| Linus Torvalds | 1da177e | 2005-04-16 15:20:36 -0700 | [diff] [blame] | 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 | |