linux/sound/soc/codecs/wm9712.c

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/*
* wm9712.c -- ALSA Soc WM9712 codec support
*
* Copyright 2006-12 Wolfson Microelectronics PLC.
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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/init.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 08:04:11 +00:00
#include <linux/slab.h>
#include <linux/mfd/wm97xx.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <linux/regmap.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/ac97/codec.h>
#include <sound/ac97/compat.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/tlv.h>
#define WM9712_VENDOR_ID 0x574d4c12
#define WM9712_VENDOR_ID_MASK 0xffffffff
struct wm9712_priv {
struct snd_ac97 *ac97;
unsigned int hp_mixer[2];
struct mutex lock;
struct wm97xx_platform_data *mfd_pdata;
};
static const struct reg_default wm9712_reg_defaults[] = {
{ 0x02, 0x8000 },
{ 0x04, 0x8000 },
{ 0x06, 0x8000 },
{ 0x08, 0x0f0f },
{ 0x0a, 0xaaa0 },
{ 0x0c, 0xc008 },
{ 0x0e, 0x6808 },
{ 0x10, 0xe808 },
{ 0x12, 0xaaa0 },
{ 0x14, 0xad00 },
{ 0x16, 0x8000 },
{ 0x18, 0xe808 },
{ 0x1a, 0x3000 },
{ 0x1c, 0x8000 },
{ 0x20, 0x0000 },
{ 0x22, 0x0000 },
{ 0x26, 0x000f },
{ 0x28, 0x0605 },
{ 0x2a, 0x0410 },
{ 0x2c, 0xbb80 },
{ 0x2e, 0xbb80 },
{ 0x32, 0xbb80 },
{ 0x34, 0x2000 },
{ 0x4c, 0xf83e },
{ 0x4e, 0xffff },
{ 0x50, 0x0000 },
{ 0x52, 0x0000 },
{ 0x56, 0xf83e },
{ 0x58, 0x0008 },
{ 0x5c, 0x0000 },
{ 0x60, 0xb032 },
{ 0x62, 0x3e00 },
{ 0x64, 0x0000 },
{ 0x76, 0x0006 },
{ 0x78, 0x0001 },
{ 0x7a, 0x0000 },
};
static bool wm9712_volatile_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AC97_REC_GAIN:
return true;
default:
return regmap_ac97_default_volatile(dev, reg);
}
}
static const struct regmap_config wm9712_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x7e,
.cache_type = REGCACHE_RBTREE,
.volatile_reg = wm9712_volatile_reg,
.reg_defaults = wm9712_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm9712_reg_defaults),
};
#define HPL_MIXER 0x0
#define HPR_MIXER 0x1
static const char *wm9712_alc_select[] = {"None", "Left", "Right", "Stereo"};
static const char *wm9712_alc_mux[] = {"Stereo", "Left", "Right", "None"};
static const char *wm9712_out3_src[] = {"Left", "VREF", "Left + Right",
"Mono"};
static const char *wm9712_spk_src[] = {"Speaker Mix", "Headphone Mix"};
static const char *wm9712_rec_adc[] = {"Stereo", "Left", "Right", "Mute"};
static const char *wm9712_base[] = {"Linear Control", "Adaptive Boost"};
static const char *wm9712_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
static const char *wm9712_mic[] = {"Mic 1", "Differential", "Mic 2",
"Stereo"};
static const char *wm9712_rec_sel[] = {"Mic", "NC", "NC", "Speaker Mixer",
"Line", "Headphone Mixer", "Phone Mixer", "Phone"};
static const char *wm9712_ng_type[] = {"Constant Gain", "Mute"};
static const char *wm9712_diff_sel[] = {"Mic", "Line"};
static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 2000, 0);
static const struct soc_enum wm9712_enum[] = {
SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9712_alc_select),
SOC_ENUM_SINGLE(AC97_VIDEO, 12, 4, wm9712_alc_mux),
SOC_ENUM_SINGLE(AC97_AUX, 9, 4, wm9712_out3_src),
SOC_ENUM_SINGLE(AC97_AUX, 8, 2, wm9712_spk_src),
SOC_ENUM_SINGLE(AC97_REC_SEL, 12, 4, wm9712_rec_adc),
SOC_ENUM_SINGLE(AC97_MASTER_TONE, 15, 2, wm9712_base),
SOC_ENUM_DOUBLE(AC97_REC_GAIN, 14, 6, 2, wm9712_rec_gain),
SOC_ENUM_SINGLE(AC97_MIC, 5, 4, wm9712_mic),
SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9712_rec_sel),
SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9712_rec_sel),
SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9712_ng_type),
SOC_ENUM_SINGLE(0x5c, 8, 2, wm9712_diff_sel),
};
static const struct snd_kcontrol_new wm9712_snd_ac97_controls[] = {
SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1),
SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1),
SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0),
SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0),
SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0),
SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0),
SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0),
SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0),
SOC_SINGLE("ALC Decay Time", AC97_CODEC_CLASS_REV, 4, 15, 0),
SOC_SINGLE("ALC Attack Time", AC97_CODEC_CLASS_REV, 0, 15, 0),
SOC_ENUM("ALC Function", wm9712_enum[0]),
SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 1),
SOC_SINGLE("ALC ZC Switch", AC97_PCI_SVID, 8, 1, 0),
SOC_SINGLE("ALC NG Switch", AC97_PCI_SVID, 7, 1, 0),
SOC_ENUM("ALC NG Type", wm9712_enum[10]),
SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 1),
SOC_SINGLE("Mic Headphone Volume", AC97_VIDEO, 12, 7, 1),
SOC_SINGLE("ALC Headphone Volume", AC97_VIDEO, 7, 7, 1),
SOC_SINGLE("Out3 Switch", AC97_AUX, 15, 1, 1),
SOC_SINGLE("Out3 ZC Switch", AC97_AUX, 7, 1, 1),
SOC_SINGLE("Out3 Volume", AC97_AUX, 0, 31, 1),
SOC_SINGLE("PCBeep Bypass Headphone Volume", AC97_PC_BEEP, 12, 7, 1),
SOC_SINGLE("PCBeep Bypass Speaker Volume", AC97_PC_BEEP, 8, 7, 1),
SOC_SINGLE("PCBeep Bypass Phone Volume", AC97_PC_BEEP, 4, 7, 1),
SOC_SINGLE("Aux Playback Headphone Volume", AC97_CD, 12, 7, 1),
SOC_SINGLE("Aux Playback Speaker Volume", AC97_CD, 8, 7, 1),
SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1),
SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1),
SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1),
SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv),
SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1,
boost_tlv),
SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1),
SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1),
SOC_SINGLE("3D Playback Volume", AC97_3D_CONTROL, 0, 15, 0),
SOC_ENUM("Bass Control", wm9712_enum[5]),
SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1),
SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1),
SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0),
SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1),
SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1),
SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),
SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 63, 0),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
SOC_SINGLE_TLV("Mic 1 Volume", AC97_MIC, 8, 31, 1, main_tlv),
SOC_SINGLE_TLV("Mic 2 Volume", AC97_MIC, 0, 31, 1, main_tlv),
SOC_SINGLE_TLV("Mic Boost Volume", AC97_MIC, 7, 1, 0, boost_tlv),
};
static const unsigned int wm9712_mixer_mute_regs[] = {
AC97_VIDEO,
AC97_PCM,
AC97_LINE,
AC97_PHONE,
AC97_CD,
AC97_PC_BEEP,
};
/* We have to create a fake left and right HP mixers because
* the codec only has a single control that is shared by both channels.
* This makes it impossible to determine the audio path.
*/
static int wm9712_hp_mixer_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
unsigned int val = ucontrol->value.integer.value[0];
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int mixer, mask, shift, old;
struct snd_soc_dapm_update update = {};
bool change;
mixer = mc->shift >> 8;
shift = mc->shift & 0xff;
mask = 1 << shift;
mutex_lock(&wm9712->lock);
old = wm9712->hp_mixer[mixer];
if (ucontrol->value.integer.value[0])
wm9712->hp_mixer[mixer] |= mask;
else
wm9712->hp_mixer[mixer] &= ~mask;
change = old != wm9712->hp_mixer[mixer];
if (change) {
update.kcontrol = kcontrol;
update.reg = wm9712_mixer_mute_regs[shift];
update.mask = 0x8000;
if ((wm9712->hp_mixer[0] & mask) ||
(wm9712->hp_mixer[1] & mask))
update.val = 0x0;
else
update.val = 0x8000;
snd_soc_dapm_mixer_update_power(dapm, kcontrol, val,
&update);
}
mutex_unlock(&wm9712->lock);
return change;
}
static int wm9712_hp_mixer_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol);
struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int shift, mixer;
mixer = mc->shift >> 8;
shift = mc->shift & 0xff;
ucontrol->value.integer.value[0] =
(wm9712->hp_mixer[mixer] >> shift) & 1;
return 0;
}
#define WM9712_HP_MIXER_CTRL(xname, xmixer, xshift) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.get = wm9712_hp_mixer_get, .put = wm9712_hp_mixer_put, \
.private_value = SOC_SINGLE_VALUE(SND_SOC_NOPM, \
(xmixer << 8) | xshift, 1, 0, 0) \
}
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPL_MIXER, 5),
WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 4),
WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPL_MIXER, 3),
WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPL_MIXER, 2),
WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 1),
WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPL_MIXER, 0),
};
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
WM9712_HP_MIXER_CTRL("PCBeep Bypass Switch", HPR_MIXER, 5),
WM9712_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 4),
WM9712_HP_MIXER_CTRL("Phone Bypass Switch", HPR_MIXER, 3),
WM9712_HP_MIXER_CTRL("Line Bypass Switch", HPR_MIXER, 2),
WM9712_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 1),
WM9712_HP_MIXER_CTRL("Mic Sidetone Switch", HPR_MIXER, 0),
};
/* Speaker Mixer */
static const struct snd_kcontrol_new wm9712_speaker_mixer_controls[] = {
SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 11, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 11, 1, 1),
SOC_DAPM_SINGLE("Phone Bypass Switch", AC97_PHONE, 14, 1, 1),
SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 14, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 14, 1, 1),
};
/* Phone Mixer */
static const struct snd_kcontrol_new wm9712_phone_mixer_controls[] = {
SOC_DAPM_SINGLE("PCBeep Bypass Switch", AC97_PC_BEEP, 7, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_CD, 7, 1, 1),
SOC_DAPM_SINGLE("Line Bypass Switch", AC97_LINE, 13, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PCM, 13, 1, 1),
SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_MIC, 14, 1, 1),
SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_MIC, 13, 1, 1),
};
/* ALC headphone mux */
static const struct snd_kcontrol_new wm9712_alc_mux_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[1]);
/* out 3 mux */
static const struct snd_kcontrol_new wm9712_out3_mux_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[2]);
/* spk mux */
static const struct snd_kcontrol_new wm9712_spk_mux_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[3]);
/* Capture to Phone mux */
static const struct snd_kcontrol_new wm9712_capture_phone_mux_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[4]);
/* Capture left select */
static const struct snd_kcontrol_new wm9712_capture_selectl_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[8]);
/* Capture right select */
static const struct snd_kcontrol_new wm9712_capture_selectr_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[9]);
/* Mic select */
static const struct snd_kcontrol_new wm9712_mic_src_controls =
SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]);
/* diff select */
static const struct snd_kcontrol_new wm9712_diff_sel_controls =
SOC_DAPM_ENUM("Route", wm9712_enum[11]);
static const struct snd_soc_dapm_widget wm9712_dapm_widgets[] = {
SND_SOC_DAPM_MUX("ALC Sidetone Mux", SND_SOC_NOPM, 0, 0,
&wm9712_alc_mux_controls),
SND_SOC_DAPM_MUX("Out3 Mux", SND_SOC_NOPM, 0, 0,
&wm9712_out3_mux_controls),
SND_SOC_DAPM_MUX("Speaker Mux", SND_SOC_NOPM, 0, 0,
&wm9712_spk_mux_controls),
SND_SOC_DAPM_MUX("Capture Phone Mux", SND_SOC_NOPM, 0, 0,
&wm9712_capture_phone_mux_controls),
SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectl_controls),
SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0,
&wm9712_capture_selectr_controls),
SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0,
&wm9712_mic_src_controls),
SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0,
&wm9712_mic_src_controls),
SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0,
&wm9712_diff_sel_controls),
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_INT_PAGING, 9, 1,
&wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls)),
SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_INT_PAGING, 8, 1,
&wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls)),
SND_SOC_DAPM_MIXER("Phone Mixer", AC97_INT_PAGING, 6, 1,
&wm9712_phone_mixer_controls[0], ARRAY_SIZE(wm9712_phone_mixer_controls)),
SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_INT_PAGING, 7, 1,
&wm9712_speaker_mixer_controls[0],
ARRAY_SIZE(wm9712_speaker_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_INT_PAGING, 14, 1),
SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_INT_PAGING, 13, 1),
SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", AC97_INT_PAGING, 12, 1),
SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", AC97_INT_PAGING, 11, 1),
SND_SOC_DAPM_PGA("Headphone PGA", AC97_INT_PAGING, 4, 1, NULL, 0),
SND_SOC_DAPM_PGA("Speaker PGA", AC97_INT_PAGING, 3, 1, NULL, 0),
SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0),
SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0),
SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1),
SND_SOC_DAPM_OUTPUT("MONOOUT"),
SND_SOC_DAPM_OUTPUT("HPOUTL"),
SND_SOC_DAPM_OUTPUT("HPOUTR"),
SND_SOC_DAPM_OUTPUT("LOUT2"),
SND_SOC_DAPM_OUTPUT("ROUT2"),
SND_SOC_DAPM_OUTPUT("OUT3"),
SND_SOC_DAPM_INPUT("LINEINL"),
SND_SOC_DAPM_INPUT("LINEINR"),
SND_SOC_DAPM_INPUT("PHONE"),
SND_SOC_DAPM_INPUT("PCBEEP"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2"),
};
static const struct snd_soc_dapm_route wm9712_audio_map[] = {
/* virtual mixer - mixes left & right channels for spk and mono */
{"AC97 Mixer", NULL, "Left DAC"},
{"AC97 Mixer", NULL, "Right DAC"},
/* Left HP mixer */
{"Left HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
{"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Left HP Mixer", "Phone Bypass Switch", "Phone PGA"},
{"Left HP Mixer", "Line Bypass Switch", "Line PGA"},
{"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
{"Left HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
{"Left HP Mixer", NULL, "ALC Sidetone Mux"},
/* Right HP mixer */
{"Right HP Mixer", "PCBeep Bypass Switch", "PCBEEP"},
{"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Right HP Mixer", "Phone Bypass Switch", "Phone PGA"},
{"Right HP Mixer", "Line Bypass Switch", "Line PGA"},
{"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
{"Right HP Mixer", "Mic Sidetone Switch", "Mic PGA"},
{"Right HP Mixer", NULL, "ALC Sidetone Mux"},
/* speaker mixer */
{"Speaker Mixer", "PCBeep Bypass Switch", "PCBEEP"},
{"Speaker Mixer", "Line Bypass Switch", "Line PGA"},
{"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
{"Speaker Mixer", "Phone Bypass Switch", "Phone PGA"},
{"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
/* Phone mixer */
{"Phone Mixer", "PCBeep Bypass Switch", "PCBEEP"},
{"Phone Mixer", "Line Bypass Switch", "Line PGA"},
{"Phone Mixer", "Aux Playback Switch", "Aux DAC"},
{"Phone Mixer", "PCM Playback Switch", "AC97 Mixer"},
{"Phone Mixer", "Mic 1 Sidetone Switch", "Mic PGA"},
{"Phone Mixer", "Mic 2 Sidetone Switch", "Mic PGA"},
/* inputs */
{"Line PGA", NULL, "LINEINL"},
{"Line PGA", NULL, "LINEINR"},
{"Phone PGA", NULL, "PHONE"},
{"Mic PGA", NULL, "MIC1"},
{"Mic PGA", NULL, "MIC2"},
/* microphones */
{"Differential Mic", NULL, "MIC1"},
{"Differential Mic", NULL, "MIC2"},
{"Left Mic Select Source", "Mic 1", "MIC1"},
{"Left Mic Select Source", "Mic 2", "MIC2"},
{"Left Mic Select Source", "Stereo", "MIC1"},
{"Left Mic Select Source", "Differential", "Differential Mic"},
{"Right Mic Select Source", "Mic 1", "MIC1"},
{"Right Mic Select Source", "Mic 2", "MIC2"},
{"Right Mic Select Source", "Stereo", "MIC2"},
{"Right Mic Select Source", "Differential", "Differential Mic"},
/* left capture selector */
{"Left Capture Select", "Mic", "MIC1"},
{"Left Capture Select", "Speaker Mixer", "Speaker Mixer"},
{"Left Capture Select", "Line", "LINEINL"},
{"Left Capture Select", "Headphone Mixer", "Left HP Mixer"},
{"Left Capture Select", "Phone Mixer", "Phone Mixer"},
{"Left Capture Select", "Phone", "PHONE"},
/* right capture selector */
{"Right Capture Select", "Mic", "MIC2"},
{"Right Capture Select", "Speaker Mixer", "Speaker Mixer"},
{"Right Capture Select", "Line", "LINEINR"},
{"Right Capture Select", "Headphone Mixer", "Right HP Mixer"},
{"Right Capture Select", "Phone Mixer", "Phone Mixer"},
{"Right Capture Select", "Phone", "PHONE"},
/* ALC Sidetone */
{"ALC Sidetone Mux", "Stereo", "Left Capture Select"},
{"ALC Sidetone Mux", "Stereo", "Right Capture Select"},
{"ALC Sidetone Mux", "Left", "Left Capture Select"},
{"ALC Sidetone Mux", "Right", "Right Capture Select"},
/* ADC's */
{"Left ADC", NULL, "Left Capture Select"},
{"Right ADC", NULL, "Right Capture Select"},
/* outputs */
{"MONOOUT", NULL, "Phone Mixer"},
{"HPOUTL", NULL, "Headphone PGA"},
{"Headphone PGA", NULL, "Left HP Mixer"},
{"HPOUTR", NULL, "Headphone PGA"},
{"Headphone PGA", NULL, "Right HP Mixer"},
/* mono mixer */
{"Mono Mixer", NULL, "Left HP Mixer"},
{"Mono Mixer", NULL, "Right HP Mixer"},
/* Out3 Mux */
{"Out3 Mux", "Left", "Left HP Mixer"},
{"Out3 Mux", "Mono", "Phone Mixer"},
{"Out3 Mux", "Left + Right", "Mono Mixer"},
{"Out 3 PGA", NULL, "Out3 Mux"},
{"OUT3", NULL, "Out 3 PGA"},
/* speaker Mux */
{"Speaker Mux", "Speaker Mix", "Speaker Mixer"},
{"Speaker Mux", "Headphone Mix", "Mono Mixer"},
{"Speaker PGA", NULL, "Speaker Mux"},
{"LOUT2", NULL, "Speaker PGA"},
{"ROUT2", NULL, "Speaker PGA"},
};
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
int reg;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x1, 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return snd_soc_component_write(component, reg, runtime->rate);
}
static int ac97_aux_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
struct snd_pcm_runtime *runtime = substream->runtime;
snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x1, 0x1);
snd_soc_component_update_bits(component, AC97_PCI_SID, 0x8000, 0x8000);
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
return snd_soc_component_write(component, AC97_PCM_SURR_DAC_RATE, runtime->rate);
}
#define WM9712_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
SNDRV_PCM_RATE_48000)
static const struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
.prepare = ac97_prepare,
};
static const struct snd_soc_dai_ops wm9712_dai_ops_aux = {
.prepare = ac97_aux_prepare,
};
static struct snd_soc_dai_driver wm9712_dai[] = {
{
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>
2010-03-17 20:15:21 +00:00
.name = "wm9712-hifi",
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
.formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9712_AC97_RATES,
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_hifi,
},
{
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>
2010-03-17 20:15:21 +00:00
.name = "wm9712-aux",
.playback = {
.stream_name = "Aux Playback",
.channels_min = 1,
.channels_max = 1,
.rates = WM9712_AC97_RATES,
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9712_dai_ops_aux,
}
};
static int wm9712_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
snd_soc_component_write(component, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF:
/* disable everything including AC link */
snd_soc_component_write(component, AC97_EXTENDED_MSTATUS, 0xffff);
snd_soc_component_write(component, AC97_POWERDOWN, 0xffff);
break;
}
return 0;
}
static int wm9712_soc_resume(struct snd_soc_component *component)
{
struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
int ret;
ret = snd_ac97_reset(wm9712->ac97, true, WM9712_VENDOR_ID,
WM9712_VENDOR_ID_MASK);
if (ret < 0)
return ret;
snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
if (ret == 0)
snd_soc_component_cache_sync(component);
return ret;
}
static int wm9712_soc_probe(struct snd_soc_component *component)
{
struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
struct regmap *regmap;
int ret;
if (wm9712->mfd_pdata) {
wm9712->ac97 = wm9712->mfd_pdata->ac97;
regmap = wm9712->mfd_pdata->regmap;
} else {
#ifdef CONFIG_SND_SOC_AC97_BUS
wm9712->ac97 = snd_soc_new_ac97_component(component, WM9712_VENDOR_ID,
WM9712_VENDOR_ID_MASK);
if (IS_ERR(wm9712->ac97)) {
ret = PTR_ERR(wm9712->ac97);
dev_err(component->dev,
"Failed to register AC97 codec: %d\n", ret);
return ret;
}
regmap = regmap_init_ac97(wm9712->ac97, &wm9712_regmap_config);
if (IS_ERR(regmap)) {
snd_soc_free_ac97_component(wm9712->ac97);
return PTR_ERR(regmap);
}
#endif
}
snd_soc_component_init_regmap(component, regmap);
/* set alc mux to none */
snd_soc_component_update_bits(component, AC97_VIDEO, 0x3000, 0x3000);
return 0;
}
static void wm9712_soc_remove(struct snd_soc_component *component)
{
#ifdef CONFIG_SND_SOC_AC97_BUS
struct wm9712_priv *wm9712 = snd_soc_component_get_drvdata(component);
if (!wm9712->mfd_pdata) {
snd_soc_component_exit_regmap(component);
snd_soc_free_ac97_component(wm9712->ac97);
}
#endif
}
static const struct snd_soc_component_driver soc_component_dev_wm9712 = {
.probe = wm9712_soc_probe,
.remove = wm9712_soc_remove,
.resume = wm9712_soc_resume,
.set_bias_level = wm9712_set_bias_level,
.controls = wm9712_snd_ac97_controls,
.num_controls = ARRAY_SIZE(wm9712_snd_ac97_controls),
.dapm_widgets = wm9712_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm9712_dapm_widgets),
.dapm_routes = wm9712_audio_map,
.num_dapm_routes = ARRAY_SIZE(wm9712_audio_map),
.suspend_bias_off = 1,
.idle_bias_on = 1,
.use_pmdown_time = 1,
.endianness = 1,
.non_legacy_dai_naming = 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>
2010-03-17 20:15:21 +00:00
static int wm9712_probe(struct platform_device *pdev)
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>
2010-03-17 20:15:21 +00:00
{
struct wm9712_priv *wm9712;
wm9712 = devm_kzalloc(&pdev->dev, sizeof(*wm9712), GFP_KERNEL);
if (wm9712 == NULL)
return -ENOMEM;
mutex_init(&wm9712->lock);
wm9712->mfd_pdata = dev_get_platdata(&pdev->dev);
platform_set_drvdata(pdev, wm9712);
return devm_snd_soc_register_component(&pdev->dev,
&soc_component_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai));
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>
2010-03-17 20:15:21 +00:00
}
static struct platform_driver wm9712_component_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>
2010-03-17 20:15:21 +00:00
.driver = {
.name = "wm9712-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>
2010-03-17 20:15:21 +00:00
},
.probe = wm9712_probe,
};
module_platform_driver(wm9712_component_driver);
MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");