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f4ee271709
Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 1 -> .idle_bias_on = 0 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> Signed-off-by: Mark Brown <broonie@kernel.org>
256 lines
5.8 KiB
C
256 lines
5.8 KiB
C
/*
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* Tobermory audio support
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*
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* Copyright 2011 Wolfson Microelectronics
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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 <sound/soc.h>
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#include <sound/soc-dapm.h>
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#include <sound/jack.h>
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#include <linux/gpio.h>
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#include <linux/module.h>
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#include "../codecs/wm8962.h"
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static int sample_rate = 44100;
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static int tobermory_set_bias_level(struct snd_soc_card *card,
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struct snd_soc_dapm_context *dapm,
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enum snd_soc_bias_level level)
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{
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struct snd_soc_pcm_runtime *rtd;
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struct snd_soc_dai *codec_dai;
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int ret;
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rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
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codec_dai = rtd->codec_dai;
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if (dapm->dev != codec_dai->dev)
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return 0;
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switch (level) {
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case SND_SOC_BIAS_PREPARE:
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if (dapm->bias_level == SND_SOC_BIAS_STANDBY) {
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ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
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WM8962_FLL_MCLK, 32768,
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sample_rate * 512);
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if (ret < 0)
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pr_err("Failed to start FLL: %d\n", ret);
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ret = snd_soc_dai_set_sysclk(codec_dai,
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WM8962_SYSCLK_FLL,
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sample_rate * 512,
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SND_SOC_CLOCK_IN);
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if (ret < 0) {
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pr_err("Failed to set SYSCLK: %d\n", ret);
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snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
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0, 0, 0);
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return ret;
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}
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}
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break;
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default:
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break;
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}
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return 0;
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}
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static int tobermory_set_bias_level_post(struct snd_soc_card *card,
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struct snd_soc_dapm_context *dapm,
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enum snd_soc_bias_level level)
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{
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struct snd_soc_pcm_runtime *rtd;
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struct snd_soc_dai *codec_dai;
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int ret;
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rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
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codec_dai = rtd->codec_dai;
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if (dapm->dev != codec_dai->dev)
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return 0;
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switch (level) {
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case SND_SOC_BIAS_STANDBY:
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ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
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32768, SND_SOC_CLOCK_IN);
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if (ret < 0) {
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pr_err("Failed to switch away from FLL: %d\n", ret);
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return ret;
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}
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ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL,
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0, 0, 0);
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if (ret < 0) {
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pr_err("Failed to stop FLL: %d\n", ret);
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return ret;
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}
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break;
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default:
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break;
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}
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dapm->bias_level = level;
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return 0;
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}
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static int tobermory_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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sample_rate = params_rate(params);
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return 0;
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}
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static struct snd_soc_ops tobermory_ops = {
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.hw_params = tobermory_hw_params,
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};
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static struct snd_soc_dai_link tobermory_dai[] = {
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{
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.name = "CPU",
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.stream_name = "CPU",
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.cpu_dai_name = "samsung-i2s.0",
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.codec_dai_name = "wm8962",
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.platform_name = "samsung-i2s.0",
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.codec_name = "wm8962.1-001a",
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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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.ops = &tobermory_ops,
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},
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};
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static const struct snd_kcontrol_new controls[] = {
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SOC_DAPM_PIN_SWITCH("Main Speaker"),
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SOC_DAPM_PIN_SWITCH("DMIC"),
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};
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static struct snd_soc_dapm_widget widgets[] = {
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SND_SOC_DAPM_HP("Headphone", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("DMIC", NULL),
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SND_SOC_DAPM_MIC("AMIC", NULL),
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SND_SOC_DAPM_SPK("Main Speaker", NULL),
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};
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static struct snd_soc_dapm_route audio_paths[] = {
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{ "Headphone", NULL, "HPOUTL" },
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{ "Headphone", NULL, "HPOUTR" },
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{ "Main Speaker", NULL, "SPKOUTL" },
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{ "Main Speaker", NULL, "SPKOUTR" },
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{ "Headset Mic", NULL, "MICBIAS" },
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{ "IN4L", NULL, "Headset Mic" },
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{ "IN4R", NULL, "Headset Mic" },
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{ "AMIC", NULL, "MICBIAS" },
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{ "IN1L", NULL, "AMIC" },
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{ "IN1R", NULL, "AMIC" },
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{ "DMIC", NULL, "MICBIAS" },
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{ "DMICDAT", NULL, "DMIC" },
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};
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static struct snd_soc_jack tobermory_headset;
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/* Headset jack detection DAPM pins */
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static struct snd_soc_jack_pin tobermory_headset_pins[] = {
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{
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.pin = "Headset Mic",
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.mask = SND_JACK_MICROPHONE,
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},
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{
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.pin = "Headphone",
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.mask = SND_JACK_MICROPHONE,
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},
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};
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static int tobermory_late_probe(struct snd_soc_card *card)
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{
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struct snd_soc_pcm_runtime *rtd;
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struct snd_soc_component *component;
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struct snd_soc_dai *codec_dai;
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int ret;
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rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name);
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component = rtd->codec_dai->component;
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codec_dai = rtd->codec_dai;
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ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK,
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32768, SND_SOC_CLOCK_IN);
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if (ret < 0)
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return ret;
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ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET |
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SND_JACK_BTN_0, &tobermory_headset,
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tobermory_headset_pins,
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ARRAY_SIZE(tobermory_headset_pins));
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if (ret)
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return ret;
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wm8962_mic_detect(component, &tobermory_headset);
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return 0;
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}
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static struct snd_soc_card tobermory = {
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.name = "Tobermory",
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.owner = THIS_MODULE,
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.dai_link = tobermory_dai,
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.num_links = ARRAY_SIZE(tobermory_dai),
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.set_bias_level = tobermory_set_bias_level,
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.set_bias_level_post = tobermory_set_bias_level_post,
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.controls = controls,
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.num_controls = ARRAY_SIZE(controls),
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.dapm_widgets = widgets,
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.num_dapm_widgets = ARRAY_SIZE(widgets),
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.dapm_routes = audio_paths,
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.num_dapm_routes = ARRAY_SIZE(audio_paths),
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.fully_routed = true,
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.late_probe = tobermory_late_probe,
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};
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static int tobermory_probe(struct platform_device *pdev)
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{
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struct snd_soc_card *card = &tobermory;
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int ret;
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card->dev = &pdev->dev;
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ret = devm_snd_soc_register_card(&pdev->dev, card);
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if (ret)
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dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
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ret);
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return ret;
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}
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static struct platform_driver tobermory_driver = {
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.driver = {
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.name = "tobermory",
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.pm = &snd_soc_pm_ops,
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},
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.probe = tobermory_probe,
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};
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module_platform_driver(tobermory_driver);
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MODULE_DESCRIPTION("Tobermory audio support");
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MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>");
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MODULE_LICENSE("GPL");
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MODULE_ALIAS("platform:tobermory");
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