From 2c3c1048746a4622d8c89a29670120dc8fab93c4 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 20:49:45 +0200 Subject: Adding upstream version 6.1.76. Signed-off-by: Daniel Baumann --- sound/pci/emu10k1/p16v.h | 284 +++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 284 insertions(+) create mode 100644 sound/pci/emu10k1/p16v.h (limited to 'sound/pci/emu10k1/p16v.h') diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h new file mode 100644 index 000000000..3cdafa311 --- /dev/null +++ b/sound/pci/emu10k1/p16v.h @@ -0,0 +1,284 @@ +/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* + * Copyright (c) by James Courtier-Dutton + * 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 + */ + +/********************************************************************************************************/ +/* 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< 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 */ + -- cgit v1.2.3