linux_dsm_epyc7002/sound/soc/codecs/ak4671.c

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/*
* ak4671.c -- audio driver for AK4671
*
* Copyright (C) 2009 Samsung Electronics Co.Ltd
* Author: Joonyoung Shim <jy0922.shim@samsung.com>
*
* This program is free software; you can redistribute it and/or modify it
* under the terms of the GNU General Public License as published by the
* Free Software Foundation; either version 2 of the License, or (at your
* option) any later version.
*
*/
#include <linux/module.h>
#include <linux/init.h>
#include <linux/i2c.h>
#include <linux/delay.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/soc.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "ak4671.h"
/* ak4671 register cache & default register settings */
static const struct reg_default ak4671_reg_defaults[] = {
{ 0x00, 0x00 }, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */
{ 0x01, 0xf6 }, /* AK4671_PLL_MODE_SELECT0 (0x01) */
{ 0x02, 0x00 }, /* AK4671_PLL_MODE_SELECT1 (0x02) */
{ 0x03, 0x02 }, /* AK4671_FORMAT_SELECT (0x03) */
{ 0x04, 0x00 }, /* AK4671_MIC_SIGNAL_SELECT (0x04) */
{ 0x05, 0x55 }, /* AK4671_MIC_AMP_GAIN (0x05) */
{ 0x06, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */
{ 0x07, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */
{ 0x08, 0xb5 }, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */
{ 0x09, 0x00 }, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */
{ 0x0a, 0x00 }, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */
{ 0x0b, 0x00 }, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */
{ 0x0c, 0x00 }, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */
{ 0x0d, 0x00 }, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */
{ 0x0e, 0x00 }, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */
{ 0x0f, 0x00 }, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */
{ 0x10, 0x00 }, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */
{ 0x11, 0x80 }, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */
{ 0x12, 0x91 }, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */
{ 0x13, 0x91 }, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */
{ 0x14, 0xe1 }, /* AK4671_ALC_REFERENCE_SELECT (0x14) */
{ 0x15, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */
{ 0x16, 0x00 }, /* AK4671_ALC_TIMER_SELECT (0x16) */
{ 0x17, 0x00 }, /* AK4671_ALC_MODE_CONTROL (0x17) */
{ 0x18, 0x02 }, /* AK4671_MODE_CONTROL1 (0x18) */
{ 0x19, 0x01 }, /* AK4671_MODE_CONTROL2 (0x19) */
{ 0x1a, 0x18 }, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */
{ 0x1b, 0x18 }, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */
{ 0x1c, 0x00 }, /* AK4671_SIDETONE_A_CONTROL (0x1c) */
{ 0x1d, 0x02 }, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */
{ 0x1e, 0x00 }, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */
{ 0x1f, 0x00 }, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */
{ 0x20, 0x00 }, /* AK4671_FIL3_COEFFICIENT2 (0x20) */
{ 0x21, 0x00 }, /* AK4671_FIL3_COEFFICIENT3 (0x21) */
{ 0x22, 0x00 }, /* AK4671_EQ_COEFFICIENT0 (0x22) */
{ 0x23, 0x00 }, /* AK4671_EQ_COEFFICIENT1 (0x23) */
{ 0x24, 0x00 }, /* AK4671_EQ_COEFFICIENT2 (0x24) */
{ 0x25, 0x00 }, /* AK4671_EQ_COEFFICIENT3 (0x25) */
{ 0x26, 0x00 }, /* AK4671_EQ_COEFFICIENT4 (0x26) */
{ 0x27, 0x00 }, /* AK4671_EQ_COEFFICIENT5 (0x27) */
{ 0x28, 0xa9 }, /* AK4671_FIL1_COEFFICIENT0 (0x28) */
{ 0x29, 0x1f }, /* AK4671_FIL1_COEFFICIENT1 (0x29) */
{ 0x2a, 0xad }, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */
{ 0x2b, 0x20 }, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */
{ 0x2c, 0x00 }, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */
{ 0x2d, 0x00 }, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */
{ 0x2e, 0x00 }, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */
{ 0x2f, 0x00 }, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */
{ 0x30, 0x00 }, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */
{ 0x32, 0x00 }, /* AK4671_E1_COEFFICIENT0 (0x32) */
{ 0x33, 0x00 }, /* AK4671_E1_COEFFICIENT1 (0x33) */
{ 0x34, 0x00 }, /* AK4671_E1_COEFFICIENT2 (0x34) */
{ 0x35, 0x00 }, /* AK4671_E1_COEFFICIENT3 (0x35) */
{ 0x36, 0x00 }, /* AK4671_E1_COEFFICIENT4 (0x36) */
{ 0x37, 0x00 }, /* AK4671_E1_COEFFICIENT5 (0x37) */
{ 0x38, 0x00 }, /* AK4671_E2_COEFFICIENT0 (0x38) */
{ 0x39, 0x00 }, /* AK4671_E2_COEFFICIENT1 (0x39) */
{ 0x3a, 0x00 }, /* AK4671_E2_COEFFICIENT2 (0x3a) */
{ 0x3b, 0x00 }, /* AK4671_E2_COEFFICIENT3 (0x3b) */
{ 0x3c, 0x00 }, /* AK4671_E2_COEFFICIENT4 (0x3c) */
{ 0x3d, 0x00 }, /* AK4671_E2_COEFFICIENT5 (0x3d) */
{ 0x3e, 0x00 }, /* AK4671_E3_COEFFICIENT0 (0x3e) */
{ 0x3f, 0x00 }, /* AK4671_E3_COEFFICIENT1 (0x3f) */
{ 0x40, 0x00 }, /* AK4671_E3_COEFFICIENT2 (0x40) */
{ 0x41, 0x00 }, /* AK4671_E3_COEFFICIENT3 (0x41) */
{ 0x42, 0x00 }, /* AK4671_E3_COEFFICIENT4 (0x42) */
{ 0x43, 0x00 }, /* AK4671_E3_COEFFICIENT5 (0x43) */
{ 0x44, 0x00 }, /* AK4671_E4_COEFFICIENT0 (0x44) */
{ 0x45, 0x00 }, /* AK4671_E4_COEFFICIENT1 (0x45) */
{ 0x46, 0x00 }, /* AK4671_E4_COEFFICIENT2 (0x46) */
{ 0x47, 0x00 }, /* AK4671_E4_COEFFICIENT3 (0x47) */
{ 0x48, 0x00 }, /* AK4671_E4_COEFFICIENT4 (0x48) */
{ 0x49, 0x00 }, /* AK4671_E4_COEFFICIENT5 (0x49) */
{ 0x4a, 0x00 }, /* AK4671_E5_COEFFICIENT0 (0x4a) */
{ 0x4b, 0x00 }, /* AK4671_E5_COEFFICIENT1 (0x4b) */
{ 0x4c, 0x00 }, /* AK4671_E5_COEFFICIENT2 (0x4c) */
{ 0x4d, 0x00 }, /* AK4671_E5_COEFFICIENT3 (0x4d) */
{ 0x4e, 0x00 }, /* AK4671_E5_COEFFICIENT4 (0x4e) */
{ 0x4f, 0x00 }, /* AK4671_E5_COEFFICIENT5 (0x4f) */
{ 0x50, 0x88 }, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */
{ 0x51, 0x88 }, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */
{ 0x52, 0x08 }, /* AK4671_EQ_CONTRO_10KHZ (0x52) */
{ 0x53, 0x00 }, /* AK4671_PCM_IF_CONTROL0 (0x53) */
{ 0x54, 0x00 }, /* AK4671_PCM_IF_CONTROL1 (0x54) */
{ 0x55, 0x00 }, /* AK4671_PCM_IF_CONTROL2 (0x55) */
{ 0x56, 0x18 }, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */
{ 0x57, 0x18 }, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */
{ 0x58, 0x00 }, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */
{ 0x59, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */
{ 0x5a, 0x00 }, /* AK4671_SAR_ADC_CONTROL (0x5a) */
};
/*
* LOUT1/ROUT1 output volume control:
* from -24 to 6 dB in 6 dB steps (mute instead of -30 dB)
*/
static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1);
/*
* LOUT2/ROUT2 output volume control:
* from -33 to 6 dB in 3 dB steps (mute instead of -33 dB)
*/
static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1);
/*
* LOUT3/ROUT3 output volume control:
* from -6 to 3 dB in 3 dB steps
*/
static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0);
/*
* Mic amp gain control:
* from -15 to 30 dB in 3 dB steps
* REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not
* available
*/
static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0);
static const struct snd_kcontrol_new ak4671_snd_controls[] = {
/* Common playback gain controls */
SOC_SINGLE_TLV("Line Output1 Playback Volume",
AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv),
SOC_SINGLE_TLV("Headphone Output2 Playback Volume",
AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv),
SOC_SINGLE_TLV("Line Output3 Playback Volume",
AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv),
/* Common capture gain controls */
SOC_DOUBLE_TLV("Mic Amp Capture Volume",
AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv),
};
/* event handlers */
static int ak4671_out2_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
switch (event) {
case SND_SOC_DAPM_POST_PMU:
snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT,
AK4671_MUTEN, AK4671_MUTEN);
break;
case SND_SOC_DAPM_PRE_PMD:
snd_soc_update_bits(codec, AK4671_LOUT2_POWER_MANAGERMENT,
AK4671_MUTEN, 0);
break;
}
return 0;
}
/* Output Mixers */
static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = {
SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0),
SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0),
SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0),
SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0),
SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0),
SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0),
};
static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = {
SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0),
SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0),
SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0),
SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0),
SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0),
SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0),
};
static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = {
SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0),
SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0),
SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0),
SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0),
SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0),
SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0),
};
static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = {
SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0),
SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0),
SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0),
SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0),
SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0),
SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0),
};
static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = {
SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0),
SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0),
SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0),
SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0),
SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0),
SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0),
};
static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = {
SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0),
SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0),
SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0),
SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0),
SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0),
SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0),
};
/* Input MUXs */
static const char *ak4671_lin_mux_texts[] =
{"LIN1", "LIN2", "LIN3", "LIN4"};
static SOC_ENUM_SINGLE_DECL(ak4671_lin_mux_enum,
AK4671_MIC_SIGNAL_SELECT, 0,
ak4671_lin_mux_texts);
static const struct snd_kcontrol_new ak4671_lin_mux_control =
SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum);
static const char *ak4671_rin_mux_texts[] =
{"RIN1", "RIN2", "RIN3", "RIN4"};
static SOC_ENUM_SINGLE_DECL(ak4671_rin_mux_enum,
AK4671_MIC_SIGNAL_SELECT, 2,
ak4671_rin_mux_texts);
static const struct snd_kcontrol_new ak4671_rin_mux_control =
SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum);
static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = {
/* Inputs */
SND_SOC_DAPM_INPUT("LIN1"),
SND_SOC_DAPM_INPUT("RIN1"),
SND_SOC_DAPM_INPUT("LIN2"),
SND_SOC_DAPM_INPUT("RIN2"),
SND_SOC_DAPM_INPUT("LIN3"),
SND_SOC_DAPM_INPUT("RIN3"),
SND_SOC_DAPM_INPUT("LIN4"),
SND_SOC_DAPM_INPUT("RIN4"),
/* Outputs */
SND_SOC_DAPM_OUTPUT("LOUT1"),
SND_SOC_DAPM_OUTPUT("ROUT1"),
SND_SOC_DAPM_OUTPUT("LOUT2"),
SND_SOC_DAPM_OUTPUT("ROUT2"),
SND_SOC_DAPM_OUTPUT("LOUT3"),
SND_SOC_DAPM_OUTPUT("ROUT3"),
/* DAC */
SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback",
AK4671_AD_DA_POWER_MANAGEMENT, 6, 0),
SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback",
AK4671_AD_DA_POWER_MANAGEMENT, 7, 0),
/* ADC */
SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture",
AK4671_AD_DA_POWER_MANAGEMENT, 4, 0),
SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture",
AK4671_AD_DA_POWER_MANAGEMENT, 5, 0),
/* PGA */
SND_SOC_DAPM_PGA("LOUT2 Mix Amp",
AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("ROUT2 Mix Amp",
AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("LIN1 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0),
SND_SOC_DAPM_PGA("RIN1 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0),
SND_SOC_DAPM_PGA("LIN2 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0),
SND_SOC_DAPM_PGA("RIN2 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0),
SND_SOC_DAPM_PGA("LIN3 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("RIN3 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("LIN4 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0),
SND_SOC_DAPM_PGA("RIN4 Mixing Circuit",
AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0),
/* Output Mixers */
SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0,
&ak4671_lout1_mixer_controls[0],
ARRAY_SIZE(ak4671_lout1_mixer_controls)),
SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0,
&ak4671_rout1_mixer_controls[0],
ARRAY_SIZE(ak4671_rout1_mixer_controls)),
SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
0, 0, &ak4671_lout2_mixer_controls[0],
ARRAY_SIZE(ak4671_lout2_mixer_controls),
ak4671_out2_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT,
1, 0, &ak4671_rout2_mixer_controls[0],
ARRAY_SIZE(ak4671_rout2_mixer_controls),
ak4671_out2_event,
SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0,
&ak4671_lout3_mixer_controls[0],
ARRAY_SIZE(ak4671_lout3_mixer_controls)),
SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0,
&ak4671_rout3_mixer_controls[0],
ARRAY_SIZE(ak4671_rout3_mixer_controls)),
/* Input MUXs */
SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0,
&ak4671_lin_mux_control),
SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0,
&ak4671_rin_mux_control),
/* Mic Power */
SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0),
/* Supply */
SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0),
};
static const struct snd_soc_dapm_route ak4671_intercon[] = {
{"DAC Left", "NULL", "PMPLL"},
{"DAC Right", "NULL", "PMPLL"},
{"ADC Left", "NULL", "PMPLL"},
{"ADC Right", "NULL", "PMPLL"},
/* Outputs */
{"LOUT1", "NULL", "LOUT1 Mixer"},
{"ROUT1", "NULL", "ROUT1 Mixer"},
{"LOUT2", "NULL", "LOUT2 Mix Amp"},
{"ROUT2", "NULL", "ROUT2 Mix Amp"},
{"LOUT3", "NULL", "LOUT3 Mixer"},
{"ROUT3", "NULL", "ROUT3 Mixer"},
{"LOUT1 Mixer", "DACL", "DAC Left"},
{"ROUT1 Mixer", "DACR", "DAC Right"},
{"LOUT2 Mixer", "DACHL", "DAC Left"},
{"ROUT2 Mixer", "DACHR", "DAC Right"},
{"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"},
{"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"},
{"LOUT3 Mixer", "DACSL", "DAC Left"},
{"ROUT3 Mixer", "DACSR", "DAC Right"},
/* Inputs */
{"LIN MUX", "LIN1", "LIN1"},
{"LIN MUX", "LIN2", "LIN2"},
{"LIN MUX", "LIN3", "LIN3"},
{"LIN MUX", "LIN4", "LIN4"},
{"RIN MUX", "RIN1", "RIN1"},
{"RIN MUX", "RIN2", "RIN2"},
{"RIN MUX", "RIN3", "RIN3"},
{"RIN MUX", "RIN4", "RIN4"},
{"LIN1", NULL, "Mic Bias"},
{"RIN1", NULL, "Mic Bias"},
{"LIN2", NULL, "Mic Bias"},
{"RIN2", NULL, "Mic Bias"},
{"ADC Left", "NULL", "LIN MUX"},
{"ADC Right", "NULL", "RIN MUX"},
/* Analog Loops */
{"LIN1 Mixing Circuit", "NULL", "LIN1"},
{"RIN1 Mixing Circuit", "NULL", "RIN1"},
{"LIN2 Mixing Circuit", "NULL", "LIN2"},
{"RIN2 Mixing Circuit", "NULL", "RIN2"},
{"LIN3 Mixing Circuit", "NULL", "LIN3"},
{"RIN3 Mixing Circuit", "NULL", "RIN3"},
{"LIN4 Mixing Circuit", "NULL", "LIN4"},
{"RIN4 Mixing Circuit", "NULL", "RIN4"},
{"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"},
{"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"},
{"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"},
{"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"},
{"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"},
{"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"},
{"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"},
{"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"},
{"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"},
{"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"},
{"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"},
{"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"},
{"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"},
{"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"},
{"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"},
{"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"},
{"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"},
{"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"},
{"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"},
{"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"},
{"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"},
{"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"},
{"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"},
{"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"},
};
static int ak4671_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
u8 fs;
fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
fs &= ~AK4671_FS;
switch (params_rate(params)) {
case 8000:
fs |= AK4671_FS_8KHZ;
break;
case 12000:
fs |= AK4671_FS_12KHZ;
break;
case 16000:
fs |= AK4671_FS_16KHZ;
break;
case 24000:
fs |= AK4671_FS_24KHZ;
break;
case 11025:
fs |= AK4671_FS_11_025KHZ;
break;
case 22050:
fs |= AK4671_FS_22_05KHZ;
break;
case 32000:
fs |= AK4671_FS_32KHZ;
break;
case 44100:
fs |= AK4671_FS_44_1KHZ;
break;
case 48000:
fs |= AK4671_FS_48KHZ;
break;
default:
return -EINVAL;
}
snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs);
return 0;
}
static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq, int dir)
{
struct snd_soc_codec *codec = dai->codec;
u8 pll;
pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0);
pll &= ~AK4671_PLL;
switch (freq) {
case 11289600:
pll |= AK4671_PLL_11_2896MHZ;
break;
case 12000000:
pll |= AK4671_PLL_12MHZ;
break;
case 12288000:
pll |= AK4671_PLL_12_288MHZ;
break;
case 13000000:
pll |= AK4671_PLL_13MHZ;
break;
case 13500000:
pll |= AK4671_PLL_13_5MHZ;
break;
case 19200000:
pll |= AK4671_PLL_19_2MHZ;
break;
case 24000000:
pll |= AK4671_PLL_24MHZ;
break;
case 26000000:
pll |= AK4671_PLL_26MHZ;
break;
case 27000000:
pll |= AK4671_PLL_27MHZ;
break;
default:
return -EINVAL;
}
snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll);
return 0;
}
static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
u8 mode;
u8 format;
/* set master/slave audio interface */
mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
mode |= AK4671_M_S;
break;
case SND_SOC_DAIFMT_CBM_CFS:
mode &= ~(AK4671_M_S);
break;
default:
return -EINVAL;
}
/* interface format */
format = snd_soc_read(codec, AK4671_FORMAT_SELECT);
format &= ~AK4671_DIF;
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
format |= AK4671_DIF_I2S_MODE;
break;
case SND_SOC_DAIFMT_LEFT_J:
format |= AK4671_DIF_MSB_MODE;
break;
case SND_SOC_DAIFMT_DSP_A:
format |= AK4671_DIF_DSP_MODE;
format |= AK4671_BCKP;
format |= AK4671_MSBS;
break;
default:
return -EINVAL;
}
/* set mode and format */
snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode);
snd_soc_write(codec, AK4671_FORMAT_SELECT, format);
return 0;
}
static int ak4671_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
case SND_SOC_BIAS_STANDBY:
snd_soc_update_bits(codec, AK4671_AD_DA_POWER_MANAGEMENT,
AK4671_PMVCM, AK4671_PMVCM);
break;
case SND_SOC_BIAS_OFF:
snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00);
break;
}
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 20:53:46 +07:00
codec->dapm.bias_level = level;
return 0;
}
#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000)
#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE
static const struct snd_soc_dai_ops ak4671_dai_ops = {
.hw_params = ak4671_hw_params,
.set_sysclk = ak4671_set_dai_sysclk,
.set_fmt = ak4671_set_dai_fmt,
};
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 ak4671_dai = {
.name = "ak4671-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 1,
.channels_max = 2,
.rates = AK4671_RATES,
.formats = AK4671_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 1,
.channels_max = 2,
.rates = AK4671_RATES,
.formats = AK4671_FORMATS,},
.ops = &ak4671_dai_ops,
};
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_codec_driver soc_codec_dev_ak4671 = {
.set_bias_level = ak4671_set_bias_level,
.controls = ak4671_snd_controls,
.num_controls = ARRAY_SIZE(ak4671_snd_controls),
.dapm_widgets = ak4671_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(ak4671_dapm_widgets),
.dapm_routes = ak4671_intercon,
.num_dapm_routes = ARRAY_SIZE(ak4671_intercon),
};
static const struct regmap_config ak4671_regmap = {
.reg_bits = 8,
.val_bits = 8,
.max_register = AK4671_SAR_ADC_CONTROL,
.reg_defaults = ak4671_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(ak4671_reg_defaults),
.cache_type = REGCACHE_RBTREE,
};
static int ak4671_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
struct regmap *regmap;
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;
regmap = devm_regmap_init_i2c(client, &ak4671_regmap);
if (IS_ERR(regmap)) {
ret = PTR_ERR(regmap);
dev_err(&client->dev, "Failed to create regmap: %d\n", ret);
return ret;
}
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
ret = snd_soc_register_codec(&client->dev,
&soc_codec_dev_ak4671, &ak4671_dai, 1);
return ret;
}
static int ak4671_i2c_remove(struct i2c_client *client)
{
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
snd_soc_unregister_codec(&client->dev);
return 0;
}
static const struct i2c_device_id ak4671_i2c_id[] = {
{ "ak4671", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id);
static struct i2c_driver ak4671_i2c_driver = {
.driver = {
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
.name = "ak4671-codec",
.owner = THIS_MODULE,
},
.probe = ak4671_i2c_probe,
.remove = ak4671_i2c_remove,
.id_table = ak4671_i2c_id,
};
module_i2c_driver(ak4671_i2c_driver);
MODULE_DESCRIPTION("ASoC AK4671 codec driver");
MODULE_AUTHOR("Joonyoung Shim <jy0922.shim@samsung.com>");
MODULE_LICENSE("GPL");