mirror of
https://mirrors.bfsu.edu.cn/git/linux.git
synced 2024-11-18 01:34:14 +08:00
5ace37bd79
Declare snd_soc_ops structures as const as they are only stored
in the ops field of a snd_soc_dai_link structure. This field is
of type const, so snd_soc_ops structures having this property
can be made const too.
The following .o files did not compile:
sound/soc/fsl/{p1022_rdk.c/p1022_ds.c/mpc8610_hpcd.c}
Signed-off-by: Bhumika Goyal <bhumirks@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
293 lines
8.6 KiB
C
293 lines
8.6 KiB
C
/*
|
|
* 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);
|
|
|
|
/* 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);
|
|
|
|
/* TODO: The SSI driver should figure this out for us */
|
|
switch (channels) {
|
|
case 2:
|
|
snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
|
|
break;
|
|
case 1:
|
|
snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 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 const 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;
|
|
|
|
/* Headphone jack detection */
|
|
snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
|
|
&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
|
|
wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
|
|
|
|
/* Microphone jack detection */
|
|
snd_soc_card_jack_new(rtd->card, "Microphone",
|
|
SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
|
|
mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
|
|
wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
|
|
SND_JACK_BTN_0);
|
|
|
|
snd_soc_dapm_force_enable_pin(&rtd->card->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-ssi.0",
|
|
.codec_name = "wm8350-codec.0-0x1a",
|
|
.init = wm1133_ev1_init,
|
|
.ops = &wm1133_ev1_ops,
|
|
.symmetric_rates = 1,
|
|
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
|
|
SND_SOC_DAIFMT_CBM_CFM,
|
|
};
|
|
|
|
static struct snd_soc_card wm1133_ev1 = {
|
|
.name = "WM1133-EV1",
|
|
.owner = THIS_MODULE,
|
|
.dai_link = &wm1133_ev1_dai,
|
|
.num_links = 1,
|
|
|
|
.dapm_widgets = wm1133_ev1_widgets,
|
|
.num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
|
|
.dapm_routes = wm1133_ev1_map,
|
|
.num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
|
|
};
|
|
|
|
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");
|