/* * h1940-uda1380.c -- ALSA Soc Audio Layer * * Copyright (c) 2010 Arnaud Patard * Copyright (c) 2010 Vasily Khoruzhick * * Based on version from Arnaud Patard * * 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 #include #include #include #include #include "regs-iis.h" #include #include #include "s3c24xx-i2s.h" static const unsigned int rates[] = { 11025, 22050, 44100, }; static const struct snd_pcm_hw_constraint_list hw_rates = { .count = ARRAY_SIZE(rates), .list = rates, }; 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; 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; 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; } /* 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(S3C_GPIO_END + 9, 1); else gpio_set_value(S3C_GPIO_END + 9, 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) { snd_soc_card_jack_new(rtd->card, "Headphone Jack", SND_JACK_HEADPHONE, &hp_jack, hp_jack_pins, ARRAY_SIZE(hp_jack_pins)); snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios), hp_jack_gpios); return 0; } static int h1940_uda1380_card_remove(struct snd_soc_card *card) { snd_soc_jack_free_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 = "s3c24xx-iis", .codec_name = "uda1380-codec.0-001a", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .ops = &h1940_ops, }, }; static struct snd_soc_card h1940_asoc = { .name = "h1940", .owner = THIS_MODULE, .remove = h1940_uda1380_card_remove, .dai_link = h1940_uda1380_dai, .num_links = ARRAY_SIZE(h1940_uda1380_dai), .dapm_widgets = uda1380_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets), .dapm_routes = audio_map, .num_dapm_routes = ARRAY_SIZE(audio_map), }; static int __init h1940_init(void) { int ret; if (!machine_is_h1940()) return -ENODEV; /* configure some gpios */ ret = gpio_request(S3C_GPIO_END + 9, "speaker-power"); if (ret) goto err_out; ret = gpio_direction_output(S3C_GPIO_END + 9, 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(S3C_GPIO_END + 9); err_out: return ret; } static void __exit h1940_exit(void) { platform_device_unregister(s3c24xx_snd_device); gpio_free(S3C_GPIO_END + 9); } module_init(h1940_init); module_exit(h1940_exit); /* Module information */ MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick"); MODULE_DESCRIPTION("ALSA SoC H1940"); MODULE_LICENSE("GPL"); mit-subject'>net: phy: Add LED mode driver for Microsemi PHYs.
LED Mode: Microsemi PHY support 2 LEDs (LED[0] and LED[1]) to display different status information that can be selected by setting LED mode. LED Mode parameter (vsc8531, led-0-mode) and (vsc8531, led-1-mode) get from Device Tree. Signed-off-by: Raju Lakkaraju <Raju.Lakkaraju@microsemi.com> Signed-off-by: David S. Miller <davem@davemloft.net>
Diffstat (limited to 'sound/soc/codecs/da7219.h')