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: 0.0.22 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 * 0.0.21 49 * Implement support for Line-in capture on SB Live 24bit. 50 * 0.0.22 51 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) 52 * 53 * 54 * This code was initally based on code from ALSA's emu10k1x.c which is: 55 * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> 56 * 57 * This program is free software; you can redistribute it and/or modify 58 * it under the terms of the GNU General Public License as published by 59 * the Free Software Foundation; either version 2 of the License, or 60 * (at your option) any later version. 61 * 62 * This program is distributed in the hope that it will be useful, 63 * but WITHOUT ANY WARRANTY; without even the implied warranty of 64 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the 65 * GNU General Public License for more details. 66 * 67 * You should have received a copy of the GNU General Public License 68 * along with this program; if not, write to the Free Software 69 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA 70 * 71 */ 72 73/************************************************************************************************/ 74/* PCI function 0 registers, address = <val> + PCIBASE0 */ 75/************************************************************************************************/ 76 77#define PTR 0x00 /* Indexed register set pointer register */ 78 /* NOTE: The CHANNELNUM and ADDRESS words can */ 79 /* be modified independently of each other. */ 80 /* CNL[1:0], ADDR[27:16] */ 81 82#define DATA 0x04 /* Indexed register set data register */ 83 /* DATA[31:0] */ 84 85#define IPR 0x08 /* Global interrupt pending register */ 86 /* Clear pending interrupts by writing a 1 to */ 87 /* the relevant bits and zero to the other bits */ 88#define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ 89#define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ 90#define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ 91#define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ 92#define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ 93#define IPR_SPI 0x00000800 /* SPI transaction completed */ 94#define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ 95#define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ 96#define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */ 97#define IPR_GPI 0x00000080 /* General Purpose input changed */ 98#define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */ 99#define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ 100#define IPR_TIMER2 0x00000010 /* 192000Hz Timer */ 101#define IPR_TIMER1 0x00000008 /* 44100Hz Timer */ 102#define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ 103#define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ 104#define IPR_PCI 0x00000001 /* PCI Bus error */ 105 106#define INTE 0x0c /* Interrupt enable register */ 107 108#define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ 109#define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ 110#define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ 111#define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ 112#define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ 113#define INTE_SPI 0x00000800 /* SPI transaction completed */ 114#define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ 115#define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ 116#define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */ 117#define INTE_GPI 0x00000080 /* General Purpose input changed */ 118#define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */ 119#define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ 120#define INTE_TIMER2 0x00000010 /* 192000Hz Timer */ 121#define INTE_TIMER1 0x00000008 /* 44100Hz Timer */ 122#define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ 123#define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ 124#define INTE_PCI 0x00000001 /* PCI Bus error */ 125 126#define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ 127#define HCFG 0x14 /* Hardware config register */ 128 /* 0x1000 causes AC3 to fails. It adds a dither bit. */ 129 130#define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */ 131#define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */ 132#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */ 133#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */ 134#define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */ 135#define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */ 136#define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */ 137#define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */ 138#define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */ 139#define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/ 140#define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/ 141#define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */ 142#define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */ 143#define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */ 144#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */ 145 /* NOTE: This should generally never be used. */ 146#define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */ 147 /* NOTE: This should generally never be used. */ 148#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */ 149 /* Should be set to 1 when the EMU10K1 is */ 150 /* completely initialized. */ 151#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ 152 /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */ 153 /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */ 154 /* SB Live 24bit: 155 * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in. 156 * bit 9 0 = Mute / 1 = Analog out. 157 * bit 10 0 = Line-in / 1 = Mic-in. 158 * bit 11 0 = ? / 1 = ? 159 * bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit. 160 * bit 13 0 = ? / 1 = ? 161 * bit 14 0 = Mute / 1 = Analog out 162 * bit 15 0 = ? / 1 = ? 163 * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit. 164 */ 165 /* 8 general purpose programmable In/Out pins. 166 * GPI [8:0] Read only. Default 0. 167 * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF) 168 * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin. 169 */ 170#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ 171 172#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ 173 174/********************************************************************************************************/ 175/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ 176/********************************************************************************************************/ 177 178/* Initally all registers from 0x00 to 0x3f have zero contents. */ 179#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ 180 /* One list entry: 4 bytes for DMA address, 181 * 4 bytes for period_size << 16. 182 * One list entry is 8 bytes long. 183 * One list entry for each period in the buffer. 184 */ 185 /* ADDR[31:0], Default: 0x0 */ 186#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ 187 /* SIZE[21:16], Default: 0x8 */ 188#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ 189 /* PTR[5:0], Default: 0x0 */ 190#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */ 191#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ 192 /* DMA[31:0], Default: 0x0 */ 193#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ 194 /* SIZE[31:16], Default: 0x0 */ 195#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ 196 /* POINTER[15:0], Default: 0x0 */ 197#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */ 198 /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */ 199#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */ 200 /* Cache size valid [5:0] */ 201#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */ 202#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */ 203 /* DMA[31:0], Default: 0x0 */ 204#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */ 205 /* SIZE[31:16], Default: 0x0 */ 206#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */ 207 /* POINTER[15:0], Default: 0x0 */ 208#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */ 209 /* Cache size valid [5:0] */ 210#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */ 211/* 0x21 - 0x3f unused */ 212#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */ 213 /* Playback (0x1<<channel_id) */ 214 /* Capture (0x100<<channel_id) */ 215 /* Playback sample rate 96000 = 0x20000 */ 216 /* Start Playback [3:0] (one bit per channel) 217 * Start Capture [11:8] (one bit per channel) 218 * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 219 * Playback mixer in enable [27:24] (one bit per channel) 220 * Playback mixer out enable [31:28] (one bit per channel) 221 */ 222/* The Digital out jack is shared with the Center/LFE Analogue output. 223 * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 224 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground 225 * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. 226 * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. 227 * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. 228 */ 229/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS 230 * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS 231 * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM. 232 * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output 233 */ 234/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel. 235 * 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. 236 */ 237#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */ 238#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */ 239#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */ 240#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */ 241 /* When Channel set to 0: */ 242#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */ 243#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */ 244#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */ 245#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */ 246#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */ 247#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */ 248#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */ 249#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */ 250#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */ 251#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */ 252#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */ 253#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */ 254#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */ 255#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */ 256#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */ 257#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */ 258#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */ 259#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */ 260#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */ 261#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */ 262#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */ 263#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */ 264#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */ 265 266 /* When Channel set to 1: */ 267#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */ 268#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */ 269#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */ 270#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */ 271#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */ 272#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */ 273#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */ 274#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ 275#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */ 276#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */ 277#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */ 278#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */ 279#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */ 280#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */ 281#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */ 282#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */ 283#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */ 284#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */ 285#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */ 286#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */ 287#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */ 288#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */ 289#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */ 290#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */ 291#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */ 292#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */ 293#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */ 294#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */ 295 296#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */ 297 /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE. 298 * But as the jack is shared, use 0xf00. 299 * The Windows2000 driver uses 0x0000000f for both digital and analog. 300 * 0xf00 introduces interesting noises onto the Center/LFE. 301 * If you turn the volume up, you hear computer noise, 302 * e.g. mouse moving, changing between app windows etc. 303 * So, I am going to set this to 0x0000000f all the time now, 304 * same as the windows driver does. 305 * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog. 306 */ 307 /* When Channel = 0: 308 * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit) 309 * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate) 310 * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass) 311 */ 312 /* When Channel = 1: 313 * SPDIF 0 User data [7:0] 314 * SPDIF 1 User data [15:8] 315 * SPDIF 0 User data [23:16] 316 * SPDIF 0 User data [31:24] 317 * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts. 318 */ 319#define WATERMARK 0x46 /* Test bit to indicate cache usage level */ 320#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS. 321 * When Channel = 0: Bits the same as SPCS channel 0. 322 * When Channel = 1: Bits the same as SPCS channel 1. 323 * When Channel = 2: 324 * SPDIF Input User data [16:0] 325 * SPDIF Input Frame count [21:16] 326 */ 327#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */ 328#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */ 329#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */ 330#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */ 331#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */ 332#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */ 333#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */ 334 /* 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 335 * Record source select for channel 0 [18:16] 336 * Record source select for channel 1 [22:20] 337 * Record source select for channel 2 [26:24] 338 * Record source select for channel 3 [30:28] 339 * 0 - SPDIF mixer output. 340 * 1 - i2s mixer output. 341 * 2 - SPDIF input. 342 * 3 - i2s input. 343 * 4 - AC97 capture. 344 * 5 - SRC output. 345 */ 346#define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */ 347#define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */ 348 349#define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */ 350#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */ 351#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */ 352#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */ 353#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */ 354 /* Channel_id's handle stereo channels. Channel X is a single mono channel */ 355 /* Host is input from the PCI bus. */ 356 /* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. 357 * Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. 358 * Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. 359 * Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. 360 * Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. 361 * Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. 362 * Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. 363 * Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. 364 */ 365 366#define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */ 367 /* SRC is input from the capture inputs. */ 368 /* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7. 369 * SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7. 370 * SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7. 371 * SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7. 372 * SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7. 373 * SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7. 374 * SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7. 375 * SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7. 376 */ 377 378#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */ 379 /* SPDIF Mixer input control: 380 * Invert SRC to SPDIF Mixer [7-0] (One bit per channel) 381 * Invert Host to SPDIF Mixer [15:8] (One bit per channel) 382 * SRC to SPDIF Mixer disable [23:16] (One bit per channel) 383 * Host to SPDIF Mixer disable [31:24] (One bit per channel) 384 */ 385#define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */ 386 /* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */ 387 /* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */ 388 /* One register for each of the 4 stereo streams. */ 389 /* SRC Right volume [7:0] 390 * SRC Left volume [15:8] 391 * Host Right volume [23:16] 392 * Host Left volume [31:24] 393 */ 394#define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */ 395 /* 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. */ 396#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */ 397 /* 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. */ 398#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */ 399 /* 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. */ 400#define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */ 401 /* 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. */ 402#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */ 403#define MIDI_UART_A_DATA 0x6c /* Midi Uart A Data */ 404#define MIDI_UART_A_CMD 0x6d /* Midi Uart A Command/Status */ 405#define MIDI_UART_B_DATA 0x6e /* Midi Uart B Data (currently unused) */ 406#define MIDI_UART_B_CMD 0x6f /* Midi Uart B Command/Status (currently unused) */ 407 408/* unique channel identifier for midi->channel */ 409 410#define CA0106_MIDI_CHAN_A 0x1 411#define CA0106_MIDI_CHAN_B 0x2 412 413/* from mpu401 */ 414 415#define CA0106_MIDI_INPUT_AVAIL 0x80 416#define CA0106_MIDI_OUTPUT_READY 0x40 417#define CA0106_MPU401_RESET 0xff 418#define CA0106_MPU401_ENTER_UART 0x3f 419#define CA0106_MPU401_ACK 0xfe 420 421#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */ 422 /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0 423 * Rate Locked [20] 424 * SPDIF Locked [21] For SPDIF channel only. 425 * Valid Audio [22] For SPDIF channel only. 426 */ 427#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */ 428 /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */ 429 /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */ 430 /* Sample rate output control register Channel=0 431 * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 432 * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) 433 * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source. 434 * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) 435 * Record mixer output enable [12:10] 436 * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 437 * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 438 * I2S output source select [18] (0=Audio from host, 1=Audio from SRC) 439 * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0) 440 * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.) 441 * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.) 442 * I2S input mode [23] (0=Slave, 1=Master) 443 * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) 444 * SPDIF output source select [26] (0=host, 1=SRC) 445 * Not used [27] 446 * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) 447 * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) 448 */ 449 /* Sample rate output control register Channel=1 450 * I2S Input 0 volume Right [7:0] 451 * I2S Input 0 volume Left [15:8] 452 * I2S Input 1 volume Right [23:16] 453 * I2S Input 1 volume Left [31:24] 454 */ 455 /* Sample rate output control register Channel=2 456 * SPDIF Input volume Right [23:16] 457 * SPDIF Input volume Left [31:24] 458 */ 459 /* Sample rate output control register Channel=3 460 * No used 461 */ 462#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */ 463#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */ 464#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */ 465#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */ 466 /* Audio output control 467 * AC97 output enable [5:0] 468 * I2S output enable [19:16] 469 * SPDIF output enable [27:24] 470 */ 471#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */ 472#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */ 473#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */ 474 /* Sets which Interrupts are enabled. */ 475 /* 0x00000001 = Half period. Playback. 476 * 0x00000010 = Full period. Playback. 477 * 0x00000100 = Half buffer. Playback. 478 * 0x00001000 = Full buffer. Playback. 479 * 0x00010000 = Half buffer. Capture. 480 * 0x00100000 = Full buffer. Capture. 481 * Capture can only do 2 periods. 482 * 0x01000000 = End audio. Playback. 483 * 0x40000000 = Half buffer Playback,Caputre xrun. 484 * 0x80000000 = Full buffer Playback,Caputre xrun. 485 */ 486#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */ 487 /* Shows which interrupts are active at the moment. */ 488 /* Same bit layout as EXTENDED_INT_MASK */ 489#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */ 490#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */ 491#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */ 492 /* Causes interrupts based on timer intervals. */ 493#define SPI 0x7a /* SPI: Serial Interface Register */ 494#define I2C_A 0x7b /* I2C Address. 32 bit */ 495#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */ 496#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */ 497//I2C values 498#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address 499#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W 500#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value 501#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag 502#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction 503#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode 504 505#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC 506#define I2C_A_ADC_READ 0x00000001 //To perform a read operation 507#define I2C_A_ADC_START 0x00000100 //Start I2C transaction 508#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort 509#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction 510#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode 511 512#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register 513#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register 514 515#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable 516#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control 517#define ADC_MASTER 0x0000000c //ADC Master Mode Control 518#define ADC_POWER 0x0000000d //ADC PowerDown Control 519#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL 520#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR 521#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1 522#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2 523#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3 524#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control 525#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control 526#define ADC_MUX 0x00000015 //ADC Mux offset 527 528#if 0 529/* FIXME: Not tested yet. */ 530#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain 531#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB 532#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute 533#define ADC_MUTE 0x000000c0 //Value to mute ADC 534#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select 535#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock 536#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter 537#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window 538#endif 539 540#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux 541#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) 542#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux 543#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux 544#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux 545 546#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ 547#define PCM_FRONT_CHANNEL 0 548#define PCM_REAR_CHANNEL 1 549#define PCM_CENTER_LFE_CHANNEL 2 550#define PCM_UNKNOWN_CHANNEL 3 551#define CONTROL_FRONT_CHANNEL 0 552#define CONTROL_REAR_CHANNEL 3 553#define CONTROL_CENTER_LFE_CHANNEL 1 554#define CONTROL_UNKNOWN_CHANNEL 2 555 556 557/* Based on WM8768 Datasheet Rev 4.2 page 32 */ 558#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */ 559#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */ 560 561#define SPI_LDA1_REG 0 /* digital attenuation */ 562#define SPI_RDA1_REG 1 563#define SPI_LDA2_REG 4 564#define SPI_RDA2_REG 5 565#define SPI_LDA3_REG 6 566#define SPI_RDA3_REG 7 567#define SPI_LDA4_REG 13 568#define SPI_RDA4_REG 14 569#define SPI_MASTDA_REG 8 570 571#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */ 572#define SPI_DA_BIT_0dB 0xff /* 0 dB */ 573#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */ 574 575#define SPI_PL_REG 2 576#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */ 577#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */ 578#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */ 579#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */ 580#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */ 581#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */ 582#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ 583#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ 584#define SPI_IZD_REG 2 585#define SPI_IZD_BIT (1<<4) /* infinite zero detect */ 586 587#define SPI_FMT_REG 3 588#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ 589#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */ 590#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */ 591#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */ 592#define SPI_LRP_REG 3 593#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */ 594#define SPI_BCP_REG 3 595#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */ 596#define SPI_IWL_REG 3 597#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */ 598#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */ 599#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */ 600#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */ 601 602#define SPI_MS_REG 10 603#define SPI_MS_BIT (1<<5) /* master mode */ 604#define SPI_RATE_REG 10 /* only applies in master mode */ 605#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */ 606#define SPI_RATE_BIT_192 (1<<6) 607#define SPI_RATE_BIT_256 (2<<6) 608#define SPI_RATE_BIT_384 (3<<6) 609#define SPI_RATE_BIT_512 (4<<6) 610#define SPI_RATE_BIT_768 (5<<6) 611 612/* They really do label the bit for the 4th channel "4" and not "3" */ 613#define SPI_DMUTE0_REG 9 614#define SPI_DMUTE1_REG 9 615#define SPI_DMUTE2_REG 9 616#define SPI_DMUTE4_REG 15 617#define SPI_DMUTE0_BIT (1<<3) 618#define SPI_DMUTE1_BIT (1<<4) 619#define SPI_DMUTE2_BIT (1<<5) 620#define SPI_DMUTE4_BIT (1<<2) 621 622#define SPI_PHASE0_REG 3 623#define SPI_PHASE1_REG 3 624#define SPI_PHASE2_REG 3 625#define SPI_PHASE4_REG 15 626#define SPI_PHASE0_BIT (1<<6) 627#define SPI_PHASE1_BIT (1<<7) 628#define SPI_PHASE2_BIT (1<<8) 629#define SPI_PHASE4_BIT (1<<3) 630 631#define SPI_PDWN_REG 2 /* power down all DACs */ 632#define SPI_PDWN_BIT (1<<2) 633#define SPI_DACD0_REG 10 /* power down individual DACs */ 634#define SPI_DACD1_REG 10 635#define SPI_DACD2_REG 10 636#define SPI_DACD4_REG 15 637#define SPI_DACD0_BIT (1<<1) 638#define SPI_DACD1_BIT (1<<2) 639#define SPI_DACD2_BIT (1<<3) 640#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */ 641 642#define SPI_PWRDNALL_REG 10 /* power down everything */ 643#define SPI_PWRDNALL_BIT (1<<4) 644 645#include "ca_midi.h" 646 647struct snd_ca0106; 648 649struct snd_ca0106_channel { 650 struct snd_ca0106 *emu; 651 int number; 652 int use; 653 void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel); 654 struct snd_ca0106_pcm *epcm; 655}; 656 657struct snd_ca0106_pcm { 658 struct snd_ca0106 *emu; 659 struct snd_pcm_substream *substream; 660 int channel_id; 661 unsigned short running; 662}; 663 664struct snd_ca0106_details { 665 u32 serial; 666 char * name; 667 int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. 668 ac97 = 1 -> Default to AC97 in. */ 669 int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in 670 gpio_type = 2 -> shared side-out/line-in. */ 671 int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume 672 controls, phone, mic, line-in and aux. */ 673 int spi_dac; /* spi_dac=1 adds the mute switch for each analog 674 output, front, rear, etc. */ 675}; 676 677// definition of the chip-specific record 678struct snd_ca0106 { 679 struct snd_card *card; 680 struct snd_ca0106_details *details; 681 struct pci_dev *pci; 682 683 unsigned long port; 684 struct resource *res_port; 685 int irq; 686 687 unsigned int serial; /* serial number */ 688 unsigned short model; /* subsystem id */ 689 690 spinlock_t emu_lock; 691 692 struct snd_ac97 *ac97; 693 struct snd_pcm *pcm[4]; 694 695 struct snd_ca0106_channel playback_channels[4]; 696 struct snd_ca0106_channel capture_channels[4]; 697 u32 spdif_bits[4]; /* s/pdif out default setup */ 698 u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */ 699 int spdif_enable; 700 int capture_source; 701 int i2c_capture_source; 702 u8 i2c_capture_volume[4][2]; 703 int capture_mic_line_in; 704 705 struct snd_dma_buffer buffer; 706 707 struct snd_ca_midi midi; 708 struct snd_ca_midi midi2; 709 710 u16 spi_dac_reg[16]; 711 712#ifdef CONFIG_PM 713#define NUM_SAVED_VOLUMES 9 714 unsigned int saved_vol[NUM_SAVED_VOLUMES]; 715#endif 716}; 717 718int snd_ca0106_mixer(struct snd_ca0106 *emu); 719int snd_ca0106_proc_init(struct snd_ca0106 * emu); 720 721unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, 722 unsigned int reg, 723 unsigned int chn); 724 725void snd_ca0106_ptr_write(struct snd_ca0106 *emu, 726 unsigned int reg, 727 unsigned int chn, 728 unsigned int data); 729 730int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); 731 732int snd_ca0106_spi_write(struct snd_ca0106 * emu, 733 unsigned int data); 734 735#ifdef CONFIG_PM 736void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip); 737void snd_ca0106_mixer_resume(struct snd_ca0106 *chip); 738#else 739#define snd_ca0106_mixer_suspend(chip) do { } while (0) 740#define snd_ca0106_mixer_resume(chip) do { } while (0) 741#endif 742