android_kernel_xiaomi_sm8350/sound/soc/davinci/davinci-evm.c
David Brownell 05d5e991a7 ASoC: Clocking fixes for davinci-evm.c
Let's have audio playback not sound like chipmunks, 'k? :)

ASP1 on the DM355 EVM uses a 27 MHz external audio clock, not
the slower clock used with ASP0 on the DM6446 EVM.

Also, that slower ASP0 clock on the DM6446 is 12.288 MHz,
not 22.5792 MHz ... 48 KHz sample rate (x256), not a double
speed 44.1 KHz sample rate (which could be done, but isn't
what the board init code now sets up).

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-01-05 17:47:17 +00:00

216 lines
5.5 KiB
C

/*
* ASoC driver for TI DAVINCI EVM platform
*
* Author: Vladimir Barinov, <vbarinov@embeddedalley.com>
* Copyright: (C) 2007 MontaVista Software, Inc., <source@mvista.com>
*
* 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.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <asm/dma.h>
#include <mach/hardware.h>
#include "../codecs/tlv320aic3x.h"
#include "davinci-pcm.h"
#include "davinci-i2s.h"
#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B | \
SND_SOC_DAIFMT_CBM_CFM | SND_SOC_DAIFMT_IB_NF)
static int evm_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;
int ret = 0;
unsigned sysclk;
/* ASP1 on DM355 EVM is clocked by an external oscillator */
if (machine_is_davinci_dm355_evm())
sysclk = 27000000;
/* ASP0 in DM6446 EVM is clocked by U55, as configured by
* board-dm644x-evm.c using GPIOs from U18. There are six
* options; here we "know" we use a 48 KHz sample rate.
*/
else if (machine_is_davinci_evm())
sysclk = 12288000;
else
return -EINVAL;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
if (ret < 0)
return ret;
/* set the codec system clock */
ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops evm_ops = {
.hw_params = evm_hw_params,
};
/* davinci-evm machine dapm widgets */
static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_LINE("Line Out", NULL),
SND_SOC_DAPM_MIC("Mic Jack", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
};
/* davinci-evm machine audio_mapnections to the codec pins */
static const struct snd_soc_dapm_route audio_map[] = {
/* Headphone connected to HPLOUT, HPROUT */
{"Headphone Jack", NULL, "HPLOUT"},
{"Headphone Jack", NULL, "HPROUT"},
/* Line Out connected to LLOUT, RLOUT */
{"Line Out", NULL, "LLOUT"},
{"Line Out", NULL, "RLOUT"},
/* Mic connected to (MIC3L | MIC3R) */
{"MIC3L", NULL, "Mic Bias 2V"},
{"MIC3R", NULL, "Mic Bias 2V"},
{"Mic Bias 2V", NULL, "Mic Jack"},
/* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
{"LINE1L", NULL, "Line In"},
{"LINE2L", NULL, "Line In"},
{"LINE1R", NULL, "Line In"},
{"LINE2R", NULL, "Line In"},
};
/* Logic for a aic3x as connected on a davinci-evm */
static int evm_aic3x_init(struct snd_soc_codec *codec)
{
/* Add davinci-evm specific widgets */
snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets,
ARRAY_SIZE(aic3x_dapm_widgets));
/* Set up davinci-evm specific audio path audio_map */
snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
/* not connected */
snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
snd_soc_dapm_disable_pin(codec, "HPLCOM");
snd_soc_dapm_disable_pin(codec, "HPRCOM");
/* always connected */
snd_soc_dapm_enable_pin(codec, "Headphone Jack");
snd_soc_dapm_enable_pin(codec, "Line Out");
snd_soc_dapm_enable_pin(codec, "Mic Jack");
snd_soc_dapm_enable_pin(codec, "Line In");
snd_soc_dapm_sync(codec);
return 0;
}
/* davinci-evm digital audio interface glue - connects codec <--> CPU */
static struct snd_soc_dai_link evm_dai = {
.name = "TLV320AIC3X",
.stream_name = "AIC3X",
.cpu_dai = &davinci_i2s_dai,
.codec_dai = &aic3x_dai,
.init = evm_aic3x_init,
.ops = &evm_ops,
};
/* davinci-evm audio machine driver */
static struct snd_soc_card snd_soc_card_evm = {
.name = "DaVinci EVM",
.platform = &davinci_soc_platform,
.dai_link = &evm_dai,
.num_links = 1,
};
/* evm audio private data */
static struct aic3x_setup_data evm_aic3x_setup = {
.i2c_bus = 0,
.i2c_address = 0x1b,
};
/* evm audio subsystem */
static struct snd_soc_device evm_snd_devdata = {
.card = &snd_soc_card_evm,
.codec_dev = &soc_codec_dev_aic3x,
.codec_data = &evm_aic3x_setup,
};
static struct resource evm_snd_resources[] = {
{
.start = DAVINCI_MCBSP_BASE,
.end = DAVINCI_MCBSP_BASE + SZ_8K - 1,
.flags = IORESOURCE_MEM,
},
};
static struct evm_snd_platform_data evm_snd_data = {
.tx_dma_ch = DM644X_DMACH_MCBSP_TX,
.rx_dma_ch = DM644X_DMACH_MCBSP_RX,
};
static struct platform_device *evm_snd_device;
static int __init evm_init(void)
{
int ret;
evm_snd_device = platform_device_alloc("soc-audio", 0);
if (!evm_snd_device)
return -ENOMEM;
platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
evm_snd_devdata.dev = &evm_snd_device->dev;
evm_snd_device->dev.platform_data = &evm_snd_data;
ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
ARRAY_SIZE(evm_snd_resources));
if (ret) {
platform_device_put(evm_snd_device);
return ret;
}
ret = platform_device_add(evm_snd_device);
if (ret)
platform_device_put(evm_snd_device);
return ret;
}
static void __exit evm_exit(void)
{
platform_device_unregister(evm_snd_device);
}
module_init(evm_init);
module_exit(evm_exit);
MODULE_AUTHOR("Vladimir Barinov");
MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver");
MODULE_LICENSE("GPL");