linux/sound/soc/codecs/wm8961.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

1157 lines
30 KiB
C

/*
* wm8961.c -- WM8961 ALSA SoC Audio driver
*
* Author: Mark Brown
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License version 2 as
* published by the Free Software Foundation.
*
* Currently unimplemented features:
* - ALC
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/init.h>
#include <linux/delay.h>
#include <linux/pm.h>
#include <linux/i2c.h>
#include <linux/platform_device.h>
#include <linux/slab.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/initval.h>
#include <sound/tlv.h>
#include "wm8961.h"
#define WM8961_MAX_REGISTER 0xFC
static u16 wm8961_reg_defaults[] = {
0x009F, /* R0 - Left Input volume */
0x009F, /* R1 - Right Input volume */
0x0000, /* R2 - LOUT1 volume */
0x0000, /* R3 - ROUT1 volume */
0x0020, /* R4 - Clocking1 */
0x0008, /* R5 - ADC & DAC Control 1 */
0x0000, /* R6 - ADC & DAC Control 2 */
0x000A, /* R7 - Audio Interface 0 */
0x01F4, /* R8 - Clocking2 */
0x0000, /* R9 - Audio Interface 1 */
0x00FF, /* R10 - Left DAC volume */
0x00FF, /* R11 - Right DAC volume */
0x0000, /* R12 */
0x0000, /* R13 */
0x0040, /* R14 - Audio Interface 2 */
0x0000, /* R15 - Software Reset */
0x0000, /* R16 */
0x007B, /* R17 - ALC1 */
0x0000, /* R18 - ALC2 */
0x0032, /* R19 - ALC3 */
0x0000, /* R20 - Noise Gate */
0x00C0, /* R21 - Left ADC volume */
0x00C0, /* R22 - Right ADC volume */
0x0120, /* R23 - Additional control(1) */
0x0000, /* R24 - Additional control(2) */
0x0000, /* R25 - Pwr Mgmt (1) */
0x0000, /* R26 - Pwr Mgmt (2) */
0x0000, /* R27 - Additional Control (3) */
0x0000, /* R28 - Anti-pop */
0x0000, /* R29 */
0x005F, /* R30 - Clocking 3 */
0x0000, /* R31 */
0x0000, /* R32 - ADCL signal path */
0x0000, /* R33 - ADCR signal path */
0x0000, /* R34 */
0x0000, /* R35 */
0x0000, /* R36 */
0x0000, /* R37 */
0x0000, /* R38 */
0x0000, /* R39 */
0x0000, /* R40 - LOUT2 volume */
0x0000, /* R41 - ROUT2 volume */
0x0000, /* R42 */
0x0000, /* R43 */
0x0000, /* R44 */
0x0000, /* R45 */
0x0000, /* R46 */
0x0000, /* R47 - Pwr Mgmt (3) */
0x0023, /* R48 - Additional Control (4) */
0x0000, /* R49 - Class D Control 1 */
0x0000, /* R50 */
0x0003, /* R51 - Class D Control 2 */
0x0000, /* R52 */
0x0000, /* R53 */
0x0000, /* R54 */
0x0000, /* R55 */
0x0106, /* R56 - Clocking 4 */
0x0000, /* R57 - DSP Sidetone 0 */
0x0000, /* R58 - DSP Sidetone 1 */
0x0000, /* R59 */
0x0000, /* R60 - DC Servo 0 */
0x0000, /* R61 - DC Servo 1 */
0x0000, /* R62 */
0x015E, /* R63 - DC Servo 3 */
0x0010, /* R64 */
0x0010, /* R65 - DC Servo 5 */
0x0000, /* R66 */
0x0001, /* R67 */
0x0003, /* R68 - Analogue PGA Bias */
0x0000, /* R69 - Analogue HP 0 */
0x0060, /* R70 */
0x01FB, /* R71 - Analogue HP 2 */
0x0000, /* R72 - Charge Pump 1 */
0x0065, /* R73 */
0x005F, /* R74 */
0x0059, /* R75 */
0x006B, /* R76 */
0x0038, /* R77 */
0x000C, /* R78 */
0x000A, /* R79 */
0x006B, /* R80 */
0x0000, /* R81 */
0x0000, /* R82 - Charge Pump B */
0x0087, /* R83 */
0x0000, /* R84 */
0x005C, /* R85 */
0x0000, /* R86 */
0x0000, /* R87 - Write Sequencer 1 */
0x0000, /* R88 - Write Sequencer 2 */
0x0000, /* R89 - Write Sequencer 3 */
0x0000, /* R90 - Write Sequencer 4 */
0x0000, /* R91 - Write Sequencer 5 */
0x0000, /* R92 - Write Sequencer 6 */
0x0000, /* R93 - Write Sequencer 7 */
0x0000, /* R94 */
0x0000, /* R95 */
0x0000, /* R96 */
0x0000, /* R97 */
0x0000, /* R98 */
0x0000, /* R99 */
0x0000, /* R100 */
0x0000, /* R101 */
0x0000, /* R102 */
0x0000, /* R103 */
0x0000, /* R104 */
0x0000, /* R105 */
0x0000, /* R106 */
0x0000, /* R107 */
0x0000, /* R108 */
0x0000, /* R109 */
0x0000, /* R110 */
0x0000, /* R111 */
0x0000, /* R112 */
0x0000, /* R113 */
0x0000, /* R114 */
0x0000, /* R115 */
0x0000, /* R116 */
0x0000, /* R117 */
0x0000, /* R118 */
0x0000, /* R119 */
0x0000, /* R120 */
0x0000, /* R121 */
0x0000, /* R122 */
0x0000, /* R123 */
0x0000, /* R124 */
0x0000, /* R125 */
0x0000, /* R126 */
0x0000, /* R127 */
0x0000, /* R128 */
0x0000, /* R129 */
0x0000, /* R130 */
0x0000, /* R131 */
0x0000, /* R132 */
0x0000, /* R133 */
0x0000, /* R134 */
0x0000, /* R135 */
0x0000, /* R136 */
0x0000, /* R137 */
0x0000, /* R138 */
0x0000, /* R139 */
0x0000, /* R140 */
0x0000, /* R141 */
0x0000, /* R142 */
0x0000, /* R143 */
0x0000, /* R144 */
0x0000, /* R145 */
0x0000, /* R146 */
0x0000, /* R147 */
0x0000, /* R148 */
0x0000, /* R149 */
0x0000, /* R150 */
0x0000, /* R151 */
0x0000, /* R152 */
0x0000, /* R153 */
0x0000, /* R154 */
0x0000, /* R155 */
0x0000, /* R156 */
0x0000, /* R157 */
0x0000, /* R158 */
0x0000, /* R159 */
0x0000, /* R160 */
0x0000, /* R161 */
0x0000, /* R162 */
0x0000, /* R163 */
0x0000, /* R164 */
0x0000, /* R165 */
0x0000, /* R166 */
0x0000, /* R167 */
0x0000, /* R168 */
0x0000, /* R169 */
0x0000, /* R170 */
0x0000, /* R171 */
0x0000, /* R172 */
0x0000, /* R173 */
0x0000, /* R174 */
0x0000, /* R175 */
0x0000, /* R176 */
0x0000, /* R177 */
0x0000, /* R178 */
0x0000, /* R179 */
0x0000, /* R180 */
0x0000, /* R181 */
0x0000, /* R182 */
0x0000, /* R183 */
0x0000, /* R184 */
0x0000, /* R185 */
0x0000, /* R186 */
0x0000, /* R187 */
0x0000, /* R188 */
0x0000, /* R189 */
0x0000, /* R190 */
0x0000, /* R191 */
0x0000, /* R192 */
0x0000, /* R193 */
0x0000, /* R194 */
0x0000, /* R195 */
0x0030, /* R196 */
0x0006, /* R197 */
0x0000, /* R198 */
0x0060, /* R199 */
0x0000, /* R200 */
0x003F, /* R201 */
0x0000, /* R202 */
0x0000, /* R203 */
0x0000, /* R204 */
0x0001, /* R205 */
0x0000, /* R206 */
0x0181, /* R207 */
0x0005, /* R208 */
0x0008, /* R209 */
0x0008, /* R210 */
0x0000, /* R211 */
0x013B, /* R212 */
0x0000, /* R213 */
0x0000, /* R214 */
0x0000, /* R215 */
0x0000, /* R216 */
0x0070, /* R217 */
0x0000, /* R218 */
0x0000, /* R219 */
0x0000, /* R220 */
0x0000, /* R221 */
0x0000, /* R222 */
0x0003, /* R223 */
0x0000, /* R224 */
0x0000, /* R225 */
0x0001, /* R226 */
0x0008, /* R227 */
0x0000, /* R228 */
0x0000, /* R229 */
0x0000, /* R230 */
0x0000, /* R231 */
0x0004, /* R232 */
0x0000, /* R233 */
0x0000, /* R234 */
0x0000, /* R235 */
0x0000, /* R236 */
0x0000, /* R237 */
0x0080, /* R238 */
0x0000, /* R239 */
0x0000, /* R240 */
0x0000, /* R241 */
0x0000, /* R242 */
0x0000, /* R243 */
0x0000, /* R244 */
0x0052, /* R245 */
0x0110, /* R246 */
0x0040, /* R247 */
0x0000, /* R248 */
0x0030, /* R249 */
0x0000, /* R250 */
0x0000, /* R251 */
0x0001, /* R252 - General test 1 */
};
struct wm8961_priv {
enum snd_soc_control_type control_type;
void *control_data;
int sysclk;
u16 reg_cache[WM8961_MAX_REGISTER];
};
static int wm8961_volatile_register(unsigned int reg)
{
switch (reg) {
case WM8961_SOFTWARE_RESET:
case WM8961_WRITE_SEQUENCER_7:
case WM8961_DC_SERVO_1:
return 1;
default:
return 0;
}
}
static int wm8961_reset(struct snd_soc_codec *codec)
{
return snd_soc_write(codec, WM8961_SOFTWARE_RESET, 0);
}
/*
* The headphone output supports special anti-pop sequences giving
* silent power up and power down.
*/
static int wm8961_hp_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
u16 hp_reg = snd_soc_read(codec, WM8961_ANALOGUE_HP_0);
u16 cp_reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_1);
u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2);
u16 dcs_reg = snd_soc_read(codec, WM8961_DC_SERVO_1);
int timeout = 500;
if (event & SND_SOC_DAPM_POST_PMU) {
/* Make sure the output is shorted */
hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT);
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Enable the charge pump */
cp_reg |= WM8961_CP_ENA;
snd_soc_write(codec, WM8961_CHARGE_PUMP_1, cp_reg);
mdelay(5);
/* Enable the PGA */
pwr_reg |= WM8961_LOUT1_PGA | WM8961_ROUT1_PGA;
snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
/* Enable the amplifier */
hp_reg |= WM8961_HPR_ENA | WM8961_HPL_ENA;
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Second stage enable */
hp_reg |= WM8961_HPR_ENA_DLY | WM8961_HPL_ENA_DLY;
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Enable the DC servo & trigger startup */
dcs_reg |=
WM8961_DCS_ENA_CHAN_HPR | WM8961_DCS_TRIG_STARTUP_HPR |
WM8961_DCS_ENA_CHAN_HPL | WM8961_DCS_TRIG_STARTUP_HPL;
dev_dbg(codec->dev, "Enabling DC servo\n");
snd_soc_write(codec, WM8961_DC_SERVO_1, dcs_reg);
do {
msleep(1);
dcs_reg = snd_soc_read(codec, WM8961_DC_SERVO_1);
} while (--timeout &&
dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR |
WM8961_DCS_TRIG_STARTUP_HPL));
if (dcs_reg & (WM8961_DCS_TRIG_STARTUP_HPR |
WM8961_DCS_TRIG_STARTUP_HPL))
dev_err(codec->dev, "DC servo timed out\n");
else
dev_dbg(codec->dev, "DC servo startup complete\n");
/* Enable the output stage */
hp_reg |= WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP;
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Remove the short on the output stage */
hp_reg |= WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT;
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
}
if (event & SND_SOC_DAPM_PRE_PMD) {
/* Short the output */
hp_reg &= ~(WM8961_HPR_RMV_SHORT | WM8961_HPL_RMV_SHORT);
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Disable the output stage */
hp_reg &= ~(WM8961_HPR_ENA_OUTP | WM8961_HPL_ENA_OUTP);
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Disable DC offset cancellation */
dcs_reg &= ~(WM8961_DCS_ENA_CHAN_HPR |
WM8961_DCS_ENA_CHAN_HPL);
snd_soc_write(codec, WM8961_DC_SERVO_1, dcs_reg);
/* Finish up */
hp_reg &= ~(WM8961_HPR_ENA_DLY | WM8961_HPR_ENA |
WM8961_HPL_ENA_DLY | WM8961_HPL_ENA);
snd_soc_write(codec, WM8961_ANALOGUE_HP_0, hp_reg);
/* Disable the PGA */
pwr_reg &= ~(WM8961_LOUT1_PGA | WM8961_ROUT1_PGA);
snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
/* Disable the charge pump */
dev_dbg(codec->dev, "Disabling charge pump\n");
snd_soc_write(codec, WM8961_CHARGE_PUMP_1,
cp_reg & ~WM8961_CP_ENA);
}
return 0;
}
static int wm8961_spk_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *kcontrol, int event)
{
struct snd_soc_codec *codec = w->codec;
u16 pwr_reg = snd_soc_read(codec, WM8961_PWR_MGMT_2);
u16 spk_reg = snd_soc_read(codec, WM8961_CLASS_D_CONTROL_1);
if (event & SND_SOC_DAPM_POST_PMU) {
/* Enable the PGA */
pwr_reg |= WM8961_SPKL_PGA | WM8961_SPKR_PGA;
snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
/* Enable the amplifier */
spk_reg |= WM8961_SPKL_ENA | WM8961_SPKR_ENA;
snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg);
}
if (event & SND_SOC_DAPM_PRE_PMD) {
/* Enable the amplifier */
spk_reg &= ~(WM8961_SPKL_ENA | WM8961_SPKR_ENA);
snd_soc_write(codec, WM8961_CLASS_D_CONTROL_1, spk_reg);
/* Enable the PGA */
pwr_reg &= ~(WM8961_SPKL_PGA | WM8961_SPKR_PGA);
snd_soc_write(codec, WM8961_PWR_MGMT_2, pwr_reg);
}
return 0;
}
static const char *adc_hpf_text[] = {
"Hi-fi", "Voice 1", "Voice 2", "Voice 3",
};
static const struct soc_enum adc_hpf =
SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text);
static const char *dac_deemph_text[] = {
"None", "32kHz", "44.1kHz", "48kHz",
};
static const struct soc_enum dac_deemph =
SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text);
static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1);
static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0);
static const DECLARE_TLV_DB_SCALE(adc_tlv, -7200, 75, 1);
static const DECLARE_TLV_DB_SCALE(sidetone_tlv, -3600, 300, 0);
static unsigned int boost_tlv[] = {
TLV_DB_RANGE_HEAD(4),
0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
1, 1, TLV_DB_SCALE_ITEM(13, 0, 0),
2, 2, TLV_DB_SCALE_ITEM(20, 0, 0),
3, 3, TLV_DB_SCALE_ITEM(29, 0, 0),
};
static const DECLARE_TLV_DB_SCALE(pga_tlv, -2325, 75, 0);
static const struct snd_kcontrol_new wm8961_snd_controls[] = {
SOC_DOUBLE_R_TLV("Headphone Volume", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME,
0, 127, 0, out_tlv),
SOC_DOUBLE_TLV("Headphone Secondary Volume", WM8961_ANALOGUE_HP_2,
6, 3, 7, 0, hp_sec_tlv),
SOC_DOUBLE_R("Headphone ZC Switch", WM8961_LOUT1_VOLUME, WM8961_ROUT1_VOLUME,
7, 1, 0),
SOC_DOUBLE_R_TLV("Speaker Volume", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME,
0, 127, 0, out_tlv),
SOC_DOUBLE_R("Speaker ZC Switch", WM8961_LOUT2_VOLUME, WM8961_ROUT2_VOLUME,
7, 1, 0),
SOC_SINGLE("Speaker AC Gain", WM8961_CLASS_D_CONTROL_2, 0, 7, 0),
SOC_SINGLE("DAC x128 OSR Switch", WM8961_ADC_DAC_CONTROL_2, 0, 1, 0),
SOC_ENUM("DAC Deemphasis", dac_deemph),
SOC_SINGLE("DAC Soft Mute Switch", WM8961_ADC_DAC_CONTROL_2, 3, 1, 0),
SOC_DOUBLE_R_TLV("Sidetone Volume", WM8961_DSP_SIDETONE_0,
WM8961_DSP_SIDETONE_1, 4, 12, 0, sidetone_tlv),
SOC_SINGLE("ADC High Pass Filter Switch", WM8961_ADC_DAC_CONTROL_1, 0, 1, 0),
SOC_ENUM("ADC High Pass Filter Mode", adc_hpf),
SOC_DOUBLE_R_TLV("Capture Volume",
WM8961_LEFT_ADC_VOLUME, WM8961_RIGHT_ADC_VOLUME,
1, 119, 0, adc_tlv),
SOC_DOUBLE_R_TLV("Capture Boost Volume",
WM8961_ADCL_SIGNAL_PATH, WM8961_ADCR_SIGNAL_PATH,
4, 3, 0, boost_tlv),
SOC_DOUBLE_R_TLV("Capture PGA Volume",
WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
0, 62, 0, pga_tlv),
SOC_DOUBLE_R("Capture PGA ZC Switch",
WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
6, 1, 1),
SOC_DOUBLE_R("Capture PGA Switch",
WM8961_LEFT_INPUT_VOLUME, WM8961_RIGHT_INPUT_VOLUME,
7, 1, 1),
};
static const char *sidetone_text[] = {
"None", "Left", "Right"
};
static const struct soc_enum dacl_sidetone =
SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text);
static const struct soc_enum dacr_sidetone =
SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text);
static const struct snd_kcontrol_new dacl_mux =
SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone);
static const struct snd_kcontrol_new dacr_mux =
SOC_DAPM_ENUM("DACR Sidetone", dacr_sidetone);
static const struct snd_soc_dapm_widget wm8961_dapm_widgets[] = {
SND_SOC_DAPM_INPUT("LINPUT"),
SND_SOC_DAPM_INPUT("RINPUT"),
SND_SOC_DAPM_SUPPLY("CLK_DSP", WM8961_CLOCKING2, 4, 0, NULL, 0),
SND_SOC_DAPM_PGA("Left Input", WM8961_PWR_MGMT_1, 5, 0, NULL, 0),
SND_SOC_DAPM_PGA("Right Input", WM8961_PWR_MGMT_1, 4, 0, NULL, 0),
SND_SOC_DAPM_ADC("ADCL", "HiFi Capture", WM8961_PWR_MGMT_1, 3, 0),
SND_SOC_DAPM_ADC("ADCR", "HiFi Capture", WM8961_PWR_MGMT_1, 2, 0),
SND_SOC_DAPM_MICBIAS("MICBIAS", WM8961_PWR_MGMT_1, 1, 0),
SND_SOC_DAPM_MUX("DACL Sidetone", SND_SOC_NOPM, 0, 0, &dacl_mux),
SND_SOC_DAPM_MUX("DACR Sidetone", SND_SOC_NOPM, 0, 0, &dacr_mux),
SND_SOC_DAPM_DAC("DACL", "HiFi Playback", WM8961_PWR_MGMT_2, 8, 0),
SND_SOC_DAPM_DAC("DACR", "HiFi Playback", WM8961_PWR_MGMT_2, 7, 0),
/* Handle as a mono path for DCS */
SND_SOC_DAPM_PGA_E("Headphone Output", SND_SOC_NOPM,
4, 0, NULL, 0, wm8961_hp_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_PGA_E("Speaker Output", SND_SOC_NOPM,
4, 0, NULL, 0, wm8961_spk_event,
SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
SND_SOC_DAPM_OUTPUT("HP_L"),
SND_SOC_DAPM_OUTPUT("HP_R"),
SND_SOC_DAPM_OUTPUT("SPK_LN"),
SND_SOC_DAPM_OUTPUT("SPK_LP"),
SND_SOC_DAPM_OUTPUT("SPK_RN"),
SND_SOC_DAPM_OUTPUT("SPK_RP"),
};
static const struct snd_soc_dapm_route audio_paths[] = {
{ "DACL", NULL, "CLK_DSP" },
{ "DACL", NULL, "DACL Sidetone" },
{ "DACR", NULL, "CLK_DSP" },
{ "DACR", NULL, "DACR Sidetone" },
{ "DACL Sidetone", "Left", "ADCL" },
{ "DACL Sidetone", "Right", "ADCR" },
{ "DACR Sidetone", "Left", "ADCL" },
{ "DACR Sidetone", "Right", "ADCR" },
{ "HP_L", NULL, "Headphone Output" },
{ "HP_R", NULL, "Headphone Output" },
{ "Headphone Output", NULL, "DACL" },
{ "Headphone Output", NULL, "DACR" },
{ "SPK_LN", NULL, "Speaker Output" },
{ "SPK_LP", NULL, "Speaker Output" },
{ "SPK_RN", NULL, "Speaker Output" },
{ "SPK_RP", NULL, "Speaker Output" },
{ "Speaker Output", NULL, "DACL" },
{ "Speaker Output", NULL, "DACR" },
{ "ADCL", NULL, "Left Input" },
{ "ADCL", NULL, "CLK_DSP" },
{ "ADCR", NULL, "Right Input" },
{ "ADCR", NULL, "CLK_DSP" },
{ "Left Input", NULL, "LINPUT" },
{ "Right Input", NULL, "RINPUT" },
};
/* Values for CLK_SYS_RATE */
static struct {
int ratio;
u16 val;
} wm8961_clk_sys_ratio[] = {
{ 64, 0 },
{ 128, 1 },
{ 192, 2 },
{ 256, 3 },
{ 384, 4 },
{ 512, 5 },
{ 768, 6 },
{ 1024, 7 },
{ 1408, 8 },
{ 1536, 9 },
};
/* Values for SAMPLE_RATE */
static struct {
int rate;
u16 val;
} wm8961_srate[] = {
{ 48000, 0 },
{ 44100, 0 },
{ 32000, 1 },
{ 22050, 2 },
{ 24000, 2 },
{ 16000, 3 },
{ 11250, 4 },
{ 12000, 4 },
{ 8000, 5 },
};
static int wm8961_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec);
int i, best, target, fs;
u16 reg;
fs = params_rate(params);
if (!wm8961->sysclk) {
dev_err(codec->dev, "MCLK has not been specified\n");
return -EINVAL;
}
/* Find the closest sample rate for the filters */
best = 0;
for (i = 0; i < ARRAY_SIZE(wm8961_srate); i++) {
if (abs(wm8961_srate[i].rate - fs) <
abs(wm8961_srate[best].rate - fs))
best = i;
}
reg = snd_soc_read(codec, WM8961_ADDITIONAL_CONTROL_3);
reg &= ~WM8961_SAMPLE_RATE_MASK;
reg |= wm8961_srate[best].val;
snd_soc_write(codec, WM8961_ADDITIONAL_CONTROL_3, reg);
dev_dbg(codec->dev, "Selected SRATE %dHz for %dHz\n",
wm8961_srate[best].rate, fs);
/* Select a CLK_SYS/fs ratio equal to or higher than required */
target = wm8961->sysclk / fs;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && target < 64) {
dev_err(codec->dev,
"SYSCLK must be at least 64*fs for DAC\n");
return -EINVAL;
}
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE && target < 256) {
dev_err(codec->dev,
"SYSCLK must be at least 256*fs for ADC\n");
return -EINVAL;
}
for (i = 0; i < ARRAY_SIZE(wm8961_clk_sys_ratio); i++) {
if (wm8961_clk_sys_ratio[i].ratio >= target)
break;
}
if (i == ARRAY_SIZE(wm8961_clk_sys_ratio)) {
dev_err(codec->dev, "Unable to generate CLK_SYS_RATE\n");
return -EINVAL;
}
dev_dbg(codec->dev, "Selected CLK_SYS_RATE of %d for %d/%d=%d\n",
wm8961_clk_sys_ratio[i].ratio, wm8961->sysclk, fs,
wm8961->sysclk / fs);
reg = snd_soc_read(codec, WM8961_CLOCKING_4);
reg &= ~WM8961_CLK_SYS_RATE_MASK;
reg |= wm8961_clk_sys_ratio[i].val << WM8961_CLK_SYS_RATE_SHIFT;
snd_soc_write(codec, WM8961_CLOCKING_4, reg);
reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0);
reg &= ~WM8961_WL_MASK;
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
reg |= 1 << WM8961_WL_SHIFT;
break;
case SNDRV_PCM_FORMAT_S24_LE:
reg |= 2 << WM8961_WL_SHIFT;
break;
case SNDRV_PCM_FORMAT_S32_LE:
reg |= 3 << WM8961_WL_SHIFT;
break;
default:
return -EINVAL;
}
snd_soc_write(codec, WM8961_AUDIO_INTERFACE_0, reg);
/* Sloping stop-band filter is recommended for <= 24kHz */
reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_2);
if (fs <= 24000)
reg |= WM8961_DACSLOPE;
else
reg &= WM8961_DACSLOPE;
snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
return 0;
}
static int wm8961_set_sysclk(struct snd_soc_dai *dai, int clk_id,
unsigned int freq,
int dir)
{
struct snd_soc_codec *codec = dai->codec;
struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec);
u16 reg = snd_soc_read(codec, WM8961_CLOCKING1);
if (freq > 33000000) {
dev_err(codec->dev, "MCLK must be <33MHz\n");
return -EINVAL;
}
if (freq > 16500000) {
dev_dbg(codec->dev, "Using MCLK/2 for %dHz MCLK\n", freq);
reg |= WM8961_MCLKDIV;
freq /= 2;
} else {
dev_dbg(codec->dev, "Using MCLK/1 for %dHz MCLK\n", freq);
reg &= WM8961_MCLKDIV;
}
snd_soc_write(codec, WM8961_CLOCKING1, reg);
wm8961->sysclk = freq;
return 0;
}
static int wm8961_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
u16 aif = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_0);
aif &= ~(WM8961_BCLKINV | WM8961_LRP |
WM8961_MS | WM8961_FORMAT_MASK);
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
aif |= WM8961_MS;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
aif |= 1;
break;
case SND_SOC_DAIFMT_I2S:
aif |= 2;
break;
case SND_SOC_DAIFMT_DSP_B:
aif |= WM8961_LRP;
case SND_SOC_DAIFMT_DSP_A:
aif |= 3;
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
case SND_SOC_DAIFMT_IB_NF:
break;
default:
return -EINVAL;
}
break;
default:
return -EINVAL;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
case SND_SOC_DAIFMT_NB_NF:
break;
case SND_SOC_DAIFMT_NB_IF:
aif |= WM8961_LRP;
break;
case SND_SOC_DAIFMT_IB_NF:
aif |= WM8961_BCLKINV;
break;
case SND_SOC_DAIFMT_IB_IF:
aif |= WM8961_BCLKINV | WM8961_LRP;
break;
default:
return -EINVAL;
}
return snd_soc_write(codec, WM8961_AUDIO_INTERFACE_0, aif);
}
static int wm8961_set_tristate(struct snd_soc_dai *dai, int tristate)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg = snd_soc_read(codec, WM8961_ADDITIONAL_CONTROL_2);
if (tristate)
reg |= WM8961_TRIS;
else
reg &= ~WM8961_TRIS;
return snd_soc_write(codec, WM8961_ADDITIONAL_CONTROL_2, reg);
}
static int wm8961_digital_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_1);
if (mute)
reg |= WM8961_DACMU;
else
reg &= ~WM8961_DACMU;
msleep(17);
return snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_1, reg);
}
static int wm8961_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg;
switch (div_id) {
case WM8961_BCLK:
reg = snd_soc_read(codec, WM8961_CLOCKING2);
reg &= ~WM8961_BCLKDIV_MASK;
reg |= div;
snd_soc_write(codec, WM8961_CLOCKING2, reg);
break;
case WM8961_LRCLK:
reg = snd_soc_read(codec, WM8961_AUDIO_INTERFACE_2);
reg &= ~WM8961_LRCLK_RATE_MASK;
reg |= div;
snd_soc_write(codec, WM8961_AUDIO_INTERFACE_2, reg);
break;
default:
return -EINVAL;
}
return 0;
}
static int wm8961_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 reg;
/* This is all slightly unusual since we have no bypass paths
* and the output amplifier structure means we can just slam
* the biases straight up rather than having to ramp them
* slowly.
*/
switch (level) {
case SND_SOC_BIAS_ON:
break;
case SND_SOC_BIAS_PREPARE:
if (codec->bias_level == SND_SOC_BIAS_STANDBY) {
/* Enable bias generation */
reg = snd_soc_read(codec, WM8961_ANTI_POP);
reg |= WM8961_BUFIOEN | WM8961_BUFDCOPEN;
snd_soc_write(codec, WM8961_ANTI_POP, reg);
/* VMID=2*50k, VREF */
reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VMIDSEL_MASK;
reg |= (1 << WM8961_VMIDSEL_SHIFT) | WM8961_VREF;
snd_soc_write(codec, WM8961_PWR_MGMT_1, reg);
}
break;
case SND_SOC_BIAS_STANDBY:
if (codec->bias_level == SND_SOC_BIAS_PREPARE) {
/* VREF off */
reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VREF;
snd_soc_write(codec, WM8961_PWR_MGMT_1, reg);
/* Bias generation off */
reg = snd_soc_read(codec, WM8961_ANTI_POP);
reg &= ~(WM8961_BUFIOEN | WM8961_BUFDCOPEN);
snd_soc_write(codec, WM8961_ANTI_POP, reg);
/* VMID off */
reg = snd_soc_read(codec, WM8961_PWR_MGMT_1);
reg &= ~WM8961_VMIDSEL_MASK;
snd_soc_write(codec, WM8961_PWR_MGMT_1, reg);
}
break;
case SND_SOC_BIAS_OFF:
break;
}
codec->bias_level = level;
return 0;
}
#define WM8961_RATES SNDRV_PCM_RATE_8000_48000
#define WM8961_FORMATS \
(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_LE)
static struct snd_soc_dai_ops wm8961_dai_ops = {
.hw_params = wm8961_hw_params,
.set_sysclk = wm8961_set_sysclk,
.set_fmt = wm8961_set_fmt,
.digital_mute = wm8961_digital_mute,
.set_tristate = wm8961_set_tristate,
.set_clkdiv = wm8961_set_clkdiv,
};
static struct snd_soc_dai_driver wm8961_dai = {
.name = "wm8961-hifi",
.playback = {
.stream_name = "HiFi Playback",
.channels_min = 1,
.channels_max = 2,
.rates = WM8961_RATES,
.formats = WM8961_FORMATS,},
.capture = {
.stream_name = "HiFi Capture",
.channels_min = 1,
.channels_max = 2,
.rates = WM8961_RATES,
.formats = WM8961_FORMATS,},
.ops = &wm8961_dai_ops,
};
static int wm8961_probe(struct snd_soc_codec *codec)
{
struct wm8961_priv *wm8961 = snd_soc_codec_get_drvdata(codec);
int ret = 0;
u16 reg;
codec->control_data = wm8961->control_data;
ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_I2C);
if (ret != 0) {
dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
return ret;
}
reg = snd_soc_read(codec, WM8961_SOFTWARE_RESET);
if (reg != 0x1801) {
dev_err(codec->dev, "Device is not a WM8961: ID=0x%x\n", reg);
return -EINVAL;
}
/* This isn't volatile - readback doesn't correspond to write */
reg = codec->hw_read(codec, WM8961_RIGHT_INPUT_VOLUME);
dev_info(codec->dev, "WM8961 family %d revision %c\n",
(reg & WM8961_DEVICE_ID_MASK) >> WM8961_DEVICE_ID_SHIFT,
((reg & WM8961_CHIP_REV_MASK) >> WM8961_CHIP_REV_SHIFT)
+ 'A');
ret = wm8961_reset(codec);
if (ret < 0) {
dev_err(codec->dev, "Failed to issue reset\n");
return ret;
}
/* Enable class W */
reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B);
reg |= WM8961_CP_DYN_PWR_MASK;
snd_soc_write(codec, WM8961_CHARGE_PUMP_B, reg);
/* Latch volume update bits (right channel only, we always
* write both out) and default ZC on. */
reg = snd_soc_read(codec, WM8961_ROUT1_VOLUME);
snd_soc_write(codec, WM8961_ROUT1_VOLUME,
reg | WM8961_LO1ZC | WM8961_OUT1VU);
snd_soc_write(codec, WM8961_LOUT1_VOLUME, reg | WM8961_LO1ZC);
reg = snd_soc_read(codec, WM8961_ROUT2_VOLUME);
snd_soc_write(codec, WM8961_ROUT2_VOLUME,
reg | WM8961_SPKRZC | WM8961_SPKVU);
snd_soc_write(codec, WM8961_LOUT2_VOLUME, reg | WM8961_SPKLZC);
reg = snd_soc_read(codec, WM8961_RIGHT_ADC_VOLUME);
snd_soc_write(codec, WM8961_RIGHT_ADC_VOLUME, reg | WM8961_ADCVU);
reg = snd_soc_read(codec, WM8961_RIGHT_INPUT_VOLUME);
snd_soc_write(codec, WM8961_RIGHT_INPUT_VOLUME, reg | WM8961_IPVU);
/* Use soft mute by default */
reg = snd_soc_read(codec, WM8961_ADC_DAC_CONTROL_2);
reg |= WM8961_DACSMM;
snd_soc_write(codec, WM8961_ADC_DAC_CONTROL_2, reg);
/* Use automatic clocking mode by default; for now this is all
* we support.
*/
reg = snd_soc_read(codec, WM8961_CLOCKING_3);
reg &= ~WM8961_MANUAL_MODE;
snd_soc_write(codec, WM8961_CLOCKING_3, reg);
wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
snd_soc_add_controls(codec, wm8961_snd_controls,
ARRAY_SIZE(wm8961_snd_controls));
snd_soc_dapm_new_controls(codec, wm8961_dapm_widgets,
ARRAY_SIZE(wm8961_dapm_widgets));
snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths));
return 0;
}
static int wm8961_remove(struct snd_soc_codec *codec)
{
wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
#ifdef CONFIG_PM
static int wm8961_suspend(struct snd_soc_codec *codec, pm_message_t state)
{
wm8961_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int wm8961_resume(struct snd_soc_codec *codec)
{
u16 *reg_cache = codec->reg_cache;
int i;
for (i = 0; i < codec->driver->reg_cache_size; i++) {
if (reg_cache[i] == wm8961_reg_defaults[i])
continue;
if (i == WM8961_SOFTWARE_RESET)
continue;
snd_soc_write(codec, i, reg_cache[i]);
}
wm8961_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
#else
#define wm8961_suspend NULL
#define wm8961_resume NULL
#endif
static struct snd_soc_codec_driver soc_codec_dev_wm8961 = {
.probe = wm8961_probe,
.remove = wm8961_remove,
.suspend = wm8961_suspend,
.resume = wm8961_resume,
.set_bias_level = wm8961_set_bias_level,
.reg_cache_size = sizeof(wm8961_reg_defaults),
.reg_word_size = sizeof(u16),
.reg_cache_default = wm8961_reg_defaults,
.volatile_register = wm8961_volatile_register,
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
static __devinit int wm8961_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct wm8961_priv *wm8961;
int ret;
wm8961 = kzalloc(sizeof(struct wm8961_priv), GFP_KERNEL);
if (wm8961 == NULL)
return -ENOMEM;
i2c_set_clientdata(i2c, wm8961);
wm8961->control_data = i2c;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_wm8961, &wm8961_dai, 1);
if (ret < 0)
kfree(wm8961);
return ret;
}
static __devexit int wm8961_i2c_remove(struct i2c_client *client)
{
snd_soc_unregister_codec(&client->dev);
kfree(i2c_get_clientdata(client));
return 0;
}
static const struct i2c_device_id wm8961_i2c_id[] = {
{ "wm8961", 0 },
{ }
};
MODULE_DEVICE_TABLE(i2c, wm8961_i2c_id);
static struct i2c_driver wm8961_i2c_driver = {
.driver = {
.name = "wm8961-codec",
.owner = THIS_MODULE,
},
.probe = wm8961_i2c_probe,
.remove = __devexit_p(wm8961_i2c_remove),
.id_table = wm8961_i2c_id,
};
#endif
static int __init wm8961_modinit(void)
{
int ret = 0;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&wm8961_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register wm8961 I2C driver: %d\n",
ret);
}
#endif
return ret;
}
module_init(wm8961_modinit);
static void __exit wm8961_exit(void)
{
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&wm8961_i2c_driver);
#endif
}
module_exit(wm8961_exit);
MODULE_DESCRIPTION("ASoC WM8961 driver");
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_LICENSE("GPL");