Merge tag 'asoc-v3.15-5' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie...
[deliverable/linux.git] / sound / soc / codecs / alc5623.c
1 /*
2 * alc5623.c -- alc562[123] ALSA Soc Audio driver
3 *
4 * Copyright 2008 Realtek Microelectronics
5 * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
6 *
7 * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
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/regmap.h>
25 #include <linux/slab.h>
26 #include <sound/core.h>
27 #include <sound/pcm.h>
28 #include <sound/pcm_params.h>
29 #include <sound/tlv.h>
30 #include <sound/soc.h>
31 #include <sound/initval.h>
32 #include <sound/alc5623.h>
33
34 #include "alc5623.h"
35
36 static int caps_charge = 2000;
37 module_param(caps_charge, int, 0);
38 MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
39
40 /* codec private data */
41 struct alc5623_priv {
42 struct regmap *regmap;
43 u8 id;
44 unsigned int sysclk;
45 unsigned int add_ctrl;
46 unsigned int jack_det_ctrl;
47 };
48
49 static inline int alc5623_reset(struct snd_soc_codec *codec)
50 {
51 return snd_soc_write(codec, ALC5623_RESET, 0);
52 }
53
54 static int amp_mixer_event(struct snd_soc_dapm_widget *w,
55 struct snd_kcontrol *kcontrol, int event)
56 {
57 /* to power-on/off class-d amp generators/speaker */
58 /* need to write to 'index-46h' register : */
59 /* so write index num (here 0x46) to reg 0x6a */
60 /* and then 0xffff/0 to reg 0x6c */
61 snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
62
63 switch (event) {
64 case SND_SOC_DAPM_PRE_PMU:
65 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
66 break;
67 case SND_SOC_DAPM_POST_PMD:
68 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
69 break;
70 }
71
72 return 0;
73 }
74
75 /*
76 * ALC5623 Controls
77 */
78
79 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
80 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
81 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
82 static const unsigned int boost_tlv[] = {
83 TLV_DB_RANGE_HEAD(3),
84 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
85 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
86 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
87 };
88 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
89
90 static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = {
91 SOC_DOUBLE_TLV("Speaker Playback Volume",
92 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
93 SOC_DOUBLE("Speaker Playback Switch",
94 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
95 SOC_DOUBLE_TLV("Headphone Playback Volume",
96 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
97 SOC_DOUBLE("Headphone Playback Switch",
98 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
99 };
100
101 static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = {
102 SOC_DOUBLE_TLV("Speaker Playback Volume",
103 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
104 SOC_DOUBLE("Speaker Playback Switch",
105 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
106 SOC_DOUBLE_TLV("Line Playback Volume",
107 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
108 SOC_DOUBLE("Line Playback Switch",
109 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
110 };
111
112 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
113 SOC_DOUBLE_TLV("Line Playback Volume",
114 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
115 SOC_DOUBLE("Line Playback Switch",
116 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
117 SOC_DOUBLE_TLV("Headphone Playback Volume",
118 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
119 SOC_DOUBLE("Headphone Playback Switch",
120 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
121 };
122
123 static const struct snd_kcontrol_new alc5623_snd_controls[] = {
124 SOC_DOUBLE_TLV("Auxout Playback Volume",
125 ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
126 SOC_DOUBLE("Auxout Playback Switch",
127 ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
128 SOC_DOUBLE_TLV("PCM Playback Volume",
129 ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
130 SOC_DOUBLE_TLV("AuxI Capture Volume",
131 ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
132 SOC_DOUBLE_TLV("LineIn Capture Volume",
133 ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
134 SOC_SINGLE_TLV("Mic1 Capture Volume",
135 ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
136 SOC_SINGLE_TLV("Mic2 Capture Volume",
137 ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
138 SOC_DOUBLE_TLV("Rec Capture Volume",
139 ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
140 SOC_SINGLE_TLV("Mic 1 Boost Volume",
141 ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
142 SOC_SINGLE_TLV("Mic 2 Boost Volume",
143 ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
144 SOC_SINGLE_TLV("Digital Boost Volume",
145 ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
146 };
147
148 /*
149 * DAPM Controls
150 */
151 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
152 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
153 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
154 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
155 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
156 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
157 };
158
159 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
160 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
161 };
162
163 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
164 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
165 };
166
167 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
168 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
169 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
170 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
171 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
172 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
173 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
174 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
175 };
176
177 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
178 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
179 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
180 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
181 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
182 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
183 };
184
185 /* Left Record Mixer */
186 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
187 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
188 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
189 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
190 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
191 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
192 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
193 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
194 };
195
196 /* Right Record Mixer */
197 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
198 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
199 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
200 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
201 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
202 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
203 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
204 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
205 };
206
207 static const char *alc5623_spk_n_sour_sel[] = {
208 "RN/-R", "RP/+R", "LN/-R", "Vmid" };
209 static const char *alc5623_hpl_out_input_sel[] = {
210 "Vmid", "HP Left Mix"};
211 static const char *alc5623_hpr_out_input_sel[] = {
212 "Vmid", "HP Right Mix"};
213 static const char *alc5623_spkout_input_sel[] = {
214 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
215 static const char *alc5623_aux_out_input_sel[] = {
216 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
217
218 /* auxout output mux */
219 static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum,
220 ALC5623_OUTPUT_MIXER_CTRL, 6,
221 alc5623_aux_out_input_sel);
222 static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
223 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
224
225 /* speaker output mux */
226 static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum,
227 ALC5623_OUTPUT_MIXER_CTRL, 10,
228 alc5623_spkout_input_sel);
229 static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
230 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
231
232 /* headphone left output mux */
233 static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum,
234 ALC5623_OUTPUT_MIXER_CTRL, 9,
235 alc5623_hpl_out_input_sel);
236 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
237 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
238
239 /* headphone right output mux */
240 static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum,
241 ALC5623_OUTPUT_MIXER_CTRL, 8,
242 alc5623_hpr_out_input_sel);
243 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
244 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
245
246 /* speaker output N select */
247 static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum,
248 ALC5623_OUTPUT_MIXER_CTRL, 14,
249 alc5623_spk_n_sour_sel);
250 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
251 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
252
253 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
254 /* Muxes */
255 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
256 &alc5623_auxout_mux_controls),
257 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
258 &alc5623_spkout_mux_controls),
259 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
260 &alc5623_hpl_out_mux_controls),
261 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
262 &alc5623_hpr_out_mux_controls),
263 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
264 &alc5623_spkoutn_mux_controls),
265
266 /* output mixers */
267 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
268 &alc5623_hp_mixer_controls[0],
269 ARRAY_SIZE(alc5623_hp_mixer_controls)),
270 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
271 &alc5623_hpr_mixer_controls[0],
272 ARRAY_SIZE(alc5623_hpr_mixer_controls)),
273 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
274 &alc5623_hpl_mixer_controls[0],
275 ARRAY_SIZE(alc5623_hpl_mixer_controls)),
276 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
277 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
278 &alc5623_mono_mixer_controls[0],
279 ARRAY_SIZE(alc5623_mono_mixer_controls)),
280 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
281 &alc5623_speaker_mixer_controls[0],
282 ARRAY_SIZE(alc5623_speaker_mixer_controls)),
283
284 /* input mixers */
285 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
286 &alc5623_captureL_mixer_controls[0],
287 ARRAY_SIZE(alc5623_captureL_mixer_controls)),
288 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
289 &alc5623_captureR_mixer_controls[0],
290 ARRAY_SIZE(alc5623_captureR_mixer_controls)),
291
292 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
293 ALC5623_PWR_MANAG_ADD2, 9, 0),
294 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
295 ALC5623_PWR_MANAG_ADD2, 8, 0),
296 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
297 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
298 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
299 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
300 ALC5623_PWR_MANAG_ADD2, 7, 0),
301 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
302 ALC5623_PWR_MANAG_ADD2, 6, 0),
303 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
304 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
305 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
306 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
307 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
308 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
309 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
310 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
311 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
312 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
313 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
314 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
315 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
316 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
317
318 SND_SOC_DAPM_OUTPUT("AUXOUTL"),
319 SND_SOC_DAPM_OUTPUT("AUXOUTR"),
320 SND_SOC_DAPM_OUTPUT("HPL"),
321 SND_SOC_DAPM_OUTPUT("HPR"),
322 SND_SOC_DAPM_OUTPUT("SPKOUT"),
323 SND_SOC_DAPM_OUTPUT("SPKOUTN"),
324 SND_SOC_DAPM_INPUT("LINEINL"),
325 SND_SOC_DAPM_INPUT("LINEINR"),
326 SND_SOC_DAPM_INPUT("AUXINL"),
327 SND_SOC_DAPM_INPUT("AUXINR"),
328 SND_SOC_DAPM_INPUT("MIC1"),
329 SND_SOC_DAPM_INPUT("MIC2"),
330 SND_SOC_DAPM_VMID("Vmid"),
331 };
332
333 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
334 static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum,
335 ALC5623_OUTPUT_MIXER_CTRL, 13,
336 alc5623_amp_names);
337 static const struct snd_kcontrol_new alc5623_amp_mux_controls =
338 SOC_DAPM_ENUM("Route", alc5623_amp_enum);
339
340 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
341 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
342 amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
343 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
344 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
345 &alc5623_amp_mux_controls),
346 };
347
348 static const struct snd_soc_dapm_route intercon[] = {
349 /* virtual mixer - mixes left & right channels */
350 {"I2S Mix", NULL, "Left DAC"},
351 {"I2S Mix", NULL, "Right DAC"},
352 {"Line Mix", NULL, "Right LineIn"},
353 {"Line Mix", NULL, "Left LineIn"},
354 {"AuxI Mix", NULL, "Left AuxI"},
355 {"AuxI Mix", NULL, "Right AuxI"},
356 {"AUXOUTL", NULL, "Left AuxOut"},
357 {"AUXOUTR", NULL, "Right AuxOut"},
358
359 /* HP mixer */
360 {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"},
361 {"HPL Mix", NULL, "HP Mix"},
362 {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"},
363 {"HPR Mix", NULL, "HP Mix"},
364 {"HP Mix", "LI2HP Playback Switch", "Line Mix"},
365 {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"},
366 {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
367 {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
368 {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"},
369
370 /* speaker mixer */
371 {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
372 {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"},
373 {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
374 {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
375 {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"},
376
377 /* mono mixer */
378 {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
379 {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
380 {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
381 {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
382 {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
383 {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
384 {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"},
385
386 /* Left record mixer */
387 {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
388 {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
389 {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
390 {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
391 {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
392 {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
393 {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
394
395 /*Right record mixer */
396 {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
397 {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"},
398 {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
399 {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
400 {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
401 {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
402 {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
403
404 /* headphone left mux */
405 {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
406 {"Left Headphone Mux", "Vmid", "Vmid"},
407
408 /* headphone right mux */
409 {"Right Headphone Mux", "HP Right Mix", "HPR Mix"},
410 {"Right Headphone Mux", "Vmid", "Vmid"},
411
412 /* speaker out mux */
413 {"SpeakerOut Mux", "Vmid", "Vmid"},
414 {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"},
415 {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"},
416 {"SpeakerOut Mux", "Mono Mix", "Mono Mix"},
417
418 /* Mono/Aux Out mux */
419 {"AuxOut Mux", "Vmid", "Vmid"},
420 {"AuxOut Mux", "HPOut Mix", "HPOut Mix"},
421 {"AuxOut Mux", "Speaker Mix", "Speaker Mix"},
422 {"AuxOut Mux", "Mono Mix", "Mono Mix"},
423
424 /* output pga */
425 {"HPL", NULL, "Left Headphone"},
426 {"Left Headphone", NULL, "Left Headphone Mux"},
427 {"HPR", NULL, "Right Headphone"},
428 {"Right Headphone", NULL, "Right Headphone Mux"},
429 {"Left AuxOut", NULL, "AuxOut Mux"},
430 {"Right AuxOut", NULL, "AuxOut Mux"},
431
432 /* input pga */
433 {"Left LineIn", NULL, "LINEINL"},
434 {"Right LineIn", NULL, "LINEINR"},
435 {"Left AuxI", NULL, "AUXINL"},
436 {"Right AuxI", NULL, "AUXINR"},
437 {"MIC1 Pre Amp", NULL, "MIC1"},
438 {"MIC2 Pre Amp", NULL, "MIC2"},
439 {"MIC1 PGA", NULL, "MIC1 Pre Amp"},
440 {"MIC2 PGA", NULL, "MIC2 Pre Amp"},
441
442 /* left ADC */
443 {"Left ADC", NULL, "Left Capture Mix"},
444
445 /* right ADC */
446 {"Right ADC", NULL, "Right Capture Mix"},
447
448 {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"},
449 {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"},
450 {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"},
451 {"SpeakerOut N Mux", "Vmid", "Vmid"},
452
453 {"SPKOUT", NULL, "SpeakerOut"},
454 {"SPKOUTN", NULL, "SpeakerOut N Mux"},
455 };
456
457 static const struct snd_soc_dapm_route intercon_spk[] = {
458 {"SpeakerOut", NULL, "SpeakerOut Mux"},
459 };
460
461 static const struct snd_soc_dapm_route intercon_amp_spk[] = {
462 {"AB Amp", NULL, "SpeakerOut Mux"},
463 {"D Amp", NULL, "SpeakerOut Mux"},
464 {"AB-D Amp Mux", "AB Amp", "AB Amp"},
465 {"AB-D Amp Mux", "D Amp", "D Amp"},
466 {"SpeakerOut", NULL, "AB-D Amp Mux"},
467 };
468
469 /* PLL divisors */
470 struct _pll_div {
471 u32 pll_in;
472 u32 pll_out;
473 u16 regvalue;
474 };
475
476 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
477 /* useful only for master mode */
478 static const struct _pll_div codec_master_pll_div[] = {
479
480 { 2048000, 8192000, 0x0ea0},
481 { 3686400, 8192000, 0x4e27},
482 { 12000000, 8192000, 0x456b},
483 { 13000000, 8192000, 0x495f},
484 { 13100000, 8192000, 0x0320},
485 { 2048000, 11289600, 0xf637},
486 { 3686400, 11289600, 0x2f22},
487 { 12000000, 11289600, 0x3e2f},
488 { 13000000, 11289600, 0x4d5b},
489 { 13100000, 11289600, 0x363b},
490 { 2048000, 16384000, 0x1ea0},
491 { 3686400, 16384000, 0x9e27},
492 { 12000000, 16384000, 0x452b},
493 { 13000000, 16384000, 0x542f},
494 { 13100000, 16384000, 0x03a0},
495 { 2048000, 16934400, 0xe625},
496 { 3686400, 16934400, 0x9126},
497 { 12000000, 16934400, 0x4d2c},
498 { 13000000, 16934400, 0x742f},
499 { 13100000, 16934400, 0x3c27},
500 { 2048000, 22579200, 0x2aa0},
501 { 3686400, 22579200, 0x2f20},
502 { 12000000, 22579200, 0x7e2f},
503 { 13000000, 22579200, 0x742f},
504 { 13100000, 22579200, 0x3c27},
505 { 2048000, 24576000, 0x2ea0},
506 { 3686400, 24576000, 0xee27},
507 { 12000000, 24576000, 0x2915},
508 { 13000000, 24576000, 0x772e},
509 { 13100000, 24576000, 0x0d20},
510 };
511
512 static const struct _pll_div codec_slave_pll_div[] = {
513
514 { 1024000, 16384000, 0x3ea0},
515 { 1411200, 22579200, 0x3ea0},
516 { 1536000, 24576000, 0x3ea0},
517 { 2048000, 16384000, 0x1ea0},
518 { 2822400, 22579200, 0x1ea0},
519 { 3072000, 24576000, 0x1ea0},
520
521 };
522
523 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
524 int source, unsigned int freq_in, unsigned int freq_out)
525 {
526 int i;
527 struct snd_soc_codec *codec = codec_dai->codec;
528 int gbl_clk = 0, pll_div = 0;
529 u16 reg;
530
531 if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
532 return -ENODEV;
533
534 /* Disable PLL power */
535 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
536 ALC5623_PWR_ADD2_PLL,
537 0);
538
539 /* pll is not used in slave mode */
540 reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
541 if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
542 return 0;
543
544 if (!freq_in || !freq_out)
545 return 0;
546
547 switch (pll_id) {
548 case ALC5623_PLL_FR_MCLK:
549 for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
550 if (codec_master_pll_div[i].pll_in == freq_in
551 && codec_master_pll_div[i].pll_out == freq_out) {
552 /* PLL source from MCLK */
553 pll_div = codec_master_pll_div[i].regvalue;
554 break;
555 }
556 }
557 break;
558 case ALC5623_PLL_FR_BCK:
559 for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
560 if (codec_slave_pll_div[i].pll_in == freq_in
561 && codec_slave_pll_div[i].pll_out == freq_out) {
562 /* PLL source from Bitclk */
563 gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
564 pll_div = codec_slave_pll_div[i].regvalue;
565 break;
566 }
567 }
568 break;
569 default:
570 return -EINVAL;
571 }
572
573 if (!pll_div)
574 return -EINVAL;
575
576 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
577 snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
578 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
579 ALC5623_PWR_ADD2_PLL,
580 ALC5623_PWR_ADD2_PLL);
581 gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
582 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
583
584 return 0;
585 }
586
587 struct _coeff_div {
588 u16 fs;
589 u16 regvalue;
590 };
591
592 /* codec hifi mclk (after PLL) clock divider coefficients */
593 /* values inspired from column BCLK=32Fs of Appendix A table */
594 static const struct _coeff_div coeff_div[] = {
595 {256*8, 0x3a69},
596 {384*8, 0x3c6b},
597 {256*4, 0x2a69},
598 {384*4, 0x2c6b},
599 {256*2, 0x1a69},
600 {384*2, 0x1c6b},
601 {256*1, 0x0a69},
602 {384*1, 0x0c6b},
603 };
604
605 static int get_coeff(struct snd_soc_codec *codec, int rate)
606 {
607 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
608 int i;
609
610 for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
611 if (coeff_div[i].fs * rate == alc5623->sysclk)
612 return i;
613 }
614 return -EINVAL;
615 }
616
617 /*
618 * Clock after PLL and dividers
619 */
620 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
621 int clk_id, unsigned int freq, int dir)
622 {
623 struct snd_soc_codec *codec = codec_dai->codec;
624 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
625
626 switch (freq) {
627 case 8192000:
628 case 11289600:
629 case 12288000:
630 case 16384000:
631 case 16934400:
632 case 18432000:
633 case 22579200:
634 case 24576000:
635 alc5623->sysclk = freq;
636 return 0;
637 }
638 return -EINVAL;
639 }
640
641 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
642 unsigned int fmt)
643 {
644 struct snd_soc_codec *codec = codec_dai->codec;
645 u16 iface = 0;
646
647 /* set master/slave audio interface */
648 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
649 case SND_SOC_DAIFMT_CBM_CFM:
650 iface = ALC5623_DAI_SDP_MASTER_MODE;
651 break;
652 case SND_SOC_DAIFMT_CBS_CFS:
653 iface = ALC5623_DAI_SDP_SLAVE_MODE;
654 break;
655 default:
656 return -EINVAL;
657 }
658
659 /* interface format */
660 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
661 case SND_SOC_DAIFMT_I2S:
662 iface |= ALC5623_DAI_I2S_DF_I2S;
663 break;
664 case SND_SOC_DAIFMT_RIGHT_J:
665 iface |= ALC5623_DAI_I2S_DF_RIGHT;
666 break;
667 case SND_SOC_DAIFMT_LEFT_J:
668 iface |= ALC5623_DAI_I2S_DF_LEFT;
669 break;
670 case SND_SOC_DAIFMT_DSP_A:
671 iface |= ALC5623_DAI_I2S_DF_PCM;
672 break;
673 case SND_SOC_DAIFMT_DSP_B:
674 iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
675 break;
676 default:
677 return -EINVAL;
678 }
679
680 /* clock inversion */
681 switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
682 case SND_SOC_DAIFMT_NB_NF:
683 break;
684 case SND_SOC_DAIFMT_IB_IF:
685 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
686 break;
687 case SND_SOC_DAIFMT_IB_NF:
688 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
689 break;
690 case SND_SOC_DAIFMT_NB_IF:
691 break;
692 default:
693 return -EINVAL;
694 }
695
696 return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
697 }
698
699 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
700 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
701 {
702 struct snd_soc_codec *codec = dai->codec;
703 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
704 int coeff, rate;
705 u16 iface;
706
707 iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
708 iface &= ~ALC5623_DAI_I2S_DL_MASK;
709
710 /* bit size */
711 switch (params_width(params)) {
712 case 16:
713 iface |= ALC5623_DAI_I2S_DL_16;
714 break;
715 case 20:
716 iface |= ALC5623_DAI_I2S_DL_20;
717 break;
718 case 24:
719 iface |= ALC5623_DAI_I2S_DL_24;
720 break;
721 case 32:
722 iface |= ALC5623_DAI_I2S_DL_32;
723 break;
724 default:
725 return -EINVAL;
726 }
727
728 /* set iface & srate */
729 snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
730 rate = params_rate(params);
731 coeff = get_coeff(codec, rate);
732 if (coeff < 0)
733 return -EINVAL;
734
735 coeff = coeff_div[coeff].regvalue;
736 dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
737 __func__, alc5623->sysclk, rate, coeff);
738 snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
739
740 return 0;
741 }
742
743 static int alc5623_mute(struct snd_soc_dai *dai, int mute)
744 {
745 struct snd_soc_codec *codec = dai->codec;
746 u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
747 u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
748
749 if (mute)
750 mute_reg |= hp_mute;
751
752 return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
753 }
754
755 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
756 | ALC5623_PWR_ADD2_DAC_REF_CIR)
757
758 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
759 | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
760
761 #define ALC5623_ADD1_POWER_EN \
762 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
763 | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
764 | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
765
766 #define ALC5623_ADD1_POWER_EN_5622 \
767 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
768 | ALC5623_PWR_ADD1_HP_OUT_AMP)
769
770 static void enable_power_depop(struct snd_soc_codec *codec)
771 {
772 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
773
774 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
775 ALC5623_PWR_ADD1_SOFTGEN_EN,
776 ALC5623_PWR_ADD1_SOFTGEN_EN);
777
778 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
779
780 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
781 ALC5623_MISC_HP_DEPOP_MODE2_EN,
782 ALC5623_MISC_HP_DEPOP_MODE2_EN);
783
784 msleep(500);
785
786 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
787
788 /* avoid writing '1' into 5622 reserved bits */
789 if (alc5623->id == 0x22)
790 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
791 ALC5623_ADD1_POWER_EN_5622);
792 else
793 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
794 ALC5623_ADD1_POWER_EN);
795
796 /* disable HP Depop2 */
797 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
798 ALC5623_MISC_HP_DEPOP_MODE2_EN,
799 0);
800
801 }
802
803 static int alc5623_set_bias_level(struct snd_soc_codec *codec,
804 enum snd_soc_bias_level level)
805 {
806 switch (level) {
807 case SND_SOC_BIAS_ON:
808 enable_power_depop(codec);
809 break;
810 case SND_SOC_BIAS_PREPARE:
811 break;
812 case SND_SOC_BIAS_STANDBY:
813 /* everything off except vref/vmid, */
814 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
815 ALC5623_PWR_ADD2_VREF);
816 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
817 ALC5623_PWR_ADD3_MAIN_BIAS);
818 break;
819 case SND_SOC_BIAS_OFF:
820 /* everything off, dac mute, inactive */
821 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
822 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
823 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
824 break;
825 }
826 codec->dapm.bias_level = level;
827 return 0;
828 }
829
830 #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
831 | SNDRV_PCM_FMTBIT_S24_LE \
832 | SNDRV_PCM_FMTBIT_S32_LE)
833
834 static const struct snd_soc_dai_ops alc5623_dai_ops = {
835 .hw_params = alc5623_pcm_hw_params,
836 .digital_mute = alc5623_mute,
837 .set_fmt = alc5623_set_dai_fmt,
838 .set_sysclk = alc5623_set_dai_sysclk,
839 .set_pll = alc5623_set_dai_pll,
840 };
841
842 static struct snd_soc_dai_driver alc5623_dai = {
843 .name = "alc5623-hifi",
844 .playback = {
845 .stream_name = "Playback",
846 .channels_min = 1,
847 .channels_max = 2,
848 .rate_min = 8000,
849 .rate_max = 48000,
850 .rates = SNDRV_PCM_RATE_8000_48000,
851 .formats = ALC5623_FORMATS,},
852 .capture = {
853 .stream_name = "Capture",
854 .channels_min = 1,
855 .channels_max = 2,
856 .rate_min = 8000,
857 .rate_max = 48000,
858 .rates = SNDRV_PCM_RATE_8000_48000,
859 .formats = ALC5623_FORMATS,},
860
861 .ops = &alc5623_dai_ops,
862 };
863
864 static int alc5623_suspend(struct snd_soc_codec *codec)
865 {
866 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
867
868 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
869 regcache_cache_only(alc5623->regmap, true);
870
871 return 0;
872 }
873
874 static int alc5623_resume(struct snd_soc_codec *codec)
875 {
876 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
877 int ret;
878
879 /* Sync reg_cache with the hardware */
880 regcache_cache_only(alc5623->regmap, false);
881 ret = regcache_sync(alc5623->regmap);
882 if (ret != 0) {
883 dev_err(codec->dev, "Failed to sync register cache: %d\n",
884 ret);
885 regcache_cache_only(alc5623->regmap, true);
886 return ret;
887 }
888
889 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
890
891 /* charge alc5623 caps */
892 if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
893 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
894 codec->dapm.bias_level = SND_SOC_BIAS_ON;
895 alc5623_set_bias_level(codec, codec->dapm.bias_level);
896 }
897
898 return 0;
899 }
900
901 static int alc5623_probe(struct snd_soc_codec *codec)
902 {
903 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
904 struct snd_soc_dapm_context *dapm = &codec->dapm;
905
906 alc5623_reset(codec);
907
908 /* power on device */
909 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
910
911 if (alc5623->add_ctrl) {
912 snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
913 alc5623->add_ctrl);
914 }
915
916 if (alc5623->jack_det_ctrl) {
917 snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
918 alc5623->jack_det_ctrl);
919 }
920
921 switch (alc5623->id) {
922 case 0x21:
923 snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls,
924 ARRAY_SIZE(alc5621_vol_snd_controls));
925 break;
926 case 0x22:
927 snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls,
928 ARRAY_SIZE(alc5622_vol_snd_controls));
929 break;
930 case 0x23:
931 snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls,
932 ARRAY_SIZE(alc5623_vol_snd_controls));
933 break;
934 default:
935 return -EINVAL;
936 }
937
938 snd_soc_add_codec_controls(codec, alc5623_snd_controls,
939 ARRAY_SIZE(alc5623_snd_controls));
940
941 snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
942 ARRAY_SIZE(alc5623_dapm_widgets));
943
944 /* set up audio path interconnects */
945 snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
946
947 switch (alc5623->id) {
948 case 0x21:
949 case 0x22:
950 snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
951 ARRAY_SIZE(alc5623_dapm_amp_widgets));
952 snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
953 ARRAY_SIZE(intercon_amp_spk));
954 break;
955 case 0x23:
956 snd_soc_dapm_add_routes(dapm, intercon_spk,
957 ARRAY_SIZE(intercon_spk));
958 break;
959 default:
960 return -EINVAL;
961 }
962
963 return 0;
964 }
965
966 /* power down chip */
967 static int alc5623_remove(struct snd_soc_codec *codec)
968 {
969 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
970 return 0;
971 }
972
973 static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
974 .probe = alc5623_probe,
975 .remove = alc5623_remove,
976 .suspend = alc5623_suspend,
977 .resume = alc5623_resume,
978 .set_bias_level = alc5623_set_bias_level,
979 };
980
981 static const struct regmap_config alc5623_regmap = {
982 .reg_bits = 8,
983 .val_bits = 16,
984 .reg_stride = 2,
985
986 .max_register = ALC5623_VENDOR_ID2,
987 .cache_type = REGCACHE_RBTREE,
988 };
989
990 /*
991 * ALC5623 2 wire address is determined by A1 pin
992 * state during powerup.
993 * low = 0x1a
994 * high = 0x1b
995 */
996 static int alc5623_i2c_probe(struct i2c_client *client,
997 const struct i2c_device_id *id)
998 {
999 struct alc5623_platform_data *pdata;
1000 struct alc5623_priv *alc5623;
1001 unsigned int vid1, vid2;
1002 int ret;
1003
1004 alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
1005 GFP_KERNEL);
1006 if (alc5623 == NULL)
1007 return -ENOMEM;
1008
1009 alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap);
1010 if (IS_ERR(alc5623->regmap)) {
1011 ret = PTR_ERR(alc5623->regmap);
1012 dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret);
1013 return ret;
1014 }
1015
1016 ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1);
1017 if (ret < 0) {
1018 dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
1019 return ret;
1020 }
1021 vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
1022
1023 ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
1024 if (ret < 0) {
1025 dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
1026 return ret;
1027 }
1028
1029 if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
1030 dev_err(&client->dev, "unknown or wrong codec\n");
1031 dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
1032 0x10ec, id->driver_data,
1033 vid1, vid2);
1034 return -ENODEV;
1035 }
1036
1037 dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
1038
1039 pdata = client->dev.platform_data;
1040 if (pdata) {
1041 alc5623->add_ctrl = pdata->add_ctrl;
1042 alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
1043 }
1044
1045 alc5623->id = vid2;
1046 switch (alc5623->id) {
1047 case 0x21:
1048 alc5623_dai.name = "alc5621-hifi";
1049 break;
1050 case 0x22:
1051 alc5623_dai.name = "alc5622-hifi";
1052 break;
1053 case 0x23:
1054 alc5623_dai.name = "alc5623-hifi";
1055 break;
1056 default:
1057 return -EINVAL;
1058 }
1059
1060 i2c_set_clientdata(client, alc5623);
1061
1062 ret = snd_soc_register_codec(&client->dev,
1063 &soc_codec_device_alc5623, &alc5623_dai, 1);
1064 if (ret != 0)
1065 dev_err(&client->dev, "Failed to register codec: %d\n", ret);
1066
1067 return ret;
1068 }
1069
1070 static int alc5623_i2c_remove(struct i2c_client *client)
1071 {
1072 snd_soc_unregister_codec(&client->dev);
1073 return 0;
1074 }
1075
1076 static const struct i2c_device_id alc5623_i2c_table[] = {
1077 {"alc5621", 0x21},
1078 {"alc5622", 0x22},
1079 {"alc5623", 0x23},
1080 {}
1081 };
1082 MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
1083
1084 /* i2c codec control layer */
1085 static struct i2c_driver alc5623_i2c_driver = {
1086 .driver = {
1087 .name = "alc562x-codec",
1088 .owner = THIS_MODULE,
1089 },
1090 .probe = alc5623_i2c_probe,
1091 .remove = alc5623_i2c_remove,
1092 .id_table = alc5623_i2c_table,
1093 };
1094
1095 module_i2c_driver(alc5623_i2c_driver);
1096
1097 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1098 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
1099 MODULE_LICENSE("GPL");
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