Revision 761bfb999868c413aabed8caa345694836ec6f11 authored by Alex Deucher on 06 August 2013, 17:34:00 UTC, committed by Alex Deucher on 07 August 2013, 21:37:19 UTC
The rlc is required for dpm to work properly, so if the rlc ucode is missing, don't enable dpm. Enabling dpm without the rlc enabled can result in hangs. Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
1 parent f61d5b4
jive_wm8750.c
/* sound/soc/samsung/jive_wm8750.c
*
* Copyright 2007,2008 Simtec Electronics
*
* Based on sound/soc/pxa/spitz.c
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* 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 <sound/soc.h>
#include <asm/mach-types.h>
#include "s3c2412-i2s.h"
#include "../codecs/wm8750.h"
static const struct snd_soc_dapm_route audio_map[] = {
{ "Headphone Jack", NULL, "LOUT1" },
{ "Headphone Jack", NULL, "ROUT1" },
{ "Internal Speaker", NULL, "LOUT2" },
{ "Internal Speaker", NULL, "ROUT2" },
{ "LINPUT1", NULL, "Line Input" },
{ "RINPUT1", NULL, "Line Input" },
};
static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
SND_SOC_DAPM_HP("Headphone Jack", NULL),
SND_SOC_DAPM_SPK("Internal Speaker", NULL),
SND_SOC_DAPM_LINE("Line In", NULL),
};
static int jive_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;
struct s3c_i2sv2_rate_calc div;
unsigned int clk = 0;
int ret = 0;
switch (params_rate(params)) {
case 8000:
case 16000:
case 48000:
case 96000:
clk = 12288000;
break;
case 11025:
case 22050:
case 44100:
clk = 11289600;
break;
}
s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
s3c_i2sv2_get_clock(cpu_dai));
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
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_CBS_CFS);
if (ret < 0)
return ret;
/* set the codec system clock for DAC and ADC */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER,
div.clk_div - 1);
if (ret < 0)
return ret;
return 0;
}
static struct snd_soc_ops jive_ops = {
.hw_params = jive_hw_params,
};
static int jive_wm8750_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
/* These endpoints are not being used. */
snd_soc_dapm_nc_pin(dapm, "LINPUT2");
snd_soc_dapm_nc_pin(dapm, "RINPUT2");
snd_soc_dapm_nc_pin(dapm, "LINPUT3");
snd_soc_dapm_nc_pin(dapm, "RINPUT3");
snd_soc_dapm_nc_pin(dapm, "OUT3");
snd_soc_dapm_nc_pin(dapm, "MONO");
return 0;
}
static struct snd_soc_dai_link jive_dai = {
.name = "wm8750",
.stream_name = "WM8750",
.cpu_dai_name = "s3c2412-i2s",
.codec_dai_name = "wm8750-hifi",
.platform_name = "s3c2412-i2s",
.codec_name = "wm8750.0-001a",
.init = jive_wm8750_init,
.ops = &jive_ops,
};
/* jive audio machine driver */
static struct snd_soc_card snd_soc_machine_jive = {
.name = "Jive",
.owner = THIS_MODULE,
.dai_link = &jive_dai,
.num_links = 1,
.dapm_widgets = wm8750_dapm_widgets,
.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
};
static struct platform_device *jive_snd_device;
static int __init jive_init(void)
{
int ret;
if (!machine_is_jive())
return 0;
printk("JIVE WM8750 Audio support\n");
jive_snd_device = platform_device_alloc("soc-audio", -1);
if (!jive_snd_device)
return -ENOMEM;
platform_set_drvdata(jive_snd_device, &snd_soc_machine_jive);
ret = platform_device_add(jive_snd_device);
if (ret)
platform_device_put(jive_snd_device);
return ret;
}
static void __exit jive_exit(void)
{
platform_device_unregister(jive_snd_device);
}
module_init(jive_init);
module_exit(jive_exit);
MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
MODULE_DESCRIPTION("ALSA SoC Jive Audio support");
MODULE_LICENSE("GPL");
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