Merge branch 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab...
[deliverable/linux.git] / sound / soc / samsung / h1940_uda1380.c
1 /*
2 * h1940-uda1380.c -- ALSA Soc Audio Layer
3 *
4 * Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
5 * Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
6 *
7 * Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
8 *
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.
13 *
14 */
15
16 #include <linux/types.h>
17 #include <linux/gpio.h>
18 #include <linux/module.h>
19
20 #include <sound/soc.h>
21 #include <sound/jack.h>
22
23 #include "regs-iis.h"
24 #include <asm/mach-types.h>
25
26 #include <mach/gpio-samsung.h>
27 #include "s3c24xx-i2s.h"
28
29 static unsigned int rates[] = {
30 11025,
31 22050,
32 44100,
33 };
34
35 static struct snd_pcm_hw_constraint_list hw_rates = {
36 .count = ARRAY_SIZE(rates),
37 .list = rates,
38 .mask = 0,
39 };
40
41 static struct snd_soc_jack hp_jack;
42
43 static struct snd_soc_jack_pin hp_jack_pins[] = {
44 {
45 .pin = "Headphone Jack",
46 .mask = SND_JACK_HEADPHONE,
47 },
48 {
49 .pin = "Speaker",
50 .mask = SND_JACK_HEADPHONE,
51 .invert = 1,
52 },
53 };
54
55 static struct snd_soc_jack_gpio hp_jack_gpios[] = {
56 {
57 .gpio = S3C2410_GPG(4),
58 .name = "hp-gpio",
59 .report = SND_JACK_HEADPHONE,
60 .invert = 1,
61 .debounce_time = 200,
62 },
63 };
64
65 static int h1940_startup(struct snd_pcm_substream *substream)
66 {
67 struct snd_pcm_runtime *runtime = substream->runtime;
68
69 return snd_pcm_hw_constraint_list(runtime, 0,
70 SNDRV_PCM_HW_PARAM_RATE,
71 &hw_rates);
72 }
73
74 static int h1940_hw_params(struct snd_pcm_substream *substream,
75 struct snd_pcm_hw_params *params)
76 {
77 struct snd_soc_pcm_runtime *rtd = substream->private_data;
78 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
79 struct snd_soc_dai *codec_dai = rtd->codec_dai;
80 int div;
81 int ret;
82 unsigned int rate = params_rate(params);
83
84 switch (rate) {
85 case 11025:
86 case 22050:
87 case 44100:
88 div = s3c24xx_i2s_get_clockrate() / (384 * rate);
89 if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
90 div++;
91 break;
92 default:
93 dev_err(rtd->dev, "%s: rate %d is not supported\n",
94 __func__, rate);
95 return -EINVAL;
96 }
97
98 /* set codec DAI configuration */
99 ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
100 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
101 if (ret < 0)
102 return ret;
103
104 /* set cpu DAI configuration */
105 ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
106 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
107 if (ret < 0)
108 return ret;
109
110 /* select clock source */
111 ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
112 SND_SOC_CLOCK_OUT);
113 if (ret < 0)
114 return ret;
115
116 /* set MCLK division for sample rate */
117 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
118 S3C2410_IISMOD_384FS);
119 if (ret < 0)
120 return ret;
121
122 /* set BCLK division for sample rate */
123 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
124 S3C2410_IISMOD_32FS);
125 if (ret < 0)
126 return ret;
127
128 /* set prescaler division for sample rate */
129 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
130 S3C24XX_PRESCALE(div, div));
131 if (ret < 0)
132 return ret;
133
134 return 0;
135 }
136
137 static struct snd_soc_ops h1940_ops = {
138 .startup = h1940_startup,
139 .hw_params = h1940_hw_params,
140 };
141
142 static int h1940_spk_power(struct snd_soc_dapm_widget *w,
143 struct snd_kcontrol *kcontrol, int event)
144 {
145 if (SND_SOC_DAPM_EVENT_ON(event))
146 gpio_set_value(S3C_GPIO_END + 9, 1);
147 else
148 gpio_set_value(S3C_GPIO_END + 9, 0);
149
150 return 0;
151 }
152
153 /* h1940 machine dapm widgets */
154 static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
155 SND_SOC_DAPM_HP("Headphone Jack", NULL),
156 SND_SOC_DAPM_MIC("Mic Jack", NULL),
157 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
158 };
159
160 /* h1940 machine audio_map */
161 static const struct snd_soc_dapm_route audio_map[] = {
162 /* headphone connected to VOUTLHP, VOUTRHP */
163 {"Headphone Jack", NULL, "VOUTLHP"},
164 {"Headphone Jack", NULL, "VOUTRHP"},
165
166 /* ext speaker connected to VOUTL, VOUTR */
167 {"Speaker", NULL, "VOUTL"},
168 {"Speaker", NULL, "VOUTR"},
169
170 /* mic is connected to VINM */
171 {"VINM", NULL, "Mic Jack"},
172 };
173
174 static struct platform_device *s3c24xx_snd_device;
175
176 static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
177 {
178 struct snd_soc_codec *codec = rtd->codec;
179 struct snd_soc_dapm_context *dapm = &codec->dapm;
180
181 snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
182 snd_soc_dapm_enable_pin(dapm, "Speaker");
183 snd_soc_dapm_enable_pin(dapm, "Mic Jack");
184
185 snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
186 &hp_jack);
187
188 snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
189 hp_jack_pins);
190
191 snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
192 hp_jack_gpios);
193
194 return 0;
195 }
196
197 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
198 static struct snd_soc_dai_link h1940_uda1380_dai[] = {
199 {
200 .name = "uda1380",
201 .stream_name = "UDA1380 Duplex",
202 .cpu_dai_name = "s3c24xx-iis",
203 .codec_dai_name = "uda1380-hifi",
204 .init = h1940_uda1380_init,
205 .platform_name = "s3c24xx-iis",
206 .codec_name = "uda1380-codec.0-001a",
207 .ops = &h1940_ops,
208 },
209 };
210
211 static struct snd_soc_card h1940_asoc = {
212 .name = "h1940",
213 .owner = THIS_MODULE,
214 .dai_link = h1940_uda1380_dai,
215 .num_links = ARRAY_SIZE(h1940_uda1380_dai),
216
217 .dapm_widgets = uda1380_dapm_widgets,
218 .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
219 .dapm_routes = audio_map,
220 .num_dapm_routes = ARRAY_SIZE(audio_map),
221 };
222
223 static int __init h1940_init(void)
224 {
225 int ret;
226
227 if (!machine_is_h1940())
228 return -ENODEV;
229
230 /* configure some gpios */
231 ret = gpio_request(S3C_GPIO_END + 9, "speaker-power");
232 if (ret)
233 goto err_out;
234
235 ret = gpio_direction_output(S3C_GPIO_END + 9, 0);
236 if (ret)
237 goto err_gpio;
238
239 s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
240 if (!s3c24xx_snd_device) {
241 ret = -ENOMEM;
242 goto err_gpio;
243 }
244
245 platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
246 ret = platform_device_add(s3c24xx_snd_device);
247
248 if (ret)
249 goto err_plat;
250
251 return 0;
252
253 err_plat:
254 platform_device_put(s3c24xx_snd_device);
255 err_gpio:
256 gpio_free(S3C_GPIO_END + 9);
257
258 err_out:
259 return ret;
260 }
261
262 static void __exit h1940_exit(void)
263 {
264 platform_device_unregister(s3c24xx_snd_device);
265 snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
266 hp_jack_gpios);
267 gpio_free(S3C_GPIO_END + 9);
268 }
269
270 module_init(h1940_init);
271 module_exit(h1940_exit);
272
273 /* Module information */
274 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
275 MODULE_DESCRIPTION("ALSA SoC H1940");
276 MODULE_LICENSE("GPL");
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