Staging
v0.7.0
v0.7.0
https://github.com/torvalds/linux
Tip revision: 8005c49d9aea74d382f474ce11afbbc7d7130bec authored by Linus Torvalds on 16 November 2015, 01:00:27 UTC
Linux 4.4-rc1
Linux 4.4-rc1
Tip revision: 8005c49
neo1973_wm8753.c
/*
* neo1973_wm8753.c -- SoC audio for Openmoko Neo1973 and Freerunner devices
*
* Copyright 2007 Openmoko Inc
* Author: Graeme Gregory <graeme@openmoko.org>
* Copyright 2007 Wolfson Microelectronics PLC.
* Author: Graeme Gregory
* graeme.gregory@wolfsonmicro.com or 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/platform_device.h>
#include <linux/gpio.h>
#include <sound/soc.h>
#include <mach/gpio-samsung.h>
#include <asm/mach-types.h>
#include "regs-iis.h"
#include "../codecs/wm8753.h"
#include "s3c24xx-i2s.h"
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;
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 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;
/* 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;
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 */
/* 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;
/* configure 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;
/* 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 gta02_speaker_enabled;
static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
gta02_speaker_enabled = ucontrol->value.integer.value[0];
gpio_set_value(S3C2410_GPJ(2), !gta02_speaker_enabled);
return 0;
}
static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
ucontrol->value.integer.value[0] = gta02_speaker_enabled;
return 0;
}
static int lm4853_event(struct snd_soc_dapm_widget *w,
struct snd_kcontrol *k, int event)
{
gpio_set_value(S3C2410_GPJ(1), SND_SOC_DAPM_EVENT_OFF(event));
return 0;
}
static const struct snd_soc_dapm_widget neo1973_wm8753_dapm_widgets[] = {
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),
SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
};
static const struct snd_soc_dapm_route neo1973_wm8753_routes[] = {
/* 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"},
/* Connect the ALC pins */
{"ACIN", NULL, "ACOP"},
/* Connections to the amp */
{"Stereo Out", NULL, "LOUT1"},
{"Stereo Out", NULL, "ROUT1"},
/* Call Speaker */
{"Handset Spk", NULL, "LOUT2"},
{"Handset Spk", NULL, "ROUT2"},
};
static const struct snd_kcontrol_new neo1973_wm8753_controls[] = {
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"),
SOC_DAPM_PIN_SWITCH("Stereo Out"),
SOC_SINGLE_BOOL_EXT("Amp Spk Switch", 0,
lm4853_get_spk,
lm4853_set_spk),
};
static int neo1973_wm8753_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_card *card = rtd->card;
/* set endpoints to default off mode */
snd_soc_dapm_disable_pin(&card->dapm, "GSM Line Out");
snd_soc_dapm_disable_pin(&card->dapm, "GSM Line In");
snd_soc_dapm_disable_pin(&card->dapm, "Headset Mic");
snd_soc_dapm_disable_pin(&card->dapm, "Handset Mic");
snd_soc_dapm_disable_pin(&card->dapm, "Stereo Out");
snd_soc_dapm_disable_pin(&card->dapm, "Handset Spk");
/* allow audio paths from the GSM modem to run during suspend */
snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line Out");
snd_soc_dapm_ignore_suspend(&card->dapm, "GSM Line In");
snd_soc_dapm_ignore_suspend(&card->dapm, "Headset Mic");
snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Mic");
snd_soc_dapm_ignore_suspend(&card->dapm, "Stereo Out");
snd_soc_dapm_ignore_suspend(&card->dapm, "Handset Spk");
return 0;
}
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-iis",
.cpu_dai_name = "s3c24xx-iis",
.codec_dai_name = "wm8753-hifi",
.codec_name = "wm8753.0-001a",
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM,
.init = neo1973_wm8753_init,
.ops = &neo1973_hifi_ops,
},
{ /* Voice via BT */
.name = "Bluetooth",
.stream_name = "Voice",
.cpu_dai_name = "bt-sco-pcm",
.codec_dai_name = "wm8753-voice",
.codec_name = "wm8753.0-001a",
.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &neo1973_voice_ops,
},
};
static struct snd_soc_aux_dev neo1973_aux_devs[] = {
{
.name = "dfbmcs320",
.codec_name = "dfbmcs320.0",
},
};
static struct snd_soc_codec_conf neo1973_codec_conf[] = {
{
.dev_name = "lm4857.0-007c",
.name_prefix = "Amp",
},
};
static const struct gpio neo1973_gta02_gpios[] = {
{ S3C2410_GPJ(2), GPIOF_OUT_INIT_HIGH, "GTA02_HP_IN" },
{ S3C2410_GPJ(1), GPIOF_OUT_INIT_HIGH, "GTA02_AMP_SHUT" },
};
static struct snd_soc_card neo1973 = {
.name = "neo1973",
.owner = THIS_MODULE,
.dai_link = neo1973_dai,
.num_links = ARRAY_SIZE(neo1973_dai),
.aux_dev = neo1973_aux_devs,
.num_aux_devs = ARRAY_SIZE(neo1973_aux_devs),
.codec_conf = neo1973_codec_conf,
.num_configs = ARRAY_SIZE(neo1973_codec_conf),
.controls = neo1973_wm8753_controls,
.num_controls = ARRAY_SIZE(neo1973_wm8753_controls),
.dapm_widgets = neo1973_wm8753_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(neo1973_wm8753_dapm_widgets),
.dapm_routes = neo1973_wm8753_routes,
.num_dapm_routes = ARRAY_SIZE(neo1973_wm8753_routes),
.fully_routed = true,
};
static struct platform_device *neo1973_snd_device;
static int __init neo1973_init(void)
{
int ret;
if (!machine_is_neo1973_gta02())
return -ENODEV;
if (machine_is_neo1973_gta02()) {
neo1973.name = "neo1973gta02";
neo1973.num_aux_devs = 1;
ret = gpio_request_array(neo1973_gta02_gpios,
ARRAY_SIZE(neo1973_gta02_gpios));
if (ret)
return ret;
}
neo1973_snd_device = platform_device_alloc("soc-audio", -1);
if (!neo1973_snd_device) {
ret = -ENOMEM;
goto err_gpio_free;
}
platform_set_drvdata(neo1973_snd_device, &neo1973);
ret = platform_device_add(neo1973_snd_device);
if (ret)
goto err_put_device;
return 0;
err_put_device:
platform_device_put(neo1973_snd_device);
err_gpio_free:
if (machine_is_neo1973_gta02()) {
gpio_free_array(neo1973_gta02_gpios,
ARRAY_SIZE(neo1973_gta02_gpios));
}
return ret;
}
module_init(neo1973_init);
static void __exit neo1973_exit(void)
{
platform_device_unregister(neo1973_snd_device);
if (machine_is_neo1973_gta02()) {
gpio_free_array(neo1973_gta02_gpios,
ARRAY_SIZE(neo1973_gta02_gpios));
}
}
module_exit(neo1973_exit);
/* Module information */
MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org, www.openmoko.org");
MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 and Frerunner");
MODULE_LICENSE("GPL");