From 64a648c2204b0c750fe49828158751183d8b5f83 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 25 Jul 2011 11:15:15 +0100 Subject: ASoC: dapm - Add DAPM stream completion event. In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ 2 files changed, 8 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e0583b7769cb..350b1b395cac 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -524,6 +524,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a32ba8..64a9dd5a69d6 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; -- cgit v1.2.3-58-ga151 From be3ea3b9e8df64acb3606055c01291f0b58876a6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Jun 2011 19:35:29 +0100 Subject: ASoC: Use new register map API for ASoC generic physical I/O Remove all the ASoC specific physical I/O code and replace it with calls into the regmap API. The bulk write code can only be used safely if all regmap calls are locked with the CODEC lock, we need to add bulk support to the regmap API or replace the code with an open coded loop (though currently it has no users...). Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 2 + sound/soc/Kconfig | 2 + sound/soc/soc-io.c | 319 ++++++---------------------------------------------- 3 files changed, 38 insertions(+), 285 deletions(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a32ba8..4d04b4b86aa1 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include @@ -576,6 +577,7 @@ struct snd_soc_codec { const void *reg_def_copy; const struct snd_soc_cache_ops *cache_ops; struct mutex cache_rw_mutex; + int val_bytes; /* dapm */ struct snd_soc_dapm_context dapm; diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 8224db5f0434..f9054f7c1d52 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -7,6 +7,8 @@ menuconfig SND_SOC select SND_PCM select AC97_BUS if SND_SOC_AC97_BUS select SND_JACK if INPUT=y || INPUT=SND + select REGMAP_I2C if I2C + select REGMAP_SPI if SPI_MASTER ---help--- If you want ASoC support, you should say Y here and also to the diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c80589..b56e1c4bb9e6 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -13,26 +13,13 @@ #include #include +#include #include #include -#ifdef CONFIG_SPI_MASTER -static int do_spi_write(void *control, const char *data, int len) -{ - struct spi_device *spi = control; - int ret; - - ret = spi_write(spi, data, len); - if (ret < 0) - return ret; - - return len; -} -#endif - -static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value, const void *data, int len) +static int hw_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) { int ret; @@ -49,13 +36,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, return 0; } - ret = codec->hw_write(codec->control_data, data, len); - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_write(codec->control_data, reg, value); } static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) @@ -69,8 +50,11 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) if (codec->cache_only) return -1; - BUG_ON(!codec->hw_read); - return codec->hw_read(codec, reg); + ret = regmap_read(codec->control_data, reg, &val); + if (ret == 0) + return val; + else + return ret; } ret = snd_soc_cache_read(codec, reg, &val); @@ -79,183 +63,18 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) return val; } -static int snd_soc_4_12_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 12) | (value & 0xffffff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_7_9_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data; - - data = cpu_to_be16((reg << 9) | (value & 0x1ff)); - - return do_hw_write(codec, reg, value, &data, 2); -} - -static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[2]; - - reg &= 0xff; - data[0] = reg; - data[1] = value & 0xff; - - return do_hw_write(codec, reg, value, data, 2); -} - -static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 val = cpu_to_be16(value); - - data[0] = reg; - memcpy(&data[1], &val, sizeof(val)); - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int do_i2c_read(struct snd_soc_codec *codec, - void *reg, int reglen, - void *data, int datalen) -{ - struct i2c_msg xfer[2]; - int ret; - struct i2c_client *client = codec->control_data; - - /* Write register */ - xfer[0].addr = client->addr; - xfer[0].flags = 0; - xfer[0].len = reglen; - xfer[0].buf = reg; - - /* Read data */ - xfer[1].addr = client->addr; - xfer[1].flags = I2C_M_RD; - xfer[1].len = datalen; - xfer[1].buf = data; - - ret = i2c_transfer(client->adapter, xfer, 2); - if (ret == 2) - return 0; - else if (ret < 0) - return ret; - else - return -EIO; -} -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_8_8_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_8_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u8 reg = r; - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 1, &data, 2); - if (ret < 0) - return 0; - return (data >> 8) | ((data & 0xff) << 8); -} -#else -#define snd_soc_8_16_read_i2c NULL -#endif - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_8_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = r; - u8 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 1); - if (ret < 0) - return 0; - return data; -} -#else -#define snd_soc_16_8_read_i2c NULL -#endif - -static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u8 data[3]; - u16 rval = cpu_to_be16(reg); - - memcpy(data, &rval, sizeof(rval)); - data[2] = value; - - return do_hw_write(codec, reg, value, data, 3); -} - -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) -static unsigned int snd_soc_16_16_read_i2c(struct snd_soc_codec *codec, - unsigned int r) -{ - u16 reg = cpu_to_be16(r); - u16 data; - int ret; - - ret = do_i2c_read(codec, ®, 2, &data, 2); - if (ret < 0) - return 0; - return be16_to_cpu(data); -} -#else -#define snd_soc_16_16_read_i2c NULL -#endif - -static int snd_soc_16_16_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - u16 data[2]; - - data[0] = cpu_to_be16(reg); - data[1] = cpu_to_be16(value); - - return do_hw_write(codec, reg, value, data, sizeof(data)); -} - /* Primitive bulk write support for soc-cache. The data pointed to by - * `data' needs to already be in the form the hardware expects - * including any leading register specific data. Any data written - * through this function will not go through the cache as it only - * handles writing to volatile or out of bounds registers. + * `data' needs to already be in the form the hardware expects. Any + * data written through this function will not go through the cache as + * it only handles writing to volatile or out of bounds registers. + * + * This is currently only supported for devices using the regmap API + * wrappers. */ -static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int reg, +static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, + unsigned int reg, const void *data, size_t len) { - int ret; - /* To ensure that we don't get out of sync with the cache, check * whether the base register is volatile or if we've directly asked * to bypass the cache. Out of bounds registers are considered @@ -266,66 +85,9 @@ static int snd_soc_hw_bulk_write_raw(struct snd_soc_codec *codec, unsigned int r && reg < codec->driver->reg_cache_size) return -EINVAL; - switch (codec->control_type) { -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - case SND_SOC_I2C: - ret = i2c_master_send(to_i2c_client(codec->dev), data, len); - break; -#endif -#if defined(CONFIG_SPI_MASTER) - case SND_SOC_SPI: - ret = spi_write(to_spi_device(codec->dev), data, len); - break; -#endif - default: - BUG(); - } - - if (ret == len) - return 0; - if (ret < 0) - return ret; - else - return -EIO; + return regmap_raw_write(codec->control_data, reg, data, len); } -static struct { - int addr_bits; - int data_bits; - int (*write)(struct snd_soc_codec *codec, unsigned int, unsigned int); - unsigned int (*read)(struct snd_soc_codec *, unsigned int); - unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); -} io_types[] = { - { - .addr_bits = 4, .data_bits = 12, - .write = snd_soc_4_12_write, - }, - { - .addr_bits = 7, .data_bits = 9, - .write = snd_soc_7_9_write, - }, - { - .addr_bits = 8, .data_bits = 8, - .write = snd_soc_8_8_write, - .i2c_read = snd_soc_8_8_read_i2c, - }, - { - .addr_bits = 8, .data_bits = 16, - .write = snd_soc_8_16_write, - .i2c_read = snd_soc_8_16_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 8, - .write = snd_soc_16_8_write, - .i2c_read = snd_soc_16_8_read_i2c, - }, - { - .addr_bits = 16, .data_bits = 16, - .write = snd_soc_16_16_write, - .i2c_read = snd_soc_16_16_read_i2c, - }, -}; - /** * snd_soc_codec_set_cache_io: Set up standard I/O functions. * @@ -349,47 +111,34 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, int addr_bits, int data_bits, enum snd_soc_control_type control) { - int i; - - for (i = 0; i < ARRAY_SIZE(io_types); i++) - if (io_types[i].addr_bits == addr_bits && - io_types[i].data_bits == data_bits) - break; - if (i == ARRAY_SIZE(io_types)) { - printk(KERN_ERR - "No I/O functions for %d bit address %d bit data\n", - addr_bits, data_bits); - return -EINVAL; - } + struct regmap_config config; - codec->write = io_types[i].write; + memset(&config, 0, sizeof(config)); + codec->write = hw_write; codec->read = hw_read; codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; + config.reg_bits = addr_bits; + config.val_bits = data_bits; + switch (control) { case SND_SOC_I2C: -#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) - codec->hw_write = (hw_write_t)i2c_master_send; -#endif - if (io_types[i].i2c_read) - codec->hw_read = io_types[i].i2c_read; - - codec->control_data = container_of(codec->dev, - struct i2c_client, - dev); + codec->control_data = regmap_init_i2c(to_i2c_client(codec->dev), + &config); break; case SND_SOC_SPI: -#ifdef CONFIG_SPI_MASTER - codec->hw_write = do_spi_write; -#endif - - codec->control_data = container_of(codec->dev, - struct spi_device, - dev); + codec->control_data = regmap_init_spi(to_spi_device(codec->dev), + &config); break; + + default: + return -EINVAL; } + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); -- cgit v1.2.3-58-ga151 From 0671da189c1d75eec5f6aba786d57d25209dd2bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 Jul 2011 12:23:37 +0100 Subject: ASoC: Add regmap as a control type Allow drivers to set up their own regmap API structures. This is mainly useful with MFDs where the core driver will have set up regmap at the minute, though it may make sense to push the existing regmap setup out of the core into the drivers. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 1 + sound/soc/soc-io.c | 4 ++++ 2 files changed, 5 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 4d04b4b86aa1..d02269437de3 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -261,6 +261,7 @@ extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, SND_SOC_SPI, + SND_SOC_REGMAP, }; enum snd_soc_compress_type { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index b56e1c4bb9e6..e471ed667fe9 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -132,6 +132,10 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, &config); break; + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ + break; + default: return -EINVAL; } -- cgit v1.2.3-58-ga151 From ddd7a26094c93a950f4b2e6b4d5865c93976372e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 15 Aug 2011 20:15:22 +0200 Subject: ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/adau1373.h | 34 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adau1373.c | 1414 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1373.h | 29 + 5 files changed, 1483 insertions(+) create mode 100644 include/sound/adau1373.h create mode 100644 sound/soc/codecs/adau1373.c create mode 100644 sound/soc/codecs/adau1373.h (limited to 'include/sound') diff --git a/include/sound/adau1373.h b/include/sound/adau1373.h new file mode 100644 index 000000000000..1b19c7666574 --- /dev/null +++ b/include/sound/adau1373.h @@ -0,0 +1,34 @@ +/* + * Analog Devices ADAU1373 Audio Codec drive + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2 or later. + */ + +#ifndef __SOUND_ADAU1373_H__ +#define __SOUND_ADAU1373_H__ + +enum adau1373_micbias_voltage { + ADAU1373_MICBIAS_2_9V = 0, + ADAU1373_MICBIAS_2_2V = 1, + ADAU1373_MICBIAS_2_6V = 2, + ADAU1373_MICBIAS_1_8V = 3, +}; + +#define ADAU1373_DRC_SIZE 13 + +struct adau1373_platform_data { + bool input_differential[4]; + bool lineout_differential; + bool lineout_ground_sense; + + unsigned int num_drc; + uint8_t drc_setting[3][ADAU1373_DRC_SIZE]; + + enum adau1373_micbias_voltage micbias1; + enum adau1373_micbias_voltage micbias2; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 665d9240c4ae..71b46c8f70d7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -139,6 +140,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAU1373 + tristate + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5119a7e2c1a8..70c1769acd15 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o @@ -100,6 +101,7 @@ obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o +obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c new file mode 100644 index 000000000000..2aa40c3731d0 --- /dev/null +++ b/sound/soc/codecs/adau1373.c @@ -0,0 +1,1414 @@ +/* + * Analog Devices ADAU1373 Audio Codec drive + * + * Copyright 2011 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2 or later. + */ + +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "adau1373.h" + +struct adau1373_dai { + unsigned int clk_src; + unsigned int sysclk; + bool enable_src; + bool master; +}; + +struct adau1373 { + struct adau1373_dai dais[3]; +}; + +#define ADAU1373_INPUT_MODE 0x00 +#define ADAU1373_AINL_CTRL(x) (0x01 + (x) * 2) +#define ADAU1373_AINR_CTRL(x) (0x02 + (x) * 2) +#define ADAU1373_LLINE_OUT(x) (0x9 + (x) * 2) +#define ADAU1373_RLINE_OUT(x) (0xa + (x) * 2) +#define ADAU1373_LSPK_OUT 0x0d +#define ADAU1373_RSPK_OUT 0x0e +#define ADAU1373_LHP_OUT 0x0f +#define ADAU1373_RHP_OUT 0x10 +#define ADAU1373_ADC_GAIN 0x11 +#define ADAU1373_LADC_MIXER 0x12 +#define ADAU1373_RADC_MIXER 0x13 +#define ADAU1373_LLINE1_MIX 0x14 +#define ADAU1373_RLINE1_MIX 0x15 +#define ADAU1373_LLINE2_MIX 0x16 +#define ADAU1373_RLINE2_MIX 0x17 +#define ADAU1373_LSPK_MIX 0x18 +#define ADAU1373_RSPK_MIX 0x19 +#define ADAU1373_LHP_MIX 0x1a +#define ADAU1373_RHP_MIX 0x1b +#define ADAU1373_EP_MIX 0x1c +#define ADAU1373_HP_CTRL 0x1d +#define ADAU1373_HP_CTRL2 0x1e +#define ADAU1373_LS_CTRL 0x1f +#define ADAU1373_EP_CTRL 0x21 +#define ADAU1373_MICBIAS_CTRL1 0x22 +#define ADAU1373_MICBIAS_CTRL2 0x23 +#define ADAU1373_OUTPUT_CTRL 0x24 +#define ADAU1373_PWDN_CTRL1 0x25 +#define ADAU1373_PWDN_CTRL2 0x26 +#define ADAU1373_PWDN_CTRL3 0x27 +#define ADAU1373_DPLL_CTRL(x) (0x28 + (x) * 7) +#define ADAU1373_PLL_CTRL1(x) (0x29 + (x) * 7) +#define ADAU1373_PLL_CTRL2(x) (0x2a + (x) * 7) +#define ADAU1373_PLL_CTRL3(x) (0x2b + (x) * 7) +#define ADAU1373_PLL_CTRL4(x) (0x2c + (x) * 7) +#define ADAU1373_PLL_CTRL5(x) (0x2d + (x) * 7) +#define ADAU1373_PLL_CTRL6(x) (0x2e + (x) * 7) +#define ADAU1373_PLL_CTRL7(x) (0x2f + (x) * 7) +#define ADAU1373_HEADDECT 0x36 +#define ADAU1373_ADC_DAC_STATUS 0x37 +#define ADAU1373_ADC_CTRL 0x3c +#define ADAU1373_DAI(x) (0x44 + (x)) +#define ADAU1373_CLK_SRC_DIV(x) (0x40 + (x) * 2) +#define ADAU1373_BCLKDIV(x) (0x47 + (x)) +#define ADAU1373_SRC_RATIOA(x) (0x4a + (x) * 2) +#define ADAU1373_SRC_RATIOB(x) (0x4b + (x) * 2) +#define ADAU1373_DEEMP_CTRL 0x50 +#define ADAU1373_SRC_DAI_CTRL(x) (0x51 + (x)) +#define ADAU1373_DIN_MIX_CTRL(x) (0x56 + (x)) +#define ADAU1373_DOUT_MIX_CTRL(x) (0x5b + (x)) +#define ADAU1373_DAI_PBL_VOL(x) (0x62 + (x) * 2) +#define ADAU1373_DAI_PBR_VOL(x) (0x63 + (x) * 2) +#define ADAU1373_DAI_RECL_VOL(x) (0x68 + (x) * 2) +#define ADAU1373_DAI_RECR_VOL(x) (0x69 + (x) * 2) +#define ADAU1373_DAC1_PBL_VOL 0x6e +#define ADAU1373_DAC1_PBR_VOL 0x6f +#define ADAU1373_DAC2_PBL_VOL 0x70 +#define ADAU1373_DAC2_PBR_VOL 0x71 +#define ADAU1373_ADC_RECL_VOL 0x72 +#define ADAU1373_ADC_RECR_VOL 0x73 +#define ADAU1373_DMIC_RECL_VOL 0x74 +#define ADAU1373_DMIC_RECR_VOL 0x75 +#define ADAU1373_VOL_GAIN1 0x76 +#define ADAU1373_VOL_GAIN2 0x77 +#define ADAU1373_VOL_GAIN3 0x78 +#define ADAU1373_HPF_CTRL 0x7d +#define ADAU1373_BASS1 0x7e +#define ADAU1373_BASS2 0x7f +#define ADAU1373_DRC(x) (0x80 + (x) * 0x10) +#define ADAU1373_3D_CTRL1 0xc0 +#define ADAU1373_3D_CTRL2 0xc1 +#define ADAU1373_FDSP_SEL1 0xdc +#define ADAU1373_FDSP_SEL2 0xdd +#define ADAU1373_FDSP_SEL3 0xde +#define ADAU1373_FDSP_SEL4 0xdf +#define ADAU1373_DIGMICCTRL 0xe2 +#define ADAU1373_DIGEN 0xeb +#define ADAU1373_SOFT_RESET 0xff + + +#define ADAU1373_PLL_CTRL6_DPLL_BYPASS BIT(1) +#define ADAU1373_PLL_CTRL6_PLL_EN BIT(0) + +#define ADAU1373_DAI_INVERT_BCLK BIT(7) +#define ADAU1373_DAI_MASTER BIT(6) +#define ADAU1373_DAI_INVERT_LRCLK BIT(4) +#define ADAU1373_DAI_WLEN_16 0x0 +#define ADAU1373_DAI_WLEN_20 0x4 +#define ADAU1373_DAI_WLEN_24 0x8 +#define ADAU1373_DAI_WLEN_32 0xc +#define ADAU1373_DAI_WLEN_MASK 0xc +#define ADAU1373_DAI_FORMAT_RIGHT_J 0x0 +#define ADAU1373_DAI_FORMAT_LEFT_J 0x1 +#define ADAU1373_DAI_FORMAT_I2S 0x2 +#define ADAU1373_DAI_FORMAT_DSP 0x3 + +#define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_32 0x03 +#define ADAU1373_BCLKDIV_64 0x02 +#define ADAU1373_BCLKDIV_128 0x01 +#define ADAU1373_BCLKDIV_256 0x00 + +#define ADAU1373_ADC_CTRL_PEAK_DETECT BIT(0) +#define ADAU1373_ADC_CTRL_RESET BIT(1) +#define ADAU1373_ADC_CTRL_RESET_FORCE BIT(2) + +#define ADAU1373_OUTPUT_CTRL_LDIFF BIT(3) +#define ADAU1373_OUTPUT_CTRL_LNFBEN BIT(2) + +#define ADAU1373_PWDN_CTRL3_PWR_EN BIT(0) + +#define ADAU1373_EP_CTRL_MICBIAS1_OFFSET 4 +#define ADAU1373_EP_CTRL_MICBIAS2_OFFSET 2 + +static const uint8_t adau1373_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x00 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x10 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x20 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, /* 0x30 */ + 0x00, 0x00, 0x00, 0x80, 0x00, 0x01, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x0a, 0x0a, 0x0a, 0x00, /* 0x40 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x08, 0x08, 0x08, 0x00, 0x00, 0x00, 0x00, /* 0x50 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x60 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0x70 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x80 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0x90 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x78, 0x18, 0x00, 0x00, 0x00, 0xc0, 0x00, 0x00, /* 0xa0 */ + 0x00, 0xc0, 0x88, 0x7a, 0xdf, 0x20, 0x00, 0x00, + 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0xff, /* 0xb0 */ + 0xff, 0xff, 0xff, 0xff, 0xff, 0x1f, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xc0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 0xd0 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 0xe0 */ + 0x00, 0x1f, 0x0f, 0x00, 0x00, +}; + +static const unsigned int adau1373_out_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 7, TLV_DB_SCALE_ITEM(-7900, 400, 1), + 8, 15, TLV_DB_SCALE_ITEM(-4700, 300, 0), + 16, 23, TLV_DB_SCALE_ITEM(-2300, 200, 0), + 24, 31, TLV_DB_SCALE_ITEM(-700, 100, 0), +}; + +static const DECLARE_TLV_DB_MINMAX(adau1373_digital_tlv, -9563, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_in_pga_tlv, -1300, 100, 1); +static const DECLARE_TLV_DB_SCALE(adau1373_ep_tlv, -600, 600, 1); + +static const DECLARE_TLV_DB_SCALE(adau1373_input_boost_tlv, 0, 2000, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_gain_boost_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adau1373_speaker_boost_tlv, 1200, 600, 0); + +static const char *adau1373_fdsp_sel_text[] = { + "None", + "Channel 1", + "Channel 2", + "Channel 3", + "Channel 4", + "Channel 5", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, + ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, + ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, + ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, + ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, + ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text); + +static const char *adau1373_hpf_cutoff_text[] = { + "3.7Hz", "50Hz", "100Hz", "150Hz", "200Hz", "250Hz", "300Hz", "350Hz", + "400Hz", "450Hz", "500Hz", "550Hz", "600Hz", "650Hz", "700Hz", "750Hz", + "800Hz", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, + ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text); + +static const char *adau1373_bass_lpf_cutoff_text[] = { + "801Hz", "1001Hz", +}; + +static const char *adau1373_bass_clip_level_text[] = { + "0.125", "0.250", "0.370", "0.500", "0.625", "0.750", "0.875", +}; + +static const unsigned int adau1373_bass_clip_level_values[] = { + 1, 2, 3, 4, 5, 6, 7, +}; + +static const char *adau1373_bass_hpf_cutoff_text[] = { + "158Hz", "232Hz", "347Hz", "520Hz", +}; + +static const unsigned int adau1373_bass_tlv[] = { + TLV_DB_RANGE_HEAD(4), + 0, 2, TLV_DB_SCALE_ITEM(-600, 600, 1), + 3, 4, TLV_DB_SCALE_ITEM(950, 250, 0), + 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, + ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text); + +static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, + ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text, + adau1373_bass_clip_level_values); + +static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, + ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text); + +static const char *adau1373_3d_level_text[] = { + "0%", "6.67%", "13.33%", "20%", "26.67%", "33.33%", + "40%", "46.67%", "53.33%", "60%", "66.67%", "73.33%", + "80%", "86.67", "99.33%", "100%" +}; + +static const char *adau1373_3d_cutoff_text[] = { + "No 3D", "0.03125 fs", "0.04583 fs", "0.075 fs", "0.11458 fs", + "0.16875 fs", "0.27083 fs" +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, + ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, + ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text); + +static const unsigned int adau1373_3d_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 7, TLV_DB_LINEAR_ITEM(-1800, -120), +}; + +static const char *adau1373_lr_mux_text[] = { + "Mute", + "Right Channel (L+R)", + "Left Channel (L+R)", + "Stereo", +}; + +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, + ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text); +static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, + ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text); + +static const struct snd_kcontrol_new adau1373_controls[] = { + SOC_DOUBLE_R_TLV("AIF1 Capture Volume", ADAU1373_DAI_RECL_VOL(0), + ADAU1373_DAI_RECR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Capture Volume", ADAU1373_DAI_RECL_VOL(1), + ADAU1373_DAI_RECR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Capture Volume", ADAU1373_DAI_RECL_VOL(2), + ADAU1373_DAI_RECR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("ADC Capture Volume", ADAU1373_ADC_RECL_VOL, + ADAU1373_ADC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DMIC Capture Volume", ADAU1373_DMIC_RECL_VOL, + ADAU1373_DMIC_RECR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("AIF1 Playback Volume", ADAU1373_DAI_PBL_VOL(0), + ADAU1373_DAI_PBR_VOL(0), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF2 Playback Volume", ADAU1373_DAI_PBL_VOL(1), + ADAU1373_DAI_PBR_VOL(1), 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("AIF3 Playback Volume", ADAU1373_DAI_PBL_VOL(2), + ADAU1373_DAI_PBR_VOL(2), 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", ADAU1373_DAC1_PBL_VOL, + ADAU1373_DAC1_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", ADAU1373_DAC2_PBL_VOL, + ADAU1373_DAC2_PBR_VOL, 0, 0xff, 1, adau1373_digital_tlv), + + SOC_DOUBLE_R_TLV("Lineout1 Playback Volume", ADAU1373_LLINE_OUT(0), + ADAU1373_RLINE_OUT(0), 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Speaker Playback Volume", ADAU1373_LSPK_OUT, + ADAU1373_RSPK_OUT, 0, 0x1f, 0, adau1373_out_tlv), + SOC_DOUBLE_R_TLV("Headphone Playback Volume", ADAU1373_LHP_OUT, + ADAU1373_RHP_OUT, 0, 0x1f, 0, adau1373_out_tlv), + + SOC_DOUBLE_R_TLV("Input 1 Capture Volume", ADAU1373_AINL_CTRL(0), + ADAU1373_AINR_CTRL(0), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 2 Capture Volume", ADAU1373_AINL_CTRL(1), + ADAU1373_AINR_CTRL(1), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 3 Capture Volume", ADAU1373_AINL_CTRL(2), + ADAU1373_AINR_CTRL(2), 0, 0x1f, 0, adau1373_in_pga_tlv), + SOC_DOUBLE_R_TLV("Input 4 Capture Volume", ADAU1373_AINL_CTRL(3), + ADAU1373_AINR_CTRL(3), 0, 0x1f, 0, adau1373_in_pga_tlv), + + SOC_SINGLE_TLV("Earpiece Playback Volume", ADAU1373_EP_CTRL, 0, 3, 0, + adau1373_ep_tlv), + + SOC_DOUBLE_TLV("AIF3 Boost Playback Volume", ADAU1373_VOL_GAIN1, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Playback Volume", ADAU1373_VOL_GAIN1, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Playback Volume", ADAU1373_VOL_GAIN1, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF3 Boost Capture Volume", ADAU1373_VOL_GAIN2, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF2 Boost Capture Volume", ADAU1373_VOL_GAIN2, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("AIF1 Boost Capture Volume", ADAU1373_VOL_GAIN2, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DMIC Boost Capture Volume", ADAU1373_VOL_GAIN3, 6, 7, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("ADC Boost Capture Volume", ADAU1373_VOL_GAIN3, 4, 5, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC2 Boost Playback Volume", ADAU1373_VOL_GAIN3, 2, 3, + 1, 0, adau1373_gain_boost_tlv), + SOC_DOUBLE_TLV("DAC1 Boost Playback Volume", ADAU1373_VOL_GAIN3, 0, 1, + 1, 0, adau1373_gain_boost_tlv), + + SOC_DOUBLE_TLV("Input 1 Boost Capture Volume", ADAU1373_ADC_GAIN, 0, 4, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 2 Boost Capture Volume", ADAU1373_ADC_GAIN, 1, 5, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 3 Boost Capture Volume", ADAU1373_ADC_GAIN, 2, 6, + 1, 0, adau1373_input_boost_tlv), + SOC_DOUBLE_TLV("Input 4 Boost Capture Volume", ADAU1373_ADC_GAIN, 3, 7, + 1, 0, adau1373_input_boost_tlv), + + SOC_DOUBLE_TLV("Speaker Boost Playback Volume", ADAU1373_LS_CTRL, 2, 3, + 1, 0, adau1373_speaker_boost_tlv), + + SOC_ENUM("Lineout1 LR Mux", adau1373_lineout1_lr_mux_enum), + SOC_ENUM("Speaker LR Mux", adau1373_speaker_lr_mux_enum), + + SOC_ENUM("HPF Cutoff", adau1373_hpf_cutoff_enum), + SOC_DOUBLE("HPF Switch", ADAU1373_HPF_CTRL, 1, 0, 1, 0), + SOC_ENUM("HPF Channel", adau1373_hpf_channel_enum), + + SOC_ENUM("Bass HPF Cutoff", adau1373_bass_hpf_cutoff_enum), + SOC_VALUE_ENUM("Bass Clip Level Threshold", + adau1373_bass_clip_level_enum), + SOC_ENUM("Bass LPF Cutoff", adau1373_bass_lpf_cutoff_enum), + SOC_DOUBLE("Bass Playback Switch", ADAU1373_BASS2, 0, 1, 1, 0), + SOC_SINGLE_TLV("Bass Playback Volume", ADAU1373_BASS2, 2, 7, 0, + adau1373_bass_tlv), + SOC_ENUM("Bass Channel", adau1373_bass_channel_enum), + + SOC_ENUM("3D Freq", adau1373_3d_cutoff_enum), + SOC_ENUM("3D Level", adau1373_3d_level_enum), + SOC_SINGLE("3D Playback Switch", ADAU1373_3D_CTRL2, 0, 1, 0), + SOC_SINGLE_TLV("3D Playback Volume", ADAU1373_3D_CTRL2, 2, 7, 0, + adau1373_3d_tlv), + SOC_ENUM("3D Channel", adau1373_bass_channel_enum), + + SOC_SINGLE("Zero Cross Switch", ADAU1373_PWDN_CTRL3, 7, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_lineout2_controls[] = { + SOC_DOUBLE_R_TLV("Lineout2 Playback Volume", ADAU1373_LLINE_OUT(1), + ADAU1373_RLINE_OUT(1), 0, 0x1f, 0, adau1373_out_tlv), + SOC_ENUM("Lineout2 LR Mux", adau1373_lineout2_lr_mux_enum), +}; + +static const struct snd_kcontrol_new adau1373_drc_controls[] = { + SOC_ENUM("DRC1 Channel", adau1373_drc1_channel_enum), + SOC_ENUM("DRC2 Channel", adau1373_drc2_channel_enum), + SOC_ENUM("DRC3 Channel", adau1373_drc3_channel_enum), +}; + +static int adau1373_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int pll_id = w->name[3] - '1'; + unsigned int val; + + if (SND_SOC_DAPM_EVENT_ON(event)) + val = ADAU1373_PLL_CTRL6_PLL_EN; + else + val = 0; + + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_PLL_EN, val); + + if (SND_SOC_DAPM_EVENT_ON(event)) + mdelay(5); + + return 0; +} + +static const char *adau1373_decimator_text[] = { + "ADC", + "DMIC1", +}; + +static const struct soc_enum adau1373_decimator_enum = + SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); + +static const struct snd_kcontrol_new adau1373_decimator_mux = + SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); + +static const struct snd_kcontrol_new adau1373_left_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_LADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_LADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_LADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_LADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_LADC_MIXER, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_adc_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC1 Switch", ADAU1373_RADC_MIXER, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Switch", ADAU1373_RADC_MIXER, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Switch", ADAU1373_RADC_MIXER, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Switch", ADAU1373_RADC_MIXER, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Switch", ADAU1373_RADC_MIXER, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("Left DAC2 Switch", _reg, 7, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC2 Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("Left DAC1 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("Right DAC1 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("Input 4 Bypass Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("Input 3 Bypass Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("Input 2 Bypass Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("Input 1 Bypass Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line1_mixer_controls, + ADAU1373_LLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line1_mixer_controls, + ADAU1373_RLINE1_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_line2_mixer_controls, + ADAU1373_LLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_line2_mixer_controls, + ADAU1373_RLINE2_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_left_spk_mixer_controls, + ADAU1373_LSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_right_spk_mixer_controls, + ADAU1373_RSPK_MIX); +static DECLARE_ADAU1373_OUTPUT_MIXER_CTRLS(adau1373_ep_mixer_controls, + ADAU1373_EP_MIX); + +static const struct snd_kcontrol_new adau1373_left_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC1 Switch", ADAU1373_LHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Left DAC2 Switch", ADAU1373_LHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_LHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_LHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_LHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_LHP_MIX, 0, 1, 0), +}; + +static const struct snd_kcontrol_new adau1373_right_hp_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC1 Switch", ADAU1373_RHP_MIX, 5, 1, 0), + SOC_DAPM_SINGLE("Right DAC2 Switch", ADAU1373_RHP_MIX, 4, 1, 0), + SOC_DAPM_SINGLE("Input 4 Bypass Switch", ADAU1373_RHP_MIX, 3, 1, 0), + SOC_DAPM_SINGLE("Input 3 Bypass Switch", ADAU1373_RHP_MIX, 2, 1, 0), + SOC_DAPM_SINGLE("Input 2 Bypass Switch", ADAU1373_RHP_MIX, 1, 1, 0), + SOC_DAPM_SINGLE("Input 1 Bypass Switch", ADAU1373_RHP_MIX, 0, 1, 0), +}; + +#define DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DMIC2 Swapped Switch", _reg, 6, 1, 0), \ + SOC_DAPM_SINGLE("DMIC2 Switch", _reg, 5, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Swapped Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("ADC/DMIC1 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("AIF3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("AIF2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("AIF1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel1_mixer_controls, + ADAU1373_DIN_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel2_mixer_controls, + ADAU1373_DIN_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel3_mixer_controls, + ADAU1373_DIN_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel4_mixer_controls, + ADAU1373_DIN_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_CHANNEL_MIXER_CTRLS(adau1373_dsp_channel5_mixer_controls, + ADAU1373_DIN_MIX_CTRL(4)); + +#define DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(_name, _reg) \ +const struct snd_kcontrol_new _name[] = { \ + SOC_DAPM_SINGLE("DSP Channel5 Switch", _reg, 4, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel4 Switch", _reg, 3, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel3 Switch", _reg, 2, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel2 Switch", _reg, 1, 1, 0), \ + SOC_DAPM_SINGLE("DSP Channel1 Switch", _reg, 0, 1, 0), \ +} + +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(0)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(1)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_aif3_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(2)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac1_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(3)); +static DECLARE_ADAU1373_DSP_OUTPUT_MIXER_CTRLS(adau1373_dac2_mixer_controls, + ADAU1373_DOUT_MIX_CTRL(4)); + +static const struct snd_soc_dapm_widget adau1373_dapm_widgets[] = { + /* Datasheet claims Left ADC is bit 6 and Right ADC is bit 7, but that + * doesn't seem to be the case. */ + SND_SOC_DAPM_ADC("Left ADC", NULL, ADAU1373_PWDN_CTRL1, 7, 0), + SND_SOC_DAPM_ADC("Right ADC", NULL, ADAU1373_PWDN_CTRL1, 6, 0), + + SND_SOC_DAPM_ADC("DMIC1", NULL, ADAU1373_DIGMICCTRL, 0, 0), + SND_SOC_DAPM_ADC("DMIC2", NULL, ADAU1373_DIGMICCTRL, 2, 0), + + SND_SOC_DAPM_VIRT_MUX("Decimator Mux", SND_SOC_NOPM, 0, 0, + &adau1373_decimator_mux), + + SND_SOC_DAPM_SUPPLY("MICBIAS2", ADAU1373_PWDN_CTRL1, 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS1", ADAU1373_PWDN_CTRL1, 4, 0, NULL, 0), + + SND_SOC_DAPM_PGA("IN4PGA", ADAU1373_PWDN_CTRL1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN3PGA", ADAU1373_PWDN_CTRL1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN2PGA", ADAU1373_PWDN_CTRL1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("IN1PGA", ADAU1373_PWDN_CTRL1, 0, 0, NULL, 0), + + SND_SOC_DAPM_DAC("Left DAC2", NULL, ADAU1373_PWDN_CTRL2, 7, 0), + SND_SOC_DAPM_DAC("Right DAC2", NULL, ADAU1373_PWDN_CTRL2, 6, 0), + SND_SOC_DAPM_DAC("Left DAC1", NULL, ADAU1373_PWDN_CTRL2, 5, 0), + SND_SOC_DAPM_DAC("Right DAC1", NULL, ADAU1373_PWDN_CTRL2, 4, 0), + + SOC_MIXER_ARRAY("Left ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_adc_mixer_controls), + SOC_MIXER_ARRAY("Right ADC Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_adc_mixer_controls), + + SOC_MIXER_ARRAY("Left Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 3, 0, + adau1373_left_line2_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout2 Mixer", ADAU1373_PWDN_CTRL2, 2, 0, + adau1373_right_line2_mixer_controls), + SOC_MIXER_ARRAY("Left Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 1, 0, + adau1373_left_line1_mixer_controls), + SOC_MIXER_ARRAY("Right Lineout1 Mixer", ADAU1373_PWDN_CTRL2, 0, 0, + adau1373_right_line1_mixer_controls), + + SOC_MIXER_ARRAY("Earpiece Mixer", ADAU1373_PWDN_CTRL3, 4, 0, + adau1373_ep_mixer_controls), + SOC_MIXER_ARRAY("Left Speaker Mixer", ADAU1373_PWDN_CTRL3, 3, 0, + adau1373_left_spk_mixer_controls), + SOC_MIXER_ARRAY("Right Speaker Mixer", ADAU1373_PWDN_CTRL3, 2, 0, + adau1373_right_spk_mixer_controls), + SOC_MIXER_ARRAY("Left Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_left_hp_mixer_controls), + SOC_MIXER_ARRAY("Right Headphone Mixer", SND_SOC_NOPM, 0, 0, + adau1373_right_hp_mixer_controls), + SND_SOC_DAPM_SUPPLY("Headphone Enable", ADAU1373_PWDN_CTRL3, 1, 0, + NULL, 0), + + SND_SOC_DAPM_SUPPLY("AIF1 CLK", ADAU1373_SRC_DAI_CTRL(0), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 CLK", ADAU1373_SRC_DAI_CTRL(1), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 CLK", ADAU1373_SRC_DAI_CTRL(2), 0, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 IN SRC", ADAU1373_SRC_DAI_CTRL(0), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF1 OUT SRC", ADAU1373_SRC_DAI_CTRL(0), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 IN SRC", ADAU1373_SRC_DAI_CTRL(1), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF2 OUT SRC", ADAU1373_SRC_DAI_CTRL(1), 1, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 IN SRC", ADAU1373_SRC_DAI_CTRL(2), 2, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("AIF3 OUT SRC", ADAU1373_SRC_DAI_CTRL(2), 1, 0, + NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIF1 IN", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1 OUT", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2 IN", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2 OUT", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF3 IN", "AIF3 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF3 OUT", "AIF3 Capture", 0, SND_SOC_NOPM, 0, 0), + + SOC_MIXER_ARRAY("DSP Channel1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel1_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel2_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel3_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel4 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel4_mixer_controls), + SOC_MIXER_ARRAY("DSP Channel5 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dsp_channel5_mixer_controls), + + SOC_MIXER_ARRAY("AIF1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif1_mixer_controls), + SOC_MIXER_ARRAY("AIF2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif2_mixer_controls), + SOC_MIXER_ARRAY("AIF3 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_aif3_mixer_controls), + SOC_MIXER_ARRAY("DAC1 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac1_mixer_controls), + SOC_MIXER_ARRAY("DAC2 Mixer", SND_SOC_NOPM, 0, 0, + adau1373_dac2_mixer_controls), + + SND_SOC_DAPM_SUPPLY("DSP", ADAU1373_DIGEN, 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine B", ADAU1373_DIGEN, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Recording Engine A", ADAU1373_DIGEN, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine B", ADAU1373_DIGEN, 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Playback Engine A", ADAU1373_DIGEN, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL1", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("PLL2", SND_SOC_NOPM, 0, 0, adau1373_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("SYSCLK1", ADAU1373_CLK_SRC_DIV(0), 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("SYSCLK2", ADAU1373_CLK_SRC_DIV(1), 7, 0, NULL, 0), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), + SND_SOC_DAPM_INPUT("AIN4L"), + SND_SOC_DAPM_INPUT("AIN4R"), + + SND_SOC_DAPM_INPUT("DMIC1DAT"), + SND_SOC_DAPM_INPUT("DMIC2DAT"), + + SND_SOC_DAPM_OUTPUT("LOUT1L"), + SND_SOC_DAPM_OUTPUT("LOUT1R"), + SND_SOC_DAPM_OUTPUT("LOUT2L"), + SND_SOC_DAPM_OUTPUT("LOUT2R"), + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("SPKL"), + SND_SOC_DAPM_OUTPUT("SPKR"), + SND_SOC_DAPM_OUTPUT("EP"), +}; + +static int adau1373_check_aif_clk(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + const char *clk; + + dai = sink->name[3] - '1'; + + if (!adau1373->dais[dai].master) + return 0; + + if (adau1373->dais[dai].clk_src == ADAU1373_CLK_SRC_PLL1) + clk = "SYSCLK1"; + else + clk = "SYSCLK2"; + + return strcmp(source->name, clk) == 0; +} + +static int adau1373_check_src(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + unsigned int dai; + + dai = sink->name[3] - '1'; + + return adau1373->dais[dai].enable_src; +} + +#define DSP_CHANNEL_MIXER_ROUTES(_sink) \ + { _sink, "DMIC2 Swapped Switch", "DMIC2" }, \ + { _sink, "DMIC2 Switch", "DMIC2" }, \ + { _sink, "ADC/DMIC1 Swapped Switch", "Decimator Mux" }, \ + { _sink, "ADC/DMIC1 Switch", "Decimator Mux" }, \ + { _sink, "AIF1 Switch", "AIF1 IN" }, \ + { _sink, "AIF2 Switch", "AIF2 IN" }, \ + { _sink, "AIF3 Switch", "AIF3 IN" } + +#define DSP_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "DSP Channel1 Switch", "DSP Channel1 Mixer" }, \ + { _sink, "DSP Channel2 Switch", "DSP Channel2 Mixer" }, \ + { _sink, "DSP Channel3 Switch", "DSP Channel3 Mixer" }, \ + { _sink, "DSP Channel4 Switch", "DSP Channel4 Mixer" }, \ + { _sink, "DSP Channel5 Switch", "DSP Channel5 Mixer" } + +#define LEFT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +#define RIGHT_OUTPUT_MIXER_ROUTES(_sink) \ + { _sink, "Right DAC2 Switch", "Right DAC2" }, \ + { _sink, "Left DAC2 Switch", "Left DAC2" }, \ + { _sink, "Right DAC1 Switch", "Right DAC1" }, \ + { _sink, "Left DAC1 Switch", "Left DAC1" }, \ + { _sink, "Input 1 Bypass Switch", "IN1PGA" }, \ + { _sink, "Input 2 Bypass Switch", "IN2PGA" }, \ + { _sink, "Input 3 Bypass Switch", "IN3PGA" }, \ + { _sink, "Input 4 Bypass Switch", "IN4PGA" } + +static const struct snd_soc_dapm_route adau1373_dapm_routes[] = { + { "Left ADC Mixer", "DAC1 Switch", "Left DAC1" }, + { "Left ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Left ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Left ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Left ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Right ADC Mixer", "DAC1 Switch", "Right DAC1" }, + { "Right ADC Mixer", "Input 1 Switch", "IN1PGA" }, + { "Right ADC Mixer", "Input 2 Switch", "IN2PGA" }, + { "Right ADC Mixer", "Input 3 Switch", "IN3PGA" }, + { "Right ADC Mixer", "Input 4 Switch", "IN4PGA" }, + + { "Left ADC", NULL, "Left ADC Mixer" }, + { "Right ADC", NULL, "Right ADC Mixer" }, + + { "Decimator Mux", "ADC", "Left ADC" }, + { "Decimator Mux", "ADC", "Right ADC" }, + { "Decimator Mux", "DMIC1", "DMIC1" }, + + DSP_CHANNEL_MIXER_ROUTES("DSP Channel1 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel2 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel3 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel4 Mixer"), + DSP_CHANNEL_MIXER_ROUTES("DSP Channel5 Mixer"), + + DSP_OUTPUT_MIXER_ROUTES("AIF1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF2 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("AIF3 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC1 Mixer"), + DSP_OUTPUT_MIXER_ROUTES("DAC2 Mixer"), + + { "AIF1 OUT", NULL, "AIF1 Mixer" }, + { "AIF2 OUT", NULL, "AIF2 Mixer" }, + { "AIF3 OUT", NULL, "AIF3 Mixer" }, + { "Left DAC1", NULL, "DAC1 Mixer" }, + { "Right DAC1", NULL, "DAC1 Mixer" }, + { "Left DAC2", NULL, "DAC2 Mixer" }, + { "Right DAC2", NULL, "DAC2 Mixer" }, + + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout1 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout1 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Lineout2 Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Lineout2 Mixer"), + LEFT_OUTPUT_MIXER_ROUTES("Left Speaker Mixer"), + RIGHT_OUTPUT_MIXER_ROUTES("Right Speaker Mixer"), + + { "Left Headphone Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Left Headphone Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Left Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Left Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Left Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Left Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + { "Right Headphone Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Right Headphone Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Right Headphone Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Right Headphone Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Right Headphone Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Right Headphone Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "Left Headphone Mixer", NULL, "Headphone Enable" }, + { "Right Headphone Mixer", NULL, "Headphone Enable" }, + + { "Earpiece Mixer", "Right DAC2 Switch", "Right DAC2" }, + { "Earpiece Mixer", "Left DAC2 Switch", "Left DAC2" }, + { "Earpiece Mixer", "Right DAC1 Switch", "Right DAC1" }, + { "Earpiece Mixer", "Left DAC1 Switch", "Left DAC1" }, + { "Earpiece Mixer", "Input 1 Bypass Switch", "IN1PGA" }, + { "Earpiece Mixer", "Input 2 Bypass Switch", "IN2PGA" }, + { "Earpiece Mixer", "Input 3 Bypass Switch", "IN3PGA" }, + { "Earpiece Mixer", "Input 4 Bypass Switch", "IN4PGA" }, + + { "LOUT1L", NULL, "Left Lineout1 Mixer" }, + { "LOUT1R", NULL, "Right Lineout1 Mixer" }, + { "LOUT2L", NULL, "Left Lineout2 Mixer" }, + { "LOUT2R", NULL, "Right Lineout2 Mixer" }, + { "SPKL", NULL, "Left Speaker Mixer" }, + { "SPKR", NULL, "Right Speaker Mixer" }, + { "HPL", NULL, "Left Headphone Mixer" }, + { "HPR", NULL, "Right Headphone Mixer" }, + { "EP", NULL, "Earpiece Mixer" }, + + { "IN1PGA", NULL, "AIN1L" }, + { "IN2PGA", NULL, "AIN2L" }, + { "IN3PGA", NULL, "AIN3L" }, + { "IN4PGA", NULL, "AIN4L" }, + { "IN1PGA", NULL, "AIN1R" }, + { "IN2PGA", NULL, "AIN2R" }, + { "IN3PGA", NULL, "AIN3R" }, + { "IN4PGA", NULL, "AIN4R" }, + + { "SYSCLK1", NULL, "PLL1" }, + { "SYSCLK2", NULL, "PLL2" }, + + { "Left DAC1", NULL, "SYSCLK1" }, + { "Right DAC1", NULL, "SYSCLK1" }, + { "Left DAC2", NULL, "SYSCLK1" }, + { "Right DAC2", NULL, "SYSCLK1" }, + { "Left ADC", NULL, "SYSCLK1" }, + { "Right ADC", NULL, "SYSCLK1" }, + + { "DSP", NULL, "SYSCLK1" }, + + { "AIF1 Mixer", NULL, "DSP" }, + { "AIF2 Mixer", NULL, "DSP" }, + { "AIF3 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "DSP" }, + { "DAC2 Mixer", NULL, "DSP" }, + { "DAC1 Mixer", NULL, "Playback Engine A" }, + { "DAC2 Mixer", NULL, "Playback Engine B" }, + { "Left ADC Mixer", NULL, "Recording Engine A" }, + { "Right ADC Mixer", NULL, "Recording Engine A" }, + + { "AIF1 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK1", adau1373_check_aif_clk }, + { "AIF1 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF2 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + { "AIF3 CLK", NULL, "SYSCLK2", adau1373_check_aif_clk }, + + { "AIF1 IN", NULL, "AIF1 CLK" }, + { "AIF1 OUT", NULL, "AIF1 CLK" }, + { "AIF2 IN", NULL, "AIF2 CLK" }, + { "AIF2 OUT", NULL, "AIF2 CLK" }, + { "AIF3 IN", NULL, "AIF3 CLK" }, + { "AIF3 OUT", NULL, "AIF3 CLK" }, + { "AIF1 IN", NULL, "AIF1 IN SRC", adau1373_check_src }, + { "AIF1 OUT", NULL, "AIF1 OUT SRC", adau1373_check_src }, + { "AIF2 IN", NULL, "AIF2 IN SRC", adau1373_check_src }, + { "AIF2 OUT", NULL, "AIF2 OUT SRC", adau1373_check_src }, + { "AIF3 IN", NULL, "AIF3 IN SRC", adau1373_check_src }, + { "AIF3 OUT", NULL, "AIF3 OUT SRC", adau1373_check_src }, + + { "DMIC1", NULL, "DMIC1DAT" }, + { "DMIC1", NULL, "SYSCLK1" }, + { "DMIC1", NULL, "Recording Engine A" }, + { "DMIC2", NULL, "DMIC2DAT" }, + { "DMIC2", NULL, "SYSCLK1" }, + { "DMIC2", NULL, "Recording Engine B" }, +}; + +static int adau1373_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 adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int div; + unsigned int freq; + unsigned int ctrl; + + freq = adau1373_dai->sysclk; + + if (freq % params_rate(params) != 0) + return -EINVAL; + + switch (freq / params_rate(params)) { + case 1024: /* sysclk / 256 */ + div = 0; + break; + case 1536: /* 2/3 sysclk / 256 */ + div = 1; + break; + case 2048: /* 1/2 sysclk / 256 */ + div = 2; + break; + case 3072: /* 1/3 sysclk / 256 */ + div = 3; + break; + case 4096: /* 1/4 sysclk / 256 */ + div = 4; + break; + case 6144: /* 1/6 sysclk / 256 */ + div = 5; + break; + case 5632: /* 2/11 sysclk / 256 */ + div = 6; + break; + default: + return -EINVAL; + } + + adau1373_dai->enable_src = (div != 0); + + snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), + ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl = ADAU1373_DAI_WLEN_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + ctrl = ADAU1373_DAI_WLEN_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + ctrl = ADAU1373_DAI_WLEN_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + ctrl = ADAU1373_DAI_WLEN_32; + break; + default: + return -EINVAL; + } + + return snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ADAU1373_DAI_WLEN_MASK, ctrl); +} + +static int adau1373_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + unsigned int ctrl; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl = ADAU1373_DAI_MASTER; + adau1373_dai->master = true; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ctrl = 0; + adau1373_dai->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl |= ADAU1373_DAI_FORMAT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl |= ADAU1373_DAI_FORMAT_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl |= ADAU1373_DAI_FORMAT_RIGHT_J; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl |= ADAU1373_DAI_FORMAT_DSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl |= ADAU1373_DAI_INVERT_BCLK; + break; + case SND_SOC_DAIFMT_NB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK; + break; + case SND_SOC_DAIFMT_IB_IF: + ctrl |= ADAU1373_DAI_INVERT_LRCLK | ADAU1373_DAI_INVERT_BCLK; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAU1373_DAI(dai->id), + ~ADAU1373_DAI_WLEN_MASK, ctrl); + + return 0; +} + +static int adau1373_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct adau1373 *adau1373 = snd_soc_codec_get_drvdata(dai->codec); + struct adau1373_dai *adau1373_dai = &adau1373->dais[dai->id]; + + switch (clk_id) { + case ADAU1373_CLK_SRC_PLL1: + case ADAU1373_CLK_SRC_PLL2: + break; + default: + return -EINVAL; + } + + adau1373_dai->sysclk = freq; + adau1373_dai->clk_src = clk_id; + + snd_soc_update_bits(dai->codec, ADAU1373_BCLKDIV(dai->id), + ADAU1373_BCLKDIV_SOURCE, clk_id << 5); + + return 0; +} + +static const struct snd_soc_dai_ops adau1373_dai_ops = { + .hw_params = adau1373_hw_params, + .set_sysclk = adau1373_set_dai_sysclk, + .set_fmt = adau1373_set_dai_fmt, +}; + +#define ADAU1373_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver adau1373_dai_driver[] = { + { + .id = 0, + .name = "adau1373-aif1", + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 1, + .name = "adau1373-aif2", + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, + { + .id = 2, + .name = "adau1373-aif3", + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .formats = ADAU1373_FORMATS, + }, + .ops = &adau1373_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int adau1373_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + unsigned int dpll_div = 0; + unsigned int x, r, n, m, i, j, mode; + + switch (pll_id) { + case ADAU1373_PLL1: + case ADAU1373_PLL2: + break; + default: + return -EINVAL; + } + + switch (source) { + case ADAU1373_PLL_SRC_BCLK1: + case ADAU1373_PLL_SRC_BCLK2: + case ADAU1373_PLL_SRC_BCLK3: + case ADAU1373_PLL_SRC_LRCLK1: + case ADAU1373_PLL_SRC_LRCLK2: + case ADAU1373_PLL_SRC_LRCLK3: + case ADAU1373_PLL_SRC_MCLK1: + case ADAU1373_PLL_SRC_MCLK2: + case ADAU1373_PLL_SRC_GPIO1: + case ADAU1373_PLL_SRC_GPIO2: + case ADAU1373_PLL_SRC_GPIO3: + case ADAU1373_PLL_SRC_GPIO4: + break; + default: + return -EINVAL; + } + + if (freq_in < 7813 || freq_in > 27000000) + return -EINVAL; + + if (freq_out < 45158000 || freq_out > 49152000) + return -EINVAL; + + /* APLL input needs to be >= 8Mhz, so in case freq_in is less we use the + * DPLL to get it there. DPLL_out = (DPLL_in / div) * 1024 */ + while (freq_in < 8000000) { + freq_in *= 2; + dpll_div++; + } + + if (freq_out % freq_in != 0) { + /* fout = fin * (r + (n/m)) / x */ + x = DIV_ROUND_UP(freq_in, 13500000); + freq_in /= x; + r = freq_out / freq_in; + i = freq_out % freq_in; + j = gcd(i, freq_in); + n = i / j; + m = freq_in / j; + x--; + mode = 1; + } else { + /* fout = fin / r */ + r = freq_out / freq_in; + n = 0; + m = 0; + x = 0; + mode = 0; + } + + if (r < 2 || r > 8 || x > 3 || m > 0xffff || n > 0xffff) + return -EINVAL; + + if (dpll_div) { + dpll_div = 11 - dpll_div; + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, 0); + } else { + snd_soc_update_bits(codec, ADAU1373_PLL_CTRL6(pll_id), + ADAU1373_PLL_CTRL6_DPLL_BYPASS, + ADAU1373_PLL_CTRL6_DPLL_BYPASS); + } + + snd_soc_write(codec, ADAU1373_DPLL_CTRL(pll_id), + (source << 4) | dpll_div); + snd_soc_write(codec, ADAU1373_PLL_CTRL1(pll_id), (m >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL2(pll_id), m & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL3(pll_id), (n >> 8) & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL4(pll_id), n & 0xff); + snd_soc_write(codec, ADAU1373_PLL_CTRL5(pll_id), + (r << 3) | (x << 1) | mode); + + /* Set sysclk to pll_rate / 4 */ + snd_soc_update_bits(codec, ADAU1373_CLK_SRC_DIV(pll_id), 0x3f, 0x09); + + return 0; +} + +static void adau1373_load_drc_settings(struct snd_soc_codec *codec, + unsigned int nr, uint8_t *drc) +{ + unsigned int i; + + for (i = 0; i < ADAU1373_DRC_SIZE; ++i) + snd_soc_write(codec, ADAU1373_DRC(nr) + i, drc[i]); +} + +static bool adau1373_valid_micbias(enum adau1373_micbias_voltage micbias) +{ + switch (micbias) { + case ADAU1373_MICBIAS_2_9V: + case ADAU1373_MICBIAS_2_2V: + case ADAU1373_MICBIAS_2_6V: + case ADAU1373_MICBIAS_1_8V: + return true; + default: + break; + } + return false; +} + +static int adau1373_probe(struct snd_soc_codec *codec) +{ + struct adau1373_platform_data *pdata = codec->dev->platform_data; + bool lineout_differential = false; + unsigned int val; + int ret; + int i; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_I2C); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + codec->dapm.idle_bias_off = true; + + if (pdata) { + if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) + return -EINVAL; + + if (!adau1373_valid_micbias(pdata->micbias1) || + !adau1373_valid_micbias(pdata->micbias2)) + return -EINVAL; + + for (i = 0; i < pdata->num_drc; ++i) { + adau1373_load_drc_settings(codec, i, + pdata->drc_setting[i]); + } + + snd_soc_add_controls(codec, adau1373_drc_controls, + pdata->num_drc); + + val = 0; + for (i = 0; i < 4; ++i) { + if (pdata->input_differential[i]) + val |= BIT(i); + } + snd_soc_write(codec, ADAU1373_INPUT_MODE, val); + + val = 0; + if (pdata->lineout_differential) + val |= ADAU1373_OUTPUT_CTRL_LDIFF; + if (pdata->lineout_ground_sense) + val |= ADAU1373_OUTPUT_CTRL_LNFBEN; + snd_soc_write(codec, ADAU1373_OUTPUT_CTRL, val); + + lineout_differential = pdata->lineout_differential; + + snd_soc_write(codec, ADAU1373_EP_CTRL, + (pdata->micbias1 << ADAU1373_EP_CTRL_MICBIAS1_OFFSET) | + (pdata->micbias2 << ADAU1373_EP_CTRL_MICBIAS2_OFFSET)); + } + + if (!lineout_differential) { + snd_soc_add_controls(codec, adau1373_lineout2_controls, + ARRAY_SIZE(adau1373_lineout2_controls)); + } + + snd_soc_write(codec, ADAU1373_ADC_CTRL, + ADAU1373_ADC_CTRL_RESET_FORCE | ADAU1373_ADC_CTRL_PEAK_DETECT); + + return 0; +} + +static int adau1373_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, ADAU1373_PWDN_CTRL3_PWR_EN); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAU1373_PWDN_CTRL3, + ADAU1373_PWDN_CTRL3_PWR_EN, 0); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static int adau1373_remove(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int adau1373_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adau1373_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adau1373_resume(struct snd_soc_codec *codec) +{ + adau1373_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + snd_soc_cache_sync(codec); + + return 0; +} + +static struct snd_soc_codec_driver adau1373_codec_driver = { + .probe = adau1373_probe, + .remove = adau1373_remove, + .suspend = adau1373_suspend, + .resume = adau1373_resume, + .set_bias_level = adau1373_set_bias_level, + .reg_cache_size = ARRAY_SIZE(adau1373_default_regs), + .reg_cache_default = adau1373_default_regs, + .reg_word_size = sizeof(uint8_t), + + .set_pll = adau1373_set_pll, + + .controls = adau1373_controls, + .num_controls = ARRAY_SIZE(adau1373_controls), + .dapm_widgets = adau1373_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1373_dapm_widgets), + .dapm_routes = adau1373_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1373_dapm_routes), +}; + +static int __devinit adau1373_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct adau1373 *adau1373; + int ret; + + adau1373 = kzalloc(sizeof(*adau1373), GFP_KERNEL); + if (!adau1373) + return -ENOMEM; + + dev_set_drvdata(&client->dev, adau1373); + + ret = snd_soc_register_codec(&client->dev, &adau1373_codec_driver, + adau1373_dai_driver, ARRAY_SIZE(adau1373_dai_driver)); + if (ret < 0) + kfree(adau1373); + + return ret; +} + +static int __devexit adau1373_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + kfree(dev_get_drvdata(&client->dev)); + return 0; +} + +static const struct i2c_device_id adau1373_i2c_id[] = { + { "adau1373", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1373_i2c_id); + +static struct i2c_driver adau1373_i2c_driver = { + .driver = { + .name = "adau1373", + .owner = THIS_MODULE, + }, + .probe = adau1373_i2c_probe, + .remove = __devexit_p(adau1373_i2c_remove), + .id_table = adau1373_i2c_id, +}; + +static int __init adau1373_init(void) +{ + return i2c_add_driver(&adau1373_i2c_driver); +} +module_init(adau1373_init); + +static void __exit adau1373_exit(void) +{ + i2c_del_driver(&adau1373_i2c_driver); +} +module_exit(adau1373_exit); + +MODULE_DESCRIPTION("ASoC ADAU1373 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1373.h b/sound/soc/codecs/adau1373.h new file mode 100644 index 000000000000..c6ab5530760c --- /dev/null +++ b/sound/soc/codecs/adau1373.h @@ -0,0 +1,29 @@ +#ifndef __ADAU1373_H__ +#define __ADAU1373_H__ + +enum adau1373_pll_src { + ADAU1373_PLL_SRC_MCLK1 = 0, + ADAU1373_PLL_SRC_BCLK1 = 1, + ADAU1373_PLL_SRC_BCLK2 = 2, + ADAU1373_PLL_SRC_BCLK3 = 3, + ADAU1373_PLL_SRC_LRCLK1 = 4, + ADAU1373_PLL_SRC_LRCLK2 = 5, + ADAU1373_PLL_SRC_LRCLK3 = 6, + ADAU1373_PLL_SRC_GPIO1 = 7, + ADAU1373_PLL_SRC_GPIO2 = 8, + ADAU1373_PLL_SRC_GPIO3 = 9, + ADAU1373_PLL_SRC_GPIO4 = 10, + ADAU1373_PLL_SRC_MCLK2 = 11, +}; + +enum adau1373_pll { + ADAU1373_PLL1 = 0, + ADAU1373_PLL2 = 1, +}; + +enum adau1373_clk_src { + ADAU1373_CLK_SRC_PLL1 = 0, + ADAU1373_CLK_SRC_PLL2 = 1, +}; + +#endif -- cgit v1.2.3-58-ga151 From 33c5f969b969c277e96cd9e9bf8472c4b8709c25 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 22 Aug 2011 18:40:30 +0100 Subject: ASoC: Allow idle_bias_off to be specified in CODEC drivers If devices can unconditionally support idle_bias_off let them flag it in their driver structure. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 2 ++ 2 files changed, 3 insertions(+) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 3fe658eea28b..6da55a17fcfd 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -633,6 +633,7 @@ struct snd_soc_codec_driver { /* codec bias level */ int (*set_bias_level)(struct snd_soc_codec *, enum snd_soc_bias_level level); + bool idle_bias_off; void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ae93aa81244c..f8f985a4f2a8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -956,6 +956,8 @@ static int soc_probe_codec(struct snd_soc_card *card, snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, driver->num_dapm_widgets); + codec->dapm.idle_bias_off = driver->idle_bias_off; + if (driver->probe) { ret = driver->probe(codec); if (ret < 0) { -- cgit v1.2.3-58-ga151 From 4a8923ba99f559b078cf584f7caad901ada0e5be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Aug 2011 19:12:49 +0100 Subject: ASoC: Allow register defaults to be larger than unsigned short Devices that need this exist; obviously the newer regmap defaults mechanism will deal with this more happily. Signed-off-by: Mark Brown --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 6da55a17fcfd..0fc8f15f1aca 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -622,7 +622,7 @@ struct snd_soc_codec_driver { int (*volatile_register)(struct snd_soc_codec *, unsigned int); int (*readable_register)(struct snd_soc_codec *, unsigned int); int (*writable_register)(struct snd_soc_codec *, unsigned int); - short reg_cache_size; + unsigned int reg_cache_size; short reg_cache_step; short reg_word_size; const void *reg_cache_default; -- cgit v1.2.3-58-ga151 From da1c6ea6cf85544292c30295c70a89e8555358bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 24 Aug 2011 20:09:01 +0100 Subject: ASoC: Allow source specification for CODEC level sysclk Similarly to PLLs/FLLs some modern CODECs provide selectable system clock sources. When the clock is the clock for a DAI we do not usually need to identify which clock is being configured so can use clk_id for the source clock but with CODEC wide system clocks we will need to specify both the clock being configured and the source. Add a source argument to the CODEC driver set_sysclk() operation to reflect this. As this operation is not as widely used as the DAI set_sysclk() operation the change is not very invasive. We probably ought to go and make the same alternation for DAIs at some point. Signed-off-by: Mark Brown --- include/sound/soc.h | 4 ++-- sound/soc/codecs/adav80x.c | 3 ++- sound/soc/codecs/wm9081.c | 4 ++-- sound/soc/samsung/speyside.c | 2 +- sound/soc/soc-core.c | 8 +++++--- 5 files changed, 12 insertions(+), 9 deletions(-) (limited to 'include/sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index 0fc8f15f1aca..24e17be38c19 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -276,7 +276,7 @@ enum snd_soc_pcm_subclass { }; int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir); + int source, unsigned int freq, int dir); int snd_soc_codec_set_pll(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); @@ -610,7 +610,7 @@ struct snd_soc_codec_driver { /* codec wide operations */ int (*set_sysclk)(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir); + int clk_id, int source, unsigned int freq, int dir); int (*set_pll)(struct snd_soc_codec *codec, int pll_id, int source, unsigned int freq_in, unsigned int freq_out); diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index 300c04b70e71..f9f08948e5e8 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -523,7 +523,8 @@ static int adav80x_hw_params(struct snd_pcm_substream *substream, } static int adav80x_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) + int clk_id, int source, + unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index a4691321f9b3..f32ab1ee9647 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1120,8 +1120,8 @@ static int wm9081_digital_mute(struct snd_soc_dai *codec_dai, int mute) return 0; } -static int wm9081_set_sysclk(struct snd_soc_codec *codec, - int clk_id, unsigned int freq, int dir) +static int wm9081_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index bfed1ff7093f..09df8afbb447 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -223,7 +223,7 @@ static int speyside_wm9081_init(struct snd_soc_dapm_context *dapm) snd_soc_dapm_nc_pin(dapm, "LINEOUT"); /* At any time the WM9081 is active it will have this clock */ - return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, + return snd_soc_codec_set_sysclk(dapm->codec, WM9081_SYSCLK_MCLK, 0, 48000 * 256, 0); } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fc7fff3604f7..4ec93d1df047 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2670,7 +2670,7 @@ int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, if (dai->driver && dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); else if (dai->codec && dai->codec->driver->set_sysclk) - return dai->codec->driver->set_sysclk(dai->codec, clk_id, + return dai->codec->driver->set_sysclk(dai->codec, clk_id, 0, freq, dir); else return -EINVAL; @@ -2681,16 +2681,18 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); * snd_soc_codec_set_sysclk - configure CODEC system or master clock. * @codec: CODEC * @clk_id: DAI specific clock ID + * @source: Source for the clock * @freq: new clock frequency in Hz * @dir: new clock direction - input/output. * * Configures the CODEC master (MCLK) or system (SYSCLK) clocking. */ int snd_soc_codec_set_sysclk(struct snd_soc_codec *codec, int clk_id, - unsigned int freq, int dir) + int source, unsigned int freq, int dir) { if (codec->driver->set_sysclk) - return codec->driver->set_sysclk(codec, clk_id, freq, dir); + return codec->driver->set_sysclk(codec, clk_id, source, + freq, dir); else return -EINVAL; } -- cgit v1.2.3-58-ga151 From 76067540c642b1a14679ab74bd027a074c23e63b Mon Sep 17 00:00:00 2001 From: Dong Aisheng Date: Wed, 7 Sep 2011 20:51:50 +0800 Subject: ASoC: mxs-saif: add record function 1. add different clkmux mode handling SAIF can use two instances to implement full duplex (playback & recording) and record saif may work on EXTMASTER mode which is using other saif's BITCLK&LRCLK. The clkmux mode could be set in pdata->init() in mach-specific code. For generic saif driver, it only needs to know who is his master and the master id is also provided in mach-specific code. 2. support playback and capture simutaneously however the sample rates can not be different due to hw limitation. Signed-off-by: Dong Aisheng Acked-by: Wolfram Sang Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- include/sound/saif.h | 16 ++++++ sound/soc/mxs/mxs-saif.c | 145 ++++++++++++++++++++++++++++++++++++++++++----- sound/soc/mxs/mxs-saif.h | 4 ++ 3 files changed, 151 insertions(+), 14 deletions(-) create mode 100644 include/sound/saif.h (limited to 'include/sound') diff --git a/include/sound/saif.h b/include/sound/saif.h new file mode 100644 index 000000000000..d0e0de7984ec --- /dev/null +++ b/include/sound/saif.h @@ -0,0 +1,16 @@ +/* + * Copyright 2011 Freescale Semiconductor, Inc. All Rights Reserved. + * + * 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 __SOUND_SAIF_H__ +#define __SOUND_SAIF_H__ + +struct mxs_saif_platform_data { + int (*init) (void); + int (*get_master_id) (unsigned int saif_id); +}; +#endif diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index af5734f6dab7..401944cf4560 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -23,10 +23,12 @@ #include #include #include +#include #include #include #include #include +#include #include #include #include @@ -36,6 +38,24 @@ static struct mxs_saif *mxs_saif[2]; +/* + * SAIF is a little different with other normal SOC DAIs on clock using. + * + * For MXS, two SAIF modules are instantiated on-chip. + * Each SAIF has a set of clock pins and can be operating in master + * mode simultaneously if they are connected to different off-chip codecs. + * Also, one of the two SAIFs can master or drive the clock pins while the + * other SAIF, in slave mode, receives clocking from the master SAIF. + * This also means that both SAIFs must operate at the same sample rate. + * + * We abstract this as each saif has a master, the master could be + * himself or other saifs. In the generic saif driver, saif does not need + * to know the different clkmux. Saif only needs to know who is his master + * and operating his master to generate the proper clock rate for him. + * The master id is provided in mach-specific layer according to different + * clkmux setting. + */ + static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { @@ -51,6 +71,17 @@ static int mxs_saif_set_dai_sysclk(struct snd_soc_dai *cpu_dai, return 0; } +/* + * Since SAIF may work on EXTMASTER mode, IOW, it's working BITCLK&LRCLK + * is provided by other SAIF, we provide a interface here to get its master + * from its master_id. + * Note that the master could be himself. + */ +static inline struct mxs_saif *mxs_saif_get_master(struct mxs_saif * saif) +{ + return mxs_saif[saif->master_id]; +} + /* * Set SAIF clock and MCLK */ @@ -60,8 +91,26 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, { u32 scr; int ret; + struct mxs_saif *master_saif; - scr = __raw_readl(saif->base + SAIF_CTRL); + dev_dbg(saif->dev, "mclk %d rate %d\n", mclk, rate); + + /* Set master saif to generate proper clock */ + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; + + dev_dbg(saif->dev, "master saif%d\n", master_saif->id); + + /* Checking if can playback and capture simutaneously */ + if (master_saif->ongoing && rate != master_saif->cur_rate) { + dev_err(saif->dev, + "can not change clock, master saif%d(rate %d) is ongoing\n", + master_saif->id, master_saif->cur_rate); + return -EINVAL; + } + + scr = __raw_readl(master_saif->base + SAIF_CTRL); scr &= ~BM_SAIF_CTRL_BITCLK_MULT_RATE; scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; @@ -75,27 +124,29 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, * * If MCLK is not used, we just set saif clk to 512*fs. */ - if (saif->mclk_in_use) { + if (master_saif->mclk_in_use) { if (mclk % 32 == 0) { scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); } else if (mclk % 48 == 0) { scr |= BM_SAIF_CTRL_BITCLK_BASE_RATE; - ret = clk_set_rate(saif->clk, 384 * rate); + ret = clk_set_rate(master_saif->clk, 384 * rate); } else { /* SAIF MCLK should be either 32x or 48x */ return -EINVAL; } } else { - ret = clk_set_rate(saif->clk, 512 * rate); + ret = clk_set_rate(master_saif->clk, 512 * rate); scr &= ~BM_SAIF_CTRL_BITCLK_BASE_RATE; } if (ret) return ret; - if (!saif->mclk_in_use) { - __raw_writel(scr, saif->base + SAIF_CTRL); + master_saif->cur_rate = rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -137,7 +188,7 @@ static int mxs_saif_set_clk(struct mxs_saif *saif, return -EINVAL; } - __raw_writel(scr, saif->base + SAIF_CTRL); + __raw_writel(scr, master_saif->base + SAIF_CTRL); return 0; } @@ -183,6 +234,7 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, struct mxs_saif *saif = mxs_saif[saif_id]; u32 stat; int ret; + struct mxs_saif *master_saif; if (!saif) return -EINVAL; @@ -195,6 +247,12 @@ int mxs_saif_get_mclk(unsigned int saif_id, unsigned int mclk, __raw_writel(BM_SAIF_CTRL_CLKGATE, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + master_saif = mxs_saif_get_master(saif); + if (saif != master_saif) { + dev_err(saif->dev, "can not get mclk from a non-master saif\n"); + return -EINVAL; + } + stat = __raw_readl(saif->base + SAIF_STAT); if (stat & BM_SAIF_STAT_BUSY) { dev_err(saif->dev, "error: busy\n"); @@ -278,10 +336,17 @@ static int mxs_saif_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) /* * Note: We simply just support master mode since SAIF TX can only * work as master. + * Here the master is relative to codec side. + * Saif internally could be slave when working on EXTMASTER mode. + * We just hide this to machine driver. */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + if (saif->id == saif->master_id) + scr &= ~BM_SAIF_CTRL_SLAVE_MODE; + else + scr |= BM_SAIF_CTRL_SLAVE_MODE; + __raw_writel(scr | scr0, saif->base + SAIF_CTRL); break; default: @@ -396,6 +461,12 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct mxs_saif *saif = snd_soc_dai_get_drvdata(cpu_dai); + struct mxs_saif *master_saif; + u32 delay; + + master_saif = mxs_saif_get_master(saif); + if (!master_saif) + return -EINVAL; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -403,10 +474,20 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: dev_dbg(cpu_dai->dev, "start\n"); - clk_enable(saif->clk); - if (!saif->mclk_in_use) + clk_enable(master_saif->clk); + if (!master_saif->mclk_in_use) + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_SET_ADDR); + + /* + * If the saif's master is not himself, we also need to enable + * itself clk for its internal basic logic to work. + */ + if (saif != master_saif) { + clk_enable(saif->clk); __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_SET_ADDR); + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { /* @@ -422,20 +503,39 @@ static int mxs_saif_trigger(struct snd_pcm_substream *substream, int cmd, __raw_readl(saif->base + SAIF_DATA); } - dev_dbg(cpu_dai->dev, "CTRL 0x%x STAT 0x%x\n", + master_saif->ongoing = 1; + + dev_dbg(saif->dev, "CTRL 0x%x STAT 0x%x\n", __raw_readl(saif->base + SAIF_CTRL), __raw_readl(saif->base + SAIF_STAT)); + dev_dbg(master_saif->dev, "CTRL 0x%x STAT 0x%x\n", + __raw_readl(master_saif->base + SAIF_CTRL), + __raw_readl(master_saif->base + SAIF_STAT)); break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: dev_dbg(cpu_dai->dev, "stop\n"); - clk_disable(saif->clk); - if (!saif->mclk_in_use) + /* wait a while for the current sample to complete */ + delay = USEC_PER_SEC / master_saif->cur_rate; + + if (!master_saif->mclk_in_use) { + __raw_writel(BM_SAIF_CTRL_RUN, + master_saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + } + clk_disable(master_saif->clk); + + if (saif != master_saif) { __raw_writel(BM_SAIF_CTRL_RUN, saif->base + SAIF_CTRL + MXS_CLR_ADDR); + udelay(delay); + clk_disable(saif->clk); + } + + master_saif->ongoing = 0; break; default: @@ -519,16 +619,33 @@ static int mxs_saif_probe(struct platform_device *pdev) { struct resource *res; struct mxs_saif *saif; + struct mxs_saif_platform_data *pdata; int ret = 0; if (pdev->id >= ARRAY_SIZE(mxs_saif)) return -EINVAL; + pdata = pdev->dev.platform_data; + if (pdata && pdata->init) { + ret = pdata->init(); + if (ret) + return ret; + } + saif = kzalloc(sizeof(*saif), GFP_KERNEL); if (!saif) return -ENOMEM; mxs_saif[pdev->id] = saif; + saif->id = pdev->id; + + saif->master_id = saif->id; + if (pdata && pdata->get_master_id) { + saif->master_id = pdata->get_master_id(saif->id); + if (saif->master_id < 0 || + saif->master_id >= ARRAY_SIZE(mxs_saif)) + return -EINVAL; + } saif->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { diff --git a/sound/soc/mxs/mxs-saif.h b/sound/soc/mxs/mxs-saif.h index 0e2ff8cdbfee..12c91e4eb941 100644 --- a/sound/soc/mxs/mxs-saif.h +++ b/sound/soc/mxs/mxs-saif.h @@ -118,6 +118,10 @@ struct mxs_saif { void __iomem *base; int irq; struct mxs_pcm_dma_params dma_param; + unsigned int id; + unsigned int master_id; + unsigned int cur_rate; + unsigned int ongoing; struct platform_device *soc_platform_pdev; u32 fifo_underrun; -- cgit v1.2.3-58-ga151 From 88e24c3a4b30a6bd361f2b5ce602667a8161b2e8 Mon Sep 17 00:00:00 2001 From: Yong Zhang Date: Thu, 22 Sep 2011 16:59:20 +0800 Subject: sound: irq: Remove IRQF_DISABLED Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled], We run all interrupt handlers with interrupts disabled and we even check and yell when an interrupt handler returns with interrupts enabled (see commit [b738a50a: genirq: Warn when handler enables interrupts]). So now this flag is a NOOP and can be removed. Signed-off-by: Yong Zhang Acked-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- include/sound/initval.h | 2 +- sound/arm/aaci.c | 2 +- sound/arm/pxa2xx-ac97-lib.c | 2 +- sound/drivers/ml403-ac97cr.c | 4 ++-- sound/drivers/mpu401/mpu401_uart.c | 2 +- sound/drivers/mtpav.c | 2 +- sound/drivers/serial-u16550.c | 2 +- sound/isa/ad1816a/ad1816a_lib.c | 2 +- sound/isa/es1688/es1688_lib.c | 2 +- sound/isa/es18xx.c | 2 +- sound/isa/gus/gus_main.c | 2 +- sound/isa/gus/gusmax.c | 2 +- sound/isa/gus/interwave.c | 2 +- sound/isa/opl3sa2.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 2 +- sound/isa/sb/sb_common.c | 2 +- sound/isa/wavefront/wavefront.c | 2 +- sound/isa/wss/wss_lib.c | 2 +- sound/mips/au1x00.c | 4 ++-- sound/pci/sis7019.c | 4 ++-- sound/ppc/snd_ps3.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/nuc900/nuc900-pcm.c | 2 +- sound/soc/samsung/ac97.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/txx9/txx9aclc-ac97.c | 2 +- sound/sparc/amd7930.c | 2 +- 28 files changed, 31 insertions(+), 31 deletions(-) (limited to 'include/sound') diff --git a/include/sound/initval.h b/include/sound/initval.h index 1daa6dff8297..f99a0d2ddfe7 100644 --- a/include/sound/initval.h +++ b/include/sound/initval.h @@ -62,7 +62,7 @@ static int snd_legacy_find_free_irq(int *irq_table) { while (*irq_table != -1) { if (!request_irq(*irq_table, snd_legacy_empty_irq_handler, - IRQF_DISABLED | IRQF_PROBE_SHARED, "ALSA Test IRQ", + IRQF_PROBE_SHARED, "ALSA Test IRQ", (void *) irq_table)) { free_irq(*irq_table, (void *) irq_table); return *irq_table; diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index d0cead38d5fb..e518d38b1c74 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -443,7 +443,7 @@ static int aaci_pcm_open(struct snd_pcm_substream *substream) mutex_lock(&aaci->irq_lock); if (!aaci->users++) { ret = request_irq(aaci->dev->irq[0], aaci_irq, - IRQF_SHARED | IRQF_DISABLED, DRIVER_NAME, aaci); + IRQF_SHARED, DRIVER_NAME, aaci); if (ret != 0) aaci->users--; } diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 88eec3847df2..8ad65352bf91 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -359,7 +359,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev) if (ret) goto err_clk2; - ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, IRQF_DISABLED, "AC97", NULL); + ret = request_irq(IRQ_AC97, pxa2xx_ac97_irq, 0, "AC97", NULL); if (ret < 0) goto err_irq; diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 5cfcb908c430..2c7a7636f472 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -1153,7 +1153,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, "0x%x done\n", (unsigned int)ml403_ac97cr->port); /* get irq */ irq = platform_get_irq(pfdev, 0); - if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + if (request_irq(irq, snd_ml403_ac97cr_irq, 0, dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", @@ -1166,7 +1166,7 @@ snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, "request (playback) irq %d done\n", ml403_ac97cr->irq); irq = platform_get_irq(pfdev, 1); - if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + if (request_irq(irq, snd_ml403_ac97cr_irq, 0, dev_name(&pfdev->dev), (void *)ml403_ac97cr)) { snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " "unable to grab IRQ %d\n", diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c index 9d01c181feca..e91698a634b2 100644 --- a/sound/drivers/mpu401/mpu401_uart.c +++ b/sound/drivers/mpu401/mpu401_uart.c @@ -577,7 +577,7 @@ int snd_mpu401_uart_new(struct snd_card *card, int device, else mpu->cport = port + 1; if (irq >= 0) { - if (request_irq(irq, snd_mpu401_uart_interrupt, IRQF_DISABLED, + if (request_irq(irq, snd_mpu401_uart_interrupt, 0, "MPU401 UART", (void *) mpu)) { snd_printk(KERN_ERR "mpu401_uart: " "unable to grab IRQ %d\n", irq); diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 5c426df87678..1eef4ccebe4b 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -589,7 +589,7 @@ static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard) return -EBUSY; } mcard->port = port; - if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) { + if (request_irq(irq, snd_mtpav_irqh, 0, "MOTU MTPAV", mcard)) { snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); return -EBUSY; } diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index a25fb7b1f441..fc1d822802c3 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -816,7 +816,7 @@ static int __devinit snd_uart16550_create(struct snd_card *card, if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (request_irq(irq, snd_uart16550_interrupt, - IRQF_DISABLED, "Serial MIDI", uart)) { + 0, "Serial MIDI", uart)) { snd_printk(KERN_WARNING "irq %d busy. Using Polling.\n", irq); } else { diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 05aef8b97e96..177eed3271bc 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -595,7 +595,7 @@ int __devinit snd_ad1816a_create(struct snd_card *card, snd_ad1816a_free(chip); return -EBUSY; } - if (request_irq(irq, snd_ad1816a_interrupt, IRQF_DISABLED, "AD1816A", (void *) chip)) { + if (request_irq(irq, snd_ad1816a_interrupt, 0, "AD1816A", (void *) chip)) { snd_printk(KERN_ERR "ad1816a: can't grab IRQ %d\n", irq); snd_ad1816a_free(chip); return -EBUSY; diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 07676200496a..d3eab6fb0866 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -661,7 +661,7 @@ int snd_es1688_create(struct snd_card *card, snd_printk(KERN_ERR "es1688: can't grab port 0x%lx\n", port + 4); return -EBUSY; } - if (request_irq(irq, snd_es1688_interrupt, IRQF_DISABLED, "ES1688", (void *) chip)) { + if (request_irq(irq, snd_es1688_interrupt, 0, "ES1688", (void *) chip)) { snd_printk(KERN_ERR "es1688: can't grab IRQ %d\n", irq); return -EBUSY; } diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index aeee8f8bf5e9..bf6ad0bf51c6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -1805,7 +1805,7 @@ static int __devinit snd_es18xx_new_device(struct snd_card *card, return -EBUSY; } - if (request_irq(irq, snd_es18xx_interrupt, IRQF_DISABLED, "ES18xx", + if (request_irq(irq, snd_es18xx_interrupt, 0, "ES18xx", (void *) card)) { snd_es18xx_free(card); snd_printk(KERN_ERR PFX "unable to grap IRQ %d\n", irq); diff --git a/sound/isa/gus/gus_main.c b/sound/isa/gus/gus_main.c index 12eb98f2f931..3167e5ac3699 100644 --- a/sound/isa/gus/gus_main.c +++ b/sound/isa/gus/gus_main.c @@ -180,7 +180,7 @@ int snd_gus_create(struct snd_card *card, snd_gus_free(gus); return -EBUSY; } - if (irq >= 0 && request_irq(irq, snd_gus_interrupt, IRQF_DISABLED, "GUS GF1", (void *) gus)) { + if (irq >= 0 && request_irq(irq, snd_gus_interrupt, 0, "GUS GF1", (void *) gus)) { snd_printk(KERN_ERR "gus: can't grab irq %d\n", irq); snd_gus_free(gus); return -EBUSY; diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c index 3e4a58b72913..c43faa057ff6 100644 --- a/sound/isa/gus/gusmax.c +++ b/sound/isa/gus/gusmax.c @@ -291,7 +291,7 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev) goto _err; } - if (request_irq(xirq, snd_gusmax_interrupt, IRQF_DISABLED, "GUS MAX", (void *)maxcard)) { + if (request_irq(xirq, snd_gusmax_interrupt, 0, "GUS MAX", (void *)maxcard)) { snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); err = -EBUSY; goto _err; diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index c7b80e4730fc..5f869a32b48c 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -684,7 +684,7 @@ static int __devinit snd_interwave_probe(struct snd_card *card, int dev) if ((err = snd_gus_initialize(gus)) < 0) return err; - if (request_irq(xirq, snd_interwave_interrupt, IRQF_DISABLED, + if (request_irq(xirq, snd_interwave_interrupt, 0, "InterWave", iwcard)) { snd_printk(KERN_ERR PFX "unable to grab IRQ %d\n", xirq); return -EBUSY; diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index de99f47770bf..bbafb0b543ea 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -667,7 +667,7 @@ static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev) err = snd_opl3sa2_detect(card); if (err < 0) return err; - err = request_irq(xirq, snd_opl3sa2_interrupt, IRQF_DISABLED, + err = request_irq(xirq, snd_opl3sa2_interrupt, 0, "OPL3-SA2", card); if (err) { snd_printk(KERN_ERR PFX "can't grab IRQ %d\n", xirq); diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 346e12baa98e..6dbbfa76b440 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -892,7 +892,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) #endif #ifdef OPTi93X error = request_irq(irq, snd_opti93x_interrupt, - IRQF_DISABLED, DEV_NAME" - WSS", chip); + 0, DEV_NAME" - WSS", chip); if (error < 0) { snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", irq); return error; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index eae6c1c0eff9..d2e19215813e 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -240,7 +240,7 @@ int snd_sbdsp_create(struct snd_card *card, if (request_irq(irq, irq_handler, (hardware == SB_HW_ALS4000 || hardware == SB_HW_CS5530) ? - IRQF_SHARED : IRQF_DISABLED, + IRQF_SHARED : 0, "SoundBlaster", (void *) chip)) { snd_printk(KERN_ERR "sb: can't grab irq %d\n", irq); snd_sbdsp_free(chip); diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 83f291d89a95..87142977335a 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -418,7 +418,7 @@ snd_wavefront_probe (struct snd_card *card, int dev) return -EBUSY; } if (request_irq(ics2115_irq[dev], snd_wavefront_ics2115_interrupt, - IRQF_DISABLED, "ICS2115", acard)) { + 0, "ICS2115", acard)) { snd_printk(KERN_ERR "unable to use ICS2115 IRQ %d\n", ics2115_irq[dev]); return -EBUSY; } diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 2a42cc377957..7277c5b7df6c 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -1833,7 +1833,7 @@ int snd_wss_create(struct snd_card *card, } chip->cport = cport; if (!(hwshare & WSS_HWSHARE_IRQ)) - if (request_irq(irq, snd_wss_interrupt, IRQF_DISABLED, + if (request_irq(irq, snd_wss_interrupt, 0, "WSS", (void *) chip)) { snd_printk(KERN_ERR "wss: can't grab IRQ %d\n", irq); snd_wss_free(chip); diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 446cf9748664..7567ebd71913 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -465,13 +465,13 @@ snd_au1000_pcm_new(struct snd_au1000 *au1000) flags = claim_dma_lock(); if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX, - "AC97 TX", au1000_dma_interrupt, IRQF_DISABLED, + "AC97 TX", au1000_dma_interrupt, 0, au1000->stream[PLAYBACK])) < 0) { release_dma_lock(flags); return -EBUSY; } if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX, - "AC97 RX", au1000_dma_interrupt, IRQF_DISABLED, + "AC97 RX", au1000_dma_interrupt, 0, au1000->stream[CAPTURE])) < 0){ release_dma_lock(flags); return -EBUSY; diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index bcf61524a13f..5ffb20b18786 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1234,7 +1234,7 @@ static int sis_resume(struct pci_dev *pci) goto error; } - if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); goto error; @@ -1340,7 +1340,7 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out_cleanup; - if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + if (request_irq(pci->irq, sis_interrupt, IRQF_SHARED, KBUILD_MODNAME, sis)) { printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); goto error_out_cleanup; diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index bc823a547550..775bd95d4be6 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -845,7 +845,7 @@ static int __devinit snd_ps3_allocate_irq(void) return ret; } - ret = request_irq(the_card.irq_no, snd_ps3_interrupt, IRQF_DISABLED, + ret = request_irq(the_card.irq_no, snd_ps3_interrupt, 0, SND_PS3_DRIVER_NAME, &the_card); if (ret) { pr_info("%s: request_irq failed (%d)\n", __func__, ret); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 7aa5b7606777..177f7137a9c8 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -211,7 +211,7 @@ static int alchemy_pcm_open(struct snd_pcm_substream *substream) /* DMA setup */ name = (s == SNDRV_PCM_STREAM_PLAYBACK) ? "audio-tx" : "audio-rx"; ctx->stream[s].dma = request_au1000_dma(dmaids[s], name, - au1000_dma_interrupt, IRQF_DISABLED, + au1000_dma_interrupt, 0, &ctx->stream[s]); set_dma_mode(ctx->stream[s].dma, get_dma_mode(ctx->stream[s].dma) & ~DMA_NC); diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index faa5e9fb1471..243d17711211 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1431,7 +1431,7 @@ static int dac33_soc_probe(struct snd_soc_codec *codec) /* Check if the IRQ number is valid and request it */ if (dac33->irq >= 0) { ret = request_irq(dac33->irq, dac33_interrupt_handler, - IRQF_TRIGGER_RISING | IRQF_DISABLED, + IRQF_TRIGGER_RISING, codec->name, codec); if (ret < 0) { dev_err(codec->dev, "Could not request IRQ%d (%d)\n", diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index e46d5516e000..865b288bd748 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -268,7 +268,7 @@ static int nuc900_dma_open(struct snd_pcm_substream *substream) nuc900_audio = nuc900_ac97_data; if (request_irq(nuc900_audio->irq_num, nuc900_dma_interrupt, - IRQF_DISABLED, "nuc900-dma", substream)) + 0, "nuc900-dma", substream)) return -EBUSY; runtime->private_data = nuc900_audio; diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index f97110e72e85..884c8a107bf9 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -444,7 +444,7 @@ static __devinit int s3c_ac97_probe(struct platform_device *pdev) } ret = request_irq(irq_res->start, s3c_ac97_irq, - IRQF_DISABLED, "AC97", NULL); + 0, "AC97", NULL); if (ret < 0) { dev_err(&pdev->dev, "ac97: interrupt request failed.\n"); goto err4; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 8e112ccffb13..1493ebf4d943 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1285,7 +1285,7 @@ static int fsi_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); dev_set_drvdata(&pdev->dev, master); - ret = request_irq(irq, &fsi_interrupt, IRQF_DISABLED, + ret = request_irq(irq, &fsi_interrupt, 0, id_entry->name, master); if (ret) { dev_err(&pdev->dev, "irq request err\n"); diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 743d07b82c06..a4e3f5501847 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -201,7 +201,7 @@ static int __devinit txx9aclc_ac97_dev_probe(struct platform_device *pdev) if (!drvdata->base) return -EBUSY; err = devm_request_irq(&pdev->dev, irq, txx9aclc_ac97_irq, - IRQF_DISABLED, dev_name(&pdev->dev), drvdata); + 0, dev_name(&pdev->dev), drvdata); if (err < 0) return err; diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index ad7d4d7d9237..f036776380b5 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -962,7 +962,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, amd7930_idle(amd); if (request_irq(irq, snd_amd7930_interrupt, - IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) { + IRQF_SHARED, "amd7930", amd)) { snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", dev, irq); snd_amd7930_free(amd); -- cgit v1.2.3-58-ga151