Merge branch 'gta02-audio' into for-2.6.32

This commit is contained in:
Mark Brown 2009-07-30 13:21:38 +01:00
commit a1daf67d72
3 changed files with 509 additions and 0 deletions

View file

@ -38,6 +38,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
Say Y if you want to add support for SoC audio on smdk2440
with the WM8753.
config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
select SND_S3C24XX_SOC_I2S
select SND_SOC_WM8753
help
This driver provides audio support for the Openmoko Neo FreeRunner
smartphone.
config SND_S3C24XX_SOC_JIVE_WM8750
tristate "SoC I2S Audio support for Jive"
depends on SND_S3C24XX_SOC && MACH_JIVE

View file

@ -16,12 +16,14 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
# S3C24XX Machine Support
snd-soc-jive-wm8750-objs := jive_wm8750.o
snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o

View file

@ -0,0 +1,498 @@
/*
* neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02)
*
* Copyright 2007 Openmoko Inc
* Author: Graeme Gregory <graeme@openmoko.org>
* Copyright 2007 Wolfson Microelectronics PLC.
* Author: Graeme Gregory <linux@wolfsonmicro.com>
* Copyright 2009 Wolfson Microelectronics
*
* 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/gpio.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/mach-types.h>
#include <plat/regs-iis.h>
#include <mach/regs-clock.h>
#include <asm/io.h>
#include <mach/gta02.h>
#include "../codecs/wm8753.h"
#include "s3c24xx-pcm.h"
#include "s3c24xx-i2s.h"
static struct snd_soc_card neo1973_gta02;
static int neo1973_gta02_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->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
unsigned int pll_out = 0, bclk = 0;
int ret = 0;
unsigned long iis_clkrate;
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,
iis_clkrate / 4, pll_out);
if (ret < 0)
return ret;
return 0;
}
static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
}
/*
* Neo1973 WM8753 HiFi DAI opserations.
*/
static struct snd_soc_ops neo1973_gta02_hifi_ops = {
.hw_params = neo1973_gta02_hifi_hw_params,
.hw_free = neo1973_gta02_hifi_hw_free,
};
static int neo1973_gta02_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->dai->codec_dai;
unsigned int pcmdiv = 0;
int ret = 0;
unsigned long iis_clkrate;
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,
iis_clkrate / 4, 12288000);
if (ret < 0)
return ret;
return 0;
}
static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
/* disable the PLL */
return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
}
static struct snd_soc_ops neo1973_gta02_voice_ops = {
.hw_params = neo1973_gta02_voice_hw_params,
.hw_free = neo1973_gta02_voice_hw_free,
};
#define LM4853_AMP 1
#define LM4853_SPK 2
static u8 lm4853_state;
/* This has no effect, it exists only to maintain compatibility with
* existing ALSA state files.
*/
static int lm4853_set_state(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int val = ucontrol->value.integer.value[0];
if (val)
lm4853_state |= LM4853_AMP;
else
lm4853_state &= ~LM4853_AMP;
return 0;
}
static int lm4853_get_state(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
return 0;
}
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
int val = ucontrol->value.integer.value[0];
if (val) {
lm4853_state |= LM4853_SPK;
gpio_set_value(GTA02_GPIO_HP_IN, 0);
} else {
lm4853_state &= ~LM4853_SPK;
gpio_set_value(GTA02_GPIO_HP_IN, 1);
}
return 0;
}
static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
return 0;
}
static int lm4853_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k,
int event)
{
gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value));
return 0;
}
static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
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("Handset Mic", NULL),
SND_SOC_DAPM_SPK("Handset Spk", NULL),
};
/* example machine audio_mapnections */
static const struct snd_soc_dapm_route audio_map[] = {
/* Connections to the lm4853 amp */
{"Stereo Out", NULL, "LOUT1"},
{"Stereo 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, "Handset Mic"},
/* Call Speaker */
{"Handset Spk", NULL, "LOUT2"},
{"Handset Spk", NULL, "ROUT2"},
/* Connect the ALC pins */
{"ACIN", NULL, "ACOP"},
};
static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
SOC_DAPM_PIN_SWITCH("Stereo Out"),
SOC_DAPM_PIN_SWITCH("GSM Line Out"),
SOC_DAPM_PIN_SWITCH("GSM Line In"),
SOC_DAPM_PIN_SWITCH("Headset Mic"),
SOC_DAPM_PIN_SWITCH("Handset Mic"),
SOC_DAPM_PIN_SWITCH("Handset Spk"),
/* This has no effect, it exists only to maintain compatibility with
* existing ALSA state files.
*/
SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
lm4853_get_state,
lm4853_set_state),
SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
lm4853_get_spk,
lm4853_set_spk),
};
/*
* This is an example machine initialisation for a wm8753 connected to a
* neo1973 GTA02.
*/
static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
{
int err;
/* set up NC codec pins */
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 gta02 specific widgets */
snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
ARRAY_SIZE(wm8753_dapm_widgets));
/* add neo1973 gta02 specific controls */
err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
ARRAY_SIZE(wm8753_neo1973_gta02_controls));
if (err < 0)
return err;
/* set up neo1973 gta02 specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* set endpoints to default off mode */
snd_soc_dapm_disable_pin(codec, "Stereo 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, "Handset Mic");
snd_soc_dapm_disable_pin(codec, "Handset Spk");
snd_soc_dapm_sync(codec);
return 0;
}
/*
* BT Codec DAI
*/
static struct snd_soc_dai bt_dai = {
.name = "Bluetooth",
.id = 0,
.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_gta02_dai[] = {
{ /* Hifi Playback - for similatious use with voice below */
.name = "WM8753",
.stream_name = "WM8753 HiFi",
.cpu_dai = &s3c24xx_i2s_dai,
.codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
.init = neo1973_gta02_wm8753_init,
.ops = &neo1973_gta02_hifi_ops,
},
{ /* Voice via BT */
.name = "Bluetooth",
.stream_name = "Voice",
.cpu_dai = &bt_dai,
.codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
.ops = &neo1973_gta02_voice_ops,
},
};
static struct snd_soc_card neo1973_gta02 = {
.name = "neo1973-gta02",
.platform = &s3c24xx_soc_platform,
.dai_link = neo1973_gta02_dai,
.num_links = ARRAY_SIZE(neo1973_gta02_dai),
};
static struct snd_soc_device neo1973_gta02_snd_devdata = {
.card = &neo1973_gta02,
.codec_dev = &soc_codec_dev_wm8753,
};
static struct platform_device *neo1973_gta02_snd_device;
static int __init neo1973_gta02_init(void)
{
int ret;
if (!machine_is_neo1973_gta02()) {
printk(KERN_INFO
"Only GTA02 is supported by this ASoC driver\n");
return -ENODEV;
}
/* register bluetooth DAI here */
ret = snd_soc_register_dai(&bt_dai);
if (ret)
return ret;
neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
if (!neo1973_gta02_snd_device)
return -ENOMEM;
platform_set_drvdata(neo1973_gta02_snd_device,
&neo1973_gta02_snd_devdata);
neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev;
ret = platform_device_add(neo1973_gta02_snd_device);
if (ret) {
platform_device_put(neo1973_gta02_snd_device);
return ret;
}
/* Initialise GPIOs used by amp */
ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
if (ret) {
pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
goto err_unregister_device;
}
ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1);
if (ret) {
pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
goto err_free_gpio_hp_in;
}
ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
if (ret) {
pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
goto err_free_gpio_hp_in;
}
ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
if (ret) {
pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
goto err_free_gpio_amp_shut;
}
return 0;
err_free_gpio_amp_shut:
gpio_free(GTA02_GPIO_AMP_SHUT);
err_free_gpio_hp_in:
gpio_free(GTA02_GPIO_HP_IN);
err_unregister_device:
platform_device_unregister(neo1973_gta02_snd_device);
return ret;
}
module_init(neo1973_gta02_init);
static void __exit neo1973_gta02_exit(void)
{
snd_soc_unregister_dai(&bt_dai);
platform_device_unregister(neo1973_gta02_snd_device);
gpio_free(GTA02_GPIO_HP_IN);
gpio_free(GTA02_GPIO_AMP_SHUT);
}
module_exit(neo1973_gta02_exit);
/* Module information */
MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
MODULE_LICENSE("GPL");