aboutsummaryrefslogtreecommitdiffstats
path: root/sound/soc/codecs/stac9766.c
blob: 2eda85ba79acd2cdd442b78fa4631542f0caffdd (plain) (blame)
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
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
/*
 * stac9766.c  --  ALSA SoC STAC9766 codec support
 *
 * Copyright 2009 Jon Smirl, Digispeaker
 * Author: Jon Smirl <jonsmirl@gmail.com>
 *
 *  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.
 *
 *  Features:-
 *
 *   o Support for AC97 Codec, S/PDIF
 */

#include <linux/init.h>
#include <linux/slab.h>
#include <linux/module.h>
#include <linux/device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/ac97_codec.h>
#include <sound/initval.h>
#include <sound/pcm_params.h>
#include <sound/soc.h>
#include <sound/tlv.h>

#include "stac9766.h"

#define STAC9766_VERSION "0.10"

/*
 * STAC9766 register cache
 */
static const u16 stac9766_reg[] = {
	0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
	0x0000, 0x0000, 0x8008, 0x8008, /* e */
	0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
	0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
	0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
	0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
	0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
	0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
	0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
	0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
	0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
	0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
};

static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
			"Line", "Stereo Mix", "Mono Mix", "Phone"};
static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
static const char *stac9766_record_all_mux[] = {"All analog",
	"Analog plus DAC"};
static const char *stac9766_boost1[] = {"0dB", "10dB"};
static const char *stac9766_boost2[] = {"0dB", "20dB"};
static const char *stac9766_stereo_mic[] = {"Off", "On"};

static const struct soc_enum stac9766_record_enum =
	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
static const struct soc_enum stac9766_mono_enum =
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
static const struct soc_enum stac9766_mic_enum =
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
static const struct soc_enum stac9766_SPDIF_enum =
	SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
static const struct soc_enum stac9766_popbypass_enum =
	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
static const struct soc_enum stac9766_record_all_enum =
	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
			stac9766_record_all_mux);
static const struct soc_enum stac9766_boost1_enum =
	SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
static const struct soc_enum stac9766_boost2_enum =
	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
static const struct soc_enum stac9766_stereo_mic_enum =
	SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);

static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);

static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
		       master_tlv),
	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
		       master_tlv),
	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),

	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),


	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),

	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),

	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),

	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),

	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
	SOC_ENUM("Record Mux", stac9766_record_enum),
	SOC_ENUM("Mono Mux", stac9766_mono_enum),
	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
};

static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
			       unsigned int val)
{
	u16 *cache = codec->reg_cache;

	if (reg > AC97_STAC_PAGE0) {
		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
		soc_ac97_ops.write(codec->ac97, reg, val);
		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
		return 0;
	}
	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
		return -EIO;

	soc_ac97_ops.write(codec->ac97, reg, val);
	cache[reg / 2] = val;
	return 0;
}

static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
				       unsigned int reg)
{
	u16 val = 0, *cache = codec->reg_cache;

	if (reg > AC97_STAC_PAGE0) {
		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
		val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
		return val;
	}
	if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
		return -EIO;

	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
		reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
		reg == AC97_VENDOR_ID2) {

		val = soc_ac97_ops.read(codec->ac97, reg);
		return val;
	}
	return cache[reg / 2];
}

static int ac97_analog_prepare(struct snd_pcm_substream *substream,
			       struct snd_soc_dai *dai)
{
	struct snd_soc_codec *codec = dai->codec;
	struct snd_pcm_runtime *runtime = substream->runtime;
	unsigned short reg, vra;

	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);

	vra |= 0x1; /* enable variable rate audio */
	vra &= ~0x4; /* disable SPDIF output */

	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);

	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
		reg = AC97_PCM_FRONT_DAC_RATE;
	else
		reg = AC97_PCM_LR_ADC_RATE;

	return stac9766_ac97_write(codec, reg, runtime->rate);
}

static int ac97_digital_prepare(struct snd_pcm_substream *substream,
				struct snd_soc_dai *dai)
{
	struct snd_soc_codec *codec = dai->codec;
	struct snd_pcm_runtime *runtime = substream->runtime;
	unsigned short reg, vra;

	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);

	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
	vra |= 0x5; /* Enable VRA and SPDIF out */

	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);

	reg = AC97_PCM_FRONT_DAC_RATE;

	return stac9766_ac97_write(codec, reg, runtime->rate);
}

static int stac9766_set_bias_level(struct snd_soc_codec *codec,
				   enum snd_soc_bias_level level)
{
	switch (level) {
	case SND_SOC_BIAS_ON: /* full On */
	case SND_SOC_BIAS_PREPARE: /* partial On */
	case SND_SOC_BIAS_STANDBY: /* Off, with power */
		stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
		break;
	case SND_SOC_BIAS_OFF: /* Off, without power */
		/* disable everything including AC link */
		stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
		break;
	}
	codec->dapm.bias_level = level;
	return 0;
}

static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
{
	if (try_warm && soc_ac97_ops.warm_reset) {
		soc_ac97_ops.warm_reset(codec->ac97);
		if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
			return 1;
	}

	soc_ac97_ops.reset(codec->ac97);
	if (soc_ac97_ops.warm_reset)
		soc_ac97_ops.warm_reset(codec->ac97);
	if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
		return -EIO;
	return 0;
}

static int stac9766_codec_suspend(struct snd_soc_codec *codec)
{
	stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
	return 0;
}

static int stac9766_codec_resume(struct snd_soc_codec *codec)
{
	u16 id, reset;

	reset = 0;
	/* give the codec an AC97 warm reset to start the link */
reset:
	if (reset > 5) {
		printk(KERN_ERR "stac9766 failed to resume");
		return -EIO;
	}
	codec->ac97->bus->ops->warm_reset(codec->ac97);
	id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
	if (id != 0x4c13) {
		stac9766_reset(codec, 0);
		reset++;
		goto reset;
	}
	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	return 0;
}

static const struct snd_soc_dai_ops stac9766_dai_ops_analog = {
	.prepare = ac97_analog_prepare,
};

static const struct snd_soc_dai_ops stac9766_dai_ops_digital = {
	.prepare = ac97_digital_prepare,
};

static struct snd_soc_dai_driver stac9766_dai[] = {
{
	.name = "stac9766-hifi-analog",
	.ac97_control = 1,

	/* stream cababilities */
	.playback = {
		.stream_name = "stac9766 analog",
		.channels_min = 1,
		.channels_max = 2,
		.rates = SNDRV_PCM_RATE_8000_48000,
		.formats = SND_SOC_STD_AC97_FMTS,
	},
	.capture = {
		.stream_name = "stac9766 analog",
		.channels_min = 1,
		.channels_max = 2,
		.rates = SNDRV_PCM_RATE_8000_48000,
		.formats = SND_SOC_STD_AC97_FMTS,
	},
	/* alsa ops */
	.ops = &stac9766_dai_ops_analog,
},
{
	.name = "stac9766-hifi-IEC958",
	.ac97_control = 1,

	/* stream cababilities */
	.playback = {
		.stream_name = "stac9766 IEC958",
		.channels_min = 1,
		.channels_max = 2,
		.rates = SNDRV_PCM_RATE_32000 | \
			SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
		.formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
	},
	/* alsa ops */
	.ops = &stac9766_dai_ops_digital,
}
};

static int stac9766_codec_probe(struct snd_soc_codec *codec)
{
	int ret = 0;

	printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);

	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
	if (ret < 0)
		goto codec_err;

	/* do a cold reset for the controller and then try
	 * a warm reset followed by an optional cold reset for codec */
	stac9766_reset(codec, 0);
	ret = stac9766_reset(codec, 1);
	if (ret < 0) {
		printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
		goto codec_err;
	}

	stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);

	snd_soc_add_codec_controls(codec, stac9766_snd_ac97_controls,
			     ARRAY_SIZE(stac9766_snd_ac97_controls));

	return 0;

codec_err:
	snd_soc_free_ac97_codec(codec);
	return ret;
}

static int stac9766_codec_remove(struct snd_soc_codec *codec)
{
	snd_soc_free_ac97_codec(codec);
	return 0;
}

static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
	.write = stac9766_ac97_write,
	.read = stac9766_ac97_read,
	.set_bias_level = stac9766_set_bias_level,
	.probe = stac9766_codec_probe,
	.remove = stac9766_codec_remove,
	.suspend = stac9766_codec_suspend,
	.resume = stac9766_codec_resume,
	.reg_cache_size = ARRAY_SIZE(stac9766_reg),
	.reg_word_size = sizeof(u16),
	.reg_cache_step = 2,
	.reg_cache_default = stac9766_reg,
};

static int stac9766_probe(struct platform_device *pdev)
{
	return snd_soc_register_codec(&pdev->dev,
			&soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
}

static int stac9766_remove(struct platform_device *pdev)
{
	snd_soc_unregister_codec(&pdev->dev);
	return 0;
}

static struct platform_driver stac9766_codec_driver = {
	.driver = {
			.name = "stac9766-codec",
			.owner = THIS_MODULE,
	},

	.probe = stac9766_probe,
	.remove = stac9766_remove,
};

module_platform_driver(stac9766_codec_driver);

MODULE_DESCRIPTION("ASoC stac9766 driver");
MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
MODULE_LICENSE("GPL");