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https://github.com/AuxXxilium/linux_dsm_epyc7002.git
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9423082951
ASoC is now supporting modern style dai_link (= snd_soc_dai_link_component) for CPU/Codec/Platform. This patch switches to use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
144 lines
3.3 KiB
C
144 lines
3.3 KiB
C
// SPDX-License-Identifier: GPL-2.0
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//
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// Copyright 2007,2008 Simtec Electronics
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//
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// Based on sound/soc/pxa/spitz.c
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// Copyright 2005 Wolfson Microelectronics PLC.
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// Copyright 2005 Openedhand Ltd.
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#include <linux/module.h>
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#include <sound/soc.h>
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#include <asm/mach-types.h>
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#include "s3c2412-i2s.h"
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#include "../codecs/wm8750.h"
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static const struct snd_soc_dapm_route audio_map[] = {
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{ "Headphone Jack", NULL, "LOUT1" },
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{ "Headphone Jack", NULL, "ROUT1" },
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{ "Internal Speaker", NULL, "LOUT2" },
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{ "Internal Speaker", NULL, "ROUT2" },
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{ "LINPUT1", NULL, "Line Input" },
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{ "RINPUT1", NULL, "Line Input" },
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};
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static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_SPK("Internal Speaker", NULL),
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SND_SOC_DAPM_LINE("Line In", NULL),
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};
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static int jive_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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struct s3c_i2sv2_rate_calc div;
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unsigned int clk = 0;
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int ret = 0;
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switch (params_rate(params)) {
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case 8000:
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case 16000:
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case 48000:
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case 96000:
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clk = 12288000;
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break;
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case 11025:
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case 22050:
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case 44100:
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clk = 11289600;
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break;
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}
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s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
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s3c_i2sv2_get_clock(cpu_dai));
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/* set the codec system clock for DAC and ADC */
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ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
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SND_SOC_CLOCK_IN);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER,
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div.clk_div - 1);
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if (ret < 0)
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return ret;
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return 0;
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}
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static const struct snd_soc_ops jive_ops = {
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.hw_params = jive_hw_params,
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};
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SND_SOC_DAILINK_DEFS(wm8750,
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DAILINK_COMP_ARRAY(COMP_CPU("s3c2412-i2s")),
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DAILINK_COMP_ARRAY(COMP_CODEC("wm8750.0-001a", "wm8750-hifi")),
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DAILINK_COMP_ARRAY(COMP_PLATFORM("s3c2412-i2s")));
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static struct snd_soc_dai_link jive_dai = {
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.name = "wm8750",
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.stream_name = "WM8750",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBS_CFS,
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.ops = &jive_ops,
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SND_SOC_DAILINK_REG(wm8750),
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};
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/* jive audio machine driver */
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static struct snd_soc_card snd_soc_machine_jive = {
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.name = "Jive",
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.owner = THIS_MODULE,
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.dai_link = &jive_dai,
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.num_links = 1,
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.dapm_widgets = wm8750_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(wm8750_dapm_widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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.fully_routed = true,
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};
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static struct platform_device *jive_snd_device;
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static int __init jive_init(void)
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{
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int ret;
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if (!machine_is_jive())
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return 0;
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printk("JIVE WM8750 Audio support\n");
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jive_snd_device = platform_device_alloc("soc-audio", -1);
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if (!jive_snd_device)
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return -ENOMEM;
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platform_set_drvdata(jive_snd_device, &snd_soc_machine_jive);
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ret = platform_device_add(jive_snd_device);
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if (ret)
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platform_device_put(jive_snd_device);
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return ret;
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}
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static void __exit jive_exit(void)
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{
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platform_device_unregister(jive_snd_device);
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}
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module_init(jive_init);
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module_exit(jive_exit);
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MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
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MODULE_DESCRIPTION("ALSA SoC Jive Audio support");
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MODULE_LICENSE("GPL");
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