Author | Tokens | Token Proportion | Commits | Commit Proportion |
---|---|---|---|---|
Jaroslav Kysela | 272 | 98.91% | 1 | 33.33% |
Lucas De Marchi | 2 | 0.73% | 1 | 33.33% |
Uwe Kleine-König | 1 | 0.36% | 1 | 33.33% |
Total | 275 | 3 |
/* * Copyright (c) by James Courtier-Dutton <James@superbug.demon.co.uk> * Driver p16v chips * Version: 0.21 * * FEATURES currently supported: * Output fixed at S32_LE, 2 channel to hw:0,0 * Rates: 44.1, 48, 96, 192. * * Changelog: * 0.8 * Use separate card based buffer for periods table. * 0.9 * Use 2 channel output streams instead of 8 channel. * (8 channel output streams might be good for ASIO type output) * Corrected speaker output, so Front -> Front etc. * 0.10 * Fixed missed interrupts. * 0.11 * Add Sound card model number and names. * Add Analog volume controls. * 0.12 * Corrected playback interrupts. Now interrupt per period, instead of half period. * 0.13 * Use single trigger for multichannel. * 0.14 * Mic capture now works at fixed: S32_LE, 96000Hz, Stereo. * 0.15 * Force buffer_size / period_size == INTEGER. * 0.16 * Update p16v.c to work with changed alsa api. * 0.17 * Update p16v.c to work with changed alsa api. Removed boot_devs. * 0.18 * Merging with snd-emu10k1 driver. * 0.19 * One stereo channel at 24bit now works. * 0.20 * Added better register defines. * 0.21 * Split from p16v.c * * * BUGS: * Some stability problems when unloading the snd-p16v kernel module. * -- * * TODO: * SPDIF out. * Find out how to change capture sample rates. E.g. To record SPDIF at 48000Hz. * Currently capture fixed at 48000Hz. * * -- * GENERAL INFO: * Model: SB0240 * P16V Chip: CA0151-DBS * Audigy 2 Chip: CA0102-IAT * AC97 Codec: STAC 9721 * ADC: Philips 1361T (Stereo 24bit) * DAC: CS4382-K (8-channel, 24bit, 192Khz) * * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ /********************************************************************************************************/ /* Audigy2 P16V pointer-offset register set, accessed through the PTR2 and DATA2 registers */ /********************************************************************************************************/ /* The sample rate of the SPDIF outputs is set by modifying a register in the EMU10K2 PTR register A_SPDIF_SAMPLERATE. * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters. */ /* Initially all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. * One list entry is 8 bytes long. * One list entry for each period in the buffer. */ #define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ #define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ #define PLAYBACK_UNKNOWN3 0x03 /* Not used */ #define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */ #define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ #define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ #define PLAYBACK_FIFO_END_ADDRESS 0x07 /* Playback FIFO end address */ #define PLAYBACK_FIFO_POINTER 0x08 /* Playback FIFO pointer and number of valid sound samples in cache */ #define PLAYBACK_UNKNOWN9 0x09 /* Not used */ #define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */ #define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */ #define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */ #define CAPTURE_FIFO_POINTER 0x13 /* Capture FIFO pointer and number of valid sound samples in cache */ #define CAPTURE_P16V_VOLUME1 0x14 /* Low: Capture volume 0xXXXX3030 */ #define CAPTURE_P16V_VOLUME2 0x15 /* High:Has no effect on capture volume */ #define CAPTURE_P16V_SOURCE 0x16 /* P16V source select. Set to 0x0700E4E5 for AC97 CAPTURE */ /* [0:1] Capture input 0 channel select. 0 = Capture output 0. * 1 = Capture output 1. * 2 = Capture output 2. * 3 = Capture output 3. * [3:2] Capture input 1 channel select. 0 = Capture output 0. * 1 = Capture output 1. * 2 = Capture output 2. * 3 = Capture output 3. * [5:4] Capture input 2 channel select. 0 = Capture output 0. * 1 = Capture output 1. * 2 = Capture output 2. * 3 = Capture output 3. * [7:6] Capture input 3 channel select. 0 = Capture output 0. * 1 = Capture output 1. * 2 = Capture output 2. * 3 = Capture output 3. * [9:8] Playback input 0 channel select. 0 = Play output 0. * 1 = Play output 1. * 2 = Play output 2. * 3 = Play output 3. * [11:10] Playback input 1 channel select. 0 = Play output 0. * 1 = Play output 1. * 2 = Play output 2. * 3 = Play output 3. * [13:12] Playback input 2 channel select. 0 = Play output 0. * 1 = Play output 1. * 2 = Play output 2. * 3 = Play output 3. * [15:14] Playback input 3 channel select. 0 = Play output 0. * 1 = Play output 1. * 2 = Play output 2. * 3 = Play output 3. * [19:16] Playback mixer output enable. 1 bit per channel. * [23:20] Capture mixer output enable. 1 bit per channel. * [26:24] FX engine channel capture 0 = 0x60-0x67. * 1 = 0x68-0x6f. * 2 = 0x70-0x77. * 3 = 0x78-0x7f. * 4 = 0x80-0x87. * 5 = 0x88-0x8f. * 6 = 0x90-0x97. * 7 = 0x98-0x9f. * [31:27] Not used. */ /* 0x1 = capture on. * 0x100 = capture off. * 0x200 = capture off. * 0x1000 = capture off. */ #define CAPTURE_RATE_STATUS 0x17 /* Capture sample rate. Read only */ /* [15:0] Not used. * [18:16] Channel 0 Detected sample rate. 0 - 44.1khz * 1 - 48 khz * 2 - 96 khz * 3 - 192 khz * 7 - undefined rate. * [19] Channel 0. 1 - Valid, 0 - Not Valid. * [22:20] Channel 1 Detected sample rate. * [23] Channel 1. 1 - Valid, 0 - Not Valid. * [26:24] Channel 2 Detected sample rate. * [27] Channel 2. 1 - Valid, 0 - Not Valid. * [30:28] Channel 3 Detected sample rate. * [31] Channel 3. 1 - Valid, 0 - Not Valid. */ /* 0x18 - 0x1f unused */ #define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played. Read only */ /* 0x21 - 0x3f unused */ #define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */ /* Playback (0x1<<channel_id) Don't touch high 16bits. */ /* Capture (0x100<<channel_id). not tested */ /* Start Playback [3:0] (one bit per channel) * Start Capture [11:8] (one bit per channel) * Record source select for channel 0 [18:16] * Record source select for channel 1 [22:20] * Record source select for channel 2 [26:24] * Record source select for channel 3 [30:28] * 0 - SPDIF channel. * 1 - I2S channel. * 2 - SRC48 channel. * 3 - SRCMulti_SPDIF channel. * 4 - SRCMulti_I2S channel. * 5 - SPDIF channel. * 6 - fxengine capture. * 7 - AC97 capture. */ /* Default 41110000. * Writing 0xffffffff hangs the PC. * Writing 0xffff0000 -> 77770000 so it must be some sort of route. * bit 0x1 starts DMA playback on channel_id 0 */ /* 0x41,42 take values from 0 - 0xffffffff, but have no effect on playback */ /* 0x43,0x48 do not remember settings */ /* 0x41-45 unused */ #define WATERMARK 0x46 /* Test bit to indicate cache level usage */ /* Values it can have while playing on channel 0. * 0000f000, 0000f004, 0000f008, 0000f00c. * Readonly. */ /* 0x47-0x4f unused */ /* 0x50-0x5f Capture cache data */ #define SRCSel 0x60 /* SRCSel. Default 0x4. Bypass P16V 0x14 */ /* [0] 0 = 10K2 audio, 1 = SRC48 mixer output. * [2] 0 = 10K2 audio, 1 = SRCMulti SPDIF mixer output. * [4] 0 = 10K2 audio, 1 = SRCMulti I2S mixer output. */ /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */ /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */ /* SRC48 and SRCMULTI sample rate select and output select. */ /* 0xffffffff -> 0xC0000015 * 0xXXXXXXX4 = Enable Front Left/Right * Enable PCMs */ /* 0x61 -> 0x6c are Volume controls */ #define PLAYBACK_VOLUME_MIXER1 0x61 /* SRC48 Low to mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER2 0x62 /* SRC48 High to mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER3 0x63 /* SRCMULTI SPDIF Low to mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER4 0x64 /* SRCMULTI SPDIF High to mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER5 0x65 /* SRCMULTI I2S Low to mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER6 0x66 /* SRCMULTI I2S High to mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER7 0x67 /* P16V Low to SRCMULTI SPDIF mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER8 0x68 /* P16V High to SRCMULTI SPDIF mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER9 0x69 /* P16V Low to SRCMULTI I2S mixer input volume control. */ /* 0xXXXX3030 = PCM0 Volume (Front). * 0x3030XXXX = PCM1 Volume (Center) */ #define PLAYBACK_VOLUME_MIXER10 0x6a /* P16V High to SRCMULTI I2S mixer input volume control. */ /* 0x3030XXXX = PCM3 Volume (Rear). */ #define PLAYBACK_VOLUME_MIXER11 0x6b /* E10K2 Low to SRC48 mixer input volume control. */ #define PLAYBACK_VOLUME_MIXER12 0x6c /* E10K2 High to SRC48 mixer input volume control. */ #define SRC48_ENABLE 0x6d /* SRC48 input audio enable */ /* SRC48 converts samples rates 44.1, 48, 96, 192 to 48 khz. */ /* [23:16] The corresponding P16V channel to SRC48 enabled if == 1. * [31:24] The corresponding E10K2 channel to SRC48 enabled. */ #define SRCMULTI_ENABLE 0x6e /* SRCMulti input audio enable. Default 0xffffffff */ /* SRCMulti converts 48khz samples rates to 44.1, 48, 96, 192 to 48. */ /* [7:0] The corresponding P16V channel to SRCMulti_I2S enabled if == 1. * [15:8] The corresponding E10K2 channel to SRCMulti I2S enabled. * [23:16] The corresponding P16V channel to SRCMulti SPDIF enabled. * [31:24] The corresponding E10K2 channel to SRCMulti SPDIF enabled. */ /* Bypass P16V 0xff00ff00 * Bitmap. 0 = Off, 1 = On. * P16V playback outputs: * 0xXXXXXXX1 = PCM0 Left. (Front) * 0xXXXXXXX2 = PCM0 Right. * 0xXXXXXXX4 = PCM1 Left. (Center/LFE) * 0xXXXXXXX8 = PCM1 Right. * 0xXXXXXX1X = PCM2 Left. (Unknown) * 0xXXXXXX2X = PCM2 Right. * 0xXXXXXX4X = PCM3 Left. (Rear) * 0xXXXXXX8X = PCM3 Right. */ #define AUDIO_OUT_ENABLE 0x6f /* Default: 000100FF */ /* [3:0] Does something, but not documented. Probably capture enable. * [7:4] Playback channels enable. not documented. * [16] AC97 output enable if == 1 * [30] 0 = SRCMulti_I2S input from fxengine 0x68-0x6f. * 1 = SRCMulti_I2S input from SRC48 output. * [31] 0 = SRCMulti_SPDIF input from fxengine 0x60-0x67. * 1 = SRCMulti_SPDIF input from SRC48 output. */ /* 0xffffffff -> C00100FF */ /* 0 -> Not playback sound, irq still running */ /* 0xXXXXXX10 = PCM0 Left/Right On. (Front) * 0xXXXXXX20 = PCM1 Left/Right On. (Center/LFE) * 0xXXXXXX40 = PCM2 Left/Right On. (Unknown) * 0xXXXXXX80 = PCM3 Left/Right On. (Rear) */ #define PLAYBACK_SPDIF_SELECT 0x70 /* Default: 12030F00 */ /* 0xffffffff -> 3FF30FFF */ /* 0x00000001 pauses stream/irq fail. */ /* All other bits do not effect playback */ #define PLAYBACK_SPDIF_SRC_SELECT 0x71 /* Default: 0000E4E4 */ /* 0xffffffff -> F33FFFFF */ /* All bits do not effect playback */ #define PLAYBACK_SPDIF_USER_DATA0 0x72 /* SPDIF out user data 0 */ #define PLAYBACK_SPDIF_USER_DATA1 0x73 /* SPDIF out user data 1 */ /* 0x74-0x75 unknown */ #define CAPTURE_SPDIF_CONTROL 0x76 /* SPDIF in control setting */ #define CAPTURE_SPDIF_STATUS 0x77 /* SPDIF in status */ #define CAPURE_SPDIF_USER_DATA0 0x78 /* SPDIF in user data 0 */ #define CAPURE_SPDIF_USER_DATA1 0x79 /* SPDIF in user data 1 */ #define CAPURE_SPDIF_USER_DATA2 0x7a /* SPDIF in user data 2 */
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