ASoC: Convert H1940 to table based init
[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
19 #include <sound/soc.h>
20 #include <sound/jack.h>
21
22 #include <plat/regs-iis.h>
23 #include <mach/h1940-latch.h>
24 #include <asm/mach-types.h>
25
26 #include "s3c24xx-i2s.h"
27
28 static unsigned int rates[] = {
29 11025,
30 22050,
31 44100,
32 };
33
34 static struct snd_pcm_hw_constraint_list hw_rates = {
35 .count = ARRAY_SIZE(rates),
36 .list = rates,
37 .mask = 0,
38 };
39
40 static struct snd_soc_jack hp_jack;
41
42 static struct snd_soc_jack_pin hp_jack_pins[] = {
43 {
44 .pin = "Headphone Jack",
45 .mask = SND_JACK_HEADPHONE,
46 },
47 {
48 .pin = "Speaker",
49 .mask = SND_JACK_HEADPHONE,
50 .invert = 1,
51 },
52 };
53
54 static struct snd_soc_jack_gpio hp_jack_gpios[] = {
55 {
56 .gpio = S3C2410_GPG(4),
57 .name = "hp-gpio",
58 .report = SND_JACK_HEADPHONE,
59 .invert = 1,
60 .debounce_time = 200,
61 },
62 };
63
64 static int h1940_startup(struct snd_pcm_substream *substream)
65 {
66 struct snd_pcm_runtime *runtime = substream->runtime;
67
68 runtime->hw.rate_min = hw_rates.list[0];
69 runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
70 runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
71
72 return snd_pcm_hw_constraint_list(runtime, 0,
73 SNDRV_PCM_HW_PARAM_RATE,
74 &hw_rates);
75 }
76
77 static int h1940_hw_params(struct snd_pcm_substream *substream,
78 struct snd_pcm_hw_params *params)
79 {
80 struct snd_soc_pcm_runtime *rtd = substream->private_data;
81 struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
82 struct snd_soc_dai *codec_dai = rtd->codec_dai;
83 int div;
84 int ret;
85 unsigned int rate = params_rate(params);
86
87 switch (rate) {
88 case 11025:
89 case 22050:
90 case 44100:
91 div = s3c24xx_i2s_get_clockrate() / (384 * rate);
92 if (s3c24xx_i2s_get_clockrate() % (384 * rate) > (192 * rate))
93 div++;
94 break;
95 default:
96 dev_err(&rtd->dev, "%s: rate %d is not supported\n",
97 __func__, rate);
98 return -EINVAL;
99 }
100
101 /* set codec DAI configuration */
102 ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
103 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
104 if (ret < 0)
105 return ret;
106
107 /* set cpu DAI configuration */
108 ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
109 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
110 if (ret < 0)
111 return ret;
112
113 /* select clock source */
114 ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_PCLK, rate,
115 SND_SOC_CLOCK_OUT);
116 if (ret < 0)
117 return ret;
118
119 /* set MCLK division for sample rate */
120 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
121 S3C2410_IISMOD_384FS);
122 if (ret < 0)
123 return ret;
124
125 /* set BCLK division for sample rate */
126 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_BCLK,
127 S3C2410_IISMOD_32FS);
128 if (ret < 0)
129 return ret;
130
131 /* set prescaler division for sample rate */
132 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
133 S3C24XX_PRESCALE(div, div));
134 if (ret < 0)
135 return ret;
136
137 return 0;
138 }
139
140 static struct snd_soc_ops h1940_ops = {
141 .startup = h1940_startup,
142 .hw_params = h1940_hw_params,
143 };
144
145 static int h1940_spk_power(struct snd_soc_dapm_widget *w,
146 struct snd_kcontrol *kcontrol, int event)
147 {
148 if (SND_SOC_DAPM_EVENT_ON(event))
149 gpio_set_value(H1940_LATCH_AUDIO_POWER, 1);
150 else
151 gpio_set_value(H1940_LATCH_AUDIO_POWER, 0);
152
153 return 0;
154 }
155
156 /* h1940 machine dapm widgets */
157 static const struct snd_soc_dapm_widget uda1380_dapm_widgets[] = {
158 SND_SOC_DAPM_HP("Headphone Jack", NULL),
159 SND_SOC_DAPM_MIC("Mic Jack", NULL),
160 SND_SOC_DAPM_SPK("Speaker", h1940_spk_power),
161 };
162
163 /* h1940 machine audio_map */
164 static const struct snd_soc_dapm_route audio_map[] = {
165 /* headphone connected to VOUTLHP, VOUTRHP */
166 {"Headphone Jack", NULL, "VOUTLHP"},
167 {"Headphone Jack", NULL, "VOUTRHP"},
168
169 /* ext speaker connected to VOUTL, VOUTR */
170 {"Speaker", NULL, "VOUTL"},
171 {"Speaker", NULL, "VOUTR"},
172
173 /* mic is connected to VINM */
174 {"VINM", NULL, "Mic Jack"},
175 };
176
177 static struct platform_device *s3c24xx_snd_device;
178
179 static int h1940_uda1380_init(struct snd_soc_pcm_runtime *rtd)
180 {
181 struct snd_soc_codec *codec = rtd->codec;
182 struct snd_soc_dapm_context *dapm = &codec->dapm;
183 int err;
184
185 snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
186 snd_soc_dapm_enable_pin(dapm, "Speaker");
187 snd_soc_dapm_enable_pin(dapm, "Mic Jack");
188
189 snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
190 &hp_jack);
191
192 snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
193 hp_jack_pins);
194
195 snd_soc_jack_add_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
196 hp_jack_gpios);
197
198 return 0;
199 }
200
201 /* s3c24xx digital audio interface glue - connects codec <--> CPU */
202 static struct snd_soc_dai_link h1940_uda1380_dai[] = {
203 {
204 .name = "uda1380",
205 .stream_name = "UDA1380 Duplex",
206 .cpu_dai_name = "s3c24xx-iis",
207 .codec_dai_name = "uda1380-hifi",
208 .init = h1940_uda1380_init,
209 .platform_name = "samsung-audio",
210 .codec_name = "uda1380-codec.0-001a",
211 .ops = &h1940_ops,
212 },
213 };
214
215 static struct snd_soc_card h1940_asoc = {
216 .name = "h1940",
217 .dai_link = h1940_uda1380_dai,
218 .num_links = ARRAY_SIZE(h1940_uda1380_dai),
219
220 .dapm_widgets = uda1380_dapm_widgets,
221 .num_dapm_widgets = ARRAY_SIZE(uda1380_dapm_widgets),
222 .dapm_routes = audio_map,
223 .num_dapm_routes = ARRAY_SIZE(audio_map),
224 };
225
226 static int __init h1940_init(void)
227 {
228 int ret;
229
230 if (!machine_is_h1940())
231 return -ENODEV;
232
233 /* configure some gpios */
234 ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
235 if (ret)
236 goto err_out;
237
238 ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 0);
239 if (ret)
240 goto err_gpio;
241
242 s3c24xx_snd_device = platform_device_alloc("soc-audio", -1);
243 if (!s3c24xx_snd_device) {
244 ret = -ENOMEM;
245 goto err_gpio;
246 }
247
248 platform_set_drvdata(s3c24xx_snd_device, &h1940_asoc);
249 ret = platform_device_add(s3c24xx_snd_device);
250
251 if (ret)
252 goto err_plat;
253
254 return 0;
255
256 err_plat:
257 platform_device_put(s3c24xx_snd_device);
258 err_gpio:
259 gpio_free(H1940_LATCH_AUDIO_POWER);
260
261 err_out:
262 return ret;
263 }
264
265 static void __exit h1940_exit(void)
266 {
267 platform_device_unregister(s3c24xx_snd_device);
268 snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
269 hp_jack_gpios);
270 gpio_free(H1940_LATCH_AUDIO_POWER);
271 }
272
273 module_init(h1940_init);
274 module_exit(h1940_exit);
275
276 /* Module information */
277 MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
278 MODULE_DESCRIPTION("ALSA SoC H1940");
279 MODULE_LICENSE("GPL");
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