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alc5623.c
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1 /*
2  * alc5623.c -- alc562[123] ALSA Soc Audio driver
3  *
4  * Copyright 2008 Realtek Microelectronics
5  * Author: flove <[email protected]> Ethan <[email protected]>
6  *
7  * Copyright 2010 Arnaud Patard <[email protected]>
8  *
9  *
10  * Based on WM8753.c
11  *
12  * This program is free software; you can redistribute it and/or modify
13  * it under the terms of the GNU General Public License version 2 as
14  * published by the Free Software Foundation.
15  *
16  */
17 
18 #include <linux/module.h>
19 #include <linux/kernel.h>
20 #include <linux/init.h>
21 #include <linux/delay.h>
22 #include <linux/pm.h>
23 #include <linux/i2c.h>
24 #include <linux/slab.h>
25 #include <sound/core.h>
26 #include <sound/pcm.h>
27 #include <sound/pcm_params.h>
28 #include <sound/tlv.h>
29 #include <sound/soc.h>
30 #include <sound/initval.h>
31 #include <sound/alc5623.h>
32 
33 #include "alc5623.h"
34 
35 static int caps_charge = 2000;
36 module_param(caps_charge, int, 0);
37 MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
38 
39 /* codec private data */
40 struct alc5623_priv {
41  enum snd_soc_control_type control_type;
42  u8 id;
43  unsigned int sysclk;
44  u16 reg_cache[ALC5623_VENDOR_ID2+2];
45  unsigned int add_ctrl;
46  unsigned int jack_det_ctrl;
47 };
48 
49 static void alc5623_fill_cache(struct snd_soc_codec *codec)
50 {
51  int i, step = codec->driver->reg_cache_step;
52  u16 *cache = codec->reg_cache;
53 
54  /* not really efficient ... */
55  codec->cache_bypass = 1;
56  for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
57  cache[i] = snd_soc_read(codec, i);
58  codec->cache_bypass = 0;
59 }
60 
61 static inline int alc5623_reset(struct snd_soc_codec *codec)
62 {
63  return snd_soc_write(codec, ALC5623_RESET, 0);
64 }
65 
66 static int amp_mixer_event(struct snd_soc_dapm_widget *w,
67  struct snd_kcontrol *kcontrol, int event)
68 {
69  /* to power-on/off class-d amp generators/speaker */
70  /* need to write to 'index-46h' register : */
71  /* so write index num (here 0x46) to reg 0x6a */
72  /* and then 0xffff/0 to reg 0x6c */
73  snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
74 
75  switch (event) {
76  case SND_SOC_DAPM_PRE_PMU:
77  snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
78  break;
79  case SND_SOC_DAPM_POST_PMD:
80  snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
81  break;
82  }
83 
84  return 0;
85 }
86 
87 /*
88  * ALC5623 Controls
89  */
90 
91 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
92 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
93 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
94 static const unsigned int boost_tlv[] = {
95  TLV_DB_RANGE_HEAD(3),
96  0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
97  1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
98  2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
99 };
100 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
101 
102 static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = {
103  SOC_DOUBLE_TLV("Speaker Playback Volume",
104  ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
105  SOC_DOUBLE("Speaker Playback Switch",
106  ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
107  SOC_DOUBLE_TLV("Headphone Playback Volume",
108  ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
109  SOC_DOUBLE("Headphone Playback Switch",
110  ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
111 };
112 
113 static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = {
114  SOC_DOUBLE_TLV("Speaker Playback Volume",
115  ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
116  SOC_DOUBLE("Speaker Playback Switch",
117  ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
118  SOC_DOUBLE_TLV("Line Playback Volume",
119  ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
120  SOC_DOUBLE("Line Playback Switch",
121  ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
122 };
123 
124 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
125  SOC_DOUBLE_TLV("Line Playback Volume",
126  ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
127  SOC_DOUBLE("Line Playback Switch",
128  ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
129  SOC_DOUBLE_TLV("Headphone Playback Volume",
130  ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
131  SOC_DOUBLE("Headphone Playback Switch",
132  ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
133 };
134 
135 static const struct snd_kcontrol_new alc5623_snd_controls[] = {
136  SOC_DOUBLE_TLV("Auxout Playback Volume",
137  ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
138  SOC_DOUBLE("Auxout Playback Switch",
139  ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
140  SOC_DOUBLE_TLV("PCM Playback Volume",
141  ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
142  SOC_DOUBLE_TLV("AuxI Capture Volume",
143  ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
144  SOC_DOUBLE_TLV("LineIn Capture Volume",
145  ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
146  SOC_SINGLE_TLV("Mic1 Capture Volume",
147  ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
148  SOC_SINGLE_TLV("Mic2 Capture Volume",
149  ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
150  SOC_DOUBLE_TLV("Rec Capture Volume",
151  ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
152  SOC_SINGLE_TLV("Mic 1 Boost Volume",
153  ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
154  SOC_SINGLE_TLV("Mic 2 Boost Volume",
155  ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
156  SOC_SINGLE_TLV("Digital Boost Volume",
157  ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
158 };
159 
160 /*
161  * DAPM Controls
162  */
163 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
164 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
165 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
166 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
167 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
168 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
169 };
170 
171 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
172 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
173 };
174 
175 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
176 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
177 };
178 
179 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
180 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
181 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
182 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
183 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
184 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
185 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
186 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
187 };
188 
189 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
190 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
191 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
192 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
193 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
194 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
195 };
196 
197 /* Left Record Mixer */
198 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
199 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
200 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
201 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
202 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
203 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
204 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
205 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
206 };
207 
208 /* Right Record Mixer */
209 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
210 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
211 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
212 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
213 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
214 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
215 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
216 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
217 };
218 
219 static const char *alc5623_spk_n_sour_sel[] = {
220  "RN/-R", "RP/+R", "LN/-R", "Vmid" };
221 static const char *alc5623_hpl_out_input_sel[] = {
222  "Vmid", "HP Left Mix"};
223 static const char *alc5623_hpr_out_input_sel[] = {
224  "Vmid", "HP Right Mix"};
225 static const char *alc5623_spkout_input_sel[] = {
226  "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
227 static const char *alc5623_aux_out_input_sel[] = {
228  "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
229 
230 /* auxout output mux */
231 static const struct soc_enum alc5623_aux_out_input_enum =
232 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
233 static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
234 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
235 
236 /* speaker output mux */
237 static const struct soc_enum alc5623_spkout_input_enum =
238 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
239 static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
240 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
241 
242 /* headphone left output mux */
243 static const struct soc_enum alc5623_hpl_out_input_enum =
244 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
245 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
246 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
247 
248 /* headphone right output mux */
249 static const struct soc_enum alc5623_hpr_out_input_enum =
250 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
251 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
252 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
253 
254 /* speaker output N select */
255 static const struct soc_enum alc5623_spk_n_sour_enum =
256 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
257 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
258 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
259 
260 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
261 /* Muxes */
262 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
263  &alc5623_auxout_mux_controls),
264 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
265  &alc5623_spkout_mux_controls),
266 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
267  &alc5623_hpl_out_mux_controls),
268 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
269  &alc5623_hpr_out_mux_controls),
270 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
271  &alc5623_spkoutn_mux_controls),
272 
273 /* output mixers */
274 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
275  &alc5623_hp_mixer_controls[0],
276  ARRAY_SIZE(alc5623_hp_mixer_controls)),
277 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
278  &alc5623_hpr_mixer_controls[0],
279  ARRAY_SIZE(alc5623_hpr_mixer_controls)),
280 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
281  &alc5623_hpl_mixer_controls[0],
282  ARRAY_SIZE(alc5623_hpl_mixer_controls)),
283 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
284 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
285  &alc5623_mono_mixer_controls[0],
286  ARRAY_SIZE(alc5623_mono_mixer_controls)),
287 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
288  &alc5623_speaker_mixer_controls[0],
289  ARRAY_SIZE(alc5623_speaker_mixer_controls)),
290 
291 /* input mixers */
292 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
293  &alc5623_captureL_mixer_controls[0],
294  ARRAY_SIZE(alc5623_captureL_mixer_controls)),
295 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
296  &alc5623_captureR_mixer_controls[0],
297  ARRAY_SIZE(alc5623_captureR_mixer_controls)),
298 
299 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
300  ALC5623_PWR_MANAG_ADD2, 9, 0),
301 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
302  ALC5623_PWR_MANAG_ADD2, 8, 0),
303 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
304 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
305 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
306 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
307  ALC5623_PWR_MANAG_ADD2, 7, 0),
308 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
309  ALC5623_PWR_MANAG_ADD2, 6, 0),
310 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
311 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
312 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
313 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
314 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
315 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
316 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
317 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
318 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
319 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
320 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
321 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
322 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
323 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
324 
325 SND_SOC_DAPM_OUTPUT("AUXOUTL"),
326 SND_SOC_DAPM_OUTPUT("AUXOUTR"),
327 SND_SOC_DAPM_OUTPUT("HPL"),
328 SND_SOC_DAPM_OUTPUT("HPR"),
329 SND_SOC_DAPM_OUTPUT("SPKOUT"),
330 SND_SOC_DAPM_OUTPUT("SPKOUTN"),
331 SND_SOC_DAPM_INPUT("LINEINL"),
332 SND_SOC_DAPM_INPUT("LINEINR"),
333 SND_SOC_DAPM_INPUT("AUXINL"),
334 SND_SOC_DAPM_INPUT("AUXINR"),
335 SND_SOC_DAPM_INPUT("MIC1"),
336 SND_SOC_DAPM_INPUT("MIC2"),
337 SND_SOC_DAPM_VMID("Vmid"),
338 };
339 
340 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
341 static const struct soc_enum alc5623_amp_enum =
342  SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
343 static const struct snd_kcontrol_new alc5623_amp_mux_controls =
344  SOC_DAPM_ENUM("Route", alc5623_amp_enum);
345 
346 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
347 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
348  amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
349 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
350 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
351  &alc5623_amp_mux_controls),
352 };
353 
354 static const struct snd_soc_dapm_route intercon[] = {
355  /* virtual mixer - mixes left & right channels */
356  {"I2S Mix", NULL, "Left DAC"},
357  {"I2S Mix", NULL, "Right DAC"},
358  {"Line Mix", NULL, "Right LineIn"},
359  {"Line Mix", NULL, "Left LineIn"},
360  {"AuxI Mix", NULL, "Left AuxI"},
361  {"AuxI Mix", NULL, "Right AuxI"},
362  {"AUXOUTL", NULL, "Left AuxOut"},
363  {"AUXOUTR", NULL, "Right AuxOut"},
364 
365  /* HP mixer */
366  {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"},
367  {"HPL Mix", NULL, "HP Mix"},
368  {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"},
369  {"HPR Mix", NULL, "HP Mix"},
370  {"HP Mix", "LI2HP Playback Switch", "Line Mix"},
371  {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"},
372  {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
373  {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
374  {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"},
375 
376  /* speaker mixer */
377  {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
378  {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"},
379  {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
380  {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
381  {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"},
382 
383  /* mono mixer */
384  {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
385  {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
386  {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
387  {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
388  {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
389  {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
390  {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"},
391 
392  /* Left record mixer */
393  {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
394  {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
395  {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
396  {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
397  {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
398  {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
399  {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
400 
401  /*Right record mixer */
402  {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
403  {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"},
404  {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
405  {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
406  {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
407  {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
408  {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
409 
410  /* headphone left mux */
411  {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
412  {"Left Headphone Mux", "Vmid", "Vmid"},
413 
414  /* headphone right mux */
415  {"Right Headphone Mux", "HP Right Mix", "HPR Mix"},
416  {"Right Headphone Mux", "Vmid", "Vmid"},
417 
418  /* speaker out mux */
419  {"SpeakerOut Mux", "Vmid", "Vmid"},
420  {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"},
421  {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"},
422  {"SpeakerOut Mux", "Mono Mix", "Mono Mix"},
423 
424  /* Mono/Aux Out mux */
425  {"AuxOut Mux", "Vmid", "Vmid"},
426  {"AuxOut Mux", "HPOut Mix", "HPOut Mix"},
427  {"AuxOut Mux", "Speaker Mix", "Speaker Mix"},
428  {"AuxOut Mux", "Mono Mix", "Mono Mix"},
429 
430  /* output pga */
431  {"HPL", NULL, "Left Headphone"},
432  {"Left Headphone", NULL, "Left Headphone Mux"},
433  {"HPR", NULL, "Right Headphone"},
434  {"Right Headphone", NULL, "Right Headphone Mux"},
435  {"Left AuxOut", NULL, "AuxOut Mux"},
436  {"Right AuxOut", NULL, "AuxOut Mux"},
437 
438  /* input pga */
439  {"Left LineIn", NULL, "LINEINL"},
440  {"Right LineIn", NULL, "LINEINR"},
441  {"Left AuxI", NULL, "AUXINL"},
442  {"Right AuxI", NULL, "AUXINR"},
443  {"MIC1 Pre Amp", NULL, "MIC1"},
444  {"MIC2 Pre Amp", NULL, "MIC2"},
445  {"MIC1 PGA", NULL, "MIC1 Pre Amp"},
446  {"MIC2 PGA", NULL, "MIC2 Pre Amp"},
447 
448  /* left ADC */
449  {"Left ADC", NULL, "Left Capture Mix"},
450 
451  /* right ADC */
452  {"Right ADC", NULL, "Right Capture Mix"},
453 
454  {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"},
455  {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"},
456  {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"},
457  {"SpeakerOut N Mux", "Vmid", "Vmid"},
458 
459  {"SPKOUT", NULL, "SpeakerOut"},
460  {"SPKOUTN", NULL, "SpeakerOut N Mux"},
461 };
462 
463 static const struct snd_soc_dapm_route intercon_spk[] = {
464  {"SpeakerOut", NULL, "SpeakerOut Mux"},
465 };
466 
467 static const struct snd_soc_dapm_route intercon_amp_spk[] = {
468  {"AB Amp", NULL, "SpeakerOut Mux"},
469  {"D Amp", NULL, "SpeakerOut Mux"},
470  {"AB-D Amp Mux", "AB Amp", "AB Amp"},
471  {"AB-D Amp Mux", "D Amp", "D Amp"},
472  {"SpeakerOut", NULL, "AB-D Amp Mux"},
473 };
474 
475 /* PLL divisors */
476 struct _pll_div {
477  u32 pll_in;
478  u32 pll_out;
479  u16 regvalue;
480 };
481 
482 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
483 /* useful only for master mode */
484 static const struct _pll_div codec_master_pll_div[] = {
485 
486  { 2048000, 8192000, 0x0ea0},
487  { 3686400, 8192000, 0x4e27},
488  { 12000000, 8192000, 0x456b},
489  { 13000000, 8192000, 0x495f},
490  { 13100000, 8192000, 0x0320},
491  { 2048000, 11289600, 0xf637},
492  { 3686400, 11289600, 0x2f22},
493  { 12000000, 11289600, 0x3e2f},
494  { 13000000, 11289600, 0x4d5b},
495  { 13100000, 11289600, 0x363b},
496  { 2048000, 16384000, 0x1ea0},
497  { 3686400, 16384000, 0x9e27},
498  { 12000000, 16384000, 0x452b},
499  { 13000000, 16384000, 0x542f},
500  { 13100000, 16384000, 0x03a0},
501  { 2048000, 16934400, 0xe625},
502  { 3686400, 16934400, 0x9126},
503  { 12000000, 16934400, 0x4d2c},
504  { 13000000, 16934400, 0x742f},
505  { 13100000, 16934400, 0x3c27},
506  { 2048000, 22579200, 0x2aa0},
507  { 3686400, 22579200, 0x2f20},
508  { 12000000, 22579200, 0x7e2f},
509  { 13000000, 22579200, 0x742f},
510  { 13100000, 22579200, 0x3c27},
511  { 2048000, 24576000, 0x2ea0},
512  { 3686400, 24576000, 0xee27},
513  { 12000000, 24576000, 0x2915},
514  { 13000000, 24576000, 0x772e},
515  { 13100000, 24576000, 0x0d20},
516 };
517 
518 static const struct _pll_div codec_slave_pll_div[] = {
519 
520  { 1024000, 16384000, 0x3ea0},
521  { 1411200, 22579200, 0x3ea0},
522  { 1536000, 24576000, 0x3ea0},
523  { 2048000, 16384000, 0x1ea0},
524  { 2822400, 22579200, 0x1ea0},
525  { 3072000, 24576000, 0x1ea0},
526 
527 };
528 
529 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
530  int source, unsigned int freq_in, unsigned int freq_out)
531 {
532  int i;
533  struct snd_soc_codec *codec = codec_dai->codec;
534  int gbl_clk = 0, pll_div = 0;
535  u16 reg;
536 
537  if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
538  return -ENODEV;
539 
540  /* Disable PLL power */
541  snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
542  ALC5623_PWR_ADD2_PLL,
543  0);
544 
545  /* pll is not used in slave mode */
546  reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
547  if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
548  return 0;
549 
550  if (!freq_in || !freq_out)
551  return 0;
552 
553  switch (pll_id) {
554  case ALC5623_PLL_FR_MCLK:
555  for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
556  if (codec_master_pll_div[i].pll_in == freq_in
557  && codec_master_pll_div[i].pll_out == freq_out) {
558  /* PLL source from MCLK */
559  pll_div = codec_master_pll_div[i].regvalue;
560  break;
561  }
562  }
563  break;
564  case ALC5623_PLL_FR_BCK:
565  for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
566  if (codec_slave_pll_div[i].pll_in == freq_in
567  && codec_slave_pll_div[i].pll_out == freq_out) {
568  /* PLL source from Bitclk */
569  gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
570  pll_div = codec_slave_pll_div[i].regvalue;
571  break;
572  }
573  }
574  break;
575  default:
576  return -EINVAL;
577  }
578 
579  if (!pll_div)
580  return -EINVAL;
581 
582  snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
583  snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
584  snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
585  ALC5623_PWR_ADD2_PLL,
586  ALC5623_PWR_ADD2_PLL);
587  gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
588  snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
589 
590  return 0;
591 }
592 
593 struct _coeff_div {
594  u16 fs;
595  u16 regvalue;
596 };
597 
598 /* codec hifi mclk (after PLL) clock divider coefficients */
599 /* values inspired from column BCLK=32Fs of Appendix A table */
600 static const struct _coeff_div coeff_div[] = {
601  {256*8, 0x3a69},
602  {384*8, 0x3c6b},
603  {256*4, 0x2a69},
604  {384*4, 0x2c6b},
605  {256*2, 0x1a69},
606  {384*2, 0x1c6b},
607  {256*1, 0x0a69},
608  {384*1, 0x0c6b},
609 };
610 
611 static int get_coeff(struct snd_soc_codec *codec, int rate)
612 {
613  struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
614  int i;
615 
616  for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
617  if (coeff_div[i].fs * rate == alc5623->sysclk)
618  return i;
619  }
620  return -EINVAL;
621 }
622 
623 /*
624  * Clock after PLL and dividers
625  */
626 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
627  int clk_id, unsigned int freq, int dir)
628 {
629  struct snd_soc_codec *codec = codec_dai->codec;
630  struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
631 
632  switch (freq) {
633  case 8192000:
634  case 11289600:
635  case 12288000:
636  case 16384000:
637  case 16934400:
638  case 18432000:
639  case 22579200:
640  case 24576000:
641  alc5623->sysclk = freq;
642  return 0;
643  }
644  return -EINVAL;
645 }
646 
647 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
648  unsigned int fmt)
649 {
650  struct snd_soc_codec *codec = codec_dai->codec;
651  u16 iface = 0;
652 
653  /* set master/slave audio interface */
654  switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
655  case SND_SOC_DAIFMT_CBM_CFM:
656  iface = ALC5623_DAI_SDP_MASTER_MODE;
657  break;
658  case SND_SOC_DAIFMT_CBS_CFS:
659  iface = ALC5623_DAI_SDP_SLAVE_MODE;
660  break;
661  default:
662  return -EINVAL;
663  }
664 
665  /* interface format */
666  switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
667  case SND_SOC_DAIFMT_I2S:
668  iface |= ALC5623_DAI_I2S_DF_I2S;
669  break;
670  case SND_SOC_DAIFMT_RIGHT_J:
671  iface |= ALC5623_DAI_I2S_DF_RIGHT;
672  break;
673  case SND_SOC_DAIFMT_LEFT_J:
674  iface |= ALC5623_DAI_I2S_DF_LEFT;
675  break;
676  case SND_SOC_DAIFMT_DSP_A:
677  iface |= ALC5623_DAI_I2S_DF_PCM;
678  break;
679  case SND_SOC_DAIFMT_DSP_B:
680  iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
681  break;
682  default:
683  return -EINVAL;
684  }
685 
686  /* clock inversion */
687  switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
688  case SND_SOC_DAIFMT_NB_NF:
689  break;
690  case SND_SOC_DAIFMT_IB_IF:
691  iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
692  break;
693  case SND_SOC_DAIFMT_IB_NF:
694  iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
695  break;
696  case SND_SOC_DAIFMT_NB_IF:
697  break;
698  default:
699  return -EINVAL;
700  }
701 
702  return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
703 }
704 
705 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
706  struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
707 {
708  struct snd_soc_codec *codec = dai->codec;
709  struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
710  int coeff, rate;
711  u16 iface;
712 
713  iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
714  iface &= ~ALC5623_DAI_I2S_DL_MASK;
715 
716  /* bit size */
717  switch (params_format(params)) {
718  case SNDRV_PCM_FORMAT_S16_LE:
719  iface |= ALC5623_DAI_I2S_DL_16;
720  break;
721  case SNDRV_PCM_FORMAT_S20_3LE:
722  iface |= ALC5623_DAI_I2S_DL_20;
723  break;
724  case SNDRV_PCM_FORMAT_S24_LE:
725  iface |= ALC5623_DAI_I2S_DL_24;
726  break;
727  case SNDRV_PCM_FORMAT_S32_LE:
728  iface |= ALC5623_DAI_I2S_DL_32;
729  break;
730  default:
731  return -EINVAL;
732  }
733 
734  /* set iface & srate */
735  snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
736  rate = params_rate(params);
737  coeff = get_coeff(codec, rate);
738  if (coeff < 0)
739  return -EINVAL;
740 
741  coeff = coeff_div[coeff].regvalue;
742  dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
743  __func__, alc5623->sysclk, rate, coeff);
744  snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
745 
746  return 0;
747 }
748 
749 static int alc5623_mute(struct snd_soc_dai *dai, int mute)
750 {
751  struct snd_soc_codec *codec = dai->codec;
752  u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
753  u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
754 
755  if (mute)
756  mute_reg |= hp_mute;
757 
758  return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
759 }
760 
761 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
762  | ALC5623_PWR_ADD2_DAC_REF_CIR)
763 
764 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
765  | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
766 
767 #define ALC5623_ADD1_POWER_EN \
768  (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
769  | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
770  | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
771 
772 #define ALC5623_ADD1_POWER_EN_5622 \
773  (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
774  | ALC5623_PWR_ADD1_HP_OUT_AMP)
775 
776 static void enable_power_depop(struct snd_soc_codec *codec)
777 {
778  struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
779 
780  snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
781  ALC5623_PWR_ADD1_SOFTGEN_EN,
782  ALC5623_PWR_ADD1_SOFTGEN_EN);
783 
784  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
785 
786  snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
787  ALC5623_MISC_HP_DEPOP_MODE2_EN,
788  ALC5623_MISC_HP_DEPOP_MODE2_EN);
789 
790  msleep(500);
791 
792  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
793 
794  /* avoid writing '1' into 5622 reserved bits */
795  if (alc5623->id == 0x22)
796  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
797  ALC5623_ADD1_POWER_EN_5622);
798  else
799  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
800  ALC5623_ADD1_POWER_EN);
801 
802  /* disable HP Depop2 */
803  snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
804  ALC5623_MISC_HP_DEPOP_MODE2_EN,
805  0);
806 
807 }
808 
809 static int alc5623_set_bias_level(struct snd_soc_codec *codec,
810  enum snd_soc_bias_level level)
811 {
812  switch (level) {
813  case SND_SOC_BIAS_ON:
814  enable_power_depop(codec);
815  break;
816  case SND_SOC_BIAS_PREPARE:
817  break;
818  case SND_SOC_BIAS_STANDBY:
819  /* everything off except vref/vmid, */
820  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
821  ALC5623_PWR_ADD2_VREF);
822  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
823  ALC5623_PWR_ADD3_MAIN_BIAS);
824  break;
825  case SND_SOC_BIAS_OFF:
826  /* everything off, dac mute, inactive */
827  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
828  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
829  snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
830  break;
831  }
832  codec->dapm.bias_level = level;
833  return 0;
834 }
835 
836 #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
837  | SNDRV_PCM_FMTBIT_S24_LE \
838  | SNDRV_PCM_FMTBIT_S32_LE)
839 
840 static const struct snd_soc_dai_ops alc5623_dai_ops = {
841  .hw_params = alc5623_pcm_hw_params,
842  .digital_mute = alc5623_mute,
843  .set_fmt = alc5623_set_dai_fmt,
844  .set_sysclk = alc5623_set_dai_sysclk,
845  .set_pll = alc5623_set_dai_pll,
846 };
847 
848 static struct snd_soc_dai_driver alc5623_dai = {
849  .name = "alc5623-hifi",
850  .playback = {
851  .stream_name = "Playback",
852  .channels_min = 1,
853  .channels_max = 2,
854  .rate_min = 8000,
855  .rate_max = 48000,
856  .rates = SNDRV_PCM_RATE_8000_48000,
857  .formats = ALC5623_FORMATS,},
858  .capture = {
859  .stream_name = "Capture",
860  .channels_min = 1,
861  .channels_max = 2,
862  .rate_min = 8000,
863  .rate_max = 48000,
864  .rates = SNDRV_PCM_RATE_8000_48000,
865  .formats = ALC5623_FORMATS,},
866 
867  .ops = &alc5623_dai_ops,
868 };
869 
870 static int alc5623_suspend(struct snd_soc_codec *codec)
871 {
872  alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
873  return 0;
874 }
875 
876 static int alc5623_resume(struct snd_soc_codec *codec)
877 {
878  int i, step = codec->driver->reg_cache_step;
879  u16 *cache = codec->reg_cache;
880 
881  /* Sync reg_cache with the hardware */
882  for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
883  snd_soc_write(codec, i, cache[i]);
884 
885  alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
886 
887  /* charge alc5623 caps */
888  if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
889  alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
890  codec->dapm.bias_level = SND_SOC_BIAS_ON;
891  alc5623_set_bias_level(codec, codec->dapm.bias_level);
892  }
893 
894  return 0;
895 }
896 
897 static int alc5623_probe(struct snd_soc_codec *codec)
898 {
899  struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
900  struct snd_soc_dapm_context *dapm = &codec->dapm;
901  int ret;
902 
903  ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
904  if (ret < 0) {
905  dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
906  return ret;
907  }
908 
909  alc5623_reset(codec);
910  alc5623_fill_cache(codec);
911 
912  /* power on device */
913  alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
914 
915  if (alc5623->add_ctrl) {
916  snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
917  alc5623->add_ctrl);
918  }
919 
920  if (alc5623->jack_det_ctrl) {
921  snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
922  alc5623->jack_det_ctrl);
923  }
924 
925  switch (alc5623->id) {
926  case 0x21:
927  snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls,
928  ARRAY_SIZE(alc5621_vol_snd_controls));
929  break;
930  case 0x22:
931  snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls,
932  ARRAY_SIZE(alc5622_vol_snd_controls));
933  break;
934  case 0x23:
935  snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls,
936  ARRAY_SIZE(alc5623_vol_snd_controls));
937  break;
938  default:
939  return -EINVAL;
940  }
941 
942  snd_soc_add_codec_controls(codec, alc5623_snd_controls,
943  ARRAY_SIZE(alc5623_snd_controls));
944 
945  snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
946  ARRAY_SIZE(alc5623_dapm_widgets));
947 
948  /* set up audio path interconnects */
949  snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
950 
951  switch (alc5623->id) {
952  case 0x21:
953  case 0x22:
954  snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
955  ARRAY_SIZE(alc5623_dapm_amp_widgets));
956  snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
957  ARRAY_SIZE(intercon_amp_spk));
958  break;
959  case 0x23:
960  snd_soc_dapm_add_routes(dapm, intercon_spk,
961  ARRAY_SIZE(intercon_spk));
962  break;
963  default:
964  return -EINVAL;
965  }
966 
967  return ret;
968 }
969 
970 /* power down chip */
971 static int alc5623_remove(struct snd_soc_codec *codec)
972 {
973  alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
974  return 0;
975 }
976 
977 static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
978  .probe = alc5623_probe,
979  .remove = alc5623_remove,
980  .suspend = alc5623_suspend,
981  .resume = alc5623_resume,
982  .set_bias_level = alc5623_set_bias_level,
983  .reg_cache_size = ALC5623_VENDOR_ID2+2,
984  .reg_word_size = sizeof(u16),
985  .reg_cache_step = 2,
986 };
987 
988 /*
989  * ALC5623 2 wire address is determined by A1 pin
990  * state during powerup.
991  * low = 0x1a
992  * high = 0x1b
993  */
994 static __devinit int alc5623_i2c_probe(struct i2c_client *client,
995  const struct i2c_device_id *id)
996 {
997  struct alc5623_platform_data *pdata;
998  struct alc5623_priv *alc5623;
999  int ret, vid1, vid2;
1000 
1001  vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
1002  if (vid1 < 0) {
1003  dev_err(&client->dev, "failed to read I2C\n");
1004  return -EIO;
1005  }
1006  vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
1007 
1008  vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
1009  if (vid2 < 0) {
1010  dev_err(&client->dev, "failed to read I2C\n");
1011  return -EIO;
1012  }
1013 
1014  if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
1015  dev_err(&client->dev, "unknown or wrong codec\n");
1016  dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
1017  0x10ec, id->driver_data,
1018  vid1, vid2);
1019  return -ENODEV;
1020  }
1021 
1022  dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
1023 
1024  alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
1025  GFP_KERNEL);
1026  if (alc5623 == NULL)
1027  return -ENOMEM;
1028 
1029  pdata = client->dev.platform_data;
1030  if (pdata) {
1031  alc5623->add_ctrl = pdata->add_ctrl;
1032  alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
1033  }
1034 
1035  alc5623->id = vid2;
1036  switch (alc5623->id) {
1037  case 0x21:
1038  alc5623_dai.name = "alc5621-hifi";
1039  break;
1040  case 0x22:
1041  alc5623_dai.name = "alc5622-hifi";
1042  break;
1043  case 0x23:
1044  alc5623_dai.name = "alc5623-hifi";
1045  break;
1046  default:
1047  return -EINVAL;
1048  }
1049 
1050  i2c_set_clientdata(client, alc5623);
1051  alc5623->control_type = SND_SOC_I2C;
1052 
1053  ret = snd_soc_register_codec(&client->dev,
1054  &soc_codec_device_alc5623, &alc5623_dai, 1);
1055  if (ret != 0)
1056  dev_err(&client->dev, "Failed to register codec: %d\n", ret);
1057 
1058  return ret;
1059 }
1060 
1061 static __devexit int alc5623_i2c_remove(struct i2c_client *client)
1062 {
1063  snd_soc_unregister_codec(&client->dev);
1064  return 0;
1065 }
1066 
1067 static const struct i2c_device_id alc5623_i2c_table[] = {
1068  {"alc5621", 0x21},
1069  {"alc5622", 0x22},
1070  {"alc5623", 0x23},
1071  {}
1072 };
1073 MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
1074 
1075 /* i2c codec control layer */
1076 static struct i2c_driver alc5623_i2c_driver = {
1077  .driver = {
1078  .name = "alc562x-codec",
1079  .owner = THIS_MODULE,
1080  },
1081  .probe = alc5623_i2c_probe,
1082  .remove = __devexit_p(alc5623_i2c_remove),
1083  .id_table = alc5623_i2c_table,
1084 };
1085 
1086 module_i2c_driver(alc5623_i2c_driver);
1087 
1088 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1089 MODULE_AUTHOR("Arnaud Patard <[email protected]>");
1090 MODULE_LICENSE("GPL");
Generated on Thu Jan 10 2013 15:07:34 for Linux Kernel by   doxygen 1.8.2