1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
|
/*
* h1940-uda1380.c -- ALSA Soc Audio Layer
*
* Copyright (c) 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
* Copyright (c) 2010 Vasily Khoruzhick <anarsoul@gmail.com>
*
* Based on version from Arnaud Patard <arnaud.patard@rtp-net.org>
*
* 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 <linux/gpio.h>
#include <sound/soc.h>
#include <sound/jack.h>
#include <plat/regs-iis.h>
#include <mach/h1940-latch.h>
#include <asm/mach-types.h>
#include "s3c24xx-i2s.h"
static unsigned int rates[] = {
11025,
22050,
44100,
};
static struct snd_pcm_hw_constraint_list hw_rates = {
.count = ARRAY_SIZE(rates),
.list = rates,
.mask = 0,
};
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;
runtime->hw.rate_min = hw_rates.list[0];
runtime->hw.rate_max = hw_rates.list[hw_rates.count - 1];
runtime->hw.rates = SNDRV_PCM_RATE_KNOT;
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;
struct snd_soc_dai *codec_dai = rtd->codec_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;
}
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* 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(H1940_LATCH_AUDIO_POWER, 1);
else
gpio_set_value(H1940_LATCH_AUDIO_POWER, 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)
{
struct snd_soc_codec *codec = rtd->codec;
struct snd_soc_dapm_context *dapm = &codec->dapm;
int err;
/* Add h1940 specific widgets */
err = snd_soc_dapm_new_controls(dapm, uda1380_dapm_widgets,
ARRAY_SIZE(uda1380_dapm_widgets));
if (err)
return err;
/* Set up h1940 specific audio path audio_mapnects */
err = snd_soc_dapm_add_routes(dapm, audio_map,
ARRAY_SIZE(audio_map));
if (err)
return err;
snd_soc_dapm_enable_pin(dapm, "Headphone Jack");
snd_soc_dapm_enable_pin(dapm, "Speaker");
snd_soc_dapm_enable_pin(dapm, "Mic Jack");
snd_soc_dapm_sync(dapm);
snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE,
&hp_jack);
snd_soc_jack_add_pins(&hp_jack, ARRAY_SIZE(hp_jack_pins),
hp_jack_pins);
snd_soc_jack_add_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 = "samsung-audio",
.codec_name = "uda1380-codec.0-001a",
.ops = &h1940_ops,
},
};
static struct snd_soc_card h1940_asoc = {
.name = "h1940",
.dai_link = h1940_uda1380_dai,
.num_links = ARRAY_SIZE(h1940_uda1380_dai),
};
static int __init h1940_init(void)
{
int ret;
if (!machine_is_h1940())
return -ENODEV;
/* configure some gpios */
ret = gpio_request(H1940_LATCH_AUDIO_POWER, "speaker-power");
if (ret)
goto err_out;
ret = gpio_direction_output(H1940_LATCH_AUDIO_POWER, 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(H1940_LATCH_AUDIO_POWER);
err_out:
return ret;
}
static void __exit h1940_exit(void)
{
platform_device_unregister(s3c24xx_snd_device);
snd_soc_jack_free_gpios(&hp_jack, ARRAY_SIZE(hp_jack_gpios),
hp_jack_gpios);
gpio_free(H1940_LATCH_AUDIO_POWER);
}
module_init(h1940_init);
module_exit(h1940_exit);
/* Module information */
MODULE_AUTHOR("Arnaud Patard, Vasily Khoruzhick");
MODULE_DESCRIPTION("ALSA SoC H1940");
MODULE_LICENSE("GPL");
|
