1 // SPDX-License-Identifier: GPL-2.0+
2 //
3 // soc-util.c -- ALSA SoC Audio Layer utility functions
4 //
5 // Copyright 2009 Wolfson Microelectronics PLC.
6 //
7 // Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
8 // Liam Girdwood <lrg@slimlogic.co.uk>
9
10 #include <linux/platform_device.h>
11 #include <linux/export.h>
12 #include <sound/core.h>
13 #include <sound/pcm.h>
14 #include <sound/pcm_params.h>
15 #include <sound/soc.h>
16
17 int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots)
18 {
19 return sample_size * channels * tdm_slots;
20 }
21 EXPORT_SYMBOL_GPL(snd_soc_calc_frame_size);
22
23 int snd_soc_params_to_frame_size(struct snd_pcm_hw_params *params)
24 {
25 int sample_size;
26
27 sample_size = snd_pcm_format_width(params_format(params));
28 if (sample_size < 0)
29 return sample_size;
30
31 return snd_soc_calc_frame_size(sample_size, params_channels(params),
32 1);
33 }
34 EXPORT_SYMBOL_GPL(snd_soc_params_to_frame_size);
35
36 int snd_soc_calc_bclk(int fs, int sample_size, int channels, int tdm_slots)
37 {
38 return fs * snd_soc_calc_frame_size(sample_size, channels, tdm_slots);
39 }
40 EXPORT_SYMBOL_GPL(snd_soc_calc_bclk);
41
42 int snd_soc_params_to_bclk(struct snd_pcm_hw_params *params)
43 {
44 int ret;
45
46 ret = snd_soc_params_to_frame_size(params);
47
48 if (ret > 0)
49 return ret * params_rate(params);
50 else
51 return ret;
52 }
53 EXPORT_SYMBOL_GPL(snd_soc_params_to_bclk);
54
55 static const struct snd_pcm_hardware dummy_dma_hardware = {
56 /* Random values to keep userspace happy when checking constraints */
57 .info = SNDRV_PCM_INFO_INTERLEAVED |
58 SNDRV_PCM_INFO_BLOCK_TRANSFER,
59 .buffer_bytes_max = 128*1024,
60 .period_bytes_min = PAGE_SIZE,
61 .period_bytes_max = PAGE_SIZE*2,
62 .periods_min = 2,
63 .periods_max = 128,
64 };
65
66 static int dummy_dma_open(struct snd_soc_component *component,
67 struct snd_pcm_substream *substream)
68 {
69 struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
70
71 /* BE's dont need dummy params */
72 if (!rtd->dai_link->no_pcm)
73 snd_soc_set_runtime_hwparams(substream, &dummy_dma_hardware);
74
75 return 0;
76 }
77
78 static const struct snd_soc_component_driver dummy_platform = {
79 .open = dummy_dma_open,
80 };
81
82 static const struct snd_soc_component_driver dummy_codec = {
83 .idle_bias_on = 1,
84 .use_pmdown_time = 1,
85 .endianness = 1,
86 .non_legacy_dai_naming = 1,
87 };
88
89 #define STUB_RATES SNDRV_PCM_RATE_8000_384000
90 #define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \
91 SNDRV_PCM_FMTBIT_U8 | \
92 SNDRV_PCM_FMTBIT_S16_LE | \
93 SNDRV_PCM_FMTBIT_U16_LE | \
94 SNDRV_PCM_FMTBIT_S24_LE | \
95 SNDRV_PCM_FMTBIT_S24_3LE | \
96 SNDRV_PCM_FMTBIT_U24_LE | \
97 SNDRV_PCM_FMTBIT_S32_LE | \
98 SNDRV_PCM_FMTBIT_U32_LE | \
99 SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE)
100
101 /*
102 * Select these from Sound Card Manually
103 * SND_SOC_POSSIBLE_DAIFMT_CBP_CFP
104 * SND_SOC_POSSIBLE_DAIFMT_CBP_CFC
105 * SND_SOC_POSSIBLE_DAIFMT_CBC_CFP
106 * SND_SOC_POSSIBLE_DAIFMT_CBC_CFC
107 */
108 static u64 dummy_dai_formats =
109 SND_SOC_POSSIBLE_DAIFMT_I2S |
110 SND_SOC_POSSIBLE_DAIFMT_RIGHT_J |
111 SND_SOC_POSSIBLE_DAIFMT_LEFT_J |
112 SND_SOC_POSSIBLE_DAIFMT_DSP_A |
113 SND_SOC_POSSIBLE_DAIFMT_DSP_B |
114 SND_SOC_POSSIBLE_DAIFMT_AC97 |
115 SND_SOC_POSSIBLE_DAIFMT_PDM |
116 SND_SOC_POSSIBLE_DAIFMT_GATED |
117 SND_SOC_POSSIBLE_DAIFMT_CONT |
118 SND_SOC_POSSIBLE_DAIFMT_NB_NF |
119 SND_SOC_POSSIBLE_DAIFMT_NB_IF |
120 SND_SOC_POSSIBLE_DAIFMT_IB_NF |
121 SND_SOC_POSSIBLE_DAIFMT_IB_IF;
122
123 static const struct snd_soc_dai_ops dummy_dai_ops = {
124 .auto_selectable_formats = &dummy_dai_formats,
125 .num_auto_selectable_formats = 1,
126 };
127
128 /*
129 * The dummy CODEC is only meant to be used in situations where there is no
130 * actual hardware.
131 *
132 * If there is actual hardware even if it does not have a control bus
133 * the hardware will still have constraints like supported samplerates, etc.
134 * which should be modelled. And the data flow graph also should be modelled
135 * using DAPM.
136 */
137 static struct snd_soc_dai_driver dummy_dai = {
138 .name = "snd-soc-dummy-dai",
139 .playback = {
140 .stream_name = "Playback",
141 .channels_min = 1,
142 .channels_max = 384,
143 .rates = STUB_RATES,
144 .formats = STUB_FORMATS,
145 },
146 .capture = {
147 .stream_name = "Capture",
148 .channels_min = 1,
149 .channels_max = 384,
150 .rates = STUB_RATES,
151 .formats = STUB_FORMATS,
152 },
153 .ops = &dummy_dai_ops,
154 };
155
156 int snd_soc_dai_is_dummy(struct snd_soc_dai *dai)
157 {
158 if (dai->driver == &dummy_dai)
159 return 1;
160 return 0;
161 }
162
163 int snd_soc_component_is_dummy(struct snd_soc_component *component)
164 {
165 return ((component->driver == &dummy_platform) ||
166 (component->driver == &dummy_codec));
167 }
168
169 static int snd_soc_dummy_probe(struct platform_device *pdev)
170 {
171 int ret;
172
173 ret = devm_snd_soc_register_component(&pdev->dev,
174 &dummy_codec, &dummy_dai, 1);
175 if (ret < 0)
176 return ret;
177
178 ret = devm_snd_soc_register_component(&pdev->dev, &dummy_platform,
179 NULL, 0);
180
181 return ret;
182 }
183
184 static struct platform_driver soc_dummy_driver = {
185 .driver = {
186 .name = "snd-soc-dummy",
187 },
188 .probe = snd_soc_dummy_probe,
189 };
190
191 static struct platform_device *soc_dummy_dev;
192
193 int __init snd_soc_util_init(void)
194 {
195 int ret;
196
197 soc_dummy_dev =
198 platform_device_register_simple("snd-soc-dummy", -1, NULL, 0);
199 if (IS_ERR(soc_dummy_dev))
200 return PTR_ERR(soc_dummy_dev);
201
202 ret = platform_driver_register(&soc_dummy_driver);
203 if (ret != 0)
204 platform_device_unregister(soc_dummy_dev);
205
206 return ret;
207 }
208
209 void __exit snd_soc_util_exit(void)
210 {
211 platform_driver_unregister(&soc_dummy_driver);
212 platform_device_unregister(soc_dummy_dev);
213 }
214
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