?  search:
summary | log | graphiclog1 | graphiclog2 | commit | commitdiff | tree | refs | edit | fork
blame | history | raw | HEAD
usbnet: increase URB reference count before usb_unlink_urb
[linux/fpc-iii.git] / sound / soc / codecs / alc5623.c
blobeecffb5489476dff195d04a9061fae68d37af638
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 <linux/platform_device.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 enum snd_soc_control_type control_type;
43 void *control_data;
44 struct mutex mutex;
45 u8 id;
46 unsigned int sysclk;
47 u16 reg_cache[ALC5623_VENDOR_ID2+2];
48 unsigned int add_ctrl;
49 unsigned int jack_det_ctrl;
50 };
51
52 static void alc5623_fill_cache(struct snd_soc_codec *codec)
53 {
54 int i, step = codec->driver->reg_cache_step;
55 u16 *cache = codec->reg_cache;
56
57 /* not really efficient ... */
58 for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
59 cache[i] = codec->hw_read(codec, i);
60 }
61
62 static inline int alc5623_reset(struct snd_soc_codec *codec)
63 {
64 return snd_soc_write(codec, ALC5623_RESET, 0);
65 }
66
67 static int amp_mixer_event(struct snd_soc_dapm_widget *w,
68 struct snd_kcontrol *kcontrol, int event)
69 {
70 /* to power-on/off class-d amp generators/speaker */
71 /* need to write to 'index-46h' register : */
72 /* so write index num (here 0x46) to reg 0x6a */
73 /* and then 0xffff/0 to reg 0x6c */
74 snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
75
76 switch (event) {
77 case SND_SOC_DAPM_PRE_PMU:
78 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
79 break;
80 case SND_SOC_DAPM_POST_PMD:
81 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
82 break;
83 }
84
85 return 0;
86 }
87
88 /*
89 * ALC5623 Controls
90 */
91
92 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
93 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
94 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
95 static const unsigned int boost_tlv[] = {
96 TLV_DB_RANGE_HEAD(3),
97 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
98 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
99 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
100 };
101 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
102
103 static const struct snd_kcontrol_new rt5621_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("Headphone Playback Volume",
109 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
110 SOC_DOUBLE("Headphone Playback Switch",
111 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
112 };
113
114 static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
115 SOC_DOUBLE_TLV("Speaker Playback Volume",
116 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
117 SOC_DOUBLE("Speaker Playback Switch",
118 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
119 SOC_DOUBLE_TLV("Line Playback Volume",
120 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
121 SOC_DOUBLE("Line Playback Switch",
122 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
123 };
124
125 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
126 SOC_DOUBLE_TLV("Line Playback Volume",
127 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
128 SOC_DOUBLE("Line Playback Switch",
129 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
130 SOC_DOUBLE_TLV("Headphone Playback Volume",
131 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
132 SOC_DOUBLE("Headphone Playback Switch",
133 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
134 };
135
136 static const struct snd_kcontrol_new alc5623_snd_controls[] = {
137 SOC_DOUBLE_TLV("Auxout Playback Volume",
138 ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
139 SOC_DOUBLE("Auxout Playback Switch",
140 ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
141 SOC_DOUBLE_TLV("PCM Playback Volume",
142 ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
143 SOC_DOUBLE_TLV("AuxI Capture Volume",
144 ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
145 SOC_DOUBLE_TLV("LineIn Capture Volume",
146 ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
147 SOC_SINGLE_TLV("Mic1 Capture Volume",
148 ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
149 SOC_SINGLE_TLV("Mic2 Capture Volume",
150 ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
151 SOC_DOUBLE_TLV("Rec Capture Volume",
152 ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
153 SOC_SINGLE_TLV("Mic 1 Boost Volume",
154 ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
155 SOC_SINGLE_TLV("Mic 2 Boost Volume",
156 ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
157 SOC_SINGLE_TLV("Digital Boost Volume",
158 ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
159 };
160
161 /*
162 * DAPM Controls
163 */
164 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
165 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
166 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
167 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
168 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
169 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
170 };
171
172 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
173 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
174 };
175
176 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
177 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
178 };
179
180 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
181 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
182 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
183 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
184 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
185 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
186 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
187 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
188 };
189
190 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
191 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
192 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
193 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
194 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
195 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
196 };
197
198 /* Left Record Mixer */
199 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
200 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
201 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
202 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
203 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
204 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
205 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
206 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
207 };
208
209 /* Right Record Mixer */
210 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
211 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
212 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
213 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
214 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
215 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
216 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
217 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
218 };
219
220 static const char *alc5623_spk_n_sour_sel[] = {
221 "RN/-R", "RP/+R", "LN/-R", "Vmid" };
222 static const char *alc5623_hpl_out_input_sel[] = {
223 "Vmid", "HP Left Mix"};
224 static const char *alc5623_hpr_out_input_sel[] = {
225 "Vmid", "HP Right Mix"};
226 static const char *alc5623_spkout_input_sel[] = {
227 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
228 static const char *alc5623_aux_out_input_sel[] = {
229 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
230
231 /* auxout output mux */
232 static const struct soc_enum alc5623_aux_out_input_enum =
233 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
234 static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
235 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
236
237 /* speaker output mux */
238 static const struct soc_enum alc5623_spkout_input_enum =
239 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
240 static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
241 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
242
243 /* headphone left output mux */
244 static const struct soc_enum alc5623_hpl_out_input_enum =
245 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
246 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
247 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
248
249 /* headphone right output mux */
250 static const struct soc_enum alc5623_hpr_out_input_enum =
251 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
252 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
253 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
254
255 /* speaker output N select */
256 static const struct soc_enum alc5623_spk_n_sour_enum =
257 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
258 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
259 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
260
261 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
262 /* Muxes */
263 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
264 &alc5623_auxout_mux_controls),
265 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
266 &alc5623_spkout_mux_controls),
267 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
268 &alc5623_hpl_out_mux_controls),
269 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
270 &alc5623_hpr_out_mux_controls),
271 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
272 &alc5623_spkoutn_mux_controls),
273
274 /* output mixers */
275 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
276 &alc5623_hp_mixer_controls[0],
277 ARRAY_SIZE(alc5623_hp_mixer_controls)),
278 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
279 &alc5623_hpr_mixer_controls[0],
280 ARRAY_SIZE(alc5623_hpr_mixer_controls)),
281 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
282 &alc5623_hpl_mixer_controls[0],
283 ARRAY_SIZE(alc5623_hpl_mixer_controls)),
284 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
285 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
286 &alc5623_mono_mixer_controls[0],
287 ARRAY_SIZE(alc5623_mono_mixer_controls)),
288 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
289 &alc5623_speaker_mixer_controls[0],
290 ARRAY_SIZE(alc5623_speaker_mixer_controls)),
291
292 /* input mixers */
293 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
294 &alc5623_captureL_mixer_controls[0],
295 ARRAY_SIZE(alc5623_captureL_mixer_controls)),
296 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
297 &alc5623_captureR_mixer_controls[0],
298 ARRAY_SIZE(alc5623_captureR_mixer_controls)),
299
300 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
301 ALC5623_PWR_MANAG_ADD2, 9, 0),
302 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
303 ALC5623_PWR_MANAG_ADD2, 8, 0),
304 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
305 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
306 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
307 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
308 ALC5623_PWR_MANAG_ADD2, 7, 0),
309 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
310 ALC5623_PWR_MANAG_ADD2, 6, 0),
311 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
312 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
313 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
314 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
315 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
316 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
317 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
318 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
319 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
320 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
321 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
322 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
323 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
324 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
325
326 SND_SOC_DAPM_OUTPUT("AUXOUTL"),
327 SND_SOC_DAPM_OUTPUT("AUXOUTR"),
328 SND_SOC_DAPM_OUTPUT("HPL"),
329 SND_SOC_DAPM_OUTPUT("HPR"),
330 SND_SOC_DAPM_OUTPUT("SPKOUT"),
331 SND_SOC_DAPM_OUTPUT("SPKOUTN"),
332 SND_SOC_DAPM_INPUT("LINEINL"),
333 SND_SOC_DAPM_INPUT("LINEINR"),
334 SND_SOC_DAPM_INPUT("AUXINL"),
335 SND_SOC_DAPM_INPUT("AUXINR"),
336 SND_SOC_DAPM_INPUT("MIC1"),
337 SND_SOC_DAPM_INPUT("MIC2"),
338 SND_SOC_DAPM_VMID("Vmid"),
339 };
340
341 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
342 static const struct soc_enum alc5623_amp_enum =
343 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
344 static const struct snd_kcontrol_new alc5623_amp_mux_controls =
345 SOC_DAPM_ENUM("Route", alc5623_amp_enum);
346
347 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
348 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
349 amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
350 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
351 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
352 &alc5623_amp_mux_controls),
353 };
354
355 static const struct snd_soc_dapm_route intercon[] = {
356 /* virtual mixer - mixes left & right channels */
357 {"I2S Mix", NULL, "Left DAC"},
358 {"I2S Mix", NULL, "Right DAC"},
359 {"Line Mix", NULL, "Right LineIn"},
360 {"Line Mix", NULL, "Left LineIn"},
361 {"AuxI Mix", NULL, "Left AuxI"},
362 {"AuxI Mix", NULL, "Right AuxI"},
363 {"AUXOUTL", NULL, "Left AuxOut"},
364 {"AUXOUTR", NULL, "Right AuxOut"},
365
366 /* HP mixer */
367 {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"},
368 {"HPL Mix", NULL, "HP Mix"},
369 {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"},
370 {"HPR Mix", NULL, "HP Mix"},
371 {"HP Mix", "LI2HP Playback Switch", "Line Mix"},
372 {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"},
373 {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
374 {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
375 {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"},
376
377 /* speaker mixer */
378 {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
379 {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"},
380 {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
381 {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
382 {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"},
383
384 /* mono mixer */
385 {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
386 {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
387 {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
388 {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
389 {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
390 {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
391 {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"},
392
393 /* Left record mixer */
394 {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
395 {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
396 {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
397 {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
398 {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
399 {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
400 {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
401
402 /*Right record mixer */
403 {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
404 {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"},
405 {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
406 {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
407 {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
408 {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
409 {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
410
411 /* headphone left mux */
412 {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
413 {"Left Headphone Mux", "Vmid", "Vmid"},
414
415 /* headphone right mux */
416 {"Right Headphone Mux", "HP Right Mix", "HPR Mix"},
417 {"Right Headphone Mux", "Vmid", "Vmid"},
418
419 /* speaker out mux */
420 {"SpeakerOut Mux", "Vmid", "Vmid"},
421 {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"},
422 {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"},
423 {"SpeakerOut Mux", "Mono Mix", "Mono Mix"},
424
425 /* Mono/Aux Out mux */
426 {"AuxOut Mux", "Vmid", "Vmid"},
427 {"AuxOut Mux", "HPOut Mix", "HPOut Mix"},
428 {"AuxOut Mux", "Speaker Mix", "Speaker Mix"},
429 {"AuxOut Mux", "Mono Mix", "Mono Mix"},
430
431 /* output pga */
432 {"HPL", NULL, "Left Headphone"},
433 {"Left Headphone", NULL, "Left Headphone Mux"},
434 {"HPR", NULL, "Right Headphone"},
435 {"Right Headphone", NULL, "Right Headphone Mux"},
436 {"Left AuxOut", NULL, "AuxOut Mux"},
437 {"Right AuxOut", NULL, "AuxOut Mux"},
438
439 /* input pga */
440 {"Left LineIn", NULL, "LINEINL"},
441 {"Right LineIn", NULL, "LINEINR"},
442 {"Left AuxI", NULL, "AUXINL"},
443 {"Right AuxI", NULL, "AUXINR"},
444 {"MIC1 Pre Amp", NULL, "MIC1"},
445 {"MIC2 Pre Amp", NULL, "MIC2"},
446 {"MIC1 PGA", NULL, "MIC1 Pre Amp"},
447 {"MIC2 PGA", NULL, "MIC2 Pre Amp"},
448
449 /* left ADC */
450 {"Left ADC", NULL, "Left Capture Mix"},
451
452 /* right ADC */
453 {"Right ADC", NULL, "Right Capture Mix"},
454
455 {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"},
456 {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"},
457 {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"},
458 {"SpeakerOut N Mux", "Vmid", "Vmid"},
459
460 {"SPKOUT", NULL, "SpeakerOut"},
461 {"SPKOUTN", NULL, "SpeakerOut N Mux"},
462 };
463
464 static const struct snd_soc_dapm_route intercon_spk[] = {
465 {"SpeakerOut", NULL, "SpeakerOut Mux"},
466 };
467
468 static const struct snd_soc_dapm_route intercon_amp_spk[] = {
469 {"AB Amp", NULL, "SpeakerOut Mux"},
470 {"D Amp", NULL, "SpeakerOut Mux"},
471 {"AB-D Amp Mux", "AB Amp", "AB Amp"},
472 {"AB-D Amp Mux", "D Amp", "D Amp"},
473 {"SpeakerOut", NULL, "AB-D Amp Mux"},
474 };
475
476 /* PLL divisors */
477 struct _pll_div {
478 u32 pll_in;
479 u32 pll_out;
480 u16 regvalue;
481 };
482
483 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
484 /* useful only for master mode */
485 static const struct _pll_div codec_master_pll_div[] = {
486
487 { 2048000, 8192000, 0x0ea0},
488 { 3686400, 8192000, 0x4e27},
489 { 12000000, 8192000, 0x456b},
490 { 13000000, 8192000, 0x495f},
491 { 13100000, 8192000, 0x0320},
492 { 2048000, 11289600, 0xf637},
493 { 3686400, 11289600, 0x2f22},
494 { 12000000, 11289600, 0x3e2f},
495 { 13000000, 11289600, 0x4d5b},
496 { 13100000, 11289600, 0x363b},
497 { 2048000, 16384000, 0x1ea0},
498 { 3686400, 16384000, 0x9e27},
499 { 12000000, 16384000, 0x452b},
500 { 13000000, 16384000, 0x542f},
501 { 13100000, 16384000, 0x03a0},
502 { 2048000, 16934400, 0xe625},
503 { 3686400, 16934400, 0x9126},
504 { 12000000, 16934400, 0x4d2c},
505 { 13000000, 16934400, 0x742f},
506 { 13100000, 16934400, 0x3c27},
507 { 2048000, 22579200, 0x2aa0},
508 { 3686400, 22579200, 0x2f20},
509 { 12000000, 22579200, 0x7e2f},
510 { 13000000, 22579200, 0x742f},
511 { 13100000, 22579200, 0x3c27},
512 { 2048000, 24576000, 0x2ea0},
513 { 3686400, 24576000, 0xee27},
514 { 12000000, 24576000, 0x2915},
515 { 13000000, 24576000, 0x772e},
516 { 13100000, 24576000, 0x0d20},
517 };
518
519 static const struct _pll_div codec_slave_pll_div[] = {
520
521 { 1024000, 16384000, 0x3ea0},
522 { 1411200, 22579200, 0x3ea0},
523 { 1536000, 24576000, 0x3ea0},
524 { 2048000, 16384000, 0x1ea0},
525 { 2822400, 22579200, 0x1ea0},
526 { 3072000, 24576000, 0x1ea0},
527
528 };
529
530 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
531 int source, unsigned int freq_in, unsigned int freq_out)
532 {
533 int i;
534 struct snd_soc_codec *codec = codec_dai->codec;
535 int gbl_clk = 0, pll_div = 0;
536 u16 reg;
537
538 if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
539 return -ENODEV;
540
541 /* Disable PLL power */
542 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
543 ALC5623_PWR_ADD2_PLL,
544 0);
545
546 /* pll is not used in slave mode */
547 reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
548 if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
549 return 0;
550
551 if (!freq_in || !freq_out)
552 return 0;
553
554 switch (pll_id) {
555 case ALC5623_PLL_FR_MCLK:
556 for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
557 if (codec_master_pll_div[i].pll_in == freq_in
558 && codec_master_pll_div[i].pll_out == freq_out) {
559 /* PLL source from MCLK */
560 pll_div = codec_master_pll_div[i].regvalue;
561 break;
562 }
563 }
564 break;
565 case ALC5623_PLL_FR_BCK:
566 for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
567 if (codec_slave_pll_div[i].pll_in == freq_in
568 && codec_slave_pll_div[i].pll_out == freq_out) {
569 /* PLL source from Bitclk */
570 gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
571 pll_div = codec_slave_pll_div[i].regvalue;
572 break;
573 }
574 }
575 break;
576 default:
577 return -EINVAL;
578 }
579
580 if (!pll_div)
581 return -EINVAL;
582
583 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
584 snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
585 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
586 ALC5623_PWR_ADD2_PLL,
587 ALC5623_PWR_ADD2_PLL);
588 gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
589 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
590
591 return 0;
592 }
593
594 struct _coeff_div {
595 u16 fs;
596 u16 regvalue;
597 };
598
599 /* codec hifi mclk (after PLL) clock divider coefficients */
600 /* values inspired from column BCLK=32Fs of Appendix A table */
601 static const struct _coeff_div coeff_div[] = {
602 {256*8, 0x3a69},
603 {384*8, 0x3c6b},
604 {256*4, 0x2a69},
605 {384*4, 0x2c6b},
606 {256*2, 0x1a69},
607 {384*2, 0x1c6b},
608 {256*1, 0x0a69},
609 {384*1, 0x0c6b},
610 };
611
612 static int get_coeff(struct snd_soc_codec *codec, int rate)
613 {
614 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
615 int i;
616
617 for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
618 if (coeff_div[i].fs * rate == alc5623->sysclk)
619 return i;
620 }
621 return -EINVAL;
622 }
623
624 /*
625 * Clock after PLL and dividers
626 */
627 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
628 int clk_id, unsigned int freq, int dir)
629 {
630 struct snd_soc_codec *codec = codec_dai->codec;
631 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
632
633 switch (freq) {
634 case 8192000:
635 case 11289600:
636 case 12288000:
637 case 16384000:
638 case 16934400:
639 case 18432000:
640 case 22579200:
641 case 24576000:
642 alc5623->sysclk = freq;
643 return 0;
644 }
645 return -EINVAL;
646 }
647
648 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
649 unsigned int fmt)
650 {
651 struct snd_soc_codec *codec = codec_dai->codec;
652 u16 iface = 0;
653
654 /* set master/slave audio interface */
655 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
656 case SND_SOC_DAIFMT_CBM_CFM:
657 iface = ALC5623_DAI_SDP_MASTER_MODE;
658 break;
659 case SND_SOC_DAIFMT_CBS_CFS:
660 iface = ALC5623_DAI_SDP_SLAVE_MODE;
661 break;
662 default:
663 return -EINVAL;
664 }
665
666 /* interface format */
667 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
668 case SND_SOC_DAIFMT_I2S:
669 iface |= ALC5623_DAI_I2S_DF_I2S;
670 break;
671 case SND_SOC_DAIFMT_RIGHT_J:
672 iface |= ALC5623_DAI_I2S_DF_RIGHT;
673 break;
674 case SND_SOC_DAIFMT_LEFT_J:
675 iface |= ALC5623_DAI_I2S_DF_LEFT;
676 break;
677 case SND_SOC_DAIFMT_DSP_A:
678 iface |= ALC5623_DAI_I2S_DF_PCM;
679 break;
680 case SND_SOC_DAIFMT_DSP_B:
681 iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
682 break;
683 default:
684 return -EINVAL;
685 }
686
687 /* clock inversion */
688 switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
689 case SND_SOC_DAIFMT_NB_NF:
690 break;
691 case SND_SOC_DAIFMT_IB_IF:
692 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
693 break;
694 case SND_SOC_DAIFMT_IB_NF:
695 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
696 break;
697 case SND_SOC_DAIFMT_NB_IF:
698 break;
699 default:
700 return -EINVAL;
701 }
702
703 return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
704 }
705
706 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
707 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
708 {
709 struct snd_soc_pcm_runtime *rtd = substream->private_data;
710 struct snd_soc_codec *codec = rtd->codec;
711 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
712 int coeff, rate;
713 u16 iface;
714
715 iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
716 iface &= ~ALC5623_DAI_I2S_DL_MASK;
717
718 /* bit size */
719 switch (params_format(params)) {
720 case SNDRV_PCM_FORMAT_S16_LE:
721 iface |= ALC5623_DAI_I2S_DL_16;
722 break;
723 case SNDRV_PCM_FORMAT_S20_3LE:
724 iface |= ALC5623_DAI_I2S_DL_20;
725 break;
726 case SNDRV_PCM_FORMAT_S24_LE:
727 iface |= ALC5623_DAI_I2S_DL_24;
728 break;
729 case SNDRV_PCM_FORMAT_S32_LE:
730 iface |= ALC5623_DAI_I2S_DL_32;
731 break;
732 default:
733 return -EINVAL;
734 }
735
736 /* set iface & srate */
737 snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
738 rate = params_rate(params);
739 coeff = get_coeff(codec, rate);
740 if (coeff < 0)
741 return -EINVAL;
742
743 coeff = coeff_div[coeff].regvalue;
744 dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
745 __func__, alc5623->sysclk, rate, coeff);
746 snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
747
748 return 0;
749 }
750
751 static int alc5623_mute(struct snd_soc_dai *dai, int mute)
752 {
753 struct snd_soc_codec *codec = dai->codec;
754 u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
755 u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
756
757 if (mute)
758 mute_reg |= hp_mute;
759
760 return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
761 }
762
763 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
764 | ALC5623_PWR_ADD2_DAC_REF_CIR)
765
766 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
767 | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
768
769 #define ALC5623_ADD1_POWER_EN \
770 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
771 | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
772 | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
773
774 #define ALC5623_ADD1_POWER_EN_5622 \
775 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
776 | ALC5623_PWR_ADD1_HP_OUT_AMP)
777
778 static void enable_power_depop(struct snd_soc_codec *codec)
779 {
780 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
781
782 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
783 ALC5623_PWR_ADD1_SOFTGEN_EN,
784 ALC5623_PWR_ADD1_SOFTGEN_EN);
785
786 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
787
788 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
789 ALC5623_MISC_HP_DEPOP_MODE2_EN,
790 ALC5623_MISC_HP_DEPOP_MODE2_EN);
791
792 msleep(500);
793
794 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
795
796 /* avoid writing '1' into 5622 reserved bits */
797 if (alc5623->id == 0x22)
798 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
799 ALC5623_ADD1_POWER_EN_5622);
800 else
801 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
802 ALC5623_ADD1_POWER_EN);
803
804 /* disable HP Depop2 */
805 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
806 ALC5623_MISC_HP_DEPOP_MODE2_EN,
807 0);
808
809 }
810
811 static int alc5623_set_bias_level(struct snd_soc_codec *codec,
812 enum snd_soc_bias_level level)
813 {
814 switch (level) {
815 case SND_SOC_BIAS_ON:
816 enable_power_depop(codec);
817 break;
818 case SND_SOC_BIAS_PREPARE:
819 break;
820 case SND_SOC_BIAS_STANDBY:
821 /* everything off except vref/vmid, */
822 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
823 ALC5623_PWR_ADD2_VREF);
824 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
825 ALC5623_PWR_ADD3_MAIN_BIAS);
826 break;
827 case SND_SOC_BIAS_OFF:
828 /* everything off, dac mute, inactive */
829 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
830 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
831 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
832 break;
833 }
834 codec->dapm.bias_level = level;
835 return 0;
836 }
837
838 #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
839 | SNDRV_PCM_FMTBIT_S24_LE \
840 | SNDRV_PCM_FMTBIT_S32_LE)
841
842 static struct snd_soc_dai_ops alc5623_dai_ops = {
843 .hw_params = alc5623_pcm_hw_params,
844 .digital_mute = alc5623_mute,
845 .set_fmt = alc5623_set_dai_fmt,
846 .set_sysclk = alc5623_set_dai_sysclk,
847 .set_pll = alc5623_set_dai_pll,
848 };
849
850 static struct snd_soc_dai_driver alc5623_dai = {
851 .name = "alc5623-hifi",
852 .playback = {
853 .stream_name = "Playback",
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 .capture = {
861 .stream_name = "Capture",
862 .channels_min = 1,
863 .channels_max = 2,
864 .rate_min = 8000,
865 .rate_max = 48000,
866 .rates = SNDRV_PCM_RATE_8000_48000,
867 .formats = ALC5623_FORMATS,},
868
869 .ops = &alc5623_dai_ops,
870 };
871
872 static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
873 {
874 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
875 return 0;
876 }
877
878 static int alc5623_resume(struct snd_soc_codec *codec)
879 {
880 int i, step = codec->driver->reg_cache_step;
881 u16 *cache = codec->reg_cache;
882
883 /* Sync reg_cache with the hardware */
884 for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
885 snd_soc_write(codec, i, cache[i]);
886
887 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
888
889 /* charge alc5623 caps */
890 if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
891 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
892 codec->dapm.bias_level = SND_SOC_BIAS_ON;
893 alc5623_set_bias_level(codec, codec->dapm.bias_level);
894 }
895
896 return 0;
897 }
898
899 static int alc5623_probe(struct snd_soc_codec *codec)
900 {
901 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
902 struct snd_soc_dapm_context *dapm = &codec->dapm;
903 int ret;
904
905 ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type);
906 if (ret < 0) {
907 dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret);
908 return ret;
909 }
910
911 alc5623_reset(codec);
912 alc5623_fill_cache(codec);
913
914 /* power on device */
915 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
916
917 if (alc5623->add_ctrl) {
918 snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
919 alc5623->add_ctrl);
920 }
921
922 if (alc5623->jack_det_ctrl) {
923 snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
924 alc5623->jack_det_ctrl);
925 }
926
927 switch (alc5623->id) {
928 case 0x21:
929 snd_soc_add_controls(codec, rt5621_vol_snd_controls,
930 ARRAY_SIZE(rt5621_vol_snd_controls));
931 break;
932 case 0x22:
933 snd_soc_add_controls(codec, rt5622_vol_snd_controls,
934 ARRAY_SIZE(rt5622_vol_snd_controls));
935 break;
936 case 0x23:
937 snd_soc_add_controls(codec, alc5623_vol_snd_controls,
938 ARRAY_SIZE(alc5623_vol_snd_controls));
939 break;
940 default:
941 return -EINVAL;
942 }
943
944 snd_soc_add_controls(codec, alc5623_snd_controls,
945 ARRAY_SIZE(alc5623_snd_controls));
946
947 snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
948 ARRAY_SIZE(alc5623_dapm_widgets));
949
950 /* set up audio path interconnects */
951 snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
952
953 switch (alc5623->id) {
954 case 0x21:
955 case 0x22:
956 snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
957 ARRAY_SIZE(alc5623_dapm_amp_widgets));
958 snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
959 ARRAY_SIZE(intercon_amp_spk));
960 break;
961 case 0x23:
962 snd_soc_dapm_add_routes(dapm, intercon_spk,
963 ARRAY_SIZE(intercon_spk));
964 break;
965 default:
966 return -EINVAL;
967 }
968
969 return ret;
970 }
971
972 /* power down chip */
973 static int alc5623_remove(struct snd_soc_codec *codec)
974 {
975 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
976 return 0;
977 }
978
979 static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
980 .probe = alc5623_probe,
981 .remove = alc5623_remove,
982 .suspend = alc5623_suspend,
983 .resume = alc5623_resume,
984 .set_bias_level = alc5623_set_bias_level,
985 .reg_cache_size = ALC5623_VENDOR_ID2+2,
986 .reg_word_size = sizeof(u16),
987 .reg_cache_step = 2,
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 int ret, vid1, vid2;
1002
1003 vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1);
1004 if (vid1 < 0) {
1005 dev_err(&client->dev, "failed to read I2C\n");
1006 return -EIO;
1007 }
1008 vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
1009
1010 vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2);
1011 if (vid2 < 0) {
1012 dev_err(&client->dev, "failed to read I2C\n");
1013 return -EIO;
1014 }
1015
1016 if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
1017 dev_err(&client->dev, "unknown or wrong codec\n");
1018 dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
1019 0x10ec, id->driver_data,
1020 vid1, vid2);
1021 return -ENODEV;
1022 }
1023
1024 dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
1025
1026 alc5623 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL);
1027 if (alc5623 == NULL)
1028 return -ENOMEM;
1029
1030 pdata = client->dev.platform_data;
1031 if (pdata) {
1032 alc5623->add_ctrl = pdata->add_ctrl;
1033 alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
1034 }
1035
1036 alc5623->id = vid2;
1037 switch (alc5623->id) {
1038 case 0x21:
1039 alc5623_dai.name = "alc5621-hifi";
1040 break;
1041 case 0x22:
1042 alc5623_dai.name = "alc5622-hifi";
1043 break;
1044 case 0x23:
1045 alc5623_dai.name = "alc5623-hifi";
1046 break;
1047 default:
1048 kfree(alc5623);
1049 return -EINVAL;
1050 }
1051
1052 i2c_set_clientdata(client, alc5623);
1053 alc5623->control_data = client;
1054 alc5623->control_type = SND_SOC_I2C;
1055 mutex_init(&alc5623->mutex);
1056
1057 ret = snd_soc_register_codec(&client->dev,
1058 &soc_codec_device_alc5623, &alc5623_dai, 1);
1059 if (ret != 0) {
1060 dev_err(&client->dev, "Failed to register codec: %d\n", ret);
1061 kfree(alc5623);
1062 }
1063
1064 return ret;
1065 }
1066
1067 static int alc5623_i2c_remove(struct i2c_client *client)
1068 {
1069 struct alc5623_priv *alc5623 = i2c_get_clientdata(client);
1070
1071 snd_soc_unregister_codec(&client->dev);
1072 kfree(alc5623);
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 = __devexit_p(alc5623_i2c_remove),
1092 .id_table = alc5623_i2c_table,
1093 };
1094
1095 static int __init alc5623_modinit(void)
1096 {
1097 int ret;
1098
1099 ret = i2c_add_driver(&alc5623_i2c_driver);
1100 if (ret != 0) {
1101 printk(KERN_ERR "%s: can't add i2c driver", __func__);
1102 return ret;
1103 }
1104
1105 return ret;
1106 }
1107 module_init(alc5623_modinit);
1108
1109 static void __exit alc5623_modexit(void)
1110 {
1111 i2c_del_driver(&alc5623_i2c_driver);
1112 }
1113 module_exit(alc5623_modexit);
1114
1115 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1116 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
1117 MODULE_LICENSE("GPL");
Linux with added FPC-III support