2 * alc5623.c -- alc562[123] ALSA Soc Audio driver
4 * Copyright 2008 Realtek Microelectronics
5 * Author: flove <flove@realtek.com> Ethan <eku@marvell.com>
7 * Copyright 2010 Arnaud Patard <arnaud.patard@rtp-net.org>
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.
18 #include <linux/module.h>
19 #include <linux/kernel.h>
20 #include <linux/init.h>
21 #include <linux/delay.h>
23 #include <linux/i2c.h>
24 #include <linux/regmap.h>
25 #include <linux/slab.h>
26 #include <sound/core.h>
27 #include <sound/pcm.h>
28 #include <sound/pcm_params.h>
29 #include <sound/tlv.h>
30 #include <sound/soc.h>
31 #include <sound/initval.h>
32 #include <sound/alc5623.h>
36 static int caps_charge = 2000;
37 module_param(caps_charge, int, 0);
38 MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)");
40 /* codec private data */
42 struct regmap *regmap;
45 unsigned int add_ctrl;
46 unsigned int jack_det_ctrl;
49 static inline int alc5623_reset(struct snd_soc_codec *codec)
51 return snd_soc_write(codec, ALC5623_RESET, 0);
54 static int amp_mixer_event(struct snd_soc_dapm_widget *w,
55 struct snd_kcontrol *kcontrol, int event)
57 /* to power-on/off class-d amp generators/speaker */
58 /* need to write to 'index-46h' register : */
59 /* so write index num (here 0x46) to reg 0x6a */
60 /* and then 0xffff/0 to reg 0x6c */
61 snd_soc_write(w->codec, ALC5623_HID_CTRL_INDEX, 0x46);
64 case SND_SOC_DAPM_PRE_PMU:
65 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0xFFFF);
67 case SND_SOC_DAPM_POST_PMD:
68 snd_soc_write(w->codec, ALC5623_HID_CTRL_DATA, 0);
79 static const DECLARE_TLV_DB_SCALE(vol_tlv, -3450, 150, 0);
80 static const DECLARE_TLV_DB_SCALE(hp_tlv, -4650, 150, 0);
81 static const DECLARE_TLV_DB_SCALE(adc_rec_tlv, -1650, 150, 0);
82 static const unsigned int boost_tlv[] = {
84 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0),
85 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0),
86 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0),
88 static const DECLARE_TLV_DB_SCALE(dig_tlv, 0, 600, 0);
90 static const struct snd_kcontrol_new alc5621_vol_snd_controls[] = {
91 SOC_DOUBLE_TLV("Speaker Playback Volume",
92 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
93 SOC_DOUBLE("Speaker Playback Switch",
94 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
95 SOC_DOUBLE_TLV("Headphone Playback Volume",
96 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
97 SOC_DOUBLE("Headphone Playback Switch",
98 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
101 static const struct snd_kcontrol_new alc5622_vol_snd_controls[] = {
102 SOC_DOUBLE_TLV("Speaker Playback Volume",
103 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
104 SOC_DOUBLE("Speaker Playback Switch",
105 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
106 SOC_DOUBLE_TLV("Line Playback Volume",
107 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
108 SOC_DOUBLE("Line Playback Switch",
109 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
112 static const struct snd_kcontrol_new alc5623_vol_snd_controls[] = {
113 SOC_DOUBLE_TLV("Line Playback Volume",
114 ALC5623_SPK_OUT_VOL, 8, 0, 31, 1, hp_tlv),
115 SOC_DOUBLE("Line Playback Switch",
116 ALC5623_SPK_OUT_VOL, 15, 7, 1, 1),
117 SOC_DOUBLE_TLV("Headphone Playback Volume",
118 ALC5623_HP_OUT_VOL, 8, 0, 31, 1, hp_tlv),
119 SOC_DOUBLE("Headphone Playback Switch",
120 ALC5623_HP_OUT_VOL, 15, 7, 1, 1),
123 static const struct snd_kcontrol_new alc5623_snd_controls[] = {
124 SOC_DOUBLE_TLV("Auxout Playback Volume",
125 ALC5623_MONO_AUX_OUT_VOL, 8, 0, 31, 1, hp_tlv),
126 SOC_DOUBLE("Auxout Playback Switch",
127 ALC5623_MONO_AUX_OUT_VOL, 15, 7, 1, 1),
128 SOC_DOUBLE_TLV("PCM Playback Volume",
129 ALC5623_STEREO_DAC_VOL, 8, 0, 31, 1, vol_tlv),
130 SOC_DOUBLE_TLV("AuxI Capture Volume",
131 ALC5623_AUXIN_VOL, 8, 0, 31, 1, vol_tlv),
132 SOC_DOUBLE_TLV("LineIn Capture Volume",
133 ALC5623_LINE_IN_VOL, 8, 0, 31, 1, vol_tlv),
134 SOC_SINGLE_TLV("Mic1 Capture Volume",
135 ALC5623_MIC_VOL, 8, 31, 1, vol_tlv),
136 SOC_SINGLE_TLV("Mic2 Capture Volume",
137 ALC5623_MIC_VOL, 0, 31, 1, vol_tlv),
138 SOC_DOUBLE_TLV("Rec Capture Volume",
139 ALC5623_ADC_REC_GAIN, 7, 0, 31, 0, adc_rec_tlv),
140 SOC_SINGLE_TLV("Mic 1 Boost Volume",
141 ALC5623_MIC_CTRL, 10, 2, 0, boost_tlv),
142 SOC_SINGLE_TLV("Mic 2 Boost Volume",
143 ALC5623_MIC_CTRL, 8, 2, 0, boost_tlv),
144 SOC_SINGLE_TLV("Digital Boost Volume",
145 ALC5623_ADD_CTRL_REG, 4, 3, 0, dig_tlv),
151 static const struct snd_kcontrol_new alc5623_hp_mixer_controls[] = {
152 SOC_DAPM_SINGLE("LI2HP Playback Switch", ALC5623_LINE_IN_VOL, 15, 1, 1),
153 SOC_DAPM_SINGLE("AUXI2HP Playback Switch", ALC5623_AUXIN_VOL, 15, 1, 1),
154 SOC_DAPM_SINGLE("MIC12HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 15, 1, 1),
155 SOC_DAPM_SINGLE("MIC22HP Playback Switch", ALC5623_MIC_ROUTING_CTRL, 7, 1, 1),
156 SOC_DAPM_SINGLE("DAC2HP Playback Switch", ALC5623_STEREO_DAC_VOL, 15, 1, 1),
159 static const struct snd_kcontrol_new alc5623_hpl_mixer_controls[] = {
160 SOC_DAPM_SINGLE("ADC2HP_L Playback Switch", ALC5623_ADC_REC_GAIN, 15, 1, 1),
163 static const struct snd_kcontrol_new alc5623_hpr_mixer_controls[] = {
164 SOC_DAPM_SINGLE("ADC2HP_R Playback Switch", ALC5623_ADC_REC_GAIN, 14, 1, 1),
167 static const struct snd_kcontrol_new alc5623_mono_mixer_controls[] = {
168 SOC_DAPM_SINGLE("ADC2MONO_L Playback Switch", ALC5623_ADC_REC_GAIN, 13, 1, 1),
169 SOC_DAPM_SINGLE("ADC2MONO_R Playback Switch", ALC5623_ADC_REC_GAIN, 12, 1, 1),
170 SOC_DAPM_SINGLE("LI2MONO Playback Switch", ALC5623_LINE_IN_VOL, 13, 1, 1),
171 SOC_DAPM_SINGLE("AUXI2MONO Playback Switch", ALC5623_AUXIN_VOL, 13, 1, 1),
172 SOC_DAPM_SINGLE("MIC12MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 13, 1, 1),
173 SOC_DAPM_SINGLE("MIC22MONO Playback Switch", ALC5623_MIC_ROUTING_CTRL, 5, 1, 1),
174 SOC_DAPM_SINGLE("DAC2MONO Playback Switch", ALC5623_STEREO_DAC_VOL, 13, 1, 1),
177 static const struct snd_kcontrol_new alc5623_speaker_mixer_controls[] = {
178 SOC_DAPM_SINGLE("LI2SPK Playback Switch", ALC5623_LINE_IN_VOL, 14, 1, 1),
179 SOC_DAPM_SINGLE("AUXI2SPK Playback Switch", ALC5623_AUXIN_VOL, 14, 1, 1),
180 SOC_DAPM_SINGLE("MIC12SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 14, 1, 1),
181 SOC_DAPM_SINGLE("MIC22SPK Playback Switch", ALC5623_MIC_ROUTING_CTRL, 6, 1, 1),
182 SOC_DAPM_SINGLE("DAC2SPK Playback Switch", ALC5623_STEREO_DAC_VOL, 14, 1, 1),
185 /* Left Record Mixer */
186 static const struct snd_kcontrol_new alc5623_captureL_mixer_controls[] = {
187 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 14, 1, 1),
188 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 13, 1, 1),
189 SOC_DAPM_SINGLE("LineInL Capture Switch", ALC5623_ADC_REC_MIXER, 12, 1, 1),
190 SOC_DAPM_SINGLE("Left AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 11, 1, 1),
191 SOC_DAPM_SINGLE("HPMixerL Capture Switch", ALC5623_ADC_REC_MIXER, 10, 1, 1),
192 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 9, 1, 1),
193 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 8, 1, 1),
196 /* Right Record Mixer */
197 static const struct snd_kcontrol_new alc5623_captureR_mixer_controls[] = {
198 SOC_DAPM_SINGLE("Mic1 Capture Switch", ALC5623_ADC_REC_MIXER, 6, 1, 1),
199 SOC_DAPM_SINGLE("Mic2 Capture Switch", ALC5623_ADC_REC_MIXER, 5, 1, 1),
200 SOC_DAPM_SINGLE("LineInR Capture Switch", ALC5623_ADC_REC_MIXER, 4, 1, 1),
201 SOC_DAPM_SINGLE("Right AuxI Capture Switch", ALC5623_ADC_REC_MIXER, 3, 1, 1),
202 SOC_DAPM_SINGLE("HPMixerR Capture Switch", ALC5623_ADC_REC_MIXER, 2, 1, 1),
203 SOC_DAPM_SINGLE("SPKMixer Capture Switch", ALC5623_ADC_REC_MIXER, 1, 1, 1),
204 SOC_DAPM_SINGLE("MonoMixer Capture Switch", ALC5623_ADC_REC_MIXER, 0, 1, 1),
207 static const char *alc5623_spk_n_sour_sel[] = {
208 "RN/-R", "RP/+R", "LN/-R", "Vmid" };
209 static const char *alc5623_hpl_out_input_sel[] = {
210 "Vmid", "HP Left Mix"};
211 static const char *alc5623_hpr_out_input_sel[] = {
212 "Vmid", "HP Right Mix"};
213 static const char *alc5623_spkout_input_sel[] = {
214 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
215 static const char *alc5623_aux_out_input_sel[] = {
216 "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"};
218 /* auxout output mux */
219 static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum,
220 ALC5623_OUTPUT_MIXER_CTRL, 6,
221 alc5623_aux_out_input_sel);
222 static const struct snd_kcontrol_new alc5623_auxout_mux_controls =
223 SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum);
225 /* speaker output mux */
226 static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum,
227 ALC5623_OUTPUT_MIXER_CTRL, 10,
228 alc5623_spkout_input_sel);
229 static const struct snd_kcontrol_new alc5623_spkout_mux_controls =
230 SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum);
232 /* headphone left output mux */
233 static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum,
234 ALC5623_OUTPUT_MIXER_CTRL, 9,
235 alc5623_hpl_out_input_sel);
236 static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls =
237 SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum);
239 /* headphone right output mux */
240 static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum,
241 ALC5623_OUTPUT_MIXER_CTRL, 8,
242 alc5623_hpr_out_input_sel);
243 static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls =
244 SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum);
246 /* speaker output N select */
247 static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum,
248 ALC5623_OUTPUT_MIXER_CTRL, 14,
249 alc5623_spk_n_sour_sel);
250 static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls =
251 SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum);
253 static const struct snd_soc_dapm_widget alc5623_dapm_widgets[] = {
255 SND_SOC_DAPM_MUX("AuxOut Mux", SND_SOC_NOPM, 0, 0,
256 &alc5623_auxout_mux_controls),
257 SND_SOC_DAPM_MUX("SpeakerOut Mux", SND_SOC_NOPM, 0, 0,
258 &alc5623_spkout_mux_controls),
259 SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0,
260 &alc5623_hpl_out_mux_controls),
261 SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0,
262 &alc5623_hpr_out_mux_controls),
263 SND_SOC_DAPM_MUX("SpeakerOut N Mux", SND_SOC_NOPM, 0, 0,
264 &alc5623_spkoutn_mux_controls),
267 SND_SOC_DAPM_MIXER("HP Mix", SND_SOC_NOPM, 0, 0,
268 &alc5623_hp_mixer_controls[0],
269 ARRAY_SIZE(alc5623_hp_mixer_controls)),
270 SND_SOC_DAPM_MIXER("HPR Mix", ALC5623_PWR_MANAG_ADD2, 4, 0,
271 &alc5623_hpr_mixer_controls[0],
272 ARRAY_SIZE(alc5623_hpr_mixer_controls)),
273 SND_SOC_DAPM_MIXER("HPL Mix", ALC5623_PWR_MANAG_ADD2, 5, 0,
274 &alc5623_hpl_mixer_controls[0],
275 ARRAY_SIZE(alc5623_hpl_mixer_controls)),
276 SND_SOC_DAPM_MIXER("HPOut Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
277 SND_SOC_DAPM_MIXER("Mono Mix", ALC5623_PWR_MANAG_ADD2, 2, 0,
278 &alc5623_mono_mixer_controls[0],
279 ARRAY_SIZE(alc5623_mono_mixer_controls)),
280 SND_SOC_DAPM_MIXER("Speaker Mix", ALC5623_PWR_MANAG_ADD2, 3, 0,
281 &alc5623_speaker_mixer_controls[0],
282 ARRAY_SIZE(alc5623_speaker_mixer_controls)),
285 SND_SOC_DAPM_MIXER("Left Capture Mix", ALC5623_PWR_MANAG_ADD2, 1, 0,
286 &alc5623_captureL_mixer_controls[0],
287 ARRAY_SIZE(alc5623_captureL_mixer_controls)),
288 SND_SOC_DAPM_MIXER("Right Capture Mix", ALC5623_PWR_MANAG_ADD2, 0, 0,
289 &alc5623_captureR_mixer_controls[0],
290 ARRAY_SIZE(alc5623_captureR_mixer_controls)),
292 SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
293 ALC5623_PWR_MANAG_ADD2, 9, 0),
294 SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
295 ALC5623_PWR_MANAG_ADD2, 8, 0),
296 SND_SOC_DAPM_MIXER("I2S Mix", ALC5623_PWR_MANAG_ADD1, 15, 0, NULL, 0),
297 SND_SOC_DAPM_MIXER("AuxI Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
298 SND_SOC_DAPM_MIXER("Line Mix", SND_SOC_NOPM, 0, 0, NULL, 0),
299 SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture",
300 ALC5623_PWR_MANAG_ADD2, 7, 0),
301 SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture",
302 ALC5623_PWR_MANAG_ADD2, 6, 0),
303 SND_SOC_DAPM_PGA("Left Headphone", ALC5623_PWR_MANAG_ADD3, 10, 0, NULL, 0),
304 SND_SOC_DAPM_PGA("Right Headphone", ALC5623_PWR_MANAG_ADD3, 9, 0, NULL, 0),
305 SND_SOC_DAPM_PGA("SpeakerOut", ALC5623_PWR_MANAG_ADD3, 12, 0, NULL, 0),
306 SND_SOC_DAPM_PGA("Left AuxOut", ALC5623_PWR_MANAG_ADD3, 14, 0, NULL, 0),
307 SND_SOC_DAPM_PGA("Right AuxOut", ALC5623_PWR_MANAG_ADD3, 13, 0, NULL, 0),
308 SND_SOC_DAPM_PGA("Left LineIn", ALC5623_PWR_MANAG_ADD3, 7, 0, NULL, 0),
309 SND_SOC_DAPM_PGA("Right LineIn", ALC5623_PWR_MANAG_ADD3, 6, 0, NULL, 0),
310 SND_SOC_DAPM_PGA("Left AuxI", ALC5623_PWR_MANAG_ADD3, 5, 0, NULL, 0),
311 SND_SOC_DAPM_PGA("Right AuxI", ALC5623_PWR_MANAG_ADD3, 4, 0, NULL, 0),
312 SND_SOC_DAPM_PGA("MIC1 PGA", ALC5623_PWR_MANAG_ADD3, 3, 0, NULL, 0),
313 SND_SOC_DAPM_PGA("MIC2 PGA", ALC5623_PWR_MANAG_ADD3, 2, 0, NULL, 0),
314 SND_SOC_DAPM_PGA("MIC1 Pre Amp", ALC5623_PWR_MANAG_ADD3, 1, 0, NULL, 0),
315 SND_SOC_DAPM_PGA("MIC2 Pre Amp", ALC5623_PWR_MANAG_ADD3, 0, 0, NULL, 0),
316 SND_SOC_DAPM_MICBIAS("Mic Bias1", ALC5623_PWR_MANAG_ADD1, 11, 0),
318 SND_SOC_DAPM_OUTPUT("AUXOUTL"),
319 SND_SOC_DAPM_OUTPUT("AUXOUTR"),
320 SND_SOC_DAPM_OUTPUT("HPL"),
321 SND_SOC_DAPM_OUTPUT("HPR"),
322 SND_SOC_DAPM_OUTPUT("SPKOUT"),
323 SND_SOC_DAPM_OUTPUT("SPKOUTN"),
324 SND_SOC_DAPM_INPUT("LINEINL"),
325 SND_SOC_DAPM_INPUT("LINEINR"),
326 SND_SOC_DAPM_INPUT("AUXINL"),
327 SND_SOC_DAPM_INPUT("AUXINR"),
328 SND_SOC_DAPM_INPUT("MIC1"),
329 SND_SOC_DAPM_INPUT("MIC2"),
330 SND_SOC_DAPM_VMID("Vmid"),
333 static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"};
334 static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum,
335 ALC5623_OUTPUT_MIXER_CTRL, 13,
337 static const struct snd_kcontrol_new alc5623_amp_mux_controls =
338 SOC_DAPM_ENUM("Route", alc5623_amp_enum);
340 static const struct snd_soc_dapm_widget alc5623_dapm_amp_widgets[] = {
341 SND_SOC_DAPM_PGA_E("D Amp", ALC5623_PWR_MANAG_ADD2, 14, 0, NULL, 0,
342 amp_mixer_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
343 SND_SOC_DAPM_PGA("AB Amp", ALC5623_PWR_MANAG_ADD2, 15, 0, NULL, 0),
344 SND_SOC_DAPM_MUX("AB-D Amp Mux", SND_SOC_NOPM, 0, 0,
345 &alc5623_amp_mux_controls),
348 static const struct snd_soc_dapm_route intercon[] = {
349 /* virtual mixer - mixes left & right channels */
350 {"I2S Mix", NULL, "Left DAC"},
351 {"I2S Mix", NULL, "Right DAC"},
352 {"Line Mix", NULL, "Right LineIn"},
353 {"Line Mix", NULL, "Left LineIn"},
354 {"AuxI Mix", NULL, "Left AuxI"},
355 {"AuxI Mix", NULL, "Right AuxI"},
356 {"AUXOUTL", NULL, "Left AuxOut"},
357 {"AUXOUTR", NULL, "Right AuxOut"},
360 {"HPL Mix", "ADC2HP_L Playback Switch", "Left Capture Mix"},
361 {"HPL Mix", NULL, "HP Mix"},
362 {"HPR Mix", "ADC2HP_R Playback Switch", "Right Capture Mix"},
363 {"HPR Mix", NULL, "HP Mix"},
364 {"HP Mix", "LI2HP Playback Switch", "Line Mix"},
365 {"HP Mix", "AUXI2HP Playback Switch", "AuxI Mix"},
366 {"HP Mix", "MIC12HP Playback Switch", "MIC1 PGA"},
367 {"HP Mix", "MIC22HP Playback Switch", "MIC2 PGA"},
368 {"HP Mix", "DAC2HP Playback Switch", "I2S Mix"},
371 {"Speaker Mix", "LI2SPK Playback Switch", "Line Mix"},
372 {"Speaker Mix", "AUXI2SPK Playback Switch", "AuxI Mix"},
373 {"Speaker Mix", "MIC12SPK Playback Switch", "MIC1 PGA"},
374 {"Speaker Mix", "MIC22SPK Playback Switch", "MIC2 PGA"},
375 {"Speaker Mix", "DAC2SPK Playback Switch", "I2S Mix"},
378 {"Mono Mix", "ADC2MONO_L Playback Switch", "Left Capture Mix"},
379 {"Mono Mix", "ADC2MONO_R Playback Switch", "Right Capture Mix"},
380 {"Mono Mix", "LI2MONO Playback Switch", "Line Mix"},
381 {"Mono Mix", "AUXI2MONO Playback Switch", "AuxI Mix"},
382 {"Mono Mix", "MIC12MONO Playback Switch", "MIC1 PGA"},
383 {"Mono Mix", "MIC22MONO Playback Switch", "MIC2 PGA"},
384 {"Mono Mix", "DAC2MONO Playback Switch", "I2S Mix"},
386 /* Left record mixer */
387 {"Left Capture Mix", "LineInL Capture Switch", "LINEINL"},
388 {"Left Capture Mix", "Left AuxI Capture Switch", "AUXINL"},
389 {"Left Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
390 {"Left Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
391 {"Left Capture Mix", "HPMixerL Capture Switch", "HPL Mix"},
392 {"Left Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
393 {"Left Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
395 /*Right record mixer */
396 {"Right Capture Mix", "LineInR Capture Switch", "LINEINR"},
397 {"Right Capture Mix", "Right AuxI Capture Switch", "AUXINR"},
398 {"Right Capture Mix", "Mic1 Capture Switch", "MIC1 Pre Amp"},
399 {"Right Capture Mix", "Mic2 Capture Switch", "MIC2 Pre Amp"},
400 {"Right Capture Mix", "HPMixerR Capture Switch", "HPR Mix"},
401 {"Right Capture Mix", "SPKMixer Capture Switch", "Speaker Mix"},
402 {"Right Capture Mix", "MonoMixer Capture Switch", "Mono Mix"},
404 /* headphone left mux */
405 {"Left Headphone Mux", "HP Left Mix", "HPL Mix"},
406 {"Left Headphone Mux", "Vmid", "Vmid"},
408 /* headphone right mux */
409 {"Right Headphone Mux", "HP Right Mix", "HPR Mix"},
410 {"Right Headphone Mux", "Vmid", "Vmid"},
412 /* speaker out mux */
413 {"SpeakerOut Mux", "Vmid", "Vmid"},
414 {"SpeakerOut Mux", "HPOut Mix", "HPOut Mix"},
415 {"SpeakerOut Mux", "Speaker Mix", "Speaker Mix"},
416 {"SpeakerOut Mux", "Mono Mix", "Mono Mix"},
418 /* Mono/Aux Out mux */
419 {"AuxOut Mux", "Vmid", "Vmid"},
420 {"AuxOut Mux", "HPOut Mix", "HPOut Mix"},
421 {"AuxOut Mux", "Speaker Mix", "Speaker Mix"},
422 {"AuxOut Mux", "Mono Mix", "Mono Mix"},
425 {"HPL", NULL, "Left Headphone"},
426 {"Left Headphone", NULL, "Left Headphone Mux"},
427 {"HPR", NULL, "Right Headphone"},
428 {"Right Headphone", NULL, "Right Headphone Mux"},
429 {"Left AuxOut", NULL, "AuxOut Mux"},
430 {"Right AuxOut", NULL, "AuxOut Mux"},
433 {"Left LineIn", NULL, "LINEINL"},
434 {"Right LineIn", NULL, "LINEINR"},
435 {"Left AuxI", NULL, "AUXINL"},
436 {"Right AuxI", NULL, "AUXINR"},
437 {"MIC1 Pre Amp", NULL, "MIC1"},
438 {"MIC2 Pre Amp", NULL, "MIC2"},
439 {"MIC1 PGA", NULL, "MIC1 Pre Amp"},
440 {"MIC2 PGA", NULL, "MIC2 Pre Amp"},
443 {"Left ADC", NULL, "Left Capture Mix"},
446 {"Right ADC", NULL, "Right Capture Mix"},
448 {"SpeakerOut N Mux", "RN/-R", "SpeakerOut"},
449 {"SpeakerOut N Mux", "RP/+R", "SpeakerOut"},
450 {"SpeakerOut N Mux", "LN/-R", "SpeakerOut"},
451 {"SpeakerOut N Mux", "Vmid", "Vmid"},
453 {"SPKOUT", NULL, "SpeakerOut"},
454 {"SPKOUTN", NULL, "SpeakerOut N Mux"},
457 static const struct snd_soc_dapm_route intercon_spk[] = {
458 {"SpeakerOut", NULL, "SpeakerOut Mux"},
461 static const struct snd_soc_dapm_route intercon_amp_spk[] = {
462 {"AB Amp", NULL, "SpeakerOut Mux"},
463 {"D Amp", NULL, "SpeakerOut Mux"},
464 {"AB-D Amp Mux", "AB Amp", "AB Amp"},
465 {"AB-D Amp Mux", "D Amp", "D Amp"},
466 {"SpeakerOut", NULL, "AB-D Amp Mux"},
476 /* Note : pll code from original alc5623 driver. Not sure of how good it is */
477 /* useful only for master mode */
478 static const struct _pll_div codec_master_pll_div[] = {
480 { 2048000, 8192000, 0x0ea0},
481 { 3686400, 8192000, 0x4e27},
482 { 12000000, 8192000, 0x456b},
483 { 13000000, 8192000, 0x495f},
484 { 13100000, 8192000, 0x0320},
485 { 2048000, 11289600, 0xf637},
486 { 3686400, 11289600, 0x2f22},
487 { 12000000, 11289600, 0x3e2f},
488 { 13000000, 11289600, 0x4d5b},
489 { 13100000, 11289600, 0x363b},
490 { 2048000, 16384000, 0x1ea0},
491 { 3686400, 16384000, 0x9e27},
492 { 12000000, 16384000, 0x452b},
493 { 13000000, 16384000, 0x542f},
494 { 13100000, 16384000, 0x03a0},
495 { 2048000, 16934400, 0xe625},
496 { 3686400, 16934400, 0x9126},
497 { 12000000, 16934400, 0x4d2c},
498 { 13000000, 16934400, 0x742f},
499 { 13100000, 16934400, 0x3c27},
500 { 2048000, 22579200, 0x2aa0},
501 { 3686400, 22579200, 0x2f20},
502 { 12000000, 22579200, 0x7e2f},
503 { 13000000, 22579200, 0x742f},
504 { 13100000, 22579200, 0x3c27},
505 { 2048000, 24576000, 0x2ea0},
506 { 3686400, 24576000, 0xee27},
507 { 12000000, 24576000, 0x2915},
508 { 13000000, 24576000, 0x772e},
509 { 13100000, 24576000, 0x0d20},
512 static const struct _pll_div codec_slave_pll_div[] = {
514 { 1024000, 16384000, 0x3ea0},
515 { 1411200, 22579200, 0x3ea0},
516 { 1536000, 24576000, 0x3ea0},
517 { 2048000, 16384000, 0x1ea0},
518 { 2822400, 22579200, 0x1ea0},
519 { 3072000, 24576000, 0x1ea0},
523 static int alc5623_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
524 int source, unsigned int freq_in, unsigned int freq_out)
527 struct snd_soc_codec *codec = codec_dai->codec;
528 int gbl_clk = 0, pll_div = 0;
531 if (pll_id < ALC5623_PLL_FR_MCLK || pll_id > ALC5623_PLL_FR_BCK)
534 /* Disable PLL power */
535 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
536 ALC5623_PWR_ADD2_PLL,
539 /* pll is not used in slave mode */
540 reg = snd_soc_read(codec, ALC5623_DAI_CONTROL);
541 if (reg & ALC5623_DAI_SDP_SLAVE_MODE)
544 if (!freq_in || !freq_out)
548 case ALC5623_PLL_FR_MCLK:
549 for (i = 0; i < ARRAY_SIZE(codec_master_pll_div); i++) {
550 if (codec_master_pll_div[i].pll_in == freq_in
551 && codec_master_pll_div[i].pll_out == freq_out) {
552 /* PLL source from MCLK */
553 pll_div = codec_master_pll_div[i].regvalue;
558 case ALC5623_PLL_FR_BCK:
559 for (i = 0; i < ARRAY_SIZE(codec_slave_pll_div); i++) {
560 if (codec_slave_pll_div[i].pll_in == freq_in
561 && codec_slave_pll_div[i].pll_out == freq_out) {
562 /* PLL source from Bitclk */
563 gbl_clk = ALC5623_GBL_CLK_PLL_SOUR_SEL_BITCLK;
564 pll_div = codec_slave_pll_div[i].regvalue;
576 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
577 snd_soc_write(codec, ALC5623_PLL_CTRL, pll_div);
578 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD2,
579 ALC5623_PWR_ADD2_PLL,
580 ALC5623_PWR_ADD2_PLL);
581 gbl_clk |= ALC5623_GBL_CLK_SYS_SOUR_SEL_PLL;
582 snd_soc_write(codec, ALC5623_GLOBAL_CLK_CTRL_REG, gbl_clk);
592 /* codec hifi mclk (after PLL) clock divider coefficients */
593 /* values inspired from column BCLK=32Fs of Appendix A table */
594 static const struct _coeff_div coeff_div[] = {
605 static int get_coeff(struct snd_soc_codec *codec, int rate)
607 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
610 for (i = 0; i < ARRAY_SIZE(coeff_div); i++) {
611 if (coeff_div[i].fs * rate == alc5623->sysclk)
618 * Clock after PLL and dividers
620 static int alc5623_set_dai_sysclk(struct snd_soc_dai *codec_dai,
621 int clk_id, unsigned int freq, int dir)
623 struct snd_soc_codec *codec = codec_dai->codec;
624 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
635 alc5623->sysclk = freq;
641 static int alc5623_set_dai_fmt(struct snd_soc_dai *codec_dai,
644 struct snd_soc_codec *codec = codec_dai->codec;
647 /* set master/slave audio interface */
648 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
649 case SND_SOC_DAIFMT_CBM_CFM:
650 iface = ALC5623_DAI_SDP_MASTER_MODE;
652 case SND_SOC_DAIFMT_CBS_CFS:
653 iface = ALC5623_DAI_SDP_SLAVE_MODE;
659 /* interface format */
660 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
661 case SND_SOC_DAIFMT_I2S:
662 iface |= ALC5623_DAI_I2S_DF_I2S;
664 case SND_SOC_DAIFMT_RIGHT_J:
665 iface |= ALC5623_DAI_I2S_DF_RIGHT;
667 case SND_SOC_DAIFMT_LEFT_J:
668 iface |= ALC5623_DAI_I2S_DF_LEFT;
670 case SND_SOC_DAIFMT_DSP_A:
671 iface |= ALC5623_DAI_I2S_DF_PCM;
673 case SND_SOC_DAIFMT_DSP_B:
674 iface |= ALC5623_DAI_I2S_DF_PCM | ALC5623_DAI_I2S_PCM_MODE;
680 /* clock inversion */
681 switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
682 case SND_SOC_DAIFMT_NB_NF:
684 case SND_SOC_DAIFMT_IB_IF:
685 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
687 case SND_SOC_DAIFMT_IB_NF:
688 iface |= ALC5623_DAI_MAIN_I2S_BCLK_POL_CTRL;
690 case SND_SOC_DAIFMT_NB_IF:
696 return snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
699 static int alc5623_pcm_hw_params(struct snd_pcm_substream *substream,
700 struct snd_pcm_hw_params *params, struct snd_soc_dai *dai)
702 struct snd_soc_codec *codec = dai->codec;
703 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
707 iface = snd_soc_read(codec, ALC5623_DAI_CONTROL);
708 iface &= ~ALC5623_DAI_I2S_DL_MASK;
711 switch (params_width(params)) {
713 iface |= ALC5623_DAI_I2S_DL_16;
716 iface |= ALC5623_DAI_I2S_DL_20;
719 iface |= ALC5623_DAI_I2S_DL_24;
722 iface |= ALC5623_DAI_I2S_DL_32;
728 /* set iface & srate */
729 snd_soc_write(codec, ALC5623_DAI_CONTROL, iface);
730 rate = params_rate(params);
731 coeff = get_coeff(codec, rate);
735 coeff = coeff_div[coeff].regvalue;
736 dev_dbg(codec->dev, "%s: sysclk=%d,rate=%d,coeff=0x%04x\n",
737 __func__, alc5623->sysclk, rate, coeff);
738 snd_soc_write(codec, ALC5623_STEREO_AD_DA_CLK_CTRL, coeff);
743 static int alc5623_mute(struct snd_soc_dai *dai, int mute)
745 struct snd_soc_codec *codec = dai->codec;
746 u16 hp_mute = ALC5623_MISC_M_DAC_L_INPUT | ALC5623_MISC_M_DAC_R_INPUT;
747 u16 mute_reg = snd_soc_read(codec, ALC5623_MISC_CTRL) & ~hp_mute;
752 return snd_soc_write(codec, ALC5623_MISC_CTRL, mute_reg);
755 #define ALC5623_ADD2_POWER_EN (ALC5623_PWR_ADD2_VREF \
756 | ALC5623_PWR_ADD2_DAC_REF_CIR)
758 #define ALC5623_ADD3_POWER_EN (ALC5623_PWR_ADD3_MAIN_BIAS \
759 | ALC5623_PWR_ADD3_MIC1_BOOST_AD)
761 #define ALC5623_ADD1_POWER_EN \
762 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN | ALC5623_PWR_ADD1_SOFTGEN_EN \
763 | ALC5623_PWR_ADD1_DEPOP_BUF_HP | ALC5623_PWR_ADD1_HP_OUT_AMP \
764 | ALC5623_PWR_ADD1_HP_OUT_ENH_AMP)
766 #define ALC5623_ADD1_POWER_EN_5622 \
767 (ALC5623_PWR_ADD1_SHORT_CURR_DET_EN \
768 | ALC5623_PWR_ADD1_HP_OUT_AMP)
770 static void enable_power_depop(struct snd_soc_codec *codec)
772 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
774 snd_soc_update_bits(codec, ALC5623_PWR_MANAG_ADD1,
775 ALC5623_PWR_ADD1_SOFTGEN_EN,
776 ALC5623_PWR_ADD1_SOFTGEN_EN);
778 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, ALC5623_ADD3_POWER_EN);
780 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
781 ALC5623_MISC_HP_DEPOP_MODE2_EN,
782 ALC5623_MISC_HP_DEPOP_MODE2_EN);
786 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, ALC5623_ADD2_POWER_EN);
788 /* avoid writing '1' into 5622 reserved bits */
789 if (alc5623->id == 0x22)
790 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
791 ALC5623_ADD1_POWER_EN_5622);
793 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1,
794 ALC5623_ADD1_POWER_EN);
796 /* disable HP Depop2 */
797 snd_soc_update_bits(codec, ALC5623_MISC_CTRL,
798 ALC5623_MISC_HP_DEPOP_MODE2_EN,
803 static int alc5623_set_bias_level(struct snd_soc_codec *codec,
804 enum snd_soc_bias_level level)
807 case SND_SOC_BIAS_ON:
808 enable_power_depop(codec);
810 case SND_SOC_BIAS_PREPARE:
812 case SND_SOC_BIAS_STANDBY:
813 /* everything off except vref/vmid, */
814 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2,
815 ALC5623_PWR_ADD2_VREF);
816 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3,
817 ALC5623_PWR_ADD3_MAIN_BIAS);
819 case SND_SOC_BIAS_OFF:
820 /* everything off, dac mute, inactive */
821 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD2, 0);
822 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD3, 0);
823 snd_soc_write(codec, ALC5623_PWR_MANAG_ADD1, 0);
826 codec->dapm.bias_level = level;
830 #define ALC5623_FORMATS (SNDRV_PCM_FMTBIT_S16_LE \
831 | SNDRV_PCM_FMTBIT_S24_LE \
832 | SNDRV_PCM_FMTBIT_S32_LE)
834 static const struct snd_soc_dai_ops alc5623_dai_ops = {
835 .hw_params = alc5623_pcm_hw_params,
836 .digital_mute = alc5623_mute,
837 .set_fmt = alc5623_set_dai_fmt,
838 .set_sysclk = alc5623_set_dai_sysclk,
839 .set_pll = alc5623_set_dai_pll,
842 static struct snd_soc_dai_driver alc5623_dai = {
843 .name = "alc5623-hifi",
845 .stream_name = "Playback",
850 .rates = SNDRV_PCM_RATE_8000_48000,
851 .formats = ALC5623_FORMATS,},
853 .stream_name = "Capture",
858 .rates = SNDRV_PCM_RATE_8000_48000,
859 .formats = ALC5623_FORMATS,},
861 .ops = &alc5623_dai_ops,
864 static int alc5623_suspend(struct snd_soc_codec *codec)
866 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
868 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
869 regcache_cache_only(alc5623->regmap, true);
874 static int alc5623_resume(struct snd_soc_codec *codec)
876 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
879 /* Sync reg_cache with the hardware */
880 regcache_cache_only(alc5623->regmap, false);
881 ret = regcache_sync(alc5623->regmap);
883 dev_err(codec->dev, "Failed to sync register cache: %d\n",
885 regcache_cache_only(alc5623->regmap, true);
889 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
891 /* charge alc5623 caps */
892 if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) {
893 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
894 codec->dapm.bias_level = SND_SOC_BIAS_ON;
895 alc5623_set_bias_level(codec, codec->dapm.bias_level);
901 static int alc5623_probe(struct snd_soc_codec *codec)
903 struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec);
904 struct snd_soc_dapm_context *dapm = &codec->dapm;
906 alc5623_reset(codec);
908 /* power on device */
909 alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
911 if (alc5623->add_ctrl) {
912 snd_soc_write(codec, ALC5623_ADD_CTRL_REG,
916 if (alc5623->jack_det_ctrl) {
917 snd_soc_write(codec, ALC5623_JACK_DET_CTRL,
918 alc5623->jack_det_ctrl);
921 switch (alc5623->id) {
923 snd_soc_add_codec_controls(codec, alc5621_vol_snd_controls,
924 ARRAY_SIZE(alc5621_vol_snd_controls));
927 snd_soc_add_codec_controls(codec, alc5622_vol_snd_controls,
928 ARRAY_SIZE(alc5622_vol_snd_controls));
931 snd_soc_add_codec_controls(codec, alc5623_vol_snd_controls,
932 ARRAY_SIZE(alc5623_vol_snd_controls));
938 snd_soc_add_codec_controls(codec, alc5623_snd_controls,
939 ARRAY_SIZE(alc5623_snd_controls));
941 snd_soc_dapm_new_controls(dapm, alc5623_dapm_widgets,
942 ARRAY_SIZE(alc5623_dapm_widgets));
944 /* set up audio path interconnects */
945 snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
947 switch (alc5623->id) {
950 snd_soc_dapm_new_controls(dapm, alc5623_dapm_amp_widgets,
951 ARRAY_SIZE(alc5623_dapm_amp_widgets));
952 snd_soc_dapm_add_routes(dapm, intercon_amp_spk,
953 ARRAY_SIZE(intercon_amp_spk));
956 snd_soc_dapm_add_routes(dapm, intercon_spk,
957 ARRAY_SIZE(intercon_spk));
966 /* power down chip */
967 static int alc5623_remove(struct snd_soc_codec *codec)
969 alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF);
973 static struct snd_soc_codec_driver soc_codec_device_alc5623 = {
974 .probe = alc5623_probe,
975 .remove = alc5623_remove,
976 .suspend = alc5623_suspend,
977 .resume = alc5623_resume,
978 .set_bias_level = alc5623_set_bias_level,
981 static const struct regmap_config alc5623_regmap = {
986 .max_register = ALC5623_VENDOR_ID2,
987 .cache_type = REGCACHE_RBTREE,
991 * ALC5623 2 wire address is determined by A1 pin
992 * state during powerup.
996 static int alc5623_i2c_probe(struct i2c_client *client,
997 const struct i2c_device_id *id)
999 struct alc5623_platform_data *pdata;
1000 struct alc5623_priv *alc5623;
1001 unsigned int vid1, vid2;
1004 alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv),
1006 if (alc5623 == NULL)
1009 alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap);
1010 if (IS_ERR(alc5623->regmap)) {
1011 ret = PTR_ERR(alc5623->regmap);
1012 dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret);
1016 ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1);
1018 dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret);
1021 vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8);
1023 ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2);
1025 dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret);
1029 if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) {
1030 dev_err(&client->dev, "unknown or wrong codec\n");
1031 dev_err(&client->dev, "Expected %x:%lx, got %x:%x\n",
1032 0x10ec, id->driver_data,
1037 dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2);
1039 pdata = client->dev.platform_data;
1041 alc5623->add_ctrl = pdata->add_ctrl;
1042 alc5623->jack_det_ctrl = pdata->jack_det_ctrl;
1046 switch (alc5623->id) {
1048 alc5623_dai.name = "alc5621-hifi";
1051 alc5623_dai.name = "alc5622-hifi";
1054 alc5623_dai.name = "alc5623-hifi";
1060 i2c_set_clientdata(client, alc5623);
1062 ret = snd_soc_register_codec(&client->dev,
1063 &soc_codec_device_alc5623, &alc5623_dai, 1);
1065 dev_err(&client->dev, "Failed to register codec: %d\n", ret);
1070 static int alc5623_i2c_remove(struct i2c_client *client)
1072 snd_soc_unregister_codec(&client->dev);
1076 static const struct i2c_device_id alc5623_i2c_table[] = {
1082 MODULE_DEVICE_TABLE(i2c, alc5623_i2c_table);
1084 /* i2c codec control layer */
1085 static struct i2c_driver alc5623_i2c_driver = {
1087 .name = "alc562x-codec",
1088 .owner = THIS_MODULE,
1090 .probe = alc5623_i2c_probe,
1091 .remove = alc5623_i2c_remove,
1092 .id_table = alc5623_i2c_table,
1095 module_i2c_driver(alc5623_i2c_driver);
1097 MODULE_DESCRIPTION("ASoC alc5621/2/3 driver");
1098 MODULE_AUTHOR("Arnaud Patard <arnaud.patard@rtp-net.org>");
1099 MODULE_LICENSE("GPL");