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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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86c3304181
Added support for EDB93xx sound with CS4271 CODEC. Features: - Playback, Capture - Sample rates from 8kHz to 96kHz tested Signed-off-by: Alexander Sverdlin <subaparts@yandex.ru> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
143 lines
3.7 KiB
C
143 lines
3.7 KiB
C
/*
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* SoC audio for EDB93xx
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*
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* Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* This driver support CS4271 codec being master or slave, working
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* in control port mode, connected either via SPI or I2C.
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* The data format accepted is I2S or left-justified.
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* DAPM support not implemented.
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*/
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#include <linux/platform_device.h>
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#include <linux/gpio.h>
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#include <sound/core.h>
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#include <sound/pcm.h>
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#include <sound/soc.h>
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#include <asm/mach-types.h>
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#include <mach/hardware.h>
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#include "ep93xx-pcm.h"
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#define edb93xx_has_audio() (machine_is_edb9301() || \
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machine_is_edb9302() || \
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machine_is_edb9302a() || \
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machine_is_edb9307a() || \
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machine_is_edb9315a())
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static int edb93xx_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 err;
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unsigned int rate = params_rate(params);
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/*
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* We set LRCLK equal to `rate' and SCLK = LRCLK * 64,
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* because our sample size is 32 bit * 2 channels.
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* I2S standard permits us to transmit more bits than
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* the codec uses.
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* MCLK = SCLK * 4 is the best recommended value,
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* but we have to fall back to ratio 2 for higher
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* sample rates.
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*/
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unsigned int mclk_rate = rate * 64 * ((rate <= 48000) ? 4 : 2);
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err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBS_CFS);
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if (err)
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return err;
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err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
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SND_SOC_DAIFMT_NB_IF |
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SND_SOC_DAIFMT_CBS_CFS);
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if (err)
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return err;
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err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
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SND_SOC_CLOCK_IN);
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if (err)
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return err;
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return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
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SND_SOC_CLOCK_OUT);
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}
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static struct snd_soc_ops edb93xx_ops = {
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.hw_params = edb93xx_hw_params,
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};
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static struct snd_soc_dai_link edb93xx_dai = {
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.name = "CS4271",
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.stream_name = "CS4271 HiFi",
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.platform_name = "ep93xx-pcm-audio",
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.cpu_dai_name = "ep93xx-i2s",
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.codec_name = "spi0.0",
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.codec_dai_name = "cs4271-hifi",
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.ops = &edb93xx_ops,
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};
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static struct snd_soc_card snd_soc_edb93xx = {
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.name = "EDB93XX",
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.dai_link = &edb93xx_dai,
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.num_links = 1,
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};
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static struct platform_device *edb93xx_snd_device;
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static int __init edb93xx_init(void)
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{
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int ret;
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if (!edb93xx_has_audio())
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return -ENODEV;
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ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
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EP93XX_SYSCON_I2SCLKDIV_ORIDE |
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EP93XX_SYSCON_I2SCLKDIV_SPOL);
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if (ret)
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return ret;
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edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
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if (!edb93xx_snd_device) {
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ret = -ENOMEM;
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goto free_i2s;
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}
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platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
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ret = platform_device_add(edb93xx_snd_device);
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if (ret)
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goto device_put;
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return 0;
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device_put:
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platform_device_put(edb93xx_snd_device);
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free_i2s:
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ep93xx_i2s_release();
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return ret;
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}
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module_init(edb93xx_init);
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static void __exit edb93xx_exit(void)
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{
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platform_device_unregister(edb93xx_snd_device);
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ep93xx_i2s_release();
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}
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module_exit(edb93xx_exit);
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MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
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MODULE_DESCRIPTION("ALSA SoC EDB93xx");
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MODULE_LICENSE("GPL");
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