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authorHaojian Zhuang <haojian.zhuang@gmail.com>2010-08-18 12:35:25 -0400
committerMark Brown <broonie@opensource.wolfsonmicro.com>2010-08-18 13:03:09 -0400
commitf213f4b51777408c12bf6b890a9bcae385f7698f (patch)
treeff05b9f6d2fd2069c5be7067e9fecaf6636a7f21 /sound
parentabfa4eae0bd2723859931631771ac275f97cada4 (diff)
ASoC: add 88pm860x codec driver
Add 88PM860x codec driver. 88PM860x codec supports two interfaces. And it also supports headset/mic/hook/short detection. Signed-off-by: Haojian Zhuang <haojian.zhuang@marvell.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound')
-rw-r--r--sound/soc/codecs/88pm860x-codec.c1486
-rw-r--r--sound/soc/codecs/88pm860x-codec.h97
-rw-r--r--sound/soc/codecs/Kconfig4
-rw-r--r--sound/soc/codecs/Makefile2
4 files changed, 1589 insertions, 0 deletions
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c
new file mode 100644
index 000000000000..01d19e9f53f9
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.c
@@ -0,0 +1,1486 @@
1/*
2 * 88pm860x-codec.c -- 88PM860x ALSA SoC Audio Driver
3 *
4 * Copyright 2010 Marvell International Ltd.
5 * Author: Haojian Zhuang <haojian.zhuang@marvell.com>
6 *
7 * This program is free software; you can redistribute it and/or modify
8 * it under the terms of the GNU General Public License version 2 as
9 * published by the Free Software Foundation.
10 */
11
12#include <linux/kernel.h>
13#include <linux/module.h>
14#include <linux/i2c.h>
15#include <linux/platform_device.h>
16#include <linux/mfd/88pm860x.h>
17#include <linux/slab.h>
18#include <sound/core.h>
19#include <sound/pcm.h>
20#include <sound/pcm_params.h>
21#include <sound/soc.h>
22#include <sound/soc-dapm.h>
23#include <sound/tlv.h>
24#include <sound/initval.h>
25#include <sound/jack.h>
26
27#include "88pm860x-codec.h"
28
29#define MAX_NAME_LEN 20
30#define REG_CACHE_SIZE 0x40
31#define REG_CACHE_BASE 0xb0
32
33/* Status Register 1 (0x01) */
34#define REG_STATUS_1 0x01
35#define MIC_STATUS (1 << 7)
36#define HOOK_STATUS (1 << 6)
37#define HEADSET_STATUS (1 << 5)
38
39/* Mic Detection Register (0x37) */
40#define REG_MIC_DET 0x37
41#define CONTINUOUS_POLLING (3 << 1)
42#define EN_MIC_DET (1 << 0)
43#define MICDET_MASK 0x07
44
45/* Headset Detection Register (0x38) */
46#define REG_HS_DET 0x38
47#define EN_HS_DET (1 << 0)
48
49/* Misc2 Register (0x42) */
50#define REG_MISC2 0x42
51#define AUDIO_PLL (1 << 5)
52#define AUDIO_SECTION_RESET (1 << 4)
53#define AUDIO_SECTION_ON (1 << 3)
54
55/* PCM Interface Register 2 (0xb1) */
56#define PCM_INF2_BCLK (1 << 6) /* Bit clock polarity */
57#define PCM_INF2_FS (1 << 5) /* Frame Sync polarity */
58#define PCM_INF2_MASTER (1 << 4) /* Master / Slave */
59#define PCM_INF2_18WL (1 << 3) /* 18 / 16 bits */
60#define PCM_GENERAL_I2S 0
61#define PCM_EXACT_I2S 1
62#define PCM_LEFT_I2S 2
63#define PCM_RIGHT_I2S 3
64#define PCM_SHORT_FS 4
65#define PCM_LONG_FS 5
66#define PCM_MODE_MASK 7
67
68/* I2S Interface Register 4 (0xbe) */
69#define I2S_EQU_BYP (1 << 6)
70
71/* DAC Offset Register (0xcb) */
72#define DAC_MUTE (1 << 7)
73#define MUTE_LEFT (1 << 6)
74#define MUTE_RIGHT (1 << 2)
75
76/* ADC Analog Register 1 (0xd0) */
77#define REG_ADC_ANA_1 0xd0
78#define MIC1BIAS_MASK 0x60
79
80/* Earpiece/Speaker Control Register 2 (0xda) */
81#define REG_EAR2 0xda
82#define RSYNC_CHANGE (1 << 2)
83
84/* Audio Supplies Register 2 (0xdc) */
85#define REG_SUPPLIES2 0xdc
86#define LDO15_READY (1 << 4)
87#define LDO15_EN (1 << 3)
88#define CPUMP_READY (1 << 2)
89#define CPUMP_EN (1 << 1)
90#define AUDIO_EN (1 << 0)
91#define SUPPLY_MASK (LDO15_EN | CPUMP_EN | AUDIO_EN)
92
93/* Audio Enable Register 1 (0xdd) */
94#define ADC_MOD_RIGHT (1 << 1)
95#define ADC_MOD_LEFT (1 << 0)
96
97/* Audio Enable Register 2 (0xde) */
98#define ADC_LEFT (1 << 5)
99#define ADC_RIGHT (1 << 4)
100
101/* DAC Enable Register 2 (0xe1) */
102#define DAC_LEFT (1 << 5)
103#define DAC_RIGHT (1 << 4)
104#define MODULATOR (1 << 3)
105
106/* Shorts Register (0xeb) */
107#define REG_SHORTS 0xeb
108#define CLR_SHORT_LO2 (1 << 7)
109#define SHORT_LO2 (1 << 6)
110#define CLR_SHORT_LO1 (1 << 5)
111#define SHORT_LO1 (1 << 4)
112#define CLR_SHORT_HS2 (1 << 3)
113#define SHORT_HS2 (1 << 2)
114#define CLR_SHORT_HS1 (1 << 1)
115#define SHORT_HS1 (1 << 0)
116
117/*
118 * This widget should be just after DAC & PGA in DAPM power-on sequence and
119 * before DAC & PGA in DAPM power-off sequence.
120 */
121#define PM860X_DAPM_OUTPUT(wname, wevent) \
122{ .id = snd_soc_dapm_pga, .name = wname, .reg = SND_SOC_NOPM, \
123 .shift = 0, .invert = 0, .kcontrols = NULL, \
124 .num_kcontrols = 0, .event = wevent, \
125 .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD, }
126
127struct pm860x_det {
128 struct snd_soc_jack *hp_jack;
129 struct snd_soc_jack *mic_jack;
130 int hp_det;
131 int mic_det;
132 int hook_det;
133 int hs_shrt;
134 int lo_shrt;
135};
136
137struct pm860x_priv {
138 unsigned int sysclk;
139 unsigned int pcmclk;
140 unsigned int dir;
141 unsigned int filter;
142 struct snd_soc_codec *codec;
143 struct i2c_client *i2c;
144 struct pm860x_chip *chip;
145 struct pm860x_det det;
146
147 int irq[4];
148 unsigned char name[4][MAX_NAME_LEN];
149 unsigned char reg_cache[REG_CACHE_SIZE];
150};
151
152/* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */
153static const DECLARE_TLV_DB_SCALE(dpga_tlv, -9450, 150, 1);
154
155/* -9dB to 0db in 3dB steps */
156static const DECLARE_TLV_DB_SCALE(adc_tlv, -900, 300, 0);
157
158/* {-23, -17, -13.5, -11, -9, -6, -3, 0}dB */
159static const unsigned int mic_tlv[] = {
160 TLV_DB_RANGE_HEAD(5),
161 0, 0, TLV_DB_SCALE_ITEM(-2300, 0, 0),
162 1, 1, TLV_DB_SCALE_ITEM(-1700, 0, 0),
163 2, 2, TLV_DB_SCALE_ITEM(-1350, 0, 0),
164 3, 3, TLV_DB_SCALE_ITEM(-1100, 0, 0),
165 4, 7, TLV_DB_SCALE_ITEM(-900, 300, 0),
166};
167
168/* {0, 0, 0, -6, 0, 6, 12, 18}dB */
169static const unsigned int aux_tlv[] = {
170 TLV_DB_RANGE_HEAD(2),
171 0, 2, TLV_DB_SCALE_ITEM(0, 0, 0),
172 3, 7, TLV_DB_SCALE_ITEM(-600, 600, 0),
173};
174
175/* {-16, -13, -10, -7, -5.2, -3,3, -2.2, 0}dB, mute instead of -16dB */
176static const unsigned int out_tlv[] = {
177 TLV_DB_RANGE_HEAD(4),
178 0, 3, TLV_DB_SCALE_ITEM(-1600, 300, 1),
179 4, 4, TLV_DB_SCALE_ITEM(-520, 0, 0),
180 5, 5, TLV_DB_SCALE_ITEM(-330, 0, 0),
181 6, 7, TLV_DB_SCALE_ITEM(-220, 220, 0),
182};
183
184static const unsigned int st_tlv[] = {
185 TLV_DB_RANGE_HEAD(8),
186 0, 1, TLV_DB_SCALE_ITEM(-12041, 602, 0),
187 2, 3, TLV_DB_SCALE_ITEM(-11087, 250, 0),
188 4, 5, TLV_DB_SCALE_ITEM(-10643, 158, 0),
189 6, 7, TLV_DB_SCALE_ITEM(-10351, 116, 0),
190 8, 9, TLV_DB_SCALE_ITEM(-10133, 92, 0),
191 10, 13, TLV_DB_SCALE_ITEM(-9958, 70, 0),
192 14, 17, TLV_DB_SCALE_ITEM(-9689, 53, 0),
193 18, 271, TLV_DB_SCALE_ITEM(-9484, 37, 0),
194};
195
196/* Sidetone Gain = M * 2^(-5-N) */
197struct st_gain {
198 unsigned int db;
199 unsigned int m;
200 unsigned int n;
201};
202
203static struct st_gain st_table[] = {
204 {-12041, 1, 15}, {-11439, 1, 14}, {-11087, 3, 15}, {-10837, 1, 13},
205 {-10643, 5, 15}, {-10485, 3, 14}, {-10351, 7, 15}, {-10235, 1, 12},
206 {-10133, 9, 15}, {-10041, 5, 14}, { -9958, 11, 15}, { -9883, 3, 13},
207 { -9813, 13, 15}, { -9749, 7, 14}, { -9689, 15, 15}, { -9633, 1, 11},
208 { -9580, 17, 15}, { -9531, 9, 14}, { -9484, 19, 15}, { -9439, 5, 13},
209 { -9397, 21, 15}, { -9356, 11, 14}, { -9318, 23, 15}, { -9281, 3, 12},
210 { -9245, 25, 15}, { -9211, 13, 14}, { -9178, 27, 15}, { -9147, 7, 13},
211 { -9116, 29, 15}, { -9087, 15, 14}, { -9058, 31, 15}, { -9031, 1, 10},
212 { -8978, 17, 14}, { -8929, 9, 13}, { -8882, 19, 14}, { -8837, 5, 12},
213 { -8795, 21, 14}, { -8754, 11, 13}, { -8716, 23, 14}, { -8679, 3, 11},
214 { -8643, 25, 14}, { -8609, 13, 13}, { -8576, 27, 14}, { -8545, 7, 12},
215 { -8514, 29, 14}, { -8485, 15, 13}, { -8456, 31, 14}, { -8429, 1, 9},
216 { -8376, 17, 13}, { -8327, 9, 12}, { -8280, 19, 13}, { -8235, 5, 11},
217 { -8193, 21, 13}, { -8152, 11, 12}, { -8114, 23, 13}, { -8077, 3, 10},
218 { -8041, 25, 13}, { -8007, 13, 12}, { -7974, 27, 13}, { -7943, 7, 11},
219 { -7912, 29, 13}, { -7883, 15, 12}, { -7854, 31, 13}, { -7827, 1, 8},
220 { -7774, 17, 12}, { -7724, 9, 11}, { -7678, 19, 12}, { -7633, 5, 10},
221 { -7591, 21, 12}, { -7550, 11, 11}, { -7512, 23, 12}, { -7475, 3, 9},
222 { -7439, 25, 12}, { -7405, 13, 11}, { -7372, 27, 12}, { -7341, 7, 10},
223 { -7310, 29, 12}, { -7281, 15, 11}, { -7252, 31, 12}, { -7225, 1, 7},
224 { -7172, 17, 11}, { -7122, 9, 10}, { -7075, 19, 11}, { -7031, 5, 9},
225 { -6989, 21, 11}, { -6948, 11, 10}, { -6910, 23, 11}, { -6873, 3, 8},
226 { -6837, 25, 11}, { -6803, 13, 10}, { -6770, 27, 11}, { -6739, 7, 9},
227 { -6708, 29, 11}, { -6679, 15, 10}, { -6650, 31, 11}, { -6623, 1, 6},
228 { -6570, 17, 10}, { -6520, 9, 9}, { -6473, 19, 10}, { -6429, 5, 8},
229 { -6386, 21, 10}, { -6346, 11, 9}, { -6307, 23, 10}, { -6270, 3, 7},
230 { -6235, 25, 10}, { -6201, 13, 9}, { -6168, 27, 10}, { -6137, 7, 8},
231 { -6106, 29, 10}, { -6077, 15, 9}, { -6048, 31, 10}, { -6021, 1, 5},
232 { -5968, 17, 9}, { -5918, 9, 8}, { -5871, 19, 9}, { -5827, 5, 7},
233 { -5784, 21, 9}, { -5744, 11, 8}, { -5705, 23, 9}, { -5668, 3, 6},
234 { -5633, 25, 9}, { -5599, 13, 8}, { -5566, 27, 9}, { -5535, 7, 7},
235 { -5504, 29, 9}, { -5475, 15, 8}, { -5446, 31, 9}, { -5419, 1, 4},
236 { -5366, 17, 8}, { -5316, 9, 7}, { -5269, 19, 8}, { -5225, 5, 6},
237 { -5182, 21, 8}, { -5142, 11, 7}, { -5103, 23, 8}, { -5066, 3, 5},
238 { -5031, 25, 8}, { -4997, 13, 7}, { -4964, 27, 8}, { -4932, 7, 6},
239 { -4902, 29, 8}, { -4873, 15, 7}, { -4844, 31, 8}, { -4816, 1, 3},
240 { -4764, 17, 7}, { -4714, 9, 6}, { -4667, 19, 7}, { -4623, 5, 5},
241 { -4580, 21, 7}, { -4540, 11, 6}, { -4501, 23, 7}, { -4464, 3, 4},
242 { -4429, 25, 7}, { -4395, 13, 6}, { -4362, 27, 7}, { -4330, 7, 5},
243 { -4300, 29, 7}, { -4270, 15, 6}, { -4242, 31, 7}, { -4214, 1, 2},
244 { -4162, 17, 6}, { -4112, 9, 5}, { -4065, 19, 6}, { -4021, 5, 4},
245 { -3978, 21, 6}, { -3938, 11, 5}, { -3899, 23, 6}, { -3862, 3, 3},
246 { -3827, 25, 6}, { -3793, 13, 5}, { -3760, 27, 6}, { -3728, 7, 4},
247 { -3698, 29, 6}, { -3668, 15, 5}, { -3640, 31, 6}, { -3612, 1, 1},
248 { -3560, 17, 5}, { -3510, 9, 4}, { -3463, 19, 5}, { -3419, 5, 3},
249 { -3376, 21, 5}, { -3336, 11, 4}, { -3297, 23, 5}, { -3260, 3, 2},
250 { -3225, 25, 5}, { -3191, 13, 4}, { -3158, 27, 5}, { -3126, 7, 3},
251 { -3096, 29, 5}, { -3066, 15, 4}, { -3038, 31, 5}, { -3010, 1, 0},
252 { -2958, 17, 4}, { -2908, 9, 3}, { -2861, 19, 4}, { -2816, 5, 2},
253 { -2774, 21, 4}, { -2734, 11, 3}, { -2695, 23, 4}, { -2658, 3, 1},
254 { -2623, 25, 4}, { -2589, 13, 3}, { -2556, 27, 4}, { -2524, 7, 2},
255 { -2494, 29, 4}, { -2464, 15, 3}, { -2436, 31, 4}, { -2408, 2, 0},
256 { -2356, 17, 3}, { -2306, 9, 2}, { -2259, 19, 3}, { -2214, 5, 1},
257 { -2172, 21, 3}, { -2132, 11, 2}, { -2093, 23, 3}, { -2056, 3, 0},
258 { -2021, 25, 3}, { -1987, 13, 2}, { -1954, 27, 3}, { -1922, 7, 1},
259 { -1892, 29, 3}, { -1862, 15, 2}, { -1834, 31, 3}, { -1806, 4, 0},
260 { -1754, 17, 2}, { -1704, 9, 1}, { -1657, 19, 2}, { -1612, 5, 0},
261 { -1570, 21, 2}, { -1530, 11, 1}, { -1491, 23, 2}, { -1454, 6, 0},
262 { -1419, 25, 2}, { -1384, 13, 1}, { -1352, 27, 2}, { -1320, 7, 0},
263 { -1290, 29, 2}, { -1260, 15, 1}, { -1232, 31, 2}, { -1204, 8, 0},
264 { -1151, 17, 1}, { -1102, 9, 0}, { -1055, 19, 1}, { -1010, 10, 0},
265 { -968, 21, 1}, { -928, 11, 0}, { -889, 23, 1}, { -852, 12, 0},
266 { -816, 25, 1}, { -782, 13, 0}, { -750, 27, 1}, { -718, 14, 0},
267 { -688, 29, 1}, { -658, 15, 0}, { -630, 31, 1}, { -602, 16, 0},
268 { -549, 17, 0}, { -500, 18, 0}, { -453, 19, 0}, { -408, 20, 0},
269 { -366, 21, 0}, { -325, 22, 0}, { -287, 23, 0}, { -250, 24, 0},
270 { -214, 25, 0}, { -180, 26, 0}, { -148, 27, 0}, { -116, 28, 0},
271 { -86, 29, 0}, { -56, 30, 0}, { -28, 31, 0}, { 0, 0, 0},
272};
273
274static int pm860x_volatile(unsigned int reg)
275{
276 BUG_ON(reg >= REG_CACHE_SIZE);
277
278 switch (reg) {
279 case PM860X_AUDIO_SUPPLIES_2:
280 return 1;
281 }
282
283 return 0;
284}
285
286static unsigned int pm860x_read_reg_cache(struct snd_soc_codec *codec,
287 unsigned int reg)
288{
289 unsigned char *cache = codec->reg_cache;
290
291 BUG_ON(reg >= REG_CACHE_SIZE);
292
293 if (pm860x_volatile(reg))
294 return cache[reg];
295
296 reg += REG_CACHE_BASE;
297
298 return pm860x_reg_read(codec->control_data, reg);
299}
300
301static int pm860x_write_reg_cache(struct snd_soc_codec *codec,
302 unsigned int reg, unsigned int value)
303{
304 unsigned char *cache = codec->reg_cache;
305
306 BUG_ON(reg >= REG_CACHE_SIZE);
307
308 if (!pm860x_volatile(reg))
309 cache[reg] = (unsigned char)value;
310
311 reg += REG_CACHE_BASE;
312
313 return pm860x_reg_write(codec->control_data, reg, value);
314}
315
316static int snd_soc_get_volsw_2r_st(struct snd_kcontrol *kcontrol,
317 struct snd_ctl_elem_value *ucontrol)
318{
319 struct soc_mixer_control *mc =
320 (struct soc_mixer_control *)kcontrol->private_value;
321 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
322 unsigned int reg = mc->reg;
323 unsigned int reg2 = mc->rreg;
324 int val[2], val2[2], i;
325
326 val[0] = snd_soc_read(codec, reg) & 0x3f;
327 val[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT) >> 4) & 0xf;
328 val2[0] = snd_soc_read(codec, reg2) & 0x3f;
329 val2[1] = (snd_soc_read(codec, PM860X_SIDETONE_SHIFT)) & 0xf;
330
331 for (i = 0; i < ARRAY_SIZE(st_table); i++) {
332 if ((st_table[i].m == val[0]) && (st_table[i].n == val[1]))
333 ucontrol->value.integer.value[0] = i;
334 if ((st_table[i].m == val2[0]) && (st_table[i].n == val2[1]))
335 ucontrol->value.integer.value[1] = i;
336 }
337 return 0;
338}
339
340static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol,
341 struct snd_ctl_elem_value *ucontrol)
342{
343 struct soc_mixer_control *mc =
344 (struct soc_mixer_control *)kcontrol->private_value;
345 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
346 unsigned int reg = mc->reg;
347 unsigned int reg2 = mc->rreg;
348 int err;
349 unsigned int val, val2;
350
351 val = ucontrol->value.integer.value[0];
352 val2 = ucontrol->value.integer.value[1];
353
354 err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m);
355 if (err < 0)
356 return err;
357 err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0xf0,
358 st_table[val].n << 4);
359 if (err < 0)
360 return err;
361
362 err = snd_soc_update_bits(codec, reg2, 0x3f, st_table[val2].m);
363 if (err < 0)
364 return err;
365 err = snd_soc_update_bits(codec, PM860X_SIDETONE_SHIFT, 0x0f,
366 st_table[val2].n);
367 return err;
368}
369
370static int snd_soc_get_volsw_2r_out(struct snd_kcontrol *kcontrol,
371 struct snd_ctl_elem_value *ucontrol)
372{
373 struct soc_mixer_control *mc =
374 (struct soc_mixer_control *)kcontrol->private_value;
375 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
376 unsigned int reg = mc->reg;
377 unsigned int reg2 = mc->rreg;
378 unsigned int shift = mc->shift;
379 int max = mc->max, val, val2;
380 unsigned int mask = (1 << fls(max)) - 1;
381
382 val = snd_soc_read(codec, reg) >> shift;
383 val2 = snd_soc_read(codec, reg2) >> shift;
384 ucontrol->value.integer.value[0] = (max - val) & mask;
385 ucontrol->value.integer.value[1] = (max - val2) & mask;
386
387 return 0;
388}
389
390static int snd_soc_put_volsw_2r_out(struct snd_kcontrol *kcontrol,
391 struct snd_ctl_elem_value *ucontrol)
392{
393 struct soc_mixer_control *mc =
394 (struct soc_mixer_control *)kcontrol->private_value;
395 struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
396 unsigned int reg = mc->reg;
397 unsigned int reg2 = mc->rreg;
398 unsigned int shift = mc->shift;
399 int max = mc->max;
400 unsigned int mask = (1 << fls(max)) - 1;
401 int err;
402 unsigned int val, val2, val_mask;
403
404 val_mask = mask << shift;
405 val = ((max - ucontrol->value.integer.value[0]) & mask);
406 val2 = ((max - ucontrol->value.integer.value[1]) & mask);
407
408 val = val << shift;
409 val2 = val2 << shift;
410
411 err = snd_soc_update_bits(codec, reg, val_mask, val);
412 if (err < 0)
413 return err;
414
415 err = snd_soc_update_bits(codec, reg2, val_mask, val2);
416 return err;
417}
418
419/* DAPM Widget Events */
420/*
421 * A lot registers are belong to RSYNC domain. It requires enabling RSYNC bit
422 * after updating these registers. Otherwise, these updated registers won't
423 * be effective.
424 */
425static int pm860x_rsync_event(struct snd_soc_dapm_widget *w,
426 struct snd_kcontrol *kcontrol, int event)
427{
428 struct snd_soc_codec *codec = w->codec;
429
430 /*
431 * In order to avoid current on the load, mute power-on and power-off
432 * should be transients.
433 * Unmute by DAC_MUTE. It should be unmuted when DAPM sequence is
434 * finished.
435 */
436 snd_soc_update_bits(codec, PM860X_DAC_OFFSET, DAC_MUTE, 0);
437 snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
438 RSYNC_CHANGE, RSYNC_CHANGE);
439 return 0;
440}
441
442static int pm860x_dac_event(struct snd_soc_dapm_widget *w,
443 struct snd_kcontrol *kcontrol, int event)
444{
445 struct snd_soc_codec *codec = w->codec;
446 unsigned int dac = 0;
447 int data;
448
449 if (!strcmp(w->name, "Left DAC"))
450 dac = DAC_LEFT;
451 if (!strcmp(w->name, "Right DAC"))
452 dac = DAC_RIGHT;
453 switch (event) {
454 case SND_SOC_DAPM_PRE_PMU:
455 if (dac) {
456 /* Auto mute in power-on sequence. */
457 dac |= MODULATOR;
458 snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
459 DAC_MUTE, DAC_MUTE);
460 snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
461 RSYNC_CHANGE, RSYNC_CHANGE);
462 /* update dac */
463 snd_soc_update_bits(codec, PM860X_DAC_EN_2,
464 dac, dac);
465 }
466 break;
467 case SND_SOC_DAPM_PRE_PMD:
468 if (dac) {
469 /* Auto mute in power-off sequence. */
470 snd_soc_update_bits(codec, PM860X_DAC_OFFSET,
471 DAC_MUTE, DAC_MUTE);
472 snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
473 RSYNC_CHANGE, RSYNC_CHANGE);
474 /* update dac */
475 data = snd_soc_read(codec, PM860X_DAC_EN_2);
476 data &= ~dac;
477 if (!(data & (DAC_LEFT | DAC_RIGHT)))
478 data &= ~MODULATOR;
479 snd_soc_write(codec, PM860X_DAC_EN_2, data);
480 }
481 break;
482 }
483 return 0;
484}
485
486static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"};
487
488static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"};
489
490static const struct soc_enum pm860x_hs1_opamp_enum =
491 SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts);
492
493static const struct soc_enum pm860x_hs2_opamp_enum =
494 SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts);
495
496static const struct soc_enum pm860x_hs1_pa_enum =
497 SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts);
498
499static const struct soc_enum pm860x_hs2_pa_enum =
500 SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts);
501
502static const struct soc_enum pm860x_lo1_opamp_enum =
503 SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts);
504
505static const struct soc_enum pm860x_lo2_opamp_enum =
506 SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts);
507
508static const struct soc_enum pm860x_lo1_pa_enum =
509 SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts);
510
511static const struct soc_enum pm860x_lo2_pa_enum =
512 SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts);
513
514static const struct soc_enum pm860x_spk_pa_enum =
515 SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts);
516
517static const struct soc_enum pm860x_ear_pa_enum =
518 SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts);
519
520static const struct soc_enum pm860x_spk_ear_opamp_enum =
521 SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts);
522
523static const struct snd_kcontrol_new pm860x_snd_controls[] = {
524 SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2,
525 PM860X_ADC_ANA_3, 6, 3, 0, adc_tlv),
526 SOC_DOUBLE_TLV("AUX Capture Volume", PM860X_ADC_ANA_3, 0, 3, 7, 0,
527 aux_tlv),
528 SOC_SINGLE_TLV("MIC1 Capture Volume", PM860X_ADC_ANA_2, 0, 7, 0,
529 mic_tlv),
530 SOC_SINGLE_TLV("MIC3 Capture Volume", PM860X_ADC_ANA_2, 3, 7, 0,
531 mic_tlv),
532 SOC_DOUBLE_R_EXT_TLV("Sidetone Volume", PM860X_SIDETONE_L_GAIN,
533 PM860X_SIDETONE_R_GAIN, 0, ARRAY_SIZE(st_table)-1,
534 0, snd_soc_get_volsw_2r_st,
535 snd_soc_put_volsw_2r_st, st_tlv),
536 SOC_SINGLE_TLV("Speaker Playback Volume", PM860X_EAR_CTRL_1,
537 0, 7, 0, out_tlv),
538 SOC_DOUBLE_R_TLV("Line Playback Volume", PM860X_LO1_CTRL,
539 PM860X_LO2_CTRL, 0, 7, 0, out_tlv),
540 SOC_DOUBLE_R_TLV("Headset Playback Volume", PM860X_HS1_CTRL,
541 PM860X_HS2_CTRL, 0, 7, 0, out_tlv),
542 SOC_DOUBLE_R_EXT_TLV("Hifi Left Playback Volume",
543 PM860X_HIFIL_GAIN_LEFT,
544 PM860X_HIFIL_GAIN_RIGHT, 0, 63, 0,
545 snd_soc_get_volsw_2r_out,
546 snd_soc_put_volsw_2r_out, dpga_tlv),
547 SOC_DOUBLE_R_EXT_TLV("Hifi Right Playback Volume",
548 PM860X_HIFIR_GAIN_LEFT,
549 PM860X_HIFIR_GAIN_RIGHT, 0, 63, 0,
550 snd_soc_get_volsw_2r_out,
551 snd_soc_put_volsw_2r_out, dpga_tlv),
552 SOC_DOUBLE_R_EXT_TLV("Lofi Playback Volume", PM860X_LOFI_GAIN_LEFT,
553 PM860X_LOFI_GAIN_RIGHT, 0, 63, 0,
554 snd_soc_get_volsw_2r_out,
555 snd_soc_put_volsw_2r_out, dpga_tlv),
556 SOC_ENUM("Headset1 Operational Amplifier Current",
557 pm860x_hs1_opamp_enum),
558 SOC_ENUM("Headset2 Operational Amplifier Current",
559 pm860x_hs2_opamp_enum),
560 SOC_ENUM("Headset1 Amplifier Current", pm860x_hs1_pa_enum),
561 SOC_ENUM("Headset2 Amplifier Current", pm860x_hs2_pa_enum),
562 SOC_ENUM("Lineout1 Operational Amplifier Current",
563 pm860x_lo1_opamp_enum),
564 SOC_ENUM("Lineout2 Operational Amplifier Current",
565 pm860x_lo2_opamp_enum),
566 SOC_ENUM("Lineout1 Amplifier Current", pm860x_lo1_pa_enum),
567 SOC_ENUM("Lineout2 Amplifier Current", pm860x_lo2_pa_enum),
568 SOC_ENUM("Speaker Operational Amplifier Current",
569 pm860x_spk_ear_opamp_enum),
570 SOC_ENUM("Speaker Amplifier Current", pm860x_spk_pa_enum),
571 SOC_ENUM("Earpiece Amplifier Current", pm860x_ear_pa_enum),
572};
573
574/*
575 * DAPM Controls
576 */
577
578/* PCM Switch / PCM Interface */
579static const struct snd_kcontrol_new pcm_switch_controls =
580 SOC_DAPM_SINGLE("Switch", PM860X_ADC_EN_2, 0, 1, 0);
581
582/* AUX1 Switch */
583static const struct snd_kcontrol_new aux1_switch_controls =
584 SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 4, 1, 0);
585
586/* AUX2 Switch */
587static const struct snd_kcontrol_new aux2_switch_controls =
588 SOC_DAPM_SINGLE("Switch", PM860X_ANA_TO_ANA, 5, 1, 0);
589
590/* Left Ex. PA Switch */
591static const struct snd_kcontrol_new lepa_switch_controls =
592 SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 2, 1, 0);
593
594/* Right Ex. PA Switch */
595static const struct snd_kcontrol_new repa_switch_controls =
596 SOC_DAPM_SINGLE("Switch", PM860X_DAC_EN_2, 1, 1, 0);
597
598/* PCM Mux / Mux7 */
599static const char *aif1_text[] = {
600 "PCM L", "PCM R",
601};
602
603static const struct soc_enum aif1_enum =
604 SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text);
605
606static const struct snd_kcontrol_new aif1_mux =
607 SOC_DAPM_ENUM("PCM Mux", aif1_enum);
608
609/* I2S Mux / Mux9 */
610static const char *i2s_din_text[] = {
611 "DIN", "DIN1",
612};
613
614static const struct soc_enum i2s_din_enum =
615 SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text);
616
617static const struct snd_kcontrol_new i2s_din_mux =
618 SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum);
619
620/* I2S Mic Mux / Mux8 */
621static const char *i2s_mic_text[] = {
622 "Ex PA", "ADC",
623};
624
625static const struct soc_enum i2s_mic_enum =
626 SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text);
627
628static const struct snd_kcontrol_new i2s_mic_mux =
629 SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum);
630
631/* ADCL Mux / Mux2 */
632static const char *adcl_text[] = {
633 "ADCR", "ADCL",
634};
635
636static const struct soc_enum adcl_enum =
637 SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text);
638
639static const struct snd_kcontrol_new adcl_mux =
640 SOC_DAPM_ENUM("ADC Left Mux", adcl_enum);
641
642/* ADCR Mux / Mux3 */
643static const char *adcr_text[] = {
644 "ADCL", "ADCR",
645};
646
647static const struct soc_enum adcr_enum =
648 SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text);
649
650static const struct snd_kcontrol_new adcr_mux =
651 SOC_DAPM_ENUM("ADC Right Mux", adcr_enum);
652
653/* ADCR EC Mux / Mux6 */
654static const char *adcr_ec_text[] = {
655 "ADCR", "EC",
656};
657
658static const struct soc_enum adcr_ec_enum =
659 SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text);
660
661static const struct snd_kcontrol_new adcr_ec_mux =
662 SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum);
663
664/* EC Mux / Mux4 */
665static const char *ec_text[] = {
666 "Left", "Right", "Left + Right",
667};
668
669static const struct soc_enum ec_enum =
670 SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text);
671
672static const struct snd_kcontrol_new ec_mux =
673 SOC_DAPM_ENUM("EC Mux", ec_enum);
674
675static const char *dac_text[] = {
676 "No input", "Right", "Left", "No input",
677};
678
679/* DAC Headset 1 Mux / Mux10 */
680static const struct soc_enum dac_hs1_enum =
681 SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text);
682
683static const struct snd_kcontrol_new dac_hs1_mux =
684 SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum);
685
686/* DAC Headset 2 Mux / Mux11 */
687static const struct soc_enum dac_hs2_enum =
688 SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text);
689
690static const struct snd_kcontrol_new dac_hs2_mux =
691 SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum);
692
693/* DAC Lineout 1 Mux / Mux12 */
694static const struct soc_enum dac_lo1_enum =
695 SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text);
696
697static const struct snd_kcontrol_new dac_lo1_mux =
698 SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum);
699
700/* DAC Lineout 2 Mux / Mux13 */
701static const struct soc_enum dac_lo2_enum =
702 SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text);
703
704static const struct snd_kcontrol_new dac_lo2_mux =
705 SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum);
706
707/* DAC Spearker Earphone Mux / Mux14 */
708static const struct soc_enum dac_spk_ear_enum =
709 SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text);
710
711static const struct snd_kcontrol_new dac_spk_ear_mux =
712 SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum);
713
714/* Headset 1 Mux / Mux15 */
715static const char *in_text[] = {
716 "Digital", "Analog",
717};
718
719static const struct soc_enum hs1_enum =
720 SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text);
721
722static const struct snd_kcontrol_new hs1_mux =
723 SOC_DAPM_ENUM("Headset1 Mux", hs1_enum);
724
725/* Headset 2 Mux / Mux16 */
726static const struct soc_enum hs2_enum =
727 SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text);
728
729static const struct snd_kcontrol_new hs2_mux =
730 SOC_DAPM_ENUM("Headset2 Mux", hs2_enum);
731
732/* Lineout 1 Mux / Mux17 */
733static const struct soc_enum lo1_enum =
734 SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text);
735
736static const struct snd_kcontrol_new lo1_mux =
737 SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum);
738
739/* Lineout 2 Mux / Mux18 */
740static const struct soc_enum lo2_enum =
741 SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text);
742
743static const struct snd_kcontrol_new lo2_mux =
744 SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum);
745
746/* Speaker Earpiece Demux */
747static const char *spk_text[] = {
748 "Earpiece", "Speaker",
749};
750
751static const struct soc_enum spk_enum =
752 SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text);
753
754static const struct snd_kcontrol_new spk_demux =
755 SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum);
756
757/* MIC Mux / Mux1 */
758static const char *mic_text[] = {
759 "Mic 1", "Mic 2",
760};
761
762static const struct soc_enum mic_enum =
763 SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text);
764
765static const struct snd_kcontrol_new mic_mux =
766 SOC_DAPM_ENUM("MIC Mux", mic_enum);
767
768static const struct snd_soc_dapm_widget pm860x_dapm_widgets[] = {
769 SND_SOC_DAPM_AIF_IN("PCM SDI", "PCM Playback", 0,
770 PM860X_ADC_EN_2, 0, 0),
771 SND_SOC_DAPM_AIF_OUT("PCM SDO", "PCM Capture", 0,
772 PM860X_PCM_IFACE_3, 1, 1),
773
774
775 SND_SOC_DAPM_AIF_IN("I2S DIN", "I2S Playback", 0,
776 PM860X_DAC_EN_2, 0, 0),
777 SND_SOC_DAPM_AIF_IN("I2S DIN1", "I2S Playback", 0,
778 PM860X_DAC_EN_2, 0, 0),
779 SND_SOC_DAPM_AIF_OUT("I2S DOUT", "I2S Capture", 0,
780 PM860X_I2S_IFACE_3, 5, 1),
781 SND_SOC_DAPM_MUX("I2S Mic Mux", SND_SOC_NOPM, 0, 0, &i2s_mic_mux),
782 SND_SOC_DAPM_MUX("ADC Left Mux", SND_SOC_NOPM, 0, 0, &adcl_mux),
783 SND_SOC_DAPM_MUX("ADC Right Mux", SND_SOC_NOPM, 0, 0, &adcr_mux),
784 SND_SOC_DAPM_MUX("EC Mux", SND_SOC_NOPM, 0, 0, &ec_mux),
785 SND_SOC_DAPM_MUX("ADCR EC Mux", SND_SOC_NOPM, 0, 0, &adcr_ec_mux),
786 SND_SOC_DAPM_SWITCH("Left EPA", SND_SOC_NOPM, 0, 0,
787 &lepa_switch_controls),
788 SND_SOC_DAPM_SWITCH("Right EPA", SND_SOC_NOPM, 0, 0,
789 &repa_switch_controls),
790
791 SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Left ADC MOD", PM860X_ADC_EN_1,
792 0, 1, 1, 0),
793 SND_SOC_DAPM_REG(snd_soc_dapm_supply, "Right ADC MOD", PM860X_ADC_EN_1,
794 1, 1, 1, 0),
795 SND_SOC_DAPM_ADC("Left ADC", NULL, PM860X_ADC_EN_2, 5, 0),
796 SND_SOC_DAPM_ADC("Right ADC", NULL, PM860X_ADC_EN_2, 4, 0),
797
798 SND_SOC_DAPM_SWITCH("AUX1 Switch", SND_SOC_NOPM, 0, 0,
799 &aux1_switch_controls),
800 SND_SOC_DAPM_SWITCH("AUX2 Switch", SND_SOC_NOPM, 0, 0,
801 &aux2_switch_controls),
802
803 SND_SOC_DAPM_MUX("MIC Mux", SND_SOC_NOPM, 0, 0, &mic_mux),
804 SND_SOC_DAPM_MICBIAS("Mic1 Bias", PM860X_ADC_ANA_1, 2, 0),
805 SND_SOC_DAPM_MICBIAS("Mic3 Bias", PM860X_ADC_ANA_1, 7, 0),
806 SND_SOC_DAPM_PGA("MIC1 Volume", PM860X_ADC_EN_1, 2, 0, NULL, 0),
807 SND_SOC_DAPM_PGA("MIC3 Volume", PM860X_ADC_EN_1, 3, 0, NULL, 0),
808 SND_SOC_DAPM_PGA("AUX1 Volume", PM860X_ADC_EN_1, 4, 0, NULL, 0),
809 SND_SOC_DAPM_PGA("AUX2 Volume", PM860X_ADC_EN_1, 5, 0, NULL, 0),
810 SND_SOC_DAPM_PGA("Sidetone PGA", PM860X_ADC_EN_2, 1, 0, NULL, 0),
811 SND_SOC_DAPM_PGA("Lofi PGA", PM860X_ADC_EN_2, 2, 0, NULL, 0),
812
813 SND_SOC_DAPM_INPUT("AUX1"),
814 SND_SOC_DAPM_INPUT("AUX2"),
815 SND_SOC_DAPM_INPUT("MIC1P"),
816 SND_SOC_DAPM_INPUT("MIC1N"),
817 SND_SOC_DAPM_INPUT("MIC2P"),
818 SND_SOC_DAPM_INPUT("MIC2N"),
819 SND_SOC_DAPM_INPUT("MIC3P"),
820 SND_SOC_DAPM_INPUT("MIC3N"),
821
822 SND_SOC_DAPM_DAC_E("Left DAC", NULL, SND_SOC_NOPM, 0, 0,
823 pm860x_dac_event,
824 SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
825 SND_SOC_DAPM_DAC_E("Right DAC", NULL, SND_SOC_NOPM, 0, 0,
826 pm860x_dac_event,
827 SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_PRE_PMD),
828
829 SND_SOC_DAPM_MUX("I2S DIN Mux", SND_SOC_NOPM, 0, 0, &i2s_din_mux),
830 SND_SOC_DAPM_MUX("DAC HS1 Mux", SND_SOC_NOPM, 0, 0, &dac_hs1_mux),
831 SND_SOC_DAPM_MUX("DAC HS2 Mux", SND_SOC_NOPM, 0, 0, &dac_hs2_mux),
832 SND_SOC_DAPM_MUX("DAC LO1 Mux", SND_SOC_NOPM, 0, 0, &dac_lo1_mux),
833 SND_SOC_DAPM_MUX("DAC LO2 Mux", SND_SOC_NOPM, 0, 0, &dac_lo2_mux),
834 SND_SOC_DAPM_MUX("DAC SP Mux", SND_SOC_NOPM, 0, 0, &dac_spk_ear_mux),
835 SND_SOC_DAPM_MUX("Headset1 Mux", SND_SOC_NOPM, 0, 0, &hs1_mux),
836 SND_SOC_DAPM_MUX("Headset2 Mux", SND_SOC_NOPM, 0, 0, &hs2_mux),
837 SND_SOC_DAPM_MUX("Lineout1 Mux", SND_SOC_NOPM, 0, 0, &lo1_mux),
838 SND_SOC_DAPM_MUX("Lineout2 Mux", SND_SOC_NOPM, 0, 0, &lo2_mux),
839 SND_SOC_DAPM_MUX("Speaker Earpiece Demux", SND_SOC_NOPM, 0, 0,
840 &spk_demux),
841
842
843 SND_SOC_DAPM_PGA("Headset1 PGA", PM860X_DAC_EN_1, 0, 0, NULL, 0),
844 SND_SOC_DAPM_PGA("Headset2 PGA", PM860X_DAC_EN_1, 1, 0, NULL, 0),
845 SND_SOC_DAPM_OUTPUT("HS1"),
846 SND_SOC_DAPM_OUTPUT("HS2"),
847 SND_SOC_DAPM_PGA("Lineout1 PGA", PM860X_DAC_EN_1, 2, 0, NULL, 0),
848 SND_SOC_DAPM_PGA("Lineout2 PGA", PM860X_DAC_EN_1, 3, 0, NULL, 0),
849 SND_SOC_DAPM_OUTPUT("LINEOUT1"),
850 SND_SOC_DAPM_OUTPUT("LINEOUT2"),
851 SND_SOC_DAPM_PGA("Earpiece PGA", PM860X_DAC_EN_1, 4, 0, NULL, 0),
852 SND_SOC_DAPM_OUTPUT("EARP"),
853 SND_SOC_DAPM_OUTPUT("EARN"),
854 SND_SOC_DAPM_PGA("Speaker PGA", PM860X_DAC_EN_1, 5, 0, NULL, 0),
855 SND_SOC_DAPM_OUTPUT("LSP"),
856 SND_SOC_DAPM_OUTPUT("LSN"),
857 SND_SOC_DAPM_REG(snd_soc_dapm_supply, "VCODEC", PM860X_AUDIO_SUPPLIES_2,
858 0, SUPPLY_MASK, SUPPLY_MASK, 0),
859
860 PM860X_DAPM_OUTPUT("RSYNC", pm860x_rsync_event),
861};
862
863static const struct snd_soc_dapm_route audio_map[] = {
864 /* supply */
865 {"Left DAC", NULL, "VCODEC"},
866 {"Right DAC", NULL, "VCODEC"},
867 {"Left ADC", NULL, "VCODEC"},
868 {"Right ADC", NULL, "VCODEC"},
869 {"Left ADC", NULL, "Left ADC MOD"},
870 {"Right ADC", NULL, "Right ADC MOD"},
871
872 /* PCM/AIF1 Inputs */
873 {"PCM SDO", NULL, "ADC Left Mux"},
874 {"PCM SDO", NULL, "ADCR EC Mux"},
875
876 /* PCM/AFI2 Outputs */
877 {"Lofi PGA", NULL, "PCM SDI"},
878 {"Lofi PGA", NULL, "Sidetone PGA"},
879 {"Left DAC", NULL, "Lofi PGA"},
880 {"Right DAC", NULL, "Lofi PGA"},
881
882 /* I2S/AIF2 Inputs */
883 {"MIC Mux", "Mic 1", "MIC1P"},
884 {"MIC Mux", "Mic 1", "MIC1N"},
885 {"MIC Mux", "Mic 2", "MIC2P"},
886 {"MIC Mux", "Mic 2", "MIC2N"},
887 {"MIC1 Volume", NULL, "MIC Mux"},
888 {"MIC3 Volume", NULL, "MIC3P"},
889 {"MIC3 Volume", NULL, "MIC3N"},
890 {"Left ADC", NULL, "MIC1 Volume"},
891 {"Right ADC", NULL, "MIC3 Volume"},
892 {"ADC Left Mux", "ADCR", "Right ADC"},
893 {"ADC Left Mux", "ADCL", "Left ADC"},
894 {"ADC Right Mux", "ADCL", "Left ADC"},
895 {"ADC Right Mux", "ADCR", "Right ADC"},
896 {"Left EPA", "Switch", "Left DAC"},
897 {"Right EPA", "Switch", "Right DAC"},
898 {"EC Mux", "Left", "Left DAC"},
899 {"EC Mux", "Right", "Right DAC"},
900 {"EC Mux", "Left + Right", "Left DAC"},
901 {"EC Mux", "Left + Right", "Right DAC"},
902 {"ADCR EC Mux", "ADCR", "ADC Right Mux"},
903 {"ADCR EC Mux", "EC", "EC Mux"},
904 {"I2S Mic Mux", "Ex PA", "Left EPA"},
905 {"I2S Mic Mux", "Ex PA", "Right EPA"},
906 {"I2S Mic Mux", "ADC", "ADC Left Mux"},
907 {"I2S Mic Mux", "ADC", "ADCR EC Mux"},
908 {"I2S DOUT", NULL, "I2S Mic Mux"},
909
910 /* I2S/AIF2 Outputs */
911 {"I2S DIN Mux", "DIN", "I2S DIN"},
912 {"I2S DIN Mux", "DIN1", "I2S DIN1"},
913 {"Left DAC", NULL, "I2S DIN Mux"},
914 {"Right DAC", NULL, "I2S DIN Mux"},
915 {"DAC HS1 Mux", "Left", "Left DAC"},
916 {"DAC HS1 Mux", "Right", "Right DAC"},
917 {"DAC HS2 Mux", "Left", "Left DAC"},
918 {"DAC HS2 Mux", "Right", "Right DAC"},
919 {"DAC LO1 Mux", "Left", "Left DAC"},
920 {"DAC LO1 Mux", "Right", "Right DAC"},
921 {"DAC LO2 Mux", "Left", "Left DAC"},
922 {"DAC LO2 Mux", "Right", "Right DAC"},
923 {"Headset1 Mux", "Digital", "DAC HS1 Mux"},
924 {"Headset2 Mux", "Digital", "DAC HS2 Mux"},
925 {"Lineout1 Mux", "Digital", "DAC LO1 Mux"},
926 {"Lineout2 Mux", "Digital", "DAC LO2 Mux"},
927 {"Headset1 PGA", NULL, "Headset1 Mux"},
928 {"Headset2 PGA", NULL, "Headset2 Mux"},
929 {"Lineout1 PGA", NULL, "Lineout1 Mux"},
930 {"Lineout2 PGA", NULL, "Lineout2 Mux"},
931 {"DAC SP Mux", "Left", "Left DAC"},
932 {"DAC SP Mux", "Right", "Right DAC"},
933 {"Speaker Earpiece Demux", "Speaker", "DAC SP Mux"},
934 {"Speaker PGA", NULL, "Speaker Earpiece Demux"},
935 {"Earpiece PGA", NULL, "Speaker Earpiece Demux"},
936
937 {"RSYNC", NULL, "Headset1 PGA"},
938 {"RSYNC", NULL, "Headset2 PGA"},
939 {"RSYNC", NULL, "Lineout1 PGA"},
940 {"RSYNC", NULL, "Lineout2 PGA"},
941 {"RSYNC", NULL, "Speaker PGA"},
942 {"RSYNC", NULL, "Speaker PGA"},
943 {"RSYNC", NULL, "Earpiece PGA"},
944 {"RSYNC", NULL, "Earpiece PGA"},
945
946 {"HS1", NULL, "RSYNC"},
947 {"HS2", NULL, "RSYNC"},
948 {"LINEOUT1", NULL, "RSYNC"},
949 {"LINEOUT2", NULL, "RSYNC"},
950 {"LSP", NULL, "RSYNC"},
951 {"LSN", NULL, "RSYNC"},
952 {"EARP", NULL, "RSYNC"},
953 {"EARN", NULL, "RSYNC"},
954};
955
956/*
957 * Use MUTE_LEFT & MUTE_RIGHT to implement digital mute.
958 * These bits can also be used to mute.
959 */
960static int pm860x_digital_mute(struct snd_soc_dai *codec_dai, int mute)
961{
962 struct snd_soc_codec *codec = codec_dai->codec;
963 int data = 0, mask = MUTE_LEFT | MUTE_RIGHT;
964
965 if (mute)
966 data = mask;
967 snd_soc_update_bits(codec, PM860X_DAC_OFFSET, mask, data);
968 snd_soc_update_bits(codec, PM860X_EAR_CTRL_2,
969 RSYNC_CHANGE, RSYNC_CHANGE);
970 return 0;
971}
972
973static int pm860x_pcm_hw_params(struct snd_pcm_substream *substream,
974 struct snd_pcm_hw_params *params,
975 struct snd_soc_dai *dai)
976{
977 struct snd_soc_codec *codec = dai->codec;
978 unsigned char inf = 0, mask = 0;
979
980 /* bit size */
981 switch (params_format(params)) {
982 case SNDRV_PCM_FORMAT_S16_LE:
983 inf &= ~PCM_INF2_18WL;
984 break;
985 case SNDRV_PCM_FORMAT_S18_3LE:
986 inf |= PCM_INF2_18WL;
987 break;
988 default:
989 return -EINVAL;
990 }
991 mask |= PCM_INF2_18WL;
992 snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
993
994 /* sample rate */
995 switch (params_rate(params)) {
996 case 8000:
997 inf = 0;
998 break;
999 case 16000:
1000 inf = 3;
1001 break;
1002 case 32000:
1003 inf = 6;
1004 break;
1005 case 48000:
1006 inf = 8;
1007 break;
1008 default:
1009 return -EINVAL;
1010 }
1011 snd_soc_update_bits(codec, PM860X_PCM_RATE, 0x0f, inf);
1012
1013 return 0;
1014}
1015
1016static int pm860x_pcm_set_dai_fmt(struct snd_soc_dai *codec_dai,
1017 unsigned int fmt)
1018{
1019 struct snd_soc_codec *codec = codec_dai->codec;
1020 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1021 unsigned char inf = 0, mask = 0;
1022 int ret = -EINVAL;
1023
1024 mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
1025
1026 /* set master/slave audio interface */
1027 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
1028 case SND_SOC_DAIFMT_CBM_CFM:
1029 case SND_SOC_DAIFMT_CBM_CFS:
1030 if (pm860x->dir == PM860X_CLK_DIR_OUT) {
1031 inf |= PCM_INF2_MASTER;
1032 ret = 0;
1033 }
1034 break;
1035 case SND_SOC_DAIFMT_CBS_CFS:
1036 if (pm860x->dir == PM860X_CLK_DIR_IN) {
1037 inf &= ~PCM_INF2_MASTER;
1038 ret = 0;
1039 }
1040 break;
1041 }
1042
1043 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
1044 case SND_SOC_DAIFMT_I2S:
1045 inf |= PCM_EXACT_I2S;
1046 ret = 0;
1047 break;
1048 }
1049 mask |= PCM_MODE_MASK;
1050 if (ret)
1051 return ret;
1052 snd_soc_update_bits(codec, PM860X_PCM_IFACE_2, mask, inf);
1053 return 0;
1054}
1055
1056static int pm860x_set_dai_sysclk(struct snd_soc_dai *codec_dai,
1057 int clk_id, unsigned int freq, int dir)
1058{
1059 struct snd_soc_codec *codec = codec_dai->codec;
1060 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1061
1062 if (dir == PM860X_CLK_DIR_OUT)
1063 pm860x->dir = PM860X_CLK_DIR_OUT;
1064 else {
1065 pm860x->dir = PM860X_CLK_DIR_IN;
1066 return -EINVAL;
1067 }
1068
1069 return 0;
1070}
1071
1072static int pm860x_i2s_hw_params(struct snd_pcm_substream *substream,
1073 struct snd_pcm_hw_params *params,
1074 struct snd_soc_dai *dai)
1075{
1076 struct snd_soc_codec *codec = dai->codec;
1077 unsigned char inf;
1078
1079 /* bit size */
1080 switch (params_format(params)) {
1081 case SNDRV_PCM_FORMAT_S16_LE:
1082 inf = 0;
1083 break;
1084 case SNDRV_PCM_FORMAT_S18_3LE:
1085 inf = PCM_INF2_18WL;
1086 break;
1087 default:
1088 return -EINVAL;
1089 }
1090 snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, PCM_INF2_18WL, inf);
1091
1092 /* sample rate */
1093 switch (params_rate(params)) {
1094 case 8000:
1095 inf = 0;
1096 break;
1097 case 11025:
1098 inf = 1;
1099 break;
1100 case 16000:
1101 inf = 3;
1102 break;
1103 case 22050:
1104 inf = 4;
1105 break;
1106 case 32000:
1107 inf = 6;
1108 break;
1109 case 44100:
1110 inf = 7;
1111 break;
1112 case 48000:
1113 inf = 8;
1114 break;
1115 default:
1116 return -EINVAL;
1117 }
1118 snd_soc_update_bits(codec, PM860X_I2S_IFACE_4, 0xf, inf);
1119
1120 return 0;
1121}
1122
1123static int pm860x_i2s_set_dai_fmt(struct snd_soc_dai *codec_dai,
1124 unsigned int fmt)
1125{
1126 struct snd_soc_codec *codec = codec_dai->codec;
1127 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1128 unsigned char inf = 0, mask = 0;
1129
1130 mask |= PCM_INF2_BCLK | PCM_INF2_FS | PCM_INF2_MASTER;
1131
1132 /* set master/slave audio interface */
1133 switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
1134 case SND_SOC_DAIFMT_CBM_CFM:
1135 if (pm860x->dir == PM860X_CLK_DIR_OUT)
1136 inf |= PCM_INF2_MASTER;
1137 else
1138 return -EINVAL;
1139 break;
1140 case SND_SOC_DAIFMT_CBS_CFS:
1141 if (pm860x->dir == PM860X_CLK_DIR_IN)
1142 inf &= ~PCM_INF2_MASTER;
1143 else
1144 return -EINVAL;
1145 break;
1146 default:
1147 return -EINVAL;
1148 }
1149
1150 switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
1151 case SND_SOC_DAIFMT_I2S:
1152 inf |= PCM_EXACT_I2S;
1153 break;
1154 default:
1155 return -EINVAL;
1156 }
1157 mask |= PCM_MODE_MASK;
1158 snd_soc_update_bits(codec, PM860X_I2S_IFACE_2, mask, inf);
1159 return 0;
1160}
1161
1162static int pm860x_set_bias_level(struct snd_soc_codec *codec,
1163 enum snd_soc_bias_level level)
1164{
1165 int data;
1166
1167 switch (level) {
1168 case SND_SOC_BIAS_ON:
1169 break;
1170
1171 case SND_SOC_BIAS_PREPARE:
1172 break;
1173
1174 case SND_SOC_BIAS_STANDBY:
1175 if (codec->bias_level == SND_SOC_BIAS_OFF) {
1176 /* Enable Audio PLL & Audio section */
1177 data = AUDIO_PLL | AUDIO_SECTION_RESET
1178 | AUDIO_SECTION_ON;
1179 pm860x_reg_write(codec->control_data, REG_MISC2, data);
1180 }
1181 break;
1182
1183 case SND_SOC_BIAS_OFF:
1184 data = AUDIO_PLL | AUDIO_SECTION_RESET | AUDIO_SECTION_ON;
1185 pm860x_set_bits(codec->control_data, REG_MISC2, data, 0);
1186 break;
1187 }
1188 codec->bias_level = level;
1189 return 0;
1190}
1191
1192static struct snd_soc_dai_ops pm860x_pcm_dai_ops = {
1193 .digital_mute = pm860x_digital_mute,
1194 .hw_params = pm860x_pcm_hw_params,
1195 .set_fmt = pm860x_pcm_set_dai_fmt,
1196 .set_sysclk = pm860x_set_dai_sysclk,
1197};
1198
1199static struct snd_soc_dai_ops pm860x_i2s_dai_ops = {
1200 .digital_mute = pm860x_digital_mute,
1201 .hw_params = pm860x_i2s_hw_params,
1202 .set_fmt = pm860x_i2s_set_dai_fmt,
1203 .set_sysclk = pm860x_set_dai_sysclk,
1204};
1205
1206#define PM860X_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
1207 SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_48000)
1208
1209static struct snd_soc_dai_driver pm860x_dai[] = {
1210 {
1211 /* DAI PCM */
1212 .name = "88pm860x-pcm",
1213 .id = 1,
1214 .playback = {
1215 .stream_name = "PCM Playback",
1216 .channels_min = 2,
1217 .channels_max = 2,
1218 .rates = PM860X_RATES,
1219 .formats = SNDRV_PCM_FORMAT_S16_LE | \
1220 SNDRV_PCM_FORMAT_S18_3LE,
1221 },
1222 .capture = {
1223 .stream_name = "PCM Capture",
1224 .channels_min = 2,
1225 .channels_max = 2,
1226 .rates = PM860X_RATES,
1227 .formats = SNDRV_PCM_FORMAT_S16_LE | \
1228 SNDRV_PCM_FORMAT_S18_3LE,
1229 },
1230 .ops = &pm860x_pcm_dai_ops,
1231 }, {
1232 /* DAI I2S */
1233 .name = "88pm860x-i2s",
1234 .id = 2,
1235 .playback = {
1236 .stream_name = "I2S Playback",
1237 .channels_min = 2,
1238 .channels_max = 2,
1239 .rates = SNDRV_PCM_RATE_8000_48000,
1240 .formats = SNDRV_PCM_FORMAT_S16_LE | \
1241 SNDRV_PCM_FORMAT_S18_3LE,
1242 },
1243 .capture = {
1244 .stream_name = "I2S Capture",
1245 .channels_min = 2,
1246 .channels_max = 2,
1247 .rates = SNDRV_PCM_RATE_8000_48000,
1248 .formats = SNDRV_PCM_FORMAT_S16_LE | \
1249 SNDRV_PCM_FORMAT_S18_3LE,
1250 },
1251 .ops = &pm860x_i2s_dai_ops,
1252 },
1253};
1254
1255static irqreturn_t pm860x_codec_handler(int irq, void *data)
1256{
1257 struct pm860x_priv *pm860x = data;
1258 int status, shrt, report = 0, mic_report = 0;
1259 int mask;
1260
1261 status = pm860x_reg_read(pm860x->i2c, REG_STATUS_1);
1262 shrt = pm860x_reg_read(pm860x->i2c, REG_SHORTS);
1263 mask = pm860x->det.hs_shrt | pm860x->det.hook_det | pm860x->det.lo_shrt
1264 | pm860x->det.hp_det;
1265
1266 if ((pm860x->det.hp_det & SND_JACK_HEADPHONE)
1267 && (status & HEADSET_STATUS))
1268 report |= SND_JACK_HEADPHONE;
1269
1270 if ((pm860x->det.mic_det & SND_JACK_MICROPHONE)
1271 && (status & MIC_STATUS))
1272 mic_report |= SND_JACK_MICROPHONE;
1273
1274 if (pm860x->det.hs_shrt && (shrt & (SHORT_HS1 | SHORT_HS2)))
1275 report |= pm860x->det.hs_shrt;
1276
1277 if (pm860x->det.hook_det && (status & HOOK_STATUS))
1278 report |= pm860x->det.hook_det;
1279
1280 if (pm860x->det.lo_shrt && (shrt & (SHORT_LO1 | SHORT_LO2)))
1281 report |= pm860x->det.lo_shrt;
1282
1283 if (report)
1284 snd_soc_jack_report(pm860x->det.hp_jack, report, mask);
1285 if (mic_report)
1286 snd_soc_jack_report(pm860x->det.mic_jack, SND_JACK_MICROPHONE,
1287 SND_JACK_MICROPHONE);
1288
1289 dev_dbg(pm860x->codec->dev, "headphone report:0x%x, mask:%x\n",
1290 report, mask);
1291 dev_dbg(pm860x->codec->dev, "microphone report:0x%x\n", mic_report);
1292 return IRQ_HANDLED;
1293}
1294
1295int pm860x_hs_jack_detect(struct snd_soc_codec *codec,
1296 struct snd_soc_jack *jack,
1297 int det, int hook, int hs_shrt, int lo_shrt)
1298{
1299 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1300 int data;
1301
1302 pm860x->det.hp_jack = jack;
1303 pm860x->det.hp_det = det;
1304 pm860x->det.hook_det = hook;
1305 pm860x->det.hs_shrt = hs_shrt;
1306 pm860x->det.lo_shrt = lo_shrt;
1307
1308 if (det & SND_JACK_HEADPHONE)
1309 pm860x_set_bits(codec->control_data, REG_HS_DET,
1310 EN_HS_DET, EN_HS_DET);
1311 /* headset short detect */
1312 if (hs_shrt) {
1313 data = CLR_SHORT_HS2 | CLR_SHORT_HS1;
1314 pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
1315 }
1316 /* Lineout short detect */
1317 if (lo_shrt) {
1318 data = CLR_SHORT_LO2 | CLR_SHORT_LO1;
1319 pm860x_set_bits(codec->control_data, REG_SHORTS, data, data);
1320 }
1321
1322 /* sync status */
1323 pm860x_codec_handler(0, pm860x);
1324 return 0;
1325}
1326EXPORT_SYMBOL_GPL(pm860x_hs_jack_detect);
1327
1328int pm860x_mic_jack_detect(struct snd_soc_codec *codec,
1329 struct snd_soc_jack *jack, int det)
1330{
1331 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1332
1333 pm860x->det.mic_jack = jack;
1334 pm860x->det.mic_det = det;
1335
1336 if (det & SND_JACK_MICROPHONE)
1337 pm860x_set_bits(codec->control_data, REG_MIC_DET,
1338 MICDET_MASK, MICDET_MASK);
1339
1340 /* sync status */
1341 pm860x_codec_handler(0, pm860x);
1342 return 0;
1343}
1344EXPORT_SYMBOL_GPL(pm860x_mic_jack_detect);
1345
1346static int pm860x_probe(struct snd_soc_codec *codec)
1347{
1348 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1349 int i, ret;
1350
1351 pm860x->codec = codec;
1352
1353 codec->control_data = pm860x->i2c;
1354
1355 for (i = 0; i < 4; i++) {
1356 ret = request_threaded_irq(pm860x->irq[i], NULL,
1357 pm860x_codec_handler, IRQF_ONESHOT,
1358 pm860x->name[i], pm860x);
1359 if (ret < 0) {
1360 dev_err(codec->dev, "Failed to request IRQ!\n");
1361 goto out_irq;
1362 }
1363 }
1364
1365 pm860x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
1366
1367 ret = pm860x_bulk_read(codec->control_data, REG_CACHE_BASE,
1368 REG_CACHE_SIZE, codec->reg_cache);
1369 if (ret < 0) {
1370 dev_err(codec->dev, "Failed to fill register cache: %d\n",
1371 ret);
1372 goto out_codec;
1373 }
1374
1375 snd_soc_add_controls(codec, pm860x_snd_controls,
1376 ARRAY_SIZE(pm860x_snd_controls));
1377 snd_soc_dapm_new_controls(codec, pm860x_dapm_widgets,
1378 ARRAY_SIZE(pm860x_dapm_widgets));
1379 snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
1380 return 0;
1381
1382out_codec:
1383 i = 3;
1384out_irq:
1385 for (; i >= 0; i--)
1386 free_irq(pm860x->irq[i], pm860x);
1387 return -EINVAL;
1388}
1389
1390static int pm860x_remove(struct snd_soc_codec *codec)
1391{
1392 struct pm860x_priv *pm860x = snd_soc_codec_get_drvdata(codec);
1393 int i;
1394
1395 for (i = 3; i >= 0; i--)
1396 free_irq(pm860x->irq[i], pm860x);
1397 pm860x_set_bias_level(codec, SND_SOC_BIAS_OFF);
1398 return 0;
1399}
1400
1401static struct snd_soc_codec_driver soc_codec_dev_pm860x = {
1402 .probe = pm860x_probe,
1403 .remove = pm860x_remove,
1404 .read = pm860x_read_reg_cache,
1405 .write = pm860x_write_reg_cache,
1406 .reg_cache_size = REG_CACHE_SIZE,
1407 .reg_word_size = sizeof(u8),
1408 .set_bias_level = pm860x_set_bias_level,
1409};
1410
1411static int __devinit pm860x_codec_probe(struct platform_device *pdev)
1412{
1413 struct pm860x_chip *chip = dev_get_drvdata(pdev->dev.parent);
1414 struct pm860x_priv *pm860x;
1415 struct resource *res;
1416 int i, ret;
1417
1418 pm860x = kzalloc(sizeof(struct pm860x_priv), GFP_KERNEL);
1419 if (pm860x == NULL)
1420 return -ENOMEM;
1421
1422 pm860x->chip = chip;
1423 pm860x->i2c = (chip->id == CHIP_PM8607) ? chip->client
1424 : chip->companion;
1425 platform_set_drvdata(pdev, pm860x);
1426
1427 for (i = 0; i < 4; i++) {
1428 res = platform_get_resource(pdev, IORESOURCE_IRQ, i);
1429 if (!res) {
1430 dev_err(&pdev->dev, "Failed to get IRQ resources\n");
1431 goto out;
1432 }
1433 pm860x->irq[i] = res->start + chip->irq_base;
1434 strncpy(pm860x->name[i], res->name, MAX_NAME_LEN);
1435 }
1436
1437 ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_pm860x,
1438 pm860x_dai, ARRAY_SIZE(pm860x_dai));
1439 if (ret) {
1440 dev_err(&pdev->dev, "Failed to register codec\n");
1441 goto out;
1442 }
1443 return ret;
1444
1445out:
1446 platform_set_drvdata(pdev, NULL);
1447 kfree(pm860x);
1448 return -EINVAL;
1449}
1450
1451static int __devexit pm860x_codec_remove(struct platform_device *pdev)
1452{
1453 struct pm860x_priv *pm860x = platform_get_drvdata(pdev);
1454
1455 snd_soc_unregister_codec(&pdev->dev);
1456 platform_set_drvdata(pdev, NULL);
1457 kfree(pm860x);
1458 return 0;
1459}
1460
1461static struct platform_driver pm860x_codec_driver = {
1462 .driver = {
1463 .name = "88pm860x-codec",
1464 .owner = THIS_MODULE,
1465 },
1466 .probe = pm860x_codec_probe,
1467 .remove = __devexit_p(pm860x_codec_remove),
1468};
1469
1470static __init int pm860x_init(void)
1471{
1472 return platform_driver_register(&pm860x_codec_driver);
1473}
1474module_init(pm860x_init);
1475
1476static __exit void pm860x_exit(void)
1477{
1478 platform_driver_unregister(&pm860x_codec_driver);
1479}
1480module_exit(pm860x_exit);
1481
1482MODULE_DESCRIPTION("ASoC 88PM860x driver");
1483MODULE_AUTHOR("Haojian Zhuang <haojian.zhuang@marvell.com>");
1484MODULE_LICENSE("GPL");
1485MODULE_ALIAS("platform:88pm860x-codec");
1486
diff --git a/sound/soc/codecs/88pm860x-codec.h b/sound/soc/codecs/88pm860x-codec.h
new file mode 100644
index 000000000000..3364ba4a3607
--- /dev/null
+++ b/sound/soc/codecs/88pm860x-codec.h
@@ -0,0 +1,97 @@
1/*
2 * 88pm860x-codec.h -- 88PM860x ALSA SoC Audio Driver
3 *
4 * Copyright 2010 Marvell International Ltd.
5 * Haojian Zhuang <haojian.zhuang@marvell.com>
6 *
7 * This program is free software; you can redistribute it and/or modify
8 * it under the terms of the GNU General Public License version 2 as
9 * published by the Free Software Foundation.
10 */
11
12#ifndef __88PM860X_H
13#define __88PM860X_H
14
15/* The offset of these registers are 0xb0 */
16#define PM860X_PCM_IFACE_1 0x00
17#define PM860X_PCM_IFACE_2 0x01
18#define PM860X_PCM_IFACE_3 0x02
19#define PM860X_PCM_RATE 0x03
20#define PM860X_EC_PATH 0x04
21#define PM860X_SIDETONE_L_GAIN 0x05
22#define PM860X_SIDETONE_R_GAIN 0x06
23#define PM860X_SIDETONE_SHIFT 0x07
24#define PM860X_ADC_OFFSET_1 0x08
25#define PM860X_ADC_OFFSET_2 0x09
26#define PM860X_DMIC_DELAY 0x0a
27
28#define PM860X_I2S_IFACE_1 0x0b
29#define PM860X_I2S_IFACE_2 0x0c
30#define PM860X_I2S_IFACE_3 0x0d
31#define PM860X_I2S_IFACE_4 0x0e
32#define PM860X_EQUALIZER_N0_1 0x0f
33#define PM860X_EQUALIZER_N0_2 0x10
34#define PM860X_EQUALIZER_N1_1 0x11
35#define PM860X_EQUALIZER_N1_2 0x12
36#define PM860X_EQUALIZER_D1_1 0x13
37#define PM860X_EQUALIZER_D1_2 0x14
38#define PM860X_LOFI_GAIN_LEFT 0x15
39#define PM860X_LOFI_GAIN_RIGHT 0x16
40#define PM860X_HIFIL_GAIN_LEFT 0x17
41#define PM860X_HIFIL_GAIN_RIGHT 0x18
42#define PM860X_HIFIR_GAIN_LEFT 0x19
43#define PM860X_HIFIR_GAIN_RIGHT 0x1a
44#define PM860X_DAC_OFFSET 0x1b
45#define PM860X_OFFSET_LEFT_1 0x1c
46#define PM860X_OFFSET_LEFT_2 0x1d
47#define PM860X_OFFSET_RIGHT_1 0x1e
48#define PM860X_OFFSET_RIGHT_2 0x1f
49#define PM860X_ADC_ANA_1 0x20
50#define PM860X_ADC_ANA_2 0x21
51#define PM860X_ADC_ANA_3 0x22
52#define PM860X_ADC_ANA_4 0x23
53#define PM860X_ANA_TO_ANA 0x24
54#define PM860X_HS1_CTRL 0x25
55#define PM860X_HS2_CTRL 0x26
56#define PM860X_LO1_CTRL 0x27
57#define PM860X_LO2_CTRL 0x28
58#define PM860X_EAR_CTRL_1 0x29
59#define PM860X_EAR_CTRL_2 0x2a
60#define PM860X_AUDIO_SUPPLIES_1 0x2b
61#define PM860X_AUDIO_SUPPLIES_2 0x2c
62#define PM860X_ADC_EN_1 0x2d
63#define PM860X_ADC_EN_2 0x2e
64#define PM860X_DAC_EN_1 0x2f
65#define PM860X_DAC_EN_2 0x31
66#define PM860X_AUDIO_CAL_1 0x32
67#define PM860X_AUDIO_CAL_2 0x33
68#define PM860X_AUDIO_CAL_3 0x34
69#define PM860X_AUDIO_CAL_4 0x35
70#define PM860X_AUDIO_CAL_5 0x36
71#define PM860X_ANA_INPUT_SEL_1 0x37
72#define PM860X_ANA_INPUT_SEL_2 0x38
73
74#define PM860X_PCM_IFACE_4 0x39
75#define PM860X_I2S_IFACE_5 0x3a
76
77#define PM860X_SHORTS 0x3b
78#define PM860X_PLL_ADJ_1 0x3c
79#define PM860X_PLL_ADJ_2 0x3d
80
81/* bits definition */
82#define PM860X_CLK_DIR_IN 0
83#define PM860X_CLK_DIR_OUT 1
84
85#define PM860X_DET_HEADSET (1 << 0)
86#define PM860X_DET_MIC (1 << 1)
87#define PM860X_DET_HOOK (1 << 2)
88#define PM860X_SHORT_HEADSET (1 << 3)
89#define PM860X_SHORT_LINEOUT (1 << 4)
90#define PM860X_DET_MASK 0x1F
91
92extern int pm860x_hs_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
93 int, int, int, int);
94extern int pm860x_mic_jack_detect(struct snd_soc_codec *, struct snd_soc_jack *,
95 int);
96
97#endif /* __88PM860X_H */
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index bfdd92b78fb6..a3cfc184ee50 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,6 +10,7 @@ config SND_SOC_I2C_AND_SPI
10 10
11config SND_SOC_ALL_CODECS 11config SND_SOC_ALL_CODECS
12 tristate "Build all ASoC CODEC drivers" 12 tristate "Build all ASoC CODEC drivers"
13 select SND_SOC_88PM860X if MFD_88PM860X
13 select SND_SOC_L3 14 select SND_SOC_L3
14 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS 15 select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
15 select SND_SOC_AD1836 if SPI_MASTER 16 select SND_SOC_AD1836 if SPI_MASTER
@@ -85,6 +86,9 @@ config SND_SOC_ALL_CODECS
85 86
86 If unsure select "N". 87 If unsure select "N".
87 88
89config SND_SOC_88PM860X
90 tristate
91
88config SND_SOC_WM_HUBS 92config SND_SOC_WM_HUBS
89 tristate 93 tristate
90 default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y 94 default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 9c3c39fd99ad..b9c43582c5bd 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,3 +1,4 @@
1snd-soc-88pm860x-objs := 88pm860x-codec.o
1snd-soc-ac97-objs := ac97.o 2snd-soc-ac97-objs := ac97.o
2snd-soc-ad1836-objs := ad1836.o 3snd-soc-ad1836-objs := ad1836.o
3snd-soc-ad193x-objs := ad193x.o 4snd-soc-ad193x-objs := ad193x.o
@@ -67,6 +68,7 @@ snd-soc-tpa6130a2-objs := tpa6130a2.o
67snd-soc-wm2000-objs := wm2000.o 68snd-soc-wm2000-objs := wm2000.o
68snd-soc-wm9090-objs := wm9090.o 69snd-soc-wm9090-objs := wm9090.o
69 70
71obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o
70obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o 72obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o
71obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o 73obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o
72obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o 74obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o
generated by cgit v1.2.2 (git 2.25.0) at 2024-12-12 04:38:28 -0500