linux/sound/soc/codecs/wm_hubs.c

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/*
* wm_hubs.c -- WM8993/4 common code
*
* Copyright 2009 Wolfson Microelectronics plc
*
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "wm8993.h"
#include "wm_hubs.h"
const DECLARE_TLV_DB_SCALE(wm_hubs_spkmix_tlv, -300, 300, 0);
EXPORT_SYMBOL_GPL(wm_hubs_spkmix_tlv);
static const DECLARE_TLV_DB_SCALE(inpga_tlv, -1650, 150, 0);
static const DECLARE_TLV_DB_SCALE(inmix_sw_tlv, 0, 3000, 0);
static const DECLARE_TLV_DB_SCALE(inmix_tlv, -1500, 300, 1);
static const DECLARE_TLV_DB_SCALE(earpiece_tlv, -600, 600, 0);
static const DECLARE_TLV_DB_SCALE(outmix_tlv, -2100, 300, 0);
static const DECLARE_TLV_DB_SCALE(spkmixout_tlv, -1800, 600, 1);
static const DECLARE_TLV_DB_SCALE(outpga_tlv, -5700, 100, 0);
static const unsigned int spkboost_tlv[] = {
TLV_DB_RANGE_HEAD(7),
0, 6, TLV_DB_SCALE_ITEM(0, 150, 0),
7, 7, TLV_DB_SCALE_ITEM(1200, 0, 0),
};
static const DECLARE_TLV_DB_SCALE(line_tlv, -600, 600, 0);
static const char *speaker_ref_text[] = {
"SPKVDD/2",
"VMID",
};
static const struct soc_enum speaker_ref =
SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text);
static const char *speaker_mode_text[] = {
"Class D",
"Class AB",
};
static const struct soc_enum speaker_mode =
SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text);
static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
unsigned int reg;
int count = 0;
unsigned int val;
unsigned long timeout;
val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1;
/* Trigger the command */
snd_soc_write(codec, WM8993_DC_SERVO_0, val);
dev_dbg(codec->dev, "Waiting for DC servo...\n");
if (hubs->dcs_done_irq) {
timeout = wait_for_completion_timeout(&hubs->dcs_done,
msecs_to_jiffies(500));
if (timeout == 0)
dev_warn(codec->dev, "No DC servo interrupt\n");
reg = snd_soc_read(codec, WM8993_DC_SERVO_0);
} else {
do {
count++;
msleep(1);
reg = snd_soc_read(codec, WM8993_DC_SERVO_0);
dev_dbg(codec->dev, "DC servo: %x\n", reg);
} while (reg & op && count < 400);
}
if (reg & op)
dev_err(codec->dev, "Timed out waiting for DC Servo %x\n",
op);
}
irqreturn_t wm_hubs_dcs_done(int irq, void *data)
{
struct wm_hubs_data *hubs = data;
complete(&hubs->dcs_done);
return IRQ_HANDLED;
}
EXPORT_SYMBOL_GPL(wm_hubs_dcs_done);
/*
* Startup calibration of the DC servo
*/
static void calibrate_dc_servo(struct snd_soc_codec *codec)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
s8 offset;
u16 reg, reg_l, reg_r, dcs_cfg;
/* If we're using a digital only path and have a previously
* callibrated DC servo offset stored then use that. */
if (hubs->class_w && hubs->class_w_dcs) {
dev_dbg(codec->dev, "Using cached DC servo offset %x\n",
hubs->class_w_dcs);
snd_soc_write(codec, WM8993_DC_SERVO_3, hubs->class_w_dcs);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
return;
}
if (hubs->series_startup) {
/* Set for 32 series updates */
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_SERIES_NO_01_MASK,
32 << WM8993_DCS_SERIES_NO_01_SHIFT);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_SERIES_0 |
WM8993_DCS_TRIG_SERIES_1);
} else {
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_STARTUP_0 |
WM8993_DCS_TRIG_STARTUP_1);
}
/* Different chips in the family support different readback
* methods.
*/
switch (hubs->dcs_readback_mode) {
case 0:
reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1)
& WM8993_DCS_INTEG_CHAN_0_MASK;
reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2)
& WM8993_DCS_INTEG_CHAN_1_MASK;
break;
case 1:
reg = snd_soc_read(codec, WM8993_DC_SERVO_3);
reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK)
>> WM8993_DCS_DAC_WR_VAL_1_SHIFT;
reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK;
break;
default:
WARN(1, "Unknown DCS readback method\n");
break;
}
dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r);
/* Apply correction to DC servo result */
if (hubs->dcs_codes) {
dev_dbg(codec->dev, "Applying %d code DC servo correction\n",
hubs->dcs_codes);
/* HPOUT1R */
offset = reg_r;
offset += hubs->dcs_codes;
dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
/* HPOUT1L */
offset = reg_l;
offset += hubs->dcs_codes;
dcs_cfg |= (u8)offset;
dev_dbg(codec->dev, "DCS result: %x\n", dcs_cfg);
/* Do it */
snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg);
wait_for_dc_servo(codec,
WM8993_DCS_TRIG_DAC_WR_0 |
WM8993_DCS_TRIG_DAC_WR_1);
} else {
dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT;
dcs_cfg |= reg_l;
}
/* Save the callibrated offset if we're in class W mode and
* therefore don't have any analogue signal mixed in. */
if (hubs->class_w)
hubs->class_w_dcs = dcs_cfg;
}
/*
* Update the DC servo calibration on gain changes
*/
static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
int ret;
ret = snd_soc_put_volsw_2r(kcontrol, ucontrol);
/* Updating the analogue gains invalidates the DC servo cache */
hubs->class_w_dcs = 0;
/* If we're applying an offset correction then updating the
* callibration would be likely to introduce further offsets. */
if (hubs->dcs_codes || hubs->no_series_update)
return ret;
/* Only need to do this if the outputs are active */
if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1)
& (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA))
snd_soc_update_bits(codec,
WM8993_DC_SERVO_0,
WM8993_DCS_TRIG_SINGLE_0 |
WM8993_DCS_TRIG_SINGLE_1,
WM8993_DCS_TRIG_SINGLE_0 |
WM8993_DCS_TRIG_SINGLE_1);
return ret;
}
static const struct snd_kcontrol_new analogue_snd_controls[] = {
SOC_SINGLE_TLV("IN1L Volume", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1L Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
SOC_SINGLE("IN1L ZC Switch", WM8993_LEFT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN1R Volume", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN1R Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 7, 1, 1),
SOC_SINGLE("IN1R ZC Switch", WM8993_RIGHT_LINE_INPUT_1_2_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2L Volume", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2L Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
SOC_SINGLE("IN2L ZC Switch", WM8993_LEFT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("IN2R Volume", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 0, 31, 0,
inpga_tlv),
SOC_SINGLE("IN2R Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 7, 1, 1),
SOC_SINGLE("IN2R ZC Switch", WM8993_RIGHT_LINE_INPUT_3_4_VOLUME, 6, 1, 0),
SOC_SINGLE_TLV("MIXINL IN2L Volume", WM8993_INPUT_MIXER3, 7, 1, 0,
inmix_sw_tlv),
SOC_SINGLE_TLV("MIXINL IN1L Volume", WM8993_INPUT_MIXER3, 4, 1, 0,
inmix_sw_tlv),
SOC_SINGLE_TLV("MIXINL Output Record Volume", WM8993_INPUT_MIXER3, 0, 7, 0,
inmix_tlv),
SOC_SINGLE_TLV("MIXINL IN1LP Volume", WM8993_INPUT_MIXER5, 6, 7, 0, inmix_tlv),
SOC_SINGLE_TLV("MIXINL Direct Voice Volume", WM8993_INPUT_MIXER5, 0, 6, 0,
inmix_tlv),
SOC_SINGLE_TLV("MIXINR IN2R Volume", WM8993_INPUT_MIXER4, 7, 1, 0,
inmix_sw_tlv),
SOC_SINGLE_TLV("MIXINR IN1R Volume", WM8993_INPUT_MIXER4, 4, 1, 0,
inmix_sw_tlv),
SOC_SINGLE_TLV("MIXINR Output Record Volume", WM8993_INPUT_MIXER4, 0, 7, 0,
inmix_tlv),
SOC_SINGLE_TLV("MIXINR IN1RP Volume", WM8993_INPUT_MIXER6, 6, 7, 0, inmix_tlv),
SOC_SINGLE_TLV("MIXINR Direct Voice Volume", WM8993_INPUT_MIXER6, 0, 6, 0,
inmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer IN2RN Volume", WM8993_OUTPUT_MIXER5, 6, 7, 1,
outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer IN2LN Volume", WM8993_OUTPUT_MIXER3, 6, 7, 1,
outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer IN2LP Volume", WM8993_OUTPUT_MIXER3, 9, 7, 1,
outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer IN1L Volume", WM8993_OUTPUT_MIXER3, 0, 7, 1,
outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer IN1R Volume", WM8993_OUTPUT_MIXER3, 3, 7, 1,
outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer Right Input Volume",
WM8993_OUTPUT_MIXER5, 3, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer Left Input Volume",
WM8993_OUTPUT_MIXER5, 0, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Left Output Mixer DAC Volume", WM8993_OUTPUT_MIXER5, 9, 7, 1,
outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer IN2LN Volume",
WM8993_OUTPUT_MIXER6, 6, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer IN2RN Volume",
WM8993_OUTPUT_MIXER4, 6, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer IN1L Volume",
WM8993_OUTPUT_MIXER4, 3, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer IN1R Volume",
WM8993_OUTPUT_MIXER4, 0, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer IN2RP Volume",
WM8993_OUTPUT_MIXER4, 9, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer Left Input Volume",
WM8993_OUTPUT_MIXER6, 3, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer Right Input Volume",
WM8993_OUTPUT_MIXER6, 6, 7, 1, outmix_tlv),
SOC_SINGLE_TLV("Right Output Mixer DAC Volume",
WM8993_OUTPUT_MIXER6, 9, 7, 1, outmix_tlv),
SOC_DOUBLE_R_TLV("Output Volume", WM8993_LEFT_OPGA_VOLUME,
WM8993_RIGHT_OPGA_VOLUME, 0, 63, 0, outpga_tlv),
SOC_DOUBLE_R("Output Switch", WM8993_LEFT_OPGA_VOLUME,
WM8993_RIGHT_OPGA_VOLUME, 6, 1, 0),
SOC_DOUBLE_R("Output ZC Switch", WM8993_LEFT_OPGA_VOLUME,
WM8993_RIGHT_OPGA_VOLUME, 7, 1, 0),
SOC_SINGLE("Earpiece Switch", WM8993_HPOUT2_VOLUME, 5, 1, 1),
SOC_SINGLE_TLV("Earpiece Volume", WM8993_HPOUT2_VOLUME, 4, 1, 1, earpiece_tlv),
SOC_SINGLE_TLV("SPKL Input Volume", WM8993_SPKMIXL_ATTENUATION,
5, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKL IN1LP Volume", WM8993_SPKMIXL_ATTENUATION,
4, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKL Output Volume", WM8993_SPKMIXL_ATTENUATION,
3, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKR Input Volume", WM8993_SPKMIXR_ATTENUATION,
5, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKR IN1RP Volume", WM8993_SPKMIXR_ATTENUATION,
4, 1, 1, wm_hubs_spkmix_tlv),
SOC_SINGLE_TLV("SPKR Output Volume", WM8993_SPKMIXR_ATTENUATION,
3, 1, 1, wm_hubs_spkmix_tlv),
SOC_DOUBLE_R_TLV("Speaker Mixer Volume",
WM8993_SPKMIXL_ATTENUATION, WM8993_SPKMIXR_ATTENUATION,
0, 3, 1, spkmixout_tlv),
SOC_DOUBLE_R_TLV("Speaker Volume",
WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
0, 63, 0, outpga_tlv),
SOC_DOUBLE_R("Speaker Switch",
WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
6, 1, 0),
SOC_DOUBLE_R("Speaker ZC Switch",
WM8993_SPEAKER_VOLUME_LEFT, WM8993_SPEAKER_VOLUME_RIGHT,
7, 1, 0),
SOC_DOUBLE_TLV("Speaker Boost Volume", WM8993_SPKOUT_BOOST, 3, 0, 7, 0,
spkboost_tlv),
SOC_ENUM("Speaker Reference", speaker_ref),
SOC_ENUM("Speaker Mode", speaker_mode),
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Headphone Volume",
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
SNDRV_CTL_ELEM_ACCESS_READWRITE,
.tlv.p = outpga_tlv,
.info = snd_soc_info_volsw_2r,
.get = snd_soc_get_volsw_2r, .put = wm8993_put_dc_servo,
.private_value = (unsigned long)&(struct soc_mixer_control) {
.reg = WM8993_LEFT_OUTPUT_VOLUME,
.rreg = WM8993_RIGHT_OUTPUT_VOLUME,
.shift = 0, .max = 63
},
},
SOC_DOUBLE_R("Headphone Switch", WM8993_LEFT_OUTPUT_VOLUME,
WM8993_RIGHT_OUTPUT_VOLUME, 6, 1, 0),
SOC_DOUBLE_R("Headphone ZC Switch", WM8993_LEFT_OUTPUT_VOLUME,
WM8993_RIGHT_OUTPUT_VOLUME, 7, 1, 0),
SOC_SINGLE("LINEOUT1N Switch", WM8993_LINE_OUTPUTS_VOLUME, 6, 1, 1),
SOC_SINGLE("LINEOUT1P Switch", WM8993_LINE_OUTPUTS_VOLUME, 5, 1, 1),
SOC_SINGLE_TLV("LINEOUT1 Volume", WM8993_LINE_OUTPUTS_VOLUME, 4, 1, 1,
line_tlv),
SOC_SINGLE("LINEOUT2N Switch", WM8993_LINE_OUTPUTS_VOLUME, 2, 1, 1),
SOC_SINGLE("LINEOUT2P Switch", WM8993_LINE_OUTPUTS_VOLUME, 1, 1, 1),
SOC_SINGLE_TLV("LINEOUT2 Volume", WM8993_LINE_OUTPUTS_VOLUME, 0, 1, 1,
line_tlv),
};
static int hp_supply_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
switch (hubs->hp_startup_mode) {
case 0:
break;
case 1:
/* Enable the headphone amp */
snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
WM8993_HPOUT1L_ENA |
WM8993_HPOUT1R_ENA,
WM8993_HPOUT1L_ENA |
WM8993_HPOUT1R_ENA);
/* Enable the second stage */
snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
WM8993_HPOUT1L_DLY |
WM8993_HPOUT1R_DLY,
WM8993_HPOUT1L_DLY |
WM8993_HPOUT1R_DLY);
break;
default:
dev_err(codec->dev, "Unknown HP startup mode %d\n",
hubs->hp_startup_mode);
break;
}
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
WM8993_CP_ENA, 0);
break;
}
return 0;
}
static int hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
unsigned int reg = snd_soc_read(codec, WM8993_ANALOGUE_HP_0);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1,
WM8993_CP_ENA, WM8993_CP_ENA);
msleep(5);
snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA);
reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY;
snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
/* Smallest supported update interval */
snd_soc_update_bits(codec, WM8993_DC_SERVO_1,
WM8993_DCS_TIMER_PERIOD_01_MASK, 1);
calibrate_dc_servo(codec);
reg |= WM8993_HPOUT1R_OUTP | WM8993_HPOUT1R_RMV_SHORT |
WM8993_HPOUT1L_OUTP | WM8993_HPOUT1L_RMV_SHORT;
snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg);
break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
WM8993_HPOUT1L_OUTP |
WM8993_HPOUT1R_OUTP |
WM8993_HPOUT1L_RMV_SHORT |
WM8993_HPOUT1R_RMV_SHORT, 0);
snd_soc_update_bits(codec, WM8993_ANALOGUE_HP_0,
WM8993_HPOUT1L_DLY |
WM8993_HPOUT1R_DLY, 0);
snd_soc_write(codec, WM8993_DC_SERVO_0, 0);
snd_soc_update_bits(codec, WM8993_POWER_MANAGEMENT_1,
WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA,
0);
break;
}
return 0;
}
static int earpiece_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *control, int event)
{
struct snd_soc_codec *codec = w->codec;
u16 reg = snd_soc_read(codec, WM8993_ANTIPOP1) & ~WM8993_HPOUT2_IN_ENA;
switch (event) {
case SND_SOC_DAPM_PRE_PMU:
reg |= WM8993_HPOUT2_IN_ENA;
snd_soc_write(codec, WM8993_ANTIPOP1, reg);
udelay(50);
break;
case SND_SOC_DAPM_POST_PMD:
snd_soc_write(codec, WM8993_ANTIPOP1, reg);
break;
default:
BUG();
break;
}
return 0;
}
static const struct snd_kcontrol_new in1l_pga[] = {
SOC_DAPM_SINGLE("IN1LP Switch", WM8993_INPUT_MIXER2, 5, 1, 0),
SOC_DAPM_SINGLE("IN1LN Switch", WM8993_INPUT_MIXER2, 4, 1, 0),
};
static const struct snd_kcontrol_new in1r_pga[] = {
SOC_DAPM_SINGLE("IN1RP Switch", WM8993_INPUT_MIXER2, 1, 1, 0),
SOC_DAPM_SINGLE("IN1RN Switch", WM8993_INPUT_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new in2l_pga[] = {
SOC_DAPM_SINGLE("IN2LP Switch", WM8993_INPUT_MIXER2, 7, 1, 0),
SOC_DAPM_SINGLE("IN2LN Switch", WM8993_INPUT_MIXER2, 6, 1, 0),
};
static const struct snd_kcontrol_new in2r_pga[] = {
SOC_DAPM_SINGLE("IN2RP Switch", WM8993_INPUT_MIXER2, 3, 1, 0),
SOC_DAPM_SINGLE("IN2RN Switch", WM8993_INPUT_MIXER2, 2, 1, 0),
};
static const struct snd_kcontrol_new mixinl[] = {
SOC_DAPM_SINGLE("IN2L Switch", WM8993_INPUT_MIXER3, 8, 1, 0),
SOC_DAPM_SINGLE("IN1L Switch", WM8993_INPUT_MIXER3, 5, 1, 0),
};
static const struct snd_kcontrol_new mixinr[] = {
SOC_DAPM_SINGLE("IN2R Switch", WM8993_INPUT_MIXER4, 8, 1, 0),
SOC_DAPM_SINGLE("IN1R Switch", WM8993_INPUT_MIXER4, 5, 1, 0),
};
static const struct snd_kcontrol_new left_output_mixer[] = {
SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER1, 7, 1, 0),
SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER1, 6, 1, 0),
SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER1, 5, 1, 0),
SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER1, 4, 1, 0),
SOC_DAPM_SINGLE("IN2LP Switch", WM8993_OUTPUT_MIXER1, 1, 1, 0),
SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER1, 3, 1, 0),
SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER1, 2, 1, 0),
SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new right_output_mixer[] = {
SOC_DAPM_SINGLE("Left Input Switch", WM8993_OUTPUT_MIXER2, 7, 1, 0),
SOC_DAPM_SINGLE("Right Input Switch", WM8993_OUTPUT_MIXER2, 6, 1, 0),
SOC_DAPM_SINGLE("IN2LN Switch", WM8993_OUTPUT_MIXER2, 5, 1, 0),
SOC_DAPM_SINGLE("IN2RN Switch", WM8993_OUTPUT_MIXER2, 4, 1, 0),
SOC_DAPM_SINGLE("IN1L Switch", WM8993_OUTPUT_MIXER2, 3, 1, 0),
SOC_DAPM_SINGLE("IN1R Switch", WM8993_OUTPUT_MIXER2, 2, 1, 0),
SOC_DAPM_SINGLE("IN2RP Switch", WM8993_OUTPUT_MIXER2, 1, 1, 0),
SOC_DAPM_SINGLE("DAC Switch", WM8993_OUTPUT_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new earpiece_mixer[] = {
SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_HPOUT2_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Left Output Switch", WM8993_HPOUT2_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Right Output Switch", WM8993_HPOUT2_MIXER, 3, 1, 0),
};
static const struct snd_kcontrol_new left_speaker_boost[] = {
SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_SPKOUT_MIXERS, 5, 1, 0),
SOC_DAPM_SINGLE("SPKL Switch", WM8993_SPKOUT_MIXERS, 4, 1, 0),
SOC_DAPM_SINGLE("SPKR Switch", WM8993_SPKOUT_MIXERS, 3, 1, 0),
};
static const struct snd_kcontrol_new right_speaker_boost[] = {
SOC_DAPM_SINGLE("Direct Voice Switch", WM8993_SPKOUT_MIXERS, 2, 1, 0),
SOC_DAPM_SINGLE("SPKL Switch", WM8993_SPKOUT_MIXERS, 1, 1, 0),
SOC_DAPM_SINGLE("SPKR Switch", WM8993_SPKOUT_MIXERS, 0, 1, 0),
};
static const struct snd_kcontrol_new line1_mix[] = {
SOC_DAPM_SINGLE("IN1R Switch", WM8993_LINE_MIXER1, 2, 1, 0),
SOC_DAPM_SINGLE("IN1L Switch", WM8993_LINE_MIXER1, 1, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new line1n_mix[] = {
SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 6, 1, 0),
SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER1, 5, 1, 0),
};
static const struct snd_kcontrol_new line1p_mix[] = {
SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER1, 0, 1, 0),
};
static const struct snd_kcontrol_new line2_mix[] = {
SOC_DAPM_SINGLE("IN2R Switch", WM8993_LINE_MIXER2, 2, 1, 0),
SOC_DAPM_SINGLE("IN2L Switch", WM8993_LINE_MIXER2, 1, 1, 0),
SOC_DAPM_SINGLE("Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
};
static const struct snd_kcontrol_new line2n_mix[] = {
SOC_DAPM_SINGLE("Left Output Switch", WM8993_LINE_MIXER2, 6, 1, 0),
SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 5, 1, 0),
};
static const struct snd_kcontrol_new line2p_mix[] = {
SOC_DAPM_SINGLE("Right Output Switch", WM8993_LINE_MIXER2, 0, 1, 0),
};
static const struct snd_soc_dapm_widget analogue_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("IN1LN"),
SND_SOC_DAPM_INPUT("IN1LP"),
SND_SOC_DAPM_INPUT("IN2LN"),
SND_SOC_DAPM_INPUT("IN2LP:VXRN"),
SND_SOC_DAPM_INPUT("IN1RN"),
SND_SOC_DAPM_INPUT("IN1RP"),
SND_SOC_DAPM_INPUT("IN2RN"),
SND_SOC_DAPM_INPUT("IN2RP:VXRP"),
SND_SOC_DAPM_MICBIAS("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0),
SND_SOC_DAPM_MICBIAS("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0),
SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0,
in1l_pga, ARRAY_SIZE(in1l_pga)),
SND_SOC_DAPM_MIXER("IN1R PGA", WM8993_POWER_MANAGEMENT_2, 4, 0,
in1r_pga, ARRAY_SIZE(in1r_pga)),
SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0,
in2l_pga, ARRAY_SIZE(in2l_pga)),
SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0,
in2r_pga, ARRAY_SIZE(in2r_pga)),
SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0,
mixinl, ARRAY_SIZE(mixinl)),
SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0,
mixinr, ARRAY_SIZE(mixinr)),
SND_SOC_DAPM_MIXER("Left Output Mixer", WM8993_POWER_MANAGEMENT_3, 5, 0,
left_output_mixer, ARRAY_SIZE(left_output_mixer)),
SND_SOC_DAPM_MIXER("Right Output Mixer", WM8993_POWER_MANAGEMENT_3, 4, 0,
right_output_mixer, ARRAY_SIZE(right_output_mixer)),
SND_SOC_DAPM_PGA("Left Output PGA", WM8993_POWER_MANAGEMENT_3, 7, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Output PGA", WM8993_POWER_MANAGEMENT_3, 6, 0, NULL, 0),
SND_SOC_DAPM_SUPPLY("Headphone Supply", SND_SOC_NOPM, 0, 0, hp_supply_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Headphone PGA", SND_SOC_NOPM, 0, 0,
NULL, 0,
hp_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0,
earpiece_mixer, ARRAY_SIZE(earpiece_mixer)),
SND_SOC_DAPM_PGA_E("Earpiece Driver", WM8993_POWER_MANAGEMENT_1, 11, 0,
NULL, 0, earpiece_event,
SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
SND_SOC_DAPM_MIXER("SPKL Boost", SND_SOC_NOPM, 0, 0,
left_speaker_boost, ARRAY_SIZE(left_speaker_boost)),
SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0,
right_speaker_boost, ARRAY_SIZE(right_speaker_boost)),
SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0,
NULL, 0),
SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0,
NULL, 0),
SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0,
line1_mix, ARRAY_SIZE(line1_mix)),
SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0,
line2_mix, ARRAY_SIZE(line2_mix)),
SND_SOC_DAPM_MIXER("LINEOUT1N Mixer", SND_SOC_NOPM, 0, 0,
line1n_mix, ARRAY_SIZE(line1n_mix)),
SND_SOC_DAPM_MIXER("LINEOUT1P Mixer", SND_SOC_NOPM, 0, 0,
line1p_mix, ARRAY_SIZE(line1p_mix)),
SND_SOC_DAPM_MIXER("LINEOUT2N Mixer", SND_SOC_NOPM, 0, 0,
line2n_mix, ARRAY_SIZE(line2n_mix)),
SND_SOC_DAPM_MIXER("LINEOUT2P Mixer", SND_SOC_NOPM, 0, 0,
line2p_mix, ARRAY_SIZE(line2p_mix)),
SND_SOC_DAPM_PGA("LINEOUT1N Driver", WM8993_POWER_MANAGEMENT_3, 13, 0,
NULL, 0),
SND_SOC_DAPM_PGA("LINEOUT1P Driver", WM8993_POWER_MANAGEMENT_3, 12, 0,
NULL, 0),
SND_SOC_DAPM_PGA("LINEOUT2N Driver", WM8993_POWER_MANAGEMENT_3, 11, 0,
NULL, 0),
SND_SOC_DAPM_PGA("LINEOUT2P Driver", WM8993_POWER_MANAGEMENT_3, 10, 0,
NULL, 0),
SND_SOC_DAPM_OUTPUT("SPKOUTLP"),
SND_SOC_DAPM_OUTPUT("SPKOUTLN"),
SND_SOC_DAPM_OUTPUT("SPKOUTRP"),
SND_SOC_DAPM_OUTPUT("SPKOUTRN"),
SND_SOC_DAPM_OUTPUT("HPOUT1L"),
SND_SOC_DAPM_OUTPUT("HPOUT1R"),
SND_SOC_DAPM_OUTPUT("HPOUT2P"),
SND_SOC_DAPM_OUTPUT("HPOUT2N"),
SND_SOC_DAPM_OUTPUT("LINEOUT1P"),
SND_SOC_DAPM_OUTPUT("LINEOUT1N"),
SND_SOC_DAPM_OUTPUT("LINEOUT2P"),
SND_SOC_DAPM_OUTPUT("LINEOUT2N"),
};
static const struct snd_soc_dapm_route analogue_routes[] = {
{ "MICBIAS1", NULL, "CLK_SYS" },
{ "MICBIAS2", NULL, "CLK_SYS" },
{ "IN1L PGA", "IN1LP Switch", "IN1LP" },
{ "IN1L PGA", "IN1LN Switch", "IN1LN" },
{ "IN1R PGA", "IN1RP Switch", "IN1RP" },
{ "IN1R PGA", "IN1RN Switch", "IN1RN" },
{ "IN2L PGA", "IN2LP Switch", "IN2LP:VXRN" },
{ "IN2L PGA", "IN2LN Switch", "IN2LN" },
{ "IN2R PGA", "IN2RP Switch", "IN2RP:VXRP" },
{ "IN2R PGA", "IN2RN Switch", "IN2RN" },
{ "Direct Voice", NULL, "IN2LP:VXRN" },
{ "Direct Voice", NULL, "IN2RP:VXRP" },
{ "MIXINL", "IN1L Switch", "IN1L PGA" },
{ "MIXINL", "IN2L Switch", "IN2L PGA" },
{ "MIXINL", NULL, "Direct Voice" },
{ "MIXINL", NULL, "IN1LP" },
{ "MIXINL", NULL, "Left Output Mixer" },
{ "MIXINR", "IN1R Switch", "IN1R PGA" },
{ "MIXINR", "IN2R Switch", "IN2R PGA" },
{ "MIXINR", NULL, "Direct Voice" },
{ "MIXINR", NULL, "IN1RP" },
{ "MIXINR", NULL, "Right Output Mixer" },
{ "ADCL", NULL, "MIXINL" },
{ "ADCR", NULL, "MIXINR" },
{ "Left Output Mixer", "Left Input Switch", "MIXINL" },
{ "Left Output Mixer", "Right Input Switch", "MIXINR" },
{ "Left Output Mixer", "IN2RN Switch", "IN2RN" },
{ "Left Output Mixer", "IN2LN Switch", "IN2LN" },
{ "Left Output Mixer", "IN2LP Switch", "IN2LP:VXRN" },
{ "Left Output Mixer", "IN1L Switch", "IN1L PGA" },
{ "Left Output Mixer", "IN1R Switch", "IN1R PGA" },
{ "Right Output Mixer", "Left Input Switch", "MIXINL" },
{ "Right Output Mixer", "Right Input Switch", "MIXINR" },
{ "Right Output Mixer", "IN2LN Switch", "IN2LN" },
{ "Right Output Mixer", "IN2RN Switch", "IN2RN" },
{ "Right Output Mixer", "IN2RP Switch", "IN2RP:VXRP" },
{ "Right Output Mixer", "IN1L Switch", "IN1L PGA" },
{ "Right Output Mixer", "IN1R Switch", "IN1R PGA" },
{ "Left Output PGA", NULL, "Left Output Mixer" },
{ "Left Output PGA", NULL, "TOCLK" },
{ "Right Output PGA", NULL, "Right Output Mixer" },
{ "Right Output PGA", NULL, "TOCLK" },
{ "Earpiece Mixer", "Direct Voice Switch", "Direct Voice" },
{ "Earpiece Mixer", "Left Output Switch", "Left Output PGA" },
{ "Earpiece Mixer", "Right Output Switch", "Right Output PGA" },
{ "Earpiece Driver", NULL, "Earpiece Mixer" },
{ "HPOUT2N", NULL, "Earpiece Driver" },
{ "HPOUT2P", NULL, "Earpiece Driver" },
{ "SPKL", "Input Switch", "MIXINL" },
{ "SPKL", "IN1LP Switch", "IN1LP" },
{ "SPKL", "Output Switch", "Left Output PGA" },
{ "SPKL", NULL, "TOCLK" },
{ "SPKR", "Input Switch", "MIXINR" },
{ "SPKR", "IN1RP Switch", "IN1RP" },
{ "SPKR", "Output Switch", "Right Output PGA" },
{ "SPKR", NULL, "TOCLK" },
{ "SPKL Boost", "Direct Voice Switch", "Direct Voice" },
{ "SPKL Boost", "SPKL Switch", "SPKL" },
{ "SPKL Boost", "SPKR Switch", "SPKR" },
{ "SPKR Boost", "Direct Voice Switch", "Direct Voice" },
{ "SPKR Boost", "SPKR Switch", "SPKR" },
{ "SPKR Boost", "SPKL Switch", "SPKL" },
{ "SPKL Driver", NULL, "SPKL Boost" },
{ "SPKL Driver", NULL, "CLK_SYS" },
{ "SPKR Driver", NULL, "SPKR Boost" },
{ "SPKR Driver", NULL, "CLK_SYS" },
{ "SPKOUTLP", NULL, "SPKL Driver" },
{ "SPKOUTLN", NULL, "SPKL Driver" },
{ "SPKOUTRP", NULL, "SPKR Driver" },
{ "SPKOUTRN", NULL, "SPKR Driver" },
{ "Left Headphone Mux", "Mixer", "Left Output PGA" },
{ "Right Headphone Mux", "Mixer", "Right Output PGA" },
{ "Headphone PGA", NULL, "Left Headphone Mux" },
{ "Headphone PGA", NULL, "Right Headphone Mux" },
{ "Headphone PGA", NULL, "CLK_SYS" },
{ "Headphone PGA", NULL, "Headphone Supply" },
{ "HPOUT1L", NULL, "Headphone PGA" },
{ "HPOUT1R", NULL, "Headphone PGA" },
{ "LINEOUT1N", NULL, "LINEOUT1N Driver" },
{ "LINEOUT1P", NULL, "LINEOUT1P Driver" },
{ "LINEOUT2N", NULL, "LINEOUT2N Driver" },
{ "LINEOUT2P", NULL, "LINEOUT2P Driver" },
};
static const struct snd_soc_dapm_route lineout1_diff_routes[] = {
{ "LINEOUT1 Mixer", "IN1L Switch", "IN1L PGA" },
{ "LINEOUT1 Mixer", "IN1R Switch", "IN1R PGA" },
{ "LINEOUT1 Mixer", "Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1 Mixer" },
{ "LINEOUT1P Driver", NULL, "LINEOUT1 Mixer" },
};
static const struct snd_soc_dapm_route lineout1_se_routes[] = {
{ "LINEOUT1N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT1P Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT1N Driver", NULL, "LINEOUT1N Mixer" },
{ "LINEOUT1P Driver", NULL, "LINEOUT1P Mixer" },
};
static const struct snd_soc_dapm_route lineout2_diff_routes[] = {
{ "LINEOUT2 Mixer", "IN2L Switch", "IN2L PGA" },
{ "LINEOUT2 Mixer", "IN2R Switch", "IN2R PGA" },
{ "LINEOUT2 Mixer", "Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2 Mixer" },
{ "LINEOUT2P Driver", NULL, "LINEOUT2 Mixer" },
};
static const struct snd_soc_dapm_route lineout2_se_routes[] = {
{ "LINEOUT2N Mixer", "Left Output Switch", "Left Output PGA" },
{ "LINEOUT2N Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2P Mixer", "Right Output Switch", "Right Output PGA" },
{ "LINEOUT2N Driver", NULL, "LINEOUT2N Mixer" },
{ "LINEOUT2P Driver", NULL, "LINEOUT2P Mixer" },
};
int wm_hubs_add_analogue_controls(struct snd_soc_codec *codec)
{
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
/* Latch volume update bits & default ZC on */
snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_1_2_VOLUME,
WM8993_IN1_VU, WM8993_IN1_VU);
snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_1_2_VOLUME,
WM8993_IN1_VU, WM8993_IN1_VU);
snd_soc_update_bits(codec, WM8993_LEFT_LINE_INPUT_3_4_VOLUME,
WM8993_IN2_VU, WM8993_IN2_VU);
snd_soc_update_bits(codec, WM8993_RIGHT_LINE_INPUT_3_4_VOLUME,
WM8993_IN2_VU, WM8993_IN2_VU);
snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_LEFT,
WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
snd_soc_update_bits(codec, WM8993_SPEAKER_VOLUME_RIGHT,
WM8993_SPKOUT_VU, WM8993_SPKOUT_VU);
snd_soc_update_bits(codec, WM8993_LEFT_OUTPUT_VOLUME,
WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC,
WM8993_HPOUT1_VU | WM8993_HPOUT1L_ZC);
snd_soc_update_bits(codec, WM8993_RIGHT_OUTPUT_VOLUME,
WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC,
WM8993_HPOUT1_VU | WM8993_HPOUT1R_ZC);
snd_soc_update_bits(codec, WM8993_LEFT_OPGA_VOLUME,
WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU,
WM8993_MIXOUTL_ZC | WM8993_MIXOUT_VU);
snd_soc_update_bits(codec, WM8993_RIGHT_OPGA_VOLUME,
WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU,
WM8993_MIXOUTR_ZC | WM8993_MIXOUT_VU);
snd_soc_add_controls(codec, analogue_snd_controls,
ARRAY_SIZE(analogue_snd_controls));
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_new_controls(dapm, analogue_dapm_widgets,
ARRAY_SIZE(analogue_dapm_widgets));
return 0;
}
EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls);
int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec,
int lineout1_diff, int lineout2_diff)
{
struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec);
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
struct snd_soc_dapm_context *dapm = &codec->dapm;
init_completion(&hubs->dcs_done);
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm, analogue_routes,
ARRAY_SIZE(analogue_routes));
if (lineout1_diff)
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm,
lineout1_diff_routes,
ARRAY_SIZE(lineout1_diff_routes));
else
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm,
lineout1_se_routes,
ARRAY_SIZE(lineout1_se_routes));
if (lineout2_diff)
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm,
lineout2_diff_routes,
ARRAY_SIZE(lineout2_diff_routes));
else
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
snd_soc_dapm_add_routes(dapm,
lineout2_se_routes,
ARRAY_SIZE(lineout2_se_routes));
return 0;
}
EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes);
int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec,
int lineout1_diff, int lineout2_diff,
int lineout1fb, int lineout2fb,
int jd_scthr, int jd_thr, int micbias1_lvl,
int micbias2_lvl)
{
if (!lineout1_diff)
snd_soc_update_bits(codec, WM8993_LINE_MIXER1,
WM8993_LINEOUT1_MODE,
WM8993_LINEOUT1_MODE);
if (!lineout2_diff)
snd_soc_update_bits(codec, WM8993_LINE_MIXER2,
WM8993_LINEOUT2_MODE,
WM8993_LINEOUT2_MODE);
/* If the line outputs are differential then we aren't presenting
* VMID as an output and can disable it.
*/
if (lineout1_diff && lineout2_diff)
ASoC: Decouple DAPM from CODECs Decoupling Dynamic Audio Power Management (DAPM) from codec devices is required when developing ASoC further. Such as for other ASoC components to have DAPM widgets or when extending DAPM to handle cross-device paths. This patch decouples DAPM related variables from struct snd_soc_codec and moves them to new struct snd_soc_dapm_context that is used to encapsulate DAPM context of a device. ASoC core and API of DAPM functions are modified to use DAPM context instead of codec. This patch does not change current functionality and a large part of changes come because of structure and internal API changes. Core implementation is from Liam Girdwood <lrg@slimlogic.co.uk> with some minor core changes, codecs and machine driver conversions from Jarkko Nikula <jhnikula@gmail.com>. Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Nicolas Ferre <nicolas.ferre@atmel.com> Cc: Manuel Lauss <manuel.lauss@googlemail.com> Cc: Mike Frysinger <vapier.adi@gmail.com> Cc: Cliff Cai <cliff.cai@analog.com> Cc: Kevin Hilman <khilman@deeprootsystems.com> Cc: Ryan Mallon <ryan@bluewatersys.com> Cc: Timur Tabi <timur@freescale.com> Cc: Sascha Hauer <s.hauer@pengutronix.de> Cc: Lars-Peter Clausen <lars@metafoo.de> Cc: Arnaud Patard (Rtp) <arnaud.patard@rtp-net.org> Cc: Wan ZongShun <mcuos.com@gmail.com> Cc: Eric Miao <eric.y.miao@gmail.com> Cc: Jassi Brar <jassi.brar@samsung.com> Cc: Daniel Gloeckner <dg@emlix.com> Cc: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-11-05 21:53:46 +08:00
codec->dapm.idle_bias_off = 1;
if (lineout1fb)
snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
WM8993_LINEOUT1_FB, WM8993_LINEOUT1_FB);
if (lineout2fb)
snd_soc_update_bits(codec, WM8993_ADDITIONAL_CONTROL,
WM8993_LINEOUT2_FB, WM8993_LINEOUT2_FB);
snd_soc_update_bits(codec, WM8993_MICBIAS,
WM8993_JD_SCTHR_MASK | WM8993_JD_THR_MASK |
WM8993_MICB1_LVL | WM8993_MICB2_LVL,
jd_scthr << WM8993_JD_SCTHR_SHIFT |
jd_thr << WM8993_JD_THR_SHIFT |
micbias1_lvl |
micbias2_lvl << WM8993_MICB2_LVL_SHIFT);
return 0;
}
EXPORT_SYMBOL_GPL(wm_hubs_handle_analogue_pdata);
MODULE_DESCRIPTION("Shared support for Wolfson hubs products");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");