kevin/linux
1
Fork 0
Code
e231cab0a4
linux/sound/soc/codecs/wm9712.c
Liam Girdwood f0fba2ad1b 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>
2010-08-12 14:00:00 +01:00

720 lines
23 KiB
C
Raw Blame History

/*
* wm9712.c -- ALSA Soc WM9712 codec support
*
* Copyright 2006 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 <linux/slab.h>
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include "wm9712.h"
#define WM9712_VERSION "0.4"
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg);
static int ac97_write(struct snd_soc_codec *codec,
unsigned int reg, unsigned int val);
/*
* WM9712 register cache
*/
static const u16 wm9712_reg[] = {
0x6174, 0x8000, 0x8000, 0x8000, /* 6 */
0x0f0f, 0xaaa0, 0xc008, 0x6808, /* e */
0xe808, 0xaaa0, 0xad00, 0x8000, /* 16 */
0xe808, 0x3000, 0x8000, 0x0000, /* 1e */
0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
0x0405, 0x0410, 0xbb80, 0xbb80, /* 2e */
0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
0x0000, 0x2000, 0x0000, 0x0000, /* 3e */
0x0000, 0x0000, 0x0000, 0x0000, /* 46 */
0x0000, 0x0000, 0xf83e, 0xffff, /* 4e */
0x0000, 0x0000, 0x0000, 0xf83e, /* 56 */
0x0008, 0x0000, 0x0000, 0x0000, /* 5e */
0xb032, 0x3e00, 0x0000, 0x0000, /* 66 */
0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
0x0001, 0x0000, 0x574d, 0x4c12, /* 7e */
0x0000, 0x0000 /* virtual hp mixers */
};
/* virtual HP mixers regs */
#define HPL_MIXER 0x80
#define HPR_MIXER 0x82
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 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("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0),
SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1),
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 ADC 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, 1),
SOC_SINGLE("Capture ZC Switch", AC97_REC_GAIN, 7, 1, 0),
SOC_SINGLE("Mic 1 Volume", AC97_MIC, 8, 31, 1),
SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
};
/* 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 mixer_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
u16 l, r, beep, line, phone, mic, pcm, aux;
l = ac97_read(w->codec, HPL_MIXER);
r = ac97_read(w->codec, HPR_MIXER);
beep = ac97_read(w->codec, AC97_PC_BEEP);
mic = ac97_read(w->codec, AC97_VIDEO);
phone = ac97_read(w->codec, AC97_PHONE);
line = ac97_read(w->codec, AC97_LINE);
pcm = ac97_read(w->codec, AC97_PCM);
aux = ac97_read(w->codec, AC97_CD);
if (l & 0x1 || r & 0x1)
ac97_write(w->codec, AC97_VIDEO, mic & 0x7fff);
else
ac97_write(w->codec, AC97_VIDEO, mic | 0x8000);
if (l & 0x2 || r & 0x2)
ac97_write(w->codec, AC97_PCM, pcm & 0x7fff);
else
ac97_write(w->codec, AC97_PCM, pcm | 0x8000);
if (l & 0x4 || r & 0x4)
ac97_write(w->codec, AC97_LINE, line & 0x7fff);
else
ac97_write(w->codec, AC97_LINE, line | 0x8000);
if (l & 0x8 || r & 0x8)
ac97_write(w->codec, AC97_PHONE, phone & 0x7fff);
else
ac97_write(w->codec, AC97_PHONE, phone | 0x8000);
if (l & 0x10 || r & 0x10)
ac97_write(w->codec, AC97_CD, aux & 0x7fff);
else
ac97_write(w->codec, AC97_CD, aux | 0x8000);
if (l & 0x20 || r & 0x20)
ac97_write(w->codec, AC97_PC_BEEP, beep & 0x7fff);
else
ac97_write(w->codec, AC97_PC_BEEP, beep | 0x8000);
return 0;
}
/* Left Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpl_mixer_controls[] = {
SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPL_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", HPL_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Phone Bypass Switch", HPL_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("Line Bypass Switch", HPL_MIXER, 2, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", HPL_MIXER, 1, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", HPL_MIXER, 0, 1, 0),
};
/* Right Headphone Mixers */
static const struct snd_kcontrol_new wm9712_hpr_mixer_controls[] = {
SOC_DAPM_SINGLE("PCBeep Bypass Switch", HPR_MIXER, 5, 1, 0),
SOC_DAPM_SINGLE("Aux Playback Switch", HPR_MIXER, 4, 1, 0),
SOC_DAPM_SINGLE("Phone Bypass Switch", HPR_MIXER, 3, 1, 0),
SOC_DAPM_SINGLE("Line Bypass Switch", HPR_MIXER, 2, 1, 0),
SOC_DAPM_SINGLE("PCM Playback Switch", HPR_MIXER, 1, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", HPR_MIXER, 0, 1, 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("Route", 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("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_E("Left HP Mixer", AC97_INT_PAGING, 9, 1,
&wm9712_hpl_mixer_controls[0], ARRAY_SIZE(wm9712_hpl_mixer_controls),
mixer_event, SND_SOC_DAPM_POST_REG),
SND_SOC_DAPM_MIXER_E("Right HP Mixer", AC97_INT_PAGING, 8, 1,
&wm9712_hpr_mixer_controls[0], ARRAY_SIZE(wm9712_hpr_mixer_controls),
mixer_event, SND_SOC_DAPM_POST_REG),
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_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 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"},
/* 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 wm9712_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, wm9712_dapm_widgets,
ARRAY_SIZE(wm9712_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
return 0;
}
static unsigned int ac97_read(struct snd_soc_codec *codec,
unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2 ||
reg == AC97_REC_GAIN)
return soc_ac97_ops.read(codec->ac97, reg);
else {
reg = reg >> 1;
if (reg >= (ARRAY_SIZE(wm9712_reg)))
return -EIO;
return cache[reg];
}
}
static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int val)
{
u16 *cache = codec->reg_cache;
if (reg < 0x7c)
soc_ac97_ops.write(codec->ac97, reg, val);
reg = reg >> 1;
if (reg < (ARRAY_SIZE(wm9712_reg)))
cache[reg] = val;
return 0;
}
static int ac97_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec =rtd->codec;
int reg;
u16 vra;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
reg = AC97_PCM_FRONT_DAC_RATE;
else
reg = AC97_PCM_LR_ADC_RATE;
return ac97_write(codec, reg, runtime->rate);
}
static int ac97_aux_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
u16 vra, xsle;
vra = ac97_read(codec, AC97_EXTENDED_STATUS);
ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
xsle = ac97_read(codec, AC97_PCI_SID);
ac97_write(codec, AC97_PCI_SID, xsle | 0x8000);
if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK)
return -ENODEV;
return ac97_write(codec, 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 struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
.prepare = ac97_prepare,
};
static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
.prepare = ac97_aux_prepare,
};
struct snd_soc_dai_driver wm9712_dai[] = {
{
.name = "wm9712-hifi",
.ac97_control = 1,
.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,
},
{
.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_codec *codec,
enum snd_soc_bias_level level)
{
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
ac97_write(codec, AC97_POWERDOWN, 0x0000);
break;
case SND_SOC_BIAS_OFF:
/* disable everything including AC link */
ac97_write(codec, AC97_EXTENDED_MSTATUS, 0xffff);
ac97_write(codec, AC97_POWERDOWN, 0xffff);
break;
}
codec->bias_level = level;
return 0;
}
static int wm9712_reset(struct snd_soc_codec *codec, int try_warm)
{
if (try_warm && soc_ac97_ops.warm_reset) {
soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, 0) == wm9712_reg[0])
return 1;
}
soc_ac97_ops.reset(codec->ac97);
if (soc_ac97_ops.warm_reset)
soc_ac97_ops.warm_reset(codec->ac97);
if (ac97_read(codec, 0) != wm9712_reg[0])
goto err;
return 0;
err:
printk(KERN_ERR "WM9712 AC97 reset failed\n");
return -EIO;
}
static int wm9712_soc_suspend(struct snd_soc_codec *codec,
pm_message_t state)
{
wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int wm9712_soc_resume(struct snd_soc_codec *codec)
{
int i, ret;
u16 *cache = codec->reg_cache;
ret = wm9712_reset(codec, 1);
if (ret < 0) {
printk(KERN_ERR "could not reset AC97 codec\n");
return ret;
}
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
if (ret == 0) {
/* Sync reg_cache with the hardware after cold reset */
for (i = 2; i < ARRAY_SIZE(wm9712_reg) << 1; i += 2) {
if (i == AC97_INT_PAGING || i == AC97_POWERDOWN ||
(i > 0x58 && i != 0x5c))
continue;
soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
}
}
return ret;
}
static int wm9712_soc_probe(struct snd_soc_codec *codec)
{
int ret = 0;
printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
if (ret < 0) {
printk(KERN_ERR "wm9712: failed to register AC97 codec\n");
return ret;
}
ret = wm9712_reset(codec, 0);
if (ret < 0) {
printk(KERN_ERR "Failed to reset WM9712: AC97 link error\n");
goto reset_err;
}
/* set alc mux to none */
ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
ARRAY_SIZE(wm9712_snd_ac97_controls));
wm9712_add_widgets(codec);
return 0;
reset_err:
snd_soc_free_ac97_codec(codec);
return ret;
}
static int wm9712_soc_remove(struct snd_soc_codec *codec)
{
snd_soc_free_ac97_codec(codec);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_wm9712 = {
.probe = wm9712_soc_probe,
.remove = wm9712_soc_remove,
.suspend = wm9712_soc_suspend,
.resume = wm9712_soc_resume,
.read = ac97_read,
.write = ac97_write,
.set_bias_level = wm9712_set_bias_level,
.reg_cache_size = sizeof(wm9712_reg),
.reg_word_size = sizeof(u16),
.reg_cache_step = 2,
.reg_cache_default = wm9712_reg,
};
static __devinit int wm9712_probe(struct platform_device *pdev)
{
return snd_soc_register_codec(&pdev->dev,
&soc_codec_dev_wm9712, wm9712_dai, ARRAY_SIZE(wm9712_dai));
}
static int __devexit wm9712_remove(struct platform_device *pdev)
{
snd_soc_unregister_codec(&pdev->dev);
return 0;
}
static struct platform_driver wm9712_codec_driver = {
.driver = {
.name = "wm9712-codec",
.owner = THIS_MODULE,
},
.probe = wm9712_probe,
.remove = __devexit_p(wm9712_remove),
};
static int __init wm9712_init(void)
{
return platform_driver_register(&wm9712_codec_driver);
}
module_init(wm9712_init);
static void __exit wm9712_exit(void)
{
platform_driver_unregister(&wm9712_codec_driver);
}
module_exit(wm9712_exit);
MODULE_DESCRIPTION("ASoC WM9711/WM9712 driver");
MODULE_AUTHOR("Liam Girdwood");
MODULE_LICENSE("GPL");
View Git Blame Copy Permalink