aboutsummaryrefslogtreecommitdiffstats
path: root/sound/soc/codecs/alc5623.c
diff options
context:
space:
mode:
authorArnaud Patard (Rtp) <arnaud.patard@rtp-net.org>2010-10-21 13:40:02 -0400
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-10-21 16:51:13 -0400
commit6f4bc952c60b26ecfcb013fb9a7e9474023e046e (patch)
tree1d85312033eeab9393d1905cf342bf8f1ac6e474 /sound/soc/codecs/alc5623.c
parent4428bc0990ba545e2ef0dea8ec1b90c256b22958 (diff)
ASoC: add support for alc562[123] codecs
This patch is adding support for alc562[123] codecs. It's based on the source code available in HP source code and other places. Signed-off-by: Arnaud Patard <arnaud.patard@rtp-net.org> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound/soc/codecs/alc5623.c')
-rw-r--r--sound/soc/codecs/alc5623.c1118
1 files changed, 1118 insertions, 0 deletions
diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c
new file mode 100644
index 000000000000..fac61744f8c7
--- /dev/null
+++ b/sound/soc/codecs/alc5623.c
@@ -0,0 +1,1118 @@
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/soc-dapm.h>
32#include <sound/initval.h>
33#include <sound/alc5623.h>
34
35#include "alc5623.h"
36
37static int caps_charge = 2000;
38module_param(caps_charge, int, 0);
39MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
40
41/* codec private data */
42struct alc5623_priv {
43 enum snd_soc_control_type control_type;
44 void *control_data;
45 struct mutex mutex;
46 u8 id;
47 unsigned int sysclk;
48 u16 reg_cache[ALC5623_VENDOR_ID2+2];
49 unsigned int add_ctrl;
50 unsigned int jack_det_ctrl;
51};
52
53static void alc5623_fill_cache(struct snd_soc_codec *codec)
54{
55 int i, step = codec->driver->reg_cache_step;
56 u16 *cache = codec->reg_cache;
57
58 /* not really efficient ... */
59 for (i = 0 ; i < codec->driver->reg_cache_size ; i += step)
60 cache[i] = codec->hw_read(codec, i);
61}
62
63static inline int alc5623_reset(struct snd_soc_codec *codec)
64{
65 return snd_soc_write(codec, ALC5623_RESET, 0);
66}
67
68static int amp_mixer_event(struct snd_soc_dapm_widget *w,
69 struct snd_kcontrol *kcontrol, int event)
70{
71 /* to power-on/off class-d amp generators/speaker */
72 /* need to write to 'index-46h' register : */
73 /* so write index num (here 0x46) to reg 0x6a */
74 /* and then 0xffff/0 to reg 0x6c */
75 snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
76
77 switch (event) {
78 case SND_SOC_DAPM_PRE_PMU:
79 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
80 break;
81 case SND_SOC_DAPM_POST_PMD:
82 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
83 break;
84 }
85
86 return 0;
87}
88
89/*
90 * ALC5623 Controls
91 */
92
93static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
94static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
95static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
96static const unsigned int boost_tlv[] = {
97 TLV_DB_RANGE_HEAD(3),
98 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
99 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
100 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
101};
102static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
103
104static const struct snd_kcontrol_new rt5621_vol_snd_controls[] = {
105 SOC_DOUBLE_TLV("Speaker Playback Volume",
106 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
107 SOC_DOUBLE("Speaker Playback Switch",
108 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
109 SOC_DOUBLE_TLV("Headphone Playback Volume",
110 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
111 SOC_DOUBLE("Headphone Playback Switch",
112 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
113};
114
115static const struct snd_kcontrol_new rt5622_vol_snd_controls[] = {
116 SOC_DOUBLE_TLV("Speaker Playback Volume",
117 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
118 SOC_DOUBLE("Speaker Playback Switch",
119 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
120 SOC_DOUBLE_TLV("Line Playback Volume",
121 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
122 SOC_DOUBLE("Line Playback Switch",
123 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
124};
125
126static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
127 SOC_DOUBLE_TLV("Line Playback Volume",
128 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
129 SOC_DOUBLE("Line Playback Switch",
130 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
131 SOC_DOUBLE_TLV("Headphone Playback Volume",
132 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
133 SOC_DOUBLE("Headphone Playback Switch",
134 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
135};
136
137static const struct snd_kcontrol_new alc5623_snd_controls[] = {
138 SOC_DOUBLE_TLV("Auxout Playback Volume",
139 ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
140 SOC_DOUBLE("Auxout Playback Switch",
141 ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
142 SOC_DOUBLE_TLV("PCM Playback Volume",
143 ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
144 SOC_DOUBLE_TLV("AuxI Capture Volume",
145 ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
146 SOC_DOUBLE_TLV("LineIn Capture Volume",
147 ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
148 SOC_SINGLE_TLV("Mic1 Capture Volume",
149 ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
150 SOC_SINGLE_TLV("Mic2 Capture Volume",
151 ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
152 SOC_DOUBLE_TLV("Rec Capture Volume",
153 ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
154 SOC_SINGLE_TLV("Mic 1 Boost Volume",
155 ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
156 SOC_SINGLE_TLV("Mic 2 Boost Volume",
157 ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
158 SOC_SINGLE_TLV("Digital Boost Volume",
159 ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
160};
161
162/*
163 * DAPM Controls
164 */
165static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
166SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
167SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
168SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
169SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
170SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
171};
172
173static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
174SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
175};
176
177static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
178SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
179};
180
181static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
182SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
183SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
184SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
185SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
186SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
187SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
188SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
189};
190
191static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
192SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
193SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
194SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
195SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
196SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
197};
198
199/* Left Record Mixer */
200static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
201SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
202SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
203SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
204SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
205SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
206SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
207SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
208};
209
210/* Right Record Mixer */
211static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
212SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
213SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
214SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
215SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
216SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
217SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
218SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
219};
220
221static const char *alc5623_spk_n_sour_sel[] = {
222 "RN/-R", "RP/+R", "LN/-R", "Vmid" };
223static const char *alc5623_hpl_out_input_sel[] = {
224 "Vmid", "HP Left Mix"};
225static const char *alc5623_hpr_out_input_sel[] = {
226 "Vmid", "HP Right Mix"};
227static const char *alc5623_spkout_input_sel[] = {
228 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
229static const char *alc5623_aux_out_input_sel[] = {
230 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
231
232/* auxout output mux */
233static const struct soc_enum alc5623_aux_out_input_enum =
234SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel);
235static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
236SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
237
238/* speaker output mux */
239static const struct soc_enum alc5623_spkout_input_enum =
240SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel);
241static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
242SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
243
244/* headphone left output mux */
245static const struct soc_enum alc5623_hpl_out_input_enum =
246SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel);
247static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
248SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
249
250/* headphone right output mux */
251static const struct soc_enum alc5623_hpr_out_input_enum =
252SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel);
253static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
254SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
255
256/* speaker output N select */
257static const struct soc_enum alc5623_spk_n_sour_enum =
258SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel);
259static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
260SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
261
262static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
263/* Muxes */
264SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
265 &alc5623_auxout_mux_controls),
266SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
267 &alc5623_spkout_mux_controls),
268SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
269 &alc5623_hpl_out_mux_controls),
270SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
271 &alc5623_hpr_out_mux_controls),
272SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
273 &alc5623_spkoutn_mux_controls),
274
275/* output mixers */
276SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
277 &alc5623_hp_mixer_controls[0],
278 ARRAY_SIZE(alc5623_hp_mixer_controls)),
279SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
280 &alc5623_hpr_mixer_controls[0],
281 ARRAY_SIZE(alc5623_hpr_mixer_controls)),
282SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
283 &alc5623_hpl_mixer_controls[0],
284 ARRAY_SIZE(alc5623_hpl_mixer_controls)),
285SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
286SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
287 &alc5623_mono_mixer_controls[0],
288 ARRAY_SIZE(alc5623_mono_mixer_controls)),
289SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
290 &alc5623_speaker_mixer_controls[0],
291 ARRAY_SIZE(alc5623_speaker_mixer_controls)),
292
293/* input mixers */
294SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
295 &alc5623_captureL_mixer_controls[0],
296 ARRAY_SIZE(alc5623_captureL_mixer_controls)),
297SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
298 &alc5623_captureR_mixer_controls[0],
299 ARRAY_SIZE(alc5623_captureR_mixer_controls)),
300
301SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
302 ALC5623_PWR_MANAG_ADD2, 9, 0),
303SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
304 ALC5623_PWR_MANAG_ADD2, 8, 0),
305SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
306SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
307SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
308SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
309 ALC5623_PWR_MANAG_ADD2, 7, 0),
310SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
311 ALC5623_PWR_MANAG_ADD2, 6, 0),
312SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
313SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
314SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
315SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
316SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
317SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
318SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
319SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
320SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
321SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
322SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
323SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
324SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
325SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
326
327SND_SOC_DAPM_OUTPUT("AUXOUTL"),
328SND_SOC_DAPM_OUTPUT("AUXOUTR"),
329SND_SOC_DAPM_OUTPUT("HPL"),
330SND_SOC_DAPM_OUTPUT("HPR"),
331SND_SOC_DAPM_OUTPUT("SPKOUT"),
332SND_SOC_DAPM_OUTPUT("SPKOUTN"),
333SND_SOC_DAPM_INPUT("LINEINL"),
334SND_SOC_DAPM_INPUT("LINEINR"),
335SND_SOC_DAPM_INPUT("AUXINL"),
336SND_SOC_DAPM_INPUT("AUXINR"),
337SND_SOC_DAPM_INPUT("MIC1"),
338SND_SOC_DAPM_INPUT("MIC2"),
339SND_SOC_DAPM_VMID("Vmid"),
340};
341
342static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
343static const struct soc_enum alc5623_amp_enum =
344 SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names);
345static const struct snd_kcontrol_new alc5623_amp_mux_controls =
346 SOC_DAPM_ENUM("Route", alc5623_amp_enum);
347
348static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
349SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
350 amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
351SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
352SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
353 &alc5623_amp_mux_controls),
354};
355
356static const struct snd_soc_dapm_route intercon[] = {
357 /* virtual mixer - mixes left & right channels */
358 {"I2S Mix", NULL, "Left DAC"},
359 {"I2S Mix", NULL, "Right DAC"},
360 {"Line Mix", NULL, "Right LineIn"},
361 {"Line Mix", NULL, "Left LineIn"},
362 {"AuxI Mix", NULL, "Left AuxI"},
363 {"AuxI Mix", NULL, "Right AuxI"},
364 {"AUXOUTL", NULL, "Left AuxOut"},
365 {"AUXOUTR", NULL, "Right AuxOut"},
366
367 /* HP mixer */
368 {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"},
369 {"HPL Mix", NULL, "HP Mix"},
370 {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"},
371 {"HPR Mix", NULL, "HP Mix"},
372 {"HP Mix", "LI2HP Playback Switch", "Line Mix"},
373 {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"},
374 {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
375 {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
376 {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"},
377
378 /* speaker mixer */
379 {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
380 {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"},
381 {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
382 {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
383 {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"},
384
385 /* mono mixer */
386 {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
387 {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
388 {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
389 {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
390 {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
391 {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
392 {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"},
393
394 /* Left record mixer */
395 {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
396 {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
397 {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
398 {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
399 {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
400 {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
401 {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
402
403 /*Right record mixer */
404 {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
405 {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"},
406 {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
407 {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
408 {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
409 {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
410 {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
411
412 /* headphone left mux */
413 {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
414 {"Left Headphone Mux", "Vmid", "Vmid"},
415
416 /* headphone right mux */
417 {"Right Headphone Mux", "HP Right Mix", "HPR Mix"},
418 {"Right Headphone Mux", "Vmid", "Vmid"},
419
420 /* speaker out mux */
421 {"SpeakerOut Mux", "Vmid", "Vmid"},
422 {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"},
423 {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"},
424 {"SpeakerOut Mux", "Mono Mix", "Mono Mix"},
425
426 /* Mono/Aux Out mux */
427 {"AuxOut Mux", "Vmid", "Vmid"},
428 {"AuxOut Mux", "HPOut Mix", "HPOut Mix"},
429 {"AuxOut Mux", "Speaker Mix", "Speaker Mix"},
430 {"AuxOut Mux", "Mono Mix", "Mono Mix"},
431
432 /* output pga */
433 {"HPL", NULL, "Left Headphone"},
434 {"Left Headphone", NULL, "Left Headphone Mux"},
435 {"HPR", NULL, "Right Headphone"},
436 {"Right Headphone", NULL, "Right Headphone Mux"},
437 {"Left AuxOut", NULL, "AuxOut Mux"},
438 {"Right AuxOut", NULL, "AuxOut Mux"},
439
440 /* input pga */
441 {"Left LineIn", NULL, "LINEINL"},
442 {"Right LineIn", NULL, "LINEINR"},
443 {"Left AuxI", NULL, "AUXINL"},
444 {"Right AuxI", NULL, "AUXINR"},
445 {"MIC1 Pre Amp", NULL, "MIC1"},
446 {"MIC2 Pre Amp", NULL, "MIC2"},
447 {"MIC1 PGA", NULL, "MIC1 Pre Amp"},
448 {"MIC2 PGA", NULL, "MIC2 Pre Amp"},
449
450 /* left ADC */
451 {"Left ADC", NULL, "Left Capture Mix"},
452
453 /* right ADC */
454 {"Right ADC", NULL, "Right Capture Mix"},
455
456 {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"},
457 {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"},
458 {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"},
459 {"SpeakerOut N Mux", "Vmid", "Vmid"},
460
461 {"SPKOUT", NULL, "SpeakerOut"},
462 {"SPKOUTN", NULL, "SpeakerOut N Mux"},
463};
464
465static const struct snd_soc_dapm_route intercon_spk[] = {
466 {"SpeakerOut", NULL, "SpeakerOut Mux"},
467};
468
469static const struct snd_soc_dapm_route intercon_amp_spk[] = {
470 {"AB Amp", NULL, "SpeakerOut Mux"},
471 {"D Amp", NULL, "SpeakerOut Mux"},
472 {"AB-D Amp Mux", "AB Amp", "AB Amp"},
473 {"AB-D Amp Mux", "D Amp", "D Amp"},
474 {"SpeakerOut", NULL, "AB-D Amp Mux"},
475};
476
477/* PLL divisors */
478struct _pll_div {
479 u32 pll_in;
480 u32 pll_out;
481 u16 regvalue;
482};
483
484/* Note : pll code from original alc5623 driver. Not sure of how good it is */
485/* usefull only for master mode */
486static const struct _pll_div codec_master_pll_div[] = {
487
488 { 2048000, 8192000, 0x0ea0},
489 { 3686400, 8192000, 0x4e27},
490 { 12000000, 8192000, 0x456b},
491 { 13000000, 8192000, 0x495f},
492 { 13100000, 8192000, 0x0320},
493 { 2048000, 11289600, 0xf637},
494 { 3686400, 11289600, 0x2f22},
495 { 12000000, 11289600, 0x3e2f},
496 { 13000000, 11289600, 0x4d5b},
497 { 13100000, 11289600, 0x363b},
498 { 2048000, 16384000, 0x1ea0},
499 { 3686400, 16384000, 0x9e27},
500 { 12000000, 16384000, 0x452b},
501 { 13000000, 16384000, 0x542f},
502 { 13100000, 16384000, 0x03a0},
503 { 2048000, 16934400, 0xe625},
504 { 3686400, 16934400, 0x9126},
505 { 12000000, 16934400, 0x4d2c},
506 { 13000000, 16934400, 0x742f},
507 { 13100000, 16934400, 0x3c27},
508 { 2048000, 22579200, 0x2aa0},
509 { 3686400, 22579200, 0x2f20},
510 { 12000000, 22579200, 0x7e2f},
511 { 13000000, 22579200, 0x742f},
512 { 13100000, 22579200, 0x3c27},
513 { 2048000, 24576000, 0x2ea0},
514 { 3686400, 24576000, 0xee27},
515 { 12000000, 24576000, 0x2915},
516 { 13000000, 24576000, 0x772e},
517 { 13100000, 24576000, 0x0d20},
518};
519
520static const struct _pll_div codec_slave_pll_div[] = {
521
522 { 1024000, 16384000, 0x3ea0},
523 { 1411200, 22579200, 0x3ea0},
524 { 1536000, 24576000, 0x3ea0},
525 { 2048000, 16384000, 0x1ea0},
526 { 2822400, 22579200, 0x1ea0},
527 { 3072000, 24576000, 0x1ea0},
528
529};
530
531static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
532 int source, unsigned int freq_in, unsigned int freq_out)
533{
534 int i;
535 struct snd_soc_codec *codec = codec_dai->codec;
536 int gbl_clk = 0, pll_div = 0;
537 u16 reg;
538
539 if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
540 return -ENODEV;
541
542 /* Disable PLL power */
543 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
544 ALC5623_PWR_ADD2_PLL,
545 0);
546
547 /* pll is not used in slave mode */
548 reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
549 if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
550 return 0;
551
552 if (!freq_in || !freq_out)
553 return 0;
554
555 switch (pll_id) {
556 case ALC5623_PLL_FR_MCLK:
557 for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
558 if (codec_master_pll_div[i].pll_in == freq_in
559 && codec_master_pll_div[i].pll_out == freq_out) {
560 /* PLL source from MCLK */
561 pll_div = codec_master_pll_div[i].regvalue;
562 break;
563 }
564 }
565 break;
566 case ALC5623_PLL_FR_BCK:
567 for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
568 if (codec_slave_pll_div[i].pll_in == freq_in
569 && codec_slave_pll_div[i].pll_out == freq_out) {
570 /* PLL source from Bitclk */
571 gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
572 pll_div = codec_slave_pll_div[i].regvalue;
573 break;
574 }
575 }
576 break;
577 default:
578 return -EINVAL;
579 }
580
581 if (!pll_div)
582 return -EINVAL;
583
584 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
585 snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
586 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
587 ALC5623_PWR_ADD2_PLL,
588 ALC5623_PWR_ADD2_PLL);
589 gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
590 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
591
592 return 0;
593}
594
595struct _coeff_div {
596 u16 fs;
597 u16 regvalue;
598};
599
600/* codec hifi mclk (after PLL) clock divider coefficients */
601/* values inspired from column BCLK=32Fs of Appendix A table */
602static const struct _coeff_div coeff_div[] = {
603 {256*8, 0x3a69},
604 {384*8, 0x3c6b},
605 {256*4, 0x2a69},
606 {384*4, 0x2c6b},
607 {256*2, 0x1a69},
608 {384*2, 0x1c6b},
609 {256*1, 0x0a69},
610 {384*1, 0x0c6b},
611};
612
613static int get_coeff(struct snd_soc_codec *codec, int rate)
614{
615 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
616 int i;
617
618 for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
619 if (coeff_div[i].fs * rate == alc5623->sysclk)
620 return i;
621 }
622 return -EINVAL;
623}
624
625/*
626 * Clock after PLL and dividers
627 */
628static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
629 int clk_id, unsigned int freq, int dir)
630{
631 struct snd_soc_codec *codec = codec_dai->codec;
632 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
633
634 switch (freq) {
635 case 8192000:
636 case 11289600:
637 case 12288000:
638 case 16384000:
639 case 16934400:
640 case 18432000:
641 case 22579200:
642 case 24576000:
643 alc5623->sysclk = freq;
644 return 0;
645 }
646 return -EINVAL;
647}
648
649static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
650 unsigned int fmt)
651{
652 struct snd_soc_codec *codec = codec_dai->codec;
653 u16 iface = 0;
654
655 /* set master/slave audio interface */
656 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
657 case SND_SOC_DAIFMT_CBM_CFM:
658 iface = ALC5623_DAI_SDP_MASTER_MODE;
659 break;
660 case SND_SOC_DAIFMT_CBS_CFS:
661 iface = ALC5623_DAI_SDP_SLAVE_MODE;
662 break;
663 default:
664 return -EINVAL;
665 }
666
667 /* interface format */
668 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
669 case SND_SOC_DAIFMT_I2S:
670 iface |= ALC5623_DAI_I2S_DF_I2S;
671 break;
672 case SND_SOC_DAIFMT_RIGHT_J:
673 iface |= ALC5623_DAI_I2S_DF_RIGHT;
674 break;
675 case SND_SOC_DAIFMT_LEFT_J:
676 iface |= ALC5623_DAI_I2S_DF_LEFT;
677 break;
678 case SND_SOC_DAIFMT_DSP_A:
679 iface |= ALC5623_DAI_I2S_DF_PCM;
680 break;
681 case SND_SOC_DAIFMT_DSP_B:
682 iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
683 break;
684 default:
685 return -EINVAL;
686 }
687
688 /* clock inversion */
689 switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
690 case SND_SOC_DAIFMT_NB_NF:
691 break;
692 case SND_SOC_DAIFMT_IB_IF:
693 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
694 break;
695 case SND_SOC_DAIFMT_IB_NF:
696 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
697 break;
698 case SND_SOC_DAIFMT_NB_IF:
699 break;
700 default:
701 return -EINVAL;
702 }
703
704 return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
705}
706
707static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
708 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
709{
710 struct snd_soc_pcm_runtime *rtd = substream->private_data;
711 struct snd_soc_codec *codec = rtd->codec;
712 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
713 int coeff, rate;
714 u16 iface;
715
716 iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
717 iface &= ~ALC5623_DAI_I2S_DL_MASK;
718
719 /* bit size */
720 switch (params_format(params)) {
721 case SNDRV_PCM_FORMAT_S16_LE:
722 iface |= ALC5623_DAI_I2S_DL_16;
723 break;
724 case SNDRV_PCM_FORMAT_S20_3LE:
725 iface |= ALC5623_DAI_I2S_DL_20;
726 break;
727 case SNDRV_PCM_FORMAT_S24_LE:
728 iface |= ALC5623_DAI_I2S_DL_24;
729 break;
730 case SNDRV_PCM_FORMAT_S32_LE:
731 iface |= ALC5623_DAI_I2S_DL_32;
732 break;
733 default:
734 return -EINVAL;
735 }
736
737 /* set iface & srate */
738 snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
739 rate = params_rate(params);
740 coeff = get_coeff(codec, rate);
741 if (coeff < 0)
742 return -EINVAL;
743
744 coeff = coeff_div[coeff].regvalue;
745 dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
746 __func__, alc5623->sysclk, rate, coeff);
747 snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
748
749 return 0;
750}
751
752static int alc5623_mute(struct snd_soc_dai *dai, int mute)
753{
754 struct snd_soc_codec *codec = dai->codec;
755 u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
756 u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
757
758 if (mute)
759 mute_reg |= hp_mute;
760
761 return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
762}
763
764#define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
765 | ALC5623_PWR_ADD2_DAC_REF_CIR)
766
767#define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
768 | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
769
770#define ALC5623_ADD1_POWER_EN \
771 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
772 | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
773 | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
774
775#define ALC5623_ADD1_POWER_EN_5622 \
776 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
777 | ALC5623_PWR_ADD1_HP_OUT_AMP)
778
779static void enable_power_depop(struct snd_soc_codec *codec)
780{
781 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
782
783 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
784 ALC5623_PWR_ADD1_SOFTGEN_EN,
785 ALC5623_PWR_ADD1_SOFTGEN_EN);
786
787 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
788
789 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
790 ALC5623_MISC_HP_DEPOP_MODE2_EN,
791 ALC5623_MISC_HP_DEPOP_MODE2_EN);
792
793 msleep(500);
794
795 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
796
797 /* avoid writing '1' into 5622 reserved bits */
798 if (alc5623->id == 0x22)
799 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
800 ALC5623_ADD1_POWER_EN_5622);
801 else
802 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
803 ALC5623_ADD1_POWER_EN);
804
805 /* disable HP Depop2 */
806 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
807 ALC5623_MISC_HP_DEPOP_MODE2_EN,
808 0);
809
810}
811
812static int alc5623_set_bias_level(struct snd_soc_codec *codec,
813 enum snd_soc_bias_level level)
814{
815 switch (level) {
816 case SND_SOC_BIAS_ON:
817 enable_power_depop(codec);
818 break;
819 case SND_SOC_BIAS_PREPARE:
820 break;
821 case SND_SOC_BIAS_STANDBY:
822 /* everything off except vref/vmid, */
823 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
824 ALC5623_PWR_ADD2_VREF);
825 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
826 ALC5623_PWR_ADD3_MAIN_BIAS);
827 break;
828 case SND_SOC_BIAS_OFF:
829 /* everything off, dac mute, inactive */
830 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
831 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
832 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
833 break;
834 }
835 codec->bias_level = level;
836 return 0;
837}
838
839#define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
840 | SNDRV_PCM_FMTBIT_S24_LE \
841 | SNDRV_PCM_FMTBIT_S32_LE)
842
843static struct snd_soc_dai_ops alc5623_dai_ops = {
844 .hw_params = alc5623_pcm_hw_params,
845 .digital_mute = alc5623_mute,
846 .set_fmt = alc5623_set_dai_fmt,
847 .set_sysclk = alc5623_set_dai_sysclk,
848 .set_pll = alc5623_set_dai_pll,
849};
850
851static struct snd_soc_dai_driver alc5623_dai = {
852 .name = "alc5623-hifi",
853 .playback = {
854 .stream_name = "Playback",
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 .capture = {
862 .stream_name = "Capture",
863 .channels_min = 1,
864 .channels_max = 2,
865 .rate_min = 8000,
866 .rate_max = 48000,
867 .rates = SNDRV_PCM_RATE_8000_48000,
868 .formats = ALC5623_FORMATS,},
869
870 .ops = &alc5623_dai_ops,
871};
872
873static int alc5623_suspend(struct snd_soc_codec *codec, pm_message_t mesg)
874{
875 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
876 return 0;
877}
878
879static int alc5623_resume(struct snd_soc_codec *codec)
880{
881 int i, step = codec->driver->reg_cache_step;
882 u16 *cache = codec->reg_cache;
883
884 /* Sync reg_cache with the hardware */
885 for (i = 2 ; i < codec->driver->reg_cache_size ; i += step)
886 snd_soc_write(codec, i, cache[i]);
887
888 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
889
890 /* charge alc5623 caps */
891 if (codec->suspend_bias_level == SND_SOC_BIAS_ON) {
892 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
893 codec->bias_level = SND_SOC_BIAS_ON;
894 alc5623_set_bias_level(codec, codec->bias_level);
895 }
896
897 return 0;
898}
899
900static int alc5623_probe(struct snd_soc_codec *codec)
901{
902 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
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 default:
929 case 0x21:
930 snd_soc_add_controls(codec, rt5621_vol_snd_controls,
931 ARRAY_SIZE(rt5621_vol_snd_controls));
932 break;
933 case 0x22:
934 snd_soc_add_controls(codec, rt5622_vol_snd_controls,
935 ARRAY_SIZE(rt5622_vol_snd_controls));
936 break;
937 case 0x23:
938 snd_soc_add_controls(codec, alc5623_vol_snd_controls,
939 ARRAY_SIZE(alc5623_vol_snd_controls));
940 break;
941 }
942
943 snd_soc_add_controls(codec, alc5623_snd_controls,
944 ARRAY_SIZE(alc5623_snd_controls));
945
946 snd_soc_dapm_new_controls(codec, alc5623_dapm_widgets,
947 ARRAY_SIZE(alc5623_dapm_widgets));
948
949 /* set up audio path interconnects */
950 snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
951
952 switch (alc5623->id) {
953 default:
954 case 0x21:
955 case 0x22:
956 snd_soc_dapm_new_controls(codec, alc5623_dapm_amp_widgets,
957 ARRAY_SIZE(alc5623_dapm_amp_widgets));
958 snd_soc_dapm_add_routes(codec, intercon_amp_spk,
959 ARRAY_SIZE(intercon_amp_spk));
960 break;
961 case 0x23:
962 snd_soc_dapm_add_routes(codec, intercon_spk,
963 ARRAY_SIZE(intercon_spk));
964 break;
965 }
966
967 return ret;
968}
969
970/* power down chip */
971static int alc5623_remove(struct snd_soc_codec *codec)
972{
973 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
974 return 0;
975}
976
977static 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 */
994static 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 = kzalloc(sizeof(struct alc5623_priv), GFP_KERNEL);
1025 if (alc5623 == NULL) {
1026 ret = -ENOMEM;
1027 goto err;
1028 }
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 default:
1045 case 0x23:
1046 alc5623_dai.name = "alc5623-hifi";
1047 break;
1048 }
1049
1050 i2c_set_clientdata(client, alc5623);
1051 alc5623->control_data = client;
1052 alc5623->control_type = SND_SOC_I2C;
1053 mutex_init(&alc5623->mutex);
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 goto err;
1060 }
1061
1062 return 0;
1063
1064err:
1065 return ret;
1066}
1067
1068static int alc5623_i2c_remove(struct i2c_client *client)
1069{
1070 struct alc5623_priv *alc5623 = i2c_get_clientdata(client);
1071
1072 snd_soc_unregister_codec(&client->dev);
1073 kfree(alc5623);
1074 return 0;
1075}
1076
1077static const struct i2c_device_id alc5623_i2c_table[] = {
1078 {"alc5621", 0x21},
1079 {"alc5622", 0x22},
1080 {"alc5623", 0x23},
1081 {}
1082};
1083MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
1084
1085/* i2c codec control layer */
1086static struct i2c_driver alc5623_i2c_driver = {
1087 .driver = {
1088 .name = "alc562x-codec",
1089 .owner = THIS_MODULE,
1090 },
1091 .probe = alc5623_i2c_probe,
1092 .remove = __devexit_p(alc5623_i2c_remove),
1093 .id_table = alc5623_i2c_table,
1094};
1095
1096static int __init alc5623_modinit(void)
1097{
1098 int ret;
1099
1100 ret = i2c_add_driver(&alc5623_i2c_driver);
1101 if (ret != 0) {
1102 printk(KERN_ERR "%s: can't add i2c driver", __func__);
1103 return ret;
1104 }
1105
1106 return ret;
1107}
1108module_init(alc5623_modinit);
1109
1110static void __exit alc5623_modexit(void)
1111{
1112 i2c_del_driver(&alc5623_i2c_driver);
1113}
1114module_exit(alc5623_modexit);
1115
1116MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1117MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
1118MODULE_LICENSE("GPL");
generated by cgit v1.2.2 (git 2.25.0) at 2024-11-16 08:34:40 -0500