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fdb1e56932
ca0106 driver code uses too generic names for its register definitions such as PTR, DATA, IPR, etc, which may eventually conflict with other code. This patch renames (some of) those register definitions with CA0106_ prefix to avoid the conflicts. Link: https://lore.kernel.org/r/20220210124227.11272-1-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai@suse.de>
727 lines
36 KiB
C
727 lines
36 KiB
C
/* SPDX-License-Identifier: GPL-2.0-or-later */
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/*
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* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
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* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
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* Version: 0.0.22
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*
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* FEATURES currently supported:
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* See ca0106_main.c for features.
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*
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* Changelog:
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* Support interrupts per period.
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* Removed noise from Center/LFE channel when in Analog mode.
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* Rename and remove mixer controls.
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* 0.0.6
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* Use separate card based DMA buffer for periods table list.
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* 0.0.7
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* Change remove and rename ctrls into lists.
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* 0.0.8
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* Try to fix capture sources.
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* 0.0.9
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* Fix AC3 output.
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* Enable S32_LE format support.
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* 0.0.10
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* Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
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* 0.0.11
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* Add Model name recognition.
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* 0.0.12
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* Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
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* Remove redundent "voice" handling.
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* 0.0.13
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* Single trigger call for multi channels.
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* 0.0.14
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* Set limits based on what the sound card hardware can do.
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* playback periods_min=2, periods_max=8
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* capture hw constraints require period_size = n * 64 bytes.
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* playback hw constraints require period_size = n * 64 bytes.
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* 0.0.15
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* Separated ca0106.c into separate functional .c files.
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* 0.0.16
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* Implement 192000 sample rate.
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* 0.0.17
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* Add support for SB0410 and SB0413.
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* 0.0.18
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* Modified Copyright message.
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* 0.0.19
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* Added I2C and SPI registers. Filled in interrupt enable.
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* 0.0.20
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* Added GPIO info for SB Live 24bit.
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* 0.0.21
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* Implement support for Line-in capture on SB Live 24bit.
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* 0.0.22
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* Add support for mute control on SB Live 24bit (cards w/ SPI DAC)
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*
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* This code was initially based on code from ALSA's emu10k1x.c which is:
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* Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com>
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*/
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/************************************************************************************************/
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/* PCI function 0 registers, address = <val> + PCIBASE0 */
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/************************************************************************************************/
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#define CA0106_PTR 0x00 /* Indexed register set pointer register */
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/* NOTE: The CHANNELNUM and ADDRESS words can */
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/* be modified independently of each other. */
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/* CNL[1:0], ADDR[27:16] */
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#define CA0106_DATA 0x04 /* Indexed register set data register */
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/* DATA[31:0] */
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#define CA0106_IPR 0x08 /* Global interrupt pending register */
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/* Clear pending interrupts by writing a 1 to */
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/* the relevant bits and zero to the other bits */
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#define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
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#define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
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#define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */
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#define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */
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#define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */
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#define IPR_SPI 0x00000800 /* SPI transaction completed */
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#define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */
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#define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */
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#define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */
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#define IPR_GPI 0x00000080 /* General Purpose input changed */
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#define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */
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#define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */
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#define IPR_TIMER2 0x00000010 /* 192000Hz Timer */
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#define IPR_TIMER1 0x00000008 /* 44100Hz Timer */
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#define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
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#define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */
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#define IPR_PCI 0x00000001 /* PCI Bus error */
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#define CA0106_INTE 0x0c /* Interrupt enable register */
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#define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */
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#define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */
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#define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */
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#define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */
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#define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */
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#define INTE_SPI 0x00000800 /* SPI transaction completed */
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#define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */
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#define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */
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#define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */
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#define INTE_GPI 0x00000080 /* General Purpose input changed */
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#define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */
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#define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */
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#define INTE_TIMER2 0x00000010 /* 192000Hz Timer */
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#define INTE_TIMER1 0x00000008 /* 44100Hz Timer */
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#define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */
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#define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */
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#define INTE_PCI 0x00000001 /* PCI Bus error */
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#define CA0106_UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */
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#define CA0106_HCFG 0x14 /* Hardware config register */
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/* 0x1000 causes AC3 to fails. It adds a dither bit. */
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#define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */
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#define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */
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#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */
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#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */
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#define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */
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#define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */
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#define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */
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#define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */
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#define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */
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#define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/
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#define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/
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#define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */
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#define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */
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#define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */
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#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */
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/* NOTE: This should generally never be used. */
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#define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */
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/* NOTE: This should generally never be used. */
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#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */
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/* Should be set to 1 when the EMU10K1 is */
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/* completely initialized. */
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#define CA0106_GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */
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/* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */
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/* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */
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/* SB Live 24bit:
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* bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in.
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* bit 9 0 = Mute / 1 = Analog out.
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* bit 10 0 = Line-in / 1 = Mic-in.
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* bit 11 0 = ? / 1 = ?
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* bit 12 0 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit.
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* bit 13 0 = ? / 1 = ?
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* bit 14 0 = Mute / 1 = Analog out
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* bit 15 0 = ? / 1 = ?
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* Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
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*/
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/* 8 general purpose programmable In/Out pins.
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* GPI [8:0] Read only. Default 0.
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* GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
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* GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
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*/
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#define CA0106_AC97DATA 0x1c /* AC97 register set data register (16 bit) */
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#define CA0106_AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
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/********************************************************************************************************/
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/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */
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/********************************************************************************************************/
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/* Initially all registers from 0x00 to 0x3f have zero contents. */
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#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
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/* One list entry: 4 bytes for DMA address,
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* 4 bytes for period_size << 16.
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* One list entry is 8 bytes long.
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* One list entry for each period in the buffer.
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*/
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/* ADDR[31:0], Default: 0x0 */
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#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
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/* SIZE[21:16], Default: 0x8 */
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#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
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/* PTR[5:0], Default: 0x0 */
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#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */
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#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA address */
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/* DMA[31:0], Default: 0x0 */
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#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
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/* SIZE[31:16], Default: 0x0 */
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#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
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/* POINTER[15:0], Default: 0x0 */
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#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */
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/* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
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#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */
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/* Cache size valid [5:0] */
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#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */
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#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
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/* DMA[31:0], Default: 0x0 */
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#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
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/* SIZE[31:16], Default: 0x0 */
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#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
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/* POINTER[15:0], Default: 0x0 */
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#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */
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/* Cache size valid [5:0] */
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#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */
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/* 0x21 - 0x3f unused */
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#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
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/* Playback (0x1<<channel_id) */
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/* Capture (0x100<<channel_id) */
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/* Playback sample rate 96000 = 0x20000 */
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/* Start Playback [3:0] (one bit per channel)
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* Start Capture [11:8] (one bit per channel)
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* Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
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* Playback mixer in enable [27:24] (one bit per channel)
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* Playback mixer out enable [31:28] (one bit per channel)
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*/
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/* The Digital out jack is shared with the Center/LFE Analogue output.
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* The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3
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* For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground
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* For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground.
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* Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red.
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* So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
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*/
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/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
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* The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
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* The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
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* Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
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*/
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/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
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* 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.
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*/
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#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */
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#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */
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#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */
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#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */
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/* When Channel set to 0: */
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#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */
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#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */
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#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */
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#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */
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#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */
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#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */
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#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */
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#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */
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#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */
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#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */
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#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */
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#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */
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#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */
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#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */
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#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */
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#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */
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#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */
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#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */
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#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */
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#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */
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#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */
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#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
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#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
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/* When Channel set to 1: */
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#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */
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#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */
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#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */
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#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */
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#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */
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#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */
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#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
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#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */
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#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */
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#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */
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#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */
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#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */
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#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */
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#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */
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#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
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/* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
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* But as the jack is shared, use 0xf00.
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* The Windows2000 driver uses 0x0000000f for both digital and analog.
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* 0xf00 introduces interesting noises onto the Center/LFE.
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* If you turn the volume up, you hear computer noise,
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* e.g. mouse moving, changing between app windows etc.
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* So, I am going to set this to 0x0000000f all the time now,
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* same as the windows driver does.
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* Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
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*/
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/* When Channel = 0:
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* Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
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* Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
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* SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
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*/
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/* When Channel = 1:
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* SPDIF 0 User data [7:0]
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* SPDIF 1 User data [15:8]
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* SPDIF 0 User data [23:16]
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* SPDIF 0 User data [31:24]
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* User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
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*/
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#define WATERMARK 0x46 /* Test bit to indicate cache usage level */
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#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS.
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* When Channel = 0: Bits the same as SPCS channel 0.
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* When Channel = 1: Bits the same as SPCS channel 1.
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* When Channel = 2:
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* SPDIF Input User data [16:0]
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* SPDIF Input Frame count [21:16]
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*/
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#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */
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#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */
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#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */
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#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */
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#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */
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#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */
|
|
#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */
|
|
/* 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
|
|
* 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 mixer output.
|
|
* 1 - i2s mixer output.
|
|
* 2 - SPDIF input.
|
|
* 3 - i2s input.
|
|
* 4 - AC97 capture.
|
|
* 5 - SRC output.
|
|
*/
|
|
#define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */
|
|
#define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */
|
|
|
|
#define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
|
|
#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
|
|
#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
|
|
#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
|
|
#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
|
|
/* Channel_id's handle stereo channels. Channel X is a single mono channel */
|
|
/* Host is input from the PCI bus. */
|
|
/* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
|
|
* Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
|
|
*/
|
|
|
|
#define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
|
|
/* SRC is input from the capture inputs. */
|
|
/* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
|
|
* SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
|
|
*/
|
|
|
|
#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */
|
|
/* SPDIF Mixer input control:
|
|
* Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
|
|
* Invert Host to SPDIF Mixer [15:8] (One bit per channel)
|
|
* SRC to SPDIF Mixer disable [23:16] (One bit per channel)
|
|
* Host to SPDIF Mixer disable [31:24] (One bit per channel)
|
|
*/
|
|
#define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
|
|
/* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
|
|
/* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
|
|
/* One register for each of the 4 stereo streams. */
|
|
/* SRC Right volume [7:0]
|
|
* SRC Left volume [15:8]
|
|
* Host Right volume [23:16]
|
|
* Host Left volume [31:24]
|
|
*/
|
|
#define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */
|
|
/* 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. */
|
|
#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */
|
|
/* 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. */
|
|
#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
|
|
/* 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. */
|
|
#define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */
|
|
/* 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. */
|
|
#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
|
|
#define MIDI_UART_A_DATA 0x6c /* Midi Uart A Data */
|
|
#define MIDI_UART_A_CMD 0x6d /* Midi Uart A Command/Status */
|
|
#define MIDI_UART_B_DATA 0x6e /* Midi Uart B Data (currently unused) */
|
|
#define MIDI_UART_B_CMD 0x6f /* Midi Uart B Command/Status (currently unused) */
|
|
|
|
/* unique channel identifier for midi->channel */
|
|
|
|
#define CA0106_MIDI_CHAN_A 0x1
|
|
#define CA0106_MIDI_CHAN_B 0x2
|
|
|
|
/* from mpu401 */
|
|
|
|
#define CA0106_MIDI_INPUT_AVAIL 0x80
|
|
#define CA0106_MIDI_OUTPUT_READY 0x40
|
|
#define CA0106_MPU401_RESET 0xff
|
|
#define CA0106_MPU401_ENTER_UART 0x3f
|
|
#define CA0106_MPU401_ACK 0xfe
|
|
|
|
#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */
|
|
/* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0
|
|
* Rate Locked [20]
|
|
* SPDIF Locked [21] For SPDIF channel only.
|
|
* Valid Audio [22] For SPDIF channel only.
|
|
*/
|
|
#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
|
|
/* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
|
|
/* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
|
|
/* Sample rate output control register Channel=0
|
|
* Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
|
|
* Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
|
|
* SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
|
|
* Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
|
|
* Record mixer output enable [12:10]
|
|
* I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
|
|
* I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
|
|
* I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
|
|
* Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
|
|
* I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
|
|
* I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
|
|
* I2S input mode [23] (0=Slave, 1=Master)
|
|
* SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
|
|
* SPDIF output source select [26] (0=host, 1=SRC)
|
|
* Not used [27]
|
|
* Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
|
|
* Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
|
|
*/
|
|
/* Sample rate output control register Channel=1
|
|
* I2S Input 0 volume Right [7:0]
|
|
* I2S Input 0 volume Left [15:8]
|
|
* I2S Input 1 volume Right [23:16]
|
|
* I2S Input 1 volume Left [31:24]
|
|
*/
|
|
/* Sample rate output control register Channel=2
|
|
* SPDIF Input volume Right [23:16]
|
|
* SPDIF Input volume Left [31:24]
|
|
*/
|
|
/* Sample rate output control register Channel=3
|
|
* No used
|
|
*/
|
|
#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
|
|
#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */
|
|
#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */
|
|
#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */
|
|
/* Audio output control
|
|
* AC97 output enable [5:0]
|
|
* I2S output enable [19:16]
|
|
* SPDIF output enable [27:24]
|
|
*/
|
|
#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */
|
|
#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */
|
|
#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */
|
|
/* Sets which Interrupts are enabled. */
|
|
/* 0x00000001 = Half period. Playback.
|
|
* 0x00000010 = Full period. Playback.
|
|
* 0x00000100 = Half buffer. Playback.
|
|
* 0x00001000 = Full buffer. Playback.
|
|
* 0x00010000 = Half buffer. Capture.
|
|
* 0x00100000 = Full buffer. Capture.
|
|
* Capture can only do 2 periods.
|
|
* 0x01000000 = End audio. Playback.
|
|
* 0x40000000 = Half buffer Playback,Caputre xrun.
|
|
* 0x80000000 = Full buffer Playback,Caputre xrun.
|
|
*/
|
|
#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */
|
|
/* Shows which interrupts are active at the moment. */
|
|
/* Same bit layout as EXTENDED_INT_MASK */
|
|
#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */
|
|
#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */
|
|
#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */
|
|
/* Causes interrupts based on timer intervals. */
|
|
#define SPI 0x7a /* SPI: Serial Interface Register */
|
|
#define I2C_A 0x7b /* I2C Address. 32 bit */
|
|
#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */
|
|
#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */
|
|
//I2C values
|
|
#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address
|
|
#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W
|
|
#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value
|
|
#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag
|
|
#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction
|
|
#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode
|
|
|
|
#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC
|
|
#define I2C_A_ADC_READ 0x00000001 //To perform a read operation
|
|
#define I2C_A_ADC_START 0x00000100 //Start I2C transaction
|
|
#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort
|
|
#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction
|
|
#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode
|
|
|
|
#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register
|
|
#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register
|
|
|
|
#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable
|
|
#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control
|
|
#define ADC_MASTER 0x0000000c //ADC Master Mode Control
|
|
#define ADC_POWER 0x0000000d //ADC PowerDown Control
|
|
#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL
|
|
#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR
|
|
#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1
|
|
#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2
|
|
#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3
|
|
#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control
|
|
#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control
|
|
#define ADC_MUX 0x00000015 //ADC Mux offset
|
|
|
|
#if 0
|
|
/* FIXME: Not tested yet. */
|
|
#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain
|
|
#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB
|
|
#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute
|
|
#define ADC_MUTE 0x000000c0 //Value to mute ADC
|
|
#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select
|
|
#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock
|
|
#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter
|
|
#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window
|
|
#endif
|
|
|
|
#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux
|
|
#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used)
|
|
#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux
|
|
#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux
|
|
#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux
|
|
|
|
#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
|
|
#define PCM_FRONT_CHANNEL 0
|
|
#define PCM_REAR_CHANNEL 1
|
|
#define PCM_CENTER_LFE_CHANNEL 2
|
|
#define PCM_UNKNOWN_CHANNEL 3
|
|
#define CONTROL_FRONT_CHANNEL 0
|
|
#define CONTROL_REAR_CHANNEL 3
|
|
#define CONTROL_CENTER_LFE_CHANNEL 1
|
|
#define CONTROL_UNKNOWN_CHANNEL 2
|
|
|
|
|
|
/* Based on WM8768 Datasheet Rev 4.2 page 32 */
|
|
#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */
|
|
#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */
|
|
|
|
#define SPI_LDA1_REG 0 /* digital attenuation */
|
|
#define SPI_RDA1_REG 1
|
|
#define SPI_LDA2_REG 4
|
|
#define SPI_RDA2_REG 5
|
|
#define SPI_LDA3_REG 6
|
|
#define SPI_RDA3_REG 7
|
|
#define SPI_LDA4_REG 13
|
|
#define SPI_RDA4_REG 14
|
|
#define SPI_MASTDA_REG 8
|
|
|
|
#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */
|
|
#define SPI_DA_BIT_0dB 0xff /* 0 dB */
|
|
#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */
|
|
|
|
#define SPI_PL_REG 2
|
|
#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */
|
|
#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */
|
|
#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */
|
|
#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */
|
|
#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */
|
|
#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */
|
|
#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */
|
|
#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */
|
|
#define SPI_IZD_REG 2
|
|
#define SPI_IZD_BIT (0<<4) /* infinite zero detect */
|
|
|
|
#define SPI_FMT_REG 3
|
|
#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */
|
|
#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */
|
|
#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */
|
|
#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */
|
|
#define SPI_LRP_REG 3
|
|
#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */
|
|
#define SPI_BCP_REG 3
|
|
#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */
|
|
#define SPI_IWL_REG 3
|
|
#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */
|
|
#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */
|
|
#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */
|
|
#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */
|
|
|
|
#define SPI_MS_REG 10
|
|
#define SPI_MS_BIT (1<<5) /* master mode */
|
|
#define SPI_RATE_REG 10 /* only applies in master mode */
|
|
#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */
|
|
#define SPI_RATE_BIT_192 (1<<6)
|
|
#define SPI_RATE_BIT_256 (2<<6)
|
|
#define SPI_RATE_BIT_384 (3<<6)
|
|
#define SPI_RATE_BIT_512 (4<<6)
|
|
#define SPI_RATE_BIT_768 (5<<6)
|
|
|
|
/* They really do label the bit for the 4th channel "4" and not "3" */
|
|
#define SPI_DMUTE0_REG 9
|
|
#define SPI_DMUTE1_REG 9
|
|
#define SPI_DMUTE2_REG 9
|
|
#define SPI_DMUTE4_REG 15
|
|
#define SPI_DMUTE0_BIT (1<<3)
|
|
#define SPI_DMUTE1_BIT (1<<4)
|
|
#define SPI_DMUTE2_BIT (1<<5)
|
|
#define SPI_DMUTE4_BIT (1<<2)
|
|
|
|
#define SPI_PHASE0_REG 3
|
|
#define SPI_PHASE1_REG 3
|
|
#define SPI_PHASE2_REG 3
|
|
#define SPI_PHASE4_REG 15
|
|
#define SPI_PHASE0_BIT (1<<6)
|
|
#define SPI_PHASE1_BIT (1<<7)
|
|
#define SPI_PHASE2_BIT (1<<8)
|
|
#define SPI_PHASE4_BIT (1<<3)
|
|
|
|
#define SPI_PDWN_REG 2 /* power down all DACs */
|
|
#define SPI_PDWN_BIT (1<<2)
|
|
#define SPI_DACD0_REG 10 /* power down individual DACs */
|
|
#define SPI_DACD1_REG 10
|
|
#define SPI_DACD2_REG 10
|
|
#define SPI_DACD4_REG 15
|
|
#define SPI_DACD0_BIT (1<<1)
|
|
#define SPI_DACD1_BIT (1<<2)
|
|
#define SPI_DACD2_BIT (1<<3)
|
|
#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */
|
|
|
|
#define SPI_PWRDNALL_REG 10 /* power down everything */
|
|
#define SPI_PWRDNALL_BIT (1<<4)
|
|
|
|
#include "ca_midi.h"
|
|
|
|
struct snd_ca0106;
|
|
|
|
struct snd_ca0106_channel {
|
|
struct snd_ca0106 *emu;
|
|
int number;
|
|
int use;
|
|
void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel);
|
|
struct snd_ca0106_pcm *epcm;
|
|
};
|
|
|
|
struct snd_ca0106_pcm {
|
|
struct snd_ca0106 *emu;
|
|
struct snd_pcm_substream *substream;
|
|
int channel_id;
|
|
unsigned short running;
|
|
};
|
|
|
|
struct snd_ca0106_details {
|
|
u32 serial;
|
|
char * name;
|
|
int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in.
|
|
ac97 = 1 -> Default to AC97 in. */
|
|
int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in
|
|
gpio_type = 2 -> shared side-out/line-in. */
|
|
int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume
|
|
controls, phone, mic, line-in and aux. */
|
|
u16 spi_dac; /* spi_dac = 0 -> no spi interface for DACs
|
|
spi_dac = 0x<front><rear><center-lfe><side>
|
|
-> specifies DAC id for each channel pair. */
|
|
};
|
|
|
|
// definition of the chip-specific record
|
|
struct snd_ca0106 {
|
|
struct snd_card *card;
|
|
const struct snd_ca0106_details *details;
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struct pci_dev *pci;
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unsigned long port;
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int irq;
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unsigned int serial; /* serial number */
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unsigned short model; /* subsystem id */
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spinlock_t emu_lock;
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struct snd_ac97 *ac97;
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struct snd_pcm *pcm[4];
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struct snd_ca0106_channel playback_channels[4];
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struct snd_ca0106_channel capture_channels[4];
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u32 spdif_bits[4]; /* s/pdif out default setup */
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u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */
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int spdif_enable;
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int capture_source;
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int i2c_capture_source;
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u8 i2c_capture_volume[4][2];
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int capture_mic_line_in;
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struct snd_dma_buffer *buffer;
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struct snd_ca_midi midi;
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struct snd_ca_midi midi2;
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u16 spi_dac_reg[16];
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#ifdef CONFIG_PM_SLEEP
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#define NUM_SAVED_VOLUMES 9
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unsigned int saved_vol[NUM_SAVED_VOLUMES];
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#endif
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};
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int snd_ca0106_mixer(struct snd_ca0106 *emu);
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int snd_ca0106_proc_init(struct snd_ca0106 * emu);
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unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu,
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unsigned int reg,
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unsigned int chn);
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void snd_ca0106_ptr_write(struct snd_ca0106 *emu,
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unsigned int reg,
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unsigned int chn,
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unsigned int data);
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int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value);
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int snd_ca0106_spi_write(struct snd_ca0106 * emu,
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unsigned int data);
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#ifdef CONFIG_PM_SLEEP
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void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip);
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void snd_ca0106_mixer_resume(struct snd_ca0106 *chip);
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#else
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#define snd_ca0106_mixer_suspend(chip) do { } while (0)
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#define snd_ca0106_mixer_resume(chip) do { } while (0)
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#endif
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