Minki/linux
1
0
Fork 1
mirror of https://github.com/torvalds/linux.git synced 2024-11-05 19:41:54 +00:00
Code Issues Packages Projects Releases Wiki Activity
7711308ae6
linux/sound/soc/s3c24xx/neo1973_wm8753.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

705 lines
18 KiB
C
Raw Blame History

/*
* neo1973_wm8753.c -- SoC audio for Neo1973
*
* Copyright 2007 Wolfson Microelectronics PLC.
* Author: Graeme Gregory
* graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
*
* 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/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <linux/i2c.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/tlv.h>
#include <asm/mach-types.h>
#include <asm/hardware/scoop.h>
#include <mach/regs-clock.h>
#include <mach/regs-gpio.h>
#include <mach/hardware.h>
#include <linux/io.h>
#include <mach/spi-gpio.h>
#include <plat/regs-iis.h>
#include "../codecs/wm8753.h"
#include "lm4857.h"
#include "s3c-dma.h"
#include "s3c24xx-i2s.h"
/* define the scenarios */
#define NEO_AUDIO_OFF 0
#define NEO_GSM_CALL_AUDIO_HANDSET 1
#define NEO_GSM_CALL_AUDIO_HEADSET 2
#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
#define NEO_STEREO_TO_SPEAKERS 4
#define NEO_STEREO_TO_HEADPHONES 5
#define NEO_CAPTURE_HANDSET 6
#define NEO_CAPTURE_HEADSET 7
#define NEO_CAPTURE_BLUETOOTH 8
static struct snd_soc_card neo1973;
static struct i2c_client *i2c;
static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
unsigned int pll_out = 0, bclk = 0;
int ret = 0;
unsigned long iis_clkrate;
pr_debug("Entered %s\n", __func__);
iis_clkrate = s3c24xx_i2s_get_clockrate();
switch (params_rate(params)) {
case 8000:
case 16000:
pll_out = 12288000;
break;
case 48000:
bclk = WM8753_BCLK_DIV_4;
pll_out = 12288000;
break;
case 96000:
bclk = WM8753_BCLK_DIV_2;
pll_out = 12288000;
break;
case 11025:
bclk = WM8753_BCLK_DIV_16;
pll_out = 11289600;
break;
case 22050:
bclk = WM8753_BCLK_DIV_8;
pll_out = 11289600;
break;
case 44100:
bclk = WM8753_BCLK_DIV_4;
pll_out = 11289600;
break;
case 88200:
bclk = WM8753_BCLK_DIV_2;
pll_out = 11289600;
break;
}
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai,
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai,
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set MCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
S3C2410_IISMOD_32FS);
if (ret < 0)
return ret;
/* set codec BCLK division for sample rate */
ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
if (ret < 0)
return ret;
/* set prescaler division for sample rate */
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
S3C24XX_PRESCALE(4, 4));
if (ret < 0)
return ret;
/* codec PLL input is PCLK/4 */
ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0,
iis_clkrate / 4, pll_out);
if (ret < 0)
return ret;
return 0;
}
static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
pr_debug("Entered %s\n", __func__);
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0, 0);
}
/*
* Neo1973 WM8753 HiFi DAI opserations.
*/
static struct snd_soc_ops neo1973_hifi_ops = {
.hw_params = neo1973_hifi_hw_params,
.hw_free = neo1973_hifi_hw_free,
};
static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
unsigned int pcmdiv = 0;
int ret = 0;
unsigned long iis_clkrate;
pr_debug("Entered %s\n", __func__);
iis_clkrate = s3c24xx_i2s_get_clockrate();
if (params_rate(params) != 8000)
return -EINVAL;
if (params_channels(params) != 1)
return -EINVAL;
pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
/* todo: gg check mode (DSP_B) against CSR datasheet */
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* set codec PCM division for sample rate */
ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
if (ret < 0)
return ret;
/* configue and enable PLL for 12.288MHz output */
ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0,
iis_clkrate / 4, 12288000);
if (ret < 0)
return ret;
return 0;
}
static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->codec_dai;
pr_debug("Entered %s\n", __func__);
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0, 0);
}
static struct snd_soc_ops neo1973_voice_ops = {
.hw_params = neo1973_voice_hw_params,
.hw_free = neo1973_voice_hw_free,
};
static int neo1973_scenario;
static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = neo1973_scenario;
return 0;
}
static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
{
pr_debug("Entered %s\n", __func__);
switch (neo1973_scenario) {
case NEO_AUDIO_OFF:
snd_soc_dapm_disable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_HANDSET:
snd_soc_dapm_enable_pin(codec, "Audio Out");
snd_soc_dapm_enable_pin(codec, "GSM Line Out");
snd_soc_dapm_enable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_enable_pin(codec, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_HEADSET:
snd_soc_dapm_enable_pin(codec, "Audio Out");
snd_soc_dapm_enable_pin(codec, "GSM Line Out");
snd_soc_dapm_enable_pin(codec, "GSM Line In");
snd_soc_dapm_enable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_GSM_CALL_AUDIO_BLUETOOTH:
snd_soc_dapm_disable_pin(codec, "Audio Out");
snd_soc_dapm_enable_pin(codec, "GSM Line Out");
snd_soc_dapm_enable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_STEREO_TO_SPEAKERS:
snd_soc_dapm_enable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_STEREO_TO_HEADPHONES:
snd_soc_dapm_enable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_CAPTURE_HANDSET:
snd_soc_dapm_disable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_enable_pin(codec, "Call Mic");
break;
case NEO_CAPTURE_HEADSET:
snd_soc_dapm_disable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_enable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
case NEO_CAPTURE_BLUETOOTH:
snd_soc_dapm_disable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
break;
default:
snd_soc_dapm_disable_pin(codec, "Audio Out");
snd_soc_dapm_disable_pin(codec, "GSM Line Out");
snd_soc_dapm_disable_pin(codec, "GSM Line In");
snd_soc_dapm_disable_pin(codec, "Headset Mic");
snd_soc_dapm_disable_pin(codec, "Call Mic");
}
snd_soc_dapm_sync(codec);
return 0;
}
static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
pr_debug("Entered %s\n", __func__);
if (neo1973_scenario == ucontrol->value.integer.value[0])
return 0;
neo1973_scenario = ucontrol->value.integer.value[0];
set_scenario_endpoints(codec, neo1973_scenario);
return 1;
}
static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
static void lm4857_write_regs(void)
{
pr_debug("Entered %s\n", __func__);
if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
printk(KERN_ERR "lm4857: i2c write failed\n");
}
static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int reg = mc->reg;
int shift = mc->shift;
int mask = mc->max;
pr_debug("Entered %s\n", __func__);
ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
return 0;
}
static int lm4857_set_reg(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct soc_mixer_control *mc =
(struct soc_mixer_control *)kcontrol->private_value;
int reg = mc->reg;
int shift = mc->shift;
int mask = mc->max;
if (((lm4857_regs[reg] >> shift) & mask) ==
ucontrol->value.integer.value[0])
return 0;
lm4857_regs[reg] &= ~(mask << shift);
lm4857_regs[reg] |= ucontrol->value.integer.value[0] << shift;
lm4857_write_regs();
return 1;
}
static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
pr_debug("Entered %s\n", __func__);
if (value)
value -= 5;
ucontrol->value.integer.value[0] = value;
return 0;
}
static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
u8 value = ucontrol->value.integer.value[0];
pr_debug("Entered %s\n", __func__);
if (value)
value += 5;
if ((lm4857_regs[LM4857_CTRL] & 0x0F) == value)
return 0;
lm4857_regs[LM4857_CTRL] &= 0xF0;
lm4857_regs[LM4857_CTRL] |= value;
lm4857_write_regs();
return 1;
}
static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
SND_SOC_DAPM_LINE("Audio Out", NULL),
SND_SOC_DAPM_LINE("GSM Line Out", NULL),
SND_SOC_DAPM_LINE("GSM Line In", NULL),
SND_SOC_DAPM_MIC("Headset Mic", NULL),
SND_SOC_DAPM_MIC("Call Mic", NULL),
};
static const struct snd_soc_dapm_route dapm_routes[] = {
/* Connections to the lm4857 amp */
{"Audio Out", NULL, "LOUT1"},
{"Audio Out", NULL, "ROUT1"},
/* Connections to the GSM Module */
{"GSM Line Out", NULL, "MONO1"},
{"GSM Line Out", NULL, "MONO2"},
{"RXP", NULL, "GSM Line In"},
{"RXN", NULL, "GSM Line In"},
/* Connections to Headset */
{"MIC1", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Headset Mic"},
/* Call Mic */
{"MIC2", NULL, "Mic Bias"},
{"MIC2N", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Call Mic"},
/* Connect the ALC pins */
{"ACIN", NULL, "ACOP"},
};
static const char *lm4857_mode[] = {
"Off",
"Call Speaker",
"Stereo Speakers",
"Stereo Speakers + Headphones",
"Headphones"
};
static const struct soc_enum lm4857_mode_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode),
};
static const char *neo_scenarios[] = {
"Off",
"GSM Handset",
"GSM Headset",
"GSM Bluetooth",
"Speakers",
"Headphones",
"Capture Handset",
"Capture Headset",
"Capture Bluetooth"
};
static const struct soc_enum neo_scenario_enum[] = {
SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios),
};
static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0);
static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0);
static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
lm4857_get_reg, lm4857_set_reg, stereo_tlv),
SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
lm4857_get_reg, lm4857_set_reg, stereo_tlv),
SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0,
lm4857_get_reg, lm4857_set_reg, mono_tlv),
SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
lm4857_get_mode, lm4857_set_mode),
SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
neo1973_get_scenario, neo1973_set_scenario),
SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
lm4857_get_reg, lm4857_set_reg),
SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
lm4857_get_reg, lm4857_set_reg),
SOC_SINGLE_EXT("Amp Fast Wakeup Playback Switch", LM4857_CTRL, 5, 1, 0,
lm4857_get_reg, lm4857_set_reg),
SOC_SINGLE_EXT("Amp Earpiece 6dB Playback Switch", LM4857_CTRL, 4, 1, 0,
lm4857_get_reg, lm4857_set_reg),
};
/*
* This is an example machine initialisation for a wm8753 connected to a
* neo1973 II. It is missing logic to detect hp/mic insertions and logic
* to re-route the audio in such an event.
*/
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
int err;
pr_debug("Entered %s\n", __func__);
/* set up NC codec pins */
snd_soc_dapm_nc_pin(codec, "LOUT2");
snd_soc_dapm_nc_pin(codec, "ROUT2");
snd_soc_dapm_nc_pin(codec, "OUT3");
snd_soc_dapm_nc_pin(codec, "OUT4");
snd_soc_dapm_nc_pin(codec, "LINE1");
snd_soc_dapm_nc_pin(codec, "LINE2");
/* Add neo1973 specific widgets */
snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
ARRAY_SIZE(wm8753_dapm_widgets));
/* set endpoints to default mode */
set_scenario_endpoints(codec, NEO_AUDIO_OFF);
/* add neo1973 specific controls */
err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
ARRAY_SIZE(8753_neo1973_controls));
if (err < 0)
return err;
/* set up neo1973 specific audio routes */
err = snd_soc_dapm_add_routes(codec, dapm_routes,
ARRAY_SIZE(dapm_routes));
snd_soc_dapm_sync(codec);
return 0;
}
/*
* BT Codec DAI
*/
static struct snd_soc_dai bt_dai = {
.name = "bluetooth-dai",
.playback = {
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
.capture = {
.channels_min = 1,
.channels_max = 1,
.rates = SNDRV_PCM_RATE_8000,
.formats = SNDRV_PCM_FMTBIT_S16_LE,},
};
static struct snd_soc_dai_link neo1973_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
.stream_name = "WM8753 HiFi",
.platform_name = "s3c24xx-pcm-audio",
.cpu_dai_name = "s3c24xx-i2s",
.codec_dai_name = "wm8753-hifi",
.codec_name = "wm8753-codec.0-0x1a",
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
{ /* Voice via BT */
.name = "Bluetooth",
.stream_name = "Voice",
.platform_name = "s3c24xx-pcm-audio",
.cpu_dai_name = "bluetooth-dai",
.codec_dai_name = "wm8753-voice",
.codec_name = "wm8753-codec.0-0x1a",
.ops = &neo1973_voice_ops,
},
};
static struct snd_soc_card neo1973 = {
.name = "neo1973",
.dai_link = neo1973_dai,
.num_links = ARRAY_SIZE(neo1973_dai),
};
static int lm4857_i2c_probe(struct i2c_client *client,
const struct i2c_device_id *id)
{
pr_debug("Entered %s\n", __func__);
i2c = client;
lm4857_write_regs();
return 0;
}
static int lm4857_i2c_remove(struct i2c_client *client)
{
pr_debug("Entered %s\n", __func__);
i2c = NULL;
return 0;
}
static u8 lm4857_state;
static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
{
pr_debug("Entered %s\n", __func__);
dev_dbg(&dev->dev, "lm4857_suspend\n");
lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
if (lm4857_state) {
lm4857_regs[LM4857_CTRL] &= 0xf0;
lm4857_write_regs();
}
return 0;
}
static int lm4857_resume(struct i2c_client *dev)
{
pr_debug("Entered %s\n", __func__);
if (lm4857_state) {
lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
lm4857_write_regs();
}
return 0;
}
static void lm4857_shutdown(struct i2c_client *dev)
{
pr_debug("Entered %s\n", __func__);
dev_dbg(&dev->dev, "lm4857_shutdown\n");
lm4857_regs[LM4857_CTRL] &= 0xf0;
lm4857_write_regs();
}
static const struct i2c_device_id lm4857_i2c_id[] = {
{ "neo1973_lm4857", 0 },
{ }
};
static struct i2c_driver lm4857_i2c_driver = {
.driver = {
.name = "LM4857 I2C Amp",
.owner = THIS_MODULE,
},
.suspend = lm4857_suspend,
.resume = lm4857_resume,
.shutdown = lm4857_shutdown,
.probe = lm4857_i2c_probe,
.remove = lm4857_i2c_remove,
.id_table = lm4857_i2c_id,
};
static struct platform_device *neo1973_snd_device;
static int __init neo1973_init(void)
{
int ret;
pr_debug("Entered %s\n", __func__);
if (!machine_is_neo1973_gta01()) {
printk(KERN_INFO
"Only GTA01 hardware supported by ASoC driver\n");
return -ENODEV;
}
neo1973_snd_device = platform_device_alloc("soc-audio", -1);
if (!neo1973_snd_device)
return -ENOMEM;
platform_set_drvdata(neo1973_snd_device, &neo1973);
ret = platform_device_add(neo1973_snd_device);
if (ret) {
platform_device_put(neo1973_snd_device);
return ret;
}
ret = i2c_add_driver(&lm4857_i2c_driver);
if (ret != 0)
platform_device_unregister(neo1973_snd_device);
return ret;
}
static void __exit neo1973_exit(void)
{
pr_debug("Entered %s\n", __func__);
i2c_del_driver(&lm4857_i2c_driver);
platform_device_unregister(neo1973_snd_device);
}
module_init(neo1973_init);
module_exit(neo1973_exit);
/* Module information */
MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
MODULE_LICENSE("GPL");
Reference in New Issue View Git Blame Copy Permalink