mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-22 23:40:25 +07:00
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
/*
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* wm1133-ev1.c - Audio for WM1133-EV1 on i.MX31ADS
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*
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* Copyright (c) 2010 Wolfson Microelectronics plc
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* Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
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*
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* Based on an earlier driver for the same hardware by Liam Girdwood.
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU General Public License as published by the
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* Free Software Foundation; either version 2 of the License, or (at your
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* option) any later version.
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*/
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#include <linux/platform_device.h>
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#include <linux/clk.h>
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#include <linux/module.h>
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#include <sound/core.h>
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#include <sound/jack.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include "imx-ssi.h"
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#include "../codecs/wm8350.h"
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#include "imx-audmux.h"
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/* There is a silicon mic on the board optionally connected via a solder pad
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* SP1. Define this to enable it.
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*/
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#undef USE_SIMIC
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struct _wm8350_audio {
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unsigned int channels;
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snd_pcm_format_t format;
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unsigned int rate;
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unsigned int sysclk;
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unsigned int bclkdiv;
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unsigned int clkdiv;
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unsigned int lr_rate;
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};
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/* in order of power consumption per rate (lowest first) */
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static const struct _wm8350_audio wm8350_audio[] = {
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/* 16bit mono modes */
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{1, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000 >> 1,
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WM8350_BCLK_DIV_48, WM8350_DACDIV_3, 16,},
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/* 16 bit stereo modes */
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{2, SNDRV_PCM_FORMAT_S16_LE, 8000, 12288000,
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WM8350_BCLK_DIV_48, WM8350_DACDIV_6, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 16000, 12288000,
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WM8350_BCLK_DIV_24, WM8350_DACDIV_3, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 32000, 12288000,
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WM8350_BCLK_DIV_12, WM8350_DACDIV_1_5, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 48000, 12288000,
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WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 96000, 24576000,
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WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 11025, 11289600,
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WM8350_BCLK_DIV_32, WM8350_DACDIV_4, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 22050, 11289600,
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WM8350_BCLK_DIV_16, WM8350_DACDIV_2, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 44100, 11289600,
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WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
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{2, SNDRV_PCM_FORMAT_S16_LE, 88200, 22579200,
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WM8350_BCLK_DIV_8, WM8350_DACDIV_1, 32,},
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/* 24bit stereo modes */
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{2, SNDRV_PCM_FORMAT_S24_LE, 48000, 12288000,
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WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
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{2, SNDRV_PCM_FORMAT_S24_LE, 96000, 24576000,
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WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
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{2, SNDRV_PCM_FORMAT_S24_LE, 44100, 11289600,
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WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
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{2, SNDRV_PCM_FORMAT_S24_LE, 88200, 22579200,
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WM8350_BCLK_DIV_4, WM8350_DACDIV_1, 64,},
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};
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static int wm1133_ev1_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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int i, found = 0;
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snd_pcm_format_t format = params_format(params);
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unsigned int rate = params_rate(params);
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unsigned int channels = params_channels(params);
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/* find the correct audio parameters */
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for (i = 0; i < ARRAY_SIZE(wm8350_audio); i++) {
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if (rate == wm8350_audio[i].rate &&
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format == wm8350_audio[i].format &&
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channels == wm8350_audio[i].channels) {
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found = 1;
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break;
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}
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}
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if (!found)
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return -EINVAL;
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/* codec FLL input is 14.75 MHz from MCLK */
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snd_soc_dai_set_pll(codec_dai, 0, 0, 14750000, wm8350_audio[i].sysclk);
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/* TODO: The SSI driver should figure this out for us */
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switch (channels) {
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case 2:
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snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 0);
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break;
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case 1:
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snd_soc_dai_set_tdm_slot(cpu_dai, 0x1, 0x1, 1, 0);
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break;
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default:
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return -EINVAL;
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}
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/* set MCLK as the codec system clock for DAC and ADC */
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snd_soc_dai_set_sysclk(codec_dai, WM8350_MCLK_SEL_PLL_MCLK,
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wm8350_audio[i].sysclk, SND_SOC_CLOCK_IN);
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/* set codec BCLK division for sample rate */
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snd_soc_dai_set_clkdiv(codec_dai, WM8350_BCLK_CLKDIV,
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wm8350_audio[i].bclkdiv);
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/* DAI is synchronous and clocked with DAC LRCLK & ADC LRC */
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snd_soc_dai_set_clkdiv(codec_dai,
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WM8350_DACLR_CLKDIV, wm8350_audio[i].lr_rate);
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snd_soc_dai_set_clkdiv(codec_dai,
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WM8350_ADCLR_CLKDIV, wm8350_audio[i].lr_rate);
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/* now configure DAC and ADC clocks */
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snd_soc_dai_set_clkdiv(codec_dai,
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WM8350_DAC_CLKDIV, wm8350_audio[i].clkdiv);
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snd_soc_dai_set_clkdiv(codec_dai,
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WM8350_ADC_CLKDIV, wm8350_audio[i].clkdiv);
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return 0;
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}
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static const struct snd_soc_ops wm1133_ev1_ops = {
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.hw_params = wm1133_ev1_hw_params,
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};
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static const struct snd_soc_dapm_widget wm1133_ev1_widgets[] = {
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#ifdef USE_SIMIC
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SND_SOC_DAPM_MIC("SiMIC", NULL),
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#endif
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SND_SOC_DAPM_MIC("Mic1 Jack", NULL),
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SND_SOC_DAPM_MIC("Mic2 Jack", NULL),
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SND_SOC_DAPM_LINE("Line In Jack", NULL),
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SND_SOC_DAPM_LINE("Line Out Jack", NULL),
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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};
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/* imx32ads soc_card audio map */
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static const struct snd_soc_dapm_route wm1133_ev1_map[] = {
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#ifdef USE_SIMIC
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/* SiMIC --> IN1LN (with automatic bias) via SP1 */
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{ "IN1LN", NULL, "Mic Bias" },
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{ "Mic Bias", NULL, "SiMIC" },
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#endif
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/* Mic 1 Jack --> IN1LN and IN1LP (with automatic bias) */
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{ "IN1LN", NULL, "Mic Bias" },
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{ "IN1LP", NULL, "Mic1 Jack" },
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{ "Mic Bias", NULL, "Mic1 Jack" },
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/* Mic 2 Jack --> IN1RN and IN1RP (with automatic bias) */
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{ "IN1RN", NULL, "Mic Bias" },
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{ "IN1RP", NULL, "Mic2 Jack" },
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{ "Mic Bias", NULL, "Mic2 Jack" },
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/* Line in Jack --> AUX (L+R) */
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{ "IN3R", NULL, "Line In Jack" },
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{ "IN3L", NULL, "Line In Jack" },
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/* Out1 --> Headphone Jack */
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{ "Headphone Jack", NULL, "OUT1R" },
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{ "Headphone Jack", NULL, "OUT1L" },
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/* Out1 --> Line Out Jack */
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{ "Line Out Jack", NULL, "OUT2R" },
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{ "Line Out Jack", NULL, "OUT2L" },
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};
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static struct snd_soc_jack hp_jack;
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static struct snd_soc_jack_pin hp_jack_pins[] = {
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{ .pin = "Headphone Jack", .mask = SND_JACK_HEADPHONE },
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};
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static struct snd_soc_jack mic_jack;
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static struct snd_soc_jack_pin mic_jack_pins[] = {
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{ .pin = "Mic1 Jack", .mask = SND_JACK_MICROPHONE },
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{ .pin = "Mic2 Jack", .mask = SND_JACK_MICROPHONE },
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};
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static int wm1133_ev1_init(struct snd_soc_pcm_runtime *rtd)
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{
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struct snd_soc_codec *codec = rtd->codec;
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/* Headphone jack detection */
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snd_soc_card_jack_new(rtd->card, "Headphone", SND_JACK_HEADPHONE,
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&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
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wm8350_hp_jack_detect(codec, WM8350_JDR, &hp_jack, SND_JACK_HEADPHONE);
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/* Microphone jack detection */
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snd_soc_card_jack_new(rtd->card, "Microphone",
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SND_JACK_MICROPHONE | SND_JACK_BTN_0, &mic_jack,
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mic_jack_pins, ARRAY_SIZE(mic_jack_pins));
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wm8350_mic_jack_detect(codec, &mic_jack, SND_JACK_MICROPHONE,
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SND_JACK_BTN_0);
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snd_soc_dapm_force_enable_pin(&rtd->card->dapm, "Mic Bias");
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return 0;
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}
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static struct snd_soc_dai_link wm1133_ev1_dai = {
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.name = "WM1133-EV1",
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.stream_name = "Audio",
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.cpu_dai_name = "imx-ssi.0",
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.codec_dai_name = "wm8350-hifi",
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.platform_name = "imx-ssi.0",
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.codec_name = "wm8350-codec.0-0x1a",
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.init = wm1133_ev1_init,
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.ops = &wm1133_ev1_ops,
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.symmetric_rates = 1,
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBM_CFM,
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};
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static struct snd_soc_card wm1133_ev1 = {
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.name = "WM1133-EV1",
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.owner = THIS_MODULE,
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.dai_link = &wm1133_ev1_dai,
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.num_links = 1,
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.dapm_widgets = wm1133_ev1_widgets,
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.num_dapm_widgets = ARRAY_SIZE(wm1133_ev1_widgets),
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.dapm_routes = wm1133_ev1_map,
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.num_dapm_routes = ARRAY_SIZE(wm1133_ev1_map),
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};
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static struct platform_device *wm1133_ev1_snd_device;
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static int __init wm1133_ev1_audio_init(void)
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{
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int ret;
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unsigned int ptcr, pdcr;
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/* SSI0 mastered by port 5 */
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ptcr = IMX_AUDMUX_V2_PTCR_SYN |
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IMX_AUDMUX_V2_PTCR_TFSDIR |
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IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT5_SSI_PINS_5) |
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IMX_AUDMUX_V2_PTCR_TCLKDIR |
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IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
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pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT5_SSI_PINS_5);
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imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, ptcr, pdcr);
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ptcr = IMX_AUDMUX_V2_PTCR_SYN;
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pdcr = IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0);
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imx_audmux_v2_configure_port(MX31_AUDMUX_PORT5_SSI_PINS_5, ptcr, pdcr);
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wm1133_ev1_snd_device = platform_device_alloc("soc-audio", -1);
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if (!wm1133_ev1_snd_device)
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return -ENOMEM;
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platform_set_drvdata(wm1133_ev1_snd_device, &wm1133_ev1);
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ret = platform_device_add(wm1133_ev1_snd_device);
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if (ret)
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platform_device_put(wm1133_ev1_snd_device);
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return ret;
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}
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module_init(wm1133_ev1_audio_init);
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static void __exit wm1133_ev1_audio_exit(void)
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{
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platform_device_unregister(wm1133_ev1_snd_device);
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}
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module_exit(wm1133_ev1_audio_exit);
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MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
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MODULE_DESCRIPTION("Audio for WM1133-EV1 on i.MX31ADS");
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MODULE_LICENSE("GPL");
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