linux_dsm_epyc7002/sound/soc/codecs/wm8961.c

991 lines
28 KiB
C
Raw Normal View History

/*
* wm8961.c -- WM8961 ALSA SoC Audio driver
*
* Copyright 2009-10 Wolfson Microelectronics, plc
*
* Author: Mark Brown
*
* 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.
*
* Currently unimplemented features:
* - ALC
*/
#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/regmap.h>
include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo <tj@kernel.org> Guess-its-ok-by: Christoph Lameter <cl@linux-foundation.org> Cc: Ingo Molnar <mingo@redhat.com> Cc: Lee Schermerhorn <Lee.Schermerhorn@hp.com>
2010-03-24 15:04:11 +07:00
#include <linux/slab.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 "wm8961.h"
#define WM8961_MAX_REGISTER 0xFC
static const struct reg_default wm8961_reg_defaults[] = {
{ 0, 0x009F }, /* R0 - Left Input volume */
{ 1, 0x009F }, /* R1 - Right Input volume */
{ 2, 0x0000 }, /* R2 - LOUT1 volume */
{ 3, 0x0000 }, /* R3 - ROUT1 volume */
{ 4, 0x0020 }, /* R4 - Clocking1 */
{ 5, 0x0008 }, /* R5 - ADC & DAC Control 1 */
{ 6, 0x0000 }, /* R6 - ADC & DAC Control 2 */
{ 7, 0x000A }, /* R7 - Audio Interface 0 */
{ 8, 0x01F4 }, /* R8 - Clocking2 */
{ 9, 0x0000 }, /* R9 - Audio Interface 1 */
{ 10, 0x00FF }, /* R10 - Left DAC volume */
{ 11, 0x00FF }, /* R11 - Right DAC volume */
{ 14, 0x0040 }, /* R14 - Audio Interface 2 */
{ 17, 0x007B }, /* R17 - ALC1 */
{ 18, 0x0000 }, /* R18 - ALC2 */
{ 19, 0x0032 }, /* R19 - ALC3 */
{ 20, 0x0000 }, /* R20 - Noise Gate */
{ 21, 0x00C0 }, /* R21 - Left ADC volume */
{ 22, 0x00C0 }, /* R22 - Right ADC volume */
{ 23, 0x0120 }, /* R23 - Additional control(1) */
{ 24, 0x0000 }, /* R24 - Additional control(2) */
{ 25, 0x0000 }, /* R25 - Pwr Mgmt (1) */
{ 26, 0x0000 }, /* R26 - Pwr Mgmt (2) */
{ 27, 0x0000 }, /* R27 - Additional Control (3) */
{ 28, 0x0000 }, /* R28 - Anti-pop */
{ 30, 0x005F }, /* R30 - Clocking 3 */
{ 32, 0x0000 }, /* R32 - ADCL signal path */
{ 33, 0x0000 }, /* R33 - ADCR signal path */
{ 40, 0x0000 }, /* R40 - LOUT2 volume */
{ 41, 0x0000 }, /* R41 - ROUT2 volume */
{ 47, 0x0000 }, /* R47 - Pwr Mgmt (3) */
{ 48, 0x0023 }, /* R48 - Additional Control (4) */
{ 49, 0x0000 }, /* R49 - Class D Control 1 */
{ 51, 0x0003 }, /* R51 - Class D Control 2 */
{ 56, 0x0106 }, /* R56 - Clocking 4 */
{ 57, 0x0000 }, /* R57 - DSP Sidetone 0 */
{ 58, 0x0000 }, /* R58 - DSP Sidetone 1 */
{ 60, 0x0000 }, /* R60 - DC Servo 0 */
{ 61, 0x0000 }, /* R61 - DC Servo 1 */
{ 63, 0x015E }, /* R63 - DC Servo 3 */
{ 65, 0x0010 }, /* R65 - DC Servo 5 */
{ 68, 0x0003 }, /* R68 - Analogue PGA Bias */
{ 69, 0x0000 }, /* R69 - Analogue HP 0 */
{ 71, 0x01FB }, /* R71 - Analogue HP 2 */
{ 72, 0x0000 }, /* R72 - Charge Pump 1 */
{ 82, 0x0000 }, /* R82 - Charge Pump B */
{ 87, 0x0000 }, /* R87 - Write Sequencer 1 */
{ 88, 0x0000 }, /* R88 - Write Sequencer 2 */
{ 89, 0x0000 }, /* R89 - Write Sequencer 3 */
{ 90, 0x0000 }, /* R90 - Write Sequencer 4 */
{ 91, 0x0000 }, /* R91 - Write Sequencer 5 */
{ 92, 0x0000 }, /* R92 - Write Sequencer 6 */
{ 93, 0x0000 }, /* R93 - Write Sequencer 7 */
{ 252, 0x0001 }, /* R252 - General test 1 */
};
struct wm8961_priv {
struct regmap *regmap;
int sysclk;
};
static bool wm8961_volatile(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM8961_SOFTWARE_RESET:
case WM8961_WRITE_SEQUENCER_7:
case WM8961_DC_SERVO_1:
return true;
default:
return false;
}
}
static bool wm8961_readable(struct device *dev, unsigned int reg)
{
switch (reg) {
case WM8961_LEFT_INPUT_VOLUME:
case WM8961_RIGHT_INPUT_VOLUME:
case WM8961_LOUT1_VOLUME:
case WM8961_ROUT1_VOLUME:
case WM8961_CLOCKING1:
case WM8961_ADC_DAC_CONTROL_1:
case WM8961_ADC_DAC_CONTROL_2:
case WM8961_AUDIO_INTERFACE_0:
case WM8961_CLOCKING2:
case WM8961_AUDIO_INTERFACE_1:
case WM8961_LEFT_DAC_VOLUME:
case WM8961_RIGHT_DAC_VOLUME:
case WM8961_AUDIO_INTERFACE_2:
case WM8961_SOFTWARE_RESET:
case WM8961_ALC1:
case WM8961_ALC2:
case WM8961_ALC3:
case WM8961_NOISE_GATE:
case WM8961_LEFT_ADC_VOLUME:
case WM8961_RIGHT_ADC_VOLUME:
case WM8961_ADDITIONAL_CONTROL_1:
case WM8961_ADDITIONAL_CONTROL_2:
case WM8961_PWR_MGMT_1:
case WM8961_PWR_MGMT_2:
case WM8961_ADDITIONAL_CONTROL_3:
case WM8961_ANTI_POP:
case WM8961_CLOCKING_3:
case WM8961_ADCL_SIGNAL_PATH:
case WM8961_ADCR_SIGNAL_PATH:
case WM8961_LOUT2_VOLUME:
case WM8961_ROUT2_VOLUME:
case WM8961_PWR_MGMT_3:
case WM8961_ADDITIONAL_CONTROL_4:
case WM8961_CLASS_D_CONTROL_1:
case WM8961_CLASS_D_CONTROL_2:
case WM8961_CLOCKING_4:
case WM8961_DSP_SIDETONE_0:
case WM8961_DSP_SIDETONE_1:
case WM8961_DC_SERVO_0:
case WM8961_DC_SERVO_1:
case WM8961_DC_SERVO_3:
case WM8961_DC_SERVO_5:
case WM8961_ANALOGUE_PGA_BIAS:
case WM8961_ANALOGUE_HP_0:
case WM8961_ANALOGUE_HP_2:
case WM8961_CHARGE_PUMP_1:
case WM8961_CHARGE_PUMP_B:
case WM8961_WRITE_SEQUENCER_1:
case WM8961_WRITE_SEQUENCER_2:
case WM8961_WRITE_SEQUENCER_3:
case WM8961_WRITE_SEQUENCER_4:
case WM8961_WRITE_SEQUENCER_5:
case WM8961_WRITE_SEQUENCER_6:
case WM8961_WRITE_SEQUENCER_7:
case WM8961_GENERAL_TEST_1:
return true;
default:
return false;
}
}
/*
* The headphone output supports special anti-pop sequences giving
* silent power up and power down.
*/
static int wm8961_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
u16 hp_reg = snd_soc_component_read32(component, WM8961_ANALOGUE_HP_0);
u16 cp_reg = snd_soc_component_read32(component, WM8961_CHARGE_PUMP_1);
u16 pwr_reg = snd_soc_component_read32(component, WM8961_PWR_MGMT_2);
u16 dcs_reg = snd_soc_component_read32(component, WM8961_DC_SERVO_1);
int timeout = 500;
if (event & SND_SOC_DAPM_POST_PMU) {
/* Make sure the output is shorted */
hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT);
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Enable the charge pump */
cp_reg |= WM8961_CP_ENA;
snd_soc_component_write(component, WM8961_CHARGE_PUMP_1, cp_reg);
mdelay(5);
/* Enable the PGA */
pwr_reg |= WM8961_LOUT1_PGA | WM8961_ROUT1_PGA;
snd_soc_component_write(component, WM8961_PWR_MGMT_2, pwr_reg);
/* Enable the amplifier */
hp_reg |= WM8961_HPR_ENA | WM8961_HPL_ENA;
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Second stage enable */
hp_reg |= WM8961_HPR_ENA_DLY | WM8961_HPL_ENA_DLY;
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Enable the DC servo & trigger startup */
dcs_reg |=
WM8961_DCS_ENA_CHAN_HPR | WM8961_DCS_TRIG_STARTUP_HPR |
WM8961_DCS_ENA_CHAN_HPL | WM8961_DCS_TRIG_STARTUP_HPL;
dev_dbg(component->dev, "Enabling DC servo\n");
snd_soc_component_write(component, WM8961_DC_SERVO_1, dcs_reg);
do {
msleep(1);
dcs_reg = snd_soc_component_read32(component, WM8961_DC_SERVO_1);
} while (--timeout &&
dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR |
WM8961_DCS_TRIG_STARTUP_HPL));
if (dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR |
WM8961_DCS_TRIG_STARTUP_HPL))
dev_err(component->dev, "DC servo timed out\n");
else
dev_dbg(component->dev, "DC servo startup complete\n");
/* Enable the output stage */
hp_reg |= WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP;
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Remove the short on the output stage */
hp_reg |= WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT;
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
}
if (event & SND_SOC_DAPM_PRE_PMD) {
/* Short the output */
hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT);
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Disable the output stage */
hp_reg &= ~(WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP);
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Disable DC offset cancellation */
dcs_reg &= ~(WM8961_DCS_ENA_CHAN_HPR |
WM8961_DCS_ENA_CHAN_HPL);
snd_soc_component_write(component, WM8961_DC_SERVO_1, dcs_reg);
/* Finish up */
hp_reg &= ~(WM8961_HPR_ENA_DLY | WM8961_HPR_ENA |
WM8961_HPL_ENA_DLY | WM8961_HPL_ENA);
snd_soc_component_write(component, WM8961_ANALOGUE_HP_0, hp_reg);
/* Disable the PGA */
pwr_reg &= ~(WM8961_LOUT1_PGA | WM8961_ROUT1_PGA);
snd_soc_component_write(component, WM8961_PWR_MGMT_2, pwr_reg);
/* Disable the charge pump */
dev_dbg(component->dev, "Disabling charge pump\n");
snd_soc_component_write(component, WM8961_CHARGE_PUMP_1,
cp_reg & ~WM8961_CP_ENA);
}
return 0;
}
static int wm8961_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
u16 pwr_reg = snd_soc_component_read32(component, WM8961_PWR_MGMT_2);
u16 spk_reg = snd_soc_component_read32(component, WM8961_CLASS_D_CONTROL_1);
if (event & SND_SOC_DAPM_POST_PMU) {
/* Enable the PGA */
pwr_reg |= WM8961_SPKL_PGA | WM8961_SPKR_PGA;
snd_soc_component_write(component, WM8961_PWR_MGMT_2, pwr_reg);
/* Enable the amplifier */
spk_reg |= WM8961_SPKL_ENA | WM8961_SPKR_ENA;
snd_soc_component_write(component, WM8961_CLASS_D_CONTROL_1, spk_reg);
}
if (event & SND_SOC_DAPM_PRE_PMD) {
/* Disable the amplifier */
spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA);
snd_soc_component_write(component, WM8961_CLASS_D_CONTROL_1, spk_reg);
/* Disable the PGA */
pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA);
snd_soc_component_write(component, WM8961_PWR_MGMT_2, pwr_reg);
}
return 0;
}
static const char *adc_hpf_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3",
};
static SOC_ENUM_SINGLE_DECL(adc_hpf,
WM8961_ADC_DAC_CONTROL_2, 7, adc_hpf_text);
static const char *dac_deemph_text[] = {
"None", "32kHz", "44.1kHz", "48kHz",
};
static SOC_ENUM_SINGLE_DECL(dac_deemph,
WM8961_ADC_DAC_CONTROL_1, 1, dac_deemph_text);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0);
static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0);
static const DECLARE_TLV_DB_RANGE(boost_tlv,
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(13, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(20, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(29, 0, 0)
);
static const DECLARE_TLV_DB_SCALE(pga_tlv, -2325, 75, 0);
static const struct snd_kcontrol_new wm8961_snd_controls[] = {
SOC_DOUBLE_R_TLV("Headphone Volume", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME,
0, 127, 0, out_tlv),
SOC_DOUBLE_TLV("Headphone Secondary Volume", WM8961_ANALOGUE_HP_2,
6, 3, 7, 0, hp_sec_tlv),
SOC_DOUBLE_R("Headphone ZC Switch", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME,
7, 1, 0),
SOC_DOUBLE_R_TLV("Speaker Volume", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME,
0, 127, 0, out_tlv),
SOC_DOUBLE_R("Speaker ZC Switch", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME,
7, 1, 0),
SOC_SINGLE("Speaker AC Gain", WM8961_CLASS_D_CONTROL_2, 0, 7, 0),
SOC_SINGLE("DAC x128 OSR Switch", WM8961_ADC_DAC_CONTROL_2, 0, 1, 0),
SOC_ENUM("DAC Deemphasis", dac_deemph),
SOC_SINGLE("DAC Soft Mute Switch", WM8961_ADC_DAC_CONTROL_2, 3, 1, 0),
SOC_DOUBLE_R_TLV("Sidetone Volume", WM8961_DSP_SIDETONE_0,
WM8961_DSP_SIDETONE_1, 4, 12, 0, sidetone_tlv),
SOC_SINGLE("ADC High Pass Filter Switch", WM8961_ADC_DAC_CONTROL_1, 0, 1, 0),
SOC_ENUM("ADC High Pass Filter Mode", adc_hpf),
SOC_DOUBLE_R_TLV("Capture Volume",
WM8961_LEFT_ADC_VOLUME, WM8961_RIGHT_ADC_VOLUME,
1, 119, 0, adc_tlv),
SOC_DOUBLE_R_TLV("Capture Boost Volume",
WM8961_ADCL_SIGNAL_PATH, WM8961_ADCR_SIGNAL_PATH,
4, 3, 0, boost_tlv),
SOC_DOUBLE_R_TLV("Capture PGA Volume",
WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
0, 62, 0, pga_tlv),
SOC_DOUBLE_R("Capture PGA ZC Switch",
WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
6, 1, 1),
SOC_DOUBLE_R("Capture PGA Switch",
WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
7, 1, 1),
};
static const char *sidetone_text[] = {
"None", "Left", "Right"
};
static SOC_ENUM_SINGLE_DECL(dacl_sidetone,
WM8961_DSP_SIDETONE_0, 2, sidetone_text);
static SOC_ENUM_SINGLE_DECL(dacr_sidetone,
WM8961_DSP_SIDETONE_1, 2, sidetone_text);
static const struct snd_kcontrol_new dacl_mux =
SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone);
static const struct snd_kcontrol_new dacr_mux =
SOC_DAPM_ENUM("DACR Sidetone", dacr_sidetone);
static const struct snd_soc_dapm_widget wm8961_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINPUT"),
SND_SOC_DAPM_INPUT("RINPUT"),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8961_CLOCKING2, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Input", WM8961_PWR_MGMT_1, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Input", WM8961_PWR_MGMT_1, 4, 0, NULL, 0),
SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", WM8961_PWR_MGMT_1, 3, 0),
SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", WM8961_PWR_MGMT_1, 2, 0),
SND_SOC_DAPM_SUPPLY("MICBIAS", WM8961_PWR_MGMT_1, 1, 0, NULL, 0),
SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &dacl_mux),
SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &dacr_mux),
SND_SOC_DAPM_DAC("DACL", "HiFi Playback", WM8961_PWR_MGMT_2, 8, 0),
SND_SOC_DAPM_DAC("DACR", "HiFi Playback", WM8961_PWR_MGMT_2, 7, 0),
/* Handle as a mono path for DCS */
SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM,
4, 0, NULL, 0, wm8961_hp_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Speaker Output", SND_SOC_NOPM,
4, 0, NULL, 0, wm8961_spk_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_OUTPUT("HP_L"),
SND_SOC_DAPM_OUTPUT("HP_R"),
SND_SOC_DAPM_OUTPUT("SPK_LN"),
SND_SOC_DAPM_OUTPUT("SPK_LP"),
SND_SOC_DAPM_OUTPUT("SPK_RN"),
SND_SOC_DAPM_OUTPUT("SPK_RP"),
};
static const struct snd_soc_dapm_route audio_paths[] = {
{ "DACL", NULL, "CLK_DSP" },
{ "DACL", NULL, "DACL Sidetone" },
{ "DACR", NULL, "CLK_DSP" },
{ "DACR", NULL, "DACR Sidetone" },
{ "DACL Sidetone", "Left", "ADCL" },
{ "DACL Sidetone", "Right", "ADCR" },
{ "DACR Sidetone", "Left", "ADCL" },
{ "DACR Sidetone", "Right", "ADCR" },
{ "HP_L", NULL, "Headphone Output" },
{ "HP_R", NULL, "Headphone Output" },
{ "Headphone Output", NULL, "DACL" },
{ "Headphone Output", NULL, "DACR" },
{ "SPK_LN", NULL, "Speaker Output" },
{ "SPK_LP", NULL, "Speaker Output" },
{ "SPK_RN", NULL, "Speaker Output" },
{ "SPK_RP", NULL, "Speaker Output" },
{ "Speaker Output", NULL, "DACL" },
{ "Speaker Output", NULL, "DACR" },
{ "ADCL", NULL, "Left Input" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Input" },
{ "ADCR", NULL, "CLK_DSP" },
{ "Left Input", NULL, "LINPUT" },
{ "Right Input", NULL, "RINPUT" },
};
/* Values for CLK_SYS_RATE */
static struct {
int ratio;
u16 val;
} wm8961_clk_sys_ratio[] = {
{ 64, 0 },
{ 128, 1 },
{ 192, 2 },
{ 256, 3 },
{ 384, 4 },
{ 512, 5 },
{ 768, 6 },
{ 1024, 7 },
{ 1408, 8 },
{ 1536, 9 },
};
/* Values for SAMPLE_RATE */
static struct {
int rate;
u16 val;
} wm8961_srate[] = {
{ 48000, 0 },
{ 44100, 0 },
{ 32000, 1 },
{ 22050, 2 },
{ 24000, 2 },
{ 16000, 3 },
{ 11250, 4 },
{ 12000, 4 },
{ 8000, 5 },
};
static int wm8961_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct wm8961_priv *wm8961 = snd_soc_component_get_drvdata(component);
int i, best, target, fs;
u16 reg;
fs = params_rate(params);
if (!wm8961->sysclk) {
dev_err(component->dev, "MCLK has not been specified\n");
return -EINVAL;
}
/* Find the closest sample rate for the filters */
best = 0;
for (i = 0; i < ARRAY_SIZE(wm8961_srate); i++) {
if (abs(wm8961_srate[i].rate - fs) <
abs(wm8961_srate[best].rate - fs))
best = i;
}
reg = snd_soc_component_read32(component, WM8961_ADDITIONAL_CONTROL_3);
reg &= ~WM8961_SAMPLE_RATE_MASK;
reg |= wm8961_srate[best].val;
snd_soc_component_write(component, WM8961_ADDITIONAL_CONTROL_3, reg);
dev_dbg(component->dev, "Selected SRATE %dHz for %dHz\n",
wm8961_srate[best].rate, fs);
/* Select a CLK_SYS/fs ratio equal to or higher than required */
target = wm8961->sysclk / fs;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && target < 64) {
dev_err(component->dev,
"SYSCLK must be at least 64*fs for DAC\n");
return -EINVAL;
}
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && target < 256) {
dev_err(component->dev,
"SYSCLK must be at least 256*fs for ADC\n");
return -EINVAL;
}
for (i = 0; i < ARRAY_SIZE(wm8961_clk_sys_ratio); i++) {
if (wm8961_clk_sys_ratio[i].ratio >= target)
break;
}
if (i == ARRAY_SIZE(wm8961_clk_sys_ratio)) {
dev_err(component->dev, "Unable to generate CLK_SYS_RATE\n");
return -EINVAL;
}
dev_dbg(component->dev, "Selected CLK_SYS_RATE of %d for %d/%d=%d\n",
wm8961_clk_sys_ratio[i].ratio, wm8961->sysclk, fs,
wm8961->sysclk / fs);
reg = snd_soc_component_read32(component, WM8961_CLOCKING_4);
reg &= ~WM8961_CLK_SYS_RATE_MASK;
reg |= wm8961_clk_sys_ratio[i].val << WM8961_CLK_SYS_RATE_SHIFT;
snd_soc_component_write(component, WM8961_CLOCKING_4, reg);
reg = snd_soc_component_read32(component, WM8961_AUDIO_INTERFACE_0);
reg &= ~WM8961_WL_MASK;
switch (params_width(params)) {
case 16:
break;
case 20:
reg |= 1 << WM8961_WL_SHIFT;
break;
case 24:
reg |= 2 << WM8961_WL_SHIFT;
break;
case 32:
reg |= 3 << WM8961_WL_SHIFT;
break;
default:
return -EINVAL;
}
snd_soc_component_write(component, WM8961_AUDIO_INTERFACE_0, reg);
/* Sloping stop-band filter is recommended for <= 24kHz */
reg = snd_soc_component_read32(component, WM8961_ADC_DAC_CONTROL_2);
if (fs <= 24000)
reg |= WM8961_DACSLOPE;
else
reg &= ~WM8961_DACSLOPE;
snd_soc_component_write(component, WM8961_ADC_DAC_CONTROL_2, reg);
return 0;
}
static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq,
int dir)
{
struct snd_soc_component *component = dai->component;
struct wm8961_priv *wm8961 = snd_soc_component_get_drvdata(component);
u16 reg = snd_soc_component_read32(component, WM8961_CLOCKING1);
if (freq > 33000000) {
dev_err(component->dev, "MCLK must be <33MHz\n");
return -EINVAL;
}
if (freq > 16500000) {
dev_dbg(component->dev, "Using MCLK/2 for %dHz MCLK\n", freq);
reg |= WM8961_MCLKDIV;
freq /= 2;
} else {
dev_dbg(component->dev, "Using MCLK/1 for %dHz MCLK\n", freq);
reg &= ~WM8961_MCLKDIV;
}
snd_soc_component_write(component, WM8961_CLOCKING1, reg);
wm8961->sysclk = freq;
return 0;
}
static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_component *component = dai->component;
u16 aif = snd_soc_component_read32(component, WM8961_AUDIO_INTERFACE_0);
aif &= ~(WM8961_BCLKINV | WM8961_LRP |
WM8961_MS | WM8961_FORMAT_MASK);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
aif |= WM8961_MS;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
aif |= 1;
break;
case SND_SOC_DAIFMT_I2S:
aif |= 2;
break;
case SND_SOC_DAIFMT_DSP_B:
aif |= WM8961_LRP;
/* fall through */
case SND_SOC_DAIFMT_DSP_A:
aif |= 3;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
case SND_SOC_DAIFMT_IB_NF:
break;
default:
return -EINVAL;
}
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_NB_IF:
aif |= WM8961_LRP;
break;
case SND_SOC_DAIFMT_IB_NF:
aif |= WM8961_BCLKINV;
break;
case SND_SOC_DAIFMT_IB_IF:
aif |= WM8961_BCLKINV | WM8961_LRP;
break;
default:
return -EINVAL;
}
return snd_soc_component_write(component, WM8961_AUDIO_INTERFACE_0, aif);
}
static int wm8961_set_tristate(struct snd_soc_dai *dai, int tristate)
{
struct snd_soc_component *component = dai->component;
u16 reg = snd_soc_component_read32(component, WM8961_ADDITIONAL_CONTROL_2);
if (tristate)
reg |= WM8961_TRIS;
else
reg &= ~WM8961_TRIS;
return snd_soc_component_write(component, WM8961_ADDITIONAL_CONTROL_2, reg);
}
static int wm8961_digital_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_component *component = dai->component;
u16 reg = snd_soc_component_read32(component, WM8961_ADC_DAC_CONTROL_1);
if (mute)
reg |= WM8961_DACMU;
else
reg &= ~WM8961_DACMU;
msleep(17);
return snd_soc_component_write(component, WM8961_ADC_DAC_CONTROL_1, reg);
}
static int wm8961_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
{
struct snd_soc_component *component = dai->component;
u16 reg;
switch (div_id) {
case WM8961_BCLK:
reg = snd_soc_component_read32(component, WM8961_CLOCKING2);
reg &= ~WM8961_BCLKDIV_MASK;
reg |= div;
snd_soc_component_write(component, WM8961_CLOCKING2, reg);
break;
case WM8961_LRCLK:
reg = snd_soc_component_read32(component, WM8961_AUDIO_INTERFACE_2);
reg &= ~WM8961_LRCLK_RATE_MASK;
reg |= div;
snd_soc_component_write(component, WM8961_AUDIO_INTERFACE_2, reg);
break;
default:
return -EINVAL;
}
return 0;
}
static int wm8961_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
u16 reg;
/* This is all slightly unusual since we have no bypass paths
* and the output amplifier structure means we can just slam
* the biases straight up rather than having to ramp them
* slowly.
*/
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_STANDBY) {
/* Enable bias generation */
reg = snd_soc_component_read32(component, WM8961_ANTI_POP);
reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
snd_soc_component_write(component, WM8961_ANTI_POP, reg);
/* VMID=2*50k, VREF */
reg = snd_soc_component_read32(component, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VMIDSEL_MASK;
reg |= (1 << WM8961_VMIDSEL_SHIFT) | WM8961_VREF;
snd_soc_component_write(component, WM8961_PWR_MGMT_1, reg);
}
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_component_get_bias_level(component) == SND_SOC_BIAS_PREPARE) {
/* VREF off */
reg = snd_soc_component_read32(component, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VREF;
snd_soc_component_write(component, WM8961_PWR_MGMT_1, reg);
/* Bias generation off */
reg = snd_soc_component_read32(component, WM8961_ANTI_POP);
reg &= ~(WM8961_BUFIOEN | WM8961_BUFDCOPEN);
snd_soc_component_write(component, WM8961_ANTI_POP, reg);
/* VMID off */
reg = snd_soc_component_read32(component, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VMIDSEL_MASK;
snd_soc_component_write(component, WM8961_PWR_MGMT_1, reg);
}
break;
case SND_SOC_BIAS_OFF:
break;
}
return 0;
}
#define WM8961_RATES SNDRV_PCM_RATE_8000_48000
#define WM8961_FORMATS \
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops wm8961_dai_ops = {
.hw_params = wm8961_hw_params,
.set_sysclk = wm8961_set_sysclk,
.set_fmt = wm8961_set_fmt,
.digital_mute = wm8961_digital_mute,
.set_tristate = wm8961_set_tristate,
.set_clkdiv = wm8961_set_clkdiv,
};
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
static struct snd_soc_dai_driver wm8961_dai = {
.name = "wm8961-hifi",
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8961_RATES,
.formats = WM8961_FORMATS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8961_RATES,
.formats = WM8961_FORMATS,},
.ops = &wm8961_dai_ops,
};
static int wm8961_probe(struct snd_soc_component *component)
{
u16 reg;
/* Enable class W */
reg = snd_soc_component_read32(component, WM8961_CHARGE_PUMP_B);
reg |= WM8961_CP_DYN_PWR_MASK;
snd_soc_component_write(component, WM8961_CHARGE_PUMP_B, reg);
/* Latch volume update bits (right channel only, we always
* write both out) and default ZC on. */
reg = snd_soc_component_read32(component, WM8961_ROUT1_VOLUME);
snd_soc_component_write(component, WM8961_ROUT1_VOLUME,
reg | WM8961_LO1ZC | WM8961_OUT1VU);
snd_soc_component_write(component, WM8961_LOUT1_VOLUME, reg | WM8961_LO1ZC);
reg = snd_soc_component_read32(component, WM8961_ROUT2_VOLUME);
snd_soc_component_write(component, WM8961_ROUT2_VOLUME,
reg | WM8961_SPKRZC | WM8961_SPKVU);
snd_soc_component_write(component, WM8961_LOUT2_VOLUME, reg | WM8961_SPKLZC);
reg = snd_soc_component_read32(component, WM8961_RIGHT_ADC_VOLUME);
snd_soc_component_write(component, WM8961_RIGHT_ADC_VOLUME, reg | WM8961_ADCVU);
reg = snd_soc_component_read32(component, WM8961_RIGHT_INPUT_VOLUME);
snd_soc_component_write(component, WM8961_RIGHT_INPUT_VOLUME, reg | WM8961_IPVU);
/* Use soft mute by default */
reg = snd_soc_component_read32(component, WM8961_ADC_DAC_CONTROL_2);
reg |= WM8961_DACSMM;
snd_soc_component_write(component, WM8961_ADC_DAC_CONTROL_2, reg);
/* Use automatic clocking mode by default; for now this is all
* we support.
*/
reg = snd_soc_component_read32(component, WM8961_CLOCKING_3);
reg &= ~WM8961_MANUAL_MODE;
snd_soc_component_write(component, WM8961_CLOCKING_3, reg);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
#ifdef CONFIG_PM
static int wm8961_resume(struct snd_soc_component *component)
{
snd_soc_component_cache_sync(component);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
return 0;
}
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
#else
#define wm8961_resume NULL
#endif
static const struct snd_soc_component_driver soc_component_dev_wm8961 = {
.probe = wm8961_probe,
.resume = wm8961_resume,
.set_bias_level = wm8961_set_bias_level,
.controls = wm8961_snd_controls,
.num_controls = ARRAY_SIZE(wm8961_snd_controls),
.dapm_widgets = wm8961_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8961_dapm_widgets),
.dapm_routes = audio_paths,
.num_dapm_routes = ARRAY_SIZE(audio_paths),
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 1,
};
static const struct regmap_config wm8961_regmap = {
.reg_bits = 8,
.val_bits = 16,
.max_register = WM8961_MAX_REGISTER,
.reg_defaults = wm8961_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm8961_reg_defaults),
.cache_type = REGCACHE_RBTREE,
.volatile_reg = wm8961_volatile,
.readable_reg = wm8961_readable,
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
};
static int wm8961_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8961_priv *wm8961;
unsigned int val;
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
int ret;
wm8961 = devm_kzalloc(&i2c->dev, sizeof(struct wm8961_priv),
GFP_KERNEL);
if (wm8961 == NULL)
return -ENOMEM;
wm8961->regmap = devm_regmap_init_i2c(i2c, &wm8961_regmap);
if (IS_ERR(wm8961->regmap))
return PTR_ERR(wm8961->regmap);
ret = regmap_read(wm8961->regmap, WM8961_SOFTWARE_RESET, &val);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret);
return ret;
}
if (val != 0x1801) {
dev_err(&i2c->dev, "Device is not a WM8961: ID=0x%x\n", val);
return -EINVAL;
}
/* This isn't volatile - readback doesn't correspond to write */
regcache_cache_bypass(wm8961->regmap, true);
ret = regmap_read(wm8961->regmap, WM8961_RIGHT_INPUT_VOLUME, &val);
regcache_cache_bypass(wm8961->regmap, false);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to read chip revision: %d\n", ret);
return ret;
}
dev_info(&i2c->dev, "WM8961 family %d revision %c\n",
(val & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT,
((val & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT)
+ 'A');
ret = regmap_write(wm8961->regmap, WM8961_SOFTWARE_RESET, 0x1801);
if (ret != 0) {
dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret);
return ret;
}
i2c_set_clientdata(i2c, wm8961);
ret = devm_snd_soc_register_component(&i2c->dev,
&soc_component_dev_wm8961, &wm8961_dai, 1);
ASoC: multi-component - ASoC Multi-Component Support This patch extends the ASoC API to allow sound cards to have more than one CODEC and more than one platform DMA controller. This is achieved by dividing some current ASoC structures that contain both driver data and device data into structures that only either contain device data or driver data. i.e. struct snd_soc_codec ---> struct snd_soc_codec (device data) +-> struct snd_soc_codec_driver (driver data) struct snd_soc_platform ---> struct snd_soc_platform (device data) +-> struct snd_soc_platform_driver (driver data) struct snd_soc_dai ---> struct snd_soc_dai (device data) +-> struct snd_soc_dai_driver (driver data) struct snd_soc_device ---> deleted This now allows ASoC to be more tightly aligned with the Linux driver model and also means that every ASoC codec, platform and (platform) DAI is a kernel device. ASoC component private data is now stored as device private data. The ASoC sound card struct snd_soc_card has also been updated to store lists of it's components rather than a pointer to a codec and platform. The PCM runtime struct soc_pcm_runtime now has pointers to all its components. This patch adds DAPM support for ASoC multi-component and removes struct snd_soc_socdev from DAPM core. All DAPM calls are now made on a card, codec or runtime PCM level basis rather than using snd_soc_socdev. Other notable multi-component changes:- * Stream operations now de-reference less structures. * close_delayed work() now runs on a DAI basis rather than looping all DAIs in a card. * PM suspend()/resume() operations can now handle N CODECs and Platforms per sound card. * Added soc_bind_dai_link() to bind the component devices to the sound card. * Added soc_dai_link_probe() and soc_dai_link_remove() to probe and remove DAI link components. * sysfs entries can now be registered per component per card. * snd_soc_new_pcms() functionailty rolled into dai_link_probe(). * snd_soc_register_codec() now does all the codec list and mutex init. This patch changes the probe() and remove() of the CODEC drivers as follows:- o Make CODEC driver a platform driver o Moved all struct snd_soc_codec list, mutex, etc initialiasation to core. o Removed all static codec pointers (drivers now support > 1 codec dev) o snd_soc_register_pcms() now done by core. o snd_soc_register_dai() folded into snd_soc_register_codec(). CS4270 portions: Acked-by: Timur Tabi <timur@freescale.com> Some TLV320aic23 and Cirrus platform fixes. Signed-off-by: Ryan Mallon <ryan@bluewatersys.com> TI CODEC and OMAP fixes Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Samsung platform and misc fixes :- Signed-off-by: Chanwoo Choi <cw00.choi@samsung.com> Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com> Signed-off-by: Kyungmin Park <kyungmin.park@samsung.com> Reviewed-by: Jassi Brar <jassi.brar@samsung.com> Signed-off-by: Seungwhan Youn <sw.youn@samsung.com> MPC8610 and PPC fixes. Signed-off-by: Timur Tabi <timur@freescale.com> i.MX fixes and some core fixes. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> J4740 platform fixes:- Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> CC: Tony Lindgren <tony@atomide.com> CC: Nicolas Ferre <nicolas.ferre@atmel.com> CC: Kevin Hilman <khilman@deeprootsystems.com> CC: Sascha Hauer <s.hauer@pengutronix.de> CC: Atsushi Nemoto <anemo@mba.ocn.ne.jp> CC: Kuninori Morimoto <morimoto.kuninori@renesas.com> CC: Daniel Gloeckner <dg@emlix.com> CC: Manuel Lauss <mano@roarinelk.homelinux.net> CC: Mike Frysinger <vapier.adi@gmail.com> CC: Arnaud Patard <apatard@mandriva.com> CC: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-03-18 03:15:21 +07:00
return ret;
}
static const struct i2c_device_id wm8961_i2c_id[] = {
{ "wm8961", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id);
static struct i2c_driver wm8961_i2c_driver = {
.driver = {
.name = "wm8961",
},
.probe = wm8961_i2c_probe,
.id_table = wm8961_i2c_id,
};
module_i2c_driver(wm8961_i2c_driver);
MODULE_DESCRIPTION("ASoC WM8961 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");