xref: /linux/sound/soc/codecs/alc5623.c (revision b85d45947951d23cb22d90caecf4c1eb81342c96)
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 <linux/of.h>
27 #include <sound/core.h>
28 #include <sound/pcm.h>
29 #include <sound/pcm_params.h>
30 #include <sound/tlv.h>
31 #include <sound/soc.h>
32 #include <sound/initval.h>
33 #include <sound/alc5623.h>
34 
35 #include "alc5623.h"
36 
37 static int caps_charge = 2000;
38 module_param(caps_charge, int, 0);
39 MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
40 
41 /* codec private data */
42 struct alc5623_priv {
43 	struct regmap *regmap;
44 	u8 id;
45 	unsigned int sysclk;
46 	unsigned int add_ctrl;
47 	unsigned int jack_det_ctrl;
48 };
49 
50 static inline int alc5623_reset(struct snd_soc_codec *codec)
51 {
52 	return snd_soc_write(codec, ALC5623_RESET, 0);
53 }
54 
55 static int amp_mixer_event(struct snd_soc_dapm_widget *w,
56 	struct snd_kcontrol *kcontrol, int event)
57 {
58 	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
59 
60 	/* to power-on/off class-d amp generators/speaker */
61 	/* need to write to 'index-46h' register :        */
62 	/* so write index num (here 0x46) to reg 0x6a     */
63 	/* and then 0xffff/0 to reg 0x6c                  */
64 	snd_soc_write(codec, ALC5623_HID_CTRL_INDEX, 0x46);
65 
66 	switch (event) {
67 	case SND_SOC_DAPM_PRE_PMU:
68 		snd_soc_write(codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
69 		break;
70 	case SND_SOC_DAPM_POST_PMD:
71 		snd_soc_write(codec, ALC5623_HID_CTRL_DATA, 0);
72 		break;
73 	}
74 
75 	return 0;
76 }
77 
78 /*
79  * ALC5623 Controls
80  */
81 
82 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
83 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
84 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
85 static const DECLARE_TLV_DB_RANGE(boost_tlv,
86 	0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
87 	1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
88 	2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0)
89 );
90 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
91 
92 static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = {
93 	SOC_DOUBLE_TLV("Speaker Playback Volume",
94 			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
95 	SOC_DOUBLE("Speaker Playback Switch",
96 			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
97 	SOC_DOUBLE_TLV("Headphone Playback Volume",
98 			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
99 	SOC_DOUBLE("Headphone Playback Switch",
100 			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
101 };
102 
103 static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = {
104 	SOC_DOUBLE_TLV("Speaker Playback Volume",
105 			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
106 	SOC_DOUBLE("Speaker Playback Switch",
107 			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
108 	SOC_DOUBLE_TLV("Line Playback Volume",
109 			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
110 	SOC_DOUBLE("Line Playback Switch",
111 			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
112 };
113 
114 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
115 	SOC_DOUBLE_TLV("Line Playback Volume",
116 			ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
117 	SOC_DOUBLE("Line Playback Switch",
118 			ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
119 	SOC_DOUBLE_TLV("Headphone Playback Volume",
120 			ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
121 	SOC_DOUBLE("Headphone Playback Switch",
122 			ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
123 };
124 
125 static const struct snd_kcontrol_new alc5623_snd_controls[] = {
126 	SOC_DOUBLE_TLV("Auxout Playback Volume",
127 			ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
128 	SOC_DOUBLE("Auxout Playback Switch",
129 			ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
130 	SOC_DOUBLE_TLV("PCM Playback Volume",
131 			ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
132 	SOC_DOUBLE_TLV("AuxI Capture Volume",
133 			ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
134 	SOC_DOUBLE_TLV("LineIn Capture Volume",
135 			ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
136 	SOC_SINGLE_TLV("Mic1 Capture Volume",
137 			ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
138 	SOC_SINGLE_TLV("Mic2 Capture Volume",
139 			ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
140 	SOC_DOUBLE_TLV("Rec Capture Volume",
141 			ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
142 	SOC_SINGLE_TLV("Mic 1 Boost Volume",
143 			ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
144 	SOC_SINGLE_TLV("Mic 2 Boost Volume",
145 			ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
146 	SOC_SINGLE_TLV("Digital Boost Volume",
147 			ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
148 };
149 
150 /*
151  * DAPM Controls
152  */
153 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
154 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
155 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
156 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
157 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
158 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
159 };
160 
161 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
162 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
163 };
164 
165 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
166 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
167 };
168 
169 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
170 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
171 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
172 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
173 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
174 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
175 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
176 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
177 };
178 
179 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
180 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
181 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
182 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
183 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
184 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
185 };
186 
187 /* Left Record Mixer */
188 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
189 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
190 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
191 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
192 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
193 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
194 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
195 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
196 };
197 
198 /* Right Record Mixer */
199 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
200 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
201 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
202 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
203 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
204 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
205 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
206 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
207 };
208 
209 static const char *alc5623_spk_n_sour_sel[] = {
210 		"RN/-R", "RP/+R", "LN/-R", "Vmid" };
211 static const char *alc5623_hpl_out_input_sel[] = {
212 		"Vmid", "HP Left Mix"};
213 static const char *alc5623_hpr_out_input_sel[] = {
214 		"Vmid", "HP Right Mix"};
215 static const char *alc5623_spkout_input_sel[] = {
216 		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
217 static const char *alc5623_aux_out_input_sel[] = {
218 		"Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
219 
220 /* auxout output mux */
221 static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum,
222 			    ALC5623_OUTPUT_MIXER_CTRL, 6,
223 			    alc5623_aux_out_input_sel);
224 static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
225 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
226 
227 /* speaker output mux */
228 static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum,
229 			    ALC5623_OUTPUT_MIXER_CTRL, 10,
230 			    alc5623_spkout_input_sel);
231 static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
232 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
233 
234 /* headphone left output mux */
235 static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum,
236 			    ALC5623_OUTPUT_MIXER_CTRL, 9,
237 			    alc5623_hpl_out_input_sel);
238 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
239 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
240 
241 /* headphone right output mux */
242 static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum,
243 			    ALC5623_OUTPUT_MIXER_CTRL, 8,
244 			    alc5623_hpr_out_input_sel);
245 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
246 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
247 
248 /* speaker output N select */
249 static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum,
250 			    ALC5623_OUTPUT_MIXER_CTRL, 14,
251 			    alc5623_spk_n_sour_sel);
252 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
253 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
254 
255 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
256 /* Muxes */
257 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
258 	&alc5623_auxout_mux_controls),
259 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
260 	&alc5623_spkout_mux_controls),
261 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
262 	&alc5623_hpl_out_mux_controls),
263 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
264 	&alc5623_hpr_out_mux_controls),
265 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
266 	&alc5623_spkoutn_mux_controls),
267 
268 /* output mixers */
269 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
270 	&alc5623_hp_mixer_controls[0],
271 	ARRAY_SIZE(alc5623_hp_mixer_controls)),
272 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
273 	&alc5623_hpr_mixer_controls[0],
274 	ARRAY_SIZE(alc5623_hpr_mixer_controls)),
275 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
276 	&alc5623_hpl_mixer_controls[0],
277 	ARRAY_SIZE(alc5623_hpl_mixer_controls)),
278 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
279 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
280 	&alc5623_mono_mixer_controls[0],
281 	ARRAY_SIZE(alc5623_mono_mixer_controls)),
282 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
283 	&alc5623_speaker_mixer_controls[0],
284 	ARRAY_SIZE(alc5623_speaker_mixer_controls)),
285 
286 /* input mixers */
287 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
288 	&alc5623_captureL_mixer_controls[0],
289 	ARRAY_SIZE(alc5623_captureL_mixer_controls)),
290 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
291 	&alc5623_captureR_mixer_controls[0],
292 	ARRAY_SIZE(alc5623_captureR_mixer_controls)),
293 
294 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
295 	ALC5623_PWR_MANAG_ADD2, 9, 0),
296 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
297 	ALC5623_PWR_MANAG_ADD2, 8, 0),
298 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
299 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
300 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
301 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
302 	ALC5623_PWR_MANAG_ADD2, 7, 0),
303 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
304 	ALC5623_PWR_MANAG_ADD2, 6, 0),
305 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
306 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
307 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
308 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
309 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
310 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
311 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
312 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
313 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
314 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
315 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
316 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
317 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
318 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
319 
320 SND_SOC_DAPM_OUTPUT("AUXOUTL"),
321 SND_SOC_DAPM_OUTPUT("AUXOUTR"),
322 SND_SOC_DAPM_OUTPUT("HPL"),
323 SND_SOC_DAPM_OUTPUT("HPR"),
324 SND_SOC_DAPM_OUTPUT("SPKOUT"),
325 SND_SOC_DAPM_OUTPUT("SPKOUTN"),
326 SND_SOC_DAPM_INPUT("LINEINL"),
327 SND_SOC_DAPM_INPUT("LINEINR"),
328 SND_SOC_DAPM_INPUT("AUXINL"),
329 SND_SOC_DAPM_INPUT("AUXINR"),
330 SND_SOC_DAPM_INPUT("MIC1"),
331 SND_SOC_DAPM_INPUT("MIC2"),
332 SND_SOC_DAPM_VMID("Vmid"),
333 };
334 
335 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
336 static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum,
337 			    ALC5623_OUTPUT_MIXER_CTRL, 13,
338 			    alc5623_amp_names);
339 static const struct snd_kcontrol_new alc5623_amp_mux_controls =
340 	SOC_DAPM_ENUM("Route", alc5623_amp_enum);
341 
342 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
343 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
344 	amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
345 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
346 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
347 	&alc5623_amp_mux_controls),
348 };
349 
350 static const struct snd_soc_dapm_route intercon[] = {
351 	/* virtual mixer - mixes left & right channels */
352 	{"I2S Mix", NULL,				"Left DAC"},
353 	{"I2S Mix", NULL,				"Right DAC"},
354 	{"Line Mix", NULL,				"Right LineIn"},
355 	{"Line Mix", NULL,				"Left LineIn"},
356 	{"AuxI Mix", NULL,				"Left AuxI"},
357 	{"AuxI Mix", NULL,				"Right AuxI"},
358 	{"AUXOUTL", NULL,				"Left AuxOut"},
359 	{"AUXOUTR", NULL,				"Right AuxOut"},
360 
361 	/* HP mixer */
362 	{"HPL Mix", "ADC2HP_L Playback Switch",		"Left Capture Mix"},
363 	{"HPL Mix", NULL,				"HP Mix"},
364 	{"HPR Mix", "ADC2HP_R Playback Switch",		"Right Capture Mix"},
365 	{"HPR Mix", NULL,				"HP Mix"},
366 	{"HP Mix", "LI2HP Playback Switch",		"Line Mix"},
367 	{"HP Mix", "AUXI2HP Playback Switch",		"AuxI Mix"},
368 	{"HP Mix", "MIC12HP Playback Switch",		"MIC1 PGA"},
369 	{"HP Mix", "MIC22HP Playback Switch",		"MIC2 PGA"},
370 	{"HP Mix", "DAC2HP Playback Switch",		"I2S Mix"},
371 
372 	/* speaker mixer */
373 	{"Speaker Mix", "LI2SPK Playback Switch",	"Line Mix"},
374 	{"Speaker Mix", "AUXI2SPK Playback Switch",	"AuxI Mix"},
375 	{"Speaker Mix", "MIC12SPK Playback Switch",	"MIC1 PGA"},
376 	{"Speaker Mix", "MIC22SPK Playback Switch",	"MIC2 PGA"},
377 	{"Speaker Mix", "DAC2SPK Playback Switch",	"I2S Mix"},
378 
379 	/* mono mixer */
380 	{"Mono Mix", "ADC2MONO_L Playback Switch",	"Left Capture Mix"},
381 	{"Mono Mix", "ADC2MONO_R Playback Switch",	"Right Capture Mix"},
382 	{"Mono Mix", "LI2MONO Playback Switch",		"Line Mix"},
383 	{"Mono Mix", "AUXI2MONO Playback Switch",	"AuxI Mix"},
384 	{"Mono Mix", "MIC12MONO Playback Switch",	"MIC1 PGA"},
385 	{"Mono Mix", "MIC22MONO Playback Switch",	"MIC2 PGA"},
386 	{"Mono Mix", "DAC2MONO Playback Switch",	"I2S Mix"},
387 
388 	/* Left record mixer */
389 	{"Left Capture Mix", "LineInL Capture Switch",	"LINEINL"},
390 	{"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
391 	{"Left Capture Mix", "Mic1 Capture Switch",	"MIC1 Pre Amp"},
392 	{"Left Capture Mix", "Mic2 Capture Switch",	"MIC2 Pre Amp"},
393 	{"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
394 	{"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
395 	{"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
396 
397 	/*Right record mixer */
398 	{"Right Capture Mix", "LineInR Capture Switch",	"LINEINR"},
399 	{"Right Capture Mix", "Right AuxI Capture Switch",	"AUXINR"},
400 	{"Right Capture Mix", "Mic1 Capture Switch",	"MIC1 Pre Amp"},
401 	{"Right Capture Mix", "Mic2 Capture Switch",	"MIC2 Pre Amp"},
402 	{"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
403 	{"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
404 	{"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
405 
406 	/* headphone left mux */
407 	{"Left Headphone Mux", "HP Left Mix",		"HPL Mix"},
408 	{"Left Headphone Mux", "Vmid",			"Vmid"},
409 
410 	/* headphone right mux */
411 	{"Right Headphone Mux", "HP Right Mix",		"HPR Mix"},
412 	{"Right Headphone Mux", "Vmid",			"Vmid"},
413 
414 	/* speaker out mux */
415 	{"SpeakerOut Mux", "Vmid",			"Vmid"},
416 	{"SpeakerOut Mux", "HPOut Mix",			"HPOut Mix"},
417 	{"SpeakerOut Mux", "Speaker Mix",		"Speaker Mix"},
418 	{"SpeakerOut Mux", "Mono Mix",			"Mono Mix"},
419 
420 	/* Mono/Aux Out mux */
421 	{"AuxOut Mux", "Vmid",				"Vmid"},
422 	{"AuxOut Mux", "HPOut Mix",			"HPOut Mix"},
423 	{"AuxOut Mux", "Speaker Mix",			"Speaker Mix"},
424 	{"AuxOut Mux", "Mono Mix",			"Mono Mix"},
425 
426 	/* output pga */
427 	{"HPL", NULL,					"Left Headphone"},
428 	{"Left Headphone", NULL,			"Left Headphone Mux"},
429 	{"HPR", NULL,					"Right Headphone"},
430 	{"Right Headphone", NULL,			"Right Headphone Mux"},
431 	{"Left AuxOut", NULL,				"AuxOut Mux"},
432 	{"Right AuxOut", NULL,				"AuxOut Mux"},
433 
434 	/* input pga */
435 	{"Left LineIn", NULL,				"LINEINL"},
436 	{"Right LineIn", NULL,				"LINEINR"},
437 	{"Left AuxI", NULL,				"AUXINL"},
438 	{"Right AuxI", NULL,				"AUXINR"},
439 	{"MIC1 Pre Amp", NULL,				"MIC1"},
440 	{"MIC2 Pre Amp", NULL,				"MIC2"},
441 	{"MIC1 PGA", NULL,				"MIC1 Pre Amp"},
442 	{"MIC2 PGA", NULL,				"MIC2 Pre Amp"},
443 
444 	/* left ADC */
445 	{"Left ADC", NULL,				"Left Capture Mix"},
446 
447 	/* right ADC */
448 	{"Right ADC", NULL,				"Right Capture Mix"},
449 
450 	{"SpeakerOut N Mux", "RN/-R",			"SpeakerOut"},
451 	{"SpeakerOut N Mux", "RP/+R",			"SpeakerOut"},
452 	{"SpeakerOut N Mux", "LN/-R",			"SpeakerOut"},
453 	{"SpeakerOut N Mux", "Vmid",			"Vmid"},
454 
455 	{"SPKOUT", NULL,				"SpeakerOut"},
456 	{"SPKOUTN", NULL,				"SpeakerOut N Mux"},
457 };
458 
459 static const struct snd_soc_dapm_route intercon_spk[] = {
460 	{"SpeakerOut", NULL,				"SpeakerOut Mux"},
461 };
462 
463 static const struct snd_soc_dapm_route intercon_amp_spk[] = {
464 	{"AB Amp", NULL,				"SpeakerOut Mux"},
465 	{"D Amp", NULL,					"SpeakerOut Mux"},
466 	{"AB-D Amp Mux", "AB Amp",			"AB Amp"},
467 	{"AB-D Amp Mux", "D Amp",			"D Amp"},
468 	{"SpeakerOut", NULL,				"AB-D Amp Mux"},
469 };
470 
471 /* PLL divisors */
472 struct _pll_div {
473 	u32 pll_in;
474 	u32 pll_out;
475 	u16 regvalue;
476 };
477 
478 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
479 /* useful only for master mode */
480 static const struct _pll_div codec_master_pll_div[] = {
481 
482 	{  2048000,  8192000,	0x0ea0},
483 	{  3686400,  8192000,	0x4e27},
484 	{ 12000000,  8192000,	0x456b},
485 	{ 13000000,  8192000,	0x495f},
486 	{ 13100000,  8192000,	0x0320},
487 	{  2048000,  11289600,	0xf637},
488 	{  3686400,  11289600,	0x2f22},
489 	{ 12000000,  11289600,	0x3e2f},
490 	{ 13000000,  11289600,	0x4d5b},
491 	{ 13100000,  11289600,	0x363b},
492 	{  2048000,  16384000,	0x1ea0},
493 	{  3686400,  16384000,	0x9e27},
494 	{ 12000000,  16384000,	0x452b},
495 	{ 13000000,  16384000,	0x542f},
496 	{ 13100000,  16384000,	0x03a0},
497 	{  2048000,  16934400,	0xe625},
498 	{  3686400,  16934400,	0x9126},
499 	{ 12000000,  16934400,	0x4d2c},
500 	{ 13000000,  16934400,	0x742f},
501 	{ 13100000,  16934400,	0x3c27},
502 	{  2048000,  22579200,	0x2aa0},
503 	{  3686400,  22579200,	0x2f20},
504 	{ 12000000,  22579200,	0x7e2f},
505 	{ 13000000,  22579200,	0x742f},
506 	{ 13100000,  22579200,	0x3c27},
507 	{  2048000,  24576000,	0x2ea0},
508 	{  3686400,  24576000,	0xee27},
509 	{ 12000000,  24576000,	0x2915},
510 	{ 13000000,  24576000,	0x772e},
511 	{ 13100000,  24576000,	0x0d20},
512 };
513 
514 static const struct _pll_div codec_slave_pll_div[] = {
515 
516 	{  1024000,  16384000,  0x3ea0},
517 	{  1411200,  22579200,	0x3ea0},
518 	{  1536000,  24576000,	0x3ea0},
519 	{  2048000,  16384000,  0x1ea0},
520 	{  2822400,  22579200,	0x1ea0},
521 	{  3072000,  24576000,	0x1ea0},
522 
523 };
524 
525 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
526 		int source, unsigned int freq_in, unsigned int freq_out)
527 {
528 	int i;
529 	struct snd_soc_codec *codec = codec_dai->codec;
530 	int gbl_clk = 0, pll_div = 0;
531 	u16 reg;
532 
533 	if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
534 		return -ENODEV;
535 
536 	/* Disable PLL power */
537 	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
538 				ALC5623_PWR_ADD2_PLL,
539 				0);
540 
541 	/* pll is not used in slave mode */
542 	reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
543 	if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
544 		return 0;
545 
546 	if (!freq_in || !freq_out)
547 		return 0;
548 
549 	switch (pll_id) {
550 	case ALC5623_PLL_FR_MCLK:
551 		for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
552 			if (codec_master_pll_div[i].pll_in == freq_in
553 			   && codec_master_pll_div[i].pll_out == freq_out) {
554 				/* PLL source from MCLK */
555 				pll_div  = codec_master_pll_div[i].regvalue;
556 				break;
557 			}
558 		}
559 		break;
560 	case ALC5623_PLL_FR_BCK:
561 		for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
562 			if (codec_slave_pll_div[i].pll_in == freq_in
563 			   && codec_slave_pll_div[i].pll_out == freq_out) {
564 				/* PLL source from Bitclk */
565 				gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
566 				pll_div = codec_slave_pll_div[i].regvalue;
567 				break;
568 			}
569 		}
570 		break;
571 	default:
572 		return -EINVAL;
573 	}
574 
575 	if (!pll_div)
576 		return -EINVAL;
577 
578 	snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
579 	snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
580 	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
581 				ALC5623_PWR_ADD2_PLL,
582 				ALC5623_PWR_ADD2_PLL);
583 	gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
584 	snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
585 
586 	return 0;
587 }
588 
589 struct _coeff_div {
590 	u16 fs;
591 	u16 regvalue;
592 };
593 
594 /* codec hifi mclk (after PLL) clock divider coefficients */
595 /* values inspired from column BCLK=32Fs of Appendix A table */
596 static const struct _coeff_div coeff_div[] = {
597 	{256*8, 0x3a69},
598 	{384*8, 0x3c6b},
599 	{256*4, 0x2a69},
600 	{384*4, 0x2c6b},
601 	{256*2, 0x1a69},
602 	{384*2, 0x1c6b},
603 	{256*1, 0x0a69},
604 	{384*1, 0x0c6b},
605 };
606 
607 static int get_coeff(struct snd_soc_codec *codec, int rate)
608 {
609 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
610 	int i;
611 
612 	for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
613 		if (coeff_div[i].fs * rate == alc5623->sysclk)
614 			return i;
615 	}
616 	return -EINVAL;
617 }
618 
619 /*
620  * Clock after PLL and dividers
621  */
622 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
623 		int clk_id, unsigned int freq, int dir)
624 {
625 	struct snd_soc_codec *codec = codec_dai->codec;
626 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
627 
628 	switch (freq) {
629 	case  8192000:
630 	case 11289600:
631 	case 12288000:
632 	case 16384000:
633 	case 16934400:
634 	case 18432000:
635 	case 22579200:
636 	case 24576000:
637 		alc5623->sysclk = freq;
638 		return 0;
639 	}
640 	return -EINVAL;
641 }
642 
643 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
644 		unsigned int fmt)
645 {
646 	struct snd_soc_codec *codec = codec_dai->codec;
647 	u16 iface = 0;
648 
649 	/* set master/slave audio interface */
650 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
651 	case SND_SOC_DAIFMT_CBM_CFM:
652 		iface = ALC5623_DAI_SDP_MASTER_MODE;
653 		break;
654 	case SND_SOC_DAIFMT_CBS_CFS:
655 		iface = ALC5623_DAI_SDP_SLAVE_MODE;
656 		break;
657 	default:
658 		return -EINVAL;
659 	}
660 
661 	/* interface format */
662 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
663 	case SND_SOC_DAIFMT_I2S:
664 		iface |= ALC5623_DAI_I2S_DF_I2S;
665 		break;
666 	case SND_SOC_DAIFMT_RIGHT_J:
667 		iface |= ALC5623_DAI_I2S_DF_RIGHT;
668 		break;
669 	case SND_SOC_DAIFMT_LEFT_J:
670 		iface |= ALC5623_DAI_I2S_DF_LEFT;
671 		break;
672 	case SND_SOC_DAIFMT_DSP_A:
673 		iface |= ALC5623_DAI_I2S_DF_PCM;
674 		break;
675 	case SND_SOC_DAIFMT_DSP_B:
676 		iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
677 		break;
678 	default:
679 		return -EINVAL;
680 	}
681 
682 	/* clock inversion */
683 	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
684 	case SND_SOC_DAIFMT_NB_NF:
685 		break;
686 	case SND_SOC_DAIFMT_IB_IF:
687 		iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
688 		break;
689 	case SND_SOC_DAIFMT_IB_NF:
690 		iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
691 		break;
692 	case SND_SOC_DAIFMT_NB_IF:
693 		break;
694 	default:
695 		return -EINVAL;
696 	}
697 
698 	return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
699 }
700 
701 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
702 		struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
703 {
704 	struct snd_soc_codec *codec = dai->codec;
705 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
706 	int coeff, rate;
707 	u16 iface;
708 
709 	iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
710 	iface &= ~ALC5623_DAI_I2S_DL_MASK;
711 
712 	/* bit size */
713 	switch (params_width(params)) {
714 	case 16:
715 		iface |= ALC5623_DAI_I2S_DL_16;
716 		break;
717 	case 20:
718 		iface |= ALC5623_DAI_I2S_DL_20;
719 		break;
720 	case 24:
721 		iface |= ALC5623_DAI_I2S_DL_24;
722 		break;
723 	case 32:
724 		iface |= ALC5623_DAI_I2S_DL_32;
725 		break;
726 	default:
727 		return -EINVAL;
728 	}
729 
730 	/* set iface & srate */
731 	snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
732 	rate = params_rate(params);
733 	coeff = get_coeff(codec, rate);
734 	if (coeff < 0)
735 		return -EINVAL;
736 
737 	coeff = coeff_div[coeff].regvalue;
738 	dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
739 		__func__, alc5623->sysclk, rate, coeff);
740 	snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
741 
742 	return 0;
743 }
744 
745 static int alc5623_mute(struct snd_soc_dai *dai, int mute)
746 {
747 	struct snd_soc_codec *codec = dai->codec;
748 	u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
749 	u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
750 
751 	if (mute)
752 		mute_reg |= hp_mute;
753 
754 	return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
755 }
756 
757 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
758 	| ALC5623_PWR_ADD2_DAC_REF_CIR)
759 
760 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
761 	| ALC5623_PWR_ADD3_MIC1_BOOST_AD)
762 
763 #define ALC5623_ADD1_POWER_EN \
764 	(ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
765 	| ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
766 	| ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
767 
768 #define ALC5623_ADD1_POWER_EN_5622 \
769 	(ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
770 	| ALC5623_PWR_ADD1_HP_OUT_AMP)
771 
772 static void enable_power_depop(struct snd_soc_codec *codec)
773 {
774 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
775 
776 	snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
777 				ALC5623_PWR_ADD1_SOFTGEN_EN,
778 				ALC5623_PWR_ADD1_SOFTGEN_EN);
779 
780 	snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
781 
782 	snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
783 				ALC5623_MISC_HP_DEPOP_MODE2_EN,
784 				ALC5623_MISC_HP_DEPOP_MODE2_EN);
785 
786 	msleep(500);
787 
788 	snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
789 
790 	/* avoid writing '1' into 5622 reserved bits */
791 	if (alc5623->id == 0x22)
792 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
793 			ALC5623_ADD1_POWER_EN_5622);
794 	else
795 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
796 			ALC5623_ADD1_POWER_EN);
797 
798 	/* disable HP Depop2 */
799 	snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
800 				ALC5623_MISC_HP_DEPOP_MODE2_EN,
801 				0);
802 
803 }
804 
805 static int alc5623_set_bias_level(struct snd_soc_codec *codec,
806 				      enum snd_soc_bias_level level)
807 {
808 	switch (level) {
809 	case SND_SOC_BIAS_ON:
810 		enable_power_depop(codec);
811 		break;
812 	case SND_SOC_BIAS_PREPARE:
813 		break;
814 	case SND_SOC_BIAS_STANDBY:
815 		/* everything off except vref/vmid, */
816 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
817 				ALC5623_PWR_ADD2_VREF);
818 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
819 				ALC5623_PWR_ADD3_MAIN_BIAS);
820 		break;
821 	case SND_SOC_BIAS_OFF:
822 		/* everything off, dac mute, inactive */
823 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
824 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
825 		snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
826 		break;
827 	}
828 	return 0;
829 }
830 
831 #define ALC5623_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE \
832 			| SNDRV_PCM_FMTBIT_S24_LE \
833 			| SNDRV_PCM_FMTBIT_S32_LE)
834 
835 static const struct snd_soc_dai_ops alc5623_dai_ops = {
836 		.hw_params = alc5623_pcm_hw_params,
837 		.digital_mute = alc5623_mute,
838 		.set_fmt = alc5623_set_dai_fmt,
839 		.set_sysclk = alc5623_set_dai_sysclk,
840 		.set_pll = alc5623_set_dai_pll,
841 };
842 
843 static struct snd_soc_dai_driver alc5623_dai = {
844 	.name = "alc5623-hifi",
845 	.playback = {
846 		.stream_name = "Playback",
847 		.channels_min = 1,
848 		.channels_max = 2,
849 		.rate_min =	8000,
850 		.rate_max =	48000,
851 		.rates = SNDRV_PCM_RATE_8000_48000,
852 		.formats = ALC5623_FORMATS,},
853 	.capture = {
854 		.stream_name = "Capture",
855 		.channels_min = 1,
856 		.channels_max = 2,
857 		.rate_min =	8000,
858 		.rate_max =	48000,
859 		.rates = SNDRV_PCM_RATE_8000_48000,
860 		.formats = ALC5623_FORMATS,},
861 
862 	.ops = &alc5623_dai_ops,
863 };
864 
865 static int alc5623_suspend(struct snd_soc_codec *codec)
866 {
867 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
868 
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 	return 0;
890 }
891 
892 static int alc5623_probe(struct snd_soc_codec *codec)
893 {
894 	struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
895 	struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
896 
897 	alc5623_reset(codec);
898 
899 	if (alc5623->add_ctrl) {
900 		snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
901 				alc5623->add_ctrl);
902 	}
903 
904 	if (alc5623->jack_det_ctrl) {
905 		snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
906 				alc5623->jack_det_ctrl);
907 	}
908 
909 	switch (alc5623->id) {
910 	case 0x21:
911 		snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls,
912 			ARRAY_SIZE(alc5621_vol_snd_controls));
913 		break;
914 	case 0x22:
915 		snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls,
916 			ARRAY_SIZE(alc5622_vol_snd_controls));
917 		break;
918 	case 0x23:
919 		snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls,
920 			ARRAY_SIZE(alc5623_vol_snd_controls));
921 		break;
922 	default:
923 		return -EINVAL;
924 	}
925 
926 	snd_soc_add_codec_controls(codec, alc5623_snd_controls,
927 			ARRAY_SIZE(alc5623_snd_controls));
928 
929 	snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
930 					ARRAY_SIZE(alc5623_dapm_widgets));
931 
932 	/* set up audio path interconnects */
933 	snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
934 
935 	switch (alc5623->id) {
936 	case 0x21:
937 	case 0x22:
938 		snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
939 					ARRAY_SIZE(alc5623_dapm_amp_widgets));
940 		snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
941 					ARRAY_SIZE(intercon_amp_spk));
942 		break;
943 	case 0x23:
944 		snd_soc_dapm_add_routes(dapm, intercon_spk,
945 					ARRAY_SIZE(intercon_spk));
946 		break;
947 	default:
948 		return -EINVAL;
949 	}
950 
951 	return 0;
952 }
953 
954 static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
955 	.probe = alc5623_probe,
956 	.suspend = alc5623_suspend,
957 	.resume = alc5623_resume,
958 	.set_bias_level = alc5623_set_bias_level,
959 	.suspend_bias_off = true,
960 };
961 
962 static const struct regmap_config alc5623_regmap = {
963 	.reg_bits = 8,
964 	.val_bits = 16,
965 	.reg_stride = 2,
966 
967 	.max_register = ALC5623_VENDOR_ID2,
968 	.cache_type = REGCACHE_RBTREE,
969 };
970 
971 /*
972  * ALC5623 2 wire address is determined by A1 pin
973  * state during powerup.
974  *    low  = 0x1a
975  *    high = 0x1b
976  */
977 static int alc5623_i2c_probe(struct i2c_client *client,
978 			     const struct i2c_device_id *id)
979 {
980 	struct alc5623_platform_data *pdata;
981 	struct alc5623_priv *alc5623;
982 	struct device_node *np;
983 	unsigned int vid1, vid2;
984 	int ret;
985 	u32 val32;
986 
987 	alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
988 			       GFP_KERNEL);
989 	if (alc5623 == NULL)
990 		return -ENOMEM;
991 
992 	alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap);
993 	if (IS_ERR(alc5623->regmap)) {
994 		ret = PTR_ERR(alc5623->regmap);
995 		dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret);
996 		return ret;
997 	}
998 
999 	ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1);
1000 	if (ret < 0) {
1001 		dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
1002 		return ret;
1003 	}
1004 
1005 	ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
1006 	if (ret < 0) {
1007 		dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
1008 		return ret;
1009 	}
1010 	vid2 >>= 8;
1011 
1012 	if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
1013 		dev_err(&client->dev, "unknown or wrong codec\n");
1014 		dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
1015 				0x10ec, id->driver_data,
1016 				vid1, vid2);
1017 		return -ENODEV;
1018 	}
1019 
1020 	dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
1021 
1022 	pdata = client->dev.platform_data;
1023 	if (pdata) {
1024 		alc5623->add_ctrl = pdata->add_ctrl;
1025 		alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
1026 	} else {
1027 		if (client->dev.of_node) {
1028 			np = client->dev.of_node;
1029 			ret = of_property_read_u32(np, "add-ctrl", &val32);
1030 			if (!ret)
1031 				alc5623->add_ctrl = val32;
1032 			ret = of_property_read_u32(np, "jack-det-ctrl", &val32);
1033 			if (!ret)
1034 				alc5623->jack_det_ctrl = val32;
1035 		}
1036 	}
1037 
1038 	alc5623->id = vid2;
1039 	switch (alc5623->id) {
1040 	case 0x21:
1041 		alc5623_dai.name = "alc5621-hifi";
1042 		break;
1043 	case 0x22:
1044 		alc5623_dai.name = "alc5622-hifi";
1045 		break;
1046 	case 0x23:
1047 		alc5623_dai.name = "alc5623-hifi";
1048 		break;
1049 	default:
1050 		return -EINVAL;
1051 	}
1052 
1053 	i2c_set_clientdata(client, alc5623);
1054 
1055 	ret =  snd_soc_register_codec(&client->dev,
1056 		&soc_codec_device_alc5623, &alc5623_dai, 1);
1057 	if (ret != 0)
1058 		dev_err(&client->dev, "Failed to register codec: %d\n", ret);
1059 
1060 	return ret;
1061 }
1062 
1063 static int alc5623_i2c_remove(struct i2c_client *client)
1064 {
1065 	snd_soc_unregister_codec(&client->dev);
1066 	return 0;
1067 }
1068 
1069 static const struct i2c_device_id alc5623_i2c_table[] = {
1070 	{"alc5621", 0x21},
1071 	{"alc5622", 0x22},
1072 	{"alc5623", 0x23},
1073 	{}
1074 };
1075 MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
1076 
1077 static const struct of_device_id alc5623_of_match[] = {
1078 	{ .compatible = "realtek,alc5623", },
1079 	{ }
1080 };
1081 MODULE_DEVICE_TABLE(of, alc5623_of_match);
1082 
1083 /*  i2c codec control layer */
1084 static struct i2c_driver alc5623_i2c_driver = {
1085 	.driver = {
1086 		.name = "alc562x-codec",
1087 		.of_match_table = of_match_ptr(alc5623_of_match),
1088 	},
1089 	.probe = alc5623_i2c_probe,
1090 	.remove =  alc5623_i2c_remove,
1091 	.id_table = alc5623_i2c_table,
1092 };
1093 
1094 module_i2c_driver(alc5623_i2c_driver);
1095 
1096 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1097 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
1098 MODULE_LICENSE("GPL");
1099