OpenCloudOS-Kernel/sound/soc/codecs/wm9713.c

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// SPDX-License-Identifier: GPL-2.0-or-later
/*
* wm9713.c -- ALSA Soc WM9713 codec support
*
* Copyright 2006-10 Wolfson Microelectronics PLC.
* Author: Liam Girdwood <lrg@slimlogic.co.uk>
*
* Features:-
*
* o Support for AC97 Codec, Voice DAC and Aux DAC
* o Support for DAPM
*/
#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 16:04:11 +08:00
#include <linux/slab.h>
#include <linux/mfd/wm97xx.h>
#include <linux/module.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/pcm_params.h>
#include <sound/tlv.h>
#include <sound/soc.h>
#include "wm9713.h"
#define WM9713_VENDOR_ID 0x574d4c13
#define WM9713_VENDOR_ID_MASK 0xffffffff
struct wm9713_priv {
struct snd_ac97 *ac97;
u32 pll_in; /* PLL input frequency */
unsigned int hp_mixer[2];
struct mutex lock;
struct wm97xx_platform_data *mfd_pdata;
};
#define HPL_MIXER 0
#define HPR_MIXER 1
static const char *wm9713_mic_mixer[] = {"Stereo", "Mic 1", "Mic 2", "Mute"};
static const char *wm9713_rec_mux[] = {"Stereo", "Left", "Right", "Mute"};
static const char *wm9713_rec_src[] =
{"Mic 1", "Mic 2", "Line", "Mono In", "Headphone", "Speaker",
"Mono Out", "Zh"};
static const char *wm9713_rec_gain[] = {"+1.5dB Steps", "+0.75dB Steps"};
static const char *wm9713_alc_select[] = {"None", "Left", "Right", "Stereo"};
static const char *wm9713_mono_pga[] = {"Vmid", "Zh", "Mono", "Inv"};
static const char *wm9713_spk_pga[] =
{"Vmid", "Zh", "Headphone", "Speaker", "Inv", "Headphone Vmid",
"Speaker Vmid", "Inv Vmid"};
static const char *wm9713_hp_pga[] = {"Vmid", "Zh", "Headphone",
"Headphone Vmid"};
static const char *wm9713_out3_pga[] = {"Vmid", "Zh", "Inv 1", "Inv 1 Vmid"};
static const char *wm9713_out4_pga[] = {"Vmid", "Zh", "Inv 2", "Inv 2 Vmid"};
static const char *wm9713_dac_inv[] =
{"Off", "Mono", "Speaker", "Left Headphone", "Right Headphone",
"Headphone Mono", "NC", "Vmid"};
static const char *wm9713_bass[] = {"Linear Control", "Adaptive Boost"};
static const char *wm9713_ng_type[] = {"Constant Gain", "Mute"};
static const char *wm9713_mic_select[] = {"Mic 1", "Mic 2 A", "Mic 2 B"};
static const char *wm9713_micb_select[] = {"MPB", "MPA"};
static const struct soc_enum wm9713_enum[] = {
SOC_ENUM_SINGLE(AC97_LINE, 3, 4, wm9713_mic_mixer), /* record mic mixer 0 */
SOC_ENUM_SINGLE(AC97_VIDEO, 14, 4, wm9713_rec_mux), /* record mux hp 1 */
SOC_ENUM_SINGLE(AC97_VIDEO, 9, 4, wm9713_rec_mux), /* record mux mono 2 */
SOC_ENUM_SINGLE(AC97_VIDEO, 3, 8, wm9713_rec_src), /* record mux left 3 */
SOC_ENUM_SINGLE(AC97_VIDEO, 0, 8, wm9713_rec_src), /* record mux right 4*/
SOC_ENUM_DOUBLE(AC97_CD, 14, 6, 2, wm9713_rec_gain), /* record step size 5 */
SOC_ENUM_SINGLE(AC97_PCI_SVID, 14, 4, wm9713_alc_select), /* alc source select 6*/
SOC_ENUM_SINGLE(AC97_REC_GAIN, 14, 4, wm9713_mono_pga), /* mono input select 7 */
SOC_ENUM_SINGLE(AC97_REC_GAIN, 11, 8, wm9713_spk_pga), /* speaker left input select 8 */
SOC_ENUM_SINGLE(AC97_REC_GAIN, 8, 8, wm9713_spk_pga), /* speaker right input select 9 */
SOC_ENUM_SINGLE(AC97_REC_GAIN, 6, 3, wm9713_hp_pga), /* headphone left input 10 */
SOC_ENUM_SINGLE(AC97_REC_GAIN, 4, 3, wm9713_hp_pga), /* headphone right input 11 */
SOC_ENUM_SINGLE(AC97_REC_GAIN, 2, 4, wm9713_out3_pga), /* out 3 source 12 */
SOC_ENUM_SINGLE(AC97_REC_GAIN, 0, 4, wm9713_out4_pga), /* out 4 source 13 */
SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 13, 8, wm9713_dac_inv), /* dac invert 1 14 */
SOC_ENUM_SINGLE(AC97_REC_GAIN_MIC, 10, 8, wm9713_dac_inv), /* dac invert 2 15 */
SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, wm9713_bass), /* bass control 16 */
SOC_ENUM_SINGLE(AC97_PCI_SVID, 5, 2, wm9713_ng_type), /* noise gate type 17 */
SOC_ENUM_SINGLE(AC97_3D_CONTROL, 12, 3, wm9713_mic_select), /* mic selection 18 */
SOC_ENUM_SINGLE_VIRT(2, wm9713_micb_select), /* mic selection 19 */
};
static const DECLARE_TLV_DB_SCALE(out_tlv, -4650, 150, 0);
static const DECLARE_TLV_DB_SCALE(main_tlv, -3450, 150, 0);
static const DECLARE_TLV_DB_SCALE(misc_tlv, -1500, 300, 0);
static const DECLARE_TLV_DB_RANGE(mic_tlv,
0, 2, TLV_DB_SCALE_ITEM(1200, 600, 0),
3, 3, TLV_DB_SCALE_ITEM(3000, 0, 0)
);
static const struct snd_kcontrol_new wm9713_snd_ac97_controls[] = {
SOC_DOUBLE_TLV("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1, out_tlv),
SOC_DOUBLE("Speaker Playback Switch", AC97_MASTER, 15, 7, 1, 1),
SOC_DOUBLE_TLV("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1,
out_tlv),
SOC_DOUBLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 7, 1, 1),
SOC_DOUBLE_TLV("Line In Volume", AC97_PC_BEEP, 8, 0, 31, 1, main_tlv),
SOC_DOUBLE_TLV("PCM Playback Volume", AC97_PHONE, 8, 0, 31, 1, main_tlv),
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 1 Preamp Volume", AC97_3D_CONTROL, 10, 3, 0, mic_tlv),
SOC_SINGLE_TLV("Mic 2 Preamp Volume", AC97_3D_CONTROL, 12, 3, 0, mic_tlv),
SOC_SINGLE("Mic Boost (+20dB) Switch", AC97_LINE, 5, 1, 0),
SOC_SINGLE("Mic Headphone Mixer Volume", AC97_LINE, 0, 7, 1),
SOC_SINGLE("Capture Switch", AC97_CD, 15, 1, 1),
SOC_ENUM("Capture Volume Steps", wm9713_enum[5]),
SOC_DOUBLE("Capture Volume", AC97_CD, 8, 0, 31, 0),
SOC_SINGLE("Capture ZC Switch", AC97_CD, 7, 1, 0),
SOC_SINGLE_TLV("Capture to Headphone Volume", AC97_VIDEO, 11, 7, 1, misc_tlv),
SOC_SINGLE("Capture to Mono Boost (+20dB) Switch", AC97_VIDEO, 8, 1, 0),
SOC_SINGLE("Capture ADC Boost (+20dB) Switch", AC97_VIDEO, 6, 1, 0),
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", wm9713_enum[6]),
SOC_SINGLE("ALC Max Volume", AC97_PCI_SVID, 11, 7, 0),
SOC_SINGLE("ALC ZC Timeout", AC97_PCI_SVID, 9, 3, 0),
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", wm9713_enum[17]),
SOC_SINGLE("ALC NG Threshold", AC97_PCI_SVID, 0, 31, 0),
SOC_DOUBLE("Speaker Playback ZC Switch", AC97_MASTER, 14, 6, 1, 0),
SOC_DOUBLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 14, 6, 1, 0),
SOC_SINGLE("Out4 Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
SOC_SINGLE("Out4 Playback ZC Switch", AC97_MASTER_MONO, 14, 1, 0),
SOC_SINGLE_TLV("Out4 Playback Volume", AC97_MASTER_MONO, 8, 31, 1, out_tlv),
SOC_SINGLE("Out3 Playback Switch", AC97_MASTER_MONO, 7, 1, 1),
SOC_SINGLE("Out3 Playback ZC Switch", AC97_MASTER_MONO, 6, 1, 0),
SOC_SINGLE_TLV("Out3 Playback Volume", AC97_MASTER_MONO, 0, 31, 1, out_tlv),
SOC_SINGLE_TLV("Mono Capture Volume", AC97_MASTER_TONE, 8, 31, 1, main_tlv),
SOC_SINGLE("Mono Playback Switch", AC97_MASTER_TONE, 7, 1, 1),
SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_TONE, 6, 1, 0),
SOC_SINGLE_TLV("Mono Playback Volume", AC97_MASTER_TONE, 0, 31, 1, out_tlv),
SOC_SINGLE_TLV("Headphone Mixer Beep Playback Volume", AC97_AUX, 12, 7, 1,
misc_tlv),
SOC_SINGLE_TLV("Speaker Mixer Beep Playback Volume", AC97_AUX, 8, 7, 1,
misc_tlv),
SOC_SINGLE_TLV("Mono Mixer Beep Playback Volume", AC97_AUX, 4, 7, 1, misc_tlv),
SOC_SINGLE_TLV("Voice Playback Headphone Volume", AC97_PCM, 12, 7, 1,
misc_tlv),
SOC_SINGLE("Voice Playback Master Volume", AC97_PCM, 8, 7, 1),
SOC_SINGLE("Voice Playback Mono Volume", AC97_PCM, 4, 7, 1),
SOC_SINGLE_TLV("Headphone Mixer Aux Playback Volume", AC97_REC_SEL, 12, 7, 1,
misc_tlv),
SOC_SINGLE_TLV("Speaker Mixer Voice Playback Volume", AC97_PCM, 8, 7, 1,
misc_tlv),
SOC_SINGLE_TLV("Speaker Mixer Aux Playback Volume", AC97_REC_SEL, 8, 7, 1,
misc_tlv),
SOC_SINGLE_TLV("Mono Mixer Voice Playback Volume", AC97_PCM, 4, 7, 1,
misc_tlv),
SOC_SINGLE_TLV("Mono Mixer Aux Playback Volume", AC97_REC_SEL, 4, 7, 1,
misc_tlv),
SOC_SINGLE("Aux Playback Headphone Volume", AC97_REC_SEL, 12, 7, 1),
SOC_SINGLE("Aux Playback Master Volume", AC97_REC_SEL, 8, 7, 1),
SOC_ENUM("Bass Control", wm9713_enum[16]),
SOC_SINGLE("Bass Cut-off Switch", AC97_GENERAL_PURPOSE, 12, 1, 1),
SOC_SINGLE("Tone Cut-off Switch", AC97_GENERAL_PURPOSE, 4, 1, 1),
SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_GENERAL_PURPOSE, 6, 1, 0),
SOC_SINGLE("Bass Volume", AC97_GENERAL_PURPOSE, 8, 15, 1),
SOC_SINGLE("Tone Volume", AC97_GENERAL_PURPOSE, 0, 15, 1),
SOC_SINGLE("3D Upper Cut-off Switch", AC97_REC_GAIN_MIC, 5, 1, 0),
SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
};
static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm);
if (WARN_ON(event != SND_SOC_DAPM_PRE_PMD))
return -EINVAL;
/* Gracefully shut down the voice interface. */
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0f00, 0x0200);
schedule_timeout_interruptible(msecs_to_jiffies(1));
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0f00, 0x0f00);
snd_soc_component_update_bits(component, AC97_EXTENDED_MID, 0x1000, 0x1000);
return 0;
}
static const unsigned int wm9713_mixer_mute_regs[] = {
AC97_PC_BEEP,
AC97_MASTER_TONE,
AC97_PHONE,
AC97_REC_SEL,
AC97_PCM,
AC97_AUX,
};
/* 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 using the current
* register map, thus we add a new (virtual) register to help determine the
* audio route within the device.
*/
static int wm9713_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 wm9713_priv *wm9713 = 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(&wm9713->lock);
old = wm9713->hp_mixer[mixer];
if (ucontrol->value.integer.value[0])
wm9713->hp_mixer[mixer] |= mask;
else
wm9713->hp_mixer[mixer] &= ~mask;
change = old != wm9713->hp_mixer[mixer];
if (change) {
update.kcontrol = kcontrol;
update.reg = wm9713_mixer_mute_regs[shift];
update.mask = 0x8000;
if ((wm9713->hp_mixer[0] & mask) ||
(wm9713->hp_mixer[1] & mask))
update.val = 0x0;
else
update.val = 0x8000;
snd_soc_dapm_mixer_update_power(dapm, kcontrol, val,
&update);
}
mutex_unlock(&wm9713->lock);
return change;
}
static int wm9713_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 wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component);
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
unsigned int mixer, shift;
mixer = mc->shift >> 8;
shift = mc->shift & 0xff;
ucontrol->value.integer.value[0] =
(wm9713->hp_mixer[mixer] >> shift) & 1;
return 0;
}
#define WM9713_HP_MIXER_CTRL(xname, xmixer, xshift) { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.info = snd_soc_info_volsw, \
.get = wm9713_hp_mixer_get, .put = wm9713_hp_mixer_put, \
.private_value = SOC_DOUBLE_VALUE(SND_SOC_NOPM, \
xshift, xmixer, 1, 0, 0) \
}
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpl_mixer_controls[] = {
WM9713_HP_MIXER_CTRL("Beep Playback Switch", HPL_MIXER, 5),
WM9713_HP_MIXER_CTRL("Voice Playback Switch", HPL_MIXER, 4),
WM9713_HP_MIXER_CTRL("Aux Playback Switch", HPL_MIXER, 3),
WM9713_HP_MIXER_CTRL("PCM Playback Switch", HPL_MIXER, 2),
WM9713_HP_MIXER_CTRL("MonoIn Playback Switch", HPL_MIXER, 1),
WM9713_HP_MIXER_CTRL("Bypass Playback Switch", HPL_MIXER, 0),
};
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9713_hpr_mixer_controls[] = {
WM9713_HP_MIXER_CTRL("Beep Playback Switch", HPR_MIXER, 5),
WM9713_HP_MIXER_CTRL("Voice Playback Switch", HPR_MIXER, 4),
WM9713_HP_MIXER_CTRL("Aux Playback Switch", HPR_MIXER, 3),
WM9713_HP_MIXER_CTRL("PCM Playback Switch", HPR_MIXER, 2),
WM9713_HP_MIXER_CTRL("MonoIn Playback Switch", HPR_MIXER, 1),
WM9713_HP_MIXER_CTRL("Bypass Playback Switch", HPR_MIXER, 0),
};
/* headphone capture mux */
static const struct snd_kcontrol_new wm9713_hp_rec_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[1]);
/* headphone mic mux */
static const struct snd_kcontrol_new wm9713_hp_mic_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[0]);
/* Speaker Mixer */
static const struct snd_kcontrol_new wm9713_speaker_mixer_controls[] = {
SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 11, 1, 1),
SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 11, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 11, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 14, 1, 1),
SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 14, 1, 1),
SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 14, 1, 1),
};
/* Mono Mixer */
static const struct snd_kcontrol_new wm9713_mono_mixer_controls[] = {
SOC_DAPM_SINGLE("Beep Playback Switch", AC97_AUX, 7, 1, 1),
SOC_DAPM_SINGLE("Voice Playback Switch", AC97_PCM, 7, 1, 1),
SOC_DAPM_SINGLE("Aux Playback Switch", AC97_REC_SEL, 7, 1, 1),
SOC_DAPM_SINGLE("PCM Playback Switch", AC97_PHONE, 13, 1, 1),
SOC_DAPM_SINGLE("MonoIn Playback Switch", AC97_MASTER_TONE, 13, 1, 1),
SOC_DAPM_SINGLE("Bypass Playback Switch", AC97_PC_BEEP, 13, 1, 1),
SOC_DAPM_SINGLE("Mic 1 Sidetone Switch", AC97_LINE, 7, 1, 1),
SOC_DAPM_SINGLE("Mic 2 Sidetone Switch", AC97_LINE, 6, 1, 1),
};
/* mono mic mux */
static const struct snd_kcontrol_new wm9713_mono_mic_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[2]);
/* mono output mux */
static const struct snd_kcontrol_new wm9713_mono_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[7]);
/* speaker left output mux */
static const struct snd_kcontrol_new wm9713_hp_spkl_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[8]);
/* speaker right output mux */
static const struct snd_kcontrol_new wm9713_hp_spkr_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[9]);
/* headphone left output mux */
static const struct snd_kcontrol_new wm9713_hpl_out_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[10]);
/* headphone right output mux */
static const struct snd_kcontrol_new wm9713_hpr_out_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[11]);
/* Out3 mux */
static const struct snd_kcontrol_new wm9713_out3_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[12]);
/* Out4 mux */
static const struct snd_kcontrol_new wm9713_out4_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[13]);
/* DAC inv mux 1 */
static const struct snd_kcontrol_new wm9713_dac_inv1_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[14]);
/* DAC inv mux 2 */
static const struct snd_kcontrol_new wm9713_dac_inv2_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[15]);
/* Capture source left */
static const struct snd_kcontrol_new wm9713_rec_srcl_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[3]);
/* Capture source right */
static const struct snd_kcontrol_new wm9713_rec_srcr_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[4]);
/* mic source */
static const struct snd_kcontrol_new wm9713_mic_sel_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[18]);
/* mic source B virtual control */
static const struct snd_kcontrol_new wm9713_micb_sel_mux_controls =
SOC_DAPM_ENUM("Route", wm9713_enum[19]);
static const struct snd_soc_dapm_widget wm9713_dapm_widgets[] = {
SND_SOC_DAPM_MUX("Capture Headphone Mux", SND_SOC_NOPM, 0, 0,
&wm9713_hp_rec_mux_controls),
SND_SOC_DAPM_MUX("Sidetone Mux", SND_SOC_NOPM, 0, 0,
&wm9713_hp_mic_mux_controls),
SND_SOC_DAPM_MUX("Capture Mono Mux", SND_SOC_NOPM, 0, 0,
&wm9713_mono_mic_mux_controls),
SND_SOC_DAPM_MUX("Mono Out Mux", SND_SOC_NOPM, 0, 0,
&wm9713_mono_mux_controls),
SND_SOC_DAPM_MUX("Left Speaker Out Mux", SND_SOC_NOPM, 0, 0,
&wm9713_hp_spkl_mux_controls),
SND_SOC_DAPM_MUX("Right Speaker Out Mux", SND_SOC_NOPM, 0, 0,
&wm9713_hp_spkr_mux_controls),
SND_SOC_DAPM_MUX("Left Headphone Out Mux", SND_SOC_NOPM, 0, 0,
&wm9713_hpl_out_mux_controls),
SND_SOC_DAPM_MUX("Right Headphone Out Mux", SND_SOC_NOPM, 0, 0,
&wm9713_hpr_out_mux_controls),
SND_SOC_DAPM_MUX("Out 3 Mux", SND_SOC_NOPM, 0, 0,
&wm9713_out3_mux_controls),
SND_SOC_DAPM_MUX("Out 4 Mux", SND_SOC_NOPM, 0, 0,
&wm9713_out4_mux_controls),
SND_SOC_DAPM_MUX("DAC Inv Mux 1", SND_SOC_NOPM, 0, 0,
&wm9713_dac_inv1_mux_controls),
SND_SOC_DAPM_MUX("DAC Inv Mux 2", SND_SOC_NOPM, 0, 0,
&wm9713_dac_inv2_mux_controls),
SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
&wm9713_rec_srcl_mux_controls),
SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
&wm9713_rec_srcr_mux_controls),
SND_SOC_DAPM_MUX("Mic A Source", SND_SOC_NOPM, 0, 0,
&wm9713_mic_sel_mux_controls),
SND_SOC_DAPM_MUX("Mic B Source", SND_SOC_NOPM, 0, 0,
&wm9713_micb_sel_mux_controls),
SND_SOC_DAPM_MIXER("Left HP Mixer", AC97_EXTENDED_MID, 3, 1,
&wm9713_hpl_mixer_controls[0], ARRAY_SIZE(wm9713_hpl_mixer_controls)),
SND_SOC_DAPM_MIXER("Right HP Mixer", AC97_EXTENDED_MID, 2, 1,
&wm9713_hpr_mixer_controls[0], ARRAY_SIZE(wm9713_hpr_mixer_controls)),
SND_SOC_DAPM_MIXER("Mono Mixer", AC97_EXTENDED_MID, 0, 1,
&wm9713_mono_mixer_controls[0], ARRAY_SIZE(wm9713_mono_mixer_controls)),
SND_SOC_DAPM_MIXER("Speaker Mixer", AC97_EXTENDED_MID, 1, 1,
&wm9713_speaker_mixer_controls[0],
ARRAY_SIZE(wm9713_speaker_mixer_controls)),
SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback", AC97_EXTENDED_MID, 7, 1),
SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback", AC97_EXTENDED_MID, 6, 1),
SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1,
wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1),
SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0),
SND_SOC_DAPM_ADC("Left HiFi ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("Right HiFi ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("Left Voice ADC", "Left Voice Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_ADC("Right Voice ADC", "Right Voice Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA("Left Headphone", AC97_EXTENDED_MSTATUS, 10, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Headphone", AC97_EXTENDED_MSTATUS, 9, 1, NULL, 0),
SND_SOC_DAPM_PGA("Left Speaker", AC97_EXTENDED_MSTATUS, 8, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Speaker", AC97_EXTENDED_MSTATUS, 7, 1, NULL, 0),
SND_SOC_DAPM_PGA("Out 3", AC97_EXTENDED_MSTATUS, 11, 1, NULL, 0),
SND_SOC_DAPM_PGA("Out 4", AC97_EXTENDED_MSTATUS, 12, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mono Out", AC97_EXTENDED_MSTATUS, 13, 1, NULL, 0),
SND_SOC_DAPM_PGA("Left Line In", AC97_EXTENDED_MSTATUS, 6, 1, NULL, 0),
SND_SOC_DAPM_PGA("Right Line In", AC97_EXTENDED_MSTATUS, 5, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mono In", AC97_EXTENDED_MSTATUS, 4, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic A PGA", AC97_EXTENDED_MSTATUS, 3, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic B PGA", AC97_EXTENDED_MSTATUS, 2, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic A Pre Amp", AC97_EXTENDED_MSTATUS, 1, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic B Pre Amp", AC97_EXTENDED_MSTATUS, 0, 1, NULL, 0),
SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_EXTENDED_MSTATUS, 14, 1),
SND_SOC_DAPM_OUTPUT("MONO"),
SND_SOC_DAPM_OUTPUT("HPL"),
SND_SOC_DAPM_OUTPUT("HPR"),
SND_SOC_DAPM_OUTPUT("SPKL"),
SND_SOC_DAPM_OUTPUT("SPKR"),
SND_SOC_DAPM_OUTPUT("OUT3"),
SND_SOC_DAPM_OUTPUT("OUT4"),
SND_SOC_DAPM_INPUT("LINEL"),
SND_SOC_DAPM_INPUT("LINER"),
SND_SOC_DAPM_INPUT("MONOIN"),
SND_SOC_DAPM_INPUT("PCBEEP"),
SND_SOC_DAPM_INPUT("MIC1"),
SND_SOC_DAPM_INPUT("MIC2A"),
SND_SOC_DAPM_INPUT("MIC2B"),
SND_SOC_DAPM_VMID("VMID"),
};
static const struct snd_soc_dapm_route wm9713_audio_map[] = {
/* left HP mixer */
{"Left HP Mixer", "Beep Playback Switch", "PCBEEP"},
{"Left HP Mixer", "Voice Playback Switch", "Voice DAC"},
{"Left HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Left HP Mixer", "Bypass Playback Switch", "Left Line In"},
{"Left HP Mixer", "PCM Playback Switch", "Left DAC"},
{"Left HP Mixer", "MonoIn Playback Switch", "Mono In"},
{"Left HP Mixer", NULL, "Capture Headphone Mux"},
/* right HP mixer */
{"Right HP Mixer", "Beep Playback Switch", "PCBEEP"},
{"Right HP Mixer", "Voice Playback Switch", "Voice DAC"},
{"Right HP Mixer", "Aux Playback Switch", "Aux DAC"},
{"Right HP Mixer", "Bypass Playback Switch", "Right Line In"},
{"Right HP Mixer", "PCM Playback Switch", "Right DAC"},
{"Right HP Mixer", "MonoIn Playback Switch", "Mono In"},
{"Right HP Mixer", NULL, "Capture Headphone Mux"},
/* virtual mixer - mixes left & right channels for spk and mono */
{"AC97 Mixer", NULL, "Left DAC"},
{"AC97 Mixer", NULL, "Right DAC"},
{"Line Mixer", NULL, "Right Line In"},
{"Line Mixer", NULL, "Left Line In"},
{"HP Mixer", NULL, "Left HP Mixer"},
{"HP Mixer", NULL, "Right HP Mixer"},
{"Capture Mixer", NULL, "Left Capture Source"},
{"Capture Mixer", NULL, "Right Capture Source"},
/* speaker mixer */
{"Speaker Mixer", "Beep Playback Switch", "PCBEEP"},
{"Speaker Mixer", "Voice Playback Switch", "Voice DAC"},
{"Speaker Mixer", "Aux Playback Switch", "Aux DAC"},
{"Speaker Mixer", "Bypass Playback Switch", "Line Mixer"},
{"Speaker Mixer", "PCM Playback Switch", "AC97 Mixer"},
{"Speaker Mixer", "MonoIn Playback Switch", "Mono In"},
/* mono mixer */
{"Mono Mixer", "Beep Playback Switch", "PCBEEP"},
{"Mono Mixer", "Voice Playback Switch", "Voice DAC"},
{"Mono Mixer", "Aux Playback Switch", "Aux DAC"},
{"Mono Mixer", "Bypass Playback Switch", "Line Mixer"},
{"Mono Mixer", "PCM Playback Switch", "AC97 Mixer"},
{"Mono Mixer", "Mic 1 Sidetone Switch", "Mic A PGA"},
{"Mono Mixer", "Mic 2 Sidetone Switch", "Mic B PGA"},
{"Mono Mixer", NULL, "Capture Mono Mux"},
/* DAC inv mux 1 */
{"DAC Inv Mux 1", "Mono", "Mono Mixer"},
{"DAC Inv Mux 1", "Speaker", "Speaker Mixer"},
{"DAC Inv Mux 1", "Left Headphone", "Left HP Mixer"},
{"DAC Inv Mux 1", "Right Headphone", "Right HP Mixer"},
{"DAC Inv Mux 1", "Headphone Mono", "HP Mixer"},
/* DAC inv mux 2 */
{"DAC Inv Mux 2", "Mono", "Mono Mixer"},
{"DAC Inv Mux 2", "Speaker", "Speaker Mixer"},
{"DAC Inv Mux 2", "Left Headphone", "Left HP Mixer"},
{"DAC Inv Mux 2", "Right Headphone", "Right HP Mixer"},
{"DAC Inv Mux 2", "Headphone Mono", "HP Mixer"},
/* headphone left mux */
{"Left Headphone Out Mux", "Headphone", "Left HP Mixer"},
/* headphone right mux */
{"Right Headphone Out Mux", "Headphone", "Right HP Mixer"},
/* speaker left mux */
{"Left Speaker Out Mux", "Headphone", "Left HP Mixer"},
{"Left Speaker Out Mux", "Speaker", "Speaker Mixer"},
{"Left Speaker Out Mux", "Inv", "DAC Inv Mux 1"},
/* speaker right mux */
{"Right Speaker Out Mux", "Headphone", "Right HP Mixer"},
{"Right Speaker Out Mux", "Speaker", "Speaker Mixer"},
{"Right Speaker Out Mux", "Inv", "DAC Inv Mux 2"},
/* mono mux */
{"Mono Out Mux", "Mono", "Mono Mixer"},
{"Mono Out Mux", "Inv", "DAC Inv Mux 1"},
/* out 3 mux */
{"Out 3 Mux", "Inv 1", "DAC Inv Mux 1"},
/* out 4 mux */
{"Out 4 Mux", "Inv 2", "DAC Inv Mux 2"},
/* output pga */
{"HPL", NULL, "Left Headphone"},
{"Left Headphone", NULL, "Left Headphone Out Mux"},
{"HPR", NULL, "Right Headphone"},
{"Right Headphone", NULL, "Right Headphone Out Mux"},
{"OUT3", NULL, "Out 3"},
{"Out 3", NULL, "Out 3 Mux"},
{"OUT4", NULL, "Out 4"},
{"Out 4", NULL, "Out 4 Mux"},
{"SPKL", NULL, "Left Speaker"},
{"Left Speaker", NULL, "Left Speaker Out Mux"},
{"SPKR", NULL, "Right Speaker"},
{"Right Speaker", NULL, "Right Speaker Out Mux"},
{"MONO", NULL, "Mono Out"},
{"Mono Out", NULL, "Mono Out Mux"},
/* input pga */
{"Left Line In", NULL, "LINEL"},
{"Right Line In", NULL, "LINER"},
{"Mono In", NULL, "MONOIN"},
{"Mic A PGA", NULL, "Mic A Pre Amp"},
{"Mic B PGA", NULL, "Mic B Pre Amp"},
/* left capture select */
{"Left Capture Source", "Mic 1", "Mic A Pre Amp"},
{"Left Capture Source", "Mic 2", "Mic B Pre Amp"},
{"Left Capture Source", "Line", "LINEL"},
{"Left Capture Source", "Mono In", "MONOIN"},
{"Left Capture Source", "Headphone", "Left HP Mixer"},
{"Left Capture Source", "Speaker", "Speaker Mixer"},
{"Left Capture Source", "Mono Out", "Mono Mixer"},
/* right capture select */
{"Right Capture Source", "Mic 1", "Mic A Pre Amp"},
{"Right Capture Source", "Mic 2", "Mic B Pre Amp"},
{"Right Capture Source", "Line", "LINER"},
{"Right Capture Source", "Mono In", "MONOIN"},
{"Right Capture Source", "Headphone", "Right HP Mixer"},
{"Right Capture Source", "Speaker", "Speaker Mixer"},
{"Right Capture Source", "Mono Out", "Mono Mixer"},
/* left ADC */
{"Left ADC", NULL, "Left Capture Source"},
{"Left Voice ADC", NULL, "Left ADC"},
{"Left HiFi ADC", NULL, "Left ADC"},
/* right ADC */
{"Right ADC", NULL, "Right Capture Source"},
{"Right Voice ADC", NULL, "Right ADC"},
{"Right HiFi ADC", NULL, "Right ADC"},
/* mic */
{"Mic A Pre Amp", NULL, "Mic A Source"},
{"Mic A Source", "Mic 1", "MIC1"},
{"Mic A Source", "Mic 2 A", "MIC2A"},
{"Mic A Source", "Mic 2 B", "Mic B Source"},
{"Mic B Pre Amp", "MPB", "Mic B Source"},
{"Mic B Source", NULL, "MIC2B"},
/* headphone capture */
{"Capture Headphone Mux", "Stereo", "Capture Mixer"},
{"Capture Headphone Mux", "Left", "Left Capture Source"},
{"Capture Headphone Mux", "Right", "Right Capture Source"},
/* mono capture */
{"Capture Mono Mux", "Stereo", "Capture Mixer"},
{"Capture Mono Mux", "Left", "Left Capture Source"},
{"Capture Mono Mux", "Right", "Right Capture Source"},
};
static bool wm9713_readable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AC97_RESET ... AC97_PCM_SURR_DAC_RATE:
case AC97_PCM_LR_ADC_RATE:
case AC97_CENTER_LFE_MASTER:
case AC97_SPDIF ... AC97_LINE1_LEVEL:
case AC97_GPIO_CFG ... 0x5c:
case AC97_CODEC_CLASS_REV ... AC97_PCI_SID:
case 0x74 ... AC97_VENDOR_ID2:
return true;
default:
return false;
}
}
static bool wm9713_writeable_reg(struct device *dev, unsigned int reg)
{
switch (reg) {
case AC97_VENDOR_ID1:
case AC97_VENDOR_ID2:
return false;
default:
return wm9713_readable_reg(dev, reg);
}
}
static const struct reg_default wm9713_reg_defaults[] = {
{ 0x02, 0x8080 }, /* Speaker Output Volume */
{ 0x04, 0x8080 }, /* Headphone Output Volume */
{ 0x06, 0x8080 }, /* Out3/OUT4 Volume */
{ 0x08, 0xc880 }, /* Mono Volume */
{ 0x0a, 0xe808 }, /* LINEIN Volume */
{ 0x0c, 0xe808 }, /* DAC PGA Volume */
{ 0x0e, 0x0808 }, /* MIC PGA Volume */
{ 0x10, 0x00da }, /* MIC Routing Control */
{ 0x12, 0x8000 }, /* Record PGA Volume */
{ 0x14, 0xd600 }, /* Record Routing */
{ 0x16, 0xaaa0 }, /* PCBEEP Volume */
{ 0x18, 0xaaa0 }, /* VxDAC Volume */
{ 0x1a, 0xaaa0 }, /* AUXDAC Volume */
{ 0x1c, 0x0000 }, /* Output PGA Mux */
{ 0x1e, 0x0000 }, /* DAC 3D control */
{ 0x20, 0x0f0f }, /* DAC Tone Control*/
{ 0x22, 0x0040 }, /* MIC Input Select & Bias */
{ 0x24, 0x0000 }, /* Output Volume Mapping & Jack */
{ 0x26, 0x7f00 }, /* Powerdown Ctrl/Stat*/
{ 0x28, 0x0405 }, /* Extended Audio ID */
{ 0x2a, 0x0410 }, /* Extended Audio Start/Ctrl */
{ 0x2c, 0xbb80 }, /* Audio DACs Sample Rate */
{ 0x2e, 0xbb80 }, /* AUXDAC Sample Rate */
{ 0x32, 0xbb80 }, /* Audio ADCs Sample Rate */
{ 0x36, 0x4523 }, /* PCM codec control */
{ 0x3a, 0x2000 }, /* SPDIF control */
{ 0x3c, 0xfdff }, /* Powerdown 1 */
{ 0x3e, 0xffff }, /* Powerdown 2 */
{ 0x40, 0x0000 }, /* General Purpose */
{ 0x42, 0x0000 }, /* Fast Power-Up Control */
{ 0x44, 0x0080 }, /* MCLK/PLL Control */
{ 0x46, 0x0000 }, /* MCLK/PLL Control */
{ 0x4c, 0xfffe }, /* GPIO Pin Configuration */
{ 0x4e, 0xffff }, /* GPIO Pin Polarity / Type */
{ 0x50, 0x0000 }, /* GPIO Pin Sticky */
{ 0x52, 0x0000 }, /* GPIO Pin Wake-Up */
/* GPIO Pin Status */
{ 0x56, 0xfffe }, /* GPIO Pin Sharing */
{ 0x58, 0x4000 }, /* GPIO PullUp/PullDown */
{ 0x5a, 0x0000 }, /* Additional Functions 1 */
{ 0x5c, 0x0000 }, /* Additional Functions 2 */
{ 0x60, 0xb032 }, /* ALC Control */
{ 0x62, 0x3e00 }, /* ALC / Noise Gate Control */
{ 0x64, 0x0000 }, /* AUXDAC input control */
{ 0x74, 0x0000 }, /* Digitiser Reg 1 */
{ 0x76, 0x0006 }, /* Digitiser Reg 2 */
{ 0x78, 0x0001 }, /* Digitiser Reg 3 */
{ 0x7a, 0x0000 }, /* Digitiser Read Back */
};
static const struct regmap_config wm9713_regmap_config = {
.reg_bits = 16,
.reg_stride = 2,
.val_bits = 16,
.max_register = 0x7e,
.cache_type = REGCACHE_RBTREE,
.reg_defaults = wm9713_reg_defaults,
.num_reg_defaults = ARRAY_SIZE(wm9713_reg_defaults),
.volatile_reg = regmap_ac97_default_volatile,
.readable_reg = wm9713_readable_reg,
.writeable_reg = wm9713_writeable_reg,
};
/* PLL divisors */
struct _pll_div {
u32 divsel:1;
u32 divctl:1;
u32 lf:1;
u32 n:4;
u32 k:24;
};
/* The size in bits of the PLL divide multiplied by 10
* to allow rounding later */
#define FIXED_PLL_SIZE ((1 << 22) * 10)
static void pll_factors(struct snd_soc_component *component,
struct _pll_div *pll_div, unsigned int source)
{
u64 Kpart;
unsigned int K, Ndiv, Nmod, target;
/* The the PLL output is always 98.304MHz. */
target = 98304000;
/* If the input frequency is over 14.4MHz then scale it down. */
if (source > 14400000) {
source >>= 1;
pll_div->divsel = 1;
if (source > 14400000) {
source >>= 1;
pll_div->divctl = 1;
} else
pll_div->divctl = 0;
} else {
pll_div->divsel = 0;
pll_div->divctl = 0;
}
/* Low frequency sources require an additional divide in the
* loop.
*/
if (source < 8192000) {
pll_div->lf = 1;
target >>= 2;
} else
pll_div->lf = 0;
Ndiv = target / source;
if ((Ndiv < 5) || (Ndiv > 12))
dev_warn(component->dev,
"WM9713 PLL N value %u out of recommended range!\n",
Ndiv);
pll_div->n = Ndiv;
Nmod = target % source;
Kpart = FIXED_PLL_SIZE * (long long)Nmod;
do_div(Kpart, source);
K = Kpart & 0xFFFFFFFF;
/* Check if we need to round */
if ((K % 10) >= 5)
K += 5;
/* Move down to proper range now rounding is done */
K /= 10;
pll_div->k = K;
}
/**
* Please note that changing the PLL input frequency may require
* resynchronisation with the AC97 controller.
*/
static int wm9713_set_pll(struct snd_soc_component *component,
int pll_id, unsigned int freq_in, unsigned int freq_out)
{
struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component);
u16 reg, reg2;
struct _pll_div pll_div;
/* turn PLL off ? */
if (freq_in == 0) {
/* disable PLL power and select ext source */
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0080, 0x0080);
snd_soc_component_update_bits(component, AC97_EXTENDED_MID, 0x0200, 0x0200);
wm9713->pll_in = 0;
return 0;
}
pll_factors(component, &pll_div, freq_in);
if (pll_div.k == 0) {
reg = (pll_div.n << 12) | (pll_div.lf << 11) |
(pll_div.divsel << 9) | (pll_div.divctl << 8);
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
} else {
/* write the fractional k to the reg 0x46 pages */
reg2 = (pll_div.n << 12) | (pll_div.lf << 11) | (1 << 10) |
(pll_div.divsel << 9) | (pll_div.divctl << 8);
/* K [21:20] */
reg = reg2 | (0x5 << 4) | (pll_div.k >> 20);
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
/* K [19:16] */
reg = reg2 | (0x4 << 4) | ((pll_div.k >> 16) & 0xf);
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
/* K [15:12] */
reg = reg2 | (0x3 << 4) | ((pll_div.k >> 12) & 0xf);
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
/* K [11:8] */
reg = reg2 | (0x2 << 4) | ((pll_div.k >> 8) & 0xf);
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
/* K [7:4] */
reg = reg2 | (0x1 << 4) | ((pll_div.k >> 4) & 0xf);
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
reg = reg2 | (0x0 << 4) | (pll_div.k & 0xf); /* K [3:0] */
snd_soc_component_write(component, AC97_LINE1_LEVEL, reg);
}
/* turn PLL on and select as source */
snd_soc_component_update_bits(component, AC97_EXTENDED_MID, 0x0200, 0x0000);
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0080, 0x0000);
wm9713->pll_in = freq_in;
/* wait 10ms AC97 link frames for the link to stabilise */
schedule_timeout_interruptible(msecs_to_jiffies(10));
return 0;
}
static int wm9713_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
int source, unsigned int freq_in, unsigned int freq_out)
{
struct snd_soc_component *component = codec_dai->component;
return wm9713_set_pll(component, pll_id, freq_in, freq_out);
}
/*
* Tristate the PCM DAI lines, tristate can be disabled by calling
* wm9713_set_dai_fmt()
*/
static int wm9713_set_dai_tristate(struct snd_soc_dai *codec_dai,
int tristate)
{
struct snd_soc_component *component = codec_dai->component;
if (tristate)
snd_soc_component_update_bits(component, AC97_CENTER_LFE_MASTER,
0x6000, 0x0000);
return 0;
}
/*
* Configure WM9713 clock dividers.
* Voice DAC needs 256 FS
*/
static int wm9713_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
int div_id, int div)
{
struct snd_soc_component *component = codec_dai->component;
switch (div_id) {
case WM9713_PCMCLK_DIV:
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0f00, div);
break;
case WM9713_CLKA_MULT:
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0002, div);
break;
case WM9713_CLKB_MULT:
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x0004, div);
break;
case WM9713_HIFI_DIV:
snd_soc_component_update_bits(component, AC97_HANDSET_RATE, 0x7000, div);
break;
case WM9713_PCMBCLK_DIV:
snd_soc_component_update_bits(component, AC97_CENTER_LFE_MASTER, 0x0e00, div);
break;
case WM9713_PCMCLK_PLL_DIV:
snd_soc_component_update_bits(component, AC97_LINE1_LEVEL,
0x007f, div | 0x60);
break;
case WM9713_HIFI_PLL_DIV:
snd_soc_component_update_bits(component, AC97_LINE1_LEVEL,
0x007f, div | 0x70);
break;
default:
return -EINVAL;
}
return 0;
}
static int wm9713_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_component *component = codec_dai->component;
u16 gpio = snd_soc_component_read32(component, AC97_GPIO_CFG) & 0xffc5;
u16 reg = 0x8000;
/* clock masters */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
reg |= 0x4000;
gpio |= 0x0010;
break;
case SND_SOC_DAIFMT_CBM_CFS:
reg |= 0x6000;
gpio |= 0x0018;
break;
case SND_SOC_DAIFMT_CBS_CFS:
reg |= 0x2000;
gpio |= 0x001a;
break;
case SND_SOC_DAIFMT_CBS_CFM:
gpio |= 0x0012;
break;
}
/* clock inversion */
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_IB_IF:
reg |= 0x00c0;
break;
case SND_SOC_DAIFMT_IB_NF:
reg |= 0x0080;
break;
case SND_SOC_DAIFMT_NB_IF:
reg |= 0x0040;
break;
}
/* DAI format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
reg |= 0x0002;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
reg |= 0x0001;
break;
case SND_SOC_DAIFMT_DSP_A:
reg |= 0x0003;
break;
case SND_SOC_DAIFMT_DSP_B:
reg |= 0x0043;
break;
}
snd_soc_component_write(component, AC97_GPIO_CFG, gpio);
snd_soc_component_write(component, AC97_CENTER_LFE_MASTER, reg);
return 0;
}
static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_component *component = dai->component;
/* enable PCM interface in master mode */
switch (params_width(params)) {
case 16:
break;
case 20:
snd_soc_component_update_bits(component, AC97_CENTER_LFE_MASTER,
0x000c, 0x0004);
break;
case 24:
snd_soc_component_update_bits(component, AC97_CENTER_LFE_MASTER,
0x000c, 0x0008);
break;
case 32:
snd_soc_component_update_bits(component, AC97_CENTER_LFE_MASTER,
0x000c, 0x000c);
break;
}
return 0;
}
static int ac97_hifi_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;
int reg;
snd_soc_component_update_bits(component, AC97_EXTENDED_STATUS, 0x0001, 0x0001);
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, 0x0001, 0x0001);
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 WM9713_RATES (SNDRV_PCM_RATE_8000 | \
SNDRV_PCM_RATE_11025 | \
SNDRV_PCM_RATE_22050 | \
SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)
#define WM9713_PCM_RATES (SNDRV_PCM_RATE_8000 | \
SNDRV_PCM_RATE_11025 | \
SNDRV_PCM_RATE_16000 | \
SNDRV_PCM_RATE_22050 | \
SNDRV_PCM_RATE_44100 | \
SNDRV_PCM_RATE_48000)
#define WM9713_PCM_FORMATS \
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
static const struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
.prepare = ac97_hifi_prepare,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
};
static const struct snd_soc_dai_ops wm9713_dai_ops_aux = {
.prepare = ac97_aux_prepare,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
};
static const struct snd_soc_dai_ops wm9713_dai_ops_voice = {
.hw_params = wm9713_pcm_hw_params,
.set_clkdiv = wm9713_set_dai_clkdiv,
.set_pll = wm9713_set_dai_pll,
.set_fmt = wm9713_set_dai_fmt,
.set_tristate = wm9713_set_dai_tristate,
};
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 04:15:21 +08:00
static struct snd_soc_dai_driver wm9713_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>
2010-03-18 04:15:21 +08:00
.name = "wm9713-hifi",
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
.formats = SND_SOC_STD_AC97_FMTS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_RATES,
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_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>
2010-03-18 04:15:21 +08:00
.name = "wm9713-aux",
.playback = {
.stream_name = "Aux Playback",
.channels_min = 1,
.channels_max = 1,
.rates = WM9713_RATES,
.formats = SND_SOC_STD_AC97_FMTS,},
.ops = &wm9713_dai_ops_aux,
},
{
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 04:15:21 +08:00
.name = "wm9713-voice",
.playback = {
.stream_name = "Voice Playback",
.channels_min = 1,
.channels_max = 1,
.rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.capture = {
.stream_name = "Voice Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM9713_PCM_RATES,
.formats = WM9713_PCM_FORMATS,},
.ops = &wm9713_dai_ops_voice,
.symmetric_rates = 1,
},
};
static int wm9713_set_bias_level(struct snd_soc_component *component,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
/* enable thermal shutdown */
snd_soc_component_update_bits(component, AC97_EXTENDED_MID, 0xe400, 0x0000);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* enable master bias and vmid */
snd_soc_component_update_bits(component, AC97_EXTENDED_MID, 0xc400, 0x0000);
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_MID, 0xffff);
snd_soc_component_write(component, AC97_EXTENDED_MSTATUS, 0xffff);
snd_soc_component_write(component, AC97_POWERDOWN, 0xffff);
break;
}
return 0;
}
static int wm9713_soc_suspend(struct snd_soc_component *component)
{
/* Disable everything except touchpanel - that will be handled
* by the touch driver and left disabled if touch is not in
* use. */
snd_soc_component_update_bits(component, AC97_EXTENDED_MID, 0x7fff,
0x7fff);
snd_soc_component_write(component, AC97_EXTENDED_MSTATUS, 0xffff);
snd_soc_component_write(component, AC97_POWERDOWN, 0x6f00);
snd_soc_component_write(component, AC97_POWERDOWN, 0xffff);
return 0;
}
static int wm9713_soc_resume(struct snd_soc_component *component)
{
struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component);
int ret;
ret = snd_ac97_reset(wm9713->ac97, true, WM9713_VENDOR_ID,
WM9713_VENDOR_ID_MASK);
if (ret < 0)
return ret;
snd_soc_component_force_bias_level(component, SND_SOC_BIAS_STANDBY);
/* do we need to re-start the PLL ? */
if (wm9713->pll_in)
wm9713_set_pll(component, 0, wm9713->pll_in, 0);
/* only synchronise the codec if warm reset failed */
if (ret == 0) {
regcache_mark_dirty(component->regmap);
snd_soc_component_cache_sync(component);
}
return ret;
}
static int wm9713_soc_probe(struct snd_soc_component *component)
{
struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component);
struct regmap *regmap = NULL;
if (wm9713->mfd_pdata) {
wm9713->ac97 = wm9713->mfd_pdata->ac97;
regmap = wm9713->mfd_pdata->regmap;
} else if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS)) {
wm9713->ac97 = snd_soc_new_ac97_component(component, WM9713_VENDOR_ID,
WM9713_VENDOR_ID_MASK);
if (IS_ERR(wm9713->ac97))
return PTR_ERR(wm9713->ac97);
regmap = regmap_init_ac97(wm9713->ac97, &wm9713_regmap_config);
if (IS_ERR(regmap)) {
snd_soc_free_ac97_component(wm9713->ac97);
return PTR_ERR(regmap);
}
} else {
return -ENXIO;
}
snd_soc_component_init_regmap(component, regmap);
/* unmute the adc - move to kcontrol */
snd_soc_component_update_bits(component, AC97_CD, 0x7fff, 0x0000);
return 0;
}
static void wm9713_soc_remove(struct snd_soc_component *component)
{
struct wm9713_priv *wm9713 = snd_soc_component_get_drvdata(component);
if (IS_ENABLED(CONFIG_SND_SOC_AC97_BUS) && !wm9713->mfd_pdata) {
snd_soc_component_exit_regmap(component);
snd_soc_free_ac97_component(wm9713->ac97);
}
}
static const struct snd_soc_component_driver soc_component_dev_wm9713 = {
.probe = wm9713_soc_probe,
.remove = wm9713_soc_remove,
.suspend = wm9713_soc_suspend,
.resume = wm9713_soc_resume,
.set_bias_level = wm9713_set_bias_level,
.controls = wm9713_snd_ac97_controls,
.num_controls = ARRAY_SIZE(wm9713_snd_ac97_controls),
.dapm_widgets = wm9713_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm9713_dapm_widgets),
.dapm_routes = wm9713_audio_map,
.num_dapm_routes = ARRAY_SIZE(wm9713_audio_map),
.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>
2010-03-18 04:15:21 +08:00
};
static int wm9713_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>
2010-03-18 04:15:21 +08:00
{
struct wm9713_priv *wm9713;
wm9713 = devm_kzalloc(&pdev->dev, sizeof(*wm9713), GFP_KERNEL);
if (wm9713 == NULL)
return -ENOMEM;
mutex_init(&wm9713->lock);
wm9713->mfd_pdata = dev_get_platdata(&pdev->dev);
platform_set_drvdata(pdev, wm9713);
return devm_snd_soc_register_component(&pdev->dev,
&soc_component_dev_wm9713, wm9713_dai, ARRAY_SIZE(wm9713_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>
2010-03-18 04:15:21 +08:00
}
static struct platform_driver wm9713_codec_driver = {
.driver = {
.name = "wm9713-codec",
},
.probe = wm9713_probe,
};
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 04:15:21 +08:00
module_platform_driver(wm9713_codec_driver);
MODULE_DESCRIPTION("ASoC WM9713/WM9714 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");