summaryrefslogtreecommitdiffstats
path: root/sound/pci/emu10k1/p16v.h
blob: 00f4817533b12bf2f1c1a259c5048e7655326620 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
/*
 *  Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
 *  Driver p16v chips
 *  Version: 0.21
 *
 *  FEATURES currently supported:
 *    Output fixed at S32_LE, 2 channel to hw:0,0
 *    Rates: 44.1, 48, 96, 192.
 *
 *  Changelog:
 *  0.8
 *    Use separate card based buffer for periods table.
 *  0.9
 *    Use 2 channel output streams instead of 8 channel.
 *       (8 channel output streams might be good for ASIO type output)
 *    Corrected speaker output, so Front -> Front etc.
 *  0.10
 *    Fixed missed interrupts.
 *  0.11
 *    Add Sound card model number and names.
 *    Add Analog volume controls.
 *  0.12
 *    Corrected playback interrupts. Now interrupt per period, instead of half period.
 *  0.13
 *    Use single trigger for multichannel.
 *  0.14
 *    Mic capture now works at fixed: S32_LE, 96000Hz, Stereo.
 *  0.15
 *    Force buffer_size / period_size == INTEGER.
 *  0.16
 *    Update p16v.c to work with changed alsa api.
 *  0.17
 *    Update p16v.c to work with changed alsa api. Removed boot_devs.
 *  0.18
 *    Merging with snd-emu10k1 driver.
 *  0.19
 *    One stereo channel at 24bit now works.
 *  0.20
 *    Added better register defines.
 *  0.21
 *    Split from p16v.c
 *
 *
 *  BUGS:
 *    Some stability problems when unloading the snd-p16v kernel module.
 *    --
 *
 *  TODO:
 *    SPDIF out.
 *    Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz.
 *    Currently capture fixed at 48000Hz.
 *
 *    --
 *  GENERAL INFO:
 *    Model: SB0240
 *    P16V Chip: CA0151-DBS
 *    Audigy 2 Chip: CA0102-IAT
 *    AC97 Codec: STAC 9721
 *    ADC: Philips 1361T (Stereo 24bit)
 *    DAC: CS4382-K (8-channel, 24bit, 192Khz)
 *
 *  This code was initally based on code from ALSA's emu10k1x.c which is:
 *  Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
 *
 *   This program is free software; you can redistribute it and/or modify
 *   it under the terms of the GNU General Public License as published by
 *   the Free Software Foundation; either version 2 of the License, or
 *   (at your option) any later version.
 *
 *   This program is distributed in the hope that it will be useful,
 *   but WITHOUT ANY WARRANTY; without even the implied warranty of
 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *   GNU General Public License for more details.
 *
 *   You should have received a copy of the GNU General Public License
 *   along with this program; if not, write to the Free Software
 *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
 *
 */

/********************************************************************************************************/
/* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers                     */
/********************************************************************************************************/
                                                                                                                           
/* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE.
 * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
 */

/* Initally all registers from 0x00 to 0x3f have zero contents. */
#define PLAYBACK_LIST_ADDR	0x00		/* Base DMA address of a list of pointers to each period/size */
						/* One list entry: 4 bytes for DMA address, 
						 * 4 bytes for period_size << 16.
						 * One list entry is 8 bytes long.
						 * One list entry for each period in the buffer.
						 */
#define PLAYBACK_LIST_SIZE	0x01		/* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000  */
#define PLAYBACK_LIST_PTR	0x02		/* Pointer to the current period being played */
#define PLAYBACK_UNKNOWN3	0x03		/* Not used */
#define PLAYBACK_DMA_ADDR	0x04		/* Playback DMA address */
#define PLAYBACK_PERIOD_SIZE	0x05		/* Playback period size. win2000 uses 0x04000000 */
#define PLAYBACK_POINTER	0x06		/* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
#define PLAYBACK_FIFO_END_ADDRESS	0x07		/* Playback FIFO end address */
#define PLAYBACK_FIFO_POINTER	0x08		/* Playback FIFO pointer and number of valid sound samples in cache */
#define PLAYBACK_UNKNOWN9	0x09		/* Not used */
#define CAPTURE_DMA_ADDR	0x10		/* Capture DMA address */
#define CAPTURE_BUFFER_SIZE	0x11		/* Capture buffer size */
#define CAPTURE_POINTER		0x12		/* Capture buffer pointer. Sample currently in ADC */
#define CAPTURE_FIFO_POINTER	0x13		/* Capture FIFO pointer and number of valid sound samples in cache */
#define CAPTURE_P16V_VOLUME1	0x14		/* Low: Capture volume 0xXXXX3030 */
#define CAPTURE_P16V_VOLUME2	0x15		/* High:Has no effect on capture volume */
#define CAPTURE_P16V_SOURCE     0x16            /* P16V source select. Set to 0x0700E4E5 for AC97 CAPTURE */
						/* [0:1] Capture input 0 channel select. 0 = Capture output 0.
						 *                                       1 = Capture output 1.
						 *                                       2 = Capture output 2.
						 *                                       3 = Capture output 3.
						 * [3:2] Capture input 1 channel select. 0 = Capture output 0.
						 *                                       1 = Capture output 1.
						 *                                       2 = Capture output 2.
						 *                                       3 = Capture output 3.
						 * [5:4] Capture input 2 channel select. 0 = Capture output 0.
						 *                                       1 = Capture output 1.
						 *                                       2 = Capture output 2.
						 *                                       3 = Capture output 3.
						 * [7:6] Capture input 3 channel select. 0 = Capture output 0.
						 *                                       1 = Capture output 1.
						 *                                       2 = Capture output 2.
						 *                                       3 = Capture output 3.
						 * [9:8] Playback input 0 channel select. 0 = Play output 0.
						 *                                        1 = Play output 1.
						 *                                        2 = Play output 2.
						 *                                        3 = Play output 3.
						 * [11:10] Playback input 1 channel select. 0 = Play output 0.
						 *                                          1 = Play output 1.
						 *                                          2 = Play output 2.
						 *                                          3 = Play output 3.
						 * [13:12] Playback input 2 channel select. 0 = Play output 0.
						 *                                          1 = Play output 1.
						 *                                          2 = Play output 2.
						 *                                          3 = Play output 3.
						 * [15:14] Playback input 3 channel select. 0 = Play output 0.
						 *                                          1 = Play output 1.
						 *                                          2 = Play output 2.
						 *                                          3 = Play output 3.
						 * [19:16] Playback mixer output enable. 1 bit per channel.
						 * [23:20] Capture mixer output enable. 1 bit per channel.
						 * [26:24] FX engine channel capture 0 = 0x60-0x67.
						 *                                   1 = 0x68-0x6f.
						 *                                   2 = 0x70-0x77.
						 *                                   3 = 0x78-0x7f.
						 *                                   4 = 0x80-0x87.
						 *                                   5 = 0x88-0x8f.
						 *                                   6 = 0x90-0x97.
						 *                                   7 = 0x98-0x9f.
						 * [31:27] Not used.
						 */

						/* 0x1 = capture on.
						 * 0x100 = capture off.
						 * 0x200 = capture off.
						 * 0x1000 = capture off.
						 */
#define CAPTURE_RATE_STATUS		0x17		/* Capture sample rate. Read only */
						/* [15:0] Not used.
						 * [18:16] Channel 0 Detected sample rate. 0 - 44.1khz
						 *                               1 - 48 khz
						 *                               2 - 96 khz
						 *                               3 - 192 khz
						 *                               7 - undefined rate.
						 * [19] Channel 0. 1 - Valid, 0 - Not Valid.
						 * [22:20] Channel 1 Detected sample rate. 
						 * [23] Channel 1. 1 - Valid, 0 - Not Valid.
						 * [26:24] Channel 2 Detected sample rate. 
						 * [27] Channel 2. 1 - Valid, 0 - Not Valid.
						 * [30:28] Channel 3 Detected sample rate. 
						 * [31] Channel 3. 1 - Valid, 0 - Not Valid.
						 */
/* 0x18 - 0x1f unused */
#define PLAYBACK_LAST_SAMPLE    0x20		/* The sample currently being played. Read only */
/* 0x21 - 0x3f unused */
#define BASIC_INTERRUPT         0x40		/* Used by both playback and capture interrupt handler */
						/* Playback (0x1<<channel_id) Don't touch high 16bits. */
						/* Capture  (0x100<<channel_id). not tested */
						/* Start Playback [3:0] (one bit per channel)
						 * Start Capture [11:8] (one bit per channel)
						 * Record source select for channel 0 [18:16]
						 * Record source select for channel 1 [22:20]
						 * Record source select for channel 2 [26:24]
						 * Record source select for channel 3 [30:28]
						 * 0 - SPDIF channel.
						 * 1 - I2S channel.
						 * 2 - SRC48 channel.
						 * 3 - SRCMulti_SPDIF channel.
						 * 4 - SRCMulti_I2S channel.
						 * 5 - SPDIF channel.
						 * 6 - fxengine capture.
						 * 7 - AC97 capture.
						 */
						/* Default 41110000.
						 * Writing 0xffffffff hangs the PC.
						 * Writing 0xffff0000 -> 77770000 so it must be some sort of route.
						 * bit 0x1 starts DMA playback on channel_id 0
						 */
/* 0x41,42 take values from 0 - 0xffffffff, but have no effect on playback */
/* 0x43,0x48 do not remember settings */
/* 0x41-45 unused */
#define WATERMARK            0x46		/* Test bit to indicate cache level usage */
						/* Values it can have while playing on channel 0. 
						 * 0000f000, 0000f004, 0000f008, 0000f00c.
						 * Readonly.
						 */
/* 0x47-0x4f unused */
/* 0x50-0x5f Capture cache data */
#define SRCSel			0x60            /* SRCSel. Default 0x4. Bypass P16V 0x14 */
						/* [0] 0 = 10K2 audio, 1 = SRC48 mixer output.
						 * [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output.
						 * [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output.
						 */
						/* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
						/* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
						/* SRC48 and SRCMULTI sample rate select and output select. */
						/* 0xffffffff -> 0xC0000015
						 * 0xXXXXXXX4 = Enable Front Left/Right
						 * Enable PCMs
						 */

/* 0x61 -> 0x6c are Volume controls */
#define PLAYBACK_VOLUME_MIXER1  0x61		/* SRC48 Low to mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER2  0x62		/* SRC48 High to mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER3  0x63		/* SRCMULTI SPDIF Low to mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER4  0x64		/* SRCMULTI SPDIF High to mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER5  0x65		/* SRCMULTI I2S Low to mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER6  0x66		/* SRCMULTI I2S High to mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER7  0x67		/* P16V Low to SRCMULTI SPDIF mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER8  0x68		/* P16V High to SRCMULTI SPDIF mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER9  0x69		/* P16V Low to SRCMULTI I2S mixer input volume control. */
						/* 0xXXXX3030 = PCM0 Volume (Front).
						 * 0x3030XXXX = PCM1 Volume (Center)
						 */
#define PLAYBACK_VOLUME_MIXER10  0x6a		/* P16V High to SRCMULTI I2S mixer input volume control. */
						/* 0x3030XXXX = PCM3 Volume (Rear). */
#define PLAYBACK_VOLUME_MIXER11  0x6b		/* E10K2 Low to SRC48 mixer input volume control. */
#define PLAYBACK_VOLUME_MIXER12 0x6c		/* E10K2 High to SRC48 mixer input volume control. */

#define SRC48_ENABLE            0x6d            /* SRC48 input audio enable */
						/* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
						/* [23:16] The corresponding P16V channel to SRC48 enabled if == 1.
						 * [31:24] The corresponding E10K2 channel to SRC48 enabled.
						 */
#define SRCMULTI_ENABLE         0x6e            /* SRCMulti input audio enable. Default 0xffffffff */
						/* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
						/* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1.
						 * [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled.
						 * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled.
						 * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled.
						 */
						/* Bypass P16V 0xff00ff00 
						 * Bitmap. 0 = Off, 1 = On.
						 * P16V playback outputs:
						 * 0xXXXXXXX1 = PCM0 Left. (Front)
						 * 0xXXXXXXX2 = PCM0 Right.
						 * 0xXXXXXXX4 = PCM1 Left. (Center/LFE)
						 * 0xXXXXXXX8 = PCM1 Right.
						 * 0xXXXXXX1X = PCM2 Left. (Unknown)
						 * 0xXXXXXX2X = PCM2 Right.
						 * 0xXXXXXX4X = PCM3 Left. (Rear)
						 * 0xXXXXXX8X = PCM3 Right.
						 */
#define AUDIO_OUT_ENABLE        0x6f            /* Default: 000100FF */
						/* [3:0] Does something, but not documented. Probably capture enable.
						 * [7:4] Playback channels enable. not documented.
						 * [16] AC97 output enable if == 1
						 * [30] 0 = SRCMulti_I2S input from fxengine 0x68-0x6f.
						 *      1 = SRCMulti_I2S input from SRC48 output.
						 * [31] 0 = SRCMulti_SPDIF input from fxengine 0x60-0x67.
						 *      1 = SRCMulti_SPDIF input from SRC48 output.
						 */
						/* 0xffffffff -> C00100FF */
						/* 0 -> Not playback sound, irq still running */
						/* 0xXXXXXX10 = PCM0 Left/Right On. (Front)
						 * 0xXXXXXX20 = PCM1 Left/Right On. (Center/LFE)
						 * 0xXXXXXX40 = PCM2 Left/Right On. (Unknown)
						 * 0xXXXXXX80 = PCM3 Left/Right On. (Rear)
						 */
#define PLAYBACK_SPDIF_SELECT     0x70          /* Default: 12030F00 */
						/* 0xffffffff -> 3FF30FFF */
						/* 0x00000001 pauses stream/irq fail. */
						/* All other bits do not effect playback */
#define PLAYBACK_SPDIF_SRC_SELECT 0x71          /* Default: 0000E4E4 */
						/* 0xffffffff -> F33FFFFF */
						/* All bits do not effect playback */
#define PLAYBACK_SPDIF_USER_DATA0 0x72		/* SPDIF out user data 0 */
#define PLAYBACK_SPDIF_USER_DATA1 0x73		/* SPDIF out user data 1 */
/* 0x74-0x75 unknown */
#define CAPTURE_SPDIF_CONTROL	0x76		/* SPDIF in control setting */
#define CAPTURE_SPDIF_STATUS	0x77		/* SPDIF in status */
#define CAPURE_SPDIF_USER_DATA0 0x78		/* SPDIF in user data 0 */
#define CAPURE_SPDIF_USER_DATA1 0x79		/* SPDIF in user data 1 */
#define CAPURE_SPDIF_USER_DATA2 0x7a		/* SPDIF in user data 2 */

OpenPOWER on IntegriCloud