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TOMOYO Linux Cross Reference
Linux/sound/soc/codecs/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 <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/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 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 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 =  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 <arnaud.patard@rtp-net.org>");
1090 MODULE_LICENSE("GPL");
1091 

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