summaryrefslogtreecommitdiff
path: root/sound/soc/blackfin/bfin-eval-adau1373.c
blob: 85ed39abe10eb2c41eedef391705edef5519b52d (plain)
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
/*
 * Machine driver for EVAL-ADAU1373 on Analog Devices bfin
 * evaluation boards.
 *
 * Copyright 2011 Analog Devices Inc.
 * Author: Lars-Peter Clausen <lars@metafoo.de>
 *
 * Licensed under the GPL-2 or later.
 */

#include <linux/module.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/pcm_params.h>

#include "../codecs/adau1373.h"

static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = {
	SND_SOC_DAPM_LINE("Line In1", NULL),
	SND_SOC_DAPM_LINE("Line In2", NULL),
	SND_SOC_DAPM_LINE("Line In3", NULL),
	SND_SOC_DAPM_LINE("Line In4", NULL),

	SND_SOC_DAPM_LINE("Line Out1", NULL),
	SND_SOC_DAPM_LINE("Line Out2", NULL),
	SND_SOC_DAPM_LINE("Stereo Out", NULL),
	SND_SOC_DAPM_HP("Headphone", NULL),
	SND_SOC_DAPM_HP("Earpiece", NULL),
	SND_SOC_DAPM_SPK("Speaker", NULL),
};

static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = {
	{ "AIN1L", NULL, "Line In1" },
	{ "AIN1R", NULL, "Line In1" },
	{ "AIN2L", NULL, "Line In2" },
	{ "AIN2R", NULL, "Line In2" },
	{ "AIN3L", NULL, "Line In3" },
	{ "AIN3R", NULL, "Line In3" },
	{ "AIN4L", NULL, "Line In4" },
	{ "AIN4R", NULL, "Line In4" },

	/* MICBIAS can be connected via a jumper to the line-in jack, since w
	   don't know which one is going to be used, just power both. */
	{ "Line In1", NULL, "MICBIAS1" },
	{ "Line In2", NULL, "MICBIAS1" },
	{ "Line In3", NULL, "MICBIAS1" },
	{ "Line In4", NULL, "MICBIAS1" },
	{ "Line In1", NULL, "MICBIAS2" },
	{ "Line In2", NULL, "MICBIAS2" },
	{ "Line In3", NULL, "MICBIAS2" },
	{ "Line In4", NULL, "MICBIAS2" },

	{ "Line Out1", NULL, "LOUT1L" },
	{ "Line Out1", NULL, "LOUT1R" },
	{ "Line Out2", NULL, "LOUT2L" },
	{ "Line Out2", NULL, "LOUT2R" },
	{ "Headphone", NULL, "HPL" },
	{ "Headphone", NULL, "HPR" },
	{ "Earpiece", NULL, "EP" },
	{ "Speaker", NULL, "SPKL" },
	{ "Stereo Out", NULL, "SPKR" },
};

static int bfin_eval_adau1373_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 ret;
	int pll_rate;

	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
	if (ret)
		return ret;

	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
			SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
	if (ret)
		return ret;

	switch (params_rate(params)) {
	case 48000:
	case 8000:
	case 12000:
	case 16000:
	case 24000:
	case 32000:
		pll_rate = 48000 * 1024;
		break;
	case 44100:
	case 7350:
	case 11025:
	case 14700:
	case 22050:
	case 29400:
		pll_rate = 44100 * 1024;
		break;
	default:
		return -EINVAL;
	}

	ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
			ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
	if (ret)
		return ret;

	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
			SND_SOC_CLOCK_IN);

	return ret;
}

static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd)
{
	struct snd_soc_dai *codec_dai = rtd->codec_dai;
	unsigned int pll_rate = 48000 * 1024;
	int ret;

	ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1,
			ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate);
	if (ret)
		return ret;

	ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate,
			SND_SOC_CLOCK_IN);

	return ret;
}
static struct snd_soc_ops bfin_eval_adau1373_ops = {
	.hw_params = bfin_eval_adau1373_hw_params,
};

static struct snd_soc_dai_link bfin_eval_adau1373_dai = {
	.name = "adau1373",
	.stream_name = "adau1373",
	.cpu_dai_name = "bfin-i2s.0",
	.codec_dai_name = "adau1373-aif1",
	.platform_name = "bfin-i2s-pcm-audio",
	.codec_name = "adau1373.0-001a",
	.ops = &bfin_eval_adau1373_ops,
	.init = bfin_eval_adau1373_codec_init,
};

static struct snd_soc_card bfin_eval_adau1373 = {
	.name = "bfin-eval-adau1373",
	.dai_link = &bfin_eval_adau1373_dai,
	.num_links = 1,

	.dapm_widgets		= bfin_eval_adau1373_dapm_widgets,
	.num_dapm_widgets	= ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets),
	.dapm_routes		= bfin_eval_adau1373_dapm_routes,
	.num_dapm_routes	= ARRAY_SIZE(bfin_eval_adau1373_dapm_routes),
};

static int bfin_eval_adau1373_probe(struct platform_device *pdev)
{
	struct snd_soc_card *card = &bfin_eval_adau1373;

	card->dev = &pdev->dev;

	return snd_soc_register_card(&bfin_eval_adau1373);
}

static int __devexit bfin_eval_adau1373_remove(struct platform_device *pdev)
{
	struct snd_soc_card *card = platform_get_drvdata(pdev);

	snd_soc_unregister_card(card);

	return 0;
}

static struct platform_driver bfin_eval_adau1373_driver = {
	.driver = {
		.name = "bfin-eval-adau1373",
		.owner = THIS_MODULE,
		.pm = &snd_soc_pm_ops,
	},
	.probe = bfin_eval_adau1373_probe,
	.remove = __devexit_p(bfin_eval_adau1373_remove),
};

module_platform_driver(bfin_eval_adau1373_driver);

MODULE_AUTHOR("Lars-Peter Clausen <lars@metafoo.de>");
MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver");
MODULE_LICENSE("GPL");
MODULE_ALIAS("platform:bfin-eval-adau1373");