ARM: dmabounce: no need to check dev->bus type in needs_bounce function
[linux-2.6/linux-acpi-2.6/ibm-acpi-2.6.git] / sound / soc / samsung / h1940_uda1380.c
blob241f55d0066070b299159c0f92f6f8836fed4ad5
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/gpio.h>
18 #include <sound/soc.h>
19 #include <sound/jack.h>
21 #include <plat/regs-iis.h>
22 #include <mach/h1940-latch.h>
23 #include <asm/mach-types.h>
25 #include "s3c24xx-i2s.h"
27 static unsigned int rates[] = {
28 11025,
29 22050,
30 44100,
33 static struct snd_pcm_hw_constraint_list hw_rates = {
34 .count = ARRAY_SIZE(rates),
35 .list = rates,
36 .mask = 0,
39 static struct snd_soc_jack hp_jack;
41 static struct snd_soc_jack_pin hp_jack_pins[] = {
43 .pin = "Headphone Jack",
44 .mask = SND_JACK_HEADPHONE,
47 .pin = "Speaker",
48 .mask = SND_JACK_HEADPHONE,
49 .invert = 1,
53 static struct snd_soc_jack_gpio hp_jack_gpios[] = {
55 .gpio = S3C2410_GPG(4),
56 .name = "hp-gpio",
57 .report = SND_JACK_HEADPHONE,
58 .invert = 1,
59 .debounce_time = 200,
63 static int h1940_startup(struct snd_pcm_substream *substream)
65 struct snd_pcm_runtime *runtime = substream->runtime;
67 runtime->hw.rate_min = hw_rates.list[0];
68 runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
69 runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
71 return snd_pcm_hw_constraint_list(runtime, 0,
72 SNDRV_PCM_HW_PARAM_RATE,
73 &hw_rates);
76 static int h1940_hw_params(struct snd_pcm_substream *substream,
77 struct snd_pcm_hw_params *params)
79 struct snd_soc_pcm_runtime *rtd = substream->private_data;
80 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
81 struct snd_soc_dai *codec_dai = rtd->codec_dai;
82 int div;
83 int ret;
84 unsigned int rate = params_rate(params);
86 switch (rate) {
87 case 11025:
88 case 22050:
89 case 44100:
90 div = s3c24xx_i2s_get_clockrate() / (384 * rate);
91 if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
92 div++;
93 break;
94 default:
95 dev_err(&rtd->dev, "%s: rate %d is not supported\n",
96 __func__, rate);
97 return -EINVAL;
100 /* set codec DAI configuration */
101 ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
102 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
103 if (ret < 0)
104 return ret;
106 /* set cpu DAI configuration */
107 ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
108 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
109 if (ret < 0)
110 return ret;
112 /* select clock source */
113 ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
114 SND_SOC_CLOCK_OUT);
115 if (ret < 0)
116 return ret;
118 /* set MCLK division for sample rate */
119 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
120 S3C2410_IISMOD_384FS);
121 if (ret < 0)
122 return ret;
124 /* set BCLK division for sample rate */
125 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
126 S3C2410_IISMOD_32FS);
127 if (ret < 0)
128 return ret;
130 /* set prescaler division for sample rate */
131 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
132 S3C24XX_PRESCALE(div, div));
133 if (ret < 0)
134 return ret;
136 return 0;
139 static struct snd_soc_ops h1940_ops = {
140 .startup = h1940_startup,
141 .hw_params = h1940_hw_params,
144 static int h1940_spk_power(struct snd_soc_dapm_widget *w,
145 struct snd_kcontrol *kcontrol, int event)
147 if (SND_SOC_DAPM_EVENT_ON(event))
148 gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
149 else
150 gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
152 return 0;
155 /* h1940 machine dapm widgets */
156 static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
157 SND_SOC_DAPM_HP("Headphone Jack", NULL),
158 SND_SOC_DAPM_MIC("Mic Jack", NULL),
159 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
162 /* h1940 machine audio_map */
163 static const struct snd_soc_dapm_route audio_map[] = {
164 /* headphone connected to VOUTLHP, VOUTRHP */
165 {"Headphone Jack", NULL, "VOUTLHP"},
166 {"Headphone Jack", NULL, "VOUTRHP"},
168 /* ext speaker connected to VOUTL, VOUTR */
169 {"Speaker", NULL, "VOUTL"},
170 {"Speaker", NULL, "VOUTR"},
172 /* mic is connected to VINM */
173 {"VINM", NULL, "Mic Jack"},
176 static struct platform_device *s3c24xx_snd_device;
178 static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
180 struct snd_soc_codec *codec = rtd->codec;
181 struct snd_soc_dapm_context *dapm = &codec->dapm;
182 int err;
184 /* Add h1940 specific widgets */
185 err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
186 ARRAY_SIZE(uda1380_dapm_widgets));
187 if (err)
188 return err;
190 /* Set up h1940 specific audio path audio_mapnects */
191 err = snd_soc_dapm_add_routes(dapm, audio_map,
192 ARRAY_SIZE(audio_map));
193 if (err)
194 return err;
196 snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
197 snd_soc_dapm_enable_pin(dapm, "Speaker");
198 snd_soc_dapm_enable_pin(dapm, "Mic Jack");
200 snd_soc_dapm_sync(dapm);
202 snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
203 &hp_jack);
205 snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
206 hp_jack_pins);
208 snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
209 hp_jack_gpios);
211 return 0;
214 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
215 static struct snd_soc_dai_link h1940_uda1380_dai[] = {
217 .name = "uda1380",
218 .stream_name = "UDA1380 Duplex",
219 .cpu_dai_name = "s3c24xx-iis",
220 .codec_dai_name = "uda1380-hifi",
221 .init = h1940_uda1380_init,
222 .platform_name = "samsung-audio",
223 .codec_name = "uda1380-codec.0-001a",
224 .ops = &h1940_ops,
228 static struct snd_soc_card h1940_asoc = {
229 .name = "h1940",
230 .dai_link = h1940_uda1380_dai,
231 .num_links = ARRAY_SIZE(h1940_uda1380_dai),
234 static int __init h1940_init(void)
236 int ret;
238 if (!machine_is_h1940())
239 return -ENODEV;
241 /* configure some gpios */
242 ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
243 if (ret)
244 goto err_out;
246 ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
247 if (ret)
248 goto err_gpio;
250 s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
251 if (!s3c24xx_snd_device) {
252 ret = -ENOMEM;
253 goto err_gpio;
256 platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
257 ret = platform_device_add(s3c24xx_snd_device);
259 if (ret)
260 goto err_plat;
262 return 0;
264 err_plat:
265 platform_device_put(s3c24xx_snd_device);
266 err_gpio:
267 gpio_free(H1940_LATCH_AUDIO_POWER);
269 err_out:
270 return ret;
273 static void __exit h1940_exit(void)
275 platform_device_unregister(s3c24xx_snd_device);
276 snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
277 hp_jack_gpios);
278 gpio_free(H1940_LATCH_AUDIO_POWER);
281 module_init(h1940_init);
282 module_exit(h1940_exit);
284 /* Module information */
285 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
286 MODULE_DESCRIPTION("ALSA SoC H1940");
287 MODULE_LICENSE("GPL");