/* * 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"); ms) the a5 restore process was changed and it was hard coded to put the user data segment address directly into a5. Which is ok for the common PIC compiled application case, but breaks the full relocation application code. We no longer use this type of signal handling mechanism and so we don't need to do anything special to save and restore a5 at all now. So remove the code that hard codes a5 to the address of the user data segment. Signed-off-by: Greg Ungerer <gerg@linux-m68k.org>
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