Author | Tokens | Token Proportion | Commits | Commit Proportion |
---|---|---|---|---|
Mark Brown | 869 | 89.59% | 14 | 66.67% |
Mengdong Lin | 75 | 7.73% | 1 | 4.76% |
Kuninori Morimoto | 7 | 0.72% | 1 | 4.76% |
Tushar Behera | 6 | 0.62% | 1 | 4.76% |
Axel Lin | 5 | 0.52% | 1 | 4.76% |
Lars-Peter Clausen | 4 | 0.41% | 1 | 4.76% |
Paul Gortmaker | 3 | 0.31% | 1 | 4.76% |
Padmavathi Venna | 1 | 0.10% | 1 | 4.76% |
Total | 970 | 21 |
/* * Tobermory audio support * * Copyright 2011 Wolfson Microelectronics * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. */ #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/jack.h> #include <linux/gpio.h> #include <linux/module.h> #include "../codecs/wm8962.h" static int sample_rate = 44100; static int tobermory_set_bias_level(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); codec_dai = rtd->codec_dai; if (dapm->dev != codec_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_PREPARE: if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, 32768, sample_rate * 512); if (ret < 0) pr_err("Failed to start FLL: %d\n", ret); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, sample_rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to set SYSCLK: %d\n", ret); snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); return ret; } } break; default: break; } return 0; } static int tobermory_set_bias_level_post(struct snd_soc_card *card, struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); codec_dai = rtd->codec_dai; if (dapm->dev != codec_dai->dev) return 0; switch (level) { case SND_SOC_BIAS_STANDBY: ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); if (ret < 0) { pr_err("Failed to switch away from FLL: %d\n", ret); return ret; } ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); if (ret < 0) { pr_err("Failed to stop FLL: %d\n", ret); return ret; } break; default: break; } dapm->bias_level = level; return 0; } static int tobermory_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { sample_rate = params_rate(params); return 0; } static struct snd_soc_ops tobermory_ops = { .hw_params = tobermory_hw_params, }; static struct snd_soc_dai_link tobermory_dai[] = { { .name = "CPU", .stream_name = "CPU", .cpu_dai_name = "samsung-i2s.0", .codec_dai_name = "wm8962", .platform_name = "samsung-i2s.0", .codec_name = "wm8962.1-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM, .ops = &tobermory_ops, }, }; static const struct snd_kcontrol_new controls[] = { SOC_DAPM_PIN_SWITCH("Main Speaker"), SOC_DAPM_PIN_SWITCH("DMIC"), }; static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_MIC("DMIC", NULL), SND_SOC_DAPM_MIC("AMIC", NULL), SND_SOC_DAPM_SPK("Main Speaker", NULL), }; static struct snd_soc_dapm_route audio_paths[] = { { "Headphone", NULL, "HPOUTL" }, { "Headphone", NULL, "HPOUTR" }, { "Main Speaker", NULL, "SPKOUTL" }, { "Main Speaker", NULL, "SPKOUTR" }, { "Headset Mic", NULL, "MICBIAS" }, { "IN4L", NULL, "Headset Mic" }, { "IN4R", NULL, "Headset Mic" }, { "AMIC", NULL, "MICBIAS" }, { "IN1L", NULL, "AMIC" }, { "IN1R", NULL, "AMIC" }, { "DMIC", NULL, "MICBIAS" }, { "DMICDAT", NULL, "DMIC" }, }; static struct snd_soc_jack tobermory_headset; /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin tobermory_headset_pins[] = { { .pin = "Headset Mic", .mask = SND_JACK_MICROPHONE, }, { .pin = "Headphone", .mask = SND_JACK_MICROPHONE, }, }; static int tobermory_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_component *component; struct snd_soc_dai *codec_dai; int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); component = rtd->codec_dai->component; codec_dai = rtd->codec_dai; ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); if (ret < 0) return ret; ret = snd_soc_card_jack_new(card, "Headset", SND_JACK_HEADSET | SND_JACK_BTN_0, &tobermory_headset, tobermory_headset_pins, ARRAY_SIZE(tobermory_headset_pins)); if (ret) return ret; wm8962_mic_detect(component, &tobermory_headset); return 0; } static struct snd_soc_card tobermory = { .name = "Tobermory", .owner = THIS_MODULE, .dai_link = tobermory_dai, .num_links = ARRAY_SIZE(tobermory_dai), .set_bias_level = tobermory_set_bias_level, .set_bias_level_post = tobermory_set_bias_level_post, .controls = controls, .num_controls = ARRAY_SIZE(controls), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, .num_dapm_routes = ARRAY_SIZE(audio_paths), .fully_routed = true, .late_probe = tobermory_late_probe, }; static int tobermory_probe(struct platform_device *pdev) { struct snd_soc_card *card = &tobermory; int ret; card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } static struct platform_driver tobermory_driver = { .driver = { .name = "tobermory", .pm = &snd_soc_pm_ops, }, .probe = tobermory_probe, }; module_platform_driver(tobermory_driver); MODULE_DESCRIPTION("Tobermory audio support"); MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:tobermory");
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