Revert "microblaze_mmu_v2: Update signal returning address"
[linux/fpc-iii.git] / sound / soc / samsung / h1940_uda1380.c
blob6e3257717c54b24c49fa840e8696ccd169612d86
1 /*
2 * h1940-uda1380.c -- ALSA Soc Audio Layer
4 * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
5 * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
7 * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
9 * This program is free software; you can redistribute it and/or modify it
10 * under the terms of the GNU General Public License as published by the
11 * Free Software Foundation; either version 2 of the License, or (at your
12 * option) any later version.
16 #include <linux/types.h>
17 #include <linux/gpio.h>
18 #include <linux/module.h>
20 #include <sound/soc.h>
21 #include <sound/jack.h>
23 #include <plat/regs-iis.h>
24 #include <mach/h1940-latch.h>
25 #include <asm/mach-types.h>
27 #include "s3c24xx-i2s.h"
29 static unsigned int rates[] = {
30 11025,
31 22050,
32 44100,
35 static struct snd_pcm_hw_constraint_list hw_rates = {
36 .count = ARRAY_SIZE(rates),
37 .list = rates,
38 .mask = 0,
41 static struct snd_soc_jack hp_jack;
43 static struct snd_soc_jack_pin hp_jack_pins[] = {
45 .pin = "Headphone Jack",
46 .mask = SND_JACK_HEADPHONE,
49 .pin = "Speaker",
50 .mask = SND_JACK_HEADPHONE,
51 .invert = 1,
55 static struct snd_soc_jack_gpio hp_jack_gpios[] = {
57 .gpio = S3C2410_GPG(4),
58 .name = "hp-gpio",
59 .report = SND_JACK_HEADPHONE,
60 .invert = 1,
61 .debounce_time = 200,
65 static int h1940_startup(struct snd_pcm_substream *substream)
67 struct snd_pcm_runtime *runtime = substream->runtime;
69 runtime->hw.rate_min = hw_rates.list[0];
70 runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
71 runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
73 return snd_pcm_hw_constraint_list(runtime, 0,
74 SNDRV_PCM_HW_PARAM_RATE,
75 &hw_rates);
78 static int h1940_hw_params(struct snd_pcm_substream *substream,
79 struct snd_pcm_hw_params *params)
81 struct snd_soc_pcm_runtime *rtd = substream->private_data;
82 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
83 struct snd_soc_dai *codec_dai = rtd->codec_dai;
84 int div;
85 int ret;
86 unsigned int rate = params_rate(params);
88 switch (rate) {
89 case 11025:
90 case 22050:
91 case 44100:
92 div = s3c24xx_i2s_get_clockrate() / (384 * rate);
93 if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
94 div++;
95 break;
96 default:
97 dev_err(&rtd->dev, "%s: rate %d is not supported\n",
98 __func__, rate);
99 return -EINVAL;
102 /* set codec DAI configuration */
103 ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
104 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
105 if (ret < 0)
106 return ret;
108 /* set cpu DAI configuration */
109 ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
110 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
111 if (ret < 0)
112 return ret;
114 /* select clock source */
115 ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
116 SND_SOC_CLOCK_OUT);
117 if (ret < 0)
118 return ret;
120 /* set MCLK division for sample rate */
121 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
122 S3C2410_IISMOD_384FS);
123 if (ret < 0)
124 return ret;
126 /* set BCLK division for sample rate */
127 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
128 S3C2410_IISMOD_32FS);
129 if (ret < 0)
130 return ret;
132 /* set prescaler division for sample rate */
133 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
134 S3C24XX_PRESCALE(div, div));
135 if (ret < 0)
136 return ret;
138 return 0;
141 static struct snd_soc_ops h1940_ops = {
142 .startup = h1940_startup,
143 .hw_params = h1940_hw_params,
146 static int h1940_spk_power(struct snd_soc_dapm_widget *w,
147 struct snd_kcontrol *kcontrol, int event)
149 if (SND_SOC_DAPM_EVENT_ON(event))
150 gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
151 else
152 gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
154 return 0;
157 /* h1940 machine dapm widgets */
158 static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
159 SND_SOC_DAPM_HP("Headphone Jack", NULL),
160 SND_SOC_DAPM_MIC("Mic Jack", NULL),
161 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
164 /* h1940 machine audio_map */
165 static const struct snd_soc_dapm_route audio_map[] = {
166 /* headphone connected to VOUTLHP, VOUTRHP */
167 {"Headphone Jack", NULL, "VOUTLHP"},
168 {"Headphone Jack", NULL, "VOUTRHP"},
170 /* ext speaker connected to VOUTL, VOUTR */
171 {"Speaker", NULL, "VOUTL"},
172 {"Speaker", NULL, "VOUTR"},
174 /* mic is connected to VINM */
175 {"VINM", NULL, "Mic Jack"},
178 static struct platform_device *s3c24xx_snd_device;
180 static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
182 struct snd_soc_codec *codec = rtd->codec;
183 struct snd_soc_dapm_context *dapm = &codec->dapm;
184 int err;
186 snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
187 snd_soc_dapm_enable_pin(dapm, "Speaker");
188 snd_soc_dapm_enable_pin(dapm, "Mic Jack");
190 snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
191 &hp_jack);
193 snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
194 hp_jack_pins);
196 snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
197 hp_jack_gpios);
199 return 0;
202 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
203 static struct snd_soc_dai_link h1940_uda1380_dai[] = {
205 .name = "uda1380",
206 .stream_name = "UDA1380 Duplex",
207 .cpu_dai_name = "s3c24xx-iis",
208 .codec_dai_name = "uda1380-hifi",
209 .init = h1940_uda1380_init,
210 .platform_name = "samsung-audio",
211 .codec_name = "uda1380-codec.0-001a",
212 .ops = &h1940_ops,
216 static struct snd_soc_card h1940_asoc = {
217 .name = "h1940",
218 .owner = THIS_MODULE,
219 .dai_link = h1940_uda1380_dai,
220 .num_links = ARRAY_SIZE(h1940_uda1380_dai),
222 .dapm_widgets = uda1380_dapm_widgets,
223 .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
224 .dapm_routes = audio_map,
225 .num_dapm_routes = ARRAY_SIZE(audio_map),
228 static int __init h1940_init(void)
230 int ret;
232 if (!machine_is_h1940())
233 return -ENODEV;
235 /* configure some gpios */
236 ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
237 if (ret)
238 goto err_out;
240 ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
241 if (ret)
242 goto err_gpio;
244 s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
245 if (!s3c24xx_snd_device) {
246 ret = -ENOMEM;
247 goto err_gpio;
250 platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
251 ret = platform_device_add(s3c24xx_snd_device);
253 if (ret)
254 goto err_plat;
256 return 0;
258 err_plat:
259 platform_device_put(s3c24xx_snd_device);
260 err_gpio:
261 gpio_free(H1940_LATCH_AUDIO_POWER);
263 err_out:
264 return ret;
267 static void __exit h1940_exit(void)
269 platform_device_unregister(s3c24xx_snd_device);
270 snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
271 hp_jack_gpios);
272 gpio_free(H1940_LATCH_AUDIO_POWER);
275 module_init(h1940_init);
276 module_exit(h1940_exit);
278 /* Module information */
279 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
280 MODULE_DESCRIPTION("ALSA SoC H1940");
281 MODULE_LICENSE("GPL");