| /* |
| * h1940-uda1380.c -- ALSA Soc Audio Layer |
| * |
| * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org> |
| * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com> |
| * |
| * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org> |
| * |
| * 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 <linux/types.h> |
| #include <linux/gpio.h> |
| |
| #include <sound/soc.h> |
| #include <sound/jack.h> |
| |
| #include <plat/regs-iis.h> |
| #include <mach/h1940-latch.h> |
| #include <asm/mach-types.h> |
| |
| #include "s3c24xx-i2s.h" |
| |
| static unsigned int rates[] = { |
| 11025, |
| 22050, |
| 44100, |
| }; |
| |
| static struct snd_pcm_hw_constraint_list hw_rates = { |
| .count = ARRAY_SIZE(rates), |
| .list = rates, |
| .mask = 0, |
| }; |
| |
| static struct snd_soc_jack hp_jack; |
| |
| static struct snd_soc_jack_pin hp_jack_pins[] = { |
| { |
| .pin = "Headphone Jack", |
| .mask = SND_JACK_HEADPHONE, |
| }, |
| { |
| .pin = "Speaker", |
| .mask = SND_JACK_HEADPHONE, |
| .invert = 1, |
| }, |
| }; |
| |
| static struct snd_soc_jack_gpio hp_jack_gpios[] = { |
| { |
| .gpio = S3C2410_GPG(4), |
| .name = "hp-gpio", |
| .report = SND_JACK_HEADPHONE, |
| .invert = 1, |
| .debounce_time = 200, |
| }, |
| }; |
| |
| static int h1940_startup(struct snd_pcm_substream *substream) |
| { |
| struct snd_pcm_runtime *runtime = substream->runtime; |
| |
| runtime->hw.rate_min = hw_rates.list[0]; |
| runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1]; |
| runtime->hw.rates = SNDRV_PCM_RATE_KNOT; |
| |
| return snd_pcm_hw_constraint_list(runtime, 0, |
| SNDRV_PCM_HW_PARAM_RATE, |
| &hw_rates); |
| } |
| |
| static int h1940_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 *cpu_dai = rtd->cpu_dai; |
| struct snd_soc_dai *codec_dai = rtd->codec_dai; |
| int div; |
| int ret; |
| unsigned int rate = params_rate(params); |
| |
| switch (rate) { |
| case 11025: |
| case 22050: |
| case 44100: |
| div = s3c24xx_i2s_get_clockrate() / (384 * rate); |
| if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate)) |
| div++; |
| break; |
| default: |
| dev_err(&rtd->dev, "%s: rate %d is not supported\n", |
| __func__, rate); |
| return -EINVAL; |
| } |
| |
| /* 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; |
| |
| /* select clock source */ |
| ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate, |
| SND_SOC_CLOCK_OUT); |
| if (ret < 0) |
| return ret; |
| |
| /* set MCLK division for sample rate */ |
| ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, |
| S3C2410_IISMOD_384FS); |
| if (ret < 0) |
| return ret; |
| |
| /* set BCLK division for sample rate */ |
| ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK, |
| S3C2410_IISMOD_32FS); |
| if (ret < 0) |
| return ret; |
| |
| /* set prescaler division for sample rate */ |
| ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, |
| S3C24XX_PRESCALE(div, div)); |
| if (ret < 0) |
| return ret; |
| |
| return 0; |
| } |
| |
| static struct snd_soc_ops h1940_ops = { |
| .startup = h1940_startup, |
| .hw_params = h1940_hw_params, |
| }; |
| |
| static int h1940_spk_power(struct snd_soc_dapm_widget *w, |
| struct snd_kcontrol *kcontrol, int event) |
| { |
| if (SND_SOC_DAPM_EVENT_ON(event)) |
| gpio_set_value(H1940_LATCH_AUDIO_POWER, 1); |
| else |
| gpio_set_value(H1940_LATCH_AUDIO_POWER, 0); |
| |
| return 0; |
| } |
| |
| /* h1940 machine dapm widgets */ |
| static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = { |
| SND_SOC_DAPM_HP("Headphone Jack", NULL), |
| SND_SOC_DAPM_MIC("Mic Jack", NULL), |
| SND_SOC_DAPM_SPK("Speaker", h1940_spk_power), |
| }; |
| |
| /* h1940 machine audio_map */ |
| static const struct snd_soc_dapm_route audio_map[] = { |
| /* headphone connected to VOUTLHP, VOUTRHP */ |
| {"Headphone Jack", NULL, "VOUTLHP"}, |
| {"Headphone Jack", NULL, "VOUTRHP"}, |
| |
| /* ext speaker connected to VOUTL, VOUTR */ |
| {"Speaker", NULL, "VOUTL"}, |
| {"Speaker", NULL, "VOUTR"}, |
| |
| /* mic is connected to VINM */ |
| {"VINM", NULL, "Mic Jack"}, |
| }; |
| |
| static struct platform_device *s3c24xx_snd_device; |
| |
| static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd) |
| { |
| struct snd_soc_codec *codec = rtd->codec; |
| struct snd_soc_dapm_context *dapm = &codec->dapm; |
| int err; |
| |
| /* Add h1940 specific widgets */ |
| err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets, |
| ARRAY_SIZE(uda1380_dapm_widgets)); |
| if (err) |
| return err; |
| |
| /* Set up h1940 specific audio path audio_mapnects */ |
| err = snd_soc_dapm_add_routes(dapm, audio_map, |
| ARRAY_SIZE(audio_map)); |
| if (err) |
| return err; |
| |
| snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); |
| snd_soc_dapm_enable_pin(dapm, "Speaker"); |
| snd_soc_dapm_enable_pin(dapm, "Mic Jack"); |
| |
| snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE, |
| &hp_jack); |
| |
| snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins), |
| hp_jack_pins); |
| |
| snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), |
| hp_jack_gpios); |
| |
| return 0; |
| } |
| |
| /* s3c24xx digital audio interface glue - connects codec <--> CPU */ |
| static struct snd_soc_dai_link h1940_uda1380_dai[] = { |
| { |
| .name = "uda1380", |
| .stream_name = "UDA1380 Duplex", |
| .cpu_dai_name = "s3c24xx-iis", |
| .codec_dai_name = "uda1380-hifi", |
| .init = h1940_uda1380_init, |
| .platform_name = "samsung-audio", |
| .codec_name = "uda1380-codec.0-001a", |
| .ops = &h1940_ops, |
| }, |
| }; |
| |
| static struct snd_soc_card h1940_asoc = { |
| .name = "h1940", |
| .dai_link = h1940_uda1380_dai, |
| .num_links = ARRAY_SIZE(h1940_uda1380_dai), |
| }; |
| |
| static int __init h1940_init(void) |
| { |
| int ret; |
| |
| if (!machine_is_h1940()) |
| return -ENODEV; |
| |
| /* configure some gpios */ |
| ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power"); |
| if (ret) |
| goto err_out; |
| |
| ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0); |
| if (ret) |
| goto err_gpio; |
| |
| s3c24xx_snd_device = platform_device_alloc("soc-audio", -1); |
| if (!s3c24xx_snd_device) { |
| ret = -ENOMEM; |
| goto err_gpio; |
| } |
| |
| platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc); |
| ret = platform_device_add(s3c24xx_snd_device); |
| |
| if (ret) |
| goto err_plat; |
| |
| return 0; |
| |
| err_plat: |
| platform_device_put(s3c24xx_snd_device); |
| err_gpio: |
| gpio_free(H1940_LATCH_AUDIO_POWER); |
| |
| err_out: |
| return ret; |
| } |
| |
| static void __exit h1940_exit(void) |
| { |
| platform_device_unregister(s3c24xx_snd_device); |
| snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), |
| hp_jack_gpios); |
| gpio_free(H1940_LATCH_AUDIO_POWER); |
| } |
| |
| module_init(h1940_init); |
| module_exit(h1940_exit); |
| |
| /* Module information */ |
| MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); |
| MODULE_DESCRIPTION("ALSA SoC H1940"); |
| MODULE_LICENSE("GPL"); |