263 строки
5.9 KiB
C
263 строки
5.9 KiB
C
/*
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* h1940-uda1380.c -- ALSA Soc Audio Layer
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*
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* Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
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* Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
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*
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* Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
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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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*/
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#include <linux/types.h>
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#include <linux/gpio.h>
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#include <linux/module.h>
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#include <sound/soc.h>
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#include <sound/jack.h>
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#include "regs-iis.h"
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#include <asm/mach-types.h>
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#include <mach/gpio-samsung.h>
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#include "s3c24xx-i2s.h"
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static unsigned int rates[] = {
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11025,
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22050,
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44100,
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};
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static struct snd_pcm_hw_constraint_list hw_rates = {
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.count = ARRAY_SIZE(rates),
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.list = rates,
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.mask = 0,
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};
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static struct snd_soc_jack hp_jack;
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static struct snd_soc_jack_pin hp_jack_pins[] = {
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{
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.pin = "Headphone Jack",
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.mask = SND_JACK_HEADPHONE,
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},
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{
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.pin = "Speaker",
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.mask = SND_JACK_HEADPHONE,
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.invert = 1,
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},
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};
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static struct snd_soc_jack_gpio hp_jack_gpios[] = {
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{
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.gpio = S3C2410_GPG(4),
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.name = "hp-gpio",
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.report = SND_JACK_HEADPHONE,
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.invert = 1,
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.debounce_time = 200,
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},
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};
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static int h1940_startup(struct snd_pcm_substream *substream)
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{
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struct snd_pcm_runtime *runtime = substream->runtime;
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return snd_pcm_hw_constraint_list(runtime, 0,
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SNDRV_PCM_HW_PARAM_RATE,
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&hw_rates);
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}
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static int h1940_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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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
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int div;
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int ret;
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unsigned int rate = params_rate(params);
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switch (rate) {
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case 11025:
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case 22050:
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case 44100:
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div = s3c24xx_i2s_get_clockrate() / (384 * rate);
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if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
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div++;
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break;
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default:
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dev_err(rtd->dev, "%s: rate %d is not supported\n",
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__func__, rate);
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return -EINVAL;
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}
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/* select clock source */
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ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
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SND_SOC_CLOCK_OUT);
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if (ret < 0)
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return ret;
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/* set MCLK division for sample rate */
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ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
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S3C2410_IISMOD_384FS);
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if (ret < 0)
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return ret;
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/* set BCLK division for sample rate */
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ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
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S3C2410_IISMOD_32FS);
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if (ret < 0)
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return ret;
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/* set prescaler division for sample rate */
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ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
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S3C24XX_PRESCALE(div, div));
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if (ret < 0)
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return ret;
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return 0;
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}
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static struct snd_soc_ops h1940_ops = {
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.startup = h1940_startup,
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.hw_params = h1940_hw_params,
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};
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static int h1940_spk_power(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *kcontrol, int event)
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{
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if (SND_SOC_DAPM_EVENT_ON(event))
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gpio_set_value(S3C_GPIO_END + 9, 1);
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else
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gpio_set_value(S3C_GPIO_END + 9, 0);
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return 0;
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}
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/* h1940 machine dapm widgets */
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static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone Jack", NULL),
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SND_SOC_DAPM_MIC("Mic Jack", NULL),
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SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
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};
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/* h1940 machine audio_map */
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static const struct snd_soc_dapm_route audio_map[] = {
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/* headphone connected to VOUTLHP, VOUTRHP */
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{"Headphone Jack", NULL, "VOUTLHP"},
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{"Headphone Jack", NULL, "VOUTRHP"},
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/* ext speaker connected to VOUTL, VOUTR */
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{"Speaker", NULL, "VOUTL"},
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{"Speaker", NULL, "VOUTR"},
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/* mic is connected to VINM */
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{"VINM", NULL, "Mic Jack"},
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};
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static struct platform_device *s3c24xx_snd_device;
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static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
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{
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snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE,
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&hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins));
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snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
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hp_jack_gpios);
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return 0;
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}
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static int h1940_uda1380_card_remove(struct snd_soc_card *card)
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{
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snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
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hp_jack_gpios);
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return 0;
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}
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/* s3c24xx digital audio interface glue - connects codec <--> CPU */
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static struct snd_soc_dai_link h1940_uda1380_dai[] = {
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{
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.name = "uda1380",
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.stream_name = "UDA1380 Duplex",
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.cpu_dai_name = "s3c24xx-iis",
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.codec_dai_name = "uda1380-hifi",
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.init = h1940_uda1380_init,
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.platform_name = "s3c24xx-iis",
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.codec_name = "uda1380-codec.0-001a",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
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SND_SOC_DAIFMT_CBS_CFS,
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.ops = &h1940_ops,
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},
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};
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static struct snd_soc_card h1940_asoc = {
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.name = "h1940",
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.owner = THIS_MODULE,
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.remove = h1940_uda1380_card_remove,
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.dai_link = h1940_uda1380_dai,
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.num_links = ARRAY_SIZE(h1940_uda1380_dai),
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.dapm_widgets = uda1380_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
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.dapm_routes = audio_map,
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.num_dapm_routes = ARRAY_SIZE(audio_map),
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};
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static int __init h1940_init(void)
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{
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int ret;
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if (!machine_is_h1940())
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return -ENODEV;
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/* configure some gpios */
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ret = gpio_request(S3C_GPIO_END + 9, "speaker-power");
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if (ret)
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goto err_out;
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ret = gpio_direction_output(S3C_GPIO_END + 9, 0);
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if (ret)
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goto err_gpio;
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s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
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if (!s3c24xx_snd_device) {
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ret = -ENOMEM;
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goto err_gpio;
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}
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platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
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ret = platform_device_add(s3c24xx_snd_device);
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if (ret)
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goto err_plat;
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return 0;
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err_plat:
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platform_device_put(s3c24xx_snd_device);
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err_gpio:
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gpio_free(S3C_GPIO_END + 9);
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err_out:
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return ret;
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}
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static void __exit h1940_exit(void)
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{
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platform_device_unregister(s3c24xx_snd_device);
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gpio_free(S3C_GPIO_END + 9);
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}
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module_init(h1940_init);
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module_exit(h1940_exit);
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/* Module information */
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MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
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MODULE_DESCRIPTION("ALSA SoC H1940");
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MODULE_LICENSE("GPL");
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