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authorLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 18:20:36 -0400
committerLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 18:20:36 -0400
commit1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch)
tree0bba044c4ce775e45a88a51686b5d9f90697ea9d /sound/pci/emu10k1/p16v.h
Linux-2.6.12-rc2v2.6.12-rc2
Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
Diffstat (limited to 'sound/pci/emu10k1/p16v.h')
-rw-r--r--sound/pci/emu10k1/p16v.h299
1 files changed, 299 insertions, 0 deletions
diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h
new file mode 100644
index 000000000000..153214940336
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+++ b/sound/pci/emu10k1/p16v.h
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1/*
2 * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk>
3 * Driver p16v chips
4 * Version: 0.21
5 *
6 * FEATURES currently supported:
7 * Output fixed at S32_LE, 2 channel to hw:0,0
8 * Rates: 44.1, 48, 96, 192.
9 *
10 * Changelog:
11 * 0.8
12 * Use separate card based buffer for periods table.
13 * 0.9
14 * Use 2 channel output streams instead of 8 channel.
15 * (8 channel output streams might be good for ASIO type output)
16 * Corrected speaker output, so Front -> Front etc.
17 * 0.10
18 * Fixed missed interrupts.
19 * 0.11
20 * Add Sound card model number and names.
21 * Add Analog volume controls.
22 * 0.12
23 * Corrected playback interrupts. Now interrupt per period, instead of half period.
24 * 0.13
25 * Use single trigger for multichannel.
26 * 0.14
27 * Mic capture now works at fixed: S32_LE, 96000Hz, Stereo.
28 * 0.15
29 * Force buffer_size / period_size == INTEGER.
30 * 0.16
31 * Update p16v.c to work with changed alsa api.
32 * 0.17
33 * Update p16v.c to work with changed alsa api. Removed boot_devs.
34 * 0.18
35 * Merging with snd-emu10k1 driver.
36 * 0.19
37 * One stereo channel at 24bit now works.
38 * 0.20
39 * Added better register defines.
40 * 0.21
41 * Split from p16v.c
42 *
43 *
44 * BUGS:
45 * Some stability problems when unloading the snd-p16v kernel module.
46 * --
47 *
48 * TODO:
49 * SPDIF out.
50 * Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz.
51 * Currently capture fixed at 48000Hz.
52 *
53 * --
54 * GENERAL INFO:
55 * Model: SB0240
56 * P16V Chip: CA0151-DBS
57 * Audigy 2 Chip: CA0102-IAT
58 * AC97 Codec: STAC 9721
59 * ADC: Philips 1361T (Stereo 24bit)
60 * DAC: CS4382-K (8-channel, 24bit, 192Khz)
61 *
62 * This code was initally based on code from ALSA's emu10k1x.c which is:
63 * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
64 *
65 * This program is free software; you can redistribute it and/or modify
66 * it under the terms of the GNU General Public License as published by
67 * the Free Software Foundation; either version 2 of the License, or
68 * (at your option) any later version.
69 *
70 * This program is distributed in the hope that it will be useful,
71 * but WITHOUT ANY WARRANTY; without even the implied warranty of
72 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
73 * GNU General Public License for more details.
74 *
75 * You should have received a copy of the GNU General Public License
76 * along with this program; if not, write to the Free Software
77 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
78 *
79 */
80
81/********************************************************************************************************/
82/* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */
83/********************************************************************************************************/
84
85/* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE.
86 * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters.
87 */
88
89/* Initally all registers from 0x00 to 0x3f have zero contents. */
90#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
91 /* One list entry: 4 bytes for DMA address,
92 * 4 bytes for period_size << 16.
93 * One list entry is 8 bytes long.
94 * One list entry for each period in the buffer.
95 */
96#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
97#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
98#define PLAYBACK_UNKNOWN3 0x03 /* Not used */
99#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
100#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
101#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
102#define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */
103#define PLAYBACK_FIFO_POINTER 0x08 /* Playback FIFO pointer and number of valid sound samples in cache */
104#define PLAYBACK_UNKNOWN9 0x09 /* Not used */
105#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
106#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
107#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
108#define CAPTURE_FIFO_POINTER 0x13 /* Capture FIFO pointer and number of valid sound samples in cache */
109#define CAPTURE_P16V_VOLUME1 0x14 /* Low: Capture volume 0xXXXX3030 */
110#define CAPTURE_P16V_VOLUME2 0x15 /* High:Has no effect on capture volume */
111#define CAPTURE_P16V_SOURCE 0x16 /* P16V source select. Set to 0x0700E4E5 for AC97 CAPTURE */
112 /* [0:1] Capture input 0 channel select. 0 = Capture output 0.
113 * 1 = Capture output 1.
114 * 2 = Capture output 2.
115 * 3 = Capture output 3.
116 * [3:2] Capture input 1 channel select. 0 = Capture output 0.
117 * 1 = Capture output 1.
118 * 2 = Capture output 2.
119 * 3 = Capture output 3.
120 * [5:4] Capture input 2 channel select. 0 = Capture output 0.
121 * 1 = Capture output 1.
122 * 2 = Capture output 2.
123 * 3 = Capture output 3.
124 * [7:6] Capture input 3 channel select. 0 = Capture output 0.
125 * 1 = Capture output 1.
126 * 2 = Capture output 2.
127 * 3 = Capture output 3.
128 * [9:8] Playback input 0 channel select. 0 = Play output 0.
129 * 1 = Play output 1.
130 * 2 = Play output 2.
131 * 3 = Play output 3.
132 * [11:10] Playback input 1 channel select. 0 = Play output 0.
133 * 1 = Play output 1.
134 * 2 = Play output 2.
135 * 3 = Play output 3.
136 * [13:12] Playback input 2 channel select. 0 = Play output 0.
137 * 1 = Play output 1.
138 * 2 = Play output 2.
139 * 3 = Play output 3.
140 * [15:14] Playback input 3 channel select. 0 = Play output 0.
141 * 1 = Play output 1.
142 * 2 = Play output 2.
143 * 3 = Play output 3.
144 * [19:16] Playback mixer output enable. 1 bit per channel.
145 * [23:20] Capture mixer output enable. 1 bit per channel.
146 * [26:24] FX engine channel capture 0 = 0x60-0x67.
147 * 1 = 0x68-0x6f.
148 * 2 = 0x70-0x77.
149 * 3 = 0x78-0x7f.
150 * 4 = 0x80-0x87.
151 * 5 = 0x88-0x8f.
152 * 6 = 0x90-0x97.
153 * 7 = 0x98-0x9f.
154 * [31:27] Not used.
155 */
156
157 /* 0x1 = capture on.
158 * 0x100 = capture off.
159 * 0x200 = capture off.
160 * 0x1000 = capture off.
161 */
162#define CAPTURE_RATE_STATUS 0x17 /* Capture sample rate. Read only */
163 /* [15:0] Not used.
164 * [18:16] Channel 0 Detected sample rate. 0 - 44.1khz
165 * 1 - 48 khz
166 * 2 - 96 khz
167 * 3 - 192 khz
168 * 7 - undefined rate.
169 * [19] Channel 0. 1 - Valid, 0 - Not Valid.
170 * [22:20] Channel 1 Detected sample rate.
171 * [23] Channel 1. 1 - Valid, 0 - Not Valid.
172 * [26:24] Channel 2 Detected sample rate.
173 * [27] Channel 2. 1 - Valid, 0 - Not Valid.
174 * [30:28] Channel 3 Detected sample rate.
175 * [31] Channel 3. 1 - Valid, 0 - Not Valid.
176 */
177/* 0x18 - 0x1f unused */
178#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played. Read only */
179/* 0x21 - 0x3f unused */
180#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
181 /* Playback (0x1<<channel_id) Don't touch high 16bits. */
182 /* Capture (0x100<<channel_id). not tested */
183 /* Start Playback [3:0] (one bit per channel)
184 * Start Capture [11:8] (one bit per channel)
185 * Record source select for channel 0 [18:16]
186 * Record source select for channel 1 [22:20]
187 * Record source select for channel 2 [26:24]
188 * Record source select for channel 3 [30:28]
189 * 0 - SPDIF channel.
190 * 1 - I2S channel.
191 * 2 - SRC48 channel.
192 * 3 - SRCMulti_SPDIF channel.
193 * 4 - SRCMulti_I2S channel.
194 * 5 - SPDIF channel.
195 * 6 - fxengine capture.
196 * 7 - AC97 capture.
197 */
198 /* Default 41110000.
199 * Writing 0xffffffff hangs the PC.
200 * Writing 0xffff0000 -> 77770000 so it must be some sort of route.
201 * bit 0x1 starts DMA playback on channel_id 0
202 */
203/* 0x41,42 take values from 0 - 0xffffffff, but have no effect on playback */
204/* 0x43,0x48 do not remember settings */
205/* 0x41-45 unused */
206#define WATERMARK 0x46 /* Test bit to indicate cache level usage */
207 /* Values it can have while playing on channel 0.
208 * 0000f000, 0000f004, 0000f008, 0000f00c.
209 * Readonly.
210 */
211/* 0x47-0x4f unused */
212/* 0x50-0x5f Capture cache data */
213#define SRCSel 0x60 /* SRCSel. Default 0x4. Bypass P16V 0x14 */
214 /* [0] 0 = 10K2 audio, 1 = SRC48 mixer output.
215 * [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output.
216 * [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output.
217 */
218 /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
219 /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
220 /* SRC48 and SRCMULTI sample rate select and output select. */
221 /* 0xffffffff -> 0xC0000015
222 * 0xXXXXXXX4 = Enable Front Left/Right
223 * Enable PCMs
224 */
225
226/* 0x61 -> 0x6c are Volume controls */
227#define PLAYBACK_VOLUME_MIXER1 0x61 /* SRC48 Low to mixer input volume control. */
228#define PLAYBACK_VOLUME_MIXER2 0x62 /* SRC48 High to mixer input volume control. */
229#define PLAYBACK_VOLUME_MIXER3 0x63 /* SRCMULTI SPDIF Low to mixer input volume control. */
230#define PLAYBACK_VOLUME_MIXER4 0x64 /* SRCMULTI SPDIF High to mixer input volume control. */
231#define PLAYBACK_VOLUME_MIXER5 0x65 /* SRCMULTI I2S Low to mixer input volume control. */
232#define PLAYBACK_VOLUME_MIXER6 0x66 /* SRCMULTI I2S High to mixer input volume control. */
233#define PLAYBACK_VOLUME_MIXER7 0x67 /* P16V Low to SRCMULTI SPDIF mixer input volume control. */
234#define PLAYBACK_VOLUME_MIXER8 0x68 /* P16V High to SRCMULTI SPDIF mixer input volume control. */
235#define PLAYBACK_VOLUME_MIXER9 0x69 /* P16V Low to SRCMULTI I2S mixer input volume control. */
236 /* 0xXXXX3030 = PCM0 Volume (Front).
237 * 0x3030XXXX = PCM1 Volume (Center)
238 */
239#define PLAYBACK_VOLUME_MIXER10 0x6a /* P16V High to SRCMULTI I2S mixer input volume control. */
240 /* 0x3030XXXX = PCM3 Volume (Rear). */
241#define PLAYBACK_VOLUME_MIXER11 0x6b /* E10K2 Low to SRC48 mixer input volume control. */
242#define PLAYBACK_VOLUME_MIXER12 0x6c /* E10K2 High to SRC48 mixer input volume control. */
243
244#define SRC48_ENABLE 0x6d /* SRC48 input audio enable */
245 /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */
246 /* [23:16] The corresponding P16V channel to SRC48 enabled if == 1.
247 * [31:24] The corresponding E10K2 channel to SRC48 enabled.
248 */
249#define SRCMULTI_ENABLE 0x6e /* SRCMulti input audio enable. Default 0xffffffff */
250 /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */
251 /* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1.
252 * [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled.
253 * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled.
254 * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled.
255 */
256 /* Bypass P16V 0xff00ff00
257 * Bitmap. 0 = Off, 1 = On.
258 * P16V playback outputs:
259 * 0xXXXXXXX1 = PCM0 Left. (Front)
260 * 0xXXXXXXX2 = PCM0 Right.
261 * 0xXXXXXXX4 = PCM1 Left. (Center/LFE)
262 * 0xXXXXXXX8 = PCM1 Right.
263 * 0xXXXXXX1X = PCM2 Left. (Unknown)
264 * 0xXXXXXX2X = PCM2 Right.
265 * 0xXXXXXX4X = PCM3 Left. (Rear)
266 * 0xXXXXXX8X = PCM3 Right.
267 */
268#define AUDIO_OUT_ENABLE 0x6f /* Default: 000100FF */
269 /* [3:0] Does something, but not documented. Probably capture enable.
270 * [7:4] Playback channels enable. not documented.
271 * [16] AC97 output enable if == 1
272 * [30] 0 = SRCMulti_I2S input from fxengine 0x68-0x6f.
273 * 1 = SRCMulti_I2S input from SRC48 output.
274 * [31] 0 = SRCMulti_SPDIF input from fxengine 0x60-0x67.
275 * 1 = SRCMulti_SPDIF input from SRC48 output.
276 */
277 /* 0xffffffff -> C00100FF */
278 /* 0 -> Not playback sound, irq still running */
279 /* 0xXXXXXX10 = PCM0 Left/Right On. (Front)
280 * 0xXXXXXX20 = PCM1 Left/Right On. (Center/LFE)
281 * 0xXXXXXX40 = PCM2 Left/Right On. (Unknown)
282 * 0xXXXXXX80 = PCM3 Left/Right On. (Rear)
283 */
284#define PLAYBACK_SPDIF_SELECT 0x70 /* Default: 12030F00 */
285 /* 0xffffffff -> 3FF30FFF */
286 /* 0x00000001 pauses stream/irq fail. */
287 /* All other bits do not effect playback */
288#define PLAYBACK_SPDIF_SRC_SELECT 0x71 /* Default: 0000E4E4 */
289 /* 0xffffffff -> F33FFFFF */
290 /* All bits do not effect playback */
291#define PLAYBACK_SPDIF_USER_DATA0 0x72 /* SPDIF out user data 0 */
292#define PLAYBACK_SPDIF_USER_DATA1 0x73 /* SPDIF out user data 1 */
293/* 0x74-0x75 unknown */
294#define CAPTURE_SPDIF_CONTROL 0x76 /* SPDIF in control setting */
295#define CAPTURE_SPDIF_STATUS 0x77 /* SPDIF in status */
296#define CAPURE_SPDIF_USER_DATA0 0x78 /* SPDIF in user data 0 */
297#define CAPURE_SPDIF_USER_DATA1 0x79 /* SPDIF in user data 1 */
298#define CAPURE_SPDIF_USER_DATA2 0x7a /* SPDIF in user data 2 */
299
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