1 /* 2 * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> 3 * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit 4 * Version: 0.0.20 5 * 6 * FEATURES currently supported: 7 * See ca0106_main.c for features. 8 * 9 * Changelog: 10 * Support interrupts per period. 11 * Removed noise from Center/LFE channel when in Analog mode. 12 * Rename and remove mixer controls. 13 * 0.0.6 14 * Use separate card based DMA buffer for periods table list. 15 * 0.0.7 16 * Change remove and rename ctrls into lists. 17 * 0.0.8 18 * Try to fix capture sources. 19 * 0.0.9 20 * Fix AC3 output. 21 * Enable S32_LE format support. 22 * 0.0.10 23 * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) 24 * 0.0.11 25 * Add Model name recognition. 26 * 0.0.12 27 * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. 28 * Remove redundent "voice" handling. 29 * 0.0.13 30 * Single trigger call for multi channels. 31 * 0.0.14 32 * Set limits based on what the sound card hardware can do. 33 * playback periods_min=2, periods_max=8 34 * capture hw constraints require period_size = n * 64 bytes. 35 * playback hw constraints require period_size = n * 64 bytes. 36 * 0.0.15 37 * Separated ca0106.c into separate functional .c files. 38 * 0.0.16 39 * Implement 192000 sample rate. 40 * 0.0.17 41 * Add support for SB0410 and SB0413. 42 * 0.0.18 43 * Modified Copyright message. 44 * 0.0.19 45 * Added I2C and SPI registers. Filled in interrupt enable. 46 * 0.0.20 47 * Added GPIO info for SB Live 24bit. 48 * 49 * 50 * This code was initally based on code from ALSA's emu10k1x.c which is: 51 * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> 52 * 53 * This program is free software; you can redistribute it and/or modify 54 * it under the terms of the GNU General Public License as published by 55 * the Free Software Foundation; either version 2 of the License, or 56 * (at your option) any later version. 57 * 58 * This program is distributed in the hope that it will be useful, 59 * but WITHOUT ANY WARRANTY; without even the implied warranty of 60 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the 61 * GNU General Public License for more details. 62 * 63 * You should have received a copy of the GNU General Public License 64 * along with this program; if not, write to the Free Software 65 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA 66 * 67 */ 68 69 /************************************************************************************************/ 70 /* PCI function 0 registers, address = <val> + PCIBASE0 */ 71 /************************************************************************************************/ 72 73 #define PTR 0x00 /* Indexed register set pointer register */ 74 /* NOTE: The CHANNELNUM and ADDRESS words can */ 75 /* be modified independently of each other. */ 76 /* CNL[1:0], ADDR[27:16] */ 77 78 #define DATA 0x04 /* Indexed register set data register */ 79 /* DATA[31:0] */ 80 81 #define IPR 0x08 /* Global interrupt pending register */ 82 /* Clear pending interrupts by writing a 1 to */ 83 /* the relevant bits and zero to the other bits */ 84 #define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ 85 #define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ 86 #define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ 87 #define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ 88 #define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ 89 #define IPR_SPI 0x00000800 /* SPI transaction completed */ 90 #define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ 91 #define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ 92 #define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */ 93 #define IPR_GPI 0x00000080 /* General Purpose input changed */ 94 #define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */ 95 #define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ 96 #define IPR_TIMER2 0x00000010 /* 192000Hz Timer */ 97 #define IPR_TIMER1 0x00000008 /* 44100Hz Timer */ 98 #define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ 99 #define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ 100 #define IPR_PCI 0x00000001 /* PCI Bus error */ 101 102 #define INTE 0x0c /* Interrupt enable register */ 103 104 #define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ 105 #define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ 106 #define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ 107 #define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ 108 #define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ 109 #define INTE_SPI 0x00000800 /* SPI transaction completed */ 110 #define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ 111 #define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ 112 #define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */ 113 #define INTE_GPI 0x00000080 /* General Purpose input changed */ 114 #define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */ 115 #define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ 116 #define INTE_TIMER2 0x00000010 /* 192000Hz Timer */ 117 #define INTE_TIMER1 0x00000008 /* 44100Hz Timer */ 118 #define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ 119 #define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ 120 #define INTE_PCI 0x00000001 /* PCI Bus error */ 121 122 #define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ 123 #define HCFG 0x14 /* Hardware config register */ 124 /* 0x1000 causes AC3 to fails. It adds a dither bit. */ 125 126 #define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */ 127 #define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */ 128 #define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */ 129 #define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */ 130 #define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */ 131 #define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */ 132 #define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */ 133 #define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */ 134 #define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */ 135 #define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/ 136 #define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/ 137 #define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */ 138 #define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */ 139 #define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */ 140 #define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */ 141 /* NOTE: This should generally never be used. */ 142 #define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */ 143 /* NOTE: This should generally never be used. */ 144 #define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */ 145 /* Should be set to 1 when the EMU10K1 is */ 146 /* completely initialized. */ 147 #define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ 148 /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */ 149 /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */ 150 /* SB Live 24bit: 151 * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in. 152 * bit 9 0 = Mute / 1 = Analog out. 153 * bit 10 0 = Line-in / 1 = Mic-in. 154 * bit 11 0 = ? / 1 = ? 155 * bit 12 0 = ? / 1 = ? 156 * bit 13 0 = ? / 1 = ? 157 * bit 14 0 = Mute / 1 = Analog out 158 * bit 15 0 = ? / 1 = ? 159 * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit. 160 */ 161 /* 8 general purpose programmable In/Out pins. 162 * GPI [8:0] Read only. Default 0. 163 * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF) 164 * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin. 165 */ 166 #define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ 167 168 #define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ 169 170 /********************************************************************************************************/ 171 /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ 172 /********************************************************************************************************/ 173 174 /* Initally all registers from 0x00 to 0x3f have zero contents. */ 175 #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ 176 /* One list entry: 4 bytes for DMA address, 177 * 4 bytes for period_size << 16. 178 * One list entry is 8 bytes long. 179 * One list entry for each period in the buffer. 180 */ 181 /* ADDR[31:0], Default: 0x0 */ 182 #define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ 183 /* SIZE[21:16], Default: 0x8 */ 184 #define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ 185 /* PTR[5:0], Default: 0x0 */ 186 #define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */ 187 #define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ 188 /* DMA[31:0], Default: 0x0 */ 189 #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ 190 /* SIZE[31:16], Default: 0x0 */ 191 #define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ 192 /* POINTER[15:0], Default: 0x0 */ 193 #define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */ 194 /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */ 195 #define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */ 196 /* Cache size valid [5:0] */ 197 #define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */ 198 #define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */ 199 /* DMA[31:0], Default: 0x0 */ 200 #define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */ 201 /* SIZE[31:16], Default: 0x0 */ 202 #define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */ 203 /* POINTER[15:0], Default: 0x0 */ 204 #define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */ 205 /* Cache size valid [5:0] */ 206 #define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */ 207 /* 0x21 - 0x3f unused */ 208 #define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */ 209 /* Playback (0x1<<channel_id) */ 210 /* Capture (0x100<<channel_id) */ 211 /* Playback sample rate 96000 = 0x20000 */ 212 /* Start Playback [3:0] (one bit per channel) 213 * Start Capture [11:8] (one bit per channel) 214 * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 215 * Playback mixer in enable [27:24] (one bit per channel) 216 * Playback mixer out enable [31:28] (one bit per channel) 217 */ 218 /* The Digital out jack is shared with the Center/LFE Analogue output. 219 * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 220 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground 221 * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. 222 * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. 223 * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. 224 */ 225 /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS 226 * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS 227 * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM. 228 * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output 229 */ 230 /* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel. 231 * A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs. 232 */ 233 #define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */ 234 #define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */ 235 #define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */ 236 #define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */ 237 /* When Channel set to 0: */ 238 #define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */ 239 #define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */ 240 #define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */ 241 #define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */ 242 #define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */ 243 #define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */ 244 #define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */ 245 #define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */ 246 #define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */ 247 #define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */ 248 #define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */ 249 #define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */ 250 #define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */ 251 #define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */ 252 #define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */ 253 #define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */ 254 #define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */ 255 #define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */ 256 #define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */ 257 #define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */ 258 #define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */ 259 #define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */ 260 #define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */ 261 262 /* When Channel set to 1: */ 263 #define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */ 264 #define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */ 265 #define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */ 266 #define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */ 267 #define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */ 268 #define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */ 269 #define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */ 270 #define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ 271 #define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ 272 #define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */ 273 #define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */ 274 #define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */ 275 #define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */ 276 #define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */ 277 #define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */ 278 #define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */ 279 #define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */ 280 #define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */ 281 #define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */ 282 #define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */ 283 #define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */ 284 #define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */ 285 #define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */ 286 #define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */ 287 #define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */ 288 #define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */ 289 #define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */ 290 #define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */ 291 #define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */ 292 293 #define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */ 294 /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE. 295 * But as the jack is shared, use 0xf00. 296 * The Windows2000 driver uses 0x0000000f for both digital and analog. 297 * 0xf00 introduces interesting noises onto the Center/LFE. 298 * If you turn the volume up, you hear computer noise, 299 * e.g. mouse moving, changing between app windows etc. 300 * So, I am going to set this to 0x0000000f all the time now, 301 * same as the windows driver does. 302 * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog. 303 */ 304 /* When Channel = 0: 305 * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit) 306 * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate) 307 * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass) 308 */ 309 /* When Channel = 1: 310 * SPDIF 0 User data [7:0] 311 * SPDIF 1 User data [15:8] 312 * SPDIF 0 User data [23:16] 313 * SPDIF 0 User data [31:24] 314 * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts. 315 */ 316 #define WATERMARK 0x46 /* Test bit to indicate cache usage level */ 317 #define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS. 318 * When Channel = 0: Bits the same as SPCS channel 0. 319 * When Channel = 1: Bits the same as SPCS channel 1. 320 * When Channel = 2: 321 * SPDIF Input User data [16:0] 322 * SPDIF Input Frame count [21:16] 323 */ 324 #define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */ 325 #define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */ 326 #define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */ 327 #define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */ 328 #define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */ 329 #define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */ 330 #define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */ 331 /* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3 332 * Record source select for channel 0 [18:16] 333 * Record source select for channel 1 [22:20] 334 * Record source select for channel 2 [26:24] 335 * Record source select for channel 3 [30:28] 336 * 0 - SPDIF mixer output. 337 * 1 - i2s mixer output. 338 * 2 - SPDIF input. 339 * 3 - i2s input. 340 * 4 - AC97 capture. 341 * 5 - SRC output. 342 */ 343 #define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */ 344 #define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */ 345 346 #define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */ 347 #define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */ 348 #define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */ 349 #define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */ 350 #define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */ 351 /* Channel_id's handle stereo channels. Channel X is a single mono channel */ 352 /* Host is input from the PCI bus. */ 353 /* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. 354 * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. 355 * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. 356 * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. 357 * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. 358 * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. 359 * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. 360 * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. 361 */ 362 363 #define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */ 364 /* SRC is input from the capture inputs. */ 365 /* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. 366 * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. 367 * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. 368 * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. 369 * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. 370 * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. 371 * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. 372 * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. 373 */ 374 375 #define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */ 376 /* SPDIF Mixer input control: 377 * Invert SRC to SPDIF Mixer [7-0] (One bit per channel) 378 * Invert Host to SPDIF Mixer [15:8] (One bit per channel) 379 * SRC to SPDIF Mixer disable [23:16] (One bit per channel) 380 * Host to SPDIF Mixer disable [31:24] (One bit per channel) 381 */ 382 #define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */ 383 /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */ 384 /* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */ 385 /* One register for each of the 4 stereo streams. */ 386 /* SRC Right volume [7:0] 387 * SRC Left volume [15:8] 388 * Host Right volume [23:16] 389 * Host Left volume [31:24] 390 */ 391 #define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */ 392 /* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ 393 #define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */ 394 /* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ 395 #define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */ 396 /* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ 397 #define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */ 398 /* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */ 399 #define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */ 400 #define UART_A_DATA 0x6c /* Uart, used in setting sample rates, bits per sample etc. */ 401 #define UART_A_CMD 0x6d /* Uart, used in setting sample rates, bits per sample etc. */ 402 #define UART_B_DATA 0x6e /* Uart, Unknown. */ 403 #define UART_B_CMD 0x6f /* Uart, Unknown. */ 404 #define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */ 405 /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0 406 * Rate Locked [20] 407 * SPDIF Locked [21] For SPDIF channel only. 408 * Valid Audio [22] For SPDIF channel only. 409 */ 410 #define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */ 411 /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */ 412 /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */ 413 /* Sample rate output control register Channel=0 414 * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 415 * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) 416 * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source. 417 * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) 418 * Record mixer output enable [12:10] 419 * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 420 * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 421 * I2S output source select [18] (0=Audio from host, 1=Audio from SRC) 422 * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0) 423 * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.) 424 * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.) 425 * I2S input mode [23] (0=Slave, 1=Master) 426 * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 427 * SPDIF output source select [26] (0=host, 1=SRC) 428 * Not used [27] 429 * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) 430 * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) 431 */ 432 /* Sample rate output control register Channel=1 433 * I2S Input 0 volume Right [7:0] 434 * I2S Input 0 volume Left [15:8] 435 * I2S Input 1 volume Right [23:16] 436 * I2S Input 1 volume Left [31:24] 437 */ 438 /* Sample rate output control register Channel=2 439 * SPDIF Input volume Right [23:16] 440 * SPDIF Input volume Left [31:24] 441 */ 442 /* Sample rate output control register Channel=3 443 * No used 444 */ 445 #define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */ 446 #define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */ 447 #define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */ 448 #define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */ 449 /* Audio output control 450 * AC97 output enable [5:0] 451 * I2S output enable [19:16] 452 * SPDIF output enable [27:24] 453 */ 454 #define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */ 455 #define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */ 456 #define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */ 457 /* Sets which Interrupts are enabled. */ 458 /* 0x00000001 = Half period. Playback. 459 * 0x00000010 = Full period. Playback. 460 * 0x00000100 = Half buffer. Playback. 461 * 0x00001000 = Full buffer. Playback. 462 * 0x00010000 = Half buffer. Capture. 463 * 0x00100000 = Full buffer. Capture. 464 * Capture can only do 2 periods. 465 * 0x01000000 = End audio. Playback. 466 * 0x40000000 = Half buffer Playback,Caputre xrun. 467 * 0x80000000 = Full buffer Playback,Caputre xrun. 468 */ 469 #define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */ 470 /* Shows which interrupts are active at the moment. */ 471 /* Same bit layout as EXTENDED_INT_MASK */ 472 #define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */ 473 #define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */ 474 #define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */ 475 /* Causes interrupts based on timer intervals. */ 476 #define SPI 0x7a /* SPI: Serial Interface Register */ 477 #define I2C_A 0x7b /* I2C Address. 32 bit */ 478 #define I2C_0 0x7c /* I2C Data Port 0. 32 bit */ 479 #define I2C_1 0x7d /* I2C Data Port 1. 32 bit */ 480 481 482 #define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ 483 #define PCM_FRONT_CHANNEL 0 484 #define PCM_REAR_CHANNEL 1 485 #define PCM_CENTER_LFE_CHANNEL 2 486 #define PCM_UNKNOWN_CHANNEL 3 487 #define CONTROL_FRONT_CHANNEL 0 488 #define CONTROL_REAR_CHANNEL 3 489 #define CONTROL_CENTER_LFE_CHANNEL 1 490 #define CONTROL_UNKNOWN_CHANNEL 2 491 492 typedef struct snd_ca0106_channel ca0106_channel_t; 493 typedef struct snd_ca0106 ca0106_t; 494 typedef struct snd_ca0106_pcm ca0106_pcm_t; 495 496 struct snd_ca0106_channel { 497 ca0106_t *emu; 498 int number; 499 int use; 500 void (*interrupt)(ca0106_t *emu, ca0106_channel_t *channel); 501 ca0106_pcm_t *epcm; 502 }; 503 504 struct snd_ca0106_pcm { 505 ca0106_t *emu; 506 snd_pcm_substream_t *substream; 507 int channel_id; 508 unsigned short running; 509 }; 510 511 // definition of the chip-specific record 512 struct snd_ca0106 { 513 snd_card_t *card; 514 struct pci_dev *pci; 515 516 unsigned long port; 517 struct resource *res_port; 518 int irq; 519 520 unsigned int revision; /* chip revision */ 521 unsigned int serial; /* serial number */ 522 unsigned short model; /* subsystem id */ 523 524 spinlock_t emu_lock; 525 526 ac97_t *ac97; 527 snd_pcm_t *pcm; 528 529 ca0106_channel_t playback_channels[4]; 530 ca0106_channel_t capture_channels[4]; 531 u32 spdif_bits[4]; /* s/pdif out setup */ 532 int spdif_enable; 533 int capture_source; 534 535 struct snd_dma_buffer buffer; 536 }; 537 538 int __devinit snd_ca0106_mixer(ca0106_t *emu); 539 int __devinit snd_ca0106_proc_init(ca0106_t * emu); 540 541 unsigned int snd_ca0106_ptr_read(ca0106_t * emu, 542 unsigned int reg, 543 unsigned int chn); 544 545 void snd_ca0106_ptr_write(ca0106_t *emu, 546 unsigned int reg, 547 unsigned int chn, 548 unsigned int data); 549 550