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kernel_optimize_test/sound/soc/codecs/adau1373.c

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ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
/*
* Analog Devices ADAU1373 Audio Codec drive
*
* Copyright 2011 Analog Devices Inc.
* Author: Lars-Peter Clausen <lars@metafoo.de>
*
* Licensed under the GPL-2 or later.
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/slab.h>
#include <linux/gcd.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/tlv.h>
#include <sound/soc.h>
#include <sound/adau1373.h>
#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[] = {
ASoC: adau1373: fix DB_RANGE size Give the correct number of entries to TLV_DB_RANGE_HEAD to prevent reading more data than actually is in the array. Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2011-11-20 22:10:27 +08:00
TLV_DB_RANGE_HEAD(3),
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
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;
ASoC: Fix setting adau1373_dai->master for SND_SOC_DAIFMT_CBS_CFS In the case of SND_SOC_DAIFMT_CBS_CFS, adau1373_dai->master should be false. Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-09-28 15:14:56 +08:00
adau1373_dai->master = false;
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
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;
}
ASoC: Cleanup duplicated const Commit 85e7652("ASoC: Constify snd_soc_dai_ops structs") accidentally introduced a few duplicated consts. This patch cleans it up. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-11-23 21:11:21 +08:00
static const struct snd_soc_dai_ops adau1373_dai_ops = {
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
.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;
}
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]);
}
ASoC: core: Add support for DAI and machine kcontrols. Currently ASoC can only add kcontrols using codec and platform component device handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for SoC card machine drivers too. This allows the kcontrol to have a direct handle to the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily get it's private data. This change makes snd_soc_add_controls() static and wraps it in the folowing calls (card and dai are new) :- snd_soc_add_card_controls() snd_soc_add_codec_controls() snd_soc_add_dai_controls() snd_soc_add_platform_controls() This patch also does a lot of small mechanical changes in individual codec drivers to replace snd_soc_add_controls() with snd_soc_add_codec_controls(). It also updates the McBSP DAI driver to use snd_soc_add_dai_controls(). Finally, it updates the existing machine drivers that register controls to either :- 1) Use snd_soc_add_card_controls() where no direct codec control is required. 2) Use snd_soc_add_codec_controls() where there is direct codec control. In the case of 1) above we also update the machine drivers to get the correct component data pointers from the kcontrol (rather than getting the machine pointer via the codec pointer). Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-02-04 01:43:09 +08:00
snd_soc_add_codec_controls(codec, adau1373_drc_controls,
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
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) {
ASoC: core: Add support for DAI and machine kcontrols. Currently ASoC can only add kcontrols using codec and platform component device handles. It's also desirable to add kcontrols for DAIs (i.e. McBSP) and for SoC card machine drivers too. This allows the kcontrol to have a direct handle to the parent ASoC component DAI/SoC Card/Platform/Codec device and hence easily get it's private data. This change makes snd_soc_add_controls() static and wraps it in the folowing calls (card and dai are new) :- snd_soc_add_card_controls() snd_soc_add_codec_controls() snd_soc_add_dai_controls() snd_soc_add_platform_controls() This patch also does a lot of small mechanical changes in individual codec drivers to replace snd_soc_add_controls() with snd_soc_add_codec_controls(). It also updates the McBSP DAI driver to use snd_soc_add_dai_controls(). Finally, it updates the existing machine drivers that register controls to either :- 1) Use snd_soc_add_card_controls() where no direct codec control is required. 2) Use snd_soc_add_codec_controls() where there is direct codec control. In the case of 1) above we also update the machine drivers to get the correct component data pointers from the kcontrol (rather than getting the machine pointer via the codec pointer). Signed-off-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-02-04 01:43:09 +08:00
snd_soc_add_codec_controls(codec, adau1373_lineout2_controls,
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
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;
}
ASoC: Drop unused state parameter from CODEC suspend callback The existence of this parameter is purely historical. None of the CODEC drivers uses it and we always pass in the same value anyway, so it should be safe to remove it. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-12-02 17:18:28 +08:00
static int adau1373_suspend(struct snd_soc_codec *codec)
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
{
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,
ASoC: Set idle_bias_off flag in snd_soc_codec_driver Since commit 33c5f969 "ASoC: Allow idle_bias_off to be specified in CODEC drivers", now we can set idle_bias_off flag in struct snd_soc_codec_driver for devices can unconditionally support idle_bias_off. Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-01-27 18:02:09 +08:00
.idle_bias_off = true,
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
.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;
ASoC: Convert adau1373 to devm_kzalloc() Signed-off-by: Axel Lin <axel.lin@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-12-26 20:52:13 +08:00
adau1373 = devm_kzalloc(&client->dev, sizeof(*adau1373), GFP_KERNEL);
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
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));
return ret;
}
static int __devexit adau1373_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&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,
};
ASoC: adau1373: Use module_i2c_driver module_i2c_driver makes the code simpler by eliminating module_init and module_exit calls. Signed-off-by: Sachin Kamat <sachin.kamat@linaro.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-08-06 19:55:58 +08:00
module_i2c_driver(adau1373_i2c_driver);
ASoC: Add ADAU1373 codec support This patch adds support for the Analog Devices ADAU1373 audio codec. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-08-16 02:15:22 +08:00
MODULE_DESCRIPTION("ASoC ADAU1373 driver");
MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_LICENSE("GPL");
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