19aed2d6cd
snd_soc_card_jack_new() allowed to create jack kcontrol without pins, but did not create kcontrols. The jack would not have kcontrols if pins were not going to be added. This renames the old snd_soc_card_jack_new() to snd_soc_card_jack_new_pins() for use when pins are provided or will be added later. The new snd_soc_card_jack_new() appropriately creates a jack for use without pins and adds a kcontrol. Signed-off-by: Akihiko Odaki <akihiko.odaki@gmail.com> Link: https://lore.kernel.org/r/20220408041114.6024-1-akihiko.odaki@gmail.com Signed-off-by: Mark Brown <broonie@kernel.org>
252 lines
5.7 KiB
C
252 lines
5.7 KiB
C
// SPDX-License-Identifier: GPL-2.0+
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//
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// Tobermory audio support
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//
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// Copyright 2011 Wolfson Microelectronics
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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]);
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codec_dai = asoc_rtd_to_codec(rtd, 0);
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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]);
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codec_dai = asoc_rtd_to_codec(rtd, 0);
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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 const struct snd_soc_ops tobermory_ops = {
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.hw_params = tobermory_hw_params,
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};
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SND_SOC_DAILINK_DEFS(cpu,
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DAILINK_COMP_ARRAY(COMP_CPU("samsung-i2s.0")),
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DAILINK_COMP_ARRAY(COMP_CODEC("wm8962.1-001a", "wm8962")),
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DAILINK_COMP_ARRAY(COMP_PLATFORM("samsung-i2s.0")));
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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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.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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SND_SOC_DAILINK_REG(cpu),
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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 const 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 const 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]);
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component = asoc_rtd_to_codec(rtd, 0)->component;
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codec_dai = asoc_rtd_to_codec(rtd, 0);
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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_pins(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_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n");
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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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