diff options
author | Vasily Khoruzhick | 2010-12-09 21:17:56 +0200 |
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committer | Mark Brown | 2010-12-10 17:40:15 +0000 |
commit | 1957668be903b2c98e1dfb295510850531eea135 (patch) | |
tree | 133bd11e73d55cbbcd55429804b0df0374ccd31c /sound/soc/samsung/h1940_uda1380.c | |
parent | 154b26aa9ec4aaeaa7258fe85a036613f3713a59 (diff) |
ASoC: Add HP iPAQ H1940 support
Add glue driver to make s3c24xx-i2s and uda1380 produce some sound on
H1940.
Signed-off-by: Vasily Khoruzhick <anarsoul@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound/soc/samsung/h1940_uda1380.c')
-rw-r--r-- | sound/soc/samsung/h1940_uda1380.c | 296 |
1 files changed, 296 insertions, 0 deletions
diff --git a/sound/soc/samsung/h1940_uda1380.c b/sound/soc/samsung/h1940_uda1380.c new file mode 100644 index 000000000000..c45f7ce14d61 --- /dev/null +++ b/sound/soc/samsung/h1940_uda1380.c @@ -0,0 +1,296 @@ +/* + * 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/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> + +#include <sound/soc.h> +#include <sound/uda1380.h> +#include <sound/jack.h> + +#include <plat/regs-iis.h> + +#include <mach/h1940-latch.h> + +#include <asm/mach-types.h> + +#include "dma.h" +#include "s3c24xx-i2s.h" +#include "../codecs/uda1380.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_dapm_sync(dapm); + + 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"); |