mm: compaction: don't depend on HUGETLB_PAGE
[linux/fpc-iii.git] / sound / soc / codecs / stac9766.c
blob78b2b50271e25893ade9c8b0ce3b0f0be6ab4e92
1 /*
2 * stac9766.c -- ALSA SoC STAC9766 codec support
4 * Copyright 2009 Jon Smirl, Digispeaker
5 * Author: Jon Smirl <jonsmirl@gmail.com>
7 * This program is free software; you can redistribute it and/or modify it
8 * under the terms of the GNU General Public License as published by the
9 * Free Software Foundation; either version 2 of the License, or (at your
10 * option) any later version.
12 * Features:-
14 * o Support for AC97 Codec, S/PDIF
17 #include <linux/init.h>
18 #include <linux/slab.h>
19 #include <linux/module.h>
20 #include <linux/device.h>
21 #include <sound/core.h>
22 #include <sound/pcm.h>
23 #include <sound/ac97_codec.h>
24 #include <sound/initval.h>
25 #include <sound/pcm_params.h>
26 #include <sound/soc.h>
27 #include <sound/tlv.h>
29 #include "stac9766.h"
31 #define STAC9766_VERSION "0.10"
34 * STAC9766 register cache
36 static const u16 stac9766_reg[] = {
37 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
38 0x0000, 0x0000, 0x8008, 0x8008, /* e */
39 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
40 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
41 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
42 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
43 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
44 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
45 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
46 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
47 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
48 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
49 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
50 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
51 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
52 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
55 static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
56 "Line", "Stereo Mix", "Mono Mix", "Phone"};
57 static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
58 static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
59 static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
60 static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
61 static const char *stac9766_record_all_mux[] = {"All analog",
62 "Analog plus DAC"};
63 static const char *stac9766_boost1[] = {"0dB", "10dB"};
64 static const char *stac9766_boost2[] = {"0dB", "20dB"};
65 static const char *stac9766_stereo_mic[] = {"Off", "On"};
67 static const struct soc_enum stac9766_record_enum =
68 SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
69 static const struct soc_enum stac9766_mono_enum =
70 SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
71 static const struct soc_enum stac9766_mic_enum =
72 SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
73 static const struct soc_enum stac9766_SPDIF_enum =
74 SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
75 static const struct soc_enum stac9766_popbypass_enum =
76 SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
77 static const struct soc_enum stac9766_record_all_enum =
78 SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
79 stac9766_record_all_mux);
80 static const struct soc_enum stac9766_boost1_enum =
81 SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
82 static const struct soc_enum stac9766_boost2_enum =
83 SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
84 static const struct soc_enum stac9766_stereo_mic_enum =
85 SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
87 static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
88 static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
89 static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
90 static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
92 static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
93 SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
94 SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
95 SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
96 master_tlv),
97 SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
98 SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
99 master_tlv),
100 SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
102 SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
103 SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
106 SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
107 SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
108 SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
109 SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
110 SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
112 SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
113 SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
114 SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
115 SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
116 SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
118 SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
119 SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
120 SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
121 SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
122 SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
123 SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
124 SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
125 SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
127 SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
128 SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
129 SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
130 SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
131 SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
133 SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
134 SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
135 SOC_ENUM("Record All Mux", stac9766_record_all_enum),
136 SOC_ENUM("Record Mux", stac9766_record_enum),
137 SOC_ENUM("Mono Mux", stac9766_mono_enum),
138 SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
141 static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
142 unsigned int val)
144 u16 *cache = codec->reg_cache;
146 if (reg > AC97_STAC_PAGE0) {
147 stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
148 soc_ac97_ops.write(codec->ac97, reg, val);
149 stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
150 return 0;
152 if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
153 return -EIO;
155 soc_ac97_ops.write(codec->ac97, reg, val);
156 cache[reg / 2] = val;
157 return 0;
160 static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
161 unsigned int reg)
163 u16 val = 0, *cache = codec->reg_cache;
165 if (reg > AC97_STAC_PAGE0) {
166 stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
167 val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
168 stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
169 return val;
171 if (reg / 2 >= ARRAY_SIZE(stac9766_reg))
172 return -EIO;
174 if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
175 reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
176 reg == AC97_VENDOR_ID2) {
178 val = soc_ac97_ops.read(codec->ac97, reg);
179 return val;
181 return cache[reg / 2];
184 static int ac97_analog_prepare(struct snd_pcm_substream *substream,
185 struct snd_soc_dai *dai)
187 struct snd_soc_codec *codec = dai->codec;
188 struct snd_pcm_runtime *runtime = substream->runtime;
189 unsigned short reg, vra;
191 vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
193 vra |= 0x1; /* enable variable rate audio */
194 vra &= ~0x4; /* disable SPDIF output */
196 stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
198 if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
199 reg = AC97_PCM_FRONT_DAC_RATE;
200 else
201 reg = AC97_PCM_LR_ADC_RATE;
203 return stac9766_ac97_write(codec, reg, runtime->rate);
206 static int ac97_digital_prepare(struct snd_pcm_substream *substream,
207 struct snd_soc_dai *dai)
209 struct snd_soc_codec *codec = dai->codec;
210 struct snd_pcm_runtime *runtime = substream->runtime;
211 unsigned short reg, vra;
213 stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
215 vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
216 vra |= 0x5; /* Enable VRA and SPDIF out */
218 stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
220 reg = AC97_PCM_FRONT_DAC_RATE;
222 return stac9766_ac97_write(codec, reg, runtime->rate);
225 static int stac9766_set_bias_level(struct snd_soc_codec *codec,
226 enum snd_soc_bias_level level)
228 switch (level) {
229 case SND_SOC_BIAS_ON: /* full On */
230 case SND_SOC_BIAS_PREPARE: /* partial On */
231 case SND_SOC_BIAS_STANDBY: /* Off, with power */
232 stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
233 break;
234 case SND_SOC_BIAS_OFF: /* Off, without power */
235 /* disable everything including AC link */
236 stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
237 break;
239 codec->dapm.bias_level = level;
240 return 0;
243 static int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
245 if (try_warm && soc_ac97_ops.warm_reset) {
246 soc_ac97_ops.warm_reset(codec->ac97);
247 if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
248 return 1;
251 soc_ac97_ops.reset(codec->ac97);
252 if (soc_ac97_ops.warm_reset)
253 soc_ac97_ops.warm_reset(codec->ac97);
254 if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
255 return -EIO;
256 return 0;
259 static int stac9766_codec_suspend(struct snd_soc_codec *codec,
260 pm_message_t state)
262 stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
263 return 0;
266 static int stac9766_codec_resume(struct snd_soc_codec *codec)
268 u16 id, reset;
270 reset = 0;
271 /* give the codec an AC97 warm reset to start the link */
272 reset:
273 if (reset > 5) {
274 printk(KERN_ERR "stac9766 failed to resume");
275 return -EIO;
277 codec->ac97->bus->ops->warm_reset(codec->ac97);
278 id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
279 if (id != 0x4c13) {
280 stac9766_reset(codec, 0);
281 reset++;
282 goto reset;
284 stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
286 return 0;
289 static struct snd_soc_dai_ops stac9766_dai_ops_analog = {
290 .prepare = ac97_analog_prepare,
293 static struct snd_soc_dai_ops stac9766_dai_ops_digital = {
294 .prepare = ac97_digital_prepare,
297 static struct snd_soc_dai_driver stac9766_dai[] = {
299 .name = "stac9766-hifi-analog",
300 .ac97_control = 1,
302 /* stream cababilities */
303 .playback = {
304 .stream_name = "stac9766 analog",
305 .channels_min = 1,
306 .channels_max = 2,
307 .rates = SNDRV_PCM_RATE_8000_48000,
308 .formats = SND_SOC_STD_AC97_FMTS,
310 .capture = {
311 .stream_name = "stac9766 analog",
312 .channels_min = 1,
313 .channels_max = 2,
314 .rates = SNDRV_PCM_RATE_8000_48000,
315 .formats = SND_SOC_STD_AC97_FMTS,
317 /* alsa ops */
318 .ops = &stac9766_dai_ops_analog,
321 .name = "stac9766-hifi-IEC958",
322 .ac97_control = 1,
324 /* stream cababilities */
325 .playback = {
326 .stream_name = "stac9766 IEC958",
327 .channels_min = 1,
328 .channels_max = 2,
329 .rates = SNDRV_PCM_RATE_32000 | \
330 SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
331 .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
333 /* alsa ops */
334 .ops = &stac9766_dai_ops_digital,
338 static int stac9766_codec_probe(struct snd_soc_codec *codec)
340 int ret = 0;
342 printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
344 ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
345 if (ret < 0)
346 goto codec_err;
348 /* do a cold reset for the controller and then try
349 * a warm reset followed by an optional cold reset for codec */
350 stac9766_reset(codec, 0);
351 ret = stac9766_reset(codec, 1);
352 if (ret < 0) {
353 printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
354 goto codec_err;
357 stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
359 snd_soc_add_controls(codec, stac9766_snd_ac97_controls,
360 ARRAY_SIZE(stac9766_snd_ac97_controls));
362 return 0;
364 codec_err:
365 snd_soc_free_ac97_codec(codec);
366 return ret;
369 static int stac9766_codec_remove(struct snd_soc_codec *codec)
371 snd_soc_free_ac97_codec(codec);
372 return 0;
375 static struct snd_soc_codec_driver soc_codec_dev_stac9766 = {
376 .write = stac9766_ac97_write,
377 .read = stac9766_ac97_read,
378 .set_bias_level = stac9766_set_bias_level,
379 .probe = stac9766_codec_probe,
380 .remove = stac9766_codec_remove,
381 .suspend = stac9766_codec_suspend,
382 .resume = stac9766_codec_resume,
383 .reg_cache_size = sizeof(stac9766_reg),
384 .reg_word_size = sizeof(u16),
385 .reg_cache_step = 2,
386 .reg_cache_default = stac9766_reg,
389 static __devinit int stac9766_probe(struct platform_device *pdev)
391 return snd_soc_register_codec(&pdev->dev,
392 &soc_codec_dev_stac9766, stac9766_dai, ARRAY_SIZE(stac9766_dai));
395 static int __devexit stac9766_remove(struct platform_device *pdev)
397 snd_soc_unregister_codec(&pdev->dev);
398 return 0;
401 static struct platform_driver stac9766_codec_driver = {
402 .driver = {
403 .name = "stac9766-codec",
404 .owner = THIS_MODULE,
407 .probe = stac9766_probe,
408 .remove = __devexit_p(stac9766_remove),
411 static int __init stac9766_init(void)
413 return platform_driver_register(&stac9766_codec_driver);
415 module_init(stac9766_init);
417 static void __exit stac9766_exit(void)
419 platform_driver_unregister(&stac9766_codec_driver);
421 module_exit(stac9766_exit);
423 MODULE_DESCRIPTION("ASoC stac9766 driver");
424 MODULE_AUTHOR("Jon Smirl <jonsmirl@gmail.com>");
425 MODULE_LICENSE("GPL");