kevin/linux
1
Fork 0
Code
e77125105b
linux/sound/soc/codecs/tlv320aic23.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

781 lines
21 KiB
C
Raw Blame History

/*
* ALSA SoC TLV320AIC23 codec driver
*
* Author: Arun KS, <arunks@mistralsolutions.com>
* Copyright: (C) 2008 Mistral Solutions Pvt Ltd.,
*
* Based on sound/soc/codecs/wm8731.c by Richard Purdie
*
* 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.
*
* Notes:
* The AIC23 is a driver for a low power stereo audio
* codec tlv320aic23
*
* The machine layer should disable unsupported inputs/outputs by
* snd_soc_dapm_disable_pin(codec, "LHPOUT"), etc.
*/
#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/tlv.h>
#include <sound/initval.h>
#include "tlv320aic23.h"
#define AIC23_VERSION "0.1"
/*
* AIC23 register cache
*/
static const u16 tlv320aic23_reg[] = {
0x0097, 0x0097, 0x00F9, 0x00F9, /* 0 */
0x001A, 0x0004, 0x0007, 0x0001, /* 4 */
0x0020, 0x0000, 0x0000, 0x0000, /* 8 */
0x0000, 0x0000, 0x0000, 0x0000, /* 12 */
};
/*
* read tlv320aic23 register cache
*/
static inline unsigned int tlv320aic23_read_reg_cache(struct snd_soc_codec
*codec, unsigned int reg)
{
u16 *cache = codec->reg_cache;
if (reg >= ARRAY_SIZE(tlv320aic23_reg))
return -1;
return cache[reg];
}
/*
* write tlv320aic23 register cache
*/
static inline void tlv320aic23_write_reg_cache(struct snd_soc_codec *codec,
u8 reg, u16 value)
{
u16 *cache = codec->reg_cache;
if (reg >= ARRAY_SIZE(tlv320aic23_reg))
return;
cache[reg] = value;
}
/*
* write to the tlv320aic23 register space
*/
static int tlv320aic23_write(struct snd_soc_codec *codec, unsigned int reg,
unsigned int value)
{
u8 data[2];
/* TLV320AIC23 has 7 bit address and 9 bits of data
* so we need to switch one data bit into reg and rest
* of data into val
*/
if (reg > 9 && reg != 15) {
printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg);
return -1;
}
data[0] = (reg << 1) | (value >> 8 & 0x01);
data[1] = value & 0xff;
tlv320aic23_write_reg_cache(codec, reg, value);
if (codec->hw_write(codec->control_data, data, 2) == 2)
return 0;
printk(KERN_ERR "%s cannot write %03x to register R%u\n", __func__,
value, reg);
return -EIO;
}
static const char *rec_src_text[] = { "Line", "Mic" };
static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"};
static const struct soc_enum rec_src_enum =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls =
SOC_DAPM_ENUM("Input Select", rec_src_enum);
static const struct soc_enum tlv320aic23_rec_src =
SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text);
static const struct soc_enum tlv320aic23_deemph =
SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text);
static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0);
static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0);
static const DECLARE_TLV_DB_SCALE(sidetone_vol_tlv, -1800, 300, 0);
static int snd_soc_tlv320aic23_put_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
u16 val, reg;
val = (ucontrol->value.integer.value[0] & 0x07);
/* linear conversion to userspace
* 000 = -6db
* 001 = -9db
* 010 = -12db
* 011 = -18db (Min)
* 100 = 0db (Max)
*/
val = (val >= 4) ? 4 : (3 - val);
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (~0x1C0);
tlv320aic23_write(codec, TLV320AIC23_ANLG, reg | (val << 6));
return 0;
}
static int snd_soc_tlv320aic23_get_volsw(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
u16 val;
val = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG) & (0x1C0);
val = val >> 6;
val = (val >= 4) ? 4 : (3 - val);
ucontrol->value.integer.value[0] = val;
return 0;
}
#define SOC_TLV320AIC23_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \
{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
SNDRV_CTL_ELEM_ACCESS_READWRITE,\
.tlv.p = (tlv_array), \
.info = snd_soc_info_volsw, .get = snd_soc_tlv320aic23_get_volsw,\
.put = snd_soc_tlv320aic23_put_volsw, \
.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
SOC_DOUBLE_R_TLV("Digital Playback Volume", TLV320AIC23_LCHNVOL,
TLV320AIC23_RCHNVOL, 0, 127, 0, out_gain_tlv),
SOC_SINGLE("Digital Playback Switch", TLV320AIC23_DIGT, 3, 1, 1),
SOC_DOUBLE_R("Line Input Switch", TLV320AIC23_LINVOL,
TLV320AIC23_RINVOL, 7, 1, 0),
SOC_DOUBLE_R_TLV("Line Input Volume", TLV320AIC23_LINVOL,
TLV320AIC23_RINVOL, 0, 31, 0, input_gain_tlv),
SOC_SINGLE("Mic Input Switch", TLV320AIC23_ANLG, 1, 1, 1),
SOC_SINGLE("Mic Booster Switch", TLV320AIC23_ANLG, 0, 1, 0),
SOC_TLV320AIC23_SINGLE_TLV("Sidetone Volume", TLV320AIC23_ANLG,
6, 4, 0, sidetone_vol_tlv),
SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
};
/* PGA Mixer controls for Line and Mic switch */
static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
SOC_DAPM_SINGLE("Mic Sidetone Switch", TLV320AIC23_ANLG, 5, 1, 0),
SOC_DAPM_SINGLE("Playback Switch", TLV320AIC23_ANLG, 4, 1, 0),
};
static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", TLV320AIC23_PWR, 3, 1),
SND_SOC_DAPM_ADC("ADC", "Capture", TLV320AIC23_PWR, 2, 1),
SND_SOC_DAPM_MUX("Capture Source", SND_SOC_NOPM, 0, 0,
&tlv320aic23_rec_src_mux_controls),
SND_SOC_DAPM_MIXER("Output Mixer", TLV320AIC23_PWR, 4, 1,
&tlv320aic23_output_mixer_controls[0],
ARRAY_SIZE(tlv320aic23_output_mixer_controls)),
SND_SOC_DAPM_PGA("Line Input", TLV320AIC23_PWR, 0, 1, NULL, 0),
SND_SOC_DAPM_PGA("Mic Input", TLV320AIC23_PWR, 1, 1, NULL, 0),
SND_SOC_DAPM_OUTPUT("LHPOUT"),
SND_SOC_DAPM_OUTPUT("RHPOUT"),
SND_SOC_DAPM_OUTPUT("LOUT"),
SND_SOC_DAPM_OUTPUT("ROUT"),
SND_SOC_DAPM_INPUT("LLINEIN"),
SND_SOC_DAPM_INPUT("RLINEIN"),
SND_SOC_DAPM_INPUT("MICIN"),
};
static const struct snd_soc_dapm_route intercon[] = {
/* Output Mixer */
{"Output Mixer", "Line Bypass Switch", "Line Input"},
{"Output Mixer", "Playback Switch", "DAC"},
{"Output Mixer", "Mic Sidetone Switch", "Mic Input"},
/* Outputs */
{"RHPOUT", NULL, "Output Mixer"},
{"LHPOUT", NULL, "Output Mixer"},
{"LOUT", NULL, "Output Mixer"},
{"ROUT", NULL, "Output Mixer"},
/* Inputs */
{"Line Input", "NULL", "LLINEIN"},
{"Line Input", "NULL", "RLINEIN"},
{"Mic Input", "NULL", "MICIN"},
/* input mux */
{"Capture Source", "Line", "Line Input"},
{"Capture Source", "Mic", "Mic Input"},
{"ADC", NULL, "Capture Source"},
};
/* AIC23 driver data */
struct aic23 {
enum snd_soc_control_type control_type;
void *control_data;
int mclk;
int requested_adc;
int requested_dac;
};
/*
* Common Crystals used
* 11.2896 Mhz /128 = *88.2k /192 = 58.8k
* 12.0000 Mhz /125 = *96k /136 = 88.235K
* 12.2880 Mhz /128 = *96k /192 = 64k
* 16.9344 Mhz /128 = 132.3k /192 = *88.2k
* 18.4320 Mhz /128 = 144k /192 = *96k
*/
/*
* Normal BOSR 0-256/2 = 128, 1-384/2 = 192
* USB BOSR 0-250/2 = 125, 1-272/2 = 136
*/
static const int bosr_usb_divisor_table[] = {
128, 125, 192, 136
};
#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7))
#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15))
static const unsigned short sr_valid_mask[] = {
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 0*/
LOWER_GROUP, /* Usb, bosr - 0*/
LOWER_GROUP|UPPER_GROUP, /* Normal, bosr - 1*/
UPPER_GROUP, /* Usb, bosr - 1*/
};
/*
* Every divisor is a factor of 11*12
*/
#define SR_MULT (11*12)
#define A(x) (SR_MULT/x)
static const unsigned char sr_adc_mult_table[] = {
A(2), A(2), A(12), A(12), 0, 0, A(3), A(1),
A(2), A(2), A(11), A(11), 0, 0, 0, A(1)
};
static const unsigned char sr_dac_mult_table[] = {
A(2), A(12), A(2), A(12), 0, 0, A(3), A(1),
A(2), A(11), A(2), A(11), 0, 0, 0, A(1)
};
static unsigned get_score(int adc, int adc_l, int adc_h, int need_adc,
int dac, int dac_l, int dac_h, int need_dac)
{
if ((adc >= adc_l) && (adc <= adc_h) &&
(dac >= dac_l) && (dac <= dac_h)) {
int diff_adc = need_adc - adc;
int diff_dac = need_dac - dac;
return abs(diff_adc) + abs(diff_dac);
}
return UINT_MAX;
}
static int find_rate(int mclk, u32 need_adc, u32 need_dac)
{
int i, j;
int best_i = -1;
int best_j = -1;
int best_div = 0;
unsigned best_score = UINT_MAX;
int adc_l, adc_h, dac_l, dac_h;
need_adc *= SR_MULT;
need_dac *= SR_MULT;
/*
* rates given are +/- 1/32
*/
adc_l = need_adc - (need_adc >> 5);
adc_h = need_adc + (need_adc >> 5);
dac_l = need_dac - (need_dac >> 5);
dac_h = need_dac + (need_dac >> 5);
for (i = 0; i < ARRAY_SIZE(bosr_usb_divisor_table); i++) {
int base = mclk / bosr_usb_divisor_table[i];
int mask = sr_valid_mask[i];
for (j = 0; j < ARRAY_SIZE(sr_adc_mult_table);
j++, mask >>= 1) {
int adc;
int dac;
int score;
if ((mask & 1) == 0)
continue;
adc = base * sr_adc_mult_table[j];
dac = base * sr_dac_mult_table[j];
score = get_score(adc, adc_l, adc_h, need_adc,
dac, dac_l, dac_h, need_dac);
if (best_score > score) {
best_score = score;
best_i = i;
best_j = j;
best_div = 0;
}
score = get_score((adc >> 1), adc_l, adc_h, need_adc,
(dac >> 1), dac_l, dac_h, need_dac);
/* prefer to have a /2 */
if ((score != UINT_MAX) && (best_score >= score)) {
best_score = score;
best_i = i;
best_j = j;
best_div = 1;
}
}
}
return (best_j << 2) | best_i | (best_div << TLV320AIC23_CLKIN_SHIFT);
}
#ifdef DEBUG
static void get_current_sample_rates(struct snd_soc_codec *codec, int mclk,
u32 *sample_rate_adc, u32 *sample_rate_dac)
{
int src = tlv320aic23_read_reg_cache(codec, TLV320AIC23_SRATE);
int sr = (src >> 2) & 0x0f;
int val = (mclk / bosr_usb_divisor_table[src & 3]);
int adc = (val * sr_adc_mult_table[sr]) / SR_MULT;
int dac = (val * sr_dac_mult_table[sr]) / SR_MULT;
if (src & TLV320AIC23_CLKIN_HALF) {
adc >>= 1;
dac >>= 1;
}
*sample_rate_adc = adc;
*sample_rate_dac = dac;
}
#endif
static int set_sample_rate_control(struct snd_soc_codec *codec, int mclk,
u32 sample_rate_adc, u32 sample_rate_dac)
{
/* Search for the right sample rate */
int data = find_rate(mclk, sample_rate_adc, sample_rate_dac);
if (data < 0) {
printk(KERN_ERR "%s:Invalid rate %u,%u requested\n",
__func__, sample_rate_adc, sample_rate_dac);
return -EINVAL;
}
tlv320aic23_write(codec, TLV320AIC23_SRATE, data);
#ifdef DEBUG
{
u32 adc, dac;
get_current_sample_rates(codec, mclk, &adc, &dac);
printk(KERN_DEBUG "actual samplerate = %u,%u reg=%x\n",
adc, dac, data);
}
#endif
return 0;
}
static int tlv320aic23_add_widgets(struct snd_soc_codec *codec)
{
snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
ARRAY_SIZE(tlv320aic23_dapm_widgets));
/* set up audio path interconnects */
snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
return 0;
}
static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
u16 iface_reg;
int ret;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
u32 sample_rate_adc = aic23->requested_adc;
u32 sample_rate_dac = aic23->requested_dac;
u32 sample_rate = params_rate(params);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
aic23->requested_dac = sample_rate_dac = sample_rate;
if (!sample_rate_adc)
sample_rate_adc = sample_rate;
} else {
aic23->requested_adc = sample_rate_adc = sample_rate;
if (!sample_rate_dac)
sample_rate_dac = sample_rate;
}
ret = set_sample_rate_control(codec, aic23->mclk, sample_rate_adc,
sample_rate_dac);
if (ret < 0)
return ret;
iface_reg =
tlv320aic23_read_reg_cache(codec,
TLV320AIC23_DIGT_FMT) & ~(0x03 << 2);
switch (params_format(params)) {
case SNDRV_PCM_FORMAT_S16_LE:
break;
case SNDRV_PCM_FORMAT_S20_3LE:
iface_reg |= (0x01 << 2);
break;
case SNDRV_PCM_FORMAT_S24_LE:
iface_reg |= (0x02 << 2);
break;
case SNDRV_PCM_FORMAT_S32_LE:
iface_reg |= (0x03 << 2);
break;
}
tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
/* set active */
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
return 0;
}
static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_codec *codec = rtd->codec;
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
/* deactivate */
if (!codec->active) {
udelay(50);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
}
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
aic23->requested_dac = 0;
else
aic23->requested_adc = 0;
}
static int tlv320aic23_mute(struct snd_soc_dai *dai, int mute)
{
struct snd_soc_codec *codec = dai->codec;
u16 reg;
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT);
if (mute)
reg |= TLV320AIC23_DACM_MUTE;
else
reg &= ~TLV320AIC23_DACM_MUTE;
tlv320aic23_write(codec, TLV320AIC23_DIGT, reg);
return 0;
}
static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai,
unsigned int fmt)
{
struct snd_soc_codec *codec = codec_dai->codec;
u16 iface_reg;
iface_reg =
tlv320aic23_read_reg_cache(codec, TLV320AIC23_DIGT_FMT) & (~0x03);
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
iface_reg |= TLV320AIC23_MS_MASTER;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
/* interface format */
switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
case SND_SOC_DAIFMT_I2S:
iface_reg |= TLV320AIC23_FOR_I2S;
break;
case SND_SOC_DAIFMT_DSP_A:
iface_reg |= TLV320AIC23_LRP_ON;
case SND_SOC_DAIFMT_DSP_B:
iface_reg |= TLV320AIC23_FOR_DSP;
break;
case SND_SOC_DAIFMT_RIGHT_J:
break;
case SND_SOC_DAIFMT_LEFT_J:
iface_reg |= TLV320AIC23_FOR_LJUST;
break;
default:
return -EINVAL;
}
tlv320aic23_write(codec, TLV320AIC23_DIGT_FMT, iface_reg);
return 0;
}
static int tlv320aic23_set_dai_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct aic23 *aic23 = snd_soc_dai_get_drvdata(codec_dai);
aic23->mclk = freq;
return 0;
}
static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
u16 reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_PWR) & 0xff7f;
switch (level) {
case SND_SOC_BIAS_ON:
/* vref/mid, osc on, dac unmute */
reg &= ~(TLV320AIC23_DEVICE_PWR_OFF | TLV320AIC23_OSC_OFF | \
TLV320AIC23_DAC_OFF);
tlv320aic23_write(codec, TLV320AIC23_PWR, reg);
break;
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
/* everything off except vref/vmid, */
tlv320aic23_write(codec, TLV320AIC23_PWR, reg | \
TLV320AIC23_CLK_OFF);
break;
case SND_SOC_BIAS_OFF:
/* everything off, dac mute, inactive */
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
tlv320aic23_write(codec, TLV320AIC23_PWR, 0xffff);
break;
}
codec->bias_level = level;
return 0;
}
#define AIC23_RATES SNDRV_PCM_RATE_8000_96000
#define AIC23_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
static struct snd_soc_dai_ops tlv320aic23_dai_ops = {
.prepare = tlv320aic23_pcm_prepare,
.hw_params = tlv320aic23_hw_params,
.shutdown = tlv320aic23_shutdown,
.digital_mute = tlv320aic23_mute,
.set_fmt = tlv320aic23_set_dai_fmt,
.set_sysclk = tlv320aic23_set_dai_sysclk,
};
static struct snd_soc_dai_driver tlv320aic23_dai = {
.name = "tlv320aic23-hifi",
.playback = {
.stream_name = "Playback",
.channels_min = 2,
.channels_max = 2,
.rates = AIC23_RATES,
.formats = AIC23_FORMATS,},
.capture = {
.stream_name = "Capture",
.channels_min = 2,
.channels_max = 2,
.rates = AIC23_RATES,
.formats = AIC23_FORMATS,},
.ops = &tlv320aic23_dai_ops,
};
static int tlv320aic23_suspend(struct snd_soc_codec *codec,
pm_message_t state)
{
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static int tlv320aic23_resume(struct snd_soc_codec *codec)
{
u16 reg;
/* Sync reg_cache with the hardware */
for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) {
u16 val = tlv320aic23_read_reg_cache(codec, reg);
tlv320aic23_write(codec, reg, val);
}
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
return 0;
}
static int tlv320aic23_probe(struct snd_soc_codec *codec)
{
struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec);
int reg;
printk(KERN_INFO "AIC23 Audio Codec %s\n", AIC23_VERSION);
codec->control_data = aic23->control_data;
codec->hw_write = (hw_write_t)i2c_master_send;
codec->hw_read = NULL;
/* Reset codec */
tlv320aic23_write(codec, TLV320AIC23_RESET, 0);
/* power on device */
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
tlv320aic23_write(codec, TLV320AIC23_DIGT, TLV320AIC23_DEEMP_44K);
/* Unmute input */
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_LINVOL);
tlv320aic23_write(codec, TLV320AIC23_LINVOL,
(reg & (~TLV320AIC23_LIM_MUTED)) |
(TLV320AIC23_LRS_ENABLED));
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_RINVOL);
tlv320aic23_write(codec, TLV320AIC23_RINVOL,
(reg & (~TLV320AIC23_LIM_MUTED)) |
TLV320AIC23_LRS_ENABLED);
reg = tlv320aic23_read_reg_cache(codec, TLV320AIC23_ANLG);
tlv320aic23_write(codec, TLV320AIC23_ANLG,
(reg) & (~TLV320AIC23_BYPASS_ON) &
(~TLV320AIC23_MICM_MUTED));
/* Default output volume */
tlv320aic23_write(codec, TLV320AIC23_LCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL &
TLV320AIC23_OUT_VOL_MASK);
tlv320aic23_write(codec, TLV320AIC23_RCHNVOL,
TLV320AIC23_DEFAULT_OUT_VOL &
TLV320AIC23_OUT_VOL_MASK);
tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
snd_soc_add_controls(codec, tlv320aic23_snd_controls,
ARRAY_SIZE(tlv320aic23_snd_controls));
tlv320aic23_add_widgets(codec);
return 0;
}
static int tlv320aic23_remove(struct snd_soc_codec *codec)
{
tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
return 0;
}
static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = {
.reg_cache_size = ARRAY_SIZE(tlv320aic23_reg),
.reg_word_size = sizeof(u16),
.reg_cache_default = tlv320aic23_reg,
.probe = tlv320aic23_probe,
.remove = tlv320aic23_remove,
.suspend = tlv320aic23_suspend,
.resume = tlv320aic23_resume,
.read = tlv320aic23_read_reg_cache,
.write = tlv320aic23_write,
.set_bias_level = tlv320aic23_set_bias_level,
};
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
/*
* If the i2c layer weren't so broken, we could pass this kind of data
* around
*/
static int tlv320aic23_codec_probe(struct i2c_client *i2c,
const struct i2c_device_id *i2c_id)
{
struct aic23 *aic23;
int ret;
if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
return -EINVAL;
aic23 = kzalloc(sizeof(struct aic23), GFP_KERNEL);
if (aic23 == NULL)
return -ENOMEM;
i2c_set_clientdata(i2c, aic23);
aic23->control_data = i2c;
aic23->control_type = SND_SOC_I2C;
ret = snd_soc_register_codec(&i2c->dev,
&soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1);
if (ret < 0)
kfree(aic23);
return ret;
}
static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c)
{
snd_soc_unregister_codec(&i2c->dev);
kfree(i2c_get_clientdata(i2c));
return 0;
}
static const struct i2c_device_id tlv320aic23_id[] = {
{"tlv320aic23", 0},
{}
};
MODULE_DEVICE_TABLE(i2c, tlv320aic23_id);
static struct i2c_driver tlv320aic23_i2c_driver = {
.driver = {
.name = "tlv320aic23-codec",
},
.probe = tlv320aic23_codec_probe,
.remove = __exit_p(tlv320aic23_i2c_remove),
.id_table = tlv320aic23_id,
};
#endif
static int __init tlv320aic23_modinit(void)
{
int ret;
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
ret = i2c_add_driver(&tlv320aic23_i2c_driver);
if (ret != 0) {
printk(KERN_ERR "Failed to register TLV320AIC23 I2C driver: %d\n",
ret);
}
#endif
return ret;
}
module_init(tlv320aic23_modinit);
static void __exit tlv320aic23_exit(void)
{
#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
i2c_del_driver(&tlv320aic23_i2c_driver);
#endif
}
module_exit(tlv320aic23_exit);
MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver");
MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
MODULE_LICENSE("GPL");
View Git Blame Copy Permalink