/* sound/soc/s3c24xx/jive_wm8750.c * * Copyright 2007,2008 Simtec Electronics * * Based on sound/soc/pxa/spitz.c * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ #include #include #include #include #include #include #include #include #include #include #include #include "s3c24xx-pcm.h" #include "s3c2412-i2s.h" #include "../codecs/wm8750.h" static const struct snd_soc_dapm_route audio_map[] = { { "Headphone Jack", NULL, "LOUT1" }, { "Headphone Jack", NULL, "ROUT1" }, { "Internal Speaker", NULL, "LOUT2" }, { "Internal Speaker", NULL, "ROUT2" }, { "LINPUT1", NULL, "Line Input" }, { "RINPUT1", NULL, "Line Input" }, }; static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_HP("Headphone Jack", NULL), SND_SOC_DAPM_SPK("Internal Speaker", NULL), SND_SOC_DAPM_LINE("Line In", NULL), }; static int jive_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct s3c_i2sv2_rate_calc div; unsigned int clk = 0; int ret = 0; switch (params_rate(params)) { case 8000: case 16000: case 48000: case 96000: clk = 12288000; break; case 11025: case 22050: case 44100: clk = 11289600; break; } s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params), s3c2412_get_iisclk()); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk, SND_SOC_CLOCK_IN); if (ret < 0) return ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div); if (ret < 0) return ret; ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER, div.clk_div - 1); if (ret < 0) return ret; return 0; } static struct snd_soc_ops jive_ops = { .hw_params = jive_hw_params, }; static int jive_wm8750_init(struct snd_soc_codec *codec) { int err; /* These endpoints are not being used. */ snd_soc_dapm_nc_pin(codec, "LINPUT2"); snd_soc_dapm_nc_pin(codec, "RINPUT2"); snd_soc_dapm_nc_pin(codec, "LINPUT3"); snd_soc_dapm_nc_pin(codec, "RINPUT3"); snd_soc_dapm_nc_pin(codec, "OUT3"); snd_soc_dapm_nc_pin(codec, "MONO"); /* Add jive specific widgets */ err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets, ARRAY_SIZE(wm8750_dapm_widgets)); if (err) { printk(KERN_ERR "%s: failed to add widgets (%d)\n", __func__, err); return err; } snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_sync(codec); return 0; } static struct snd_soc_dai_link jive_dai = { .name = "wm8750", .stream_name = "WM8750", .cpu_dai = &s3c2412_i2s_dai, .codec_dai = &wm8750_dai, .init = jive_wm8750_init, .ops = &jive_ops, }; /* jive audio machine driver */ static struct snd_soc_card snd_soc_machine_jive = { .name = "Jive", .platform = &s3c24xx_soc_platform, .dai_link = &jive_dai, .num_links = 1, }; /* jive audio private data */ static struct wm8750_setup_data jive_wm8750_setup = { }; /* jive audio subsystem */ static struct snd_soc_device jive_snd_devdata = { .card = &snd_soc_machine_jive, .codec_dev = &soc_codec_dev_wm8750, .codec_data = &jive_wm8750_setup, }; static struct platform_device *jive_snd_device; static int __init jive_init(void) { int ret; if (!machine_is_jive()) return 0; printk("JIVE WM8750 Audio support\n"); jive_snd_device = platform_device_alloc("soc-audio", -1); if (!jive_snd_device) return -ENOMEM; platform_set_drvdata(jive_snd_device, &jive_snd_devdata); jive_snd_devdata.dev = &jive_snd_device->dev; ret = platform_device_add(jive_snd_device); if (ret) platform_device_put(jive_snd_device); return ret; } static void __exit jive_exit(void) { platform_device_unregister(jive_snd_device); } module_init(jive_init); module_exit(jive_exit); MODULE_AUTHOR("Ben Dooks "); MODULE_DESCRIPTION("ALSA SoC Jive Audio support"); MODULE_LICENSE("GPL");