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
|
// SPDX-License-Identifier: GPL-2.0-only
/*
* bytcht_nocodec.c - ASoc Machine driver for MinnowBoard Max and Up
* to make I2S signals observable on the Low-Speed connector. Audio codec
* is not managed by ASoC/DAPM
*
* Copyright (C) 2015-2017 Intel Corp
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*
* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
*/
#include <linux/module.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include "../atom/sst-atom-controls.h"
static const struct snd_soc_dapm_widget widgets[] = {
SND_SOC_DAPM_MIC("Mic", NULL),
SND_SOC_DAPM_SPK("Speaker", NULL),
};
static const struct snd_kcontrol_new controls[] = {
SOC_DAPM_PIN_SWITCH("Mic"),
SOC_DAPM_PIN_SWITCH("Speaker"),
};
static const struct snd_soc_dapm_route audio_map[] = {
{"ssp2 Tx", NULL, "codec_out0"},
{"ssp2 Tx", NULL, "codec_out1"},
{"codec_in0", NULL, "ssp2 Rx"},
{"codec_in1", NULL, "ssp2 Rx"},
{"ssp2 Rx", NULL, "Mic"},
{"Speaker", NULL, "ssp2 Tx"},
};
static int codec_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
int ret;
/* The DSP will convert the FE rate to 48k, stereo, 24bits */
rate->min = rate->max = 48000;
channels->min = channels->max = 2;
/* set SSP2 to 24-bit */
params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
/*
* Default mode for SSP configuration is TDM 4 slot, override config
* with explicit setting to I2S 2ch 24-bit. The word length is set with
* dai_set_tdm_slot() since there is no other API exposed
*/
ret = snd_soc_dai_set_fmt(rtd->cpu_dai,
SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0) {
dev_err(rtd->dev, "can't set format to I2S, err %d\n", ret);
return ret;
}
ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 24);
if (ret < 0) {
dev_err(rtd->dev, "can't set I2S config, err %d\n", ret);
return ret;
}
return 0;
}
static const unsigned int rates_48000[] = {
48000,
};
static const struct snd_pcm_hw_constraint_list constraints_48000 = {
.count = ARRAY_SIZE(rates_48000),
.list = rates_48000,
};
static int aif1_startup(struct snd_pcm_substream *substream)
{
return snd_pcm_hw_constraint_list(substream->runtime, 0,
SNDRV_PCM_HW_PARAM_RATE,
&constraints_48000);
}
static struct snd_soc_ops aif1_ops = {
.startup = aif1_startup,
};
SND_SOC_DAILINK_DEF(dummy,
DAILINK_COMP_ARRAY(COMP_DUMMY()));
SND_SOC_DAILINK_DEF(media,
DAILINK_COMP_ARRAY(COMP_CPU("media-cpu-dai")));
SND_SOC_DAILINK_DEF(deepbuffer,
DAILINK_COMP_ARRAY(COMP_CPU("deepbuffer-cpu-dai")));
SND_SOC_DAILINK_DEF(ssp2_port,
DAILINK_COMP_ARRAY(COMP_CPU("ssp2-port")));
SND_SOC_DAILINK_DEF(platform,
DAILINK_COMP_ARRAY(COMP_PLATFORM("sst-mfld-platform")));
static struct snd_soc_dai_link dais[] = {
[MERR_DPCM_AUDIO] = {
.name = "Audio Port",
.stream_name = "Audio",
.ignore_suspend = 1,
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.dpcm_capture = 1,
.ops = &aif1_ops,
SND_SOC_DAILINK_REG(media, dummy, platform),
},
[MERR_DPCM_DEEP_BUFFER] = {
.name = "Deep-Buffer Audio Port",
.stream_name = "Deep-Buffer Audio",
.ignore_suspend = 1,
.nonatomic = true,
.dynamic = 1,
.dpcm_playback = 1,
.ops = &aif1_ops,
SND_SOC_DAILINK_REG(deepbuffer, dummy, platform),
},
/* CODEC<->CODEC link */
/* back ends */
{
.name = "SSP2-LowSpeed Connector",
.id = 0,
.no_pcm = 1,
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
| SND_SOC_DAIFMT_CBS_CFS,
.be_hw_params_fixup = codec_fixup,
.ignore_suspend = 1,
.nonatomic = true,
.dpcm_playback = 1,
.dpcm_capture = 1,
SND_SOC_DAILINK_REG(ssp2_port, dummy, platform),
},
};
/* SoC card */
static struct snd_soc_card bytcht_nocodec_card = {
.name = "bytcht-nocodec",
.owner = THIS_MODULE,
.dai_link = dais,
.num_links = ARRAY_SIZE(dais),
.dapm_widgets = widgets,
.num_dapm_widgets = ARRAY_SIZE(widgets),
.dapm_routes = audio_map,
.num_dapm_routes = ARRAY_SIZE(audio_map),
.controls = controls,
.num_controls = ARRAY_SIZE(controls),
.fully_routed = true,
};
static int snd_bytcht_nocodec_mc_probe(struct platform_device *pdev)
{
int ret_val = 0;
/* register the soc card */
bytcht_nocodec_card.dev = &pdev->dev;
ret_val = devm_snd_soc_register_card(&pdev->dev, &bytcht_nocodec_card);
if (ret_val) {
dev_err(&pdev->dev, "devm_snd_soc_register_card failed %d\n",
ret_val);
return ret_val;
}
platform_set_drvdata(pdev, &bytcht_nocodec_card);
return ret_val;
}
static struct platform_driver snd_bytcht_nocodec_mc_driver = {
.driver = {
.name = "bytcht_nocodec",
},
.probe = snd_bytcht_nocodec_mc_probe,
};
module_platform_driver(snd_bytcht_nocodec_mc_driver);
MODULE_DESCRIPTION("ASoC Intel(R) Baytrail/Cherrytrail Nocodec Machine driver");
MODULE_AUTHOR("Pierre-Louis Bossart <pierre-louis.bossart at linux.intel.com>");
MODULE_LICENSE("GPL v2");
MODULE_ALIAS("platform:bytcht_nocodec");
|
