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linux/sound/soc/fsl/wm1133-ev1.c
Shawn Guo a23dc69482 ASoC: imx: merge sound/soc/imx into sound/soc/fsl 
Freescale PowerPC and ARM/IMX families share the same SSI IP block.
The patch merges sound/soc/imx into sound/soc/fsl, so that the possible
code sharing and consolidation can happen.

This is a plain merge, except that menuconfig SND_POWERPC_SOC is added
in Kconfig for PowerPC platform as a correspondence to SND_IMX_SOC for
IMX platform.

Signed-off-by: Shawn Guo <shawn.guo@linaro.org>
Acked-by: Sascha Hauer <s.hauer@pengutronix.de>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2012-04-01 11:28:26 +01:00

305 lines
8.9 KiB
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/*
* wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
*
* Copyright (c) 2010 Wolfson Microelectronics plc
* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
*
* Based on an earlier driver for the same hardware by Liam Girdwood.
*
* 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/platform_device.h>
#include <linux/clk.h>
#include <linux/module.h>
#include <sound/core.h>
#include <sound/jack.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "imx-ssi.h"
#include "../codecs/wm8350.h"
#include "imx-audmux.h"
/* There is a silicon mic on the board optionally connected via a solder pad
* SP1. Define this to enable it.
*/
#undef USE_SIMIC
struct _wm8350_audio {
unsigned int channels;
snd_pcm_format_t format;
unsigned int rate;
unsigned int sysclk;
unsigned int bclkdiv;
unsigned int clkdiv;
unsigned int lr_rate;
};
/* in order of power consumption per rate (lowest first) */
static const struct _wm8350_audio wm8350_audio[] = {
/* 16bit mono modes */
{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
/* 16 bit stereo modes */
{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
/* 24bit stereo modes */
{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
};
static int wm1133_ev1_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;
int i, found = 0;
snd_pcm_format_t format = params_format(params);
unsigned int rate = params_rate(params);
unsigned int channels = params_channels(params);
u32 dai_format;
/* find the correct audio parameters */
for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
if (rate == wm8350_audio[i].rate &&
format == wm8350_audio[i].format &&
channels == wm8350_audio[i].channels) {
found = 1;
break;
}
}
if (!found)
return -EINVAL;
/* codec FLL input is 14.75 MHz from MCLK */
snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
dai_format = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBM_CFM;
/* set codec DAI configuration */
snd_soc_dai_set_fmt(codec_dai, dai_format);
/* set cpu DAI configuration */
snd_soc_dai_set_fmt(cpu_dai, dai_format);
/* TODO: The SSI driver should figure this out for us */
switch (channels) {
case 2:
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffc, 0xffffffc, 2, 0);
break;
case 1:
snd_soc_dai_set_tdm_slot(cpu_dai, 0xffffffe, 0xffffffe, 1, 0);
break;
default:
return -EINVAL;
}
/* set MCLK as the codec system clock for DAC and ADC */
snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
/* set codec BCLK division for sample rate */
snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
wm8350_audio[i].bclkdiv);
/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
/* now configure DAC and ADC clocks */
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
snd_soc_dai_set_clkdiv(codec_dai,
WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
return 0;
}
static struct snd_soc_ops wm1133_ev1_ops = {
.hw_params = wm1133_ev1_hw_params,
};
static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
#ifdef USE_SIMIC
SND_SOC_DAPM_MIC("SiMIC", NULL),
#endif
SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
SND_SOC_DAPM_LINE("Line In Jack", NULL),
SND_SOC_DAPM_LINE("Line Out Jack", NULL),
SND_SOC_DAPM_HP("Headphone Jack", NULL),
};
/* imx32ads soc_card audio map */
static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
#ifdef USE_SIMIC
/* SiMIC --> IN1LN (with automatic bias) via SP1 */
{ "IN1LN", NULL, "Mic Bias" },
{ "Mic Bias", NULL, "SiMIC" },
#endif
/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
{ "IN1LN", NULL, "Mic Bias" },
{ "IN1LP", NULL, "Mic1 Jack" },
{ "Mic Bias", NULL, "Mic1 Jack" },
/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
{ "IN1RN", NULL, "Mic Bias" },
{ "IN1RP", NULL, "Mic2 Jack" },
{ "Mic Bias", NULL, "Mic2 Jack" },
/* Line in Jack --> AUX (L+R) */
{ "IN3R", NULL, "Line In Jack" },
{ "IN3L", NULL, "Line In Jack" },
/* Out1 --> Headphone Jack */
{ "Headphone Jack", NULL, "OUT1R" },
{ "Headphone Jack", NULL, "OUT1L" },
/* Out1 --> Line Out Jack */
{ "Line Out Jack", NULL, "OUT2R" },
{ "Line Out Jack", NULL, "OUT2L" },
};
static struct snd_soc_jack hp_jack;
static struct snd_soc_jack_pin hp_jack_pins[] = {
{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
};
static struct snd_soc_jack mic_jack;
static struct snd_soc_jack_pin mic_jack_pins[] = {
{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
};
static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
snd_soc_dapm_new_controls(dapm, wm1133_ev1_widgets,
ARRAY_SIZE(wm1133_ev1_widgets));
snd_soc_dapm_add_routes(dapm, wm1133_ev1_map,
ARRAY_SIZE(wm1133_ev1_map));
/* Headphone jack detection */
snd_soc_jack_new(codec, "Headphone", SND_JACK_HEADPHONE, &hp_jack);
snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
hp_jack_pins);
wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
/* Microphone jack detection */
snd_soc_jack_new(codec, "Microphone",
SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack);
snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins),
mic_jack_pins);
wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
SND_JACK_BTN_0);
snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
return 0;
}
static struct snd_soc_dai_link wm1133_ev1_dai = {
.name = "WM1133-EV1",
.stream_name = "Audio",
.cpu_dai_name = "imx-ssi.0",
.codec_dai_name = "wm8350-hifi",
.platform_name = "imx-fiq-pcm-audio.0",
.codec_name = "wm8350-codec.0-0x1a",
.init = wm1133_ev1_init,
.ops = &wm1133_ev1_ops,
.symmetric_rates = 1,
};
static struct snd_soc_card wm1133_ev1 = {
.name = "WM1133-EV1",
.owner = THIS_MODULE,
.dai_link = &wm1133_ev1_dai,
.num_links = 1,
};
static struct platform_device *wm1133_ev1_snd_device;
static int __init wm1133_ev1_audio_init(void)
{
int ret;
unsigned int ptcr, pdcr;
/* SSI0 mastered by port 5 */
ptcr = IMX_AUDMUX_V2_PTCR_SYN |
IMX_AUDMUX_V2_PTCR_TFSDIR |
IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
IMX_AUDMUX_V2_PTCR_TCLKDIR |
IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
ptcr = IMX_AUDMUX_V2_PTCR_SYN;
pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
if (!wm1133_ev1_snd_device)
return -ENOMEM;
platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
ret = platform_device_add(wm1133_ev1_snd_device);
if (ret)
platform_device_put(wm1133_ev1_snd_device);
return ret;
}
module_init(wm1133_ev1_audio_init);
static void __exit wm1133_ev1_audio_exit(void)
{
platform_device_unregister(wm1133_ev1_snd_device);
}
module_exit(wm1133_ev1_audio_exit);
MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
MODULE_LICENSE("GPL");
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