("LDO DAC", RT5668_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5668_ASRC_1, + RT5668_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5668_ASRC_1, + RT5668_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5668_ASRC_1, + RT5668_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5668_PWR_ANLG_3, + RT5668_PWR_CBJ_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("BST2 Power", SND_SOC_NOPM, 0, 0, + rt5668_bst2_power, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5668_recmix1l, + ARRAY_SIZE(rt5668_recmix1l)), + SND_SOC_DAPM_MIXER("RECMIX1R", SND_SOC_NOPM, 0, 0, rt5668_recmix1r, + ARRAY_SIZE(rt5668_recmix1r)), + SND_SOC_DAPM_SUPPLY("RECMIX1R Power", RT5668_PWR_ANLG_2, + RT5668_PWR_RECMIX2_SHIFT, 0, NULL, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC R Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_R1_SHIFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_PGA("STO1 ADC L1", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_L1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R1", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_R1_SRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_L2_SRC_SHIFT, 1, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC R2", RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC_R2_SRC_SHIFT, 1, NULL, 0), + + SND_SOC_DAPM_MUX("STO1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcl_mux), + SND_SOC_DAPM_MUX("STO1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcr_mux), + + /* ADC Mix */ + SND_SOC_DAPM_MIXER("STO1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_adc_r_mix, ARRAY_SIZE(rt5668_sto1_adc_r_mix)), + + /* Analog DAC Clock */ + SND_SOC_DAPM_SUPPLY("DAC Clock", RT5668_CHOP_DAC, + RT5668_CKGEN_DAC1_SHIFT, 0, NULL, 0), + + /* Headphone out */ + SND_SOC_DAPM_SWITCH("HPO Playback", SND_SOC_NOPM, 0, 0, + &rt5668_hpo_switch), +}; + +static const struct snd_soc_dapm_widget rt5663_specific_dapm_widgets[] = { + /* System Clock Pre Divider Gating */ + SND_SOC_DAPM_SUPPLY("Pre Div Power", SND_SOC_NOPM, 0, 0, + rt5663_pre_div_power, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD), + + /* LDO */ + SND_SOC_DAPM_SUPPLY("LDO ADC", RT5668_PWR_DIG_1, + RT5668_PWR_LDO_DACREF_SHIFT, 0, NULL, 0), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY("I2S ASRC", RT5668_ASRC_1, + RT5663_I2S1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC ASRC", RT5668_ASRC_1, + RT5663_DAC_STO1_ASRC_SHIFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC ASRC", RT5668_ASRC_1, + RT5663_ADC_STO1_ASRC_SHIFT, 0, NULL, 0), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* STO ADC */ + SND_SOC_DAPM_PGA("STO1 ADC L1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("STO1 ADC L2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog DAC source */ + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, &rt5668_alg_dacl_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, &rt5668_alg_dacr_mux), +}; + +static const struct snd_soc_dapm_route rt5668_dapm_routes[] = { + /* PLL */ + { "I2S", NULL, "PLL", rt5668_is_sys_clk_from_pll }, + + /* ASRC */ + { "STO1 ADC Filter", NULL, "ADC ASRC", rt5668_is_using_asrc }, + { "STO1 DAC Filter", NULL, "DAC ASRC", rt5668_is_using_asrc }, + { "I2S", NULL, "I2S ASRC", rt5668_i2s_use_asrc }, + + { "ADC L", NULL, "ADC L Power" }, + { "ADC L", NULL, "ADC Clock" }, + + { "STO1 ADC L2", NULL, "STO1 DAC MIXL" }, + + { "STO1 ADC MIXL", "ADC1 Switch", "STO1 ADC L1" }, + { "STO1 ADC MIXL", "ADC2 Switch", "STO1 ADC L2" }, + { "STO1 ADC MIXL", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXL" }, + { "IF ADC", NULL, "IF1 ADC1" }, + { "AIFTX", NULL, "IF ADC" }, + { "AIFTX", NULL, "I2S" }, + + { "AIFRX", NULL, "I2S" }, + { "IF DAC", NULL, "AIFRX" }, + { "IF1 DAC1 L", NULL, "IF DAC" }, + { "IF1 DAC1 R", NULL, "IF DAC" }, + + { "ADDA MIXL", "ADC L Switch", "STO1 ADC MIXL" }, + { "ADDA MIXL", "DAC L Switch", "IF1 DAC1 L" }, + { "ADDA MIXL", NULL, "STO1 DAC Filter" }, + { "ADDA MIXL", NULL, "STO1 DAC L Power" }, + { "ADDA MIXR", "DAC R Switch", "IF1 DAC1 R" }, + { "ADDA MIXR", NULL, "STO1 DAC Filter" }, + { "ADDA MIXR", NULL, "STO1 DAC R Power" }, + + { "DAC L1", NULL, "ADDA MIXL" }, + { "DAC R1", NULL, "ADDA MIXR" }, + + { "STO1 DAC MIXL", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXL", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC L Power" }, + { "STO1 DAC MIXL", NULL, "STO1 DAC Filter" }, + { "STO1 DAC MIXR", "DAC R Switch", "DAC R1" }, + { "STO1 DAC MIXR", "DAC L Switch", "DAC L1" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC R Power" }, + { "STO1 DAC MIXR", NULL, "STO1 DAC Filter" }, + + { "HP Amp", NULL, "DAC L" }, + { "HP Amp", NULL, "DAC R" }, +}; + +static const struct snd_soc_dapm_route rt5668_specific_dapm_routes[] = { + { "MICBIAS1", NULL, "LDO2" }, + { "MICBIAS2", NULL, "LDO2" }, + + { "BST1 CBJ", NULL, "IN1P" }, + { "BST1 CBJ", NULL, "IN1N" }, + { "BST1 CBJ", NULL, "CBJ Power" }, + + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + { "BST2", NULL, "BST2 Power" }, + + { "RECMIX1L", "BST2 Switch", "BST2" }, + { "RECMIX1L", "BST1 CBJ Switch", "BST1 CBJ" }, + { "RECMIX1L", NULL, "RECMIX1L Power" }, + { "RECMIX1R", "BST2 Switch", "BST2" }, + { "RECMIX1R", NULL, "RECMIX1R Power" }, + + { "ADC L", NULL, "RECMIX1L" }, + { "ADC R", NULL, "RECMIX1R" }, + { "ADC R", NULL, "ADC R Power" }, + { "ADC R", NULL, "ADC Clock" }, + + { "STO1 ADC L Mux", "ADC L", "ADC L" }, + { "STO1 ADC L Mux", "ADC R", "ADC R" }, + { "STO1 ADC L1", NULL, "STO1 ADC L Mux" }, + + { "STO1 ADC R Mux", "ADC L", "ADC L" }, + { "STO1 ADC R Mux", "ADC R", "ADC R" }, + { "STO1 ADC R1", NULL, "STO1 ADC R Mux" }, + { "STO1 ADC R2", NULL, "STO1 DAC MIXR" }, + + { "STO1 ADC MIXR", "ADC1 Switch", "STO1 ADC R1" }, + { "STO1 ADC MIXR", "ADC2 Switch", "STO1 ADC R2" }, + { "STO1 ADC MIXR", NULL, "STO1 ADC Filter" }, + + { "IF1 ADC1", NULL, "STO1 ADC MIXR" }, + + { "ADDA MIXR", "ADC R Switch", "STO1 ADC MIXR" }, + + { "DAC L", NULL, "STO1 DAC MIXL" }, + { "DAC L", NULL, "LDO DAC" }, + { "DAC L", NULL, "DAC Clock" }, + { "DAC R", NULL, "STO1 DAC MIXR" }, + { "DAC R", NULL, "LDO DAC" }, + { "DAC R", NULL, "DAC Clock" }, + + { "HPO Playback", "Switch", "HP Amp" }, + { "HPOL", NULL, "HPO Playback" }, + { "HPOR", NULL, "HPO Playback" }, +}; + +static const struct snd_soc_dapm_route rt5663_specific_dapm_routes[] = { + { "I2S", NULL, "Pre Div Power" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "RECMIX1L Power" }, + + { "ADC L", NULL, "BST1" }, + + { "STO1 ADC L1", NULL, "ADC L" }, + + { "DAC L Mux", "DAC L", "DAC L1" }, + { "DAC L Mux", "STO DAC MIXL", "STO1 DAC MIXL" }, + { "DAC R Mux", "DAC R", "DAC R1"}, + { "DAC R Mux", "STO DAC MIXR", "STO1 DAC MIXR" }, + + { "DAC L", NULL, "DAC L Mux" }, + { "DAC R", NULL, "DAC R Mux" }, + + { "HPOL", NULL, "HP Amp" }, + { "HPOR", NULL, "HP Amp" }, +}; + +static int rt5668_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 rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, frame_size; + + rt5668->lrck = params_rate(params); + pre_div = rl6231_get_clk_info(rt5668->sysclk, rt5668->lrck); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting %d for DAI %d\n", + rt5668->lrck, dai->id); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5668->lrck * (32 << (frame_size > 32) ? 1 : 0), rt5668->lrck); + dev_dbg(dai->dev, "pre_div is %d for iis %d\n", pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5668_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5668_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5668_I2S_DL_8; + break; + default: + return -EINVAL; + } + + mask_clk = RT5668_I2S_PD1_MASK; + val_clk = pre_div << RT5668_I2S_PD1_SHIFT; + snd_soc_update_bits(codec, RT5668_I2S1_SDP, + RT5668_I2S_DL_MASK, val_len); + + snd_soc_update_bits(codec, RT5668_ADDA_CLK_1, mask_clk, val_clk); + + return 0; +} + +static int rt5668_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5668_I2S_MS_S; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5668_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5668_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5668_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5668_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, RT5668_I2S1_SDP, RT5668_I2S_MS_MASK | + RT5668_I2S_BP_MASK | RT5668_I2S_DF_MASK, reg_val); + + return 0; +} + +static int rt5668_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5668->sysclk && clk_id == rt5668->sysclk_src) + return 0; + + switch (clk_id) { + case RT5668_SCLK_S_MCLK: + reg_val |= RT5668_SCLK_SRC_MCLK; + break; + case RT5668_SCLK_S_PLL1: + reg_val |= RT5668_SCLK_SRC_PLL1; + break; + case RT5668_SCLK_S_RCCLK: + reg_val |= RT5668_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5668_GLB_CLK, RT5668_SCLK_SRC_MASK, + reg_val); + rt5668->sysclk = freq; + rt5668->sysclk_src = clk_id; + + dev_dbg(codec->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5668_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 rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + struct rl6231_pll_code pll_code; + int ret; + int mask, shift, val; + + if (source == rt5668->pll_src && freq_in == rt5668->pll_in && + freq_out == rt5668->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5668->pll_in = 0; + rt5668->pll_out = 0; + snd_soc_update_bits(codec, RT5668_GLB_CLK, + RT5668_SCLK_SRC_MASK, RT5668_SCLK_SRC_MCLK); + return 0; + } + + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + mask = RT5668_PLL1_SRC_MASK; + shift = RT5668_PLL1_SRC_SHIFT; + } else { + mask = RT5663_PLL1_SRC_MASK; + shift = RT5663_PLL1_SRC_SHIFT; + } + + switch (source) { + case RT5668_PLL1_S_MCLK: + val = 0x0; + break; + case RT5668_PLL1_S_BCLK1: + val = 0x1; + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5668_GLB_CLK, mask, (val << shift)); + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", pll_code.m_bp, + (pll_code.m_bp ? 0 : pll_code.m_code), pll_code.n_code, + pll_code.k_code); + + snd_soc_write(codec, RT5668_PLL_1, + pll_code.n_code << RT5668_PLL_N_SHIFT | pll_code.k_code); + snd_soc_write(codec, RT5668_PLL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5668_PLL_M_SHIFT | + pll_code.m_bp << RT5668_PLL_M_BP_SHIFT); + + rt5668->pll_in = freq_in; + rt5668->pll_out = freq_out; + rt5668->pll_src = source; + + return 0; +} + +static int rt5668_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 rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0, reg; + + if (rx_mask || tx_mask) + val |= RT5668_TDM_MODE_TDM; + + switch (slots) { + case 4: + val |= RT5668_TDM_IN_CH_4; + val |= RT5668_TDM_OUT_CH_4; + break; + case 6: + val |= RT5668_TDM_IN_CH_6; + val |= RT5668_TDM_OUT_CH_6; + break; + case 8: + val |= RT5668_TDM_IN_CH_8; + val |= RT5668_TDM_OUT_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + switch (slot_width) { + case 20: + val |= RT5668_TDM_IN_LEN_20; + val |= RT5668_TDM_OUT_LEN_20; + break; + case 24: + val |= RT5668_TDM_IN_LEN_24; + val |= RT5668_TDM_OUT_LEN_24; + break; + case 32: + val |= RT5668_TDM_IN_LEN_32; + val |= RT5668_TDM_OUT_LEN_32; + break; + case 16: + break; + default: + return -EINVAL; + } + + if (rt5668->codec_type == CODEC_TYPE_RT5668) + reg = RT5668_TDM_1; + else + reg = RT5663_TDM_1; + + snd_soc_update_bits(codec, reg, RT5668_TDM_MODE_MASK | + RT5668_TDM_IN_CH_MASK | RT5668_TDM_OUT_CH_MASK | + RT5668_TDM_IN_LEN_MASK | RT5668_TDM_OUT_LEN_MASK, val); + + return 0; +} + +static int rt5668_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "%s ratio = %d\n", __func__, ratio); + + if (rt5668->codec_type == CODEC_TYPE_RT5663) { + switch (ratio) { + case 32: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_16); + break; + case 40: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_20); + break; + case 48: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_24); + break; + case 64: + snd_soc_update_bits(codec, RT5663_TDM_5, + RT5663_TDM_LENGTN_MASK, + RT5663_TDM_LENGTN_32); + break; + default: + dev_err(codec->dev, "Invalid ratio!\n"); + return -EINVAL; + } + } + + return 0; +} + +static int rt5668_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_1, + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK, + RT5668_PWR_FV1 | RT5668_PWR_FV2); + break; + + case SND_SOC_BIAS_PREPARE: + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_EN); + snd_soc_update_bits(codec, RT5668_MCLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_AUTO | + RT5668_PWR_CLK_DET_EN); + } + break; + + case SND_SOC_BIAS_STANDBY: + if (rt5668->codec_type == CODEC_TYPE_RT5668) + snd_soc_update_bits(codec, RT5668_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, + RT5668_DIG_GATE_CTRL_DIS); + snd_soc_update_bits(codec, RT5668_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1_MASK | RT5668_PWR_FV2_MASK | + RT5668_PWR_MB_MASK, RT5668_PWR_VREF1 | + RT5668_PWR_VREF2 | RT5668_PWR_MB); + usleep_range(10000, 10005); + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + snd_soc_update_bits(codec, RT5668_MCLK_DET, + RT5668_EN_ANA_CLK_DET_MASK | + RT5668_PWR_CLK_DET_MASK, + RT5668_EN_ANA_CLK_DET_DIS | + RT5668_PWR_CLK_DET_DIS); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, RT5668_PWR_ANLG_1, + RT5668_PWR_VREF1_MASK | RT5668_PWR_VREF2_MASK | + RT5668_PWR_FV1 | RT5668_PWR_FV2, 0x0); + break; + + default: + break; + } + + return 0; +} + +static int rt5668_probe(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + rt5668->codec = codec; + + switch (rt5668->codec_type) { + case CODEC_TYPE_RT5668: + snd_soc_dapm_new_controls(dapm, + rt5668_specific_dapm_widgets, + ARRAY_SIZE(rt5668_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5668_specific_dapm_routes, + ARRAY_SIZE(rt5668_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5668_specific_controls, + ARRAY_SIZE(rt5668_specific_controls)); + break; + case CODEC_TYPE_RT5663: + snd_soc_dapm_new_controls(dapm, + rt5663_specific_dapm_widgets, + ARRAY_SIZE(rt5663_specific_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, + rt5663_specific_dapm_routes, + ARRAY_SIZE(rt5663_specific_dapm_routes)); + snd_soc_add_codec_controls(codec, rt5663_specific_controls, + ARRAY_SIZE(rt5663_specific_controls)); + break; + } + + return 0; +} + +static int rt5668_remove(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + regmap_write(rt5668->regmap, RT5668_RESET, 0); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5668_suspend(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5668->regmap, true); + regcache_mark_dirty(rt5668->regmap); + + return 0; +} + +static int rt5668_resume(struct snd_soc_codec *codec) +{ + struct rt5668_priv *rt5668 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5668->regmap, false); + regcache_sync(rt5668->regmap); + + return 0; +} +#else +#define rt5668_suspend NULL +#define rt5668_resume NULL +#endif + +#define RT5668_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5668_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5668_aif_dai_ops = { + .hw_params = rt5668_hw_params, + .set_fmt = rt5668_set_dai_fmt, + .set_sysclk = rt5668_set_dai_sysclk, + .set_pll = rt5668_set_dai_pll, + .set_tdm_slot = rt5668_set_tdm_slot, + .set_bclk_ratio = rt5668_set_bclk_ratio, +}; + +struct snd_soc_dai_driver rt5668_dai[] = { + { + .name = "rt5668-aif", + .id = RT5668_AIF, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .capture = { + .stream_name = "AIF Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .ops = &rt5668_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5668 = { + .probe = rt5668_probe, + .remove = rt5668_remove, + .suspend = rt5668_suspend, + .resume = rt5668_resume, + .set_bias_level = rt5668_set_bias_level, + .idle_bias_off = true, + .controls = rt5668_snd_controls, + .num_controls = ARRAY_SIZE(rt5668_snd_controls), + .dapm_widgets = rt5668_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5668_dapm_widgets), + .dapm_routes = rt5668_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5668_dapm_routes), +}; + +static const struct regmap_config rt5668_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x07fa, + .volatile_reg = rt5668_volatile_register, + .readable_reg = rt5668_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5668_reg, + .num_reg_defaults = ARRAY_SIZE(rt5668_reg), +}; + +static const struct regmap_config rt5663_regmap = { + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .volatile_reg = rt5663_volatile_register, + .readable_reg = rt5663_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5663_reg, + .num_reg_defaults = ARRAY_SIZE(rt5663_reg), +}; + +static const struct regmap_config temp_regmap = { + .name = "nocache", + .reg_bits = 16, + .val_bits = 16, + .use_single_rw = true, + .max_register = 0x03f3, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt5668_i2c_id[] = { + { "rt5668", 0 }, + { "rt5663", 0 }, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5668_of_match[] = { + { .compatible = "realtek,rt5668", }, + { .compatible = "realtek,rt5663", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5668_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt5668_acpi_match[] = { + { "10EC5668", 0}, + { "10EC5663", 0}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5668_acpi_match); +#endif + +static void rt5668_calibrate(struct rt5668_priv *rt5668) +{ + regmap_write(rt5668->regmap, 0x0111, 0xa402); + regmap_write(rt5668->regmap, 0x0061, 0x0100); + regmap_write(rt5668->regmap, 0x0010, 0x4040); + regmap_write(rt5668->regmap, 0x00fa, 0x0001); + regmap_write(rt5668->regmap, 0x0094, 0x0380); + regmap_write(rt5668->regmap, 0x0080, 0x8000); + regmap_write(rt5668->regmap, 0x0073, 0x1000); + regmap_write(rt5668->regmap, 0x013a, 0x3030); + regmap_write(rt5668->regmap, 0x013c, 0x3c05); + regmap_write(rt5668->regmap, 0x0063, 0xa23e); + msleep(40); + regmap_write(rt5668->regmap, 0x0063, 0xf23e); + regmap_write(rt5668->regmap, 0x01df, 0x0321); + regmap_write(rt5668->regmap, 0x01de, 0xfc00); + msleep(500); +} + +static void rt5663_calibrate(struct rt5668_priv *rt5668) +{ + regmap_write(rt5668->regmap, 0x00fa, 0x8001); + regmap_write(rt5668->regmap, 0x0063, 0xa2b8); + msleep(50); + regmap_write(rt5668->regmap, 0x0062, 0x8400); + regmap_write(rt5668->regmap, 0x0061, 0x8df8); + regmap_write(rt5668->regmap, 0x0064, 0x0003); + regmap_write(rt5668->regmap, 0x0065, 0x018c); + regmap_write(rt5668->regmap, 0x0073, 0x0000); + regmap_write(rt5668->regmap, 0x006e, 0xffff); + regmap_write(rt5668->regmap, 0x006f, 0xffff); + regmap_write(rt5668->regmap, 0x008f, 0x3003); + regmap_write(rt5668->regmap, 0x008e, 0x003b); + regmap_write(rt5668->regmap, 0x0091, 0x1e32); + regmap_write(rt5668->regmap, 0x0092, 0x1371); + regmap_write(rt5668->regmap, 0x0112, 0x3b0b); + regmap_write(rt5668->regmap, 0x002a, 0x2080); + regmap_write(rt5668->regmap, 0x002d, 0x000c); + regmap_write(rt5668->regmap, 0x0202, 0xabba); + regmap_write(rt5668->regmap, 0x0125, 0x2224); + regmap_write(rt5668->regmap, 0x0002, 0x8088); + regmap_write(rt5668->regmap, 0x0168, 0x0017); + regmap_write(rt5668->regmap, 0x0169, 0x0017); + regmap_write(rt5668->regmap, 0x0026, 0x4040); + regmap_write(rt5668->regmap, 0x013b, 0x3000); + regmap_write(rt5668->regmap, 0x0010, 0x0005); + regmap_write(rt5668->regmap, 0x0074, 0xc000); + regmap_write(rt5668->regmap, 0x00ec, 0x3320); + regmap_write(rt5668->regmap, 0x01df, 0x00c9); + regmap_write(rt5668->regmap, 0x02fa, 0x004c); + regmap_write(rt5668->regmap, 0x0108, 0x7766); + regmap_write(rt5668->regmap, 0x0111, 0x4702); + regmap_write(rt5668->regmap, 0x0063, 0xf2b8); + regmap_write(rt5668->regmap, 0x0118, 0x0007); + msleep(200); + regmap_write(rt5668->regmap, 0x01dd, 0x0069); + regmap_write(rt5668->regmap, 0x01e0, 0x06c2); + regmap_write(rt5668->regmap, 0x01de, 0x7b00); + regmap_write(rt5668->regmap, 0x01de, 0xfb00); + msleep(500); +} + +static int rt5668_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5668_priv *rt5668; + int ret; + unsigned int val; + struct regmap *regmap; + + rt5668 = devm_kzalloc(&i2c->dev, sizeof(struct rt5668_priv), + GFP_KERNEL); + + if (rt5668 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5668); + + regmap = devm_regmap_init_i2c(i2c, &temp_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c->dev, "Failed to allocate temp register map: %d\n", + ret); + return ret; + } + regmap_read(regmap, RT5668_VENDOR_ID_2, &val); + switch (val) { + case RT5668_DEVICE_ID: + rt5668->regmap = devm_regmap_init_i2c(i2c, &rt5668_regmap); + rt5668->codec_type = CODEC_TYPE_RT5668; + break; + case RT5663_DEVICE_ID: + rt5668->regmap = devm_regmap_init_i2c(i2c, &rt5663_regmap); + rt5668->codec_type = CODEC_TYPE_RT5663; + break; + default: + dev_err(&i2c->dev, + "Device with ID register %#x is not rt5663 or rt5668\n", + val); + return -ENODEV; + } + + if (IS_ERR(rt5668->regmap)) { + ret = PTR_ERR(rt5668->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* reset and calibrate */ + regmap_write(rt5668->regmap, RT5668_RESET, 0); + regcache_cache_bypass(rt5668->regmap, true); + if (rt5668->codec_type == CODEC_TYPE_RT5668) + rt5668_calibrate(rt5668); + else + rt5663_calibrate(rt5668); + regcache_cache_bypass(rt5668->regmap, false); + regmap_write(rt5668->regmap, RT5668_RESET, 0); + dev_dbg(&i2c->dev, "calibrate done\n"); + + /* GPIO1 as IRQ */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_1, RT5668_GP1_PIN_MASK, + RT5668_GP1_PIN_IRQ); + /* 4btn inline command debounce */ + regmap_update_bits(rt5668->regmap, RT5668_4BTN_IL_CMD_1, + RT5668_4BTN_CLK_DEB_MASK, RT5668_4BTN_CLK_DEB_65MS); + + if (rt5668->codec_type == CODEC_TYPE_RT5668) { + regmap_write(rt5668->regmap, RT5668_BIAS_CUR_8, 0xa402); + /* JD1 */ + regmap_update_bits(rt5668->regmap, RT5668_AUTO_RC_CLK, + RT5668_IRQ_POW_SAV_MASK | RT5668_IRQ_POW_SAV_JD1_MASK, + RT5668_IRQ_POW_SAV_EN | RT5668_IRQ_POW_SAV_JD1_EN); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_2, + RT5668_PWR_JD1_MASK, RT5668_PWR_JD1); + regmap_update_bits(rt5668->regmap, RT5668_IRQ_1, + RT5668_EN_CB_JD_MASK, RT5668_EN_CB_JD_EN); + + regmap_update_bits(rt5668->regmap, RT5668_HP_LOGIC_2, + RT5668_HP_SIG_SRC1_MASK, RT5668_HP_SIG_SRC1_REG); + regmap_update_bits(rt5668->regmap, RT5668_CBJ_TYPE_1, + RT5668_VREF_BIAS_MASK | RT5668_CBJ_DET_MASK | + RT5668_DET_TYPE_MASK, RT5668_VREF_BIAS_REG | + RT5668_CBJ_DET_EN | RT5668_DET_TYPE_QFN); + /* Set GPIO4 and GPIO8 as input for combo jack */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_2, + RT5668_GP4_PIN_CONF_MASK, RT5668_GP4_PIN_CONF_INPUT); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_3, + RT5668_GP8_PIN_CONF_MASK, RT5668_GP8_PIN_CONF_INPUT); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_LDO1_DVO_MASK | RT5668_AMP_HP_MASK, + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + } else { + regmap_update_bits(rt5668->regmap, RT5668_DIG_MISC, + RT5668_DIG_GATE_CTRL_MASK, RT5668_DIG_GATE_CTRL_EN); + regmap_update_bits(rt5668->regmap, RT5668_HP_CHARGE_PUMP_1, + RT5663_SI_HP_MASK | RT5668_OSW_HP_L_MASK | + RT5668_OSW_HP_R_MASK, RT5663_SI_HP_EN | + RT5668_OSW_HP_L_DIS | RT5668_OSW_HP_R_DIS); + regmap_update_bits(rt5668->regmap, RT5663_DUMMY_1, + RT5663_EMB_CLK_MASK | RT5663_HPA_CPL_BIAS_MASK | + RT5663_HPA_CPR_BIAS_MASK, RT5663_EMB_CLK_EN | + RT5663_HPA_CPL_BIAS_1 | RT5663_HPA_CPR_BIAS_1); + regmap_update_bits(rt5668->regmap, RT5663_CBJ_1, + RT5663_INBUF_CBJ_BST1_MASK | RT5663_CBJ_SENSE_BST1_MASK, + RT5663_INBUF_CBJ_BST1_ON | RT5663_CBJ_SENSE_BST1_L); + regmap_update_bits(rt5668->regmap, RT5668_IL_CMD_2, + RT5663_PWR_MIC_DET_MASK, RT5663_PWR_MIC_DET_ON); + /* BST1 power on for JD */ + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_2, + RT5668_PWR_BST1_MASK, RT5668_PWR_BST1_ON); + regmap_update_bits(rt5668->regmap, RT5663_EM_JACK_TYPE_1, + RT5663_CBJ_DET_MASK | RT5663_EXT_JD_MASK | + RT5663_POL_EXT_JD_MASK, RT5663_CBJ_DET_EN | + RT5663_EXT_JD_EN | RT5663_POL_EXT_JD_EN); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB_MASK | RT5668_LDO1_DVO_MASK | + RT5668_AMP_HP_MASK, RT5668_PWR_MB | + RT5668_LDO1_DVO_0_9V | RT5668_AMP_HP_3X); + regmap_update_bits(rt5668->regmap, RT5668_AUTO_RC_CLK, + RT5668_IRQ_POW_SAV_MASK, RT5668_IRQ_POW_SAV_EN); + regmap_update_bits(rt5668->regmap, RT5668_IRQ_1, + RT5663_EN_IRQ_JD1_MASK, RT5663_EN_IRQ_JD1_EN); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_2, + RT5663_GP1_PIN_CONF_MASK, RT5663_GP1_PIN_CONF_OUTPUT); + /* DACREF LDO control */ + regmap_update_bits(rt5668->regmap, RT5663_DACREF_LDO, 0x3e0e, + 0x3a0a); + /* RECMIX */ + regmap_update_bits(rt5668->regmap, RT5663_RECMIX, + RT5663_RECMIX1_BST1_MASK, RT5663_RECMIX1_BST1_UNMUTE); + /* ADCDAT L/L */ + regmap_update_bits(rt5668->regmap, RT5668_TDM_1, + RT5663_DATA_SWAP_ADCDAT1_MASK, + RT5663_DATA_SWAP_ADCDAT1_LL); + } + + setup_timer(&rt5668->btn_check_timer, rt5668_btn_check_callback, + (unsigned long)rt5668); + INIT_DELAYED_WORK(&rt5668->jack_detect_work, rt5668_jack_detect_work); + + if (i2c->irq) { + ret = request_threaded_irq(i2c->irq, NULL, rt5668_irq, + IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5668", rt5668); + if (ret) + dev_err(&i2c->dev, "%s Failed to reguest IRQ: %d\n", + __func__, ret); + } + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5668, + rt5668_dai, ARRAY_SIZE(rt5668_dai)); + + if (ret) { + if (i2c->irq) + free_irq(i2c->irq, rt5668); + } + + return ret; +} + +static int rt5668_i2c_remove(struct i2c_client *i2c) +{ + struct rt5668_priv *rt5668 = i2c_get_clientdata(i2c); + + if (i2c->irq) + free_irq(i2c->irq, rt5668); + + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +void rt5668_i2c_shutdown(struct i2c_client *client) +{ + struct rt5668_priv *rt5668 = i2c_get_clientdata(client); + + regmap_write(rt5668->regmap, RT5668_RESET, 0); +} + +struct i2c_driver rt5668_i2c_driver = { + .driver = { + .name = "rt5668", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5668_acpi_match), + .of_match_table = of_match_ptr(rt5668_of_match), + }, + .probe = rt5668_i2c_probe, + .remove = rt5668_i2c_remove, + .shutdown = rt5668_i2c_shutdown, + .id_table = rt5668_i2c_id, +}; +module_i2c_driver(rt5668_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5668 driver"); +MODULE_AUTHOR("John Lin <john.lin(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5668.h b/sound/soc/codecs/rt5668.h new file mode 100644 index 0000000..717d5af --- /dev/null +++ b/sound/soc/codecs/rt5668.h @@ -0,0 +1,781 @@ +/* + * rt5668.h -- RT5668 ALSA SoC audio driver + * + * Copyright 2015 Realtek Microelectronics + * Author: Jack Yu <jack.yu(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 __RT5668_H__ +#define __RT5668_H__ + +/* Info */ +#define RT5668_RESET 0x0000 +#define RT5668_VENDOR_ID 0x00fd +#define RT5668_VENDOR_ID_1 0x00fe +#define RT5668_VENDOR_ID_2 0x00ff + +#define RT5668_HP_AMP_1 0x0002 +#define RT5668_HP_AMP_2 0x0003 +#define RT5668_HPL_GAIN 0x0005 +#define RT5668_HPR_GAIN 0x0006 + +#define RT5668_AEC_BST 0x000b +#define RT5668_IN1_IN2 0x000c +#define RT5668_CBJ_TYPE_1 0x0010 +#define RT5668_CBJ_TYPE_2 0x0011 +#define RT5668_CBJ_TYPE_3 0x0012 + +/* I/O - ADC/DAC/DMIC */ +#define RT5668_DAC1_DIG_VOL 0x0019 +#define RT5668_STO1_ADC_DIG_VOL 0x001c +#define RT5668_STO1_BOOST 0x001f +#define RT5668_HP_IMP_GAIN_1 0x0022 +#define RT5668_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5668_STO1_ADC_MIXER 0x0026 +#define RT5668_AD_DA_MIXER 0x0029 +#define RT5668_STO_DAC_MIXER 0x002a + +/*Mixer - Analog*/ +#define RT5668_RECMIX1L 0x003c +#define RT5668_RECMIX1R 0x003e + +/* Power */ +#define RT5668_PWR_DIG_1 0x0061 +#define RT5668_PWR_DIG_2 0x0062 +#define RT5668_PWR_ANLG_1 0x0063 +#define RT5668_PWR_ANLG_2 0x0064 +#define RT5668_PWR_ANLG_3 0x0065 +#define RT5668_PWR_MIXER 0x0066 +#define RT5668_PWR_VOL 0x0067 + +#define RT5668_MCLK_DET 0x006b +/* Format - ADC/DAC */ +#define RT5668_I2S1_SDP 0x0070 +#define RT5668_ADDA_CLK_1 0x0073 +/* Format - TDM Control */ +#define RT5668_TDM_1 0x0078 +#define RT5668_TDM_2 0x0079 +#define RT5668_TDM_3 0x007a +#define RT5668_TDM_4 0x007b +#define RT5668_TDM_5 0x007c +#define RT5668_TDM_6 0x007d +#define RT5668_TDM_7 0x007e +#define RT5668_TDM_8 0x007f + +/* Function - Analog */ +#define RT5668_GLB_CLK 0x0080 +#define RT5668_PLL_1 0x0081 +#define RT5668_PLL_2 0x0082 +#define RT5668_ASRC_1 0x0083 +#define RT5668_ASRC_2 0x0084 +#define RT5668_ASRC_3 0x0085 +#define RT5668_ASRC_8 0x008a +#define RT5668_ASRC_9 0x008b +#define RT5668_ASRC_11 0x008c +#define RT5668_DEPOP_1 0x008e +#define RT5668_DEPOP_2 0x008f +#define RT5668_HP_CHARGE_PUMP_1 0x0091 +#define RT5668_HP_CHARGE_PUMP_2 0x0092 +#define RT5668_MICBIAS_2 0x0094 +#define RT5668_AUTO_RC_CLK 0x009f + +/* Function - Digital */ +#define RT5668_ADC_EQ_1 0x00ae +#define RT5668_ADC_EQ_2 0x00af + +#define RT5668_IRQ_1 0x00b6 +#define RT5668_IRQ_2 0x00b7 +#define RT5668_IRQ_3 0x00b8 +#define RT5668_IRQ_4 0x00b9 +#define RT5668_IRQ_5 0x00ba +#define RT5668_IRQ_6 0x00bb +#define RT5668_INT_ST_1 0x00be +#define RT5668_GPIO_1 0x00c0 +#define RT5668_GPIO_2 0x00c1 +#define RT5668_GPIO_3 0x00c2 +#define RT5668_IL_CMD_1 0x00db +#define RT5668_IL_CMD_2 0x00dc +#define RT5668_IL_CMD_3 0x00dd +#define RT5668_IL_CMD_4 0x00de +#define RT5668_4BTN_IL_CMD_1 0x00df +#define RT5668_4BTN_IL_CMD_2 0x00e0 +#define RT5668_4BTN_IL_CMD_3 0x00e1 +/* General Control */ +#define RT5668_DIG_MISC 0x00fa + +#define RT5668_BIAS_CUR_8 0x0111 +#define RT5668_CHOP_DAC 0x013a +#define RT5668_CHOP_ADC 0x013b +#define RT5668_CALIB_ADC 0x013c +#define RT5668_HP_LOGIC_1 0x01da +#define RT5668_HP_LOGIC_2 0x01db +#define RT5668_HP_CALIB_1 0x01de +#define RT5668_HP_CALIB_2 0x01df +#define RT5668_HP_CALIB_3 0x01e0 +#define RT5668_HP_CALIB_4 0x01e1 +#define RT5668_HP_CALIB_5 0x01e2 +#define RT5668_HP_CALIB_6 0x01e3 +#define RT5668_HP_CALIB_7 0x01e4 +#define RT5668_HP_CALIB_8 0x01e5 +#define RT5668_HP_CALIB_9 0x01e6 +#define RT5668_HP_CALIB_10 0x01e7 + +/* Headphone Amp Control 2 (0x0003) */ +#define RT5668_EN_DAC_HPO_MASK (0x1 << 14) +#define RT5668_EN_DAC_HPO_SHIFT 14 +#define RT5668_EN_DAC_HPO_DIS (0x0 << 14) +#define RT5668_EN_DAC_HPO_EN (0x1 << 14) + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5668_GAIN_HP (0x1f << 8) +#define RT5668_GAIN_HP_SHIFT 8 + +/* AEC BST Control (0x000b) */ +#define RT5668_GAIN_CBJ_MASK (0xf << 8) +#define RT5668_GAIN_CBJ_SHIFT 8 + +/* IN1 Control / MIC GND REF (0x000c) */ +#define RT5668_IN1_DF_MASK (0x1 << 15) +#define RT5668_IN1_DF_SHIFT 15 + +/* Combo Jack and Type Detection Control 1 (0x0010) */ +#define RT5668_CBJ_DET_MASK (0x1 << 15) +#define RT5668_CBJ_DET_SHIFT 15 +#define RT5668_CBJ_DET_DIS (0x0 << 15) +#define RT5668_CBJ_DET_EN (0x1 << 15) +#define RT5668_DET_TYPE_MASK (0x1 << 12) +#define RT5668_DET_TYPE_SHIFT 12 +#define RT5668_DET_TYPE_WLCSP (0x0 << 12) +#define RT5668_DET_TYPE_QFN (0x1 << 12) +#define RT5668_VREF_BIAS_MASK (0x1 << 6) +#define RT5668_VREF_BIAS_SHIFT 6 +#define RT5668_VREF_BIAS_FSM (0x0 << 6) +#define RT5668_VREF_BIAS_REG (0x1 << 6) + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5668_RECMIX1L_BST1_CBJ (0x1 << 7) +#define RT5668_RECMIX1L_BST1_CBJ_SHIFT 7 +#define RT5668_RECMIX1L_BST2 (0x1 << 4) +#define RT5668_RECMIX1L_BST2_SHIFT 4 + +/* REC Right Mixer Control 2 (0x003e) */ +#define RT5668_RECMIX1R_BST2 (0x1 << 4) +#define RT5668_RECMIX1R_BST2_SHIFT 4 + +/* DAC1 Digital Volume (0x0019) */ +#define RT5668_DAC_L1_VOL_MASK (0xff << 8) +#define RT5668_DAC_L1_VOL_SHIFT 8 +#define RT5668_DAC_R1_VOL_MASK (0xff) +#define RT5668_DAC_R1_VOL_SHIFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5668_ADC_L_MUTE_MASK (0x1 << 15) +#define RT5668_ADC_L_MUTE_SHIFT 15 +#define RT5668_ADC_L_VOL_MASK (0x7f << 8) +#define RT5668_ADC_L_VOL_SHIFT 8 +#define RT5668_ADC_R_MUTE_MASK (0x1 << 7) +#define RT5668_ADC_R_MUTE_SHIFT 7 +#define RT5668_ADC_R_VOL_MASK (0x7f) +#define RT5668_ADC_R_VOL_SHIFT 0 + +/* Stereo ADC Mixer Control (0x0026) */ +#define RT5668_M_STO1_ADC_L1 (0x1 << 15) +#define RT5668_M_STO1_ADC_L1_SHIFT 15 +#define RT5668_M_STO1_ADC_L2 (0x1 << 14) +#define RT5668_M_STO1_ADC_L2_SHIFT 14 +#define RT5668_STO1_ADC_L1_SRC (0x1 << 13) +#define RT5668_STO1_ADC_L1_SRC_SHIFT 13 +#define RT5668_STO1_ADC_L2_SRC (0x1 << 12) +#define RT5668_STO1_ADC_L2_SRC_SHIFT 12 +#define RT5668_STO1_ADC_L_SRC (0x3 << 10) +#define RT5668_STO1_ADC_L_SRC_SHIFT 10 +#define RT5668_M_STO1_ADC_R1 (0x1 << 7) +#define RT5668_M_STO1_ADC_R1_SHIFT 7 +#define RT5668_M_STO1_ADC_R2 (0x1 << 6) +#define RT5668_M_STO1_ADC_R2_SHIFT 6 +#define RT5668_STO1_ADC_R1_SRC (0x1 << 5) +#define RT5668_STO1_ADC_R1_SRC_SHIFT 5 +#define RT5668_STO1_ADC_R2_SRC (0x1 << 4) +#define RT5668_STO1_ADC_R2_SRC_SHIFT 4 +#define RT5668_STO1_ADC_R_SRC (0x3 << 2) +#define RT5668_STO1_ADC_R_SRC_SHIFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5668_M_ADCMIX_L (0x1 << 15) +#define RT5668_M_ADCMIX_L_SHIFT 15 +#define RT5668_M_DAC1_L (0x1 << 14) +#define RT5668_M_DAC1_L_SHIFT 14 +#define RT5668_M_ADCMIX_R (0x1 << 7) +#define RT5668_M_ADCMIX_R_SHIFT 7 +#define RT5668_M_DAC1_R (0x1 << 6) +#define RT5668_M_DAC1_R_SHIFT 6 + +/* Stereo DAC Mixer Control (0x002a) */ +#define RT5668_M_DAC_L1_STO_L (0x1 << 15) +#define RT5668_M_DAC_L1_STO_L_SHIFT 15 +#define RT5668_M_DAC_R1_STO_L (0x1 << 13) +#define RT5668_M_DAC_R1_STO_L_SHIFT 13 +#define RT5668_M_DAC_L1_STO_R (0x1 << 7) +#define RT5668_M_DAC_L1_STO_R_SHIFT 7 +#define RT5668_M_DAC_R1_STO_R (0x1 << 5) +#define RT5668_M_DAC_R1_STO_R_SHIFT 5 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5668_PWR_I2S1 (0x1 << 15) +#define RT5668_PWR_I2S1_SHIFT 15 +#define RT5668_PWR_DAC_L1 (0x1 << 11) +#define RT5668_PWR_DAC_L1_SHIFT 11 +#define RT5668_PWR_DAC_R1 (0x1 << 10) +#define RT5668_PWR_DAC_R1_SHIFT 10 +#define RT5668_PWR_LDO_DACREF_MASK (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_SHIFT 8 +#define RT5668_PWR_LDO_DACREF_ON (0x1 << 8) +#define RT5668_PWR_LDO_DACREF_DOWN (0x0 << 8) +#define RT5668_PWR_LDO_SHIFT 8 +#define RT5668_PWR_ADC_L1 (0x1 << 4) +#define RT5668_PWR_ADC_L1_SHIFT 4 +#define RT5668_PWR_ADC_R1 (0x1 << 3) +#define RT5668_PWR_ADC_R1_SHIFT 3 + +/* Power Management for Digital 2 (0x0062) */ +#define RT5668_PWR_ADC_S1F (0x1 << 15) +#define RT5668_PWR_ADC_S1F_SHIFT 15 +#define RT5668_PWR_DAC_S1F (0x1 << 10) +#define RT5668_PWR_DAC_S1F_SHIFT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5668_PWR_VREF1 (0x1 << 15) +#define RT5668_PWR_VREF1_MASK (0x1 << 15) +#define RT5668_PWR_VREF1_SHIFT 15 +#define RT5668_PWR_FV1 (0x1 << 14) +#define RT5668_PWR_FV1_MASK (0x1 << 14) +#define RT5668_PWR_FV1_SHIFT 14 +#define RT5668_PWR_VREF2 (0x1 << 13) +#define RT5668_PWR_VREF2_MASK (0x1 << 13) +#define RT5668_PWR_VREF2_SHIFT 13 +#define RT5668_PWR_FV2 (0x1 << 12) +#define RT5668_PWR_FV2_MASK (0x1 << 12) +#define RT5668_PWR_FV2_SHIFT 12 +#define RT5668_PWR_MB (0x1 << 9) +#define RT5668_PWR_MB_MASK (0x1 << 9) +#define RT5668_PWR_MB_SHIFT 9 +#define RT5668_AMP_HP_MASK (0x3 << 2) +#define RT5668_AMP_HP_SHIFT 2 +#define RT5668_AMP_HP_1X (0x0 << 2) +#define RT5668_AMP_HP_3X (0x1 << 2) +#define RT5668_AMP_HP_5X (0x3 << 2) +#define RT5668_LDO1_DVO_MASK (0x3) +#define RT5668_LDO1_DVO_SHIFT 0 +#define RT5668_LDO1_DVO_0_9V (0x0) +#define RT5668_LDO1_DVO_1_0V (0x1) +#define RT5668_LDO1_DVO_1_2V (0x2) +#define RT5668_LDO1_DVO_1_4V (0x3) + +/* Power Management for Analog 2 (0x0064) */ +#define RT5668_PWR_BST1 (0x1 << 15) +#define RT5668_PWR_BST1_MASK (0x1 << 15) +#define RT5668_PWR_BST1_SHIFT 15 +#define RT5668_PWR_BST1_OFF (0x0 << 15) +#define RT5668_PWR_BST1_ON (0x1 << 15) +#define RT5668_PWR_BST2 (0x1 << 14) +#define RT5668_PWR_BST2_MASK (0x1 << 14) +#define RT5668_PWR_BST2_SHIFT 14 +#define RT5668_PWR_MB1 (0x1 << 11) +#define RT5668_PWR_MB1_SHIFT 11 +#define RT5668_PWR_MB2 (0x1 << 10) +#define RT5668_PWR_MB2_SHIFT 10 +#define RT5668_PWR_BST2_OP (0x1 << 6) +#define RT5668_PWR_BST2_OP_MASK (0x1 << 6) +#define RT5668_PWR_BST2_OP_SHIFT 6 +#define RT5668_PWR_JD1 (0x1 << 3) +#define RT5668_PWR_JD1_MASK (0x1 << 3) +#define RT5668_PWR_JD1_SHIFT 3 +#define RT5668_PWR_JD2 (0x1 << 2) +#define RT5668_PWR_JD2_MASK (0x1 << 2) +#define RT5668_PWR_JD2_SHIFT 2 +#define RT5668_PWR_RECMIX1 (0x1 << 1) +#define RT5668_PWR_RECMIX1_SHIFT 1 +#define RT5668_PWR_RECMIX2 (0x1) +#define RT5668_PWR_RECMIX2_SHIFT 0 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5668_PWR_CBJ_MASK (0x1 << 9) +#define RT5668_PWR_CBJ_SHIFT 9 +#define RT5668_PWR_CBJ_OFF (0x0 << 9) +#define RT5668_PWR_CBJ_ON (0x1 << 9) +#define RT5668_PWR_PLL (0x1 << 6) +#define RT5668_PWR_PLL_SHIFT 6 +#define RT5668_PWR_LDO2 (0x1 << 2) +#define RT5668_PWR_LDO2_SHIFT 2 + +/* Power Management for Volume (0x0067) */ +#define RT5668_PWR_MIC_DET (0x1 << 5) +#define RT5668_PWR_MIC_DET_SHIFT 5 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5668_EN_ANA_CLK_DET_MASK (0x1 << 15) +#define RT5668_EN_ANA_CLK_DET_SHIFT 15 +#define RT5668_EN_ANA_CLK_DET_DIS (0x0 << 15) +#define RT5668_EN_ANA_CLK_DET_AUTO (0x1 << 15) +#define RT5668_PWR_CLK_DET_MASK (0x1) +#define RT5668_PWR_CLK_DET_SHIFT 0 +#define RT5668_PWR_CLK_DET_DIS (0x0) +#define RT5668_PWR_CLK_DET_EN (0x1) + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5668_I2S_MS_MASK (0x1 << 15) +#define RT5668_I2S_MS_SHIFT 15 +#define RT5668_I2S_MS_M (0x0 << 15) +#define RT5668_I2S_MS_S (0x1 << 15) +#define RT5668_I2S_BP_MASK (0x1 << 8) +#define RT5668_I2S_BP_SHIFT 8 +#define RT5668_I2S_BP_NOR (0x0 << 8) +#define RT5668_I2S_BP_INV (0x1 << 8) +#define RT5668_I2S_DL_MASK (0x3 << 4) +#define RT5668_I2S_DL_SHIFT 4 +#define RT5668_I2S_DL_16 (0x0 << 4) +#define RT5668_I2S_DL_20 (0x1 << 4) +#define RT5668_I2S_DL_24 (0x2 << 4) +#define RT5668_I2S_DL_8 (0x3 << 4) +#define RT5668_I2S_DF_MASK (0x7) +#define RT5668_I2S_DF_SHIFT 0 +#define RT5668_I2S_DF_I2S (0x0) +#define RT5668_I2S_DF_LEFT (0x1) +#define RT5668_I2S_DF_PCM_A (0x2) +#define RT5668_I2S_DF_PCM_B (0x3) +#define RT5668_I2S_DF_PCM_A_N (0x6) +#define RT5668_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5668_I2S_PD1_MASK (0x7 << 12) +#define RT5668_I2S_PD1_SHIFT 12 +#define RT5668_M_I2S_DIV_MASK (0x7 << 8) +#define RT5668_M_I2S_DIV_SHIFT 8 +#define RT5668_CLK_SRC_MASK (0x3 << 4) +#define RT5668_CLK_SRC_MCLK (0x0 << 4) +#define RT5668_CLK_SRC_PLL_OUT (0x1 << 4) +#define RT5668_CLK_SRC_DIV (0x2 << 4) +#define RT5668_CLK_SRC_RC (0x3 << 4) +#define RT5668_DAC_OSR_MASK (0x3 << 2) +#define RT5668_DAC_OSR_SHIFT 2 +#define RT5668_DAC_OSR_128 (0x0 << 2) +#define RT5668_DAC_OSR_64 (0x1 << 2) +#define RT5668_DAC_OSR_32 (0x2 << 2) +#define RT5668_ADC_OSR_MASK (0x3) +#define RT5668_ADC_OSR_SHIFT 0 +#define RT5668_ADC_OSR_128 (0x0) +#define RT5668_ADC_OSR_64 (0x1) +#define RT5668_ADC_OSR_32 (0x2) + +/* TDM1 control 1 (0x0078) */ +#define RT5668_TDM_MODE_MASK (0x1 << 15) +#define RT5668_TDM_MODE_SHIFT 15 +#define RT5668_TDM_MODE_I2S (0x0 << 15) +#define RT5668_TDM_MODE_TDM (0x1 << 15) +#define RT5668_TDM_IN_CH_MASK (0x3 << 10) +#define RT5668_TDM_IN_CH_SHIFT 10 +#define RT5668_TDM_IN_CH_2 (0x0 << 10) +#define RT5668_TDM_IN_CH_4 (0x1 << 10) +#define RT5668_TDM_IN_CH_6 (0x2 << 10) +#define RT5668_TDM_IN_CH_8 (0x3 << 10) +#define RT5668_TDM_OUT_CH_MASK (0x3 << 8) +#define RT5668_TDM_OUT_CH_SHIFT 8 +#define RT5668_TDM_OUT_CH_2 (0x0 << 8) +#define RT5668_TDM_OUT_CH_4 (0x1 << 8) +#define RT5668_TDM_OUT_CH_6 (0x2 << 8) +#define RT5668_TDM_OUT_CH_8 (0x3 << 8) +#define RT5668_TDM_IN_LEN_MASK (0x3 << 6) +#define RT5668_TDM_IN_LEN_SHIFT 6 +#define RT5668_TDM_IN_LEN_16 (0x0 << 6) +#define RT5668_TDM_IN_LEN_20 (0x1 << 6) +#define RT5668_TDM_IN_LEN_24 (0x2 << 6) +#define RT5668_TDM_IN_LEN_32 (0x3 << 6) +#define RT5668_TDM_OUT_LEN_MASK (0x3 << 4) +#define RT5668_TDM_OUT_LEN_SHIFT 4 +#define RT5668_TDM_OUT_LEN_16 (0x0 << 4) +#define RT5668_TDM_OUT_LEN_20 (0x1 << 4) +#define RT5668_TDM_OUT_LEN_24 (0x2 << 4) +#define RT5668_TDM_OUT_LEN_32 (0x3 << 4) + +/* Global Clock Control (0x0080) */ +#define RT5668_SCLK_SRC_MASK (0x3 << 14) +#define RT5668_SCLK_SRC_SHIFT 14 +#define RT5668_SCLK_SRC_MCLK (0x0 << 14) +#define RT5668_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5668_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5668_PLL1_SRC_MASK (0x7 << 8) +#define RT5668_PLL1_SRC_SHIFT 8 +#define RT5668_PLL1_SRC_MCLK (0x0 << 8) +#define RT5668_PLL1_SRC_BCLK1 (0x1 << 8) +#define RT5668_PLL1_PD_MASK (0x1 << 4) +#define RT5668_PLL1_PD_SHIFT 4 + +#define RT5668_PLL_INP_MAX 40000000 +#define RT5668_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5668_PLL_N_MAX 0x001ff +#define RT5668_PLL_N_MASK (RT5668_PLL_N_MAX << 7) +#define RT5668_PLL_N_SHIFT 7 +#define RT5668_PLL_K_MAX 0x001f +#define RT5668_PLL_K_MASK (RT5668_PLL_K_MAX) +#define RT5668_PLL_K_SHIFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5668_PLL_M_MAX 0x00f +#define RT5668_PLL_M_MASK (RT5668_PLL_M_MAX << 12) +#define RT5668_PLL_M_SHIFT 12 +#define RT5668_PLL_M_BP (0x1 << 11) +#define RT5668_PLL_M_BP_SHIFT 11 + +/* PLL tracking mode 1 (0x0083) */ +#define RT5668_I2S1_ASRC_MASK (0x1 << 13) +#define RT5668_I2S1_ASRC_SHIFT 13 +#define RT5668_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5668_DAC_STO1_ASRC_SHIFT 12 +#define RT5668_ADC_STO1_ASRC_MASK (0x1 << 4) +#define RT5668_ADC_STO1_ASRC_SHIFT 4 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5668_DA_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_DA_STO1_TRACK_SHIFT 12 +#define RT5668_DA_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_DA_STO1_TRACK_I2S1 (0x1 << 12) + +/* PLL tracking mode 3 (0x0085)*/ +#define RT5668_AD_STO1_TRACK_MASK (0x7 << 12) +#define RT5668_AD_STO1_TRACK_SHIFT 12 +#define RT5668_AD_STO1_TRACK_SYSCLK (0x0 << 12) +#define RT5668_AD_STO1_TRACK_I2S1 (0x1 << 12) + +/* HPOUT Charge pump control 1 (0x0091) */ +#define RT5668_OSW_HP_L_MASK (0x1 << 11) +#define RT5668_OSW_HP_L_SHIFT 11 +#define RT5668_OSW_HP_L_EN (0x1 << 11) +#define RT5668_OSW_HP_L_DIS (0x0 << 11) +#define RT5668_OSW_HP_R_MASK (0x1 << 10) +#define RT5668_OSW_HP_R_SHIFT 10 +#define RT5668_OSW_HP_R_EN (0x1 << 10) +#define RT5668_OSW_HP_R_DIS (0x0 << 10) +#define RT5668_SEL_PM_HP_MASK (0x3 << 8) +#define RT5668_SEL_PM_HP_SHIFT 8 +#define RT5668_SEL_PM_HP_0_6 (0x0 << 8) +#define RT5668_SEL_PM_HP_0_9 (0x1 << 8) +#define RT5668_SEL_PM_HP_1_8 (0x2 << 8) +#define RT5668_SEL_PM_HP_HIGH (0x3 << 8) +#define RT5668_OVCD_HP_MASK (0x1 << 2) +#define RT5668_OVCD_HP_SHIFT 2 +#define RT5668_OVCD_HP_EN (0x1 << 2) +#define RT5668_OVCD_HP_DIS (0x0 << 2) + +/* RC Clock Control (0x0094) */ +#define RT5668_DIG_25M_CLK_MASK (0x1 << 9) +#define RT5668_DIG_25M_CLK_SHIFT 9 +#define RT5668_DIG_25M_CLK_DIS (0x0 << 9) +#define RT5668_DIG_25M_CLK_EN (0x1 << 9) +#define RT5668_DIG_1M_CLK_MASK (0x1 << 8) +#define RT5668_DIG_1M_CLK_SHIFT 8 +#define RT5668_DIG_1M_CLK_DIS (0x0 << 8) +#define RT5668_DIG_1M_CLK_EN (0x1 << 8) + +/* Auto Turn On 1M RC CLK (0x009f) */ +#define RT5668_IRQ_POW_SAV_MASK (0x1 << 15) +#define RT5668_IRQ_POW_SAV_SHIFT 15 +#define RT5668_IRQ_POW_SAV_DIS (0x0 << 15) +#define RT5668_IRQ_POW_SAV_EN (0x1 << 15) +#define RT5668_IRQ_POW_SAV_JD1_MASK (0x1 << 14) +#define RT5668_IRQ_POW_SAV_JD1_SHIFT 14 +#define RT5668_IRQ_POW_SAV_JD1_DIS (0x0 << 14) +#define RT5668_IRQ_POW_SAV_JD1_EN (0x1 << 14) + +/* IRQ Control 1 (0x00b6) */ +#define RT5668_EN_CB_JD_MASK (0x1 << 3) +#define RT5668_EN_CB_JD_SHIFT 3 +#define RT5668_EN_CB_JD_EN (0x1 << 3) +#define RT5668_EN_CB_JD_DIS (0x0 << 3) + +/* IRQ Control 3 (0x00b8) */ +#define RT5668_EN_IRQ_INLINE_MASK (0x1 << 6) +#define RT5668_EN_IRQ_INLINE_SHIFT 6 +#define RT5668_EN_IRQ_INLINE_BYP (0x0 << 6) +#define RT5668_EN_IRQ_INLINE_NOR (0x1 << 6) + +/* GPIO Control 1 (0x00c0) */ +#define RT5668_GP1_PIN_MASK (0x1 << 15) +#define RT5668_GP1_PIN_SHIFT 15 +#define RT5668_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5668_GP1_PIN_IRQ (0x1 << 15) + +/* GPIO Control 2 (0x00c1) */ +#define RT5668_GP4_PIN_CONF_MASK (0x1 << 5) +#define RT5668_GP4_PIN_CONF_SHIFT 5 +#define RT5668_GP4_PIN_CONF_INPUT (0x0 << 5) +#define RT5668_GP4_PIN_CONF_OUTPUT (0x1 << 5) + +/* GPIO Control 2 (0x00c2) */ +#define RT5668_GP8_PIN_CONF_MASK (0x1 << 13) +#define RT5668_GP8_PIN_CONF_SHIFT 13 +#define RT5668_GP8_PIN_CONF_INPUT (0x0 << 13) +#define RT5668_GP8_PIN_CONF_OUTPUT (0x1 << 13) + +/* 4 Buttons Inline Command Function 1 (0x00df) */ +#define RT5668_4BTN_CLK_DEB_MASK (0x3 << 2) +#define RT5668_4BTN_CLK_DEB_SHIFT 2 +#define RT5668_4BTN_CLK_DEB_8MS (0x0 << 2) +#define RT5668_4BTN_CLK_DEB_16MS (0x1 << 2) +#define RT5668_4BTN_CLK_DEB_32MS (0x2 << 2) +#define RT5668_4BTN_CLK_DEB_65MS (0x3 << 2) + +/* Inline Command Function 6 (0x00e0) */ +#define RT5668_EN_4BTN_INL_MASK (0x1 << 15) +#define RT5668_EN_4BTN_INL_SHIFT 15 +#define RT5668_EN_4BTN_INL_DIS (0x0 << 15) +#define RT5668_EN_4BTN_INL_EN (0x1 << 15) +#define RT5668_RESET_4BTN_INL_MASK (0x1 << 14) +#define RT5668_RESET_4BTN_INL_SHIFT 14 +#define RT5668_RESET_4BTN_INL_RESET (0x0 << 14) +#define RT5668_RESET_4BTN_INL_NOR (0x1 << 14) + +/* Digital Misc Control (0x00fa) */ +#define RT5668_DIG_GATE_CTRL_MASK 0x1 +#define RT5668_DIG_GATE_CTRL_SHIFT (0) +#define RT5668_DIG_GATE_CTRL_DIS 0x0 +#define RT5668_DIG_GATE_CTRL_EN 0x1 + +/* Chopper and Clock control for DAC L (0x013a)*/ +#define RT5668_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5668_CKXEN_DAC1_SHIFT 13 +#define RT5668_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5668_CKGEN_DAC1_SHIFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5668_CKXEN_ADCC_MASK (0x1 << 13) +#define RT5668_CKXEN_ADCC_SHIFT 13 +#define RT5668_CKGEN_ADCC_MASK (0x1 << 12) +#define RT5668_CKGEN_ADCC_SHIFT 12 + +/* HP Behavior Logic Control 2 (0x01db) */ +#define RT5668_HP_SIG_SRC1_MASK (0x3) +#define RT5668_HP_SIG_SRC1_SHIFT 0 +#define RT5668_HP_SIG_SRC1_HP_DC (0x0) +#define RT5668_HP_SIG_SRC1_HP_CALIB (0x1) +#define RT5668_HP_SIG_SRC1_REG (0x2) +#define RT5668_HP_SIG_SRC1_SILENCE (0x3) + +/* RT5663 specific register */ +#define RT5663_RECMIX 0x0010 +#define RT5663_BYPASS_STO_DAC 0x002d +#define RT5663_PRE_DIV_GATING_1 0x006e +#define RT5663_PRE_DIV_GATING_2 0x006f +#define RT5663_TDM_1 0x0077 +#define RT5663_TDM_5 0x007b +#define RT5663_EM_JACK_TYPE_1 0x00e6 +#define RT5663_EM_JACK_TYPE_2 0x00e7 +#define RT5663_ANA_BIAS_CUR_1 0x0108 +#define RT5663_DACREF_LDO 0x0112 +#define RT5663_VREF_RECMIX 0x0118 +#define RT5663_CHARGE_PUMP_1 0x0125 +#define RT5663_CHARGE_PUMP_2 0x0128 +#define RT5663_CHOP_DAC_R 0x013d +#define RT5663_STO_DRE_9 0x0168 +#define RT5663_STO_DRE_10 0x0169 +#define RT5663_HP_BIAS 0x0202 +#define RT5663_CBJ_1 0x0250 +#define RT5663_CBJ_2 0x0251 +#define RT5663_DUMMY_1 0x02fa + +/* RT5663: RECMIX Control (0x0010) */ +#define RT5663_RECMIX1_BST1_MASK (0x1) +#define RT5663_RECMIX1_BST1_SHIFT 0 +#define RT5663_RECMIX1_BST1_UNMUTE (0x0) +#define RT5663_RECMIX1_BST1_MUTE (0x1) + +/* RT5663: Bypass Stereo1 DAC Mixer Control (0x002d) */ +#define RT5663_DACL1_SRC_MASK (0x1 << 3) +#define RT5663_DACL1_SRC_SHIFT 3 +#define RT5663_DACR1_SRC_MASK (0x1 << 2) +#define RT5663_DACR1_SRC_SHIFT 2 + +/* RT5663: TDM control 2 (0x0078) */ +#define RT5663_DATA_SWAP_ADCDAT1_MASK (0x3 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_SHIFT 14 +#define RT5663_DATA_SWAP_ADCDAT1_LR (0x0 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RL (0x1 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_LL (0x2 << 14) +#define RT5663_DATA_SWAP_ADCDAT1_RR (0x3 << 14) + +/* RT5663: TDM control 5 (0x007b) */ +#define RT5663_TDM_LENGTN_MASK (0x3) +#define RT5663_TDM_LENGTN_SHIFT 0 +#define RT5663_TDM_LENGTN_16 (0x0) +#define RT5663_TDM_LENGTN_20 (0x1) +#define RT5663_TDM_LENGTN_24 (0x2) +#define RT5663_TDM_LENGTN_32 (0x3) + +/* RT5663: Global Clock Control (0x0080) */ +#define RT5663_PLL1_SRC_MASK (0x7 << 11) +#define RT5663_PLL1_SRC_SHIFT 11 +#define RT5663_PLL1_SRC_MCLK (0x0 << 11) +#define RT5663_PLL1_SRC_BCLK1 (0x1 << 11) + +/* PLL tracking mode 1 (0x0083) */ +#define RT5663_I2S1_ASRC_MASK (0x1 << 11) +#define RT5663_I2S1_ASRC_SHIFT 11 +#define RT5663_DAC_STO1_ASRC_MASK (0x1 << 10) +#define RT5663_DAC_STO1_ASRC_SHIFT 10 +#define RT5663_ADC_STO1_ASRC_MASK (0x1 << 3) +#define RT5663_ADC_STO1_ASRC_SHIFT 3 + +/* PLL tracking mode 2 (0x0084)*/ +#define RT5663_AD_STO1_TRACK_MASK (0x7) +#define RT5663_AD_STO1_TRACK_SHIFT 0 +#define RT5663_AD_STO1_TRACK_SYSCLK (0x0) +#define RT5663_AD_STO1_TRACK_I2S1 (0x1) + +/* RT5663: HPOUT Charge pump control 1 (0x0091) */ +#define RT5663_SI_HP_MASK (0x1 << 12) +#define RT5663_SI_HP_SHIFT 12 +#define RT5663_SI_HP_EN (0x1 << 12) +#define RT5663_SI_HP_DIS (0x0 << 12) + +/* RT5663: GPIO Control 2 (0x00b6) */ +#define RT5663_GP1_PIN_CONF_MASK (0x1 << 2) +#define RT5663_GP1_PIN_CONF_SHIFT 2 +#define RT5663_GP1_PIN_CONF_OUTPUT (0x1 << 2) +#define RT5663_GP1_PIN_CONF_INPUT (0x0 << 2) + +/* RT5663: GPIO Control 2 (0x00b7) */ +#define RT5663_EN_IRQ_INLINE_MASK (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_SHIFT 3 +#define RT5663_EN_IRQ_INLINE_NOR (0x1 << 3) +#define RT5663_EN_IRQ_INLINE_BYP (0x0 << 3) + +/* RT5663: IRQ Control 1 (0x00c1) */ +#define RT5663_EN_IRQ_JD1_MASK (0x1 << 6) +#define RT5663_EN_IRQ_JD1_SHIFT 6 +#define RT5663_EN_IRQ_JD1_EN (0x1 << 6) +#define RT5663_EN_IRQ_JD1_DIS (0x0 << 6) + +/* RT5663: Inline Command Function 2 (0x00dc) */ +#define RT5663_PWR_MIC_DET_MASK (0x1) +#define RT5663_PWR_MIC_DET_SHIFT 0 +#define RT5663_PWR_MIC_DET_ON (0x1) +#define RT5663_PWR_MIC_DET_OFF (0x0) + +/* RT5663: Embeeded Jack and Type Detection Control 1 (0x00e6)*/ +#define RT5663_CBJ_DET_MASK (0x1 << 15) +#define RT5663_CBJ_DET_SHIFT 15 +#define RT5663_CBJ_DET_DIS (0x0 << 15) +#define RT5663_CBJ_DET_EN (0x1 << 15) +#define RT5663_EXT_JD_MASK (0x1 << 11) +#define RT5663_EXT_JD_SHIFT 11 +#define RT5663_EXT_JD_EN (0x1 << 11) +#define RT5663_EXT_JD_DIS (0x0 << 11) +#define RT5663_POL_EXT_JD_MASK (0x1 << 10) +#define RT5663_POL_EXT_JD_SHIFT 10 +#define RT5663_POL_EXT_JD_EN (0x1 << 10) +#define RT5663_POL_EXT_JD_DIS (0x0 << 10) + +/* RT5663: DACREF LDO Control (0x0112)*/ +#define RT5663_PWR_LDO_DACREFL_MASK (0x1 << 9) +#define RT5663_PWR_LDO_DACREFL_SHIFT 9 +#define RT5663_PWR_LDO_DACREFR_MASK (0x1 << 1) +#define RT5663_PWR_LDO_DACREFR_SHIFT 1 + +/* RT5663: Stereo Dynamic Range Enhancement Control 9 (0x0168, 0x0169)*/ +#define RT5663_DRE_GAIN_HP_MASK (0x1f) +#define RT5663_DRE_GAIN_HP_SHIFT 0 + +/* RT5663: Combo Jack Control (0x0250) */ +#define RT5663_INBUF_CBJ_BST1_MASK (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_SHIFT 11 +#define RT5663_INBUF_CBJ_BST1_ON (0x1 << 11) +#define RT5663_INBUF_CBJ_BST1_OFF (0x0 << 11) +#define RT5663_CBJ_SENSE_BST1_MASK (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_SHIFT 10 +#define RT5663_CBJ_SENSE_BST1_L (0x1 << 10) +#define RT5663_CBJ_SENSE_BST1_R (0x0 << 10) + +/* RT5663: Combo Jack Control (0x0251) */ +#define RT5663_GAIN_BST1_MASK (0xf) +#define RT5663_GAIN_BST1_SHIFT 0 + +/* RT5663: Dummy register 1 (0x02fa) */ +#define RT5663_EMB_CLK_MASK (0x1 << 9) +#define RT5663_EMB_CLK_SHIFT 9 +#define RT5663_EMB_CLK_EN (0x1 << 9) +#define RT5663_EMB_CLK_DIS (0x0 << 9) +#define RT5663_HPA_CPL_BIAS_MASK (0x7 << 6) +#define RT5663_HPA_CPL_BIAS_SHIFT 6 +#define RT5663_HPA_CPL_BIAS_0_5 (0x0 << 6) +#define RT5663_HPA_CPL_BIAS_1 (0x1 << 6) +#define RT5663_HPA_CPL_BIAS_2 (0x2 << 6) +#define RT5663_HPA_CPL_BIAS_3 (0x3 << 6) +#define RT5663_HPA_CPL_BIAS_4_1 (0x4 << 6) +#define RT5663_HPA_CPL_BIAS_4_2 (0x5 << 6) +#define RT5663_HPA_CPL_BIAS_6 (0x6 << 6) +#define RT5663_HPA_CPL_BIAS_8 (0x7 << 6) +#define RT5663_HPA_CPR_BIAS_MASK (0x7 << 3) +#define RT5663_HPA_CPR_BIAS_SHIFT 3 +#define RT5663_HPA_CPR_BIAS_0_5 (0x0 << 3) +#define RT5663_HPA_CPR_BIAS_1 (0x1 << 3) +#define RT5663_HPA_CPR_BIAS_2 (0x2 << 3) +#define RT5663_HPA_CPR_BIAS_3 (0x3 << 3) +#define RT5663_HPA_CPR_BIAS_4_1 (0x4 << 3) +#define RT5663_HPA_CPR_BIAS_4_2 (0x5 << 3) +#define RT5663_HPA_CPR_BIAS_6 (0x6 << 3) +#define RT5663_HPA_CPR_BIAS_8 (0x7 << 3) +#define RT5663_DUMMY_BIAS_MASK (0x7) +#define RT5663_DUMMY_BIAS_SHIFT 0 +#define RT5663_DUMMY_BIAS_0_5 (0x0) +#define RT5663_DUMMY_BIAS_1 (0x1) +#define RT5663_DUMMY_BIAS_2 (0x2) +#define RT5663_DUMMY_BIAS_3 (0x3) +#define RT5663_DUMMY_BIAS_4_1 (0x4) +#define RT5663_DUMMY_BIAS_4_2 (0x5) +#define RT5663_DUMMY_BIAS_6 (0x6) +#define RT5663_DUMMY_BIAS_8 (0x7) + + +/* System Clock Source */ +enum { + RT5668_SCLK_S_MCLK, + RT5668_SCLK_S_PLL1, + RT5668_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5668_PLL1_S_MCLK, + RT5668_PLL1_S_BCLK1, +}; + +enum { + RT5668_AIF, + RT5668_AIFS, +}; + +/* asrc clock source */ +enum { + RT5668_CLK_SEL_SYS = 0x0, + RT5668_CLK_SEL_I2S1_ASRC = 0x1, +}; + +/* filter mask */ +enum { + RT5668_DA_STEREO_FILTER = 0x1, + RT5668_AD_STEREO_FILTER = 0x2, +}; + +int rt5668_headset_detect(struct snd_soc_codec *codec, int jack_insert); +int rt5668_button_detect(struct snd_soc_codec *codec); +int rt5668_set_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *hs_jack); +int rt5668_sel_asrc_clk_src(struct snd_soc_codec *codec, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5668_H__ */ -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v3 0/3] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 13 May '16

13 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver and hdmi codec (runs on HDA link) The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are closed, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and rest two are dependent on it so Takashi if you are okay, pls ACK it and we can merge thru ASoC tree Changes in v3: - rename the varaible for dmas and make it bool - change the logic of checking links for powering off dmas as suggested by takashi Changes in v2: - use mutex instead of spinlock Vinod Koul (3): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link mangement ASoC: hdac_hdmi: add link management include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 66 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hdmi.c | 32 ++++++++++++++++-- sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 +++++++++++++++++++ 6 files changed, 146 insertions(+), 3 deletions(-) -- 1.9.1

3 7

[alsa-devel] [PATCH v2 0/3] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 13 May '16

13 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver and hdmi codec (runs on HDA link) The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are closed, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and rest two are dependent on it so Takashi if you are okay, pls ACK it and we can merge thru ASoC tree Changes in v2: - use mutex instead of spinlock Vinod Koul (3): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link management ASoC: hdac_hdmi: add link management include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/codecs/hdac_hdmi.c | 32 +++++++++++++++++-- sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 ++++++++++++++++++++ 6 files changed, 142 insertions(+), 3 deletions(-) -- 1.9.1

3 9

[alsa-devel] [PATCH] ASoC: tlv320aic32x4: Properly implement the positive and negative pins into the mixers
by Jeremy McDermond 13 May '16

13 May '16

The TLV320AIC32x4 has a very flexible mixer on the inputs to the ADCs. Each mixer has an available set of available pins that can be connected to the ADC positive and negative pins via three different resistor values. This allows for configuration of differential inputs as well as doing level manipulation between sources going into the mixers. The current code only provides positive pins and I implemented the resistors in an earlier patch. It turns out that it appears to more accurately model what's happening to implement each of the pins as a MUX rather than on/off switches and a mixer. This way each pin can be set to its desired resistor value. Since there are no switches, the mixer is no longer necessary in the DAPM path. I set the DAPM paths such that the "off" position of any of the MUXes turns the path off. This should allow for any input confiuration available on the codec. Signed-off-by: Jeremy McDermond <nh6z(a)nh6z.net> --- sound/soc/codecs/tlv320aic32x4.c | 210 ++++++++++++++++++++++++++++++--------- 1 file changed, 161 insertions(+), 49 deletions(-) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 0eb8acc..85d4978 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -187,36 +187,67 @@ static const char * const resistor_text[] = { "Off", "10 kOhm", "20 kOhm", "40 kOhm", }; -static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, - resistor_text); - -static const struct snd_kcontrol_new left_input_mixer_controls[] = { - SOC_DAPM_ENUM("IN1_L P Switch", in1l_lpga_p_enum), - SOC_DAPM_ENUM("IN2_L P Switch", in2l_lpga_p_enum), - SOC_DAPM_ENUM("IN3_L P Switch", in3l_lpga_p_enum), - SOC_DAPM_ENUM("IN1_R P Switch", in1r_lpga_p_enum), +/* Left mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1l_lpga_p_enum, AIC32X4_LMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_lpga_p_enum, AIC32X4_LMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_lpga_p_enum, AIC32X4_LMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1r_lpga_p_enum, AIC32X4_LMICPGAPIN, 0, resistor_text); + +static SOC_ENUM_SINGLE_DECL(cml_lpga_n_enum, AIC32X4_LMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_lpga_n_enum, AIC32X4_LMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_lpga_n_enum, AIC32X4_LMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L L+ Switch", in1l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L L+ Switch", in2l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in3l_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L L+ Switch", in3l_lpga_p_enum), +}; +static const struct snd_kcontrol_new in1r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R L+ Switch", in1r_lpga_p_enum), +}; +static const struct snd_kcontrol_new cml_to_lmixer_controls[] = { + SOC_DAPM_ENUM("CM_L L- Switch", cml_lpga_n_enum), +}; +static const struct snd_kcontrol_new in2r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R L- Switch", in2r_lpga_n_enum), +}; +static const struct snd_kcontrol_new in3r_to_lmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R L- Switch", in3r_lpga_n_enum), }; -static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, - resistor_text); -static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, - resistor_text); - -static const struct snd_kcontrol_new right_input_mixer_controls[] = { - SOC_DAPM_ENUM("IN1_R P Switch", in1r_rpga_p_enum), - SOC_DAPM_ENUM("IN2_R P Switch", in2r_rpga_p_enum), - SOC_DAPM_ENUM("IN3_R P Switch", in3r_rpga_p_enum), - SOC_DAPM_ENUM("IN2_L P Switch", in2l_rpga_p_enum), +/* Right mixer pins */ +static SOC_ENUM_SINGLE_DECL(in1r_rpga_p_enum, AIC32X4_RMICPGAPIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2r_rpga_p_enum, AIC32X4_RMICPGAPIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3r_rpga_p_enum, AIC32X4_RMICPGAPIN, 2, resistor_text); +static SOC_ENUM_SINGLE_DECL(in2l_rpga_p_enum, AIC32X4_RMICPGAPIN, 0, resistor_text); +static SOC_ENUM_SINGLE_DECL(cmr_rpga_n_enum, AIC32X4_RMICPGANIN, 6, resistor_text); +static SOC_ENUM_SINGLE_DECL(in1l_rpga_n_enum, AIC32X4_RMICPGANIN, 4, resistor_text); +static SOC_ENUM_SINGLE_DECL(in3l_rpga_n_enum, AIC32X4_RMICPGANIN, 2, resistor_text); + +static const struct snd_kcontrol_new in1r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_R R+ Switch", in1r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_R R+ Switch", in2r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in3r_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_R R+ Switch", in3r_rpga_p_enum), +}; +static const struct snd_kcontrol_new in2l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN2_L R+ Switch", in2l_rpga_p_enum), +}; +static const struct snd_kcontrol_new cmr_to_rmixer_controls[] = { + SOC_DAPM_ENUM("CM_R R- Switch", cmr_rpga_n_enum), +}; +static const struct snd_kcontrol_new in1l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN1_L R- Switch", in1l_rpga_n_enum), +}; +static const struct snd_kcontrol_new in3l_to_rmixer_controls[] = { + SOC_DAPM_ENUM("IN3_L R- Switch", in3l_rpga_n_enum), }; static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { @@ -240,14 +271,39 @@ static const struct snd_soc_dapm_widget aic32x4_dapm_widgets[] = { &lor_output_mixer_controls[0], ARRAY_SIZE(lor_output_mixer_controls)), SND_SOC_DAPM_PGA("LOR Power", AIC32X4_OUTPWRCTL, 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Left Input Mixer", SND_SOC_NOPM, 0, 0, - &left_input_mixer_controls[0], - ARRAY_SIZE(left_input_mixer_controls)), - SND_SOC_DAPM_MIXER("Right Input Mixer", SND_SOC_NOPM, 0, 0, - &right_input_mixer_controls[0], - ARRAY_SIZE(right_input_mixer_controls)), - SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", AIC32X4_ADCSETUP, 6, 0), + SND_SOC_DAPM_MUX("IN1_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Right Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_rmixer_controls), + SND_SOC_DAPM_MUX("CM_R to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cmr_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN1_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_rmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Right Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_rmixer_controls), + + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", AIC32X4_ADCSETUP, 7, 0), + SND_SOC_DAPM_MUX("IN1_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in2l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_L to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in3l_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN1_R to Left Mixer Positive Resistor", SND_SOC_NOPM, 0, 0, + in1r_to_lmixer_controls), + SND_SOC_DAPM_MUX("CM_L to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + cml_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN2_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in2r_to_lmixer_controls), + SND_SOC_DAPM_MUX("IN3_R to Left Mixer Negative Resistor", SND_SOC_NOPM, 0, 0, + in3r_to_lmixer_controls), + SND_SOC_DAPM_MICBIAS("Mic Bias", AIC32X4_MICBIAS, 6, 0), SND_SOC_DAPM_OUTPUT("HPL"), @@ -287,21 +343,77 @@ static const struct snd_soc_dapm_route aic32x4_dapm_routes[] = { {"LOR Power", NULL, "LOR Output Mixer"}, {"LOR", NULL, "LOR Power"}, - /* Left input */ - {"Left Input Mixer", "IN1_L P Switch", "IN1_L"}, - {"Left Input Mixer", "IN2_L P Switch", "IN2_L"}, - {"Left Input Mixer", "IN3_L P Switch", "IN3_L"}, - {"Left Input Mixer", "IN1_R P Switch", "IN1_R"}, - - {"Left ADC", NULL, "Left Input Mixer"}, - /* Right Input */ - {"Right Input Mixer", "IN1_R P Switch", "IN1_R"}, - {"Right Input Mixer", "IN2_R P Switch", "IN2_R"}, - {"Right Input Mixer", "IN3_R P Switch", "IN3_R"}, - {"Right Input Mixer", "IN2_L P Switch", "IN2_L"}, - - {"Right ADC", NULL, "Right Input Mixer"}, + {"Right ADC", NULL, "IN1_R to Right Mixer Positive Resistor"}, + {"IN1_R to Right Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Right Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Right ADC", NULL, "IN2_R to Right Mixer Positive Resistor"}, + {"IN2_R to Right Mixer Positive Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Right Mixer Positive Resistor", "40 kOhm", "IN2_R"}, + + {"Right ADC", NULL, "IN3_R to Right Mixer Positive Resistor"}, + {"IN3_R to Right Mixer Positive Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Right Mixer Positive Resistor", "40 kOhm", "IN3_R"}, + + {"Right ADC", NULL, "IN2_L to Right Mixer Positive Resistor"}, + {"IN2_L to Right Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Right Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Right ADC", NULL, "CM_R to Right Mixer Negative Resistor"}, + {"CM_R to Right Mixer Negative Resistor", "10 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "20 kOhm", "CM_R"}, + {"CM_R to Right Mixer Negative Resistor", "40 kOhm", "CM_R"}, + + {"Right ADC", NULL, "IN1_L to Right Mixer Negative Resistor"}, + {"IN1_L to Right Mixer Negative Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Right Mixer Negative Resistor", "40 kOhm", "IN1_L"}, + + {"Right ADC", NULL, "IN3_L to Right Mixer Negative Resistor"}, + {"IN3_L to Right Mixer Negative Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Right Mixer Negative Resistor", "40 kOhm", "IN3_L"}, + + /* Left Input */ + {"Left ADC", NULL, "IN1_L to Left Mixer Positive Resistor"}, + {"IN1_L to Left Mixer Positive Resistor", "10 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "20 kOhm", "IN1_L"}, + {"IN1_L to Left Mixer Positive Resistor", "40 kOhm", "IN1_L"}, + + {"Left ADC", NULL, "IN2_L to Left Mixer Positive Resistor"}, + {"IN2_L to Left Mixer Positive Resistor", "10 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "20 kOhm", "IN2_L"}, + {"IN2_L to Left Mixer Positive Resistor", "40 kOhm", "IN2_L"}, + + {"Left ADC", NULL, "IN3_L to Left Mixer Positive Resistor"}, + {"IN3_L to Left Mixer Positive Resistor", "10 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "20 kOhm", "IN3_L"}, + {"IN3_L to Left Mixer Positive Resistor", "40 kOhm", "IN3_L"}, + + {"Left ADC", NULL, "IN1_R to Left Mixer Positive Resistor"}, + {"IN1_R to Left Mixer Positive Resistor", "10 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "20 kOhm", "IN1_R"}, + {"IN1_R to Left Mixer Positive Resistor", "40 kOhm", "IN1_R"}, + + {"Left ADC", NULL, "CM_L to Left Mixer Negative Resistor"}, + {"CM_L to Left Mixer Negative Resistor", "10 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "20 kOhm", "CM_L"}, + {"CM_L to Left Mixer Negative Resistor", "40 kOhm", "CM_L"}, + + {"Left ADC", NULL, "IN2_R to Left Mixer Negative Resistor"}, + {"IN2_R to Left Mixer Negative Resistor", "10 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "20 kOhm", "IN2_R"}, + {"IN2_R to Left Mixer Negative Resistor", "40 kOhm", "IN2_R"}, + + {"Left ADC", NULL, "IN3_R to Left Mixer Negative Resistor"}, + {"IN3_R to Left Mixer Negative Resistor", "10 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "20 kOhm", "IN3_R"}, + {"IN3_R to Left Mixer Negative Resistor", "40 kOhm", "IN3_R"}, }; static const struct regmap_range_cfg aic32x4_regmap_pages[] = { -- 2.5.0

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. ;) Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++++++++++++++-------- 1 file changed, 63 insertions(+), 11 deletions(-) -- 1.9.1

5 11

[alsa-devel] [PATCH 0/5] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Bugs are detected and fixed on a imx6sl platform with linux 4.3, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (5): ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 69 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 60 insertions(+), 9 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v4 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 13 May '16

13 May '16

Version 4: - rebase on top of v4.6-rcX + alsa-broonie/for-next. - The only change with v3 concerns the patch 6/6, since CCSR_SSI_SOR is now already included in the list of volatile registers. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 80 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 69 insertions(+), 11 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 0/7] ARM/ASoC: OMAP3: Fix McBSP2/3 sidetone support
by Peter Ujfalusi 13 May '16

13 May '16

Hi, 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(-) -- 2.8.1

3 13

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: note buggy I2S support when the codec masters LRCLK
by Peter Rosin 13 May '16

13 May '16

The only difference in register settings between I2S and DSP mode A, for the SND_SOC_DAIFMT_CBM_CFM case, is the start condition (which is SSC_START_FALLING_RF for I2S and SSC_START_RISING_RF for DSP mode A). As this is not the only difference between the two formats, it is an indicator that something is seriously wrong. And it is the I2S support that is broken. While the start condition is correct for the left channel word in the I2S case, it is not correct that the right channel word follows immediately after the left channel word. The start of the right channel word should be triggered by a rising edge on LRCLK in the I2S case, something which simply does not happen. The correct thing to do would be to simply remove the alleged support for I2S/CBM_CFM, but that is likely to cause regressions. So, just make a note about the buggy I2S support in the source. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 4ab1e2013238..3108c6e20ad5 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -558,7 +558,15 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CBM_CFM: - /* I2S format, CODEC supplies BCLK and LRC clocks. */ + /* + * I2S format, CODEC supplies BCLK and LRC clocks. + * NOTE! This is not really I2S, as the SSC does not allow + * for any extra empty BCLK cycles after the left channel + * word. I.e. the SSC is only looking at the falling edge + * of LRCLK and ignores the rising edge. + * In reality, this is DSP mode A with inverted LRCLK. + * Use DSP mode A instead, if your codec supports it. + */ rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_FALLING_RF) -- 2.1.4

2 3

[alsa-devel] [PATCH 0/9] ASoC: es8328 codec improvements
by John Keeping 13 May '16

13 May '16

The first 4 patches are bug fixes, then the next 4 add support for more sample rates and formats. The final patch I'm slightly unsure about since the hardware does allow different playback and capture rates, but it only allows 8kHz and 48kHz or 8.0182kHz and 44.1kHz, all other rates have to be symmetric. All of these patches have been tested on a Radxa Rock2 Square with rockchip-i2s. John Keeping (9): ASoC: es8328: Move clock setup to hw_params ASoC: es8328: Fix ADC format setup ASoC: es8328: Fix mask for VMIDSEL ASoC: es8328: Use single R/W for regmap ASoC: es8328: Use more suitable definition for mic bias ASoC: es8328: Move sample size setup to hw_params ASoC: es8328: Support more sample formats ASoC: es8328: Support more sample rates ASoC: es8328: Set symmetric rates sound/soc/codecs/es8328.c | 198 +++++++++++++++++++++++++++++++++++----------- sound/soc/codecs/es8328.h | 23 ++++-- 2 files changed, 168 insertions(+), 53 deletions(-) -- 2.8.0.rc4.238.g874082a

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[alsa-devel] [PATCH 1/2] ASoC: rt298: fix null deref on acpi driver data
by Vinod Koul 13 May '16

13 May '16

ACPI driver data can be NULL so we need to check that before dereference the driver data. Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/rt298.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 68cf8d5a174f..45dc5fa24961 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -1179,7 +1179,7 @@ static int rt298_i2c_probe(struct i2c_client *i2c, /* enable jack combo mode on supported devices */ acpiid = acpi_match_device(dev->driver->acpi_match_table, dev); - if (acpiid) { + if (acpiid && acpiid->driver_data) { rt298->pdata = *(struct rt298_platform_data *) acpiid->driver_data; } -- 1.9.1

3 5

[alsa-devel] [PATCH] rcar: src: skip disabled-SRC nodes
by Sergei Shtylyov 13 May '16

13 May '16

The current device tree representation of the R-Car Sample Rate Converters (SRC) assumes that they are numbered consecutively, starting from 0. Alas, this is not the case with the R8A7794 SoC where SRC0 isn't present. In order to keep the existing device trees working, I'm suggesting to use a disabled node for SRC0. Teach the SRC probe to just skip disabled nodes. Signed-off-by: Sergei Shtylyov <sergei.shtylyov(a)cogentembedded.com> --- The patch is against the 'for-next' branch of Mark Brown's 'sound.git' repo. sound/soc/sh/rcar/src.c | 4 ++++ sound/soc/sh/rcar/src.c | 4 ++++ 1 file changed, 4 insertions(+) Index: sound/sound/soc/sh/rcar/src.c =================================================================== --- sound.orig/sound/soc/sh/rcar/src.c +++ sound/sound/soc/sh/rcar/src.c @@ -572,6 +572,9 @@ int rsnd_src_probe(struct rsnd_priv *pri i = 0; for_each_child_of_node(node, np) { + if (!of_device_is_available(np)) + goto skip; + src = rsnd_src_get(priv, i); snprintf(name, RSND_SRC_NAME_SIZE, "%s.%d", @@ -595,6 +598,7 @@ int rsnd_src_probe(struct rsnd_priv *pri if (ret) goto rsnd_src_probe_done; +skip: i++; }

3 3

[alsa-devel] [PATCH] ASoC: rsnd: count .probe/.remove for rsnd_mod_call()
by Kuninori Morimoto 13 May '16

13 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_mod_call is counting its calling count to avoid unbalanced function pair calling for error cases (ex init <-> quit). SSI parent is now controlled as "mod" on current rsnd driver. Because of this reason, SSI .remove function will be called twice if it was used as SSI parent when user tried unbind. But probe/remove pair were not counted. This patch counts probe/remove functions to avoid it. Special thans Hiep Reported-by: Hiep Cao Minh <cm-hiep(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/rsnd.h | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index fc89a67..a8f61d7 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -276,8 +276,9 @@ struct rsnd_mod { /* * status * - * 0xH0000CB0 + * 0xH0000CBA * + * A 0: probe 1: remove * B 0: init 1: quit * C 0: start 1: stop * @@ -287,19 +288,19 @@ struct rsnd_mod { * H 0: fallback * H 0: hw_params */ +#define __rsnd_mod_shift_probe 0 +#define __rsnd_mod_shift_remove 0 #define __rsnd_mod_shift_init 4 #define __rsnd_mod_shift_quit 4 #define __rsnd_mod_shift_start 8 #define __rsnd_mod_shift_stop 8 -#define __rsnd_mod_shift_probe 28 /* always called */ -#define __rsnd_mod_shift_remove 28 /* always called */ #define __rsnd_mod_shift_irq 28 /* always called */ #define __rsnd_mod_shift_pcm_new 28 /* always called */ #define __rsnd_mod_shift_fallback 28 /* always called */ #define __rsnd_mod_shift_hw_params 28 /* always called */ -#define __rsnd_mod_add_probe 0 -#define __rsnd_mod_add_remove 0 +#define __rsnd_mod_add_probe 1 +#define __rsnd_mod_add_remove -1 #define __rsnd_mod_add_init 1 #define __rsnd_mod_add_quit -1 #define __rsnd_mod_add_start 1 @@ -310,7 +311,7 @@ struct rsnd_mod { #define __rsnd_mod_add_hw_params 0 #define __rsnd_mod_call_probe 0 -#define __rsnd_mod_call_remove 0 +#define __rsnd_mod_call_remove 1 #define __rsnd_mod_call_init 0 #define __rsnd_mod_call_quit 1 #define __rsnd_mod_call_start 0 -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: MAX9860: new driver
by Peter Rosin 13 May '16

13 May '16

This is a driver for the MAX9860 Mono Audio Voice Codec. https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf This driver does not support sidetone since the DVST register field is backwards with the mute near the maximum level instead of the minimum. Signed-off-by: Peter Rosin <peda(a)axentia.se> --- MAINTAINERS | 7 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9860.c | 761 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/max9860.h | 162 ++++++++++ 5 files changed, 938 insertions(+) create mode 100644 sound/soc/codecs/max9860.c create mode 100644 sound/soc/codecs/max9860.h I have yet to find a good spot to add scanning for a mute TLV value needed for the backwards sidetone register field (DVST). Hence, still no sidetone support... New in v2: - Add comment about fall through when Integer Clock Mode is not possible. - Drop export of max9860_probe. - Ignore clk docs and read the mclk rate w/o enabling the clock. - Manage the DVDDIO supply. Cheers, Peter diff --git a/MAINTAINERS b/MAINTAINERS index a727d9959ecd..0803396b10f4 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -7000,6 +7000,13 @@ F: Documentation/devicetree/bindings/i2c/max6697.txt F: drivers/hwmon/max6697.c F: include/linux/platform_data/max6697.h +MAX9860 MONO AUDIO VOICE CODEC DRIVER +M: Peter Rosin <peda(a)axentia.se> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/max9860.txt +F: sound/soc/codecs/max9860* + MAXIM MUIC CHARGER DRIVERS FOR EXYNOS BASED BOARDS M: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> L: linux-pm(a)vger.kernel.org diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7ef3a0c16478..241a23c9aa9f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -83,6 +83,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C select SND_SOC_MAX9850 if I2C + select SND_SOC_MAX9860 if I2C select SND_SOC_MAX9768 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX @@ -534,6 +535,11 @@ config SND_SOC_MAX98926 config SND_SOC_MAX9850 tristate +config SND_SOC_MAX9860 + tristate "Maxim MAX9860 Mono Audio Voice Codec" + depends on I2C + select REGMAP_I2C + config SND_SOC_PCM1681 tristate "Texas Instruments PCM1681 CODEC" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 185a712a7fe7..e435f4df1787 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -78,6 +78,7 @@ snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o snd-soc-max9850-objs := max9850.o +snd-soc-max9860-objs := max9860.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o snd-soc-nau8825-objs := nau8825.o @@ -287,6 +288,7 @@ obj-$(CONFIG_SND_SOC_MAX9867) += snd-soc-max9867.o obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX98926) += snd-soc-max98926.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MAX9860) += snd-soc-max9860.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o diff --git a/sound/soc/codecs/max9860.c b/sound/soc/codecs/max9860.c new file mode 100644 index 000000000000..ee30832b0afb --- /dev/null +++ b/sound/soc/codecs/max9860.c @@ -0,0 +1,761 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf + * + * The driver does not support sidetone since the DVST register field is + * backwards with the mute near the maximum level instead of the minimum. + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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. + * + * 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 <linux/init.h> +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/kernel.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/regulator/consumer.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> + +#include "max9860.h" + +struct max9860_priv { + struct regmap *regmap; + struct regulator *dvddio; + struct notifier_block dvddio_nb; + u8 psclk; + unsigned long pclk_rate; + int fmt; +}; + +static int max9860_dvddio_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct max9860_priv *max9860 = container_of(nb, struct max9860_priv, + dvddio_nb); + if (event & REGULATOR_EVENT_DISABLE) { + regcache_mark_dirty(max9860->regmap); + regcache_cache_only(max9860->regmap, true); + } + + return 0; +} + +static const struct reg_default max9860_reg_defaults[] = { + { MAX9860_PWRMAN, 0x00 }, + { MAX9860_INTEN, 0x00 }, + { MAX9860_SYSCLK, 0x00 }, + { MAX9860_AUDIOCLKHIGH, 0x00 }, + { MAX9860_AUDIOCLKLOW, 0x00 }, + { MAX9860_IFC1A, 0x00 }, + { MAX9860_IFC1B, 0x00 }, + { MAX9860_VOICEFLTR, 0x00 }, + { MAX9860_DACATTN, 0x00 }, + { MAX9860_ADCLEVEL, 0x00 }, + { MAX9860_DACGAIN, 0x00 }, + { MAX9860_MICGAIN, 0x00 }, + { MAX9860_MICADC, 0x00 }, + { MAX9860_NOISEGATE, 0x00 }, +}; + +static bool max9860_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + case MAX9860_REVISION: + return true; + } + + return false; +} + +static bool max9860_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTEN ... MAX9860_MICGAIN: + case MAX9860_MICADC ... MAX9860_PWRMAN: + return true; + } + + return false; +} + +static bool max9860_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + case MAX9860_MICREADBACK: + return true; + } + + return false; +} + +static bool max9860_precious(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX9860_INTRSTATUS: + return true; + } + + return false; +} + +const struct regmap_config max9860_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = max9860_readable, + .writeable_reg = max9860_writeable, + .volatile_reg = max9860_volatile, + .precious_reg = max9860_precious, + + .max_register = MAX9860_MAX_REGISTER, + .reg_defaults = max9860_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(max9860_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const DECLARE_TLV_DB_SCALE(dva_tlv, -9100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dvg_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -1200, 100, 0); +static const DECLARE_TLV_DB_RANGE(pam_tlv, + 0, MAX9860_PAM_MAX - 1, TLV_DB_SCALE_ITEM(-2000, 2000, 1), + MAX9860_PAM_MAX, MAX9860_PAM_MAX, TLV_DB_SCALE_ITEM(3000, 0, 0)); +static const DECLARE_TLV_DB_SCALE(pgam_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(anth_tlv, -7600, 400, 1); +static const DECLARE_TLV_DB_SCALE(agcth_tlv, -1800, 100, 0); + +static const char * const agchld_text[] = { + "AGC Disabled", "50ms", "100ms", "400ms" +}; + +static SOC_ENUM_SINGLE_DECL(agchld_enum, MAX9860_MICADC, + MAX9860_AGCHLD_SHIFT, agchld_text); + +static const char * const agcsrc_text[] = { + "Left ADC", "Left/Right ADC" +}; + +static SOC_ENUM_SINGLE_DECL(agcsrc_enum, MAX9860_MICADC, + MAX9860_AGCSRC_SHIFT, agcsrc_text); + +static const char * const agcatk_text[] = { + "3ms", "12ms", "50ms", "200ms" +}; + +static SOC_ENUM_SINGLE_DECL(agcatk_enum, MAX9860_MICADC, + MAX9860_AGCATK_SHIFT, agcatk_text); + +static const char * const agcrls_text[] = { + "78ms", "156ms", "312ms", "625ms", + "1.25s", "2.5s", "5s", "10s" +}; + +static SOC_ENUM_SINGLE_DECL(agcrls_enum, MAX9860_MICADC, + MAX9860_AGCRLS_SHIFT, agcrls_text); + +static const char * const filter_text[] = { + "Disabled", + "Elliptical HP 217Hz notch (16kHz)", + "Butterworth HP 500Hz (16kHz)", + "Elliptical HP 217Hz notch (8kHz)", + "Butterworth HP 500Hz (8kHz)", + "Butterworth HP 200Hz (48kHz)" +}; + +static SOC_ENUM_SINGLE_DECL(avflt_enum, MAX9860_VOICEFLTR, + MAX9860_AVFLT_SHIFT, filter_text); + +static SOC_ENUM_SINGLE_DECL(dvflt_enum, MAX9860_VOICEFLTR, + MAX9860_DVFLT_SHIFT, filter_text); + +static const struct snd_kcontrol_new max9860_controls[] = { +SOC_SINGLE_TLV("Master Playback Volume", MAX9860_DACATTN, + MAX9860_DVA_SHIFT, MAX9860_DVA_MUTE, 1, dva_tlv), +SOC_SINGLE_TLV("DAC Gain Volume", MAX9860_DACGAIN, + MAX9860_DVG_SHIFT, MAX9860_DVG_MAX, 0, dvg_tlv), +SOC_DOUBLE_TLV("Line Capture Volume", MAX9860_ADCLEVEL, + MAX9860_ADCLL_SHIFT, MAX9860_ADCRL_SHIFT, MAX9860_ADCxL_MIN, 1, + adc_tlv), + +SOC_ENUM("AGC Hold Time", agchld_enum), +SOC_ENUM("AGC/Noise Gate Source", agcsrc_enum), +SOC_ENUM("AGC Attack Time", agcatk_enum), +SOC_ENUM("AGC Release Time", agcrls_enum), + +SOC_SINGLE_TLV("Noise Gate Threshold Volume", MAX9860_NOISEGATE, + MAX9860_ANTH_SHIFT, MAX9860_ANTH_MAX, 0, anth_tlv), +SOC_SINGLE_TLV("AGC Signal Threshold Volume", MAX9860_NOISEGATE, + MAX9860_AGCTH_SHIFT, MAX9860_AGCTH_MIN, 1, agcth_tlv), + +SOC_SINGLE_TLV("Mic PGA Volume", MAX9860_MICGAIN, + MAX9860_PGAM_SHIFT, MAX9860_PGAM_MIN, 1, pgam_tlv), +SOC_SINGLE_TLV("Mic Preamp Volume", MAX9860_MICGAIN, + MAX9860_PAM_SHIFT, MAX9860_PAM_MAX, 0, pam_tlv), + +SOC_ENUM("ADC Filter", avflt_enum), +SOC_ENUM("DAC Filter", dvflt_enum), +}; + +static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("MICL"), +SND_SOC_DAPM_INPUT("MICR"), + +SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), +SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), + +SND_SOC_DAPM_OUTPUT("OUT"), + +SND_SOC_DAPM_SUPPLY("Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_REGULATOR_SUPPLY("AVDD", 0, 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DVDD", 0, 0), +SND_SOC_DAPM_CLOCK_SUPPLY("mclk"), +}; + +static const struct snd_soc_dapm_route max9860_dapm_routes[] = { + { "ADCL", NULL, "MICL" }, + { "ADCR", NULL, "MICR" }, + { "AIFOUTL", NULL, "ADCL" }, + { "AIFOUTR", NULL, "ADCR" }, + + { "DAC", NULL, "AIFINL" }, + { "DAC", NULL, "AIFINR" }, + { "OUT", NULL, "DAC" }, + + { "Supply", NULL, "AVDD" }, + { "Supply", NULL, "DVDD" }, + { "Supply", NULL, "mclk" }, + + { "DAC", NULL, "Supply" }, + { "ADCL", NULL, "Supply" }, + { "ADCR", NULL, "Supply" }, +}; + +static int max9860_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBS_CFS: + return 0; + + default: + return -EINVAL; + } +} + +static int max9860_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 max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + u8 master; + u8 ifc1a = 0; + u8 ifc1b = 0; + u8 sysclk = 0; + unsigned long n; + int ret; + + dev_dbg(codec->dev, "hw_params %u Hz, %u channels\n", + params_rate(params), + params_channels(params)); + + if (params_channels(params) == 2) + ifc1b |= MAX9860_ST; + + switch (max9860->fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + case SND_SOC_DAIFMT_CBM_CFM: + master = MAX9860_MASTER; + break; + default: + return -EINVAL; + } + ifc1a |= master; + + if (master) { + if (params_width(params) * params_channels(params) > 48) + ifc1b |= MAX9860_BSEL_64X; + else + ifc1b |= MAX9860_BSEL_48X; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ifc1a |= MAX9860_DDLY; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_LEFT_J: + ifc1a |= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_DSP_A: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_A works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_DDLY | MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + ifc1b |= MAX9860_ADLY; + break; + case SND_SOC_DAIFMT_DSP_B: + if (params_width(params) != 16) { + dev_err(codec->dev, + "DSP_B works for 16 bits per sample only.\n"); + return -EINVAL; + } + ifc1a |= MAX9860_WCI | MAX9860_HIZ | MAX9860_TDM; + break; + default: + return -EINVAL; + } + + switch (max9860->fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_NB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + break; + case SND_SOC_DAIFMT_IB_IF: + switch (max9860->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + return -EINVAL; + } + ifc1a ^= MAX9860_WCI; + /* fall through */ + case SND_SOC_DAIFMT_IB_NF: + ifc1a ^= MAX9860_DBCI; + ifc1b ^= MAX9860_ABCI; + break; + default: + return -EINVAL; + } + + dev_dbg(codec->dev, "IFC1A %02x\n", ifc1a); + ret = regmap_write(max9860->regmap, MAX9860_IFC1A, ifc1a); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1A: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "IFC1B %02x\n", ifc1b); + ret = regmap_write(max9860->regmap, MAX9860_IFC1B, ifc1b); + if (ret) { + dev_err(codec->dev, "Failed to set IFC1B: %d\n", ret); + return ret; + } + + /* + * Check if Integer Clock Mode is possible, but avoid it in slave mode + * since we then do not know if lrclk is derived from pclk and the + * datasheet mentions that the frequencies have to match exactly in + * order for this to work. + */ + if (params_rate(params) == 8000 || params_rate(params) == 16000) { + if (master) { + switch (max9860->pclk_rate) { + case 12000000: + sysclk = MAX9860_FREQ_12MHZ; + break; + case 13000000: + sysclk = MAX9860_FREQ_13MHZ; + break; + case 19200000: + sysclk = MAX9860_FREQ_19_2MHZ; + break; + default: + /* + * Integer Clock Mode not possible. Leave + * sysclk at zero and fall through to the + * code below for PLL mode. + */ + break; + } + + if (sysclk && params_rate(params) == 16000) + sysclk |= MAX9860_16KHZ; + } + } + + /* + * Largest possible n: + * 65536 * 96 * 48kHz / 10MHz -> 30199 + * Smallest possible n: + * 65536 * 96 * 8kHz / 20MHz -> 2517 + * Both fit nicely in the available 15 bits, no need to apply any mask. + */ + n = DIV_ROUND_CLOSEST_ULL(65536ULL * 96 * params_rate(params), + max9860->pclk_rate); + + if (!sysclk) { + /* PLL mode */ + if (params_rate(params) > 24000) + sysclk |= MAX9860_16KHZ; + + if (!master) + n |= 1; /* trigger rapid pll lock mode */ + } + + sysclk |= max9860->psclk; + dev_dbg(codec->dev, "SYSCLK %02x\n", sysclk); + ret = regmap_write(max9860->regmap, + MAX9860_SYSCLK, sysclk); + if (ret) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + dev_dbg(codec->dev, "N %lu\n", n); + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKHIGH, n >> 8); + if (ret) { + dev_err(codec->dev, "Failed to set NHI: %d\n", ret); + return ret; + } + ret = regmap_write(max9860->regmap, + MAX9860_AUDIOCLKLOW, n & 0xff); + if (ret) { + dev_err(codec->dev, "Failed to set NLO: %d\n", ret); + return ret; + } + + if (!master) { + dev_dbg(codec->dev, "Enable PLL\n"); + ret = regmap_update_bits(max9860->regmap, MAX9860_AUDIOCLKHIGH, + MAX9860_PLL, MAX9860_PLL); + if (ret) { + dev_err(codec->dev, "Failed to enable PLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int max9860_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct max9860_priv *max9860 = snd_soc_codec_get_drvdata(codec); + + max9860->fmt = fmt; + return 0; +} + +static const struct snd_soc_dai_ops max9860_dai_ops = { + .startup = max9860_dai_startup, + .hw_params = max9860_hw_params, + .set_fmt = max9860_set_fmt, +}; + +static struct snd_soc_dai_driver max9860_dai = { + .name = "max9860-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + }, + .ops = &max9860_dai_ops, + .symmetric_rates = 1, +}; + +static int max9860_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct max9860_priv *max9860 = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, MAX9860_SHDN); + if (ret) { + dev_err(codec->dev, "Failed to remove SHDN: %d\n", ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(max9860->regmap, MAX9860_PWRMAN, + MAX9860_SHDN, 0); + if (ret) { + dev_err(codec->dev, "Failed to request SHDN: %d\n", + ret); + return ret; + } + break; + } + + return 0; +} + +static struct snd_soc_codec_driver max9860_codec_driver = { + .set_bias_level = max9860_set_bias_level, + .idle_bias_off = true, + + .controls = max9860_controls, + .num_controls = ARRAY_SIZE(max9860_controls), + .dapm_widgets = max9860_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max9860_dapm_widgets), + .dapm_routes = max9860_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(max9860_dapm_routes), +}; + +#ifdef CONFIG_PM +static int max9860_suspend(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, MAX9860_PSCLK_OFF); + if (ret) { + dev_err(dev, "Failed to disable clock: %d\n", ret); + return ret; + } + + regulator_disable(max9860->dvddio); + + return 0; +} + +static int max9860_resume(struct device *dev) +{ + struct max9860_priv *max9860 = dev_get_drvdata(dev); + int ret; + + ret = regulator_enable(max9860->dvddio); + if (ret) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + regcache_cache_only(max9860->regmap, false); + ret = regcache_sync(max9860->regmap); + if (ret) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(max9860->regmap, MAX9860_SYSCLK, + MAX9860_PSCLK, max9860->psclk); + if (ret) { + dev_err(dev, "Failed to enable clock: %d\n", ret); + return ret; + } + + return 0; +} +#endif + +const struct dev_pm_ops max9860_pm_ops = { + SET_RUNTIME_PM_OPS(max9860_suspend, max9860_resume, NULL) +}; + +static int max9860_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860; + int ret; + struct clk *mclk; + unsigned long mclk_rate; + int i; + int intr; + + max9860 = devm_kzalloc(dev, sizeof(struct max9860_priv), GFP_KERNEL); + if (!max9860) + return -ENOMEM; + + max9860->dvddio = devm_regulator_get(dev, "DVDDIO"); + if (IS_ERR(max9860->dvddio)) { + ret = PTR_ERR(max9860->dvddio); + dev_err(dev, "Failed to get DVDDIO supply: %d\n", ret); + return ret; + } + + max9860->dvddio_nb.notifier_call = max9860_dvddio_event; + + ret = regulator_register_notifier(max9860->dvddio, &max9860->dvddio_nb); + if (ret) + dev_err(dev, "Failed to register DVDDIO notifier: %d\n", ret); + + ret = regulator_enable(max9860->dvddio); + if (ret != 0) { + dev_err(dev, "Failed to enable DVDDIO: %d\n", ret); + return ret; + } + + max9860->regmap = devm_regmap_init_i2c(i2c, &max9860_regmap); + if (IS_ERR(max9860->regmap)) { + ret = PTR_ERR(max9860->regmap); + goto err_regulator; + } + + dev_set_drvdata(dev, max9860); + + /* + * mclk has to be in the 10MHz to 60MHz range. + * psclk is used to scale mclk into pclk so that + * pclk is in the 10MHz to 20MHz range. + */ + mclk = clk_get(dev, "mclk"); + + if (IS_ERR(mclk)) { + ret = PTR_ERR(mclk); + dev_err(dev, "Failed to get MCLK: %d\n", ret); + goto err_regulator; + } + + mclk_rate = clk_get_rate(mclk); + clk_put(mclk); + + if (mclk_rate > 60000000 || mclk_rate < 10000000) { + dev_err(dev, "Bad mclk %luHz (needs 10MHz - 60MHz)\n", + mclk_rate); + ret = -EINVAL; + goto err_regulator; + } + if (mclk_rate >= 40000000) + max9860->psclk = 3; + else if (mclk_rate >= 20000000) + max9860->psclk = 2; + else + max9860->psclk = 1; + max9860->pclk_rate = mclk_rate >> (max9860->psclk - 1); + max9860->psclk <<= MAX9860_PSCLK_SHIFT; + dev_dbg(dev, "mclk %lu pclk %lu\n", mclk_rate, max9860->pclk_rate); + + regcache_cache_bypass(max9860->regmap, true); + for (i = 0; i < max9860_regmap.num_reg_defaults; ++i) { + ret = regmap_write(max9860->regmap, + max9860_regmap.reg_defaults[i].reg, + max9860_regmap.reg_defaults[i].def); + if (ret) { + dev_err(dev, "Failed to initialize register %u: %d\n", + max9860_regmap.reg_defaults[i].reg, ret); + goto err_regulator; + } + } + regcache_cache_bypass(max9860->regmap, false); + + ret = regmap_read(max9860->regmap, MAX9860_INTRSTATUS, &intr); + if (ret) { + dev_err(dev, "Failed to clear INTRSTATUS: %d\n", ret); + goto err_regulator; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &max9860_codec_driver, + &max9860_dai, 1); + if (ret) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_regulator: + regulator_disable(max9860->dvddio); + return ret; +} + +static int max9860_remove(struct i2c_client *i2c) +{ + struct device *dev = &i2c->dev; + struct max9860_priv *max9860 = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + regulator_disable(max9860->dvddio); + return 0; +} + +static const struct i2c_device_id max9860_i2c_id[] = { + { "max9860", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, max9860_i2c_id); + +static const struct of_device_id max9860_of_match[] = { + { .compatible = "maxim,max9860", }, + { } +}; +MODULE_DEVICE_TABLE(of, max9860_of_match); + +static struct i2c_driver max9860_i2c_driver = { + .probe = max9860_probe, + .remove = max9860_remove, + .id_table = max9860_i2c_id, + .driver = { + .name = "max9860", + .of_match_table = max9860_of_match, + .pm = &max9860_pm_ops, + }, +}; + +module_i2c_driver(max9860_i2c_driver); + +MODULE_DESCRIPTION("ASoC MAX9860 Mono Audio Voice Codec driver"); +MODULE_AUTHOR("Peter Rosin <peda(a)axentia.se>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/max9860.h b/sound/soc/codecs/max9860.h new file mode 100644 index 000000000000..22041bd67a7d --- /dev/null +++ b/sound/soc/codecs/max9860.h @@ -0,0 +1,162 @@ +/* + * Driver for the MAX9860 Mono Audio Voice Codec + * + * Author: Peter Rosin <peda(a)axentia.s> + * Copyright 2016 Axentia Technologies + * + * 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. + * + * 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. + */ + +#ifndef _SND_SOC_MAX9860 +#define _SND_SOC_MAX9860 + +#define MAX9860_INTRSTATUS 0x00 +#define MAX9860_MICREADBACK 0x01 +#define MAX9860_INTEN 0x02 +#define MAX9860_SYSCLK 0x03 +#define MAX9860_AUDIOCLKHIGH 0x04 +#define MAX9860_AUDIOCLKLOW 0x05 +#define MAX9860_IFC1A 0x06 +#define MAX9860_IFC1B 0x07 +#define MAX9860_VOICEFLTR 0x08 +#define MAX9860_DACATTN 0x09 +#define MAX9860_ADCLEVEL 0x0a +#define MAX9860_DACGAIN 0x0b +#define MAX9860_MICGAIN 0x0c +#define MAX9860_RESERVED 0x0d +#define MAX9860_MICADC 0x0e +#define MAX9860_NOISEGATE 0x0f +#define MAX9860_PWRMAN 0x10 +#define MAX9860_REVISION 0xff + +#define MAX9860_MAX_REGISTER 0xff + +/* INTRSTATUS */ +#define MAX9860_CLD 0x80 +#define MAX9860_SLD 0x40 +#define MAX9860_ULK 0x20 + +/* MICREADBACK */ +#define MAX9860_NG 0xe0 +#define MAX9860_AGC 0x1f + +/* INTEN */ +#define MAX9860_ICLD 0x80 +#define MAX9860_ISLD 0x40 +#define MAX9860_IULK 0x20 + +/* SYSCLK */ +#define MAX9860_PSCLK 0x30 +#define MAX9860_PSCLK_OFF 0x00 +#define MAX9860_PSCLK_SHIFT 4 +#define MAX9860_FREQ 0x06 +#define MAX9860_FREQ_NORMAL 0x00 +#define MAX9860_FREQ_12MHZ 0x02 +#define MAX9860_FREQ_13MHZ 0x04 +#define MAX9860_FREQ_19_2MHZ 0x06 +#define MAX9860_16KHZ 0x01 + +/* AUDIOCLKHIGH */ +#define MAX9860_PLL 0x80 +#define MAX9860_NHI 0x7f + +/* AUDIOCLKLOW */ +#define MAX9860_NLO 0xff + +/* IFC1A */ +#define MAX9860_MASTER 0x80 +#define MAX9860_WCI 0x40 +#define MAX9860_DBCI 0x20 +#define MAX9860_DDLY 0x10 +#define MAX9860_HIZ 0x08 +#define MAX9860_TDM 0x04 + +/* IFC1B */ +#define MAX9860_ABCI 0x20 +#define MAX9860_ADLY 0x10 +#define MAX9860_ST 0x08 +#define MAX9860_BSEL 0x07 +#define MAX9860_BSEL_OFF 0x00 +#define MAX9860_BSEL_64X 0x01 +#define MAX9860_BSEL_48X 0x02 +#define MAX9860_BSEL_PCLK_2 0x04 +#define MAX9860_BSEL_PCLK_4 0x05 +#define MAX9860_BSEL_PCLK_8 0x06 +#define MAX9860_BSEL_PCLK_16 0x07 + +/* VOICEFLTR */ +#define MAX9860_AVFLT 0xf0 +#define MAX9860_AVFLT_SHIFT 4 +#define MAX9860_AVFLT_COUNT 6 +#define MAX9860_DVFLT 0x0f +#define MAX9860_DVFLT_SHIFT 0 +#define MAX9860_DVFLT_COUNT 6 + +/* DACATTN */ +#define MAX9860_DVA 0xfe +#define MAX9860_DVA_SHIFT 1 +#define MAX9860_DVA_MUTE 0x5e + +/* ADCLEVEL */ +#define MAX9860_ADCRL 0xf0 +#define MAX9860_ADCRL_SHIFT 4 +#define MAX9860_ADCLL 0x0f +#define MAX9860_ADCLL_SHIFT 0 +#define MAX9860_ADCxL_MIN 15 + +/* DACGAIN */ +#define MAX9860_DVG 0x60 +#define MAX9860_DVG_SHIFT 5 +#define MAX9860_DVG_MAX 3 +#define MAX9860_DVST 0x1f +#define MAX9860_DVST_SHIFT 0 +#define MAX9860_DVST_MIN 31 + +/* MICGAIN */ +#define MAX9860_PAM 0x60 +#define MAX9860_PAM_SHIFT 5 +#define MAX9860_PAM_MAX 3 +#define MAX9860_PGAM 0x1f +#define MAX9860_PGAM_SHIFT 0 +#define MAX9860_PGAM_MIN 20 + +/* MICADC */ +#define MAX9860_AGCSRC 0x80 +#define MAX9860_AGCSRC_SHIFT 7 +#define MAX9860_AGCSRC_COUNT 2 +#define MAX9860_AGCRLS 0x70 +#define MAX9860_AGCRLS_SHIFT 4 +#define MAX9860_AGCRLS_COUNT 8 +#define MAX9860_AGCATK 0x0c +#define MAX9860_AGCATK_SHIFT 2 +#define MAX9860_AGCATK_COUNT 4 +#define MAX9860_AGCHLD 0x03 +#define MAX9860_AGCHLD_OFF 0x00 +#define MAX9860_AGCHLD_SHIFT 0 +#define MAX9860_AGCHLD_COUNT 4 + +/* NOISEGATE */ +#define MAX9860_ANTH 0xf0 +#define MAX9860_ANTH_SHIFT 4 +#define MAX9860_ANTH_MAX 15 +#define MAX9860_AGCTH 0x0f +#define MAX9860_AGCTH_SHIFT 0 +#define MAX9860_AGCTH_MIN 15 + +/* PWRMAN */ +#define MAX9860_SHDN 0x80 +#define MAX9860_DACEN 0x08 +#define MAX9860_DACEN_SHIFT 3 +#define MAX9860_ADCLEN 0x02 +#define MAX9860_ADCLEN_SHIFT 1 +#define MAX9860_ADCREN 0x01 +#define MAX9860_ADCREN_SHIFT 0 + +#endif /* _SND_SOC_MAX9860 */ -- 2.1.4

1 1

[alsa-devel] [PATCH] ALSA: au88x0: Fix zero clear of stream->resources
by Takashi Iwai 12 May '16

12 May '16

There are a few calls of memset() to stream->resources, but they all are called in a wrong size, sizeof(unsigned char) * VORTEX_RESOURCE_LAST, while this field is a u32 array. This may leave the memories not zero-cleared. Fix it by replacing them with a simpler sizeof(stream->resources) instead. Reported-by: David Binderman <dcb314(a)hotmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/au88x0/au88x0_core.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 4667c3232b7f..4a054d720112 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2151,8 +2151,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if (stream->type != VORTEX_PCM_A3D) { @@ -2162,7 +2161,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXIN)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } @@ -2175,8 +2174,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_A3D)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); dev_err(vortex->card->dev, "out of A3D sources. Sorry\n"); return -EBUSY; @@ -2290,8 +2288,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, VORTEX_RESOURCE_MIXOUT)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } if ((src[i] = @@ -2299,8 +2296,7 @@ vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, stream->resources, en, VORTEX_RESOURCE_SRC)) < 0) { memset(stream->resources, 0, - sizeof(unsigned char) * - VORTEX_RESOURCE_LAST); + sizeof(stream->resources)); return -EBUSY; } } -- 2.8.2

1 0

[alsa-devel] sound/pci/au88x0/au88x0_core.c: many calls to memset with not-plausible size ?
by David Binderman 12 May '16

12 May '16

Hello there, sound/pci/au88x0/au88x0_core.c:2153:6: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] Source code is memset(stream->resources, 0, sizeof(unsigned char) * VORTEX_RESOURCE_LAST); but sound/pci/au88x0/au88x0.h: u32 resources[5]; Something like memset(stream->resources, 0, sizeof(stream->resources[0]) * VORTEX_RESOURCE_LAST); Some duplicates: sound/pci/au88x0/au88x0_core.c:2163:7: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2177:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2292:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] sound/pci/au88x0/au88x0_core.c:2301:5: warning: ‘memset’ used with length equal to number of elements without multiplication by element size [-Wmemset-elt-size] Regards David Binderman

2 2

[alsa-devel] [PATCH] ASoC: wm8985: rework and fix the clock calculation
by Petr Kulhavy 12 May '16

12 May '16

The clock calculation has several issues: 1) if PLL is used in master mode the BCLK output runs at double the speed 2) de-facto only 44.1kHz and 48kHz sampling rates are supported, other rates like 8kHz, 12kHz, 24kHz fail to find the proper BCLK divider 3) the wm8985->sysclk variable has a misleading name and is used wrongly in the clock calculation in wm8985_hw_params() which is the root cause for (1) 4) wm8985->bclk is used only in wm8985_hw_params() and therefore no need to store it in the wm8985_priv structure Therefore the clock calculation is rewritten in more clean and proper way: - move wm8985_priv->bclk as a local variable into mw8985_hw_params() - new variable wm8985_priv->pllout holds the actual frequency that is input to the MCLKDIV post-divider - move wm8985_priv->sysclk as a local variable into mw8985_hw_params() - sysclk is now always calculated as 256 * fs - the MCLKDIV is looked up as pllout/sysclk - fs_ratios[] is replaced by simpler mclk_divs[] lookup table With this patch all rates: 8, 11.025, 12, 16, 22.05, 24, 32, 44.1 and 48kHz work properly and generate the correct BCLK. Signed-off-by: Petr Kulhavy <petr(a)barix.com> --- sound/soc/codecs/wm8985.c | 59 ++++++++++++++++++++++++----------------------- 1 file changed, 30 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 18f2babe1090..628127aa3c96 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -181,22 +181,11 @@ static const int volume_update_regs[] = { struct wm8985_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8985_NUM_SUPPLIES]; - unsigned int sysclk; - unsigned int bclk; + unsigned int pllout; /* input rate to the MCLKDIV divider */ }; -static const struct { - int div; - int ratio; -} fs_ratios[] = { - { 10, 128 }, - { 15, 192 }, - { 20, 256 }, - { 30, 384 }, - { 40, 512 }, - { 60, 768 }, - { 80, 1024 }, - { 120, 1536 } +static const int mclk_divs[] = { + 10, 15, 20, 30, 40, 60, 80, 120 }; static const int srates[] = { 48000, 32000, 24000, 16000, 12000, 8000 }; @@ -693,15 +682,18 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec; struct wm8985_priv *wm8985; u16 blen, srate_idx; - unsigned int tmp; int srate_best; + unsigned int bclk; /* bit clock matching the current params */ + unsigned int sysclk; /* the actual sysclk after post division */ codec = dai->codec; wm8985 = snd_soc_codec_get_drvdata(codec); - wm8985->bclk = snd_soc_params_to_bclk(params); - if ((int)wm8985->bclk < 0) - return wm8985->bclk; + /* always use 256 * fs in order to get best filter quality */ + sysclk = 256 * params_rate(params); + bclk = snd_soc_params_to_bclk(params); + if ((int)bclk < 0) + return bclk; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -742,29 +734,28 @@ static int wm8985_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, WM8985_ADDITIONAL_CONTROL, WM8985_SR_MASK, srate_idx << WM8985_SR_SHIFT); - dev_dbg(dai->dev, "Target BCLK = %uHz\n", wm8985->bclk); - dev_dbg(dai->dev, "SYSCLK = %uHz\n", wm8985->sysclk); + dev_dbg(dai->dev, "Target BCLK = %uHz\n", bclk); + dev_dbg(dai->dev, "SYSCLK = %uHz\n", sysclk); - for (i = 0; i < ARRAY_SIZE(fs_ratios); ++i) { - if (wm8985->sysclk / params_rate(params) - == fs_ratios[i].ratio) + /* select the appropriate mclk divider */ + for (i = 0; i < ARRAY_SIZE(mclk_divs); ++i) { + if (wm8985->pllout / mclk_divs[i] * 10 + == sysclk) break; } - if (i == ARRAY_SIZE(fs_ratios)) { + if (i == ARRAY_SIZE(mclk_divs)) { dev_err(dai->dev, "Unable to configure MCLK ratio %u/%u\n", - wm8985->sysclk, params_rate(params)); + wm8985->pllout, sysclk); return -EINVAL; } - dev_dbg(dai->dev, "MCLK ratio = %dfs\n", fs_ratios[i].ratio); snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, WM8985_MCLKDIV_MASK, i << WM8985_MCLKDIV_SHIFT); /* select the appropriate bclk divider */ - tmp = (wm8985->sysclk / fs_ratios[i].div) * 10; for (i = 0; i < ARRAY_SIZE(bclk_divs); ++i) { - if (wm8985->bclk == tmp / bclk_divs[i]) + if (bclk == sysclk / bclk_divs[i]) break; } @@ -786,6 +777,10 @@ struct pll_div { }; #define FIXED_PLL_SIZE ((1ULL << 24) * 10) +/* + * source = MCLK input of the chip + * target = the f2 coming out of the PLL before /4 divider + */ static int pll_factors(struct pll_div *pll_div, unsigned int target, unsigned int source) { @@ -872,17 +867,23 @@ static int wm8985_set_sysclk(struct snd_soc_dai *dai, WM8985_CLKSEL_MASK, 0); snd_soc_update_bits(codec, WM8985_POWER_MANAGEMENT_1, WM8985_PLLEN_MASK, 0); + wm8985->pllout = freq; break; case WM8985_CLKSRC_PLL: snd_soc_update_bits(codec, WM8985_CLOCK_GEN_CONTROL, WM8985_CLKSEL_MASK, WM8985_CLKSEL); + /* + * in order to run the PLL within the recommended 90MHz + * operating range the wm8985_set_pll() configures the PLL + * to output double the required frequency + */ + wm8985->pllout = 2 * freq; break; default: dev_err(dai->dev, "Unknown clock source %d\n", clk_id); return -EINVAL; } - wm8985->sysclk = freq; return 0; } -- 1.9.1

2 4

[alsa-devel] [PATCH 1/2] ASoC: nau8825: non-clock jack detection for power saving at standby
by John Hsu 12 May '16

12 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 | 281 +++++++++++++++++++++++++++++++++++---------- sound/soc/codecs/nau8825.h | 9 +- 2 files changed, 226 insertions(+), 64 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..9386624 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), @@ -614,8 +623,10 @@ 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); + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. + */ return 0; } @@ -642,6 +653,22 @@ 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 +683,67 @@ 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); + + /* Mask all interruptions except jack insertion interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0xfffe); + + /* Enable insertion interruption only and bypass de-bounce circuit */ + 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) @@ -753,7 +841,10 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) struct regmap *regmap = nau8825->regmap; int active_irq, clear_irq = 0, event = 0, event_mask = 0; - regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + if (regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq)) { + dev_err(nau8825->dev, "failed to clear interruption\n"); + return IRQ_NONE; + } if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == NAU8825_JACK_EJECTION_DETECTED) { @@ -789,6 +880,26 @@ 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 +1049,8 @@ 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, @@ -958,11 +1069,10 @@ 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. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Move these configuration about inter- + * ruption at auto mode to auto irq setup function. */ - regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); - nau8825_restart_jack_detection(nau8825->regmap); return 0; } @@ -1134,7 +1244,26 @@ static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, struct regmap *regmap = nau8825->regmap; int ret; + if (!nau8825_is_jack_inserted(nau8825->regmap) && + clk_id != NAU8825_CLK_DIS) { + /* For power saving less 1mW, the jack type detection inter- + * ruption changes to non-clock architecture. Therefore, the + * clock should be disabled and not allowed to config any clock + * when no headset connected. + */ + dev_warn(nau8825->dev, "Souldn't enable any clock when no headset connected\n"); + return 0; + } + switch (clk_id) { + case NAU8825_CLK_DIS: + /* Clock provided externally and disable internal VCO clock */ + 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); + + break; case NAU8825_CLK_MCLK: regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); @@ -1209,6 +1338,30 @@ 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 insertion interruption only and bypass de-bounce circuit */ + 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); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, + NAU8825_IRQ_INSERT_DIS, 0); + + return 0; +} + static int nau8825_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -1238,11 +1391,22 @@ 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 +1416,50 @@ 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); + if (!nau8825_is_jack_inserted(nau8825->regmap)) { + /* Check the jack plug status directly. If the headset is un- + * plugged during S3 when the chip has no power, there will + * be no jack detection irq event after restarting jack detec- + * tion, because the chip just thinks no headset has ever + * been plugged in. + */ + nau8825_eject_jack(nau8825); + snd_soc_jack_report(nau8825->jack, 0, SND_JACK_HEADSET); + } + 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), @@ -1351,29 +1553,13 @@ 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. + /* Change jack type detection interruption to non-clock architecture + * for power saving less 1mW. Because resume and initiation all need + * to enable the mechanism, move these configuration to resume setup + * function. */ - 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, @@ -1442,36 +1628,6 @@ 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 +1654,6 @@ 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 12

[alsa-devel] i.MX6 SGTL5000 hangs with EIO error
by Ross Wille 12 May '16

12 May '16

Hello, I am doing an ALSA capture on a Wandboard Quad (i.MX6 quad core with SGTL5000 codec chip) from the LINEIN1 jack. I'm using a 4.6.0-rc3-armv7-x0 kernel. Sometimes when the audio glitches (for example, when I plug/unplug the audio cable or adjust my signal generator) the snd_pcm_readi() function will start returning -5 (EIO). Once this happens, the only way I've been able to recover is to reboot the computer. Calling snd_pcm_close(), snd_pcm_prepare(), snd_pcm_start(), etc. doesn't help. When in this state, running arecord returns IO errors as well. It's interesting that, on rare occasions, I must do a power cycle in order to recover. When a reboot is not effective I've noticed that the capture device doesn't appear in /proc/asound/devices. I don't believe my specific ALSA settings are important, but I'm calling snd_pcm_readi() with ALSA set to a sample rate of 48000, format of S16_LE, channels=1, frames per period of 960 (20 mS periods), and 4 periods per buffer. This same problem happens on two different Wandboards, so I don't think it's a defective board or chip. It has happened on older kernels as well. Any ideas? Thank you! Wille

1 0

[alsa-devel] Intel Cherry Trail and NAU8824 codec
by TheKit 12 May '16

12 May '16

Hello, I'm trying to get sound working in Linux on Cube iWork8 Ultimate tablet. It has Windows/Android dualboot, but no kernel sources for Android, unfortunately. From dmesg I see that NAU8824 codec is used, which I can't find any drivers for, except proposed patch that wasn't merged: https://lkml.org/lkml/2015/3/4/765 But, as I understand, additional code (like https://github.com/kirananto/ZENFONE2/blob/master/sound/soc/intel/board/cht…) is needed to bridge it with Cherry Trail SoC. Any suggestions how to go about this? Relevant lines of dmesg from stock Android kernel (where sound works): <6>[ 8.665477] nau8824_i2c_probe enter <3>[ 8.666609] snd_intel_sst: sst_platform_get_resources_edk <3>[ 8.672489] snd_intel_sst: LPE base: 0x91600000 size:0x200000snd_intel_sst: IRAM base: 0x916c0000 <3>[ 8.682557] snd_intel_sst: DRAM base: 0x91700000snd_intel_sst: SHIM base: 0x91740000 <3>[ 8.691298] snd_intel_sst: Mailbox base: 0x91744000snd_intel_sst: DDR base: 0x20000000 <6>[ 8.700428] snd_intel_sst: Enter: sst_audio_platform_init <6>[ 8.700646] snd_intel_sst: audio:set_cht_platform_config <6>[ 8.700900] snd_soc_sst_platform: sst_platform_probe is deferred <6>[ 8.700918] platform sst-platform: Driver sst-platform requests probe deferral <3>[ 8.700927] snd_intel_sst: board_name:Cherry Trail CR <6>[ 8.706611] snd_intel_sst: Registering machine device cht_nau88l24 <3>[ 8.707127] snd_intel_sst: Requesting FW fw_sst_22a8.bin now... <4>[ 8.713815] intel_sst_acpi 808622A8:00: Direct firmware load failed with error -2 <4>[ 8.724734] intel_sst_acpi 808622A8:00: Falling back to user helper <6>[ 8.789741] platform cht_nau88l24: Driver cht_nau88l24 requests probe deferral <6>[ 8.789967] cht_cr_nau88l24: Cherrytrail Machine Driver cht_nau8824 registerd <6>[ 9.077571] snd_soc_sst_platform: sst_platform_probe is deferred <3>[ 10.856689] sst-platform sst-platform: ASoC: no sink widget found for Audio Capture <3>[ 10.856693] sst-platform sst-platform: ASoC: Failed to add route media3_out -> direct -> Audio Capture <6>[ 10.873799] nau8824_codec_probe enter <6>[ 11.062658] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Headset-cpu-dai mapping ok <6>[ 11.062754] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Deepbuffer-cpu-dai mapping ok <6>[ 11.062902] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Voip-cpu-dai mapping ok <6>[ 11.063035] cht_cr_nau88l24: cht_hs_detection:Check hs insertion after 500 msec <6>[ 11.063035] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Lowlatency-cpu-dai mapping ok <6>[ 11.063383] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> Probe-cpu-dai mapping ok <6>[ 11.063596] cht_nau88l24 cht_nau88l24: nau8824-aif1 <-> ssp2-port mapping ok <6>[ 11.063638] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> snd-soc-dummy-dai mapping ok <6>[ 11.063674] cht_nau88l24 cht_nau88l24: snd-soc-dummy-dai <-> snd-soc-dummy-dai mapping ok <3>[ 11.063742] cht_nau88l24 cht_nau88l24: ASoC: Failed to add Headset Mic Switch: -16 <6>[ 11.095628] input: cherrytrailaud Headset Jack as /devices/platform/cht_nau88l24/sound/card1/input9 <6>[ 11.096265] cht_cr_nau88l24: snd_cht_mc_probe successful

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: add context information to tracepoints
by Takashi Sakamoto 11 May '16

11 May '16

In current implementation, packet processing is done in both of software IRQ contexts of IR/IT contexts and process contexts. This is usual interrupt handling of IR/IT context for 1394 OHCI. (in hardware IRQ context) irq_handler() (drivers/firewire/ohci.c) ->tasklet_schedule() (in software IRQ context) handle_it_packet() or handle_ir_packet_per_buffer() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() However, we have another chance for packet processing. It's done in PCM frame handling via ALSA PCM interfaces. (in process context) ioctl(i.e. SNDRV_PCM_IOCTL_HWSYNC) ->snd_pcm_hwsync() (sound/core/pcm_native.c) ->snd_pcm_update_hw_ptr() (sound/core/pcm_lib.c) ->snd_pcm_update_hw_ptr0() ->struct snd_pcm_ops.pointer() = amdtp_stream_pcm_pointer() ->fw_iso_context_flush_completions() (drivers/firewire/core-iso.c) ->struct fw_card_driver.flush_iso_completions() = ohci_flush_iso_completions() (drivers/firewire/ohci.c) ->flush_iso_completions() ->struct fw_iso_context.callback.sc() = out_stream_callback() or in_stream_callback() This design is for a better granularity of PCM pointer. When ioctl(2) is executed with some commands for ALSA PCM interface, queued packets are handled at first. Then, the latest number of handled PCM frames is reported. The number can represent PCM frames transferred in most near isochronous cycle. Current tracepoints include no information to distinguish running contexts. When tracing the interval of software IRQ context, this is not good. This commit adds more information for current context. Additionally, the index of packet processed in one context is added in a case that packet processing is executed in continuous context of the same kind, As a result, the output includes 11 fields with additional two fields to commit 0c95c1d6197f ("ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data"): 17131.9186: out_packet: 07 7494 ffc0 ffc1 00 000700c0 9001a496 058 45 1 13 17131.9186: out_packet: 07 7495 ffc0 ffc1 00 000700c8 9001ba00 058 46 1 14 17131.9186: out_packet: 07 7496 ffc0 ffc1 00 000700d0 9001ffff 002 47 1 15 17131.9189: out_packet: 07 7497 ffc0 ffc1 00 000700d0 9001d36a 058 00 0 00 17131.9189: out_packet: 07 7498 ffc0 ffc1 00 000700d8 9001e8d4 058 01 0 01 17131.9189: out_packet: 07 7499 ffc0 ffc1 00 000700e0 9001023e 058 02 0 00 17131.9206: in_packet: 07 7447 ffc1 ffc0 01 3f070072 9001783d 058 32 1 00 17131.9206: in_packet: 07 7448 ffc1 ffc0 01 3f070072 90ffffff 002 33 1 01 17131.9206: in_packet: 07 7449 ffc1 ffc0 01 3f07007a 900191a8 058 34 1 02 (Here, some common fields are omitted so that a line is within 80 characters.) The legend is: - The second of cycle scheduled for the packet - The count of cycle scheduled for the packet - The ID of node as source (hex) - The ID of node as destination (hex) - The value of isochronous channel - The first quadlet of CIP header (hex) - The second quadlet of CIP header (hex) - The number of included quadlets - The index of packet in a buffer maintained by this module - 0 in process context, 1 in IRQ context - The index of packet processed in the context Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream-trace.h | 28 ++++++++++++++++++++-------- sound/firewire/amdtp-stream.c | 14 ++++++++------ 2 files changed, 28 insertions(+), 14 deletions(-) diff --git a/sound/firewire/amdtp-stream-trace.h b/sound/firewire/amdtp-stream-trace.h index 425d1d7..1622579 100644 --- a/sound/firewire/amdtp-stream-trace.h +++ b/sound/firewire/amdtp-stream-trace.h @@ -14,8 +14,8 @@ #include <linux/tracepoint.h> TRACE_EVENT(in_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets), - TP_ARGS(s, cycles, cip_header, payload_quadlets), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, u32 cip_header[2], unsigned int payload_quadlets, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_quadlets, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -25,6 +25,8 @@ TRACE_EVENT(in_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -36,10 +38,12 @@ TRACE_EVENT(in_packet, __entry->cip_header0 = cip_header[0]; __entry->cip_header1 = cip_header[1]; __entry->payload_quadlets = payload_quadlets; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -48,12 +52,14 @@ TRACE_EVENT(in_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); TRACE_EVENT(out_packet, - TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length), - TP_ARGS(s, cycles, cip_header, payload_length), + TP_PROTO(const struct amdtp_stream *s, u32 cycles, __be32 *cip_header, unsigned int payload_length, unsigned int index), + TP_ARGS(s, cycles, cip_header, payload_length, index), TP_STRUCT__entry( __field(unsigned int, second) __field(unsigned int, cycle) @@ -63,6 +69,8 @@ TRACE_EVENT(out_packet, __field(u32, cip_header0) __field(u32, cip_header1) __field(unsigned int, payload_quadlets) + __field(unsigned int, packet_index) + __field(bool, irq) __field(unsigned int, index) ), TP_fast_assign( @@ -74,10 +82,12 @@ TRACE_EVENT(out_packet, __entry->cip_header0 = be32_to_cpu(cip_header[0]); __entry->cip_header1 = be32_to_cpu(cip_header[1]); __entry->payload_quadlets = payload_length / 4; - __entry->index = s->packet_index; + __entry->packet_index = s->packet_index; + __entry->irq = in_interrupt(); + __entry->index = index; ), TP_printk( - "%02u %04u %04x %04x %02d %08x %08x %03u %02u", + "%02u %04u %04x %04x %02d %08x %08x %03u %02u %01u %02u", __entry->second, __entry->cycle, __entry->src, @@ -86,6 +96,8 @@ TRACE_EVENT(out_packet, __entry->cip_header0, __entry->cip_header1, __entry->payload_quadlets, + __entry->packet_index, + __entry->irq, __entry->index) ); diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index e7163bd..db30fac4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -411,7 +411,8 @@ static inline int queue_in_packet(struct amdtp_stream *s) amdtp_stream_get_max_payload(s)); } -static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) +static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle, + unsigned int index) { __be32 *buffer; unsigned int syt; @@ -436,7 +437,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff; payload_length = 8 + data_blocks * 4 * s->data_block_quadlets; - trace_out_packet(s, cycle, buffer, payload_length); + trace_out_packet(s, cycle, buffer, payload_length, index); if (queue_out_packet(s, payload_length) < 0) return -EIO; @@ -450,7 +451,8 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) } static int handle_in_packet(struct amdtp_stream *s, - unsigned int payload_quadlets, unsigned int cycle) + unsigned int payload_quadlets, unsigned int cycle, + unsigned int index) { __be32 *buffer; u32 cip_header[2]; @@ -465,7 +467,7 @@ static int handle_in_packet(struct amdtp_stream *s, cip_header[0] = be32_to_cpu(buffer[0]); cip_header[1] = be32_to_cpu(buffer[1]); - trace_in_packet(s, cycle, cip_header, payload_quadlets); + trace_in_packet(s, cycle, cip_header, payload_quadlets, index); /* * This module supports 'Two-quadlet CIP header with SYT field'. @@ -604,7 +606,7 @@ static void out_stream_callback(struct fw_iso_context *context, u32 tstamp, for (i = 0; i < packets; ++i) { cycle = increment_cycle_count(cycle, 1); - if (handle_out_packet(s, cycle) < 0) { + if (handle_out_packet(s, cycle, i) < 0) { s->packet_index = -1; amdtp_stream_pcm_abort(s); return; @@ -651,7 +653,7 @@ static void in_stream_callback(struct fw_iso_context *context, u32 tstamp, break; } - if (handle_in_packet(s, payload_quadlets, cycle) < 0) + if (handle_in_packet(s, payload_quadlets, cycle, i) < 0) break; } -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
by Takashi Sakamoto 11 May '16

11 May '16

On most of audio and music units on IEEE 1394 bus which ALSA firewire stack supports (or plans to support), CIP with two quadlets header is used. Thus, there's no cases to queue packets with blank payload. If such packets are going to be queued, it means that they're for skips of the cycle. This commit simplifies helper functions to queue a packet. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 22 +++++++++++----------- 1 file changed, 11 insertions(+), 11 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 830a95c..e7163bd 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -370,9 +370,8 @@ static void pcm_period_tasklet(unsigned long data) snd_pcm_period_elapsed(pcm); } -static int queue_packet(struct amdtp_stream *s, - unsigned int header_length, - unsigned int payload_length, bool skip) +static int queue_packet(struct amdtp_stream *s, unsigned int header_length, + unsigned int payload_length) { struct fw_iso_packet p = {0}; int err = 0; @@ -383,8 +382,10 @@ static int queue_packet(struct amdtp_stream *s, p.interrupt = IS_ALIGNED(s->packet_index + 1, INTERRUPT_INTERVAL); p.tag = TAG_CIP; p.header_length = header_length; - p.payload_length = (!skip) ? payload_length : 0; - p.skip = skip; + if (payload_length > 0) + p.payload_length = payload_length; + else + p.skip = true; err = fw_iso_context_queue(s->context, &p, &s->buffer.iso_buffer, s->buffer.packets[s->packet_index].offset); if (err < 0) { @@ -399,16 +400,15 @@ end: } static inline int queue_out_packet(struct amdtp_stream *s, - unsigned int payload_length, bool skip) + unsigned int payload_length) { - return queue_packet(s, OUT_PACKET_HEADER_SIZE, - payload_length, skip); + return queue_packet(s, OUT_PACKET_HEADER_SIZE, payload_length); } static inline int queue_in_packet(struct amdtp_stream *s) { return queue_packet(s, IN_PACKET_HEADER_SIZE, - amdtp_stream_get_max_payload(s), false); + amdtp_stream_get_max_payload(s)); } static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) @@ -438,7 +438,7 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int cycle) trace_out_packet(s, cycle, buffer, payload_length); - if (queue_out_packet(s, payload_length, false) < 0) + if (queue_out_packet(s, payload_length) < 0) return -EIO; pcm = ACCESS_ONCE(s->pcm); @@ -764,7 +764,7 @@ int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed) if (s->direction == AMDTP_IN_STREAM) err = queue_in_packet(s); else - err = queue_out_packet(s, 0, true); + err = queue_out_packet(s, 0); if (err < 0) goto err_context; } while (s->packet_index > 0); -- 2.7.4

3 3

[alsa-devel] [RFC][PATCH] ALSA: firewire-lib: permit process context only to flush queued packets for better PCM period granularity
by Takashi Sakamoto 11 May '16

11 May '16

Hi Clemens, Could I request your comment to this patch? It is to solve a race condition, but the condition is quite rare. Practically, it might have a less meanings, except for better program. And I think there's another race condition against processing each packets by calling out/in_stream_callback(), but I cannot observe the race. Software IRQ contexts of IR/IT contexts and process contexts are under the race condition, however I can see no problems related to it in my several trials in multi-core machine. I have no idea about the reason that packet sequence is processed correctly between the software IRQ contexts and the process contexts without any lock primitives. Do you have some ideas about it? Regards Takashi Sakamoto ----- 8< ----- Subject: ALSA: firewire-lib: permit process context only to flush queued packets for better PCM period granularity These three commits were merged to improve PCM pointer granularity. commit 76fb87894828 ("ALSA: firewire-lib: taskletize the snd_pcm_period_elapsed() call") commit e9148dddc3c7 ("ALSA: firewire-lib: flush completed packets when reading PCM position") commit 92b862c7d685 ("ALSA: firewire-lib: optimize packet flushing") The point of them is to handle queued packets not only in software IRQ context of IR/IT contexts, but also in process context. This idea introduced a cyclic call of 'struct snd_pcm_ops.pointer()'. The last commit adds 'pointer_flush' member to 'struct amdtp_stream' to avoid the situation. On the other hand, This solution is weak at race condition between the process context and the software IRQ context of IR/IT contexts. Practically, this race is not so critical because it influences process context to skip flushing queued packets and to get worse granularity of PCM pointer. This is quite rare because 'struct snd_pcm_ops.pointer()' is frequently called by process contexts while IT/IR contexts run less frequently. However, it's better to solve the race. The similar solution can be achieved by 'in_interrupt()' macro. This commit applies the macro to solve the race condition against 'pointer_flush'. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/amdtp-stream.c | 23 +++++++++++++++++------ sound/firewire/amdtp-stream.h | 1 - 2 files changed, 17 insertions(+), 7 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 4484242..132c7f4 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -248,7 +248,6 @@ void amdtp_stream_pcm_prepare(struct amdtp_stream *s) tasklet_kill(&s->period_tasklet); s->pcm_buffer_pointer = 0; s->pcm_period_pointer = 0; - s->pointer_flush = true; } EXPORT_SYMBOL(amdtp_stream_pcm_prepare); @@ -353,7 +352,6 @@ static void update_pcm_pointers(struct amdtp_stream *s, s->pcm_period_pointer += frames; if (s->pcm_period_pointer >= pcm->runtime->period_size) { s->pcm_period_pointer -= pcm->runtime->period_size; - s->pointer_flush = false; tasklet_hi_schedule(&s->period_tasklet); } } @@ -798,11 +796,24 @@ EXPORT_SYMBOL(amdtp_stream_start); */ unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s) { - /* this optimization is allowed to be racy */ - if (s->pointer_flush && amdtp_stream_running(s)) + /* + * This function is called in software IRQ context of period_tasklet or + * process context. + * + * When the software IRQ context was scheduled by software IRQ context + * of IR/IT contexts, queued packets were already handled. Therefore, + * no need to flush the queue in buffer anymore. + * + * When the process context reach here, some packets wll be queued in + * the buffer. These packets should be handled immediately to keep + * better granularity of PCM pointer. + * + * Later, the process context will sometimes schedules software IRQ + * context of the period_tasklet. Then, no need to flush the queue by + * the same reason as described for IR/IT contexts. + */ + if (!in_interrupt() && amdtp_stream_running(s)) fw_iso_context_flush_completions(s->context); - else - s->pointer_flush = true; return ACCESS_ONCE(s->pcm_buffer_pointer); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 8775704..ab697cc 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -132,7 +132,6 @@ struct amdtp_stream { struct tasklet_struct period_tasklet; unsigned int pcm_buffer_pointer; unsigned int pcm_period_pointer; - bool pointer_flush; /* To wait for first packet. */ bool callbacked; -- 2.7.4

3 7

[alsa-devel] [PATCH] ALSA: usb-audio: Yet another Phoneix Audio device quirk
by Takashi Iwai 11 May '16

11 May '16

Phoenix Audio has yet another device with another id (even a different vendor id, 0556:0014) that requires the same quirk for the sample rate. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=110221 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 7b6dca2d7488..220764cd60bd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1137,6 +1137,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0x0415): /* Plantronics BT-300 */ case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ + case uSB_ID(0x0556, 0x0014): /* Phoenix Audio TMX320VC */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ -- 2.8.2

3 3

[alsa-devel] [PATCH] ASoC: twl6040: Select LPPLL during standby
by Peter Ujfalusi 11 May '16

11 May '16

When the codec is in standby we do not need to keep the HPPLL active. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/twl6040.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index bc3de2e844e6..7a744b581cc1 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -824,7 +824,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, { struct twl6040 *twl6040 = codec->control_data; struct twl6040_data *priv = snd_soc_codec_get_drvdata(codec); - int ret; + int ret = 0; switch (level) { case SND_SOC_BIAS_ON: @@ -832,12 +832,16 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - if (priv->codec_powered) + if (priv->codec_powered) { + /* Select low power PLL in standby */ + ret = twl6040_set_pll(twl6040, TWL6040_SYSCLK_SEL_LPPLL, + 32768, 19200000); break; + } ret = twl6040_power(twl6040, 1); if (ret) - return ret; + break; priv->codec_powered = 1; @@ -853,7 +857,7 @@ static int twl6040_set_bias_level(struct snd_soc_codec *codec, break; } - return 0; + return ret; } static int twl6040_startup(struct snd_pcm_substream *substream, -- 2.8.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix printk formatting
by Joonas Lahtinen 11 May '16

11 May '16

Format number after 0x in hex. Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Joonas Lahtinen <joonas.lahtinen(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/haswell/sst-haswell-pcm.c b/sound/soc/intel/haswell/sst-haswell-pcm.c index 1aa819c..994256b 100644 --- a/sound/soc/intel/haswell/sst-haswell-pcm.c +++ b/sound/soc/intel/haswell/sst-haswell-pcm.c @@ -445,7 +445,7 @@ static int create_adsp_page_table(struct snd_pcm_substream *substream, pages = snd_sgbuf_aligned_pages(size); - dev_dbg(rtd->dev, "generating page table for %p size 0x%zu pages %d\n", + dev_dbg(rtd->dev, "generating page table for %p size 0x%zx pages %d\n", dma_area, size, pages); for (i = 0; i < pages; i++) { -- 2.5.5

2 1

[alsa-devel] [PATCH RFT] ASoC: max98371 Remove duplicate entry in max98371_reg
by Axel Lin 11 May '16

11 May '16

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/max98371.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index 21fae09..cf0a39b 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -107,7 +107,6 @@ static const struct reg_default max98371_reg[] = { { 0x39, 0x00 }, { 0x3A, 0x00 }, { 0x3B, 0x00 }, - { 0x3B, 0x00 }, { 0x3C, 0x00 }, { 0x3D, 0x00 }, { 0x3E, 0x00 }, -- 2.5.0

2 1

[alsa-devel] [PATCH v5 0/4] ASoC: davinci-mcbsp: add binding for McBSP
by Petr Kulhavy 11 May '16

11 May '16

This series of patches adds devicetree support for the davinci McBSP audio interface. Changes in version 5 of the patch: * rebased to the current master of git://git.kernel.org/pub/scm/linux/kernel/git/nsekhar/linux-davinci.git * independent changes submitted separately (3/7 and 7/7 of the version 4) * 5/7 of the version 4 removed as already applied * corrected mcbsp0 and mcbsp1 nodes in the DTS file: using correct address and sound@ prefix

3 7

[alsa-devel] [PATCH alsa-lib] pcm: dmix: Fix doubly resume of slave PCM
by Takashi Iwai 11 May '16

11 May '16

The dmix plugin and co may trigger the resume for each instance in snd_pcm_direct_resume(). It means that the slave PCM gets resumed or re-prepared/started by each opened dmix stream, and this may end up with the doubly triggers even though the slave PCM has been already resumed by another dmix stream. For avoiding this conflicts, check the slave PCM state and resume only when it's still in the suspended state. Meanwhile we keep the shadow state updated no matter whether the slave was triggered or not. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e28738b0de96..ac082f1a73b2 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -841,6 +841,12 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) int err; snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); + /* resume only when the slave PCM is still in suspended state */ + if (snd_pcm_state(dmix->spcm) != SND_PCM_STATE_SUSPENDED) { + err = 0; + goto out; + } + err = snd_pcm_resume(dmix->spcm); if (err == -ENOSYS) { /* FIXME: error handling? */ @@ -848,6 +854,7 @@ int snd_pcm_direct_resume(snd_pcm_t *pcm) snd_pcm_start(dmix->spcm); err = 0; } + out: dmix->state = snd_pcm_state(dmix->spcm); snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); return err; -- 2.8.2

1 0

[alsa-devel] ASoC: DAPM and differential inputs
by Jeremy McDermond 10 May '16

10 May '16

I’ve been working on the tlv320aic32x4 driver for a while now to add some features for some projects I’m working on using the chip. I’m trying to wrap my head around DAPM and getting it to work with the variety of inputs on the aic32x4. Basically, the codec has a set of registers that control the positive and negative inputs to each of the right and left input mixers. The only setting is what value of resistor to connect between the input pin and a particular mixer (none, 10k, 20k, 40k). For example, you can have a differential pair between IN1_R and IN2_R, or between IN1_R and IN3_R, or single ended by setting the negative pin to the common. So, there’s a lot of flexibility and not just a differential “mode”. The registers would even allow you to set up two negative pins, although I’m not sure why that would be useful. I’m currently setting up DAPM so that both the positive and negative pins are wired to a single mixer. This doesn’t seem to be working since only the negative pin seems to cause the path to be enabled. Do I have to set up another “Negative Right” and “Negative Left” mixer to handle the negative pins? Am I missing something else? -- Jeremy McDermond (NH6Z) Xenotropic Systems mcdermj(a)xenotropic.com

1 0

[alsa-devel] [PATCH] MAINTAINERS: Add myself as reviewer of FSL/NXP SoC sound drivers
by Fabio Estevam 10 May '16

10 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> I would like to help reviewing FSL/NXP SoC sound drivers. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 43b85c1..f896605 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -4846,6 +4846,7 @@ FREESCALE SOC SOUND DRIVERS M: Timur Tabi <timur(a)tabi.org> M: Nicolin Chen <nicoleotsuka(a)gmail.com> M: Xiubo Li <Xiubo.Lee(a)gmail.com> +R: Fabio Estevam <fabio.estevam(a)nxp.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) L: linuxppc-dev(a)lists.ozlabs.org S: Maintained -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: simple-card: Add pm callbacks to platform driver
by Peter Ujfalusi 10 May '16

10 May '16

Set snd_soc_pm_ops for the pm ops to make sure that the ASoC level of PM operations are going to happen. This is needed to get suspend/resume working correctly when the audio is using simple-card. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, we have moved most of the TI boards to use simple-card and I have noticed that we have quite bad regression because of it caused by simple-card not having the pm ops configured. This will render audio to not recover after suspend/resume when the audio IP is loosing context. Regards, Peter sound/soc/generic/simple-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c6111e7a6d93..52a1e1ff3c62 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -658,6 +658,7 @@ MODULE_DEVICE_TABLE(of, asoc_simple_of_match); static struct platform_driver asoc_simple_card = { .driver = { .name = "asoc-simple-card", + .pm = &snd_soc_pm_ops, .of_match_table = asoc_simple_of_match, }, .probe = asoc_simple_card_probe, -- 2.8.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Do not allow multiple streams in one direction
by Peter Ujfalusi 10 May '16

10 May '16

Make sure that the user can not start multiple streams with the same direction. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index c4cd8e4fff9a..50523e744483 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1252,11 +1252,15 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, int i, dir; int tdm_slots = mcasp->tdm_slots; - if (mcasp->tdm_mask[substream->stream]) - tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + /* Do not allow more then one stream per direction */ + if (mcasp->substreams[substream->stream]) + return -EBUSY; mcasp->substreams[substream->stream] = substream; + if (mcasp->tdm_mask[substream->stream]) + tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) return 0; -- 2.8.1

2 1

[alsa-devel] [PATCH 0/4] ASoC: davinci-mcasp: Master mode clock divider correction
by Peter Ujfalusi 10 May '16

10 May '16

Hi, If the McASP is used as clock master and the reference clock is AUXCLK we can have additional level of divider. The BCLK divider is limited to maximum 32, if the desired bclk can not be reached with this, the AUXCLK divider also needs to be used. Regards, Peter --- Peter Ujfalusi (4): ASoC: davinci-mcasp: Use defines for clkdiv IDs ASoC: davinci-mcasp: Change __davinci_mcasp_set_clkdiv() first parameter ASoC: davinci-mcasp: Restructure the davinci_mcasp_calc_clk_div() ASoC: davinci-mcasp: Calculate AUXCLK divider when setting up master clocks sound/soc/davinci/davinci-mcasp.c | 82 +++++++++++++++++++++++++-------------- sound/soc/davinci/davinci-mcasp.h | 5 +++ 2 files changed, 57 insertions(+), 30 deletions(-) -- 2.8.2

2 8

[alsa-devel] [PATCH] ASoC: rsnd: don't use prohibited number to PDMACHCRn.SRS
by Kuninori Morimoto 10 May '16

10 May '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_dmapp_get_id() returns 0xFF as error code if system used strange connection. It will be used as PDMACHCRn.SRS, but 0xFF is prohibited number. In order not to use prohibited number, this patch indicates error message and returns 0x00 (same as SSI00) in error case. Special thanks to Dung-san. Reported-by: Nguyen Viet Dung <nv-dung(a)jinso.co.jp> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/dma.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 7658e8f..6bc93cb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -316,11 +316,15 @@ static u32 rsnd_dmapp_get_id(struct rsnd_dai_stream *io, size = ARRAY_SIZE(gen2_id_table_cmd); } - if (!entry) - return 0xFF; + if ((!entry) || (size <= id)) { + struct device *dev = rsnd_priv_to_dev(rsnd_io_to_priv(io)); - if (size <= id) - return 0xFF; + dev_err(dev, "unknown connection (%s[%d])\n", + rsnd_mod_name(mod), rsnd_mod_id(mod)); + + /* use non-prohibited SRS number as error */ + return 0x00; /* SSI00 */ + } return entry[id]; } -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] ASoC: Intel: Add Broxton Machine
by Vinod Koul 10 May '16

10 May '16

Broxton uses SSP5 for audio on Intel board, so add the 6 SSPs and then machine driver which uses RT298 codec Pardha Saradhi K (1): ASoC: Intel: Skylake: Add more SSP DAIs Ramesh Babu (1): ASoC: Intel: Add Broxton-P machine driver sound/soc/intel/Kconfig | 15 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bxt_rt298.c | 353 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 72 ++++++++ 4 files changed, 442 insertions(+) create mode 100644 sound/soc/intel/boards/bxt_rt298.c -- 1.9.1

2 4

[alsa-devel] [PATCH 0/5] ASoC: da7213: clocking related improvements for codec
by Adam Thomson 10 May '16

10 May '16

This is a collection of updates with regards to clocking on da7213 codec, including power efficiency & stability improvements. Patches are based on v4.6-rc7 kernel. Adam Thomson (5): ASoC: da7213: Add DAI DAPM event to control DAI clocks ASoC: da7213: Add checking of SRM lock status before enabling DAI ASoC: da7213: Default PC counter to free-running when DAI disabled ASoC: da7213: Update PLL ranges to improve locking at frequency boundary ASoC: da7213: Allow PLL disable/bypass when using 32KHz sysclk sound/soc/codecs/da7213.c | 99 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/da7213.h | 35 ++++++++++------- 2 files changed, 102 insertions(+), 32 deletions(-) -- 1.9.3

2 12

[alsa-devel] [PATCH 0/7] ALSA: drop reuse of incoming packet parameter for outgoing packet parameter
by Takashi Sakamoto 10 May '16

10 May '16

Hi, This patchset updates my former RFC, and go for merging to Linux 4.7. [alsa-devel] [RFC][PATCH 0/6] ALSA: firewire: drop reusing incoming packet parameters for outgoing packet parameters http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107290.html The aim of this patchset is to remove a feature from firewire-lib module. I call it as 'full duplex streaming with timestamp synchronization'. The feature is based on our understanding of IEC 61883-6. In the specification, a function block should reuse some parameters of incoming packets for outgoing packets. Current ALSA firewire stack partly implement it. However, actual audio and music units on IEEE 1394 bus don't expect corresponding drivers to satisfy the feature. On the other hand, the feature brings inconveniences to this module. When this feature is enabled, a software IRQ context processes packets for both of IR/IT contexts. This consumes more time over local CPUs than handling one context. Furthermore, cycle count for outgoing packets becomes improper because they're for incoming packets. For these reasons, this patchset removes the feature from ALSA firewire stack. Changes: - The last patch is newly added. Regards Takashi Sakamoto (7): ALSA: bebob: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: fireworks: drop reuse of incoming packet parameter for ougoing packet parameter ALSA: firewire-tascam: drop reuse of incoming packet parameter for outgoing packet parameter ALSA: firewire-lib: handle IT/IR contexts in each software interrupt context ALSA: firewire-lib: code cleanup for incoming packet handling ALSA: firewire-lib: code cleanup for outgoing packet handling ALSA: firewire-lib: enable the same feature as CIP_SKIP_INIT_DBC_CHECK flag sound/firewire/amdtp-stream.c | 98 +++++++++------------------ sound/firewire/amdtp-stream.h | 36 ++-------- sound/firewire/bebob/bebob.h | 1 - sound/firewire/bebob/bebob_stream.c | 101 +++++++--------------------- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/fireworks/fireworks.h | 1 - sound/firewire/fireworks/fireworks_stream.c | 84 +++++------------------ sound/firewire/oxfw/oxfw-stream.c | 3 +- sound/firewire/tascam/tascam-stream.c | 26 ++++--- 9 files changed, 94 insertions(+), 258 deletions(-) -- 2.7.4

2 8

[alsa-devel] [PATCH] ALSA: pcm: Bail out when chmap is already present
by Takashi Iwai 10 May '16

10 May '16

When snd_pcm_add_chmap_ctls() is called to the PCM stream to which a chmap has been already assigned, it returns as an error due to the conflicting snd_ctl_add() result. However, this also clears the already assigned chmap_kctl field via pcm_chmap_ctl_private_free(), and becomes inconsistent in the later operation. This patch adds the check of the conflicting chmap kctl before actually trying to allocate / assign. The check failure is treated as a kernel warning, as the double call of snd_pcm_add_chmap_ctls() is basically a driver bug and having the stack trace would help developers to figure out the bad code path. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0aca39762ed0..bb1261591a1f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -2595,6 +2595,8 @@ int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream, }; int err; + if (WARN_ON(pcm->streams[stream].chmap_kctl)) + return -EBUSY; info = kzalloc(sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; -- 2.8.2

1 0

[alsa-devel] [PATCH 3/4] ASoC: ab8500-codec: remove platform data based probe
by Arnd Bergmann 10 May '16

10 May '16

The ux500 platform hasn't used board files for a long time, and nothing defines a ab8500_codec_platform_data, so we can just remove the probing based on that and always use device tree properties directly. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/ab8500-codec.c | 33 ++++----------------------------- 1 file changed, 4 insertions(+), 29 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 8b1d0c1a7839..2fc89155f14a 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2464,45 +2464,20 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) struct device *dev = codec->dev; struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); - struct ab8500_platform_data *pdata; + struct ab8500_codec_platform_data codec_pdata; struct filter_control *fc; int status; dev_dbg(dev, "%s: Enter.\n", __func__); - /* Setup AB8500 according to board-settings */ - pdata = dev_get_platdata(dev->parent); + ab8500_codec_of_probe(dev, np, &codec_pdata); - if (np) { - if (!pdata) - pdata = devm_kzalloc(dev, - sizeof(struct ab8500_platform_data), - GFP_KERNEL); - - if (pdata && !pdata->codec) - pdata->codec - = devm_kzalloc(dev, - sizeof(struct ab8500_codec_platform_data), - GFP_KERNEL); - - if (!(pdata && pdata->codec)) - return -ENOMEM; - - ab8500_codec_of_probe(dev, np, pdata->codec); - - } else { - if (!(pdata && pdata->codec)) { - dev_err(dev, "No codec platform data or DT found\n"); - return -EINVAL; - } - } - - status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + status = ab8500_audio_setup_mics(codec, &codec_pdata.amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); return status; } - status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + status = ab8500_audio_set_ear_cmv(codec, codec_pdata.ear_cmv); if (status < 0) { pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", __func__, status); -- 2.7.0

1 0

[alsa-devel] bug: reconfig broken on kernel 4.5
by Jochen Henneberg 10 May '16

10 May '16

Hi, it looks as if 'reconfig' feature is broken since the sysfs attributes have been moved to device attributes. In hda_sysfs.c:reconfig_codec() the device is cleared with snd_hda_codec_reset() which calls snd_hdac_device_unregister() which removes the sysfs entries (including the currently in-use 'reconfig'), the re-registration fails afterwards in snd_hda_codec_build_controls(). I am not sure when things got broken, but snd_hda_codec_reset() called from reconfig_codec() unregisters the device and removes the sysfs entries where in the past it only stripped all configurations (pcms, controls ...) from the device. Regards, Jochen

2 4

[alsa-devel] [PATCH] ALSA: Fix subwoofer pin on ASUS N751 and N551
by Takashi Iwai 10 May '16

10 May '16

From: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Subwoofer does not work out of the box on ASUS N751/N551 laptops. This patch fixes it. Patch tested on N751 laptop. N551 part is not tested, but according to [1] and [2] this laptop requires similar changes, so I included them in the patch. 1. https://github.com/honsiorovskyi/asus-n551-hda-fix 2. https://bugs.launchpad.net/ubuntu/+source/alsa-tools/+bug/1405691 Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117781 Signed-off-by: Yura Pakhuchiy <pakhuchiy(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1b5cd19876fc..4918ffa5ba68 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6426,6 +6426,7 @@ enum { ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, ALC662_FIXUP_ASUS_Nx50, + ALC668_FIXUP_ASUS_Nx51, }; static const struct hda_fixup alc662_fixups[] = { @@ -6672,6 +6673,15 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC662_FIXUP_BASS_1A }, + [ALC668_FIXUP_ASUS_Nx51] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x1a, 0x90170151}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC662_FIXUP_BASS_CHMAP, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6700,6 +6710,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x129d, "Asus N750", ALC662_FIXUP_ASUS_Nx50), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), + SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 2.8.2

1 0

[alsa-devel] Kernel config issue: SND_HDA_RECONFIG depends on SND_HDA_HWDEP
by Jochen Henneberg 09 May '16

09 May '16

Hi, with kernel 4.5 (and it looks as if this has not changed since then) the SND_HDA_RECONFIG sysfs entries still show up in the hwdep devices. Enabling SND_HDA_RECONFIG without SND_HDA_HWDEP is worth nothing, but the dependency has been removed from Kconfig. The commits from 13aeaf68 / 'Manage each codec instance individually' read as if the sysfs entries should show up at the codec driver but they do not. Regards, Jochen

2 5

[alsa-devel] [PATCH 0/5] Fix poll error returns
by Charles Keepax 09 May '16

09 May '16

We return negative values from the file_operations poll callback in a few places, this callback returns an unsigned int and is expected to only return the poll specific flags. This series fixes up these issues and makes a couple of small tidy ups to the code around the edges of those functions. I noticed this issue whilst doing some additional testing on my propagation of compressed stream error series, but I decided to push these up separately as it is worth getting the fixes in now and not tying them up with that patch chain which is taking longer to get merged. Also I included the first patch of that chain (Replace complex if statement with switch) because it is a trivial tidy up and might as well get merged now as well. Thanks, Charles Charles Keepax (5): ALSA: pcm: Fix poll error return codes ALSA: compress: Use snd_compr_get_poll on error path ALSA: compress: Remove pointless NULL check ALSA: compress: Fix poll error return codes ALSA: compress: Replace complex if statement with switch sound/core/compress_offload.c | 25 ++++++++++++------------- sound/core/pcm_native.c | 4 ++-- 2 files changed, 14 insertions(+), 15 deletions(-) -- 2.1.4

5 12

[alsa-devel] [PATCH 0/3 v3] firewire-lib: support tracepoints to dump a part of packet data
by Takashi Sakamoto 09 May '16

09 May '16

Hi, This patchset updates my RFCv2, and go for merging. [alsa-devel] [RFC][PATCH 0/3 v2] firewire-lib: support Linux tracing framework to dump a part of packet data http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html The motivation of this patchset is to check the sequence of packet from/to audio and music units on IEEE 1394 bus. When the units have some quirks or ALSA firewire stack includes some bugs, we can work to seek them with Linux tracing framework. Changes: - Improve commit comments. - Improve Python3 script in commit comment for NODATA/NOINFO packets. - Move the position of trace_out_acket() so that the index of packet starts 0. Takashi Sakamoto (3): ALSA: firewire-lib: compute the value of second field in cycle count for IT context ALSA: firewire-lib: compute the value of second field in cycle count for IR context ALSA: firewire-lib: add tracepoints to dump a part of isochronous packet data sound/firewire/Makefile | 3 ++ sound/firewire/amdtp-stream-trace.h | 98 +++++++++++++++++++++++++++++++++++++ sound/firewire/amdtp-stream.c | 77 ++++++++++++++++++++++------- 3 files changed, 160 insertions(+), 18 deletions(-) create mode 100644 sound/firewire/amdtp-stream-trace.h -- 2.7.4

2 4

[alsa-devel] [PATCH alsa-lib] conf/ucm: ROCKCHIP-I2S: add Rockchip I2S UCM config.
by Enric Balletbo i Serra 09 May '16

09 May '16

Taken from the ChromeOS sources, this configuration was tested on Veyron Jerry based Chromebook from Google. Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- src/conf/ucm/Makefile.am | 2 +- src/conf/ucm/ROCKCHIP-I2S/HiFi.conf | 94 +++++++++++++++++++++++++++++ src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf | 6 ++ 3 files changed, 101 insertions(+), 1 deletion(-) create mode 100644 src/conf/ucm/ROCKCHIP-I2S/HiFi.conf create mode 100644 src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index e6a6325..d1d51e5 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 +SUBDIRS=DAISY-I2S PandaBoard PandaBoardES SDP4430 tegraalc5632 PAZ00 GoogleNyan broadwell-rt286 ROCKCHIP-I2S diff --git a/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf b/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf new file mode 100644 index 0000000..7bfe995 --- /dev/null +++ b/src/conf/ucm/ROCKCHIP-I2S/HiFi.conf @@ -0,0 +1,94 @@ +SectionVerb { + Value { + OutputDspName "speaker_eq" + MinBufferLevel "512" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Headphone Left Switch' off" + cset "name='Headphone Right Switch' off" + cset "name='Digital EQ 3 Band Switch' off" + cset "name='Digital EQ 5 Band Switch' off" + cset "name='Digital EQ 7 Band Switch' off" + cset "name='Biquad Switch' off" + cset "name='Filter Mode' Music" + cset "name='ADC Oversampling Rate' 0" + + cset "name='DMIC Mux' DMIC" + cset "name='MIC2 Mux' IN34" + cset "name='Right ADC Mixer MIC2 Switch' on" + cset "name='Left ADC Mixer MIC2 Switch' on" + cset "name='MIC2 Volume' 20" + cset "name='Headset Mic Switch' off" + cset "name='Int Mic Switch' on" + + cset "name='ADCR Boost Volume' 4" + cset "name='ADCL Boost Volume' 4" + cset "name='ADCR Volume' 11" + cset "name='ADCL Volume' 11" + + cset "name='Left Speaker Mixer Left DAC Switch' on" + cset "name='Right Speaker Mixer Right DAC Switch' on" + cset "name='Speaker Left Mixer Volume' 2" + cset "name='Speaker Right Mixer Volume' 2" + cset "name='Record Path DC Blocking' on" + cset "name='Playback Path DC Blocking' on" + + cset "name='Speaker Left Switch' on" + cset "name='Speaker Right Switch' on" + cset "name='Speaker Switch' on" + ] + + DisableSequence [ + ] +} + +SectionDevice."Headphone".0 { + Value { + JackName "ROCKCHIP-I2S Headset Jack" + OutputDspName "" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Speaker Switch' off" + cset "name='Headphone Left Switch' on" + cset "name='Headphone Right Switch' on" + ] + DisableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Headphone Left Switch' off" + cset "name='Headphone Right Switch' off" + cset "name='Speaker Switch' on" + ] +} + +SectionDevice."Mic".0 { + Value { + JackName "ROCKCHIP-I2S Headset Jack" + } + + EnableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Int Mic Switch' off" + cset "name='DMIC Mux' ADC" + cset "name='Headset Mic Switch' on" + cset "name='Record Path DC Blocking' on" + ] + + DisableSequence [ + cdev "hw:ROCKCHIPI2S" + + cset "name='Headset Mic Switch' off" + cset "name='DMIC Mux' DMIC" + cset "name='Int Mic Switch' on" + cset "name='Record Path DC Blocking' off" + ] +} diff --git a/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf b/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf new file mode 100644 index 0000000..0c399b0 --- /dev/null +++ b/src/conf/ucm/ROCKCHIP-I2S/ROCKCHIP-I2S.conf @@ -0,0 +1,6 @@ +Comment "Rockchip card" + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Default" +} -- 2.1.0

2 1

[alsa-devel] [PATCH] pcm: softvol: fix conversion of TLVs min_db and max_dB value
by Jörg Krause 09 May '16

09 May '16

Both, min_dB and max_dB, are floating type whereas the TLV is (always) unsigned. The problem with the conversion of a negative floating-point number into an unsigned integer is, that the behavior is undefined. This may, depending on the platform, result in a wrong TLV, i.e. for the default values of min_dB (-51dB) and max_dB (0dB), alsactl generates the following state on an ARM cpu build with GCC: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } With the fix applied, alsactl stores the correct TLV: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Also tested for different combinations of min_dB and max_dB other than the default values. Replaces: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107733.html Fixes: http://mailman.alsa-project.org/pipermail/alsa-devel/2016-May/107628.html Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 802aa4b..061080d 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -658,8 +658,8 @@ static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo) unsigned int tlv[4]; tlv[0] = SND_CTL_TLVT_DB_SCALE; tlv[1] = 2 * sizeof(int); - tlv[2] = svol->min_dB * 100; - tlv[3] = (svol->max_dB - svol->min_dB) * 100 / svol->max_val; + tlv[2] = (int)(svol->min_dB * 100); + tlv[3] = (int)((svol->max_dB - svol->min_dB) * 100 / svol->max_val); return snd_ctl_elem_tlv_write(svol->ctl, &cinfo->id, tlv); } -- 2.8.2

2 1

[alsa-devel] [PATCH] ASoC: pcm: allow changing the playback/capture rates for symmetric links
by Peter Rosin 09 May '16

09 May '16

The below program fails on a dai link with symmetric rates without this patch. The patch makes it work. #include <sys/soundcard.h> #include <unistd.h> #include <fcntl.h> #include <stropts.h> #include <stdio.h> int main(void) { int fd; int format; int channels; int speed; if ((fd = open("/dev/dsp", O_WRONLY, 0)) == -1) { perror("open"); return 1; } format = AFMT_S16_LE; if (ioctl(fd, SNDCTL_DSP_SETFMT, &format) == -1) { perror("SNDCTL_DSP_SETFMT"); return 1; } channels = 2; if (ioctl(fd, SNDCTL_DSP_CHANNELS, &channels) == -1) { perror("SNDCTL_DSP_CHANNELS"); return 1; } speed = 22050; if (ioctl(fd, SNDCTL_DSP_SPEED, &speed) == -1) { perror("SNDCTL_DSP_SPEED"); return 1; } return 0; } Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/soc-pcm.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4fa86..1e876ff23524 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -237,6 +237,14 @@ static int soc_pcm_params_symmetry(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai = rtd->cpu_dai; unsigned int rate, channels, sample_bits, symmetry, i; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (!cpu_dai->capture_active) + return 0; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (!cpu_dai->playback_active) + return 0; + } + rate = params_rate(params); channels = params_channels(params); sample_bits = snd_pcm_format_physical_width(params_format(params)); -- 2.1.4

3 12

[alsa-devel] [PATCH 0/2] wm5110: fixes to ANC clock control
by Richard Fitzgerald 09 May '16

09 May '16

The register for ANC clock control has a combination of volatile and non-readback bits and these two patches fix the way we write to this register. Both patches must be taken together to ensure correct operation. Richard Fitzgerald (2): mfd: wm5110: ARIZONA_CLOCK_CONTROL should be volatile ASoC: arizona: No need to update_bits when writing AEC clock control drivers/mfd/wm5110-tables.c | 1 + sound/soc/codecs/arizona.c | 3 +-- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.9.1

3 4

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake: Add DSP firmware manifest parsing
by Vinod Koul 09 May '16

09 May '16

From: Shreyas NC <shreyas.nc(a)intel.com> Module params like module_id and loadable flag can be changed in the DSP Firmware. These are kept in the firmware manifest and driver should read these values from this manifest. So, add support to parse the DSP firmware manifest and read these module params. Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Changes in v3: add bound check for firmware file sound/soc/intel/skylake/Makefile | 2 +- sound/soc/intel/skylake/skl-dsp-parse.c | 118 ++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-dsp-parse.h | 107 +++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 4 ++ sound/soc/intel/skylake/skl-sst.c | 11 ++- sound/soc/intel/skylake/skl-topology.c | 5 ++ 6 files changed, 245 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.c create mode 100644 sound/soc/intel/skylake/skl-dsp-parse.h diff --git a/sound/soc/intel/skylake/Makefile b/sound/soc/intel/skylake/Makefile index c28f5d0e1d99..e2fb5b763b21 100644 --- a/sound/soc/intel/skylake/Makefile +++ b/sound/soc/intel/skylake/Makefile @@ -5,6 +5,6 @@ obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl.o # Skylake IPC Support snd-soc-skl-ipc-objs := skl-sst-ipc.o skl-sst-dsp.o skl-sst-cldma.o \ - skl-sst.o bxt-sst.o + skl-sst.o skl-dsp-parse.o bxt-sst.o obj-$(CONFIG_SND_SOC_INTEL_SKYLAKE) += snd-soc-skl-ipc.o diff --git a/sound/soc/intel/skylake/skl-dsp-parse.c b/sound/soc/intel/skylake/skl-dsp-parse.c new file mode 100644 index 000000000000..a3cd5d7ab246 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.c @@ -0,0 +1,118 @@ +/* + * skl-dsp-parse.c - Implements DSP firmware parsing + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * 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 <linux/module.h> +#include <linux/types.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include "../common/sst-dsp.h" +#include "../common/sst-dsp-priv.h" +#include "skl-tplg-interface.h" +#include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" + +/* + * Get the module id for the module by checking + * the table for the UUID for the module + */ +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config) +{ + int i, num; + struct uuid_tbl *tbl; + uuid_le *uuid_mod; + + tbl = ctx->tbl; + num = ctx->num_modules; + uuid_mod = (uuid_le *)uuid; + + for (i = 0; i < num; i++) { + if (uuid_le_cmp(*uuid_mod, tbl[i].uuid) == 0) { + dfw_config->module_id = tbl[i].module_id; + dfw_config->is_loadable = tbl[i].is_loadable; + return 0; + } + } + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_skl_get_module_info); + +/* + * Parse the firmware binary to get the UUID, module id + * and loadable flags + */ +int snd_skl_parse_fw_bin(struct sst_dsp *ctx) +{ + struct adsp_fw_hdr *adsp_hdr; + struct adsp_module_entry *mod_entry; + int i, num_entry; + uuid_le *uuid_bin; + const char *buf; + struct skl_sst *skl = ctx->thread_context; + struct uuid_tbl *tbl; + + /* Get the FW pointer to derive ADSP header */ + buf = ctx->fw->data; + + adsp_hdr = (struct adsp_fw_hdr *)(buf + SKL_ADSP_FW_BIN_HDR_OFFSET); + + mod_entry = (struct adsp_module_entry *) + (buf + SKL_ADSP_FW_BIN_HDR_OFFSET + adsp_hdr->header_len); + + num_entry = adsp_hdr->num_module_entries; + + tbl = devm_kzalloc(ctx->dev, + num_entry * sizeof(struct uuid_tbl), GFP_KERNEL); + + if (!tbl) + return -ENOMEM; + + /* + * Read the UUID(GUID) from FW Manifest. + * The 16 byte UUID is of the format: + * XXXXXXXX-XXXX-XXXX-XXXX-XXXXXXXXXX + * Populate the UUID table to store module_id + * and loadable flags for the module. + */ + + for (i = 0; i < num_entry; i++, mod_entry++) { + + /* + * we check if current pointer is larger than file size from + * base value to check excceding the file while parsing + */ + if ((const char *)mod_entry >= buf + ctx->fw->size) { + dev_err(ctx->dev, + "Exceeds file bound: Entry %d Ptr %p\n", + i, mod_entry); + + return -EIO; + } + + uuid_bin = (uuid_le *)mod_entry->uuid.id; + memcpy(&tbl[i].uuid, uuid_bin, sizeof(tbl[i].uuid)); + + tbl[i].module_id = i; + tbl[i].is_loadable = mod_entry->type.load_type; + } + + skl->tbl = tbl; + skl->num_modules = num_entry; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_skl_parse_fw_bin); diff --git a/sound/soc/intel/skylake/skl-dsp-parse.h b/sound/soc/intel/skylake/skl-dsp-parse.h new file mode 100644 index 000000000000..cfdb24c45035 --- /dev/null +++ b/sound/soc/intel/skylake/skl-dsp-parse.h @@ -0,0 +1,107 @@ +/* + * skl-dsp-parse.h + * + * Copyright (C) 2016 Intel Corp + * Author: Shreyas NC <shreyas.nc(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as version 2, as + * published by the Free Software Foundation. + * + * 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 <asm/types.h> +#include <linux/types.h> +#include <linux/uuid.h> +#include "skl-tplg-interface.h" +#include "../common/sst-dsp-priv.h" + +#define SKL_ADSP_FW_BIN_HDR_OFFSET 0x284 +#define UUID_STR_SIZE 37 +#define DEFAULT_HASH_SHA256_LEN 32 + +struct skl_dfw_module_mod { + char name[100]; + struct skl_dfw_module skl_dfw_mod; +}; + +struct UUID { + u8 id[16]; +}; + +union seg_flags { + u32 ul; + struct { + u32 contents : 1; + u32 alloc : 1; + u32 load : 1; + u32 read_only : 1; + u32 code : 1; + u32 data : 1; + u32 _rsvd0 : 2; + u32 type : 4; + u32 _rsvd1 : 4; + u32 length : 16; + } r; +} __packed; + +struct segment_desc { + union seg_flags flags; + u32 v_base_addr; + u32 file_offset; +}; + +struct module_type { + u32 load_type : 4; + u32 auto_start : 1; + u32 domain_ll : 1; + u32 domain_dp : 1; + u32 rsvd_ : 25; +} __packed; + +struct adsp_module_entry { + u32 struct_id; + u8 name[8]; + struct UUID uuid; + struct module_type type; + u8 hash1[DEFAULT_HASH_SHA256_LEN]; + u32 entry_point; + u16 cfg_offset; + u16 cfg_count; + u32 affinity_mask; + u16 instance_max_count; + u16 instance_bss_size; + struct segment_desc segments[3]; +} __packed; + +struct adsp_fw_hdr { + u32 header_id; + u32 header_len; + u8 name[8]; + u32 preload_page_count; + u32 fw_image_flags; + u32 feature_mask; + u16 major_version; + u16 minor_version; + u16 hotfix_version; + u16 build_version; + u32 num_module_entries; + u32 hw_buf_base_addr; + u32 hw_buf_length; + u32 load_offset; +} __packed; + +struct uuid_tbl { + uuid_le uuid; + int module_id; + int is_loadable; +}; + +int snd_skl_parse_fw_bin(struct sst_dsp *ctx); +int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, + struct skl_dfw_module *dfw_config); diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index d59d1ba62a43..67d56be8cb77 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -60,6 +60,10 @@ struct skl_sst { void (*enable_miscbdcge)(struct device *dev, bool enable); /*Is CGCTL.MISCBDCGE disabled*/ bool miscbdcg_disabled; + + /* Populate module information */ + struct uuid_tbl *tbl; + int num_modules; }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl-sst.c b/sound/soc/intel/skylake/skl-sst.c index bec4a7c486fd..df7964fd09c3 100644 --- a/sound/soc/intel/skylake/skl-sst.c +++ b/sound/soc/intel/skylake/skl-sst.c @@ -25,6 +25,7 @@ #include "../common/sst-dsp-priv.h" #include "../common/sst-ipc.h" #include "skl-sst-ipc.h" +#include "skl-dsp-parse.h" #define SKL_BASEFW_TIMEOUT 300 #define SKL_INIT_TIMEOUT 1000 @@ -70,7 +71,7 @@ static int skl_transfer_firmware(struct sst_dsp *ctx, static int skl_load_base_firmware(struct sst_dsp *ctx) { - int ret = 0, i; + int ret = 0, i, err; struct skl_sst *skl = ctx->thread_context; u32 reg; @@ -84,6 +85,14 @@ static int skl_load_base_firmware(struct sst_dsp *ctx) skl_dsp_disable_core(ctx); return -EIO; } + + err = snd_skl_parse_fw_bin(ctx); + if (err < 0) { + dev_err(ctx->dev, + "Firmware parsing err: %d\n", ret); + release_firmware(ctx->fw); + return err; + } } ret = skl_dsp_boot(ctx); diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index a1f4478fabcb..f690b016203c 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -28,6 +28,7 @@ #include "skl-tplg-interface.h" #include "../common/sst-dsp.h" #include "../common/sst-dsp-priv.h" +#include "skl-dsp-parse.h" #define SKL_CH_FIXUP_MASK (1 << 0) #define SKL_RATE_FIXUP_MASK (1 << 1) @@ -1566,6 +1567,10 @@ static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, w->priv = mconfig; memcpy(&mconfig->guid, &dfw_config->uuid, 16); + ret = snd_skl_get_module_info(skl->skl_sst, mconfig->guid, dfw_config); + if (ret < 0) + return ret; + mconfig->id.module_id = dfw_config->module_id; mconfig->id.instance_id = dfw_config->instance_id; mconfig->mcps = dfw_config->max_mcps; -- 1.9.1

2 2

[alsa-devel] [PATCH v2 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 09 May '16

09 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. History: v2: Fix memory leak on realloc failure. The original obj pointer is reserved and so it can be freed at last by tplg_elem_free(). Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 26 +++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 133 insertions(+), 58 deletions(-) -- 2.5.0

2 7

[alsa-devel] [PATCH v2 1/2] ALSA: hda: fix the missing ptr initialization
by Vinod Koul 09 May '16

09 May '16

ebus is a member of extended device and was never initialized, so do this at device creation. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..64de0a3d6d93 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); -- 1.9.1

2 4

[alsa-devel] [PATCH] ALSA: dice: add support for M-Audio Profire 610 and perhaps 2626
by Takashi Sakamoto 09 May '16

09 May '16

M-Audio Profire 610 has an unexpected value in version field of its config ROM, thus ALSA dice driver is not assigned to the model due to a mismatch of modalias. This commit adds an entry to support the model. I expect the entry is also for Profire 2626. I note that Profire 610 uses TCD2220 (so-called Dice Jr.), and supports a part of Extended Application Protocol (EAP). $ cd linux-firewire-utils/src $ ./crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ------------------------------------------------------------ 400 04047689 bus_info_length 4, crc_length 4, crc 30345 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000d6c04 company_id 000d6c | 410 04400002 device_id 0404400002 | EUI-64 000d6c0404400002 root directory ------------------------------------------------------------ 414 000695fe directory_length 6, crc 38398 418 03000d6c vendor 41c 8100000a --> descriptor leaf at 444 420 17000011 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ------------------------------------------------------------ 430 0004fb14 directory_length 4, crc 64276 434 12000d6c specifier id 438 130100d1 version 43c 17000011 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ------------------------------------------------------------ 444 0004b8e4 leaf_length 4, crc 47332 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 4d2d4175 "M-Au" 454 64696f00 "dio" descriptor leaf at 458 ------------------------------------------------------------ 458 00053128 leaf_length 5, crc 12584 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 50726f46 "ProF" 468 69726520 "ire " 46c 36313000 "610" descriptor leaf at 470 ------------------------------------------------------------ 470 00053128 leaf_length 5, crc 12584 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 50726f46 "ProF" 480 69726520 "ire " 484 36313000 "610" $ cat /proc/asound/card1/dice sections: global: offset 10, size 90 tx: offset 100, size 142 rx: offset 242, size 282 ext_sync: offset 524, size 4 unused2: offset 0, size 0 global: owner: ffc0:000100000000 notification: 00000040 nick name: FW610 clock select: internal 48000 enable: 1 status: locked 48000 ext status: 00000040 sample rate: 48000 version: 1.0.4.0 clock caps: 32000 44100 48000 88200 96000 176400 192000 aes1 aes4 aes adat tdif wc arx1 arx2 internal clock source names: SPDIF\AES34\AES56\TOS\AES_ANY\ADAT\ADAT_AUX\Word Clock\Unused\Unused\Unused\Unused\Internal\\ ... Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/dice.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index 96fe68f4..25e9f77 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -317,6 +317,13 @@ static const struct ieee1394_device_id dice_id_table[] = { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, }, + /* M-Audio Profire 610/2626 has a different value in version field. */ + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_SPECIFIER_ID, + .vendor_id = 0x000d6c, + .specifier_id = 0x000d6c, + }, { } }; MODULE_DEVICE_TABLE(ieee1394, dice_id_table); -- 2.7.4

2 4

[alsa-devel] [PATCH] pcm: softvol: fix signedness of tlv data type
by Jörg Krause 08 May '16

08 May '16

For the softvol plugin the TLV data are written wrong, e.g. for the default values of min_dB = -51 dB and max_dB = 0 dB, alsactl generates the following state: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } As both min_dB and max_dB can be negative numbers, the tlv type must not be unsigned. This fixes the TLV and the correct state is generated: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Also tested for different combinations of min_dB and max_dB other than the default values. Signed-off-by: Jörg Krause <joerg.krause(a)embedded.rocks> --- src/pcm/pcm_softvol.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_softvol.c b/src/pcm/pcm_softvol.c index 802aa4b..1748667 100644 --- a/src/pcm/pcm_softvol.c +++ b/src/pcm/pcm_softvol.c @@ -655,7 +655,7 @@ static void snd_pcm_softvol_dump(snd_pcm_t *pcm, snd_output_t *out) static int add_tlv_info(snd_pcm_softvol_t *svol, snd_ctl_elem_info_t *cinfo) { - unsigned int tlv[4]; + int tlv[4]; tlv[0] = SND_CTL_TLVT_DB_SCALE; tlv[1] = 2 * sizeof(int); tlv[2] = svol->min_dB * 100; @@ -768,7 +768,7 @@ static int softvol_load_control(snd_pcm_t *pcm, snd_pcm_softvol_t *svol, } } else if (svol->max_val > 1) { /* check TLV availability */ - unsigned int tlv[4]; + int tlv[4]; err = snd_ctl_elem_tlv_read(svol->ctl, &cinfo->id, tlv, sizeof(tlv)); if (err < 0) add_tlv_info(svol, cinfo); -- 2.8.2

2 2

[alsa-devel] [PATCH 0/6] topology: complete PCM parsing and code refactoring
by mengdong.lin＠linux.intel.com 08 May '16

08 May '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> There is no ABI change in this series. This series fixes bugs and missing info of PCM (Front-end DAI & DAI link) defined by the text conf file. There is also some code refactoring, as a preparation for Back-end DAI support and ABI update which is under internal test and review atm. Mengdong Lin (6): topology: Set manifest size for ABI topology: Refactor functions to parse and build streams topology: Use generic pointer to realloc buffer for private data topology: Fix pcm ID & name parsing topology: Parse front-end DAI name and ID for the PCM topology: Update PCM configurations in Broadwell text conf file include/sound/asoc.h | 2 +- include/topology.h | 4 + src/conf/topology/broadwell/broadwell.conf | 34 ++++++--- src/topology/data.c | 24 ++---- src/topology/parser.c | 4 +- src/topology/pcm.c | 119 ++++++++++++++++++++++------- src/topology/tplg_local.h | 2 +- 7 files changed, 131 insertions(+), 58 deletions(-) -- 2.5.0

5 11

Re: [alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Kailang 08 May '16

08 May '16

Hi Takashi, Could you receive this mail? Thanks. BR, Kailang ________________________________ From: Kailang Sent: Wednesday, May 04, 2016 4:13 PM To: Takashi Iwai (tiwai(a)suse.de) Cc: (alsa-devel(a)alsa-project.org) Subject: New codecs support for ALC234/ALC274/ALC294 Hi Takashi, We have new codecs for ALC234/ALC274/ALC294. Attach file is the support patch. BR, Kailang

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Re: [alsa-devel] ALSA: hda - Fix white noise on Asus UX501VW headset
by Takashi Iwai 08 May '16

08 May '16

On Thu, 05 May 2016 16:08:02 +0200, Kaho Ng wrote: > > For reducing the noise from the headset output on ASUS UX501VW, > call the existing fixup, alc_fixup_headset_mode_alc668(), additionally. > > Thread: https://bbs.archlinux.org/viewtopic.php?id=209554 > > Signed-off-by: Kaho Ng <ngkaho1234(a)gmail.com> > --- > diff -ruNp a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > --- a/sound/pci/hda/patch_realtek.c 2016-04-20 14:44:02.000000000 +0800 > +++ b/sound/pci/hda/patch_realtek.c 2016-05-03 16:31:30.815677146 +0800 > @@ -6424,6 +6424,7 @@ enum { > ALC668_FIXUP_DELL_DISABLE_AAMIX, > ALC668_FIXUP_DELL_XPS13, > ALC662_FIXUP_ASUS_Nx50, > + ALC668_FIXUP_ASUS_UX501VW, > }; > > static const struct hda_fixup alc662_fixups[] = { > @@ -6670,6 +6671,10 @@ static const struct hda_fixup alc662_fix > .chained = true, > .chain_id = ALC662_FIXUP_BASS_1A > }, > + [ALC668_FIXUP_ASUS_UX501VW] = { > + .type = HDA_FIXUP_FUNC, > + .v.func = alc_fixup_headset_mode_alc668, > + }, > }; There is already a fixup entry (ALC668_FIXUP_HEADSET_MODE) doing the very same thing, so you don't have to redefine it. > static const struct snd_pci_quirk alc662_fixup_tbl[] = { > @@ -6692,6 +6697,7 @@ static const struct snd_pci_quirk alc662 > SND_PCI_QUIRK(0x1028, 0x0698, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x1028, 0x069f, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), > SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), > + SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_ASUS_UX501VW), And just add the quirk entry like SND_PCI_QUIRK(0x1043, 0x1080, "Asus UX501VW", ALC668_FIXUP_HEADSET_MODE), Could you resubmit the fix patch after adjusting like this? thanks, Takashi

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Re: [alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Takashi Iwai 08 May '16

08 May '16

On Wed, 04 May 2016 10:12:41 +0200, Kailang wrote: > > Hi Takashi, > > We have new codecs for ALC234/ALC274/ALC294. > Attach file is the support patch. Thanks, applied now. Takashi

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[alsa-devel] [patch] ALSA: isa/wavefront: prevent some out of bound writes
by Dan Carpenter 08 May '16

08 May '16

"header->number" can be up to USHRT_MAX and it comes from the ioctl so it needs to be capped. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69f76ff..718d5e3 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -785,6 +785,9 @@ wavefront_send_patch (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading patch %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->patch_status)) + return -EINVAL; + dev->patch_status[header->number] |= WF_SLOT_FILLED; bptr = buf; @@ -809,6 +812,9 @@ wavefront_send_program (snd_wavefront_t *dev, wavefront_patch_info *header) DPRINT (WF_DEBUG_LOAD_PATCH, "downloading program %d\n", header->number); + if (header->number >= ARRAY_SIZE(dev->prog_status)) + return -EINVAL; + dev->prog_status[header->number] = WF_SLOT_USED; /* XXX need to zero existing SLOT_USED bit for program_status[i] @@ -898,6 +904,9 @@ wavefront_send_sample (snd_wavefront_t *dev, header->number = x; } + if (header->number >= WF_MAX_SAMPLE) + return -EINVAL; + if (header->size) { /* XXX it's a debatable point whether or not RDONLY semantics

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Re: [alsa-devel] [PATCH] infoleak fix1 in timer
by Takashi Iwai 08 May '16

08 May '16

On Tue, 03 May 2016 22:44:07 +0200, Kangjie Lu wrote: > > The stack object “tread” has a total size of 32 bytes. Its field > “event” and “val” both contain 4 bytes padding. These 8 bytes > padding bytes are sent to user without being initialized. > > Signed-off-by: Kangjie Lu <kjlu(a)gatech.edu> Thanks, I applied now all three patches. But I rephrased the subjects to more meaningful ones. Better to give a bit more informative subject line at the next time. Takashi

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[alsa-devel] [PATCH] ALSA: compress: fix some typo
by Vinod Koul 08 May '16

08 May '16

Noticed two typos in Documentation, so fix them up Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- Documentation/sound/alsa/compress_offload.txt | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/compress_offload.txt b/Documentation/sound/alsa/compress_offload.txt index 630c492c3dc2..8ba556a131c3 100644 --- a/Documentation/sound/alsa/compress_offload.txt +++ b/Documentation/sound/alsa/compress_offload.txt @@ -149,7 +149,7 @@ Gapless Playback ================ When playing thru an album, the decoders have the ability to skip the encoder delay and padding and directly move from one track content to another. The end -user can perceive this as gapless playback as we dont have silence while +user can perceive this as gapless playback as we don't have silence while switching from one track to another Also, there might be low-intensity noises due to encoding. Perfect gapless is @@ -184,7 +184,7 @@ Sequence flow for gapless would be: - Fill data of the first track - Trigger start - User-space finished sending all, -- Indicaite next track data by sending set_next_track +- Indicate next track data by sending set_next_track - Set metadata of the next track - then call partial_drain to flush most of buffer in DSP - Fill data of the next track -- 1.9.1

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[alsa-devel] sound: deadlock involving snd_hrtimer_callback
by Dmitry Vyukov 08 May '16

08 May '16

Hi Takashi, I've incorporated your hrtimer fixes (but also updated to ddce192106e4f984123884f8e878f66ace94b573) and now I am seeing lots of the following deadlock messages: [ INFO: possible circular locking dependency detected ] 4.6.0-rc4+ #351 Not tainted ------------------------------------------------------- swapper/0/0 is trying to acquire lock: (&(&timer->lock)->rlock){-.-...}, at: [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 but task is already holding lock: (&(&stime->lock)->rlock){-.....}, at: [<ffffffff85383d3f>] snd_hrtimer_callback+0x4f/0x2b0 sound/core/hrtimer.c:54 which lock already depends on the new lock. the existing dependency chain (in reverse order) is: -> #1 (&(&stime->lock)->rlock){-.....}: [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff85383baa>] snd_hrtimer_start+0x4a/0xf0 sound/core/hrtimer.c:112 [<ffffffff85379c74>] snd_timer_start1+0x2b4/0x5a0 sound/core/timer.c:457 [<ffffffff8537d33d>] snd_timer_start+0x5d/0xa0 sound/core/timer.c:571 [< inline >] seq_timer_start sound/core/seq/seq_timer.c:393 [<ffffffff853fc260>] snd_seq_timer_start+0x1a0/0x2b0 sound/core/seq/seq_timer.c:405 [< inline >] snd_seq_queue_process_event sound/core/seq/seq_queue.c:687 [<ffffffff853f6f14>] snd_seq_control_queue+0x304/0x8b0 sound/core/seq/seq_queue.c:748 [<ffffffff853fcbb5>] event_input_timer+0x25/0x30 sound/core/seq/seq_system.c:118 [<ffffffff853ef7d4>] snd_seq_deliver_single_event.constprop.11+0x3f4/0x740 sound/core/seq/seq_clientmgr.c:636 [<ffffffff853efc38>] snd_seq_deliver_event+0x118/0x800 sound/core/seq/seq_clientmgr.c:833 [<ffffffff853f1026>] snd_seq_kernel_client_dispatch+0x126/0x170 sound/core/seq/seq_clientmgr.c:2418 [<ffffffff85405c2b>] send_timer_event.isra.0+0x10b/0x150 sound/core/seq/oss/seq_oss_timer.c:153 [<ffffffff8540615a>] snd_seq_oss_timer_start+0x1ca/0x310 sound/core/seq/oss/seq_oss_timer.c:174 [< inline >] old_event sound/core/seq/oss/seq_oss_event.c:125 [<ffffffff854095ff>] snd_seq_oss_process_event+0xa1f/0x2ce0 sound/core/seq/oss/seq_oss_event.c:100 [< inline >] insert_queue sound/core/seq/oss/seq_oss_rw.c:179 [<ffffffff8540c201>] snd_seq_oss_write+0x321/0x810 sound/core/seq/oss/seq_oss_rw.c:148 [<ffffffff85403f99>] odev_write+0x59/0xa0 sound/core/seq/oss/seq_oss.c:177 [<ffffffff817f3fc3>] __vfs_write+0x113/0x4b0 fs/read_write.c:529 [<ffffffff817f5a97>] vfs_write+0x167/0x4a0 fs/read_write.c:578 [< inline >] SYSC_write fs/read_write.c:625 [<ffffffff817f98d1>] SyS_write+0x111/0x220 fs/read_write.c:617 [<ffffffff867c0440>] entry_SYSCALL_64_fastpath+0x23/0xc1 arch/x86/entry/entry_64.S:207 -> #0 (&(&timer->lock)->rlock){-.-...}: [< inline >] check_prev_add kernel/locking/lockdep.c:1823 [< inline >] check_prevs_add kernel/locking/lockdep.c:1933 [< inline >] validate_chain kernel/locking/lockdep.c:2238 [<ffffffff81474035>] __lock_acquire+0x3625/0x4d00 kernel/locking/lockdep.c:3298 [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [< inline >] arch_safe_halt ./arch/x86/include/asm/paravirt.h:118 [<ffffffff8120f402>] default_idle+0x52/0x370 arch/x86/kernel/process.c:307 [<ffffffff81210bba>] arch_cpu_idle+0xa/0x10 arch/x86/kernel/process.c:298 [<ffffffff81454c28>] default_idle_call+0x48/0xa0 kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call kernel/sched/idle.c:151 [< inline >] cpu_idle_loop kernel/sched/idle.c:242 [<ffffffff8145520f>] cpu_startup_entry+0x58f/0x7b0 kernel/sched/idle.c:291 [<ffffffff8679ac6d>] rest_init+0x18d/0x1a0 init/main.c:408 [<ffffffff88ce7759>] start_kernel+0x63a/0x660 init/main.c:661 [<ffffffff88ce6364>] x86_64_start_reservations+0x38/0x3a arch/x86/kernel/head64.c:195 [<ffffffff88ce64be>] x86_64_start_kernel+0x158/0x167 arch/x86/kernel/head64.c:176 other info that might help us debug this: Possible unsafe locking scenario: CPU0 CPU1 ---- ---- lock(&(&stime->lock)->rlock); lock(&(&timer->lock)->rlock); lock(&(&stime->lock)->rlock); lock(&(&timer->lock)->rlock); *** DEADLOCK *** 1 lock held by swapper/0/0: #0: (&(&stime->lock)->rlock){-.....}, at: [<ffffffff85383d3f>] snd_hrtimer_callback+0x4f/0x2b0 sound/core/hrtimer.c:54 stack backtrace: CPU: 0 PID: 0 Comm: swapper/0 Not tainted 4.6.0-rc4+ #351 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88003ec07a50 ffffffff82c8f92f ffffffff00000000 fffffbfff0fd64b8 ffffffff8997bad0 ffffffff8997bad0 ffffffff8998aa70 ffffffff87e27510 ffffffff87e26d00 ffff88003ec07aa0 ffffffff8146cf58 Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82c8f92f>] dump_stack+0x12e/0x18f lib/dump_stack.c:51 [<ffffffff8146cf58>] print_circular_bug+0x288/0x340 kernel/locking/lockdep.c:1196 [< inline >] check_prev_add kernel/locking/lockdep.c:1823 [< inline >] check_prevs_add kernel/locking/lockdep.c:1933 [< inline >] validate_chain kernel/locking/lockdep.c:2238 [<ffffffff81474035>] __lock_acquire+0x3625/0x4d00 kernel/locking/lockdep.c:3298 [<ffffffff81477693>] lock_acquire+0x1e3/0x460 kernel/locking/lockdep.c:3677 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:112 [<ffffffff867c005f>] _raw_spin_lock_irqsave+0x9f/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 [< inline >] arch_safe_halt ./arch/x86/include/asm/paravirt.h:118 [<ffffffff8120f402>] default_idle+0x52/0x370 arch/x86/kernel/process.c:307 [<ffffffff81210bba>] arch_cpu_idle+0xa/0x10 arch/x86/kernel/process.c:298 [<ffffffff81454c28>] default_idle_call+0x48/0xa0 kernel/sched/idle.c:93 [< inline >] cpuidle_idle_call kernel/sched/idle.c:151 [< inline >] cpu_idle_loop kernel/sched/idle.c:242 [<ffffffff8145520f>] cpu_startup_entry+0x58f/0x7b0 kernel/sched/idle.c:291 [<ffffffff8679ac6d>] rest_init+0x18d/0x1a0 init/main.c:408 [<ffffffff88ce7759>] start_kernel+0x63a/0x660 init/main.c:661 [<ffffffff88ce6364>] x86_64_start_reservations+0x38/0x3a arch/x86/kernel/head64.c:195 [<ffffffff88ce64be>] x86_64_start_kernel+0x158/0x167 arch/x86/kernel/head64.c:176 They are also followed by rcu stalls and actual spinlock lockups: BUG: spinlock lockup suspected on CPU#3, syz-fuzzer/7040 lock: 0xffff880032ae2bd8, .magic: dead4ead, .owner: syz-executor/11206, .owner_cpu: 0 CPU: 3 PID: 7040 Comm: syz-fuzzer Not tainted 4.6.0-rc4+ #351 Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS Bochs 01/01/2011 ffffffff87eb25c0 ffff88006d507c40 ffffffff82c8f92f ffffffff2c963698 fffffbfff0fd64b8 ffff880032ae2bd8 ffff88002c963040 ffff880061238040 ffff880032ae2be0 000000009a9d0630 ffff88006d507c78 ffffffff8147f58d Call Trace: <IRQ> [< inline >] __dump_stack lib/dump_stack.c:15 <IRQ> [<ffffffff82c8f92f>] dump_stack+0x12e/0x18f lib/dump_stack.c:51 [<ffffffff8147f58d>] spin_dump+0x14d/0x280 kernel/locking/spinlock_debug.c:67 [< inline >] __spin_lock_debug kernel/locking/spinlock_debug.c:117 [<ffffffff8147f8ad>] do_raw_spin_lock+0x15d/0x2b0 kernel/locking/spinlock_debug.c:137 [< inline >] __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:119 [<ffffffff867c0067>] _raw_spin_lock_irqsave+0xa7/0xd0 kernel/locking/spinlock.c:159 [<ffffffff8537a749>] snd_timer_interrupt+0xa9/0xd30 sound/core/timer.c:701 [<ffffffff85383e75>] snd_hrtimer_callback+0x185/0x2b0 sound/core/hrtimer.c:59 [< inline >] __run_hrtimer kernel/time/hrtimer.c:1242 [<ffffffff814d1071>] __hrtimer_run_queues+0x331/0xe90 kernel/time/hrtimer.c:1306 [<ffffffff814d3a42>] hrtimer_interrupt+0x182/0x430 kernel/time/hrtimer.c:1340 [<ffffffff8125aa52>] local_apic_timer_interrupt+0x72/0xe0 arch/x86/kernel/apic/apic.c:907 [<ffffffff867c2d59>] smp_apic_timer_interrupt+0x79/0xa0 arch/x86/kernel/apic/apic.c:931 [<ffffffff867c10ac>] apic_timer_interrupt+0x8c/0xa0 arch/x86/entry/entry_64.S:454 <EOI>

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[alsa-devel] Unlikely big volume range Plantronics BT300
by Dennis Kadioglu 08 May '16

08 May '16

Hi, when I connect the bluetooth BT300 of my Plantronics Headset I can see the following warning: [172855.985539] usb 2-1.3: Warning! Unlikely big volume range (=720), cval->res is probably wrong. [172855.985544] usb 2-1.3: [5] FU [PCM Playback Volume] ch = 1, val = -10240/1280/16 As far as I understand it, the problem may be solved by applying a quirk which is already applied for Logitech webcams. Can somebody help me in finding out the correct value for cval->res. I have no clue which calculations are necessary for that. Thanks in advance. Dennis Kadioglu

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Re: [alsa-devel] WM5102 - Help to make baytrail machine driver work
by Pierre-Louis Bossart 07 May '16

07 May '16

On 5/4/16 8:21 PM, Paulo Sergio wrote: > Hi list, > > I've been trying to make Lenovo Yoga 2 (Model 1051F) sound work on > Android-x86/ Linux. [1] > > Based on already discussed info from Christian Hartmann with members > (see [2] and [3] ) of this list and some efforts (basically patch pxa2xx > with Mika Westerberg changes, set the correct gpio int values {246 for > reset / 405 for ldoena and 342 for irq_gpio} and set LDO1 vcc suppliers) > we could have wm5102 initialized through spi / regmap (arizona code on > drivers/mfd) and register the DAI's. > > I also ported a machine driver code from a kernel Lenovo made available > for a similar hardware (Lenovo yoga 2 1050 - uses the same baytrail > Z3745 and wolfson wm5102 audio) [4] There are quite a few references in the machine driver to concepts related to Clovertrail+ and early versions of Baytrail. I am fairly convinced the mentions of 'BYT_COMMS_BT' or 'modem mixing' are no longer supported with the DPCM platform driver/firmware you used. To enable this codec it would be simpler to start with a newer machine driver derived from the bytcr_rt5640 one, with just a basic SSP0-aif1 link for starters (+ MCLK). Better yet, do this work with a MinnowBoard and an WM5102 eval board and test all this with more control on how the hardware is setup. If you stick to working directly on the Lenovo device, make sure you use all the DSP controls set in the UCM files used for baytrail/baytrail-CR (see github.com/plbossart/UCM). > > Although machine driver is registered, we still don't have sound. Tried > some mixer settings / routes used on raspberry pi, but none help. > Application (in this case alsa / Android players) returns a "Device or > resource busy" when trying to ioctl pcm device: > > 04-30 13:05:07.877 1784 2078 E audio_hw_primary: out_write: error -1 - cannot prepare channel: Device or resource busy > > # This is raised here (external/tinyalsa/pcm.c): > > if (ioctl(pcm->fd, SNDRV_PCM_IOCTL_PREPARE) < 0) > > return oops(pcm, errno, "cannot prepare channel"); > > > Note: As pointed here [5] had to change the acpi_ipc_irq_index to 0 on > sound/soc/intel/atom/sst/sst_acpi.c because this is the 1st ACPI entry > listed on DSDT > I'm also trying to use a fw version that uses SSP0 [6] instead of SSP2 > used as default - but no sound with both > > > I'd like some help from you in order to make this work. I've collect > several info which can be found here [7] > - /proc/asound/cards > - /proc/interrupts > - amixer controls list > - dmesg > - DSDT file (decoded with iasl) > > > Source code with changes made so far is available on github [8] > > Any hint is appreciated > > Thanks, > Paulo Sérgio Travaglia (pstglia) > > > [1] - https://groups.google.com/forum/#!forum/android-x86 > [2] - http://www.spinics.net/lists/linux-acpi/msg58680.html > [3] - https://goo.gl/vBKOeI > [4] - http://goo.gl/zriwbw > [5] - https://bugzilla.kernel.org/show_bug.cgi?id=86581 > [6] - http://thread.gmane.org/gmane.linux.alsa.devel/134554/focus=140634 > [7] > - https://drive.google.com/folderview?id=0BxO6THtB865fWjVHLTFfV25ORjQ&usp=sha… > [8] - https://github.com/pstglia/linux

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[alsa-devel] [PATCH 0/7] ASoC: Mediatek: Add support for MT2701 SOC
by Garlic Tseng 06 May '16

06 May '16

This patch adds basic support for Mediatek AFE for MT2701 SoC. The patch is based on broonie tree "for-next" branch. Change since RFC v2: add BT function refine Kconfig and Makefile add prefix 'mt8173' and 'mt2701' for relative drivers. Garlic Tseng (7): ASoC: mediatek: Refine mt8173 driver and change config option ASoC: mediatek: add documents for mt2701 ASoC: mediatek: add clock and irq control for 2701 platform driver ASoC: mediatek: add mt2701 platform driver implementation. ASoC: bt-sco: extend rate and add a general compatible string ASoC: mediatek: add BT implementation ASoC: mediatek: Add mt2701-cs42448 driver and config option. Documentation/devicetree/bindings/sound/bt-sco.txt | 2 +- .../devicetree/bindings/sound/mt2701-afe-pcm.txt | 150 ++ .../devicetree/bindings/sound/mt2701-cs42448.txt | 43 + sound/soc/codecs/bt-sco.c | 5 +- sound/soc/mediatek/Kconfig | 35 +- sound/soc/mediatek/Makefile | 9 +- sound/soc/mediatek/mt2701/Makefile | 20 + sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c | 344 ++++ sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h | 33 + sound/soc/mediatek/mt2701/mt2701-afe-common.h | 228 +++ sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 1825 ++++++++++++++++++++ sound/soc/mediatek/mt2701/mt2701-cs42448.c | 412 +++++ sound/soc/mediatek/mt2701/mt2701-irq.c | 74 + sound/soc/mediatek/mt2701/mt2701-irq.h | 26 + sound/soc/mediatek/mt2701/mt2701-reg.h | 195 +++ sound/soc/mediatek/mt8173/Makefile | 7 + sound/soc/mediatek/mt8173/mt8173-afe-common.h | 101 ++ .../{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} | 488 +++--- sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 101 -- 23 files changed, 3740 insertions(+), 368 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/mt2701-cs42448.txt create mode 100644 sound/soc/mediatek/mt2701/Makefile create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-clock-ctrl.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-common.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-afe-pcm.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-cs42448.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-irq.c create mode 100644 sound/soc/mediatek/mt2701/mt2701-irq.h create mode 100644 sound/soc/mediatek/mt2701/mt2701-reg.h create mode 100644 sound/soc/mediatek/mt8173/Makefile create mode 100644 sound/soc/mediatek/mt8173/mt8173-afe-common.h rename sound/soc/mediatek/{mtk-afe-pcm.c => mt8173/mt8173-afe-pcm.c} (67%) rename sound/soc/mediatek/{ => mt8173}/mt8173-max98090.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5514.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650-rt5676.c (99%) rename sound/soc/mediatek/{ => mt8173}/mt8173-rt5650.c (99%) delete mode 100644 sound/soc/mediatek/mtk-afe-common.h -- 1.9.1

3 13

[alsa-devel] New codecs support for ALC234/ALC274/ALC294
by Kailang 06 May '16

06 May '16

Hi Takashi, We have new codecs for ALC234/ALC274/ALC294. Attach file is the support patch. BR, Kailang

1 0

[alsa-devel] Errors iterating over PCM stream hints and opening based on NAME/IOID
by Trent Reed 06 May '16

06 May '16

Hey All, I'm trying to understand ALSA, having a little trouble with some errors that I can't ration over. Gist Repro: https://gist.github.com/TReed0803/353ce27d37bc1ebe965854ea88c8b97b Whenever I run the above simple code (iterate over PCM stream hints, open for Playback/Capture depending on IOID; output also provided at link above) I notice that I sometimes hit errors. I'm having trouble understanding these errors, because as I understand I'm using the ALSA library correctly. For example, dmix is telling me that it "Only supports playback streams", well then, why does the hint IOID return that it supports capture as well? (same for dsnoop). More confusing are the "ALSA lib pcm_route.c:867:(find_matching_chmap) Found no matching channel map" errors. Can someone help me understand these issues, and perhaps what I'm doing wrong here? Would this point to something wrong with an installed package which perhaps configures these hints incorrectly? (Assuming that is probably the case, but I'd like to know for sure). I'm on a fairly new install of Ubuntu 16.04 LTS. Thanks, - Trent Reed

1 0

[alsa-devel] Dice packet sequence quirk and ALSA firewire stack in Linux 4.6
by Takashi Sakamoto 06 May '16

06 May '16

Hi all, In this developing cycle for Linux 4.7, I concentrate on working to check packet sequence of ALSA firewire stack, because I ensure that the stack causes Dice-based models to generate sound including periodical noise or to generate no sound in the worst case. As of the end of developing cycle for Linux 4.6, I assumed that this is a bug of ALSA firewire stack[1]. However, the assumption is denied by my analysis with added tracepoints of Linux tracing framework[2]. Then, I concluded that the issue comes from quirks of Dice in an aspect of processing presentation timestamp. I'm under further investigation and have seen some of the quirks. In this message, I describe the summary of superficial issues from the quirks, the status of ALSA firewire stack in Linux 4.6 for Dice-based models and direction of my work for ALSA firewire stack in this developing cycle. Besides, resources and my memo for this work is available in this URL. I wish to upload it to alsa-project.org, not in my server: http://sakamocchi.jp/media/20160505-dice-packet-dump.tar.xz (233MB archive file compressed by xz, expands 6.7 GB text files.) Summary - Dice ASICs/firmwares/drivers communicate with packet sequences which is partly not compliant to IEC 61883-6. - Dice ASICs/firmwares/drivers doesn't sometimes transfer data blocks as the same as sampling transfer frequency. - Calculation of presentation timestamp is not based on duration of data blocks in a packet. This way is different from implementation of ALSA firewire stack. - The value of estimated 'transfer delay' is against IEEE 1394 spec. As a result: - ALSA dice driver causes these issues. ALSA dice driver in Linux kernels since 3.13 have included this issue. - To fix this issue, ALSA firewire stack require timestamp generator and data block calculator specific for Dice-based models. At least, three functions should be changed; amdtp_stream_start(), calculate_data_blocks(), calculate_syt() in 'sound/firewire/amdtp-stream.c'. - The work requires much changes, thus it's impossible for Linux 4.6. ALSA firewire stack in Linux 4.6 will be shipped with these issues. - We have a short time till merge window of Linux 4.7 (Just one week). I'll work for merging my proposed patchset[2][3][4], without fixing the issues. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-March/105608.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/106950.html [3] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107290.html [4] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-April/107531.html Regards Takashi Sakamoto

1 0

[alsa-devel] I-MX6 & I2s output
by nick83ola 05 May '16

05 May '16

Hello to All I have an IMX6 board and I need a simple way to capture/put an I2s master signal I'm working with yocto and kernel version 3.14.28-1.0.0 I have on my development board an sgtl5000 codec but it work only with I2S slave mode I've found the driver fsl-asoc-card.c from more recent kernel and I've modified it But no matter what I do i cannot set I2S to master mode Please Help -- P.S. Le informazioni trasmesse attraverso la presente comunicazione sono di esclusiva spettanza dell'effettivo destinatario. Nel caso in cui le stesse raggiungessero, per qualunque motivo, soggetti non interessati, questi ultimi vorranno darne immediata notizia al mittente. In ogni caso, eventuali soggetti diversi dai legittimi destinatari della presente comunicazione e dei dati contenuti negli allegati, possono essere sanzionati ai sensi del T.U. sul trattamento dei dati personali d.lgs. 196/2003, sia ai sensi dell'art. 616 del Codice Penale che disciplina la violazione del segreto sulla corrispondenza.

1 0

[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Move compr_attach/attached functions
by Charles Keepax 05 May '16

05 May '16

Move wm_adsp_compr_attach and wm_adsp_compr_attached functions so they will stay logically grouped with similar functions after some additional changes. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 630ebcd..42fc469 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2452,6 +2452,25 @@ void wm_adsp2_remove(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_remove); +static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) +{ + return compr->buf != NULL; +} + +static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) +{ + /* + * Note this will be more complex once each DSP can support multiple + * streams + */ + if (!compr->dsp->buffer) + return -EINVAL; + + compr->buf = compr->dsp->buffer; + + return 0; +} + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; @@ -2810,25 +2829,6 @@ static int wm_adsp_buffer_free(struct wm_adsp *dsp) return 0; } -static inline int wm_adsp_compr_attached(struct wm_adsp_compr *compr) -{ - return compr->buf != NULL; -} - -static int wm_adsp_compr_attach(struct wm_adsp_compr *compr) -{ - /* - * Note this will be more complex once each DSP can support multiple - * streams - */ - if (!compr->dsp->buffer) - return -EINVAL; - - compr->buf = compr->dsp->buffer; - - return 0; -} - int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) { struct wm_adsp_compr *compr = stream->runtime->private_data; -- 2.1.4

2 3

[alsa-devel] [PATCH 1/2] Revert "ASoC: rockchip: i2s: remove unused variables"
by John Keeping 05 May '16

05 May '16

This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38. It turns out that the commit that made these variables unused is wrong so we're about to revert it. Bring back the variables in prepration. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..34743ec61c49 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -77,7 +84,11 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START, I2S_XFER_TXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); @@ -115,7 +126,11 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_RXS_START, I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); -- 2.8.0.rc4.238.g874082a

3 5

[alsa-devel] [PATCH] ASoC: rt5645: polling jd status in all conditions
by Bard Liao 05 May '16

05 May '16

We only polling jd status when rt5645->pdata.jd_invert is true. However, it should be done at all time since there will be no interrupt for jd if we press a headset button and remove the headset at the same time. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dff706a..3c6594d 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3286,10 +3286,8 @@ static void rt5645_jack_detect_work(struct work_struct *work) if (btn_type == 0)/* button release */ report = rt5645->jack_type; else { - if (rt5645->pdata.jd_invert) { - mod_timer(&rt5645->btn_check_timer, - msecs_to_jiffies(100)); - } + mod_timer(&rt5645->btn_check_timer, + msecs_to_jiffies(100)); } break; @@ -3816,9 +3814,9 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (rt5645->pdata.jd_invert) { regmap_update_bits(rt5645->regmap, RT5645_IRQ_CTRL2, RT5645_JD_1_1_MASK, RT5645_JD_1_1_INV); - setup_timer(&rt5645->btn_check_timer, - rt5645_btn_check_callback, (unsigned long)rt5645); } + setup_timer(&rt5645->btn_check_timer, + rt5645_btn_check_callback, (unsigned long)rt5645); INIT_DELAYED_WORK(&rt5645->jack_detect_work, rt5645_jack_detect_work); INIT_DELAYED_WORK(&rt5645->rcclock_work, rt5645_rcclock_work); -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v3 1/2] ASoC: fsl_sai: Introduce a compatible string for MX6UL
by Fabio Estevam 05 May '16

05 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> MX6UL may need to configure the General Purpose Register 1 (GPR1), so it is better to add a new compatible string to differentiate. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v2: - Add binding doc entry for mx6ul (Nicolin) Documentation/devicetree/bindings/sound/fsl-sai.txt | 4 ++-- sound/soc/fsl/fsl_sai.c | 4 +++- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d7..777b941 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,8 +7,8 @@ codec/DSP interfaces. Required properties: - - compatible : Compatible list, contains "fsl,vf610-sai" or - "fsl,imx6sx-sai". + - compatible : Compatible list, contains "fsl,vf610-sai", + "fsl,imx6sx-sai" or "fsl,imx6ul-sai" - reg : Offset and length of the register set for the device. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..d8b673f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 1.9.1

3 5

[alsa-devel] [PATCH 0/2] ASoC: Memeory leak fixes
by Vinod Koul 05 May '16

05 May '16

This series fixes two memeory leaks, first in topology core to freeup the objects in widget creation and second one on Skylake driver for NHLT table cleanup Jeeja KP (2): ASoC: topology: Fix memory leak in widget creation ASoC: Intel: Skylake: Fix memory leak in nhlt init sound/soc/intel/skylake/skl-nhlt.c | 15 +++++++++------ sound/soc/intel/skylake/skl.c | 5 ++++- sound/soc/intel/skylake/skl.h | 6 +++--- sound/soc/soc-topology.c | 2 ++ 4 files changed, 18 insertions(+), 10 deletions(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v2 1/2] ASoC: fsl_sai: Introduce a compatible string for MX6UL
by Fabio Estevam 05 May '16

05 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> MX6UL may need to configure the General Purpose Register 1 (GPR1), so it is better to add a new compatible string to differentiate. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Changes since v1: - Newly introduced in this series. sound/soc/fsl/fsl_sai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..d8b673f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -797,7 +797,8 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; - if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai") || + of_device_is_compatible(pdev->dev.of_node, "fsl,imx6ul-sai")) sai->sai_on_imx = true; sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); @@ -898,6 +899,7 @@ static int fsl_sai_probe(struct platform_device *pdev) static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, + { .compatible = "fsl,imx6ul-sai", }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC: fsl_sai: Allow the SAI driver to turn on SAI_MCLK
by Fabio Estevam 04 May '16

04 May '16

From: Fabio Estevam <fabio.estevam(a)nxp.com> On mx6ul the General Purpose Register 1 (GPR1) contains the following bits for enabling the output of the SAI MCLKs: SAI1_MCLK_DIR, SAI2_MCLK_DIR, SAI3_MCLK_DIR Introduce "gpr" and "fsl,sai-enable-mclk" optional properties to allow the enablement of the SAI_MCLK outputs. Tested on a imx6ul-evk board. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- .../devicetree/bindings/sound/fsl-sai.txt | 8 +++++ include/linux/mfd/syscon/imx6q-iomuxc-gpr.h | 3 ++ sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++++ 3 files changed, 48 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 044e5d7..86755bb 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -48,6 +48,14 @@ Required properties: receive data by following their own bit clocks and frame sync clocks separately. +Optional properties (for mx6ul): + + - gpr : The phandle to the General Purpose Register (GPR) + node. + + - fsl,sai-enable-mclk : This is a boolean property. If present, indicates + that SAI will output the SAI MCLK clock. + Note: - If both fsl,sai-asynchronous and fsl,sai-synchronous-rx are absent, the default synchronous mode (sync Rx with Tx) will be used, which means both diff --git a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h index 238c8db..401f97e 100644 --- a/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h +++ b/include/linux/mfd/syscon/imx6q-iomuxc-gpr.h @@ -447,5 +447,8 @@ #define IMX6UL_GPR1_ENET2_CLK_OUTPUT (0x1 << 18) #define IMX6UL_GPR1_ENET_CLK_DIR (0x3 << 17) #define IMX6UL_GPR1_ENET_CLK_OUTPUT (0x3 << 17) +#define IMX6UL_GPR1_SAI1_MCLK_DIR (0x1 << 19) +#define IMX6UL_GPR1_SAI2_MCLK_DIR (0x1 << 20) +#define IMX6UL_GPR1_SAI3_MCLK_DIR (0x1 << 21) #endif /* __LINUX_IMX6Q_IOMUXC_GPR_H */ diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 0754df7..311325e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -21,6 +21,8 @@ #include <sound/core.h> #include <sound/dmaengine_pcm.h> #include <sound/pcm_params.h> +#include <linux/mfd/syscon.h> +#include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> #include "fsl_sai.h" #include "imx-pcm.h" @@ -785,11 +787,14 @@ static const struct regmap_config fsl_sai_regmap_config = { static int fsl_sai_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; + struct device_node *gpr_np = of_parse_phandle(np, "gpr", 0); struct fsl_sai *sai; + struct regmap *gpr; struct resource *res; void __iomem *base; char tmp[8]; int irq, ret, i; + u32 index; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -877,6 +882,38 @@ static int fsl_sai_probe(struct platform_device *pdev) fsl_sai_dai.symmetric_samplebits = 0; } + if (of_find_property(np, "fsl,sai-enable-mclk", NULL)) { + ret = of_property_read_u32(np, "sai-index", &index); + if (ret) { + dev_err(&pdev->dev, "could not read sai-index\n"); + return ret; + } + + gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(gpr)) { + dev_err(&pdev->dev, "could not find gpr node\n"); + return PTR_ERR(gpr); + } + + switch (index) { + case 1: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI1_MCLK_DIR, + IMX6UL_GPR1_SAI1_MCLK_DIR); + break; + case 2: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI2_MCLK_DIR, + IMX6UL_GPR1_SAI2_MCLK_DIR); + break; + case 3: + regmap_update_bits(gpr, IOMUXC_GPR1, + IMX6UL_GPR1_SAI3_MCLK_DIR, + IMX6UL_GPR1_SAI3_MCLK_DIR); + break; + } + } + sai->dma_params_rx.addr = res->start + FSL_SAI_RDR; sai->dma_params_tx.addr = res->start + FSL_SAI_TDR; sai->dma_params_rx.maxburst = FSL_SAI_MAXBURST_RX; -- 1.9.1

2 8

[alsa-devel] [PATCH 0/2 v6] Add I2S audio support for ARC AXS10x boards
by Jose Abreu 04 May '16

04 May '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. 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) 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. 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 (2): ASoC: dwc: Add custom PCM driver ASoC: dwc: Update DOCUMENTATION for I2S Driver .../devicetree/bindings/sound/designware-i2s.txt | 9 +- sound/soc/dwc/Kconfig | 9 + sound/soc/dwc/Makefile | 1 + sound/soc/dwc/designware.h | 71 +++++++ sound/soc/dwc/designware_i2s.c | 94 ++++++--- sound/soc/dwc/designware_pcm.c | 228 +++++++++++++++++++++ 6 files changed, 383 insertions(+), 29 deletions(-) create mode 100644 sound/soc/dwc/designware.h create mode 100644 sound/soc/dwc/designware_pcm.c -- 1.9.1

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[alsa-devel] [PATCH v2 1/2] ASoC: rockchip: Revert "ASoC: rockchip: i2s: remove unused variables"
by John Keeping 04 May '16

04 May '16

This reverts commit 5938448b99275cba95167c3f9d39ca9225fdad38. It turns out that the commit that made these variables unused is wrong so we're about to revert it. Bring back the variables in prepration. Signed-off-by: John Keeping <john(a)metanate.com> --- v2: Add "ASoC: rockchip: " prefix to subject sound/soc/rockchip/rockchip_i2s.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 2f8e20416bd3..34743ec61c49 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -34,6 +34,13 @@ struct rk_i2s_dev { struct regmap *regmap; +/* + * Used to indicate the tx/rx status. + * I2S controller hopes to start the tx and rx together, + * also to stop them when they are both try to stop. +*/ + bool tx_start; + bool rx_start; bool is_master_mode; }; @@ -77,7 +84,11 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_TXS_START, I2S_XFER_TXS_START); + + i2s->tx_start = true; } else { + i2s->tx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); @@ -115,7 +126,11 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_XFER, I2S_XFER_RXS_START, I2S_XFER_RXS_START); + + i2s->rx_start = true; } else { + i2s->rx_start = false; + regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); -- 2.8.0.rc4.238.g874082a

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[alsa-devel] [PATCH v6 0/4] Propagate errors out from compressed streams
by Charles Keepax 04 May '16

04 May '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 v5: - Add a debug print to note which state we transition to in snd_compr_stop_error It might be easiest to put the ASoC changes through Mark's tree as there are some merge conflicts otherwise, as such I have based this series on Mark's tree. 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 | 5 +++ sound/core/compress_offload.c | 81 ++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 11 +++++- sound/soc/soc-compress.c | 5 ++- 4 files changed, 93 insertions(+), 9 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH v2 1/2] ASoC: dapm: support user-defined stop condition in dai_get_connected_widgets
by Piotr Stankiewicz 04 May '16

04 May '16

Certain situations may warrant examining DAPM paths only to a certain arbitrary point, as opposed to always following them to the end. For instance, when establishing a connection between a front-end DAI link and a back-end DAI link in a DPCM path, it does not make sense to walk the DAPM graph beyond the first widget associated with a back-end link. This patch introduces a mechanism which lets a user of dai_get_connected_widgets supply a function which will be called for every node during the graph walk. When invoked, this function can execute arbitrary logic to decide whether the walk, given a DAPM widget and walk direction, should be terminated at that point or continued as normal. Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 5 +++- sound/soc/soc-dapm.c | 68 +++++++++++++++++++++++++++++++++++++--------- sound/soc/soc-pcm.c | 3 +- 3 files changed, 61 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 9706946..fe5bc93 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -357,6 +357,7 @@ struct snd_soc_dapm_context; struct regulator; struct snd_soc_dapm_widget_list; struct snd_soc_dapm_update; +enum snd_soc_dapm_direction; int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -450,7 +451,9 @@ void dapm_mark_endpoints_dirty(struct snd_soc_card *card); /* dapm path query */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list); + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)); struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 801ae1a..075f0d7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1073,7 +1073,11 @@ static int dapm_widget_list_create(struct snd_soc_dapm_widget_list **list, */ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, struct list_head *list, enum snd_soc_dapm_direction dir, - int (*fn)(struct snd_soc_dapm_widget *, struct list_head *)) + int (*fn)(struct snd_soc_dapm_widget *, struct list_head *, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)), + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { enum snd_soc_dapm_direction rdir = SND_SOC_DAPM_DIR_REVERSE(dir); struct snd_soc_dapm_path *path; @@ -1088,6 +1092,13 @@ 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) { + con = custom_stop_condition(widget, dir); + if (con >= 0) + return con; + con = 0; + } + if ((widget->is_ep & SND_SOC_DAPM_DIR_TO_EP(dir)) && widget->connected) { widget->endpoints[dir] = snd_soc_dapm_suspend_check(widget); return widget->endpoints[dir]; @@ -1106,7 +1117,7 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, if (path->connect) { path->walking = 1; - con += fn(path->node[dir], list); + con += fn(path->node[dir], list, custom_stop_condition); path->walking = 0; } } @@ -1119,23 +1130,41 @@ static __always_inline int is_connected_ep(struct snd_soc_dapm_widget *widget, /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_OUT, - is_connected_output_ep); + is_connected_output_ep, custom_stop_condition); } /* * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. + * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. */ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, - struct list_head *list) + struct list_head *list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *i, + enum snd_soc_dapm_direction)) { return is_connected_ep(widget, list, SND_SOC_DAPM_DIR_IN, - is_connected_input_ep); + is_connected_input_ep, custom_stop_condition); } /** @@ -1143,15 +1172,26 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, * @dai: the soc DAI. * @stream: stream direction. * @list: list of active widgets for this stream. + * @custom_stop_condition: (optional) a function meant to stop the widget graph + * walk based on custom logic. * * Queries DAPM graph as to whether an valid audio stream path exists for * the initial stream specified by name. This takes into account * current mixer and mux kcontrol settings. Creates list of valid widgets. * + * Optionally, can be supplied with a function acting as a stopping condition. + * This function takes the dapm widget currently being examined and the walk + * direction as an arguments, and should return an integer, as follows: + * - >=0: if the walk is to be stopped (this should indicate the number of + * endpoints the widget has), + * - <0 : if the walk is to be continued as normal. + * * Returns the number of valid paths or negative error. */ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, - struct snd_soc_dapm_widget_list **list) + struct snd_soc_dapm_widget_list **list, + int (*custom_stop_condition)(struct snd_soc_dapm_widget *, + enum snd_soc_dapm_direction)) { struct snd_soc_card *card = dai->component->card; struct snd_soc_dapm_widget *w; @@ -1171,9 +1211,11 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, } if (stream == SNDRV_PCM_STREAM_PLAYBACK) - paths = is_connected_output_ep(dai->playback_widget, &widgets); + paths = is_connected_output_ep(dai->playback_widget, &widgets, + custom_stop_condition); else - paths = is_connected_input_ep(dai->capture_widget, &widgets); + paths = is_connected_input_ep(dai->capture_widget, &widgets, + custom_stop_condition); /* Drop starting point */ list_del(widgets.next); @@ -1268,8 +1310,8 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) DAPM_UPDATE_STAT(w, power_checks); - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); return out != 0 && in != 0; } @@ -1928,8 +1970,8 @@ static ssize_t dapm_widget_power_read_file(struct file *file, in = 0; out = 0; } else { - in = is_connected_input_ep(w, NULL); - out = is_connected_output_ep(w, NULL); + in = is_connected_input_ep(w, NULL, NULL); + out = is_connected_output_ep(w, NULL, NULL); } ret = snprintf(buf, PAGE_SIZE, "%s: %s%s in %d out %d", diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index aa99dac..c2b0aa8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1294,7 +1294,8 @@ int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int paths; /* get number of valid DAI paths and their widgets */ - paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list); + paths = snd_soc_dapm_dai_get_connected_widgets(cpu_dai, stream, list, + NULL); dev_dbg(fe->dev, "ASoC: found %d audio %s paths\n", paths, stream ? "capture" : "playback"); -- 1.7.10.4

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[alsa-devel] Get dB value using softvol plugin
by Jörg Krause 04 May '16

04 May '16

Hi, I configured my softvol plugin with a min_dB value of -6.0. However, if I store the state with alsactl store, the comment section shows a dbmin value of 0. I figured out that the softvol plugin adds the tlv info when it adds the user ctl and stores the db values in snd_pcm_softvol_t. Why differ the values of the tlv and the values of snd_pcm_softvol_t? My goal is to get information of the current dB value used by the softvol plugin, e.g. -6.0dB for a raw volume value of 255 and -51.0dB for 0. However, I only get values from +0 to +51dB using snd_ctl_convert_to_dB(). Do I miss something? Note, that I started from a clean asound.state. --- The state produced by alsactl: control.1 { iface MIXER name Master value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '00000001000000080000000000000014' dbmin 0 dbmax 5100 dbvalue.0 5100 dbvalue.1 5100 } } --- The output of aplay -D softvol -v: Playing WAVE '/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Soft volume PCM Control: Master min_dB: -57 max_dB: -6 resolution: 256 --- My asound.conf: pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "hw:0" } control { name "Master" card 0 } min_dB -57.0 max_dB -6.0 } ctl.!default { type hw card 0 } ctl.softvol { type hw card 0 } --- Best regards Jörg Krause

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Re: [alsa-devel] [Alsa-user] How do I run arecord on raw digital mic buffer?
by Clemens Ladisch 04 May '16

04 May '16

Rob Nertney wrote: >>> is the best method to build a sound card and a capture PCM? >> >> As compared to what? Any other method will not allow your hardware to >> be used for audio recording. > > As compared to my driver configuring the DMA to capture the data into > a known buffer, and then directly accessing the buffer to wrap it in > a WAV format. I should have rephrased my initial question as, "my goal > is to simply wrap multichannel digital mic input in WAV format; is > ALSA/arecord the most efficient way to do this?" If you bypass ALSA, you have to write all the software yourself, and you will not be able to use your hardware for anything else. > I should make my buffer at least 2x the size of a period. Yes. (You don't need periods if you don't use interrupts to wait for data being available.) > I have a buffer ready, under which function does it get passed back to > the ALSA framework? Normally, ALSA knows about the buffer, and the driver just reports the current position. > * dma completes/fires period-based IRQ > * <<question>> buffer is transferred in function X (?) to Y (?) When a period is finished, you must call snd_pcm_period_elapsed(). ALSA will then call your .pointer callback to know where in the buffer the hardware has finished writing. (Please not that .pointer can also be called at other times.) Regards, Clemens

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[alsa-devel] [patch] ASoC: hdac_hdmi: Potential NULL deref in hdac_hdmi_get_spk_alloc()
by Dan Carpenter 03 May '16

03 May '16

We intended || here instead of &&. The original code potentially leads to a NULL dereference. Fixes: 2889099eb8cd ('ASoC: hdac_hdmi: Register chmap controls and ops') Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 3043120..7e7ff1b 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1609,7 +1609,7 @@ static int hdac_hdmi_get_spk_alloc(struct hdac_device *hdac, int pcm_idx) struct hdac_hdmi_pcm *pcm = get_hdmi_pcm_from_id(hdmi, pcm_idx); struct hdac_hdmi_pin *pin = pcm->pin; - if (!pin && !pin->eld.eld_valid) + if (!pin || !pin->eld.eld_valid) return 0; return pin->eld.info.spk_alloc;

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Re: [alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by sugar 03 May '16

03 May '16

Hi John, On 4/30/2016 15:00, John Keeping Wrote: > Hi Enric, > > On Fri, Apr 29, 2016 at 04:59:27PM +0200, Enric Balletbo Serra wrote: >> 2015-12-09 11:32 GMT+01:00 John Keeping <john(a)metanate.com>: >>> If we only clear the tx/rx state when both are disabled it is not >>> possible to start/stop one multiple times while the other is running. >>> Since the two are independently controlled, treat them as such and >>> remove the false dependency between capture and playback. >>> >>> Signed-off-by: John Keeping <john(a)metanate.com> >>> --- >>> sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- >>> 1 file changed, 32 insertions(+), 40 deletions(-) >>> >>> diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c >>> index 83b1b9c..acc6225 100644 >>> --- a/sound/soc/rockchip/rockchip_i2s.c >>> +++ b/sound/soc/rockchip/rockchip_i2s.c >>> @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >>> I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); >>> >>> regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >>> + I2S_XFER_TXS_START, >>> + I2S_XFER_TXS_START); >>> >>> i2s->tx_start = true; >>> } else { >>> @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >>> regmap_update_bits(i2s->regmap, I2S_DMACR, >>> I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); >>> >>> - if (!i2s->rx_start) { >>> - regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | >>> - I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_STOP | >>> - I2S_XFER_RXS_STOP); >>> + regmap_update_bits(i2s->regmap, I2S_XFER, >>> + I2S_XFER_TXS_START, >>> + I2S_XFER_TXS_STOP); >>> >>> - regmap_update_bits(i2s->regmap, I2S_CLR, >>> - I2S_CLR_TXC | I2S_CLR_RXC, >>> - I2S_CLR_TXC | I2S_CLR_RXC); >>> + regmap_update_bits(i2s->regmap, I2S_CLR, >>> + I2S_CLR_TXC, >>> + I2S_CLR_TXC); >>> >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> >>> - /* Should wait for clear operation to finish */ >>> - while (val) { >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> - retry--; >>> - if (!retry) { >>> - dev_warn(i2s->dev, "fail to clear\n"); >>> - break; >>> - } >>> + /* Should wait for clear operation to finish */ >>> + while (val & I2S_CLR_TXC) { >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> + retry--; >>> + if (!retry) { >>> + dev_warn(i2s->dev, "fail to clear\n"); >>> + break; >>> } >>> } >>> } >>> @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >>> I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); >>> >>> regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >>> + I2S_XFER_RXS_START, >>> + I2S_XFER_RXS_START); >>> >>> i2s->rx_start = true; >>> } else { >>> @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >>> regmap_update_bits(i2s->regmap, I2S_DMACR, >>> I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); >>> >>> - if (!i2s->tx_start) { >>> - regmap_update_bits(i2s->regmap, I2S_XFER, >>> - I2S_XFER_TXS_START | >>> - I2S_XFER_RXS_START, >>> - I2S_XFER_TXS_STOP | >>> - I2S_XFER_RXS_STOP); >>> + regmap_update_bits(i2s->regmap, I2S_XFER, >>> + I2S_XFER_RXS_START, >>> + I2S_XFER_RXS_STOP); >>> >>> - regmap_update_bits(i2s->regmap, I2S_CLR, >>> - I2S_CLR_TXC | I2S_CLR_RXC, >>> - I2S_CLR_TXC | I2S_CLR_RXC); >>> + regmap_update_bits(i2s->regmap, I2S_CLR, >>> + I2S_CLR_RXC, >>> + I2S_CLR_RXC); >>> >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> >>> - /* Should wait for clear operation to finish */ >>> - while (val) { >>> - regmap_read(i2s->regmap, I2S_CLR, &val); >>> - retry--; >>> - if (!retry) { >>> - dev_warn(i2s->dev, "fail to clear\n"); >>> - break; >>> - } >>> + /* Should wait for clear operation to finish */ >>> + while (val & I2S_CLR_RXC) { >>> + regmap_read(i2s->regmap, I2S_CLR, &val); >>> + retry--; >>> + if (!retry) { >>> + dev_warn(i2s->dev, "fail to clear\n"); >>> + break; >>> } >>> } >>> } >>> -- >>> 2.6.3.462.gbe2c914 >> >> Using my Veyron Jerry Chromebook wiht latest kernel I found that the >> speakers doesn't work. Bisect points to this patch as the offending >> commit. However your changes looks reasonable the fact is that >> reverting the patch makes the audio work again on my device. I need to >> dig a bit more into the issue, but meanwhile, any idea on what is >> happening ? Can I ask which device did you test? > > I tested this on a Radxa Rock2 square. This change fixed an issue I > found when running separate processes for playback and recording (I was > testing with arecord and speaker-test IIRC) but I don't think that > should be relevant. I just read through the patch again and I can't see > anything obviously wrong so I'm afraid I don't have any idea what's > causing the problem you're seeing. > Do you have test playback/capture at the same time? I think you may reproduce this issue if you do as follows: step 1: tinyplay test.wav -D 0 -d 0 -p 1024 -n 4 & step 2: tinycap r.wav -D 0 -d 0 -p 1024 -n 4 & you will find that there is no music now. because of this problem, we enable tx/rx at the same time to WA this issue. May I know what's the issue that you met before? > > John > > _______________________________________________ > Linux-rockchip mailing list > Linux-rockchip(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-rockchip > > >

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[alsa-devel] [PATCH v3 0/6] ASoC: fsl_ssi: Fixing various channel slips and bad samples insertions
by Arnaud Mouiche 03 May '16

03 May '16

This series of patch is an attempt to fix the fsl_ssi driver to use it in TDM mode (DSP A or B) with a large number of channels/slots. Version 3: - rebase + fix + test on top of linux 4.4 + 'for-next' Version 2: - fixing a missing patch - checkpatch.pl - ... well, learning how to send a patchset. Sorry. Bugs are detected and fixed on a imx6sl platform with linux 4.4 + 'for-next' branch, where 2 SSI interfaces are used. SSI3 configured as a master of the bus, SSI2 as a slave. Similar tests were done on a imx6solo platform by Caleb Crome (He will propose a patch concerning fifo depth to fix DMA xrun at highest bitrates) Various loopback scenario are tested: * scenario 1: SSI3 (master) with TXD to RXD loopback | |---> TXD --\ SSI3 |<--- RXD --/ (master) |---> SYNC |---> BCLK | * scenario 2: SSI3 connected to SSI2 | |---> TXD --------\ SSI3 |<--- RXD -----\ | (master) |---> SYNC --\ | | |---> BCLK | | | | | | | | | | | | | |<--- BCLK | | | SSI2 |<--- SYNC --/ | | (slave) |---> TXD ------/ | |<--- RXD --------/ | A test software (called atest) was developed and available. [1] It basically generate/check specials frames with S16_LE sequence samples NNN0, NNN1, NNN2 ... NNNC with NNN = frame number, and C the number of channels-1 Patch 1: Limitation fix. Prerequisite to use the fsl_ssi driver with up to 32 channels / slots Patch 2: Bug fix. Prerequisite to setup a relative high bitclk for our tests in 8ch / 16bits / 48kHz Patch 3: Simply save a 'dev' reference inside ssi_private for dev_err() purpose. (ssi_private is only available in lot of places, and ssi_private->pdev is deprecated and NULL indeed) Patch 4: Fix playback samples being dropped because the TX fifo was not ready at the time the DMA starts filling (only in case where playback is started AFTER the capture) Detected in loopback scenario 1, with following script (> 80 % of reproducibility) $ atest -D SSI3 -c 8 -r 48000 capture play dbg: set period size: 960 dbg: SSI3: capture_start dbg: SSI3: playback_start err: invalid frame after 1 null frames err: 0000 0000 0013 0014 0015 0016 0017 0020 dbg: SIGINT total number of sequence errors: 21119 here we see that samples 0000 0001 ... 0012 are not present in the output. In addition, this the number of samples dropped is not a multiple of the number of channel, we have a a channel slip. Patch 5: This is the Caleb Crome's case [2], where the SSI starts to generate samples while the TX FIFO is not filled by the DMA yet. Void samples can be inserted before DMA streaming. Detected in loopback scenario 2, with following script (reproducibility < 1%) $ atest -D SSI3 -c 8 -r 48000 capture & $ sleep 0.2 $ atest -d 1 -D SSI2 -c 8 -r 48000 play dbg: SSI3: capture_start dbg: set period size: 960 dbg: SSI2: playback_start dbg: start a 1 seconds duration timer err: invalid frame after 11568 null frames err: 0000 0010 0011 0012 0013 0014 0015 0016 err: 0017 0020 0021 0022 0023 0024 0025 0026 Patch 6: Deals with Capture restart whereas Playback is still running (or the opposite, Playback restart whereas Capture is till running). Capture restart whereas Playback case is detected in loopback scenario 1 (reproducibility 100%). A playback session is running in background, and 2 consecutive Captures are performed. The first is fine, the second fails. We see at the 2nd capture startup that we receive 15 samples still pending in the RX FIFO from the previous capture session. => We are receiving invalid samples + slips the channels $ atest -D SSI3 -c 8 -r 48000 play & dbg: SSI3: playback_start $ atest -d 1 -D SSI3 -c 8 -r 48000 capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer warn: Valid frame after 1 null frames warn: 3a90 3a91 3a92 3a93 3a94 3a95 3a96 3a97 total number of sequence errors: 0 $ atest -I 5 -d 1 -D SSI3 -c 8 -r 48000 capture # restart the capture dbg: SSI3: capture_start dbg: start a 1 seconds duration timer err: invalid frame after 1 null frames err: 8dd6 8dd7 8de0 8de1 8de2 8de3 8de4 8de5 err: 8de6 8de7 8df0 8df1 8df2 8df3 8df4 c0c0 err: c0c1 c0c2 c0c3 c0c4 c0c5 c0c6 c0c7 c0d0 err: c0d1 c0d2 c0d3 c0d4 c0d5 c0d6 c0d7 c0e0 err: c0e1 c0e2 c0e3 c0e4 c0e5 c0e6 c0e7 c0f0 Playback restart whereas Capture case is also detected with loopback scenario 1 (reproducibility 100%), capturing continuously, and playing by periods of time. $ atest -a -D SSI3 -c 8 -r 48000 capture play -r 1000,200 dbg: SSI3: capture_start dbg: SSI3: playback_start dbg: SSI3: will stop every 1000 ms during 200 ms warn: Valid frame after 4 null frames warn: 0010 0011 0012 0013 0014 0015 0016 0017 warn: SSI3: PT_W4_STOP warn: SSI3: PT_W4_RESTART err: invalid frame after 1602 null frames err: eaa1 eaa2 eaa3 eaa4 eaa5 eaa6 eaa7 eab0 err: eab1 eab2 eab3 eab4 5810 5811 5812 5813 dbg: stop on first error total number of sequence errors: 143 Both cases are resolved by using the mostly undocumented SOR.RX_CLR and SOR TX_CLR (we can find the documentation in IMX51 reference manual at section 56.3.3.15). Arnaud [1] https://github.com/amouiche/atest [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-October/099221.ht… Arnaud Mouiche (6): ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: Fix samples being dropped as Playback startup ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. sound/soc/fsl/fsl_ssi.c | 81 ++++++++++++++++++++++++++++++++++++++++++------- 1 file changed, 70 insertions(+), 11 deletions(-) -- 1.9.1

6 20

[alsa-devel] iMX6UL EVK with kernel 4.6
by Petr Kulhavy 02 May '16

02 May '16

Hi, I'm trying to run the latest mainline kernel 4.6 on the iMX6UL EVK. The board mostly works and the only problem is audio. Even though the sound card is detected and the ALSA mixer can be controlled there is no MCLK coming out of the SAI2 interface. All the related peripherals in the DTS (pinmux, SAI2, sound card, codec) seem to be set-up correctly, I also tried out various things like disabling the ASRC, but without success and I'm slowly running out of ideas. Could somebody give me a hint please, what is different to the Freescale kernel branch where the MCLK works? By the way, why aren't the Freescale patches merged to the mainline kernel? Thanks Petr

2 14

Re: [alsa-devel] bytcr_rt5640.c:(.text+0x1593b4): undefined reference to `sst_acpi_find_name_from_hid'
by Randy Dunlap 02 May '16

02 May '16

On 03/29/16 07:21, Borislav Petkov wrote: > Hi guys, > > I'm seeing a bunch of those errors today on randconfig builds: > > sound/built-in.o: In function `snd_byt_rt5640_mc_probe': > bytcr_rt5640.c:(.text+0x1593b4): undefined reference to `sst_acpi_find_name_from_hid' > make: *** [vmlinux] Error 1 > > ERROR: "sst_acpi_find_name_from_hid" [sound/soc/intel/boards/snd-soc-sst-bytcr-rt5640.ko] undefined! > make[1]: *** [__modpost] Error 1 > make: *** [modules] Error 2 > make: *** Waiting for unfinished jobs.... > > Some of the failing configs attached. > I have 2 patches for this. I'll send them right away. -- ~Randy

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Add support for TLV based binary controls
by Charles Keepax 02 May '16

02 May '16

This patch adds support for the arbitrary length TLV based binary controls. This allows users to properly access controls that are more than 512 bytes in length. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 128 ++++++++++++++++++++++++++++++++++++++------- 1 file changed, 108 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 5699a81..630ebcd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -160,6 +160,8 @@ #define ADSP2_RAM_RDY_SHIFT 0 #define ADSP2_RAM_RDY_WIDTH 1 +#define ADSP_MAX_STD_CTRL_SIZE 512 + struct wm_adsp_buf { struct list_head list; void *buf; @@ -435,6 +437,7 @@ struct wm_coeff_ctl { size_t len; unsigned int set:1; struct snd_kcontrol *kcontrol; + struct soc_bytes_ext bytes_ext; unsigned int flags; }; @@ -711,10 +714,17 @@ static void wm_adsp2_show_fw_status(struct wm_adsp *dsp) be16_to_cpu(scratch[3])); } +static inline struct wm_coeff_ctl *bytes_ext_to_ctl(struct soc_bytes_ext *ext) +{ + return container_of(ext, struct wm_coeff_ctl, bytes_ext); +} + static int wm_coeff_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *uinfo) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; uinfo->count = ctl->len; @@ -763,7 +773,9 @@ static int wm_coeff_write_control(struct wm_coeff_ctl *ctl, static int wm_coeff_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -780,6 +792,29 @@ static int wm_coeff_put(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_put(struct snd_kcontrol *kctl, + const unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (copy_from_user(ctl->cache, bytes, size)) { + ret = -EFAULT; + } else { + ctl->set = 1; + if (ctl->enabled) + ret = wm_coeff_write_control(ctl, ctl->cache, size); + } + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, void *buf, size_t len) { @@ -822,7 +857,9 @@ static int wm_coeff_read_control(struct wm_coeff_ctl *ctl, static int wm_coeff_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *ucontrol) { - struct wm_coeff_ctl *ctl = (struct wm_coeff_ctl *)kctl->private_value; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); char *p = ucontrol->value.bytes.data; int ret = 0; @@ -845,12 +882,72 @@ static int wm_coeff_get(struct snd_kcontrol *kctl, return ret; } +static int wm_coeff_tlv_get(struct snd_kcontrol *kctl, + unsigned int __user *bytes, unsigned int size) +{ + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *)kctl->private_value; + struct wm_coeff_ctl *ctl = bytes_ext_to_ctl(bytes_ext); + int ret = 0; + + mutex_lock(&ctl->dsp->pwr_lock); + + if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) { + if (ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + else + ret = -EPERM; + } else { + if (!ctl->flags && ctl->enabled) + ret = wm_coeff_read_control(ctl, ctl->cache, size); + } + + if (!ret && copy_to_user(bytes, ctl->cache, size)) + ret = -EFAULT; + + mutex_unlock(&ctl->dsp->pwr_lock); + + return ret; +} + struct wmfw_ctl_work { struct wm_adsp *dsp; struct wm_coeff_ctl *ctl; struct work_struct work; }; +static unsigned int wmfw_convert_flags(unsigned int in, unsigned int len) +{ + unsigned int out, rd, wr, vol; + + if (len > ADSP_MAX_STD_CTRL_SIZE) { + rd = SNDRV_CTL_ELEM_ACCESS_TLV_READ; + wr = SNDRV_CTL_ELEM_ACCESS_TLV_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + } else { + rd = SNDRV_CTL_ELEM_ACCESS_READ; + wr = SNDRV_CTL_ELEM_ACCESS_WRITE; + vol = SNDRV_CTL_ELEM_ACCESS_VOLATILE; + + out = 0; + } + + if (in) { + if (in & WMFW_CTL_FLAG_READABLE) + out |= rd; + if (in & WMFW_CTL_FLAG_WRITEABLE) + out |= wr; + if (in & WMFW_CTL_FLAG_VOLATILE) + out |= vol; + } else { + out |= rd | wr | vol; + } + + return out; +} + static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) { struct snd_kcontrol_new *kcontrol; @@ -868,19 +965,15 @@ static int wmfw_add_ctl(struct wm_adsp *dsp, struct wm_coeff_ctl *ctl) kcontrol->info = wm_coeff_info; kcontrol->get = wm_coeff_get; kcontrol->put = wm_coeff_put; - kcontrol->private_value = (unsigned long)ctl; + kcontrol->iface = SNDRV_CTL_ELEM_IFACE_MIXER; + kcontrol->tlv.c = snd_soc_bytes_tlv_callback; + kcontrol->private_value = (unsigned long)&ctl->bytes_ext; - if (ctl->flags) { - if (ctl->flags & WMFW_CTL_FLAG_WRITEABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_WRITE; - if (ctl->flags & WMFW_CTL_FLAG_READABLE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_READ; - if (ctl->flags & WMFW_CTL_FLAG_VOLATILE) - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } else { - kcontrol->access = SNDRV_CTL_ELEM_ACCESS_READWRITE; - kcontrol->access |= SNDRV_CTL_ELEM_ACCESS_VOLATILE; - } + ctl->bytes_ext.max = ctl->len; + ctl->bytes_ext.get = wm_coeff_tlv_get; + ctl->bytes_ext.put = wm_coeff_tlv_put; + + kcontrol->access = wmfw_convert_flags(ctl->flags, ctl->len); ret = snd_soc_add_card_controls(dsp->card, kcontrol, 1); if (ret < 0) @@ -1039,11 +1132,6 @@ static int wm_adsp_create_control(struct wm_adsp *dsp, ctl->flags = flags; ctl->offset = offset; - if (len > 512) { - adsp_warn(dsp, "Truncating control %s from %d\n", - ctl->name, len); - len = 512; - } ctl->len = len; ctl->cache = kzalloc(ctl->len, GFP_KERNEL); if (!ctl->cache) { -- 2.1.4

2 1

[alsa-devel] Volume control with mute in the wrong end
by Peter Rosin 02 May '16

02 May '16

Hi! I'm writing a driver for the max9860 codec https://datasheets.maximintegrated.com/en/ds/MAX9860.pdf and it has a digital sidetone control where it can mix in the left channel ADC into the (mono) DAC output. The register that controls the sidetone level makes me wonder how I should wire up DAPM for it. You set the gain for the sidetone in the same register field as you enable it. I have a problem though, since the mute value is at the wrong and of the scale, i.e. you turn the volume up and up and up, and when you reach "max" it is off. I did try to declare a TLV range like so: /* The dvst field has its mute in the wrong end. Sigh. */ static const DECLARE_TLV_DB_RANGE(dvst_tlv, 0, 30, TLV_DB_SCALE_ITEM(-6000, 200, 0), 31, 31, TLV_DB_SCALE_ITEM(0, 0, 1)); and then hook it that up with a dapm mixer: static const struct snd_kcontrol_new max9860_mixer_controls[] = { SOC_DAPM_SINGLE_TLV("Sidetone Volume", MAX9860_DACGAIN, MAX9860_DVST_SHIFT, 31, 1, dvst_tlv), }; static const struct snd_soc_dapm_widget max9860_dapm_widgets[] = { SND_SOC_DAPM_INPUT("MICL"), SND_SOC_DAPM_INPUT("MICR"), SND_SOC_DAPM_ADC("ADCL", NULL, MAX9860_PWRMAN, MAX9860_ADCLEN_SHIFT, 0), SND_SOC_DAPM_ADC("ADCR", NULL, MAX9860_PWRMAN, MAX9860_ADCREN_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIFOUTL", "Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIFOUTR", "Capture", 1, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 0, max9860_mixer_controls, ARRAY_SIZE(max9860_mixer_controls)), SND_SOC_DAPM_AIF_IN("AIFINL", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIFINR", "Playback", 1, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC", NULL, MAX9860_PWRMAN, MAX9860_DACEN_SHIFT, 0), SND_SOC_DAPM_OUTPUT("OUT"), }; static const struct snd_soc_dapm_route max9860_dapm_routes[] = { { "ADCL", NULL, "MICL" }, { "ADCR", NULL, "MICR" }, { "AIFOUTL", NULL, "ADCL" }, { "AIFOUTR", NULL, "ADCR" }, { "OUT", NULL, "DAC" }, { "DAC", NULL, "Mixer" }, { "Mixer", "Sidetone Volume", "ADCL" }, { "Mixer", NULL, "AIFINL" }, { "Mixer", NULL, "AIFINR" }, }; But DAPM and some tools (e.g. alsamixer) do not properly recognize the mute value at the wrong end on the scale for the sidetone. So, the sidetone path is never off (in the dapm sense). I can use amixer to set the special mute value, and it states -99999.99dB, so I think the tlv declaration is correct, at least in theory. But as I said alsamixer is thoroughly confused by the control and behaves rather strange... If I change the tlv declaration so that DAPM thinks the mute is in the natural end of the scale, the paths are off in the dapm sense when the sidetone is muted (but the sidetone is not in fact muted, of course). Is there anything I can do to make DAPM handle this insane register field better? Cheers, Peter

1 0

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix overwriting of ahclkx
by J.D. Schroeder 02 May '16

02 May '16

From: Jim Lodes <jim.lodes(a)garmin.com> The mcasp davinci_mcasp_set_dai_fmt function was overriding ahclkx input/output status that had already been set by the davinci_mcasp_set_sysclk function. This commit removes clearing of the ahclkx input/output status from davinci_mcasp_set_dai_fmt. Signed-off-by: Jim Lodes <jim.lodes(a)garmin.com> Signed-off-by: J.D. Schroeder <jay.schroeder(a)garmin.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e132498..a1197ad 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -489,7 +489,7 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE); mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, - ACLKX | AHCLKX | AFSX | ACLKR | AHCLKR | AFSR); + ACLKX | AFSX | ACLKR | AHCLKR | AFSR); mcasp->bclk_master = 0; break; default: -- 1.9.1

3 2

[alsa-devel] [PATCH 0/5] ASoC: Intel: Skylake: Wov optimizations
by Vinod Koul 02 May '16

02 May '16

This series starts with two minor fixes on driver to fix a memory leak and one NULL pointer exception and then adds power optimization to driver for WOV case by suspending the stream DMA and keeping only DSP pipe and endpoint On during that case Dharageswari.R (2): ASoC: Intel: Skylake: Fix the NULL pointer exception in dsp_clean up ASoC: Intel: Skylake: Prevent sending Set DMA Control IPC if the widget is "On" Jeeja KP (2): ASoC: Intel: Skylake: Suspend PCMs when marked as active suspend ASoC: Intel: Boards: remove ignore_suspend for WoV streams Mousumi Jana (1): ASoC: Intel: Skylake: Fix memory leak during init instance sound/soc/intel/boards/skl_nau88l25_max98357a.c | 1 - sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 1 - sound/soc/intel/boards/skl_rt286.c | 1 - sound/soc/intel/skylake/skl-messages.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 32 ++++++++++++++++--------- sound/soc/intel/skylake/skl-sst-dsp.c | 2 -- sound/soc/intel/skylake/skl-sst.c | 4 ++++ 7 files changed, 26 insertions(+), 17 deletions(-) -- 1.9.1

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Re: [alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by Enric Balletbo Serra 02 May '16

02 May '16

Hi John, Thanks for answer. 2016-04-30 9:00 GMT+02:00 John Keeping <john(a)keeping.me.uk>: > Hi Enric, > > On Fri, Apr 29, 2016 at 04:59:27PM +0200, Enric Balletbo Serra wrote: >> 2015-12-09 11:32 GMT+01:00 John Keeping <john(a)metanate.com>: >> > If we only clear the tx/rx state when both are disabled it is not >> > possible to start/stop one multiple times while the other is running. >> > Since the two are independently controlled, treat them as such and >> > remove the false dependency between capture and playback. >> > >> > Signed-off-by: John Keeping <john(a)metanate.com> >> > --- >> > sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- >> > 1 file changed, 32 insertions(+), 40 deletions(-) >> > >> > diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c >> > index 83b1b9c..acc6225 100644 >> > --- a/sound/soc/rockchip/rockchip_i2s.c >> > +++ b/sound/soc/rockchip/rockchip_i2s.c >> > @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >> > I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); >> > >> > regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >> > + I2S_XFER_TXS_START, >> > + I2S_XFER_TXS_START); >> > >> > i2s->tx_start = true; >> > } else { >> > @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) >> > regmap_update_bits(i2s->regmap, I2S_DMACR, >> > I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); >> > >> > - if (!i2s->rx_start) { >> > - regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | >> > - I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_STOP | >> > - I2S_XFER_RXS_STOP); >> > + regmap_update_bits(i2s->regmap, I2S_XFER, >> > + I2S_XFER_TXS_START, >> > + I2S_XFER_TXS_STOP); >> > >> > - regmap_update_bits(i2s->regmap, I2S_CLR, >> > - I2S_CLR_TXC | I2S_CLR_RXC, >> > - I2S_CLR_TXC | I2S_CLR_RXC); >> > + regmap_update_bits(i2s->regmap, I2S_CLR, >> > + I2S_CLR_TXC, >> > + I2S_CLR_TXC); >> > >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > >> > - /* Should wait for clear operation to finish */ >> > - while (val) { >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > - retry--; >> > - if (!retry) { >> > - dev_warn(i2s->dev, "fail to clear\n"); >> > - break; >> > - } >> > + /* Should wait for clear operation to finish */ >> > + while (val & I2S_CLR_TXC) { >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > + retry--; >> > + if (!retry) { >> > + dev_warn(i2s->dev, "fail to clear\n"); >> > + break; >> > } >> > } >> > } >> > @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >> > I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); >> > >> > regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_START | I2S_XFER_RXS_START); >> > + I2S_XFER_RXS_START, >> > + I2S_XFER_RXS_START); >> > >> > i2s->rx_start = true; >> > } else { >> > @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) >> > regmap_update_bits(i2s->regmap, I2S_DMACR, >> > I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); >> > >> > - if (!i2s->tx_start) { >> > - regmap_update_bits(i2s->regmap, I2S_XFER, >> > - I2S_XFER_TXS_START | >> > - I2S_XFER_RXS_START, >> > - I2S_XFER_TXS_STOP | >> > - I2S_XFER_RXS_STOP); >> > + regmap_update_bits(i2s->regmap, I2S_XFER, >> > + I2S_XFER_RXS_START, >> > + I2S_XFER_RXS_STOP); >> > >> > - regmap_update_bits(i2s->regmap, I2S_CLR, >> > - I2S_CLR_TXC | I2S_CLR_RXC, >> > - I2S_CLR_TXC | I2S_CLR_RXC); >> > + regmap_update_bits(i2s->regmap, I2S_CLR, >> > + I2S_CLR_RXC, >> > + I2S_CLR_RXC); >> > >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > >> > - /* Should wait for clear operation to finish */ >> > - while (val) { >> > - regmap_read(i2s->regmap, I2S_CLR, &val); >> > - retry--; >> > - if (!retry) { >> > - dev_warn(i2s->dev, "fail to clear\n"); >> > - break; >> > - } >> > + /* Should wait for clear operation to finish */ >> > + while (val & I2S_CLR_RXC) { >> > + regmap_read(i2s->regmap, I2S_CLR, &val); >> > + retry--; >> > + if (!retry) { >> > + dev_warn(i2s->dev, "fail to clear\n"); >> > + break; >> > } >> > } >> > } >> > -- >> > 2.6.3.462.gbe2c914 >> >> Using my Veyron Jerry Chromebook wiht latest kernel I found that the >> speakers doesn't work. Bisect points to this patch as the offending >> commit. However your changes looks reasonable the fact is that >> reverting the patch makes the audio work again on my device. I need to >> dig a bit more into the issue, but meanwhile, any idea on what is >> happening ? Can I ask which device did you test? > > I tested this on a Radxa Rock2 square. This change fixed an issue I > found when running separate processes for playback and recording (I was > testing with arecord and speaker-test IIRC) but I don't think that > should be relevant. I just read through the patch again and I can't see > anything obviously wrong so I'm afraid I don't have any idea what's > causing the problem you're seeing. > By chance, do you have a git repo containing the changes required to test the sound on Radxa Rock2? Guess this is not in mainline. Thanks, Enric > > John

1 0

[alsa-devel] [PATCH 0/2] ASoC: Intel: Skylake: Add HDA link management
by Vinod Koul 02 May '16

02 May '16

HDA links are always kept On, this patch adds link management routines in hdac extended core and it's use in Skylake driver. One more patch to add support in codec will follow this The aim of serial is to shut down the links when not in use and keep them active when a user is active. The user can activate link by taking reference to the link when required. When all links are shutdown, the command DMAs are turned off as well. On first link power up, the command DMAs are turned on first. First patch touches HDA core, and second one is dependent on it so Takashi if you are okay, pls ACK it and we cna merge thru ASoC tree Vinod Koul (2): ALSA: hdac: add link pm and ref counting ASoC: Intel: Skylake: add link mangement include/sound/hdaudio_ext.h | 13 ++++++++ sound/hda/ext/hdac_ext_bus.c | 3 ++ sound/hda/ext/hdac_ext_controller.c | 62 +++++++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-pcm.c | 1 - sound/soc/intel/skylake/skl.c | 34 ++++++++++++++++++++ 5 files changed, 112 insertions(+), 1 deletion(-) -- 1.9.1

2 4

[alsa-devel] [PATCH v4 0/3]: conf: topology: Add the conf for SKL-rt286
by Vinod Koul 02 May '16

02 May '16

This series adds SKL-i2c conf for simple PCM topology graph Changes in v4: - add back the pre-compiled binary blobs - remove running the binary on intsall step Changes in v3: - remove bin blobs, now they are generated in install step - hook the tool compilation to alsa-lib and run it on install - add u8 in alsa type_compat.h Shreyas NC (3): conf: topology: Add Skylake i2s conf Add u8 in type_compat.h conf: topology: Generate Private data binary blobs configure.ac | 2 + include/sound/type_compat.h | 2 + src/conf/topology/Makefile.am | 2 +- src/conf/topology/sklrt286/Makefile.am | 5 + src/conf/topology/sklrt286/codec0_in-cpr-1.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_out-cpr-4.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec0_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec1_out-cpr-5.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/codec1_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/data/Makefile.am | 4 + src/conf/topology/sklrt286/data/pvt.c | 1815 ++++++++++++++++++++ src/conf/topology/sklrt286/data/pvt_data.c | 90 + src/conf/topology/sklrt286/data/pvt_local.h | 9 + .../topology/sklrt286/data/skl-tplg-interface.h | 232 +++ .../topology/sklrt286/dmic01_hifi_in-cpr-3.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_in-cpr-0.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_in-mi.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_out-cpr-6.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/media0_out-mo.bin | Bin 0 -> 4244 bytes src/conf/topology/sklrt286/skl_i2s.conf | 342 ++++ 28 files changed, 2502 insertions(+), 1 deletion(-) create mode 100644 src/conf/topology/sklrt286/Makefile.am create mode 100644 src/conf/topology/sklrt286/codec0_in-cpr-1.bin create mode 100644 src/conf/topology/sklrt286/codec0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-cpr-4.bin create mode 100644 src/conf/topology/sklrt286/codec0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-cpr-5.bin create mode 100644 src/conf/topology/sklrt286/codec1_out-mo.bin create mode 100644 src/conf/topology/sklrt286/data/Makefile.am create mode 100644 src/conf/topology/sklrt286/data/pvt.c create mode 100644 src/conf/topology/sklrt286/data/pvt_data.c create mode 100644 src/conf/topology/sklrt286/data/pvt_local.h create mode 100644 src/conf/topology/sklrt286/data/skl-tplg-interface.h create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-cpr-3.bin create mode 100644 src/conf/topology/sklrt286/dmic01_hifi_in-mi.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-7.bin create mode 100644 src/conf/topology/sklrt286/hdmi1_pt_out-cpr-8.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-10.bin create mode 100644 src/conf/topology/sklrt286/hdmi2_pt_out-cpr-9.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-11.bin create mode 100644 src/conf/topology/sklrt286/hdmi3_pt_out-cpr-12.bin create mode 100644 src/conf/topology/sklrt286/media0_in-cpr-0.bin create mode 100644 src/conf/topology/sklrt286/media0_in-mi.bin create mode 100644 src/conf/topology/sklrt286/media0_out-cpr-6.bin create mode 100644 src/conf/topology/sklrt286/media0_out-mo.bin create mode 100644 src/conf/topology/sklrt286/skl_i2s.conf -- 1.9.1

2 5

[alsa-devel] [PATCH 0/4] Decouple X86_32 dependency from the ISA Kconfig option
by William Breathitt Gray 02 May '16

02 May '16

This patchset is based on top of commit 3a3a5fece6f2 ("fs: kernfs: Replace CURRENT_TIME by current_fs_time()") of the driver-core-testing branch of the driver-core repository. The introduction of the ISA_BUS option in commit b3c1be1b789c ("base: isa: Remove X86_32 dependency") blocks the compilation of ISA drivers on non-x86 platforms. The ISA_BUS configuration option should not be necessary if the X86_32 dependency can be decoupled from the ISA configuration option. This patchset both removes the ISA_BUS configuration option entirely and decouples the X86_32 dependency from the ISA configuration option. The PNPBIOS driver requires preprocessor defines (located in include/asm/segment.h) only declared if the architecture is set to X86_32. If the architecture is set to X86_64, the PNPBIOS driver will not build properly. The X86 dependecy for the PNPBIOS configuration option is changed to an explicit X86_32 dependency in order to prevent an attempt to build for an unsupported architecture. Changes to the ISA SSCAPE and SCSI ULTRASTOR drivers are also included. The relevant patches simply fix format string identifier mismatches exposed during an attempted X86_64 compilation after the decoupling of the X86_32 dependency from the ISA configuration option. These patches fix compilation warnings rather than errors, but the solutions were so trivial that I decided to include them in this patchset. If it would be inappropriate to include them in this patchset, let me know and I will rebase to remove the relevant patches. William Breathitt Gray (4): pnp: pnpbios: Add explicit X86_32 dependency to PNPBIOS sound: isa: sscape: Use correct format identifier for size_t scsi: ultrastor: Use correct format identifier for kernel pointer isa: Remove the ISA_BUS Kconfig option arch/x86/Kconfig | 10 ++-------- drivers/base/Makefile | 2 +- drivers/pnp/pnpbios/Kconfig | 2 +- drivers/scsi/ultrastor.c | 8 ++++---- include/linux/isa.h | 2 +- sound/isa/sscape.c | 2 +- 6 files changed, 10 insertions(+), 16 deletions(-) -- 2.7.3

5 15

[alsa-devel] strange link on alsa-project.org, possibly a spam link
by humbert.olivier.1＠free.fr 30 Apr '16

30 Apr '16

Hi all just to let you know that there is a strange link in http://www.alsa-project.org/main/index.php/Main_Page See this screenshot : http://pix.toile-libre.org/upload/original/1461883624.png This "strange link" is under the white pointing-down arrow on the blue background circle It is linking to a "cheap holiday" booking webpage and someone on alsa@freenode suggested it looks to be one of those 'domain not used' redirects which I do agree. This link has been introduced back in August 2007 the 10th following this change : http://www.alsa-project.org/main/index.php?title=Main_Page&diff=698&oldid=6… I thought you might want to know. All the best to you all and thanks for the work done on ALSA. Cheers, Olivier

2 2

[alsa-devel] [PATCH 1/2] ASoC: rockchip: i2s: separate capture and playback
by John Keeping 29 Apr '16

29 Apr '16

If we only clear the tx/rx state when both are disabled it is not possible to start/stop one multiple times while the other is running. Since the two are independently controlled, treat them as such and remove the false dependency between capture and playback. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/rockchip/rockchip_i2s.c | 72 +++++++++++++++++---------------------- 1 file changed, 32 insertions(+), 40 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index 83b1b9c..acc6225 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -82,8 +82,8 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_TXS_START, + I2S_XFER_TXS_START); i2s->tx_start = true; } else { @@ -92,27 +92,23 @@ static void rockchip_snd_txctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_TDE_ENABLE, I2S_DMACR_TDE_DISABLE); - if (!i2s->rx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_TXS_START, + I2S_XFER_TXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_TXC, + I2S_CLR_TXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_TXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } @@ -128,8 +124,8 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_ENABLE); regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | I2S_XFER_RXS_START, - I2S_XFER_TXS_START | I2S_XFER_RXS_START); + I2S_XFER_RXS_START, + I2S_XFER_RXS_START); i2s->rx_start = true; } else { @@ -138,27 +134,23 @@ static void rockchip_snd_rxctrl(struct rk_i2s_dev *i2s, int on) regmap_update_bits(i2s->regmap, I2S_DMACR, I2S_DMACR_RDE_ENABLE, I2S_DMACR_RDE_DISABLE); - if (!i2s->tx_start) { - regmap_update_bits(i2s->regmap, I2S_XFER, - I2S_XFER_TXS_START | - I2S_XFER_RXS_START, - I2S_XFER_TXS_STOP | - I2S_XFER_RXS_STOP); + regmap_update_bits(i2s->regmap, I2S_XFER, + I2S_XFER_RXS_START, + I2S_XFER_RXS_STOP); - regmap_update_bits(i2s->regmap, I2S_CLR, - I2S_CLR_TXC | I2S_CLR_RXC, - I2S_CLR_TXC | I2S_CLR_RXC); + regmap_update_bits(i2s->regmap, I2S_CLR, + I2S_CLR_RXC, + I2S_CLR_RXC); - regmap_read(i2s->regmap, I2S_CLR, &val); + regmap_read(i2s->regmap, I2S_CLR, &val); - /* Should wait for clear operation to finish */ - while (val) { - regmap_read(i2s->regmap, I2S_CLR, &val); - retry--; - if (!retry) { - dev_warn(i2s->dev, "fail to clear\n"); - break; - } + /* Should wait for clear operation to finish */ + while (val & I2S_CLR_RXC) { + regmap_read(i2s->regmap, I2S_CLR, &val); + retry--; + if (!retry) { + dev_warn(i2s->dev, "fail to clear\n"); + break; } } } -- 2.6.3.462.gbe2c914

3 5

[alsa-devel] [PATCH] ALSA: usb-audio: Quirk for yet another Phoenix Audio devices (v2)
by Takashi Iwai 29 Apr '16

29 Apr '16

Phoenix Audio MT202pcs (1de7:0114) and MT202exe (1de7:0013) need the same workaround as TMX320 for avoiding the firmware bug. It fixes the frequent error about the sample rate inquiries and the slow device probe as consequence. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0adfd9537cf7..7b6dca2d7488 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1138,7 +1138,9 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x047F, 0xAA05): /* Plantronics DA45 */ case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ + case USB_ID(0x1de7, 0x0013): /* Phoenix Audio MT202exe */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ + case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- 2.8.1

1 0

[alsa-devel] [PATCH] ALSA: usb-audio: Quirk for yet another Phoenix Audio device
by Takashi Iwai 29 Apr '16

29 Apr '16

Phoenix Audio MT202pcs (1de7:0114) needs the same workaround as TMX320 for avoiding the firmware bug. It fixes the frequent error about the sample rate inquiries and the slow device probe as consequence. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=117321 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 0adfd9537cf7..c07facfe6699 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1139,6 +1139,7 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) case USB_ID(0x04D8, 0xFEEA): /* Benchmark DAC1 Pre */ case USB_ID(0x074D, 0x3553): /* Outlaw RR2150 (Micronas UAC3553B) */ case USB_ID(0x1de7, 0x0014): /* Phoenix Audio TMX320 */ + case USB_ID(0x1de7, 0x0114): /* Phoenix Audio MT202pcs */ case USB_ID(0x21B4, 0x0081): /* AudioQuest DragonFly */ return true; } -- 2.8.1

1 1

[alsa-devel] [PATCH 1/1] ASoC: fsl_ssi: add CCSR_SSI_SOR to volatile register list
by Caleb Crome 29 Apr '16

29 Apr '16

Due to caching, SOR wasn't written when it should have been. This patch simply adds SOR to the volatile list. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 216e3cb..2f3bf9c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -151,6 +151,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; -- 1.9.1

2 3

[alsa-devel] [PATCH] ALSA: usb-audio: Limit retrying sample rate reads
by Takashi Iwai 29 Apr '16

29 Apr '16

There are many USB audio devices with buggy firmware that don't react with the sample rate reading properly. This often results in the flood of error messages and slowing down the operation. The sample rate read back is basically only for confirming the sample rate setup, and it's not critically important. As a compromise, in this patch, we stop the sample rate read back once when the device gives errors more than tolerance (twice, as of now). This should improve most of error cases while we still can catch the firmware bugginess. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/clock.c | 4 ++++ sound/usb/usbaudio.h | 1 + 2 files changed, 5 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 7ccbcaf6a147..26dd5f20f149 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -309,6 +309,9 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, * support reading */ if (snd_usb_get_sample_rate_quirk(chip)) return 0; + /* the firmware is likely buggy, don't repeat to fail too many times */ + if (chip->sample_rate_read_error > 2) + return 0; if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN, @@ -316,6 +319,7 @@ static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface, data, sizeof(data))) < 0) { dev_err(&dev->dev, "%d:%d: cannot get freq at ep %#x\n", iface, fmt->altsetting, ep); + chip->sample_rate_read_error++; return 0; /* some devices don't support reading */ } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index a161c7c1b126..89b6853030de 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -50,6 +50,7 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + int sample_rate_read_error; struct list_head pcm_list; /* list of pcm streams */ struct list_head ep_list; /* list of audio-related endpoints */ -- 2.8.1

1 0

[alsa-devel] [PATCH v2 1/1] ASoC: fsl_ssi: add CCSR_SSI_SOR to volatile register list
by Caleb Crome 29 Apr '16

29 Apr '16

The CCSR_SSI_SOR is a register that clears the TX and/or the RX fifo on the i.MX SSI port. The fsl_ssi_trigger writes this register in order to clear the fifo at trigger time. However, since the CCSR_SSI_SOR register is not in the volatile list, the caching mechanism prevented the register write in the trigger function. This caused the fifo to not be cleared (because the value was unchanged from the last time the register was written), and thus causes the channels in both TDM or simple I2S mode to slip and be in the wrong time slots on SSI restart. This has gone unnoticed for so long because with simple stereo mode, the consequence is that left and right are swapped, which isn't that noticeable. However, it's catestrophic in some systems that require the channels to be in the right slots. Signed-off-by: Caleb Crome <caleb(a)crome.org> Suggested-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 216e3cb..2f3bf9c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -151,6 +151,7 @@ static bool fsl_ssi_volatile_reg(struct device *dev, unsigned int reg) case CCSR_SSI_SACDAT: case CCSR_SSI_SATAG: case CCSR_SSI_SACCST: + case CCSR_SSI_SOR: return true; default: return false; -- 1.9.1

3 2

[alsa-devel] sound on asus x205ta
by Michele Curti 29 Apr '16

29 Apr '16

On Thu, Mar 03, 2016 at 10:20:55AM -0600, Pierre-Louis Bossart wrote: > On 3/3/16 5:27 AM, Vinod Koul wrote: > >On Thu, Mar 03, 2016 at 12:11:10PM +0100, Michele Curti wrote: > >>Hi Vinod, > >> > >>>>[ 7.970937] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 > >>>>[ 7.970964] intel_sst_acpi 80860F28:00: fw returned err -16 > >>>Yes this is a problem, which FW are you using > > > >Have you tried the recent kernels, Pierre did a bunch of update and got many > >things working. > > > >As for timeout, that simply means we have send a message to DSP which failed > >to get a response. Perhpas the message had some bad element of FW had a > >bug/crashed... > > The firmware timeout is due to a change in the BIOS for BYT-CR devices that > impacts the IPC interrupt number, I have a set of patches that fix this on > Asus T100TAF but I still can't get any sound out. > Hi! So, it seems I got dsp interrupts after the timeout errors. I added dev_dbg(drv->dev, "Interrupt received\n"); to the intel_sst_interrupt_mrfld() function [zio@asus ~]$ dmesg | grep "Interrupt received" | wc -l 105 [zio@asus ~]$ cat /proc/interrupts | grep sst 1: 105 0 0 0 IO-APIC 29-fasteoi intel_sst_driver [zio@asus ~]$ dmesg | grep intel_sst [ 3.003961] intel_sst_acpi 80860F28:00: LPE base: 0x90a00000 size:0x200000 [ 3.003970] intel_sst_acpi 80860F28:00: IRAM base: 0x90ac0000 [ 3.004002] intel_sst_acpi 80860F28:00: DRAM base: 0x90b00000 [ 3.004013] intel_sst_acpi 80860F28:00: SHIM base: 0x90b40000 [ 3.004024] intel_sst_acpi 80860F28:00: Mailbox base: 0x90b44000 [ 3.004033] intel_sst_acpi 80860F28:00: DDR base: 0x20000000 [ 3.004260] intel_sst_acpi 80860F28:00: Got drv data max stream 25 [ 3.004388] intel_sst_acpi 80860F28:00: Registered IRQ 0x1 [ 3.004396] intel_sst_acpi 80860F28:00: Requesting FW intel/fw_sst_0f28_ssp0.bin now... [ 3.004792] intel_sst_acpi 80860F28:00: LPE is already in RESET state, No action [ 3.024222] intel_sst_acpi 80860F28:00: Enter [ 3.024232] intel_sst_acpi 80860F28:00: Request Fw completed [ 3.024507] intel_sst_acpi 80860F28:00: copied fw to ffff88007a900000 [ 3.024514] intel_sst_acpi 80860F28:00: phys: 7a900000 [ 3.024933] intel_sst_acpi 80860F28:00: Enter [ 3.024943] intel_sst_acpi 80860F28:00: header sign=$SST\xffffff96\xffffffb4 [ 3.024949] intel_sst_acpi 80860F28:00: module sign $SST\xffffff82\xffffffb4 [ 3.024955] intel_sst_acpi 80860F28:00: module entrypoint 0x0 [ 4.905820] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 4.905833] intel_sst_acpi 80860F28:00: fw returned err -16 [ 5.905864] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 5.905891] intel_sst_acpi 80860F28:00: fw returned err -16 [ 6.905818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 6.905838] intel_sst_acpi 80860F28:00: fw returned err -16 [ 7.905823] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 7.905849] intel_sst_acpi 80860F28:00: fw returned err -16 [ 8.905820] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 8.905847] intel_sst_acpi 80860F28:00: fw returned err -16 [ 9.905830] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 9.905857] intel_sst_acpi 80860F28:00: fw returned err -16 [ 10.905828] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 10.905855] intel_sst_acpi 80860F28:00: fw returned err -16 [ 11.905826] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 11.905853] intel_sst_acpi 80860F28:00: fw returned err -16 [ 12.905828] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 12.905854] intel_sst_acpi 80860F28:00: fw returned err -16 [ 13.905819] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 13.905843] intel_sst_acpi 80860F28:00: fw returned err -16 [ 14.925811] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 14.925836] intel_sst_acpi 80860F28:00: fw returned err -16 [ 15.925823] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 15.925850] intel_sst_acpi 80860F28:00: fw returned err -16 [ 16.925819] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 16.925844] intel_sst_acpi 80860F28:00: fw returned err -16 [ 17.925818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 17.925841] intel_sst_acpi 80860F28:00: fw returned err -16 [ 18.925821] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 18.925845] intel_sst_acpi 80860F28:00: fw returned err -16 [ 19.925810] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 19.925834] intel_sst_acpi 80860F28:00: fw returned err -16 [ 20.925788] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 20.925809] intel_sst_acpi 80860F28:00: fw returned err -16 [ 21.925821] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 21.925846] intel_sst_acpi 80860F28:00: fw returned err -16 [ 22.920830] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 22.920856] intel_sst_acpi 80860F28:00: fw returned err -16 [ 23.884836] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 23.884859] intel_sst_acpi 80860F28:00: fw returned err -16 [ 24.853161] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 24.853174] intel_sst_acpi 80860F28:00: fw returned err -16 [ 25.825400] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 25.825427] intel_sst_acpi 80860F28:00: fw returned err -16 [ 26.800986] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 26.801008] intel_sst_acpi 80860F28:00: fw returned err -16 [ 27.779566] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 27.779592] intel_sst_acpi 80860F28:00: fw returned err -16 [ 28.760784] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 28.760809] intel_sst_acpi 80860F28:00: fw returned err -16 [ 29.744302] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 29.744328] intel_sst_acpi 80860F28:00: fw returned err -16 [ 30.729854] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 30.729881] intel_sst_acpi 80860F28:00: fw returned err -16 [ 31.717175] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 31.717202] intel_sst_acpi 80860F28:00: fw returned err -16 [ 32.706075] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 32.706102] intel_sst_acpi 80860F28:00: fw returned err -16 [ 33.696343] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 33.696367] intel_sst_acpi 80860F28:00: fw returned err -16 [ 34.687818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 34.687844] intel_sst_acpi 80860F28:00: fw returned err -16 [ 35.680357] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 35.680383] intel_sst_acpi 80860F28:00: fw returned err -16 [ 36.673818] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 36.673844] intel_sst_acpi 80860F28:00: fw returned err -16 [ 37.668111] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 37.668135] intel_sst_acpi 80860F28:00: fw returned err -16 [ 38.663091] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 38.663118] intel_sst_acpi 80860F28:00: fw returned err -16 [ 39.658680] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 39.658691] intel_sst_acpi 80860F28:00: fw returned err -16 [ 40.654812] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 40.654824] intel_sst_acpi 80860F28:00: fw returned err -16 [ 41.651466] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 41.651479] intel_sst_acpi 80860F28:00: fw returned err -16 [ 42.648513] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 [ 42.648525] intel_sst_acpi 80860F28:00: fw returned err -16 [ 43.645942] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 43.645952] intel_sst_acpi 80860F28:00: fw returned err -16 [ 45.052845] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 45.052873] intel_sst_acpi 80860F28:00: fw returned err -16 [ 46.050948] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 46.050960] intel_sst_acpi 80860F28:00: fw returned err -16 [ 47.049320] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 47.049331] intel_sst_acpi 80860F28:00: fw returned err -16 [ 48.047885] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 48.047896] intel_sst_acpi 80860F28:00: fw returned err -16 [ 49.046627] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 49.046638] intel_sst_acpi 80860F28:00: fw returned err -16 [ 50.045542] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 50.045553] intel_sst_acpi 80860F28:00: fw returned err -16 [ 51.044603] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 51.044632] intel_sst_acpi 80860F28:00: fw returned err -16 [ 52.043787] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 52.043815] intel_sst_acpi 80860F28:00: fw returned err -16 [ 53.043030] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 53.043056] intel_sst_acpi 80860F28:00: fw returned err -16 [ 54.042669] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 54.042696] intel_sst_acpi 80860F28:00: fw returned err -16 [ 55.045310] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 55.045337] intel_sst_acpi 80860F28:00: fw returned err -16 [ 56.047789] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 56.047815] intel_sst_acpi 80860F28:00: fw returned err -16 [ 57.050159] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 57.050185] intel_sst_acpi 80860F28:00: fw returned err -16 [ 58.052350] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 58.052362] intel_sst_acpi 80860F28:00: fw returned err -16 [ 59.054494] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 59.054521] intel_sst_acpi 80860F28:00: fw returned err -16 [ 60.056485] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 60.056511] intel_sst_acpi 80860F28:00: fw returned err -16 [ 61.058360] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 61.058385] intel_sst_acpi 80860F28:00: fw returned err -16 [ 62.060160] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 62.060189] intel_sst_acpi 80860F28:00: fw returned err -16 [ 63.061829] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 63.061840] intel_sst_acpi 80860F28:00: fw returned err -16 [ 64.063443] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 64.063469] intel_sst_acpi 80860F28:00: fw returned err -16 [ 65.064970] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 65.064997] intel_sst_acpi 80860F28:00: fw returned err -16 [ 66.066390] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 66.066403] intel_sst_acpi 80860F28:00: fw returned err -16 [ 67.067803] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 67.067829] intel_sst_acpi 80860F28:00: fw returned err -16 [ 68.069108] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 68.069134] intel_sst_acpi 80860F28:00: fw returned err -16 [ 69.070353] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 69.070380] intel_sst_acpi 80860F28:00: fw returned err -16 [ 70.071530] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 70.071553] intel_sst_acpi 80860F28:00: fw returned err -16 [ 71.072665] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 71.072692] intel_sst_acpi 80860F28:00: fw returned err -16 [ 72.077040] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 72.077050] intel_sst_acpi 80860F28:00: fw returned err -16 [ 73.078102] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 73.078128] intel_sst_acpi 80860F28:00: fw returned err -16 [ 74.079073] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 74.079099] intel_sst_acpi 80860F28:00: fw returned err -16 [ 75.080034] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 75.080060] intel_sst_acpi 80860F28:00: fw returned err -16 [ 76.080922] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 76.080949] intel_sst_acpi 80860F28:00: fw returned err -16 [ 77.081768] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 77.081794] intel_sst_acpi 80860F28:00: fw returned err -16 [ 78.082592] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 78.082617] intel_sst_acpi 80860F28:00: fw returned err -16 [ 79.083395] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 79.083422] intel_sst_acpi 80860F28:00: fw returned err -16 [ 80.084155] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 80.084181] intel_sst_acpi 80860F28:00: fw returned err -16 [ 81.084883] intel_sst_acpi 80860F28:00: Wait timed-out condition:0x0, msg_id:0x2 fw_state 0x3 [ 81.084910] intel_sst_acpi 80860F28:00: fw returned err -16 [ 81.187135] intel_sst_acpi 80860F28:00: sst_load_fw [ 81.187162] intel_sst_acpi 80860F28:00: sst: Resetting the DSP in mrfld [ 81.187174] intel_sst_acpi 80860F28:00: value:0x1e40001 [ 81.187184] intel_sst_acpi 80860F28:00: value:0x1e40007 [ 81.187193] intel_sst_acpi 80860F28:00: value:0x1e40006 [ 81.229015] intel_sst_acpi 80860F28:00: config written to DCCM [ 81.229024] intel_sst_acpi 80860F28:00: sst: Starting the DSP in mrfld LALALALA [ 81.229030] intel_sst_acpi 80860F28:00: value:0x1e40006 [ 81.229035] intel_sst_acpi 80860F28:00: value:0x1e40007 [ 81.229041] intel_sst_acpi 80860F28:00: sst: Starting the DSP_merrifield:1e40802 [ 81.239680] intel_sst_acpi 80860F28:00: Interrupt received [ 81.239999] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240040] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240113] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240165] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240232] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240252] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240322] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240357] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240423] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240451] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240517] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240538] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240601] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240632] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240695] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240719] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240785] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240817] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240883] intel_sst_acpi 80860F28:00: Interrupt received [ 81.240956] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241029] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241099] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241176] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241218] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241298] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241340] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241413] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241451] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241475] intel_sst_acpi 80860F28:00: FW sent error response 0x5000e [ 81.241528] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241552] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241580] intel_sst_acpi 80860F28:00: FW sent error response 0x5000e [ 81.241638] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241661] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241692] intel_sst_acpi 80860F28:00: FW sent error response 0x5000e [ 81.241734] intel_sst_acpi 80860F28:00: Interrupt received [ 81.241999] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242069] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242096] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242145] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242298] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242340] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242400] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242431] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242550] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242570] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242689] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242807] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242828] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242888] intel_sst_acpi 80860F28:00: Interrupt received [ 81.242919] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243012] intel_sst_acpi 80860F28:00: FW sent async error msg: [ 81.243029] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243048] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243163] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243277] intel_sst_acpi 80860F28:00: fw load successful!!! [ 81.243279] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243307] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243384] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243460] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243503] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243541] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243614] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243653] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243684] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243715] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243757] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243789] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243820] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243860] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243911] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243953] intel_sst_acpi 80860F28:00: Interrupt received [ 81.243994] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244036] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244078] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244120] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244170] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244213] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244257] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244298] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244340] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244382] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244423] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244472] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244517] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244639] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244769] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244792] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244915] intel_sst_acpi 80860F28:00: Interrupt received [ 81.244956] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245058] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245187] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245316] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245424] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245479] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245532] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245579] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245623] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245672] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245706] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245748] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245786] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245816] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245880] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245931] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245967] intel_sst_acpi 80860F28:00: Interrupt received [ 81.245997] intel_sst_acpi 80860F28:00: Interrupt received [ 81.246050] intel_sst_acpi 80860F28:00: Interrupt received [ 84.180360] intel_sst_acpi 80860F28:00: Interrupt received [ 84.180479] intel_sst_acpi 80860F28:00: sst: Resetting the DSP in mrfld [ 84.180507] intel_sst_acpi 80860F28:00: value:0x1e4080a [ 84.180524] intel_sst_acpi 80860F28:00: value:0x1e40807 [ 84.180539] intel_sst_acpi 80860F28:00: value:0x1e40806 Am I doing something wrong? Any hints? Thanks, Michele My changes: Signed-off-by: Michele Curti <michele.curti(a)gmail.com> --- sound/soc/codecs/rt5645.c | 2 + sound/soc/intel/Kconfig | 13 ++ sound/soc/intel/atom/sst/sst.c | 3 + sound/soc/intel/atom/sst/sst_acpi.c | 4 +- sound/soc/intel/atom/sst/sst_loader.c | 1 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcr_rt5645.c | 332 ++++++++++++++++++++++++++++++++++ 7 files changed, 356 insertions(+), 1 deletion(-) create mode 100644 sound/soc/intel/boards/bytcr_rt5645.c diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 7af5e73..c48e5c8 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3524,6 +3524,7 @@ static const struct regmap_config temp_regmap = { static const struct i2c_device_id rt5645_i2c_id[] = { { "rt5645", 0 }, + { "rt5648", 0 }, { "rt5650", 0 }, { } }; @@ -3532,6 +3533,7 @@ MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); #ifdef CONFIG_ACPI static const struct acpi_device_id rt5645_acpi_match[] = { { "10EC5645", 0 }, + { "10EC5648", 0 }, { "10EC5650", 0 }, {}, }; diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index b3e6c23..46b7f8b 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -108,6 +108,19 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH Say Y if you have such a device If unsure select "N". +config SND_SOC_INTEL_BYTCR_RT5645_MACH + tristate "ASoC Audio driver for Intel Baytrail and Baytrail-CR with RT5645 codec" + depends on X86 && I2C + select SND_SOC_RT5645 + select SND_SST_MFLD_PLATFORM + select SND_SST_IPC_ACPI + select SND_SOC_INTEL_SST_MATCH if ACPI + help + This adds support for ASoC machine driver for Intel(R) Baytrail and Baytrail-CR + platforms with RT5645 audio codec. + 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 diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index a4b458e..3ffdb2e 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -19,6 +19,7 @@ * * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */ +#define DEBUG 1 #include <linux/module.h> #include <linux/fs.h> #include <linux/interrupt.h> @@ -58,6 +59,8 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) unsigned int size = 0; struct intel_sst_drv *drv = (struct intel_sst_drv *) context; irqreturn_t retval = IRQ_HANDLED; + + dev_dbg(drv->dev, "Interrupt received\n"); /* Interrupt arrived, check src */ isr.full = sst_shim_read64(drv->shim, SST_ISRX); diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 3bc4b63..38447d2 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -109,7 +109,7 @@ static const struct sst_res_info byt_rvp_res_info = { .mbox_size = 0x1000, .acpi_lpe_res_index = 0, .acpi_ddr_index = 2, - .acpi_ipc_irq_index = 5, + .acpi_ipc_irq_index = 0, }; static struct sst_platform_info byt_rvp_platform_data = { @@ -327,6 +327,8 @@ static struct sst_acpi_mach sst_acpi_bytcr[] = { &byt_rvp_platform_data }, {"INTCCFFD", "bytcr_rt5640", "intel/fw_sst_0f28.bin", "bytcr_rt5640", NULL, &byt_rvp_platform_data }, + {"10EC5648", "bytcr_rt5645", "intel/fw_sst_0f28_ssp0.bin", "bytcr_rt5645", NULL, + &byt_rvp_platform_data }, {"10EC5651", "bytcr_rt5651", "intel/fw_sst_0f28.bin", "bytcr_rt5651", NULL, &byt_rvp_platform_data }, {}, diff --git a/sound/soc/intel/atom/sst/sst_loader.c b/sound/soc/intel/atom/sst/sst_loader.c index 3391714..9b750ed 100644 --- a/sound/soc/intel/atom/sst/sst_loader.c +++ b/sound/soc/intel/atom/sst/sst_loader.c @@ -22,6 +22,7 @@ * This file contains all dsp controlling functions like firmware download, * setting/resetting dsp cores, etc */ +#define DEBUG 1 #include <linux/pci.h> #include <linux/delay.h> #include <linux/fs.h> diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 3310c0f..af24d39 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -3,6 +3,7 @@ snd-soc-sst-byt-rt5640-mach-objs := byt-rt5640.o snd-soc-sst-byt-max98090-mach-objs := byt-max98090.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o +snd-soc-sst-bytcr-rt5645-objs := bytcr_rt5645.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o +obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5645_MACH) += snd-soc-sst-bytcr-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5651_MACH) += snd-soc-sst-bytcr-rt5651.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o diff --git a/sound/soc/intel/boards/bytcr_rt5645.c b/sound/soc/intel/boards/bytcr_rt5645.c new file mode 100644 index 0000000..4974d6b --- /dev/null +++ b/sound/soc/intel/boards/bytcr_rt5645.c @@ -0,0 +1,332 @@ +/* + * bytcr_rt5645.c - ASoc Machine driver for Intel Byt CR platform + * (derived from bytcr_rt5640.c) + * + * Copyright (C) 2015 Intel Corp + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * 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; version 2 of the License. + * + * 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 <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/acpi.h> +#include <linux/device.h> +#include <linux/dmi.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include "../../codecs/rt5645.h" +#include "../atom/sst-atom-controls.h" + +static const struct snd_soc_dapm_widget byt_rt5645_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), +}; + +static const struct snd_soc_dapm_route byt_rt5645_audio_map[] = { + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, + + {"Headset Mic", NULL, "micbias1"}, /* lowercase for rt5645 */ + {"IN2P", NULL, "Headset Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Speaker", NULL, "LOUTL"}, + {"Speaker", NULL, "LOUTR"}, +}; + +static const struct snd_soc_dapm_route byt_rt5645_intmic_dmic1_map[] = { + {"DMIC1", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5645_intmic_dmic2_map[] = { + {"DMIC2", NULL, "Internal Mic"}, +}; + +static const struct snd_soc_dapm_route byt_rt5645_intmic_in1_map[] = { + {"Internal Mic", NULL, "micbias1"}, + {"IN1P", NULL, "Internal Mic"}, +}; + +enum { + BYT_RT5645_DMIC1_MAP, + BYT_RT5645_DMIC2_MAP, + BYT_RT5645_IN1_MAP, +}; + +#define BYT_RT5645_MAP(quirk) ((quirk) & 0xff) +#define BYT_RT5645_DMIC_EN BIT(16) + +static unsigned long byt_rt5645_quirk = BYT_RT5645_DMIC1_MAP | + BYT_RT5645_DMIC_EN; + +static const struct snd_kcontrol_new byt_rt5645_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Internal Mic"), + SOC_DAPM_PIN_SWITCH("Speaker"), +}; + +static int byt_rt5645_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + snd_soc_dai_set_bclk_ratio(codec_dai, 50); + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec clock %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_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; +} + +static const struct dmi_system_id byt_rt5645_quirk_table[] = { + {} +}; + +static int byt_rt5645_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + const struct snd_soc_dapm_route *custom_map; + int num_routes; + + card->dapm.idle_bias_off = true; + + dmi_check_system(byt_rt5645_quirk_table); + switch (BYT_RT5645_MAP(byt_rt5645_quirk)) { + case BYT_RT5645_IN1_MAP: + custom_map = byt_rt5645_intmic_in1_map; + num_routes = ARRAY_SIZE(byt_rt5645_intmic_in1_map); + break; + case BYT_RT5645_DMIC2_MAP: + custom_map = byt_rt5645_intmic_dmic2_map; + num_routes = ARRAY_SIZE(byt_rt5645_intmic_dmic2_map); + break; + default: + custom_map = byt_rt5645_intmic_dmic1_map; + num_routes = ARRAY_SIZE(byt_rt5645_intmic_dmic1_map); + } + + ret = snd_soc_add_card_controls(card, byt_rt5645_controls, + ARRAY_SIZE(byt_rt5645_controls)); + if (ret) { + dev_err(card->dev, "unable to add card controls\n"); + return ret; + } + snd_soc_dapm_ignore_suspend(&card->dapm, "Headphone"); + snd_soc_dapm_ignore_suspend(&card->dapm, "Speaker"); + + return ret; +} + +static const struct snd_soc_pcm_stream byt_rt5645_dai_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, +}; + +static int byt_rt5645_codec_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + int ret; + + /* The DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + + /* + * Default mode for SSP configuration is TDM 4 slot, override config + * with explicit setting to I2S 2ch 24-bit. The word length is set with + * dai_set_tdm_slot() since there is no other API exposed + */ + ret = snd_soc_dai_set_fmt(rtd->cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_IF | + SND_SOC_DAIFMT_CBS_CFS + ); + + if (ret < 0) { + dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set I2S config, err %d\n", ret); + return ret; + } + + return 0; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int byt_rt5645_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops byt_rt5645_aif1_ops = { + .startup = byt_rt5645_aif1_startup, +}; + +static struct snd_soc_ops byt_rt5645_be_ssp2_ops = { + .hw_params = byt_rt5645_aif1_hw_params, +}; + +static struct snd_soc_dai_link byt_rt5645_dais[] = { + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &byt_rt5645_aif1_ops, + }, + [MERR_DPCM_DEEP_BUFFER] = { + .name = "Deep-Buffer Audio Port", + .stream_name = "Deep-Buffer Audio", + .cpu_dai_name = "deepbuffer-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .ignore_suspend = 1, + .nonatomic = true, + .dynamic = 1, + .dpcm_playback = 1, + .ops = &byt_rt5645_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + /* CODEC<->CODEC link */ + /* back ends */ + { + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5648:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = byt_rt5645_codec_fixup, + .ignore_suspend = 1, + .nonatomic = true, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = byt_rt5645_init, + .ops = &byt_rt5645_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card byt_rt5645_card = { + .name = "bytcr-rt5645", + .owner = THIS_MODULE, + .dai_link = byt_rt5645_dais, + .num_links = ARRAY_SIZE(byt_rt5645_dais), + .dapm_widgets = byt_rt5645_widgets, + .num_dapm_widgets = ARRAY_SIZE(byt_rt5645_widgets), + .dapm_routes = byt_rt5645_audio_map, + .num_dapm_routes = ARRAY_SIZE(byt_rt5645_audio_map), + .fully_routed = true, +}; + +static int snd_byt_rt5645_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + byt_rt5645_card.dev = &pdev->dev; + + ret_val = devm_snd_soc_register_card(&pdev->dev, &byt_rt5645_card); + + if (ret_val) { + dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n", + ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &byt_rt5645_card); + return ret_val; +} + +static struct platform_driver snd_byt_rt5645_mc_driver = { + .driver = { + .name = "bytcr_rt5645", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_byt_rt5645_mc_probe, +}; + +module_platform_driver(snd_byt_rt5645_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver for RT5645"); +MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:bytcr_rt5645"); -- 2.8.0

1 0

[alsa-devel] [PATCH v2 0/9] ASoC: hdac_hdmi: Add multichannel support
by Subhransu S. Prusty 28 Apr '16

28 Apr '16

This series parses ELD to identify multichannel capability of the sync, and uses HDA framework to programs codec's channel map registers. Channel map controls are registered in patch 8, with which user can specify a specific channel order. chmap controls are registered inside jack_init callback. As PCMs need to be registered before registering chmap controls, jack_init call in the machine driver is moved to late_probe instead of init callback. Skylake driver is modified and buffer size calculation is fixed for multichannel support. Please note, patch 2 adds a small macro to read channel counts from cap bits in HDA headers. This needs Takashi's ack, so that it can be merged through ASoC. Patch 7 has dependency on commit#44fde3b ("ALSA: hda - Update chmap tlv to report sink's capability) which is merged in Takashi's tree. As Takashi suggested Mark may pull Takashi's for-next branch before applying this or may send a pull request, so that Takashi pull's Mark's tree before applying these with Mark's ack. Mark, please check how you want this to be merged. Changes in v2: * Dropped drm change and accommodate the change in driver * Fix one auto build test error Subhransu S. Prusty (9): ASoC: hdac_hdmi: parse eld for channel map capability ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params sound/hda/local.h | 10 ++ sound/soc/codecs/hdac_hdmi.c | 165 ++++++++++++++++++++++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 74 ++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++- sound/soc/intel/boards/skl_rt286.c | 48 ++++++- sound/soc/intel/skylake/skl-pcm.c | 14 +- sound/soc/intel/skylake/skl-topology.c | 21 ++- 7 files changed, 381 insertions(+), 24 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH v3 0/9] ASoC: hdac_hdmi: Add multichannel support
by Subhransu S. Prusty 28 Apr '16

28 Apr '16

This series parses ELD to identify multichannel capability of the sync, and uses HDA framework to programs codec's channel map registers. Channel map controls are registered in patch 8, with which user can specify a specific channel order. chmap controls are registered inside jack_init callback. As PCMs need to be registered before registering chmap controls, jack_init call in the machine driver is moved to late_probe instead of init callback. Skylake driver is modified and buffer size calculation is fixed for multichannel support. Please note, patch 2 adds a small macro to read channel counts from cap bits in HDA headers. This needs Takashi's ack, so that it can be merged through ASoC. Patch 7 has dependency on commit#44fde3b ("ALSA: hda - Update chmap tlv to report sink's capability) which is merged in Takashi's tree. As Takashi suggested Mark may pull Takashi's for-next branch before applying this or may send a pull request, so that Takashi pull's Mark's tree before applying these with Mark's ack. Mark, please check how you want this to be merged. Changes in v3: * As suggested, simplified channel calculation expression in get channel helper * Removed condition in max channel assignment expression Subhransu S. Prusty (9): ASoC: hdac_hdmi: parse eld for channel map capability ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params sound/hda/local.h | 10 ++ sound/soc/codecs/hdac_hdmi.c | 163 ++++++++++++++++++++++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 74 ++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++- sound/soc/intel/boards/skl_rt286.c | 48 ++++++- sound/soc/intel/skylake/skl-pcm.c | 14 +- sound/soc/intel/skylake/skl-topology.c | 21 ++- 7 files changed, 379 insertions(+), 24 deletions(-) -- 1.9.1

3 15

[alsa-devel] [PATCH 00/11] ASoC: hdac_hdmi: Add multichannel support
by Subhransu S. Prusty 28 Apr '16

28 Apr '16

This series parses ELD to identify multichannel capability of the sync, and uses HDA framework to programs codec's channel map registers. Channel map controls are registered in patch 8, with which user can specify a specific channel order. chmap controls are registered inside jack_init callback. As PCMs need to be registered before registering chmap controls, jack_init call in the machine driver is moved to late_probe instead of init callback. Skylake driver is modified and buffer size calculation is fixed for multichannel support. Please note, patch 2 and 4 adds small macros to read speaker allocation from ELD and channel counts from cap bits respectively in DRM and HDA headers. We have CCed DRM folks here. Please ack patch 1, so that we can have this series merged through sound trees. Patch 3 needs Takashi's ack, so that it can be merged through ASoC. Patch 9 has dependency on commit#44fde3b ("ALSA: hda - Update chmap tlv to report sink's capability) which is merged in Takashi's tree. So this may be merged in ASoC with Takashi's Ack and pulling in changes from Takashi's tree or alternatively it can be directly merged in Takashi's with Mark's Ack Takashi/Mark, please decide how you guys want this to be merged. Cc: Jani Nikula <jani.nikula(a)intel.com> Cc: David Airlie <airlied(a)linux.ie> Cc: dri-devel(a)lists.freedesktop.org Cc: Daniel Vetter <daniel.vetter(a)intel.com> Subhransu S. Prusty (11): ASoC: Intel: Skylake: Fix ibs/obs calc for non-integral sampling rates drm/edid: Add API to get speaker allocation ASoC: hdac_hdmi: parse eld for channel map capability ALSA: hda - add helper to get channels from cap bits ASoC: hdac_hdmi: Add multichannel support ASoC: skl_rt286: Fix to support hdmi channel map support ASoC: Intel: boards: Update skl_nau88l25_max98357a driver to support chmap ASoC: Intel: boards: Update skl_nau88l25_ssm4567 driver to support chmap ASoC: hdac_hdmi: Register chmap controls and ops ASoC: Intel: Skylake: Add multichannel support for HDMI ASoC: Intel: Skylake: Update channel map based on runtime params include/drm/drm_edid.h | 9 ++ sound/hda/local.h | 10 ++ sound/soc/codecs/hdac_hdmi.c | 165 ++++++++++++++++++++++-- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 74 ++++++++++- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 73 ++++++++++- sound/soc/intel/boards/skl_rt286.c | 48 ++++++- sound/soc/intel/skylake/skl-pcm.c | 14 +- sound/soc/intel/skylake/skl-topology.c | 49 +++++-- 8 files changed, 409 insertions(+), 33 deletions(-) -- 1.9.1

4 24

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 28 Apr '16

28 Apr '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- Changed some controls to TLV as Mark suggested. .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 442 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 67 ++++ 5 files changed, 531 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 913c83c..ba6c585 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -491,6 +492,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f39dee7..9f3587b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..21fae09 --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,442 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * 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/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_min_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 101, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_max_gain, + 0, 1, TLV_DB_SCALE_ITEM(537, 66, 0), + 2, 3, TLV_DB_SCALE_ITEM(677, 82, 0), + 4, 5, TLV_DB_SCALE_ITEM(852, 104, 0), + 6, 7, TLV_DB_SCALE_ITEM(1072, 131, 0), + 8, 9, TLV_DB_SCALE_ITEM(1350, 165, 0), + 10, 11, TLV_DB_SCALE_ITEM(1699, 208, 0), +); + +static const DECLARE_TLV_DB_RANGE(max98371_dht_rot_gain, + 0, 1, TLV_DB_SCALE_ITEM(-50, -50, 0), + 2, 6, TLV_DB_SCALE_ITEM(-100, -100, 0), + 7, 8, TLV_DB_SCALE_ITEM(-800, -200, 0), + 9, 11, TLV_DB_SCALE_ITEM(-1200, -300, 0), + 12, 13, TLV_DB_SCALE_ITEM(-2000, -200, 0), + 14, 15, TLV_DB_SCALE_ITEM(-2500, -500, 0), +); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker DHT Max Volume", MAX98371_GAIN, + 0, (1<<MAX98371_DHT_MAX_WIDTH)-1, 0, + max98371_dht_max_gain), + SOC_SINGLE_TLV("Speaker DHT Min Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_GAIN_WIDTH)-1, 0, + max98371_dht_min_gain), + SOC_SINGLE_TLV("Speaker DHT Rotation Volume", MAX98371_DHT_GAIN, + 0, (1<<MAX98371_DHT_ROT_WIDTH)-1, 0, + max98371_dht_rot_gain), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_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 max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..9f63309 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,67 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_DHT_MAX_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_DHT_GAIN 0x31 +#define MAX98371_DHT_GAIN_WIDTH 0x4 +#define MAX98371_DHT_ROT_WIDTH 0x4 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda: fix the missing ptr initialization
by Vinod Koul 28 Apr '16

28 Apr '16

ebus is a member of extended device and was never initialized, so do this at device creation. Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/hda/ext/hdac_ext_bus.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/hda/ext/hdac_ext_bus.c b/sound/hda/ext/hdac_ext_bus.c index 2433f7c81472..64de0a3d6d93 100644 --- a/sound/hda/ext/hdac_ext_bus.c +++ b/sound/hda/ext/hdac_ext_bus.c @@ -144,6 +144,7 @@ int snd_hdac_ext_bus_device_init(struct hdac_ext_bus *ebus, int addr) if (!edev) return -ENOMEM; hdev = &edev->hdac; + edev->ebus = ebus; snprintf(name, sizeof(name), "ehdaudio%dD%d", ebus->idx, addr); -- 1.9.1

2 5

[alsa-devel] [PATCH v2] ASoC: rt5651: remove unexisting widget
by Bard Liao 28 Apr '16

28 Apr '16

IN Muxes are not existing on rt5651. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 43 ------------------------------------------- 1 file changed, 43 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 7a61970..979e3e9 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -586,44 +586,6 @@ static const struct snd_kcontrol_new hpo_r_mute_control = SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, RT5651_R_MUTE_SFT, 1, 1); -/* INL/R source */ -static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl_enum, RT5651_INL1_INR1_VOL, - RT5651_INL_SEL_SFT, rt5651_inl_src); - -static const struct snd_kcontrol_new rt5651_inl1_mux = - SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); - -static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr1_enum, RT5651_INL1_INR1_VOL, - RT5651_INR_SEL_SFT, rt5651_inr1_src); - -static const struct snd_kcontrol_new rt5651_inr1_mux = - SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); - -static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inl2_enum, RT5651_INL2_INR2_VOL, - RT5651_INL_SEL_SFT, rt5651_inl2_src); - -static const struct snd_kcontrol_new rt5651_inl2_mux = - SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); - -static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; - -static SOC_ENUM_SINGLE_DECL( - rt5651_inr2_enum, RT5651_INL2_INR2_VOL, - RT5651_INR_SEL_SFT, rt5651_inr2_src); - -static const struct snd_kcontrol_new rt5651_inr2_mux = - SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); - - /* Stereo ADC source */ static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; @@ -955,11 +917,6 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- 1.8.1.1.439.g50a6b54

1 0

[alsa-devel] [PATCH] ASoC: rt5651: remove unexisting widget
by Bard Liao 28 Apr '16

28 Apr '16

IN Muxes are not existing on rt5651. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5651.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index 7a61970..b812c06 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -955,11 +955,6 @@ static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { RT5651_PWR_IN2_L_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, RT5651_PWR_IN2_R_BIT, 0, NULL, 0), - /* IN Mux */ - SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), - SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), - SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), - SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), /* REC Mixer */ SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: nau8825: apply BIQ to reduce recording signal DC offset
by John Hsu 28 Apr '16

28 Apr '16

Enable biquad filter as HPF in the ADC path when headset detected. It helps to reduce recording signal DC offset. The biquad filter parameters are applied for the ADC path when jack is inserted. The biquad filter is cancelled when jack is ejected. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8825.c | 59 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/nau8825.h | 7 ++++++ 2 files changed, 66 insertions(+) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index f35019a..962d2f0 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -156,6 +156,20 @@ static const struct reg_default nau8825_reg_defaults[] = { { NAU8825_REG_CHARGE_PUMP, 0x0 }, }; +/* Biquad filter coefficients for A1, A2, B0, B1, and B3 */ +static const struct reg_default nau8825_adc_biq_reg[] = { + { NAU8825_REG_BIQ_COF1, 0x009B }, + { NAU8825_REG_BIQ_COF2, 0x0006 }, + { NAU8825_REG_BIQ_COF3, 0xff66 }, + { NAU8825_REG_BIQ_COF4, 0x0000 }, + { NAU8825_REG_BIQ_COF5, 0xffb3 }, + { NAU8825_REG_BIQ_COF6, 0x0000 }, + { NAU8825_REG_BIQ_COF7, 0x009A }, + { NAU8825_REG_BIQ_COF8, 0x0006 }, + { NAU8825_REG_BIQ_COF9, 0xffb3 }, + { NAU8825_REG_BIQ_COF10, 0x8000 }, +}; + static bool nau8825_readable_reg(struct device *dev, unsigned int reg) { switch (reg) { @@ -217,12 +231,46 @@ 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; } } +static void nau8825_adc_biq_apply(struct nau8825 *nau8825) +{ + int i; + + /* Enable the BIQ parameter write/updated. The control is valid only + * when change from 0 to 1. The driver applies the BIQ function into + * the ADC path only. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN | NAU8825_BIQ_PATH_MASK, + NAU8825_BIQ_PATH_ADC); + for (i = 0; i < ARRAY_SIZE(nau8825_adc_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_adc_biq_reg[i].reg, + nau8825_adc_biq_reg[i].def); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, NAU8825_BIQ_WRT_EN); +} + +static void nau8825_adc_biq_cancel(struct nau8825 *nau8825) +{ + int i; + + /* Disable the BIQ parameter write/updated. The control is valid only + * when change from 0 to 1. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, 0); + for (i = 0; i < ARRAY_SIZE(nau8825_adc_biq_reg); i++) + regmap_write(nau8825->regmap, nau8825_adc_biq_reg[i].reg, 0); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIQ_CTRL, + NAU8825_BIQ_WRT_EN, NAU8825_BIQ_WRT_EN); +} + static int nau8825_adc_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { @@ -647,6 +695,9 @@ static void nau8825_eject_jack(struct nau8825 *nau8825) struct snd_soc_dapm_context *dapm = nau8825->dapm; struct regmap *regmap = nau8825->regmap; + /* Cancel biquad filter in ADC path */ + nau8825_adc_biq_cancel(nau8825); + snd_soc_dapm_disable_pin(dapm, "SAR"); snd_soc_dapm_disable_pin(dapm, "MICBIAS"); /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ @@ -737,6 +788,14 @@ static int nau8825_jack_insert(struct nau8825 *nau8825) break; } + if (type & SND_JACK_MICROPHONE) { + /* Apply biquad filter into ADC path. It helps to avoid the DC + * offset in recoding signal. The function will keep asleep + * without clock and work up automatically when clock provided. + */ + nau8825_adc_biq_apply(nau8825); + } + /* Leaving HPOL/R grounded after jack insert by default. They will be * ungrounded as part of the widget power up sequence at the beginning * of playback to reduce pop. diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h index 9e6cb62..f3b4acb 100644 --- a/sound/soc/codecs/nau8825.h +++ b/sound/soc/codecs/nau8825.h @@ -231,6 +231,13 @@ #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_EN (1 << 4) +#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

1 0

[alsa-devel] [PATCH 1/2] ASoC: dmaengine_pcm: Add support for packed transfers
by Matthias Reichl 27 Apr '16

27 Apr '16

dmaengine_pcm currently only supports setups where FIFO reads/writes correspond to exactly one sample, eg 16-bit sample data is transferred via 16-bit FIFO accesses, 32-bit data via 32-bit accesses. This patch adds support for setups with fixed width FIFOs where multiple samples are packed into a larger word. For example setups with a 32-bit wide FIFO register that expect 16-bit sample transfers to be done with the left+right sample data packed into a 32-bit word. Support for packed transfers is controlled via the SND_DMAENGINE_PCM_DAI_FLAG_PACK flag in snd_dmaengine_dai_dma_data.flags If this flag is set dmaengine_pcm doesn't put any restriction on the supported formats and sets the DMA transfer width to undefined. This means control over the constraints is now transferred to the DAI driver and it's responsible to provide proper configuration and check for possible corner cases that aren't handled by the ALSA core. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- include/sound/dmaengine_pcm.h | 12 ++++++++ sound/core/pcm_dmaengine.c | 11 +++++-- sound/soc/soc-generic-dmaengine-pcm.c | 57 +++++++++++++++++++++-------------- 3 files changed, 55 insertions(+), 25 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index f86ef5e..67be244 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -51,6 +51,16 @@ struct dma_chan *snd_dmaengine_pcm_request_channel(dma_filter_fn filter_fn, void *filter_data); struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream); +/* + * The DAI supports packed transfers, eg 2 16-bit samples in a 32-bit word. + * If this flag is set the dmaengine driver won't put any restriction on + * the supported sample formats and set the DMA transfer size to undefined. + * The DAI driver is responsible to disable any unsupported formats in it's + * configuration and catch corner cases that are not already handled in + * the ALSA core. + */ +#define SND_DMAENGINE_PCM_DAI_FLAG_PACK BIT(0) + /** * struct snd_dmaengine_dai_dma_data - DAI DMA configuration data * @addr: Address of the DAI data source or destination register. @@ -63,6 +73,7 @@ struct dma_chan *snd_dmaengine_pcm_get_chan(struct snd_pcm_substream *substream) * requesting the DMA channel. * @chan_name: Custom channel name to use when requesting DMA channel. * @fifo_size: FIFO size of the DAI controller in bytes + * @flags: PCM_DAI flags, only SND_DMAENGINE_PCM_DAI_FLAG_PACK for now */ struct snd_dmaengine_dai_dma_data { dma_addr_t addr; @@ -72,6 +83,7 @@ struct snd_dmaengine_dai_dma_data { void *filter_data; const char *chan_name; unsigned int fifo_size; + unsigned int flags; }; void snd_dmaengine_pcm_set_config_from_dai_data( diff --git a/sound/core/pcm_dmaengine.c b/sound/core/pcm_dmaengine.c index 697c166..8eb58c7 100644 --- a/sound/core/pcm_dmaengine.c +++ b/sound/core/pcm_dmaengine.c @@ -106,8 +106,9 @@ EXPORT_SYMBOL_GPL(snd_hwparams_to_dma_slave_config); * direction of the substream. If the substream is a playback stream the dst * fields will be initialized, if it is a capture stream the src fields will be * initialized. The {dst,src}_addr_width field will only be initialized if the - * addr_width field of the DAI DMA data struct is not equal to - * DMA_SLAVE_BUSWIDTH_UNDEFINED. + * SND_DMAENGINE_PCM_DAI_FLAG_PACK flag is set or if the addr_width field of + * the DAI DMA data struct is not equal to DMA_SLAVE_BUSWIDTH_UNDEFINED. If + * both conditions are met the latter takes priority. */ void snd_dmaengine_pcm_set_config_from_dai_data( const struct snd_pcm_substream *substream, @@ -117,11 +118,17 @@ void snd_dmaengine_pcm_set_config_from_dai_data( if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { slave_config->dst_addr = dma_data->addr; slave_config->dst_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->dst_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->dst_addr_width = dma_data->addr_width; } else { slave_config->src_addr = dma_data->addr; slave_config->src_maxburst = dma_data->maxburst; + if (dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK) + slave_config->src_addr_width = + DMA_SLAVE_BUSWIDTH_UNDEFINED; if (dma_data->addr_width != DMA_SLAVE_BUSWIDTH_UNDEFINED) slave_config->src_addr_width = dma_data->addr_width; } diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906..6cef397 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } -- 2.1.4

5 7

[alsa-devel] [PATCH] ASoC: Add max98371 codec driver
by anish kumar 27 Apr '16

27 Apr '16

Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- .../devicetree/bindings/sound/max98371.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/max98371.c | 419 +++++++++++++++++++++ sound/soc/codecs/max98371.h | 64 ++++ 5 files changed, 505 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98371.txt create mode 100644 sound/soc/codecs/max98371.c create mode 100644 sound/soc/codecs/max98371.h diff --git a/Documentation/devicetree/bindings/sound/max98371.txt b/Documentation/devicetree/bindings/sound/max98371.txt new file mode 100644 index 0000000..6c28523 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98371.txt @@ -0,0 +1,17 @@ +max98371 codec + +This device supports I2C mode only. + +Required properties: + +- compatible : "maxim,max98371" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + max98371: max98371@0x31 { + compatible = "maxim,max98371"; + reg = <0x31>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 913c83c..ba6c585 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98090 if I2C select SND_SOC_MAX98095 if I2C select SND_SOC_MAX98357A if GPIOLIB + select SND_SOC_MAX98371 if I2C select SND_SOC_MAX9867 if I2C select SND_SOC_MAX98925 if I2C select SND_SOC_MAX98926 if I2C @@ -491,6 +492,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX98357A tristate +config SND_SOC_MAX98371 + tristate + config SND_SOC_MAX9867 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f39dee7..9f3587b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -70,6 +70,7 @@ snd-soc-max98088-objs := max98088.o snd-soc-max98090-objs := max98090.o snd-soc-max98095-objs := max98095.o snd-soc-max98357a-objs := max98357a.o +snd-soc-max98371-objs := max98371.o snd-soc-max9867-objs := max9867.o snd-soc-max98925-objs := max98925.o snd-soc-max98926-objs := max98926.o diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c new file mode 100644 index 0000000..ce735db --- /dev/null +++ b/sound/soc/codecs/max98371.c @@ -0,0 +1,419 @@ +/* + * max98371.c -- ALSA SoC Stereo MAX98371 driver + * + * Copyright 2015-16 Maxim Integrated Products + * + * 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/i2c.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include "max98371.h" + +static const char *const monomix_text[] = { + "Left", "Right", "LeftRightDiv2", +}; + +static const char *const voltage_level_text[] = { + "9.5dB", "10.5dB", "11.5dB", "11.5dB", "12.5dB", + "13.5dB", "14.5dB", "15.5dB", "16.5dB", "17.5dB", + "18.5dB", "19.5dB", "20.5dB" +}; + +static const char *const hpf_cutoff_txt[] = { + "Disable", "DC Block", "50Hz", + "100Hz", "200Hz", "400Hz", "800Hz", +}; + +static SOC_ENUM_SINGLE_DECL(max98371_voltage, MAX98371_GAIN, 0, + voltage_level_text); + +static SOC_ENUM_SINGLE_DECL(max98371_monomix, MAX98371_MONOMIX_CFG, 0, + monomix_text); + +static SOC_ENUM_SINGLE_DECL(max98371_hpf_cutoff, MAX98371_HPF, 0, + hpf_cutoff_txt); + +static const struct reg_default max98371_reg[] = { + { 0x01, 0x00 }, + { 0x02, 0x00 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x00 }, + { 0x09, 0x00 }, + { 0x0A, 0x00 }, + { 0x10, 0x06 }, + { 0x11, 0x08 }, + { 0x14, 0x80 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1C, 0x00 }, + { 0x1D, 0x00 }, + { 0x1E, 0x00 }, + { 0x1F, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x00 }, + { 0x28, 0x00 }, + { 0x29, 0x00 }, + { 0x2A, 0x00 }, + { 0x2B, 0x00 }, + { 0x2C, 0x00 }, + { 0x2D, 0x00 }, + { 0x2E, 0x0B }, + { 0x31, 0x00 }, + { 0x32, 0x18 }, + { 0x33, 0x00 }, + { 0x34, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x00 }, + { 0x3A, 0x00 }, + { 0x3B, 0x00 }, + { 0x3B, 0x00 }, + { 0x3C, 0x00 }, + { 0x3D, 0x00 }, + { 0x3E, 0x00 }, + { 0x3F, 0x00 }, + { 0x40, 0x00 }, + { 0x41, 0x00 }, + { 0x42, 0x00 }, + { 0x43, 0x00 }, + { 0x4A, 0x00 }, + { 0x4B, 0x00 }, + { 0x4C, 0x00 }, + { 0x4D, 0x00 }, + { 0x4E, 0x00 }, + { 0x50, 0x00 }, + { 0x51, 0x00 }, + { 0x55, 0x00 }, + { 0x58, 0x00 }, + { 0x59, 0x00 }, + { 0x5C, 0x00 }, + { 0xFF, 0x43 }, +}; + +static bool max98371_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_IRQ_CLEAR1: + case MAX98371_IRQ_CLEAR2: + case MAX98371_IRQ_CLEAR3: + case MAX98371_VERSION: + return true; + default: + return false; + } +} + +static bool max98371_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case MAX98371_SOFT_RESET: + return false; + default: + return true; + } +}; + +static const DECLARE_TLV_DB_RANGE(max98371_gain_tlv, + 0, 7, TLV_DB_SCALE_ITEM(0, 50, 0), + 8, 10, TLV_DB_SCALE_ITEM(400, 100, 0) +); + +static const DECLARE_TLV_DB_RANGE(max98371_noload_gain_tlv, + 0, 11, TLV_DB_SCALE_ITEM(950, 100, 0), +); + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -6300, 50, 1); + +static const struct snd_kcontrol_new max98371_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Volume", MAX98371_GAIN, + MAX98371_GAIN_SHIFT, (1<<MAX98371_GAIN_WIDTH)-1, 0, + max98371_gain_tlv), + SOC_SINGLE_TLV("Digital Volume", MAX98371_DIGITAL_GAIN, 0, + (1<<MAX98371_DIGITAL_GAIN_WIDTH)-1, 1, digital_tlv), + SOC_SINGLE_TLV("Speaker No-load Volume", MAX98371_GAIN, + 0, (1<<MAX98371_NO_LOAD_WIDTH)-1, 0, + max98371_noload_gain_tlv), + SOC_SINGLE("DHT Attack Step", MAX98371_DHT, MAX98371_DHT_STEP, 3, 0), + SOC_SINGLE("DHT Attack Rate", MAX98371_DHT, 0, 7, 0), + SOC_ENUM("Voltage Level", max98371_voltage), + SOC_ENUM("Monomix Select", max98371_monomix), + SOC_ENUM("HPF Cutoff", max98371_hpf_cutoff), +}; + +static int max98371_dai_set_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + dev_err(codec->dev, "DAI clock mode unsupported"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + val |= 0; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val |= MAX98371_DAI_RIGHT; + break; + case SND_SOC_DAIFMT_LEFT_J: + val |= MAX98371_DAI_LEFT; + break; + default: + dev_err(codec->dev, "DAI wrong mode unsupported"); + return -EINVAL; + } + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MODE_MASK, val); + return 0; +} + +static int max98371_dai_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 max98371_priv *max98371 = snd_soc_codec_get_drvdata(codec); + int blr_clk_ratio, ch_size, channels = params_channels(params); + int rate = params_rate(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_16); + ch_size = 16; + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + regmap_update_bits(max98371->regmap, MAX98371_FMT, + MAX98371_FMT_MASK, MAX98371_DAI_CHANSZ_32); + ch_size = 32; + break; + default: + return -EINVAL; + } + + /* BCLK/LRCLK ratio calculation */ + blr_clk_ratio = channels * ch_size; + switch (blr_clk_ratio) { + case 32: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_32); + break; + case 48: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_48); + break; + case 64: + regmap_update_bits(max98371->regmap, + MAX98371_DAI_CLK, + MAX98371_DAI_BSEL_MASK, MAX98371_DAI_BSEL_64); + break; + default: + return -EINVAL; + } + + switch (rate) { + case 32000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_32); + break; + case 44100: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_44); + break; + case 48000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_48); + break; + case 88200: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_88); + break; + case 96000: + regmap_update_bits(max98371->regmap, + MAX98371_SPK_SR, + MAX98371_SPK_SR_MASK, MAX98371_SPK_SR_96); + break; + default: + return -EINVAL; + } + + /* enabling both the RX channels*/ + regmap_update_bits(max98371->regmap, MAX98371_MONOMIX_SRC, + MAX98371_MONOMIX_SRC_MASK, MONOMIX_RX_0_1); + regmap_update_bits(max98371->regmap, MAX98371_DAI_CHANNEL, + MAX98371_CHANNEL_MASK, MAX98371_CHANNEL_MASK); + return 0; +} + +static const struct snd_soc_dapm_widget max98371_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, MAX98371_SPK_ENABLE, 0, 0), + SND_SOC_DAPM_SUPPLY("Global Enable", MAX98371_GLOBAL_ENABLE, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPK_OUT"), +}; + +static const struct snd_soc_dapm_route max98371_audio_map[] = { + {"DAC", NULL, "HiFi Playback"}, + {"SPK_OUT", NULL, "DAC"}, + {"SPK_OUT", NULL, "Global Enable"}, +}; + +#define MAX98371_RATES SNDRV_PCM_RATE_8000_48000 +#define MAX98371_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE) + +static const struct snd_soc_dai_ops max98371_dai_ops = { + .set_fmt = max98371_dai_set_fmt, + .hw_params = max98371_dai_hw_params, +}; + +static struct snd_soc_dai_driver max98371_dai[] = { + { + .name = "max98371-aif1", + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = MAX98371_FORMATS, + }, + .ops = &max98371_dai_ops, + } +}; + +static const struct snd_soc_codec_driver max98371_codec = { + .controls = max98371_snd_controls, + .num_controls = ARRAY_SIZE(max98371_snd_controls), + .dapm_routes = max98371_audio_map, + .num_dapm_routes = ARRAY_SIZE(max98371_audio_map), + .dapm_widgets = max98371_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(max98371_dapm_widgets), +}; + +static const struct regmap_config max98371_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = MAX98371_VERSION, + .reg_defaults = max98371_reg, + .num_reg_defaults = ARRAY_SIZE(max98371_reg), + .volatile_reg = max98371_volatile_register, + .readable_reg = max98371_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + +static int max98371_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct max98371_priv *max98371; + int ret, reg; + + max98371 = devm_kzalloc(&i2c->dev, + sizeof(*max98371), GFP_KERNEL); + if (!max98371) + return -ENOMEM; + + i2c_set_clientdata(i2c, max98371); + max98371->regmap = devm_regmap_init_i2c(i2c, &max98371_regmap); + if (IS_ERR(max98371->regmap)) { + ret = PTR_ERR(max98371->regmap); + dev_err(&i2c->dev, + "Failed to allocate regmap: %d\n", ret); + return ret; + } + + ret = regmap_read(max98371->regmap, MAX98371_VERSION, &reg); + if (ret < 0) { + dev_info(&i2c->dev, "device error %d\n", ret); + return ret; + } + dev_info(&i2c->dev, "device version %x\n", reg); + + ret = snd_soc_register_codec(&i2c->dev, &max98371_codec, + max98371_dai, ARRAY_SIZE(max98371_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + return ret; + } + return ret; +} + +static int max98371_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id max98371_i2c_id[] = { + { "max98371", 0 }, +}; + +MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); + +static const struct of_device_id max98371_of_match[] = { + { .compatible = "maxim,max98371", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98371_of_match); + +static struct i2c_driver max98371_i2c_driver = { + .driver = { + .name = "max98371", + .owner = THIS_MODULE, + .pm = NULL, + .of_match_table = of_match_ptr(max98371_of_match), + }, + .probe = max98371_i2c_probe, + .remove = max98371_i2c_remove, + .id_table = max98371_i2c_id, +}; + +module_i2c_driver(max98371_i2c_driver); + +MODULE_AUTHOR("anish kumar <yesanishhere(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC MAX98371 driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/max98371.h b/sound/soc/codecs/max98371.h new file mode 100644 index 0000000..c75b259 --- /dev/null +++ b/sound/soc/codecs/max98371.h @@ -0,0 +1,64 @@ +/* + * max98371.h -- MAX98371 ALSA SoC Audio driver + * + * Copyright 2011-2012 Maxim Integrated Products + * + * 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 _MAX98371_H +#define _MAX98371_H + +#define MAX98371_IRQ_CLEAR1 0x01 +#define MAX98371_IRQ_CLEAR2 0x02 +#define MAX98371_IRQ_CLEAR3 0x03 +#define MAX98371_DAI_CLK 0x10 +#define MAX98371_DAI_BSEL_MASK 0xF +#define MAX98371_DAI_BSEL_32 2 +#define MAX98371_DAI_BSEL_48 3 +#define MAX98371_DAI_BSEL_64 4 +#define MAX98371_SPK_SR 0x11 +#define MAX98371_SPK_SR_MASK 0xF +#define MAX98371_SPK_SR_32 6 +#define MAX98371_SPK_SR_44 7 +#define MAX98371_SPK_SR_48 8 +#define MAX98371_SPK_SR_88 10 +#define MAX98371_SPK_SR_96 11 +#define MAX98371_DAI_CHANNEL 0x15 +#define MAX98371_CHANNEL_MASK 0x3 +#define MAX98371_MONOMIX_SRC 0x18 +#define MAX98371_MONOMIX_CFG 0x19 +#define MAX98371_HPF 0x1C +#define MAX98371_MONOMIX_SRC_MASK 0xFF +#define MONOMIX_RX_0_1 ((0x1)<<(4)) +#define M98371_DAI_CHANNEL_I2S 0x3 +#define MAX98371_DIGITAL_GAIN 0x2D +#define MAX98371_DIGITAL_GAIN_WIDTH 0x7 +#define MAX98371_GAIN 0x2E +#define MAX98371_GAIN_SHIFT 0x4 +#define MAX98371_GAIN_WIDTH 0x4 +#define MAX98371_NO_LOAD_WIDTH 4 +#define MAX98371_FMT 0x14 +#define MAX98371_CHANSZ_WIDTH 6 +#define MAX98371_FMT_MASK ((0x3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_FMT_MODE_MASK ((0x7)<<(3)) +#define MAX98371_DAI_LEFT ((0x1)<<(3)) +#define MAX98371_DAI_RIGHT ((0x2)<<(3)) +#define MAX98371_DAI_CHANSZ_16 ((1)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_24 ((2)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DAI_CHANSZ_32 ((3)<<(MAX98371_CHANSZ_WIDTH)) +#define MAX98371_DHT 0x32 +#define MAX98371_DHT_STEP 0x3 +#define MAX98371_SPK_ENABLE 0x4A +#define MAX98371_GLOBAL_ENABLE 0x50 +#define MAX98371_SOFT_RESET 0x51 +#define MAX98371_VERSION 0xFF + + +struct max98371_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; +}; +#endif -- 1.9.3

2 7

[alsa-devel] [PATCH] ASoC: atmel_ssc_dai: read DSP mode A data on rising edges of bclk
by Peter Rosin 27 Apr '16

27 Apr '16

Signed-off-by: Peter Rosin <peda(a)axentia.se> --- sound/soc/atmel/atmel_ssc_dai.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 6dca95aa8a82..4ab1e2013238 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -652,7 +652,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) | SSC_BF(RCMR_STTDLY, 1) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, SSC_CKS_DIV); @@ -692,7 +692,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) | SSC_BF(RCMR_START, SSC_START_RISING_RF) - | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) + | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? SSC_CKS_PIN : SSC_CKS_CLOCK); -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm_adsp: free memory when unloaded or closed
by Richard Fitzgerald 27 Apr '16

27 Apr '16

The patch adds a wm_adsp2_remove() function to ensure that memory is freed when the driver is unloaded or shut down. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 20 ++++++++++++++++++++ sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 21 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b6470de..c350a52 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1039,6 +1039,13 @@ static void wm_adsp_ctl_work(struct work_struct *work) kfree(ctl_work); } +static void wm_adsp_free_ctl_blk(struct wm_coeff_ctl *ctl) +{ + kfree(ctl->cache); + kfree(ctl->name); + kfree(ctl); +} + static int wm_adsp_create_control(struct wm_adsp *dsp, const struct wm_adsp_alg_region *alg_region, unsigned int offset, unsigned int len, @@ -2434,6 +2441,19 @@ int wm_adsp2_init(struct wm_adsp *dsp) } EXPORT_SYMBOL_GPL(wm_adsp2_init); +void wm_adsp2_remove(struct wm_adsp *dsp) +{ + struct wm_coeff_ctl *ctl; + + while (!list_empty(&dsp->ctl_list)) { + ctl = list_first_entry(&dsp->ctl_list, struct wm_coeff_ctl, + list); + list_del(&ctl->list); + wm_adsp_free_ctl_blk(ctl); + } +} +EXPORT_SYMBOL_GPL(wm_adsp2_remove); + int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) { struct wm_adsp_compr *compr; diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index b61cb57..feb61e2 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -92,6 +92,7 @@ extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; int wm_adsp1_init(struct wm_adsp *dsp); int wm_adsp2_init(struct wm_adsp *dsp); +void wm_adsp2_remove(struct wm_adsp *dsp); int wm_adsp2_codec_probe(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp2_codec_remove(struct wm_adsp *dsp, struct snd_soc_codec *codec); int wm_adsp1_event(struct snd_soc_dapm_widget *w, -- 1.9.1

2 5

[alsa-devel] [PATCH v4 0/2] ASoC: codecs: add support for TAS5720 digital amplifier
by Andreas Dannenberg 27 Apr '16

27 Apr '16

Minor update to driver for TI's TAS5720L/M digital audio amplifiers incorporating previous feedback (thanks Mark Brown and Andrew F Davis), see below change log for details. Also re-tested different aspects of the driver to make sure it still builds/works. -- Andreas Dannenberg Texas Instruments Inc Changes since v3: - Rework fault condition reporting to use dev_crit() to pay justice to the severity of the issues, also shorten the error string for more clarity/brevity - Fix Kconfig help entry to no longer mention the interrupt signal that was used previously - Simplify hw_params checks as the driver in its current form only supports left-justified audio formats and the codec is tolerant regarding the number of bits that are actually received. - Clarify DT bindings to make the power supply properties required - Remove .owner assignment from I2C driver struct as it is no longer needed (taken care off by the Kernel) Changes since v2: - Switched fault handling from using interrupts to polling mode - Remove interrupt related description from DT bindings doc - Remove unlikely() to simplify an expression - Remove unnesseary tas5720_set_dai_sysclk() function stub Changes since v1: - Simplified DT interrupt documentation (Thanks Rob Herring) - Fixed potential race condition during codec probe where deferred work that's used by the threaded IRQ handler was setup after the IRQ was initialized (would lead to an issue when the TAS5720 was already throwing interrupts at the time of probe) Andreas Dannenberg (2): ASoC: codecs: add TA5720 digital amplifier DT bindings ASoC: codecs: add support for TAS5720 digital amplifier .../devicetree/bindings/sound/tas5720.txt | 25 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5720.c | 620 +++++++++++++++++++++ sound/soc/codecs/tas5720.h | 90 +++ 5 files changed, 745 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5720.txt create mode 100644 sound/soc/codecs/tas5720.c create mode 100644 sound/soc/codecs/tas5720.h -- 2.6.4

2 3

[alsa-devel] [PATCH 1/2] ASoC: topology: Check failure to create a widget
by mengdong.lin＠linux.intel.com 27 Apr '16

27 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> Stop loading topology info if error happens when creating a widget. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index bde79f7..e38664e 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1500,9 +1500,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; ret = soc_tplg_dapm_widget_create(tplg, widget); - if (ret < 0) + if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", widget->name); + return ret; + } } return 0; -- 2.5.0

2 1

[alsa-devel] [PATCH 2/2] ASoC: topology: Check size mismatch of ABI objects before parsing
by mengdong.lin＠linux.intel.com 27 Apr '16

27 Apr '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> If size mismatch of manifest, ABI headers or elements is found, stop parsing topology info and return the error. New fields may be append to the tail of ABI objects which will cause object size to increase. If user space and kernel use different versions of ABI, size mismatch will be detected here. Signed-off-by: Mengdong Lin <mengdong.lin(a)linux.intel.com> diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e38664e..2261585 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1023,6 +1023,11 @@ static int soc_tplg_kcontrol_elems_load(struct soc_tplg *tplg, control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; + if (control_hdr->size != sizeof(*control_hdr)) { + dev_err(tplg->dev, "ASoC: invalid control size\n"); + return -EINVAL; + } + switch (control_hdr->ops.info) { case SND_SOC_TPLG_CTL_VOLSW: case SND_SOC_TPLG_CTL_STROBE: @@ -1499,6 +1504,11 @@ static int soc_tplg_dapm_widget_elems_load(struct soc_tplg *tplg, for (i = 0; i < count; i++) { widget = (struct snd_soc_tplg_dapm_widget *) tplg->pos; + if (widget->size != sizeof(*widget)) { + dev_err(tplg->dev, "ASoC: invalid widget size\n"); + return -EINVAL; + } + ret = soc_tplg_dapm_widget_create(tplg, widget); if (ret < 0) { dev_err(tplg->dev, "ASoC: failed to load widget %s\n", @@ -1651,8 +1661,6 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, if (tplg->pass != SOC_TPLG_PASS_PCM_DAI) return 0; - pcm = (struct snd_soc_tplg_pcm *)tplg->pos; - if (soc_tplg_check_elem_count(tplg, sizeof(struct snd_soc_tplg_pcm), count, hdr->payload_size, "PCM DAI")) { @@ -1662,7 +1670,13 @@ static int soc_tplg_pcm_elems_load(struct soc_tplg *tplg, } /* create the FE DAIs and DAI links */ + pcm = (struct snd_soc_tplg_pcm *)tplg->pos; for (i = 0; i < count; i++) { + if (pcm->size != sizeof(*pcm)) { + dev_err(tplg->dev, "ASoC: invalid pcm size\n"); + return -EINVAL; + } + soc_tplg_pcm_create(tplg, pcm); pcm++; } @@ -1682,6 +1696,11 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, return 0; manifest = (struct snd_soc_tplg_manifest *)tplg->pos; + if (manifest->size != sizeof(*manifest)) { + dev_err(tplg->dev, "ASoC: invalid manifest size\n"); + return -EINVAL; + } + tplg->pos += sizeof(struct snd_soc_tplg_manifest); if (tplg->comp && tplg->ops && tplg->ops->manifest) @@ -1698,6 +1717,14 @@ static int soc_valid_header(struct soc_tplg *tplg, if (soc_tplg_get_hdr_offset(tplg) >= tplg->fw->size) return 0; + if (hdr->size != sizeof(*hdr)) { + dev_err(tplg->dev, + "ASoC: invalid header size for type %d at offset 0x%lx size 0x%zx.\n", + hdr->type, soc_tplg_get_hdr_offset(tplg), + tplg->fw->size); + return -EINVAL; + } + /* big endian firmware objects not supported atm */ if (hdr->magic == cpu_to_be32(SND_SOC_TPLG_MAGIC)) { dev_err(tplg->dev, -- 2.5.0

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[alsa-devel] [GIT PULL] ASoC fixes for v4.6
by Mark Brown 27 Apr '16

27 Apr '16

The following changes since commit 02da2d72174c61988eb4456b53f405e3ebdebce4: Linux 4.6-rc5 (2016-04-24 16:17:05 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.6-rc5 for you to fetch changes up to 8c0f55100478649e3c0133684aa4aebab77ce10f: Merge remote-tracking branches 'asoc/fix/rt5640' and 'asoc/fix/wm8962' into asoc-linus (2016-04-26 19:25:18 +0100) ---------------------------------------------------------------- ASoC: Fixes for v4.6 This is a fairly large collection of fixes but almost all driver specific ones, especially to the new Intel drivers which have had a lot of recent development. The one core fix is a change to the debugfs code to avoid crashes in some relatively unusual configurations. ---------------------------------------------------------------- Arnd Bergmann (2): ASoC: cs35l32: avoid uninitialized variable access ASoC: rt5616: add I2C dependency Ben Zhang (1): ASoC: nau8825: Fix jack detection across suspend Charles Keepax (3): ASoC: wm8962: Correct typo when setting DSPCLK rate ASoC: arizona: Free speaker thermal IRQs in CODEC remove ASoC: wm5102: Free compressed IRQ in CODEC remove Dan Carpenter (1): ASoC: Intel: sst: fix a loop timeout in sst_hsw_stream_reset() Jeeja KP (2): ASoC: Intel: Skylake: Fix for unloading module only when it is loaded ASoC: Intel: Skylake: Fix DSP resource de-allocation Mark Brown (5): ASoC: dapm: Make sure we have a card when displaying component widgets Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/intel' into asoc-linus Merge remote-tracking branches 'asoc/fix/arizona', 'asoc/fix/cs35l32', 'asoc/fix/hdac', 'asoc/fix/nau8825' and 'asoc/fix/rt5616' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5640' and 'asoc/fix/wm8962' into asoc-linus Subhransu S. Prusty (4): ASoC: hdac_hdmi: Fix to use dev_pm ops instead soc pm ASoC: hdac_hdmi: Fix codec power state in S3 during playback ASoC: Intel: Skylake: Fix to turn OFF codec power when entering S3 ASoC: Intel: Skylake: Fix ibs/obs calc for non-integral sampling rates Sugar Zhang (1): ASoC: rt5640: Correct the digital interface data select Vinod Koul (7): ALSA: hda: use list macro for parsing on cleanup ASoC: Intel: Skylake: free codec objects on removal ASoC: Intel: Skylake: Freeup properly on skl_dsp_free ASoC: Intel: Skylake: Unmap the address last ASoC: Intel: Skylake: Call i915 exit last ASoC: Intel: Skylake: remove call to pci_dev_put ASoC: Intel: Skylake: Fix kbuild dependency sound/hda/ext/hdac_ext_stream.c | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.c | 12 +++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs35l32.c | 17 +++- sound/soc/codecs/cs47l24.c | 3 + sound/soc/codecs/hdac_hdmi.c | 94 ++++++++++------------ sound/soc/codecs/nau8825.c | 126 +++++++++++++++++------------- sound/soc/codecs/rt5640.c | 2 +- sound/soc/codecs/rt5640.h | 36 ++++----- sound/soc/codecs/wm5102.c | 5 ++ sound/soc/codecs/wm5110.c | 2 + sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8997.c | 2 + sound/soc/codecs/wm8998.c | 2 + sound/soc/intel/Kconfig | 1 - sound/soc/intel/haswell/sst-haswell-ipc.c | 2 +- sound/soc/intel/skylake/skl-sst-dsp.c | 5 ++ sound/soc/intel/skylake/skl-topology.c | 42 ++++++---- sound/soc/intel/skylake/skl-topology.h | 8 +- sound/soc/intel/skylake/skl.c | 32 +++++--- sound/soc/soc-dapm.c | 7 ++ 22 files changed, 245 insertions(+), 163 deletions(-)

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[alsa-devel] [PATCH v7 0/9] ASoC: MT8173 HDMI codec support
by Philipp Zabel 27 Apr '16

27 Apr '16

Since the hdmi-codec patches are now applied [1], I have merged the hdmi-codec interface into the initial MT8173 HDMI [2] driver submission. This series now only enables the HDMI dai-links in the mt8173-rt5650-rt5676 and mt8173-rt5650 machine drivers and adds hdmi-codec ELD support and the still up in the air hdmi notifications between DRM encoder driver and hdmi-codec. Changes since v6: - Dropped "drm/mediatek: hdmi: Add audio interface to the hdmi-codec driver", patch now merged into MT8173 HDMI encoder patch. - Fixed jack detection in case HDMI is already connected at boot and the hdmi-codec driver registers the notifier block after the HDMI_CONNECTED notification is issued. [1] commit 09184118a8ab ("ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders") of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/hdmi The mediatek drm patches apply on top of [2] https://patchwork.kernel.org/patch/8887051/ ("drm/mediatek: Add HDMI support") and need this patch: [3] https://patchwork.kernel.org/patch/8091531/ ("video: hdmi: add helper function for N and CTS") regards Philipp Koro Chen (1): ASoC: mediatek: Add HDMI dai-links in the mt8173-rt5650-rt5676 machine driver PC Liao (1): ASoC: mediatek: Add HDMI dai-links to the mt8173-rt5650 machine driver Philipp Zabel (7): ASoC: hdmi-codec: Add ELD control video: rmk's HDMI notification prototype ASoC: hdmi-codec: Use HDMI notifications to add jack support ASoC: mediatek: Add jack detection support to mt8173-rt5650-rt5676 machine driver ASoC: mediatek: Add jack detection support to the mt8173-rt5650 machine driver drm/mediatek: hdmi: issue notifications drm/mediatek: hdmi: use helper function for N and CTS calculation .../bindings/sound/mt8173-rt5650-rt5676.txt | 5 +- .../devicetree/bindings/sound/mt8173-rt5650.txt | 5 +- drivers/gpu/drm/mediatek/mtk_cec.c | 11 ++ drivers/gpu/drm/mediatek/mtk_hdmi.c | 88 ++++----------- drivers/video/Kconfig | 3 + drivers/video/Makefile | 1 + drivers/video/hdmi-notifier.c | 61 ++++++++++ include/linux/hdmi-notifier.h | 44 ++++++++ include/sound/hdmi-codec.h | 6 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdmi-codec.c | 124 +++++++++++++++++++-- sound/soc/mediatek/Kconfig | 2 + sound/soc/mediatek/mt8173-rt5650-rt5676.c | 48 ++++++++ sound/soc/mediatek/mt8173-rt5650.c | 47 ++++++++ 14 files changed, 370 insertions(+), 76 deletions(-) create mode 100644 drivers/video/hdmi-notifier.c create mode 100644 include/linux/hdmi-notifier.h -- 2.8.0.rc3

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