From 1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Sat, 16 Apr 2005 15:20:36 -0700 Subject: Linux-2.6.12-rc2 Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip! --- sound/pci/ca0106/Makefile | 3 + sound/pci/ca0106/ca0106.h | 549 +++++++++++++++++ sound/pci/ca0106/ca0106_main.c | 1283 +++++++++++++++++++++++++++++++++++++++ sound/pci/ca0106/ca0106_mixer.c | 634 +++++++++++++++++++ sound/pci/ca0106/ca0106_proc.c | 436 +++++++++++++ 5 files changed, 2905 insertions(+) create mode 100644 sound/pci/ca0106/Makefile create mode 100644 sound/pci/ca0106/ca0106.h create mode 100644 sound/pci/ca0106/ca0106_main.c create mode 100644 sound/pci/ca0106/ca0106_mixer.c create mode 100644 sound/pci/ca0106/ca0106_proc.c (limited to 'sound/pci/ca0106') diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile new file mode 100644 index 000000000000..89c6ceee21f3 --- /dev/null +++ b/sound/pci/ca0106/Makefile @@ -0,0 +1,3 @@ +snd-ca0106-objs := ca0106_main.o ca0106_proc.o ca0106_mixer.o + +obj-$(CONFIG_SND_CA0106) += snd-ca0106.o diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h new file mode 100644 index 000000000000..deb028851056 --- /dev/null +++ b/sound/pci/ca0106/ca0106.h @@ -0,0 +1,549 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.20 + * + * FEATURES currently supported: + * See ca0106_main.c for features. + * + * Changelog: + * Support interrupts per period. + * Removed noise from Center/LFE channel when in Analog mode. + * Rename and remove mixer controls. + * 0.0.6 + * Use separate card based DMA buffer for periods table list. + * 0.0.7 + * Change remove and rename ctrls into lists. + * 0.0.8 + * Try to fix capture sources. + * 0.0.9 + * Fix AC3 output. + * Enable S32_LE format support. + * 0.0.10 + * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) + * 0.0.11 + * Add Model name recognition. + * 0.0.12 + * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. + * Remove redundent "voice" handling. + * 0.0.13 + * Single trigger call for multi channels. + * 0.0.14 + * Set limits based on what the sound card hardware can do. + * playback periods_min=2, periods_max=8 + * capture hw constraints require period_size = n * 64 bytes. + * playback hw constraints require period_size = n * 64 bytes. + * 0.0.15 + * Separated ca0106.c into separate functional .c files. + * 0.0.16 + * Implement 192000 sample rate. + * 0.0.17 + * Add support for SB0410 and SB0413. + * 0.0.18 + * Modified Copyright message. + * 0.0.19 + * Added I2C and SPI registers. Filled in interrupt enable. + * 0.0.20 + * Added GPIO info for SB Live 24bit. + * + * + * This code was initally based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes + * + * 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 + * + */ + +/************************************************************************************************/ +/* PCI function 0 registers, address = + PCIBASE0 */ +/************************************************************************************************/ + +#define PTR 0x00 /* Indexed register set pointer register */ + /* NOTE: The CHANNELNUM and ADDRESS words can */ + /* be modified independently of each other. */ + /* CNL[1:0], ADDR[27:16] */ + +#define DATA 0x04 /* Indexed register set data register */ + /* DATA[31:0] */ + +#define IPR 0x08 /* Global interrupt pending register */ + /* Clear pending interrupts by writing a 1 to */ + /* the relevant bits and zero to the other bits */ +#define IPR_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ +#define IPR_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ +#define IPR_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ +#define IPR_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ +#define IPR_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ +#define IPR_SPI 0x00000800 /* SPI transaction completed */ +#define IPR_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ +#define IPR_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ +#define IPR_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x76 */ +#define IPR_GPI 0x00000080 /* General Purpose input changed */ +#define IPR_SRC_LOCKED 0x00000040 /* SRC lock status changed */ +#define IPR_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ +#define IPR_TIMER2 0x00000010 /* 192000Hz Timer */ +#define IPR_TIMER1 0x00000008 /* 44100Hz Timer */ +#define IPR_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ +#define IPR_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ +#define IPR_PCI 0x00000001 /* PCI Bus error */ + +#define INTE 0x0c /* Interrupt enable register */ + +#define INTE_MIDI_RX_B 0x00020000 /* MIDI UART-B Receive buffer non-empty */ +#define INTE_MIDI_TX_B 0x00010000 /* MIDI UART-B Transmit buffer empty */ +#define INTE_SPDIF_IN_USER 0x00004000 /* SPDIF input user data has 16 more bits */ +#define INTE_SPDIF_OUT_USER 0x00002000 /* SPDIF output user data needs 16 more bits */ +#define INTE_SPDIF_OUT_FRAME 0x00001000 /* SPDIF frame about to start */ +#define INTE_SPI 0x00000800 /* SPI transaction completed */ +#define INTE_I2C_EEPROM 0x00000400 /* I2C EEPROM transaction completed */ +#define INTE_I2C_DAC 0x00000200 /* I2C DAC transaction completed */ +#define INTE_AI 0x00000100 /* Audio pending register changed. See PTR reg 0x75 */ +#define INTE_GPI 0x00000080 /* General Purpose input changed */ +#define INTE_SRC_LOCKED 0x00000040 /* SRC lock status changed */ +#define INTE_SPDIF_STATUS 0x00000020 /* SPDIF status changed */ +#define INTE_TIMER2 0x00000010 /* 192000Hz Timer */ +#define INTE_TIMER1 0x00000008 /* 44100Hz Timer */ +#define INTE_MIDI_RX_A 0x00000004 /* MIDI UART-A Receive buffer non-empty */ +#define INTE_MIDI_TX_A 0x00000002 /* MIDI UART-A Transmit buffer empty */ +#define INTE_PCI 0x00000001 /* PCI Bus error */ + +#define UNKNOWN10 0x10 /* Unknown ??. Defaults to 0 */ +#define HCFG 0x14 /* Hardware config register */ + /* 0x1000 causes AC3 to fails. It adds a dither bit. */ + +#define HCFG_STAC 0x10000000 /* Special mode for STAC9460 Codec. */ +#define HCFG_CAPTURE_I2S_BYPASS 0x08000000 /* 1 = bypass I2S input async SRC. */ +#define HCFG_CAPTURE_SPDIF_BYPASS 0x04000000 /* 1 = bypass SPDIF input async SRC. */ +#define HCFG_PLAYBACK_I2S_BYPASS 0x02000000 /* 0 = I2S IN mixer output, 1 = I2S IN1. */ +#define HCFG_FORCE_LOCK 0x01000000 /* For test only. Force input SRC tracker to lock. */ +#define HCFG_PLAYBACK_ATTENUATION 0x00006000 /* Playback attenuation mask. 0 = 0dB, 1 = 6dB, 2 = 12dB, 3 = Mute. */ +#define HCFG_PLAYBACK_DITHER 0x00001000 /* 1 = Add dither bit to all playback channels. */ +#define HCFG_PLAYBACK_S32_LE 0x00000800 /* 1 = S32_LE, 0 = S16_LE */ +#define HCFG_CAPTURE_S32_LE 0x00000400 /* 1 = S32_LE, 0 = S16_LE (S32_LE current not working) */ +#define HCFG_8_CHANNEL_PLAY 0x00000200 /* 1 = 8 channels, 0 = 2 channels per substream.*/ +#define HCFG_8_CHANNEL_CAPTURE 0x00000100 /* 1 = 8 channels, 0 = 2 channels per substream.*/ +#define HCFG_MONO 0x00000080 /* 1 = I2S Input mono */ +#define HCFG_I2S_OUTPUT 0x00000010 /* 1 = I2S Output disabled */ +#define HCFG_AC97 0x00000008 /* 0 = AC97 1.0, 1 = AC97 2.0 */ +#define HCFG_LOCK_PLAYBACK_CACHE 0x00000004 /* 1 = Cancel bustmaster accesses to soundcache */ + /* NOTE: This should generally never be used. */ +#define HCFG_LOCK_CAPTURE_CACHE 0x00000002 /* 1 = Cancel bustmaster accesses to soundcache */ + /* NOTE: This should generally never be used. */ +#define HCFG_AUDIOENABLE 0x00000001 /* 0 = CODECs transmit zero-valued samples */ + /* Should be set to 1 when the EMU10K1 is */ + /* completely initialized. */ +#define GPIO 0x18 /* Defaults: 005f03a3-Analog, 005f02a2-SPDIF. */ + /* Here pins 0,1,2,3,4,,6 are output. 5,7 are input */ + /* For the Audigy LS, pin 0 (or bit 8) controls the SPDIF/Analog jack. */ + /* SB Live 24bit: + * bit 8 0 = SPDIF in and out / 1 = Analog (Mic or Line)-in. + * bit 9 0 = Mute / 1 = Analog out. + * bit 10 0 = Line-in / 1 = Mic-in. + * bit 11 0 = ? / 1 = ? + * bit 12 0 = ? / 1 = ? + * bit 13 0 = ? / 1 = ? + * bit 14 0 = Mute / 1 = Analog out + * bit 15 0 = ? / 1 = ? + * Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit. + */ + /* 8 general purpose programmable In/Out pins. + * GPI [8:0] Read only. Default 0. + * GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF) + * GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin. + */ +#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */ + +#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */ + +/********************************************************************************************************/ +/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ +/********************************************************************************************************/ + +/* Initally 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. + */ + /* ADDR[31:0], Default: 0x0 */ +#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */ + /* SIZE[21:16], Default: 0x8 */ +#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */ + /* PTR[5:0], Default: 0x0 */ +#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */ +#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */ + /* DMA[31:0], Default: 0x0 */ +#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */ + /* SIZE[31:16], Default: 0x0 */ +#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */ + /* POINTER[15:0], Default: 0x0 */ +#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */ + /* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */ +#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */ + /* Cache size valid [5:0] */ +#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */ +#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */ + /* DMA[31:0], Default: 0x0 */ +#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */ + /* SIZE[31:16], Default: 0x0 */ +#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */ + /* POINTER[15:0], Default: 0x0 */ +#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */ + /* Cache size valid [5:0] */ +#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */ +/* 0x21 - 0x3f unused */ +#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */ + /* Playback (0x1< Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground + * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. + * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. + * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. + */ +/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS + * The Rear SPDIF can be used for Stereo PCM and also AC3/DTS + * The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM. + * Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output + */ +/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel. + * 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. + */ +#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */ +#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */ +#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */ +#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */ + /* When Channel set to 0: */ +#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */ +#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */ +#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */ +#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */ +#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */ +#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */ +#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */ +#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */ +#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */ +#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */ +#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */ +#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */ +#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */ +#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */ +#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */ +#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */ +#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */ +#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */ +#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */ +#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */ +#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */ +#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */ +#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */ + + /* When Channel set to 1: */ +#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */ +#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */ +#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */ +#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */ +#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */ +#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */ +#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */ +#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ +#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */ +#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */ +#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */ +#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */ +#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */ +#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */ +#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */ + +#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */ + /* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE. + * But as the jack is shared, use 0xf00. + * The Windows2000 driver uses 0x0000000f for both digital and analog. + * 0xf00 introduces interesting noises onto the Center/LFE. + * If you turn the volume up, you hear computer noise, + * e.g. mouse moving, changing between app windows etc. + * So, I am going to set this to 0x0000000f all the time now, + * same as the windows driver does. + * Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog. + */ + /* When Channel = 0: + * Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit) + * Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate) + * SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass) + */ + /* When Channel = 1: + * SPDIF 0 User data [7:0] + * SPDIF 1 User data [15:8] + * SPDIF 0 User data [23:16] + * SPDIF 0 User data [31:24] + * User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts. + */ +#define WATERMARK 0x46 /* Test bit to indicate cache usage level */ +#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS. + * When Channel = 0: Bits the same as SPCS channel 0. + * When Channel = 1: Bits the same as SPCS channel 1. + * When Channel = 2: + * SPDIF Input User data [16:0] + * SPDIF Input Frame count [21:16] + */ +#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */ +#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */ +#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */ +#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */ +#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */ +#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 UART_A_DATA 0x6c /* Uart, used in setting sample rates, bits per sample etc. */ +#define UART_A_CMD 0x6d /* Uart, used in setting sample rates, bits per sample etc. */ +#define UART_B_DATA 0x6e /* Uart, Unknown. */ +#define UART_B_CMD 0x6f /* Uart, Unknown. */ +#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_0 0x7c /* I2C Data Port 0. 32 bit */ +#define I2C_1 0x7d /* I2C Data Port 1. 32 bit */ + + +#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 + +typedef struct snd_ca0106_channel ca0106_channel_t; +typedef struct snd_ca0106 ca0106_t; +typedef struct snd_ca0106_pcm ca0106_pcm_t; + +struct snd_ca0106_channel { + ca0106_t *emu; + int number; + int use; + void (*interrupt)(ca0106_t *emu, ca0106_channel_t *channel); + ca0106_pcm_t *epcm; +}; + +struct snd_ca0106_pcm { + ca0106_t *emu; + snd_pcm_substream_t *substream; + int channel_id; + unsigned short running; +}; + +// definition of the chip-specific record +struct snd_ca0106 { + snd_card_t *card; + struct pci_dev *pci; + + unsigned long port; + struct resource *res_port; + int irq; + + unsigned int revision; /* chip revision */ + unsigned int serial; /* serial number */ + unsigned short model; /* subsystem id */ + + spinlock_t emu_lock; + + ac97_t *ac97; + snd_pcm_t *pcm; + + ca0106_channel_t playback_channels[4]; + ca0106_channel_t capture_channels[4]; + u32 spdif_bits[4]; /* s/pdif out setup */ + int spdif_enable; + int capture_source; + + struct snd_dma_buffer buffer; +}; + +int __devinit snd_ca0106_mixer(ca0106_t *emu); +int __devinit snd_ca0106_proc_init(ca0106_t * emu); + +unsigned int snd_ca0106_ptr_read(ca0106_t * emu, + unsigned int reg, + unsigned int chn); + +void snd_ca0106_ptr_write(ca0106_t *emu, + unsigned int reg, + unsigned int chn, + unsigned int data); + diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c new file mode 100644 index 000000000000..82533b45bc8c --- /dev/null +++ b/sound/pci/ca0106/ca0106_main.c @@ -0,0 +1,1283 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.22 + * + * FEATURES currently supported: + * Front, Rear and Center/LFE. + * Surround40 and Surround51. + * Capture from MIC an LINE IN input. + * SPDIF digital playback of PCM stereo and AC3/DTS works. + * (One can use a standard mono mini-jack to one RCA plugs cable. + * or one can use a standard stereo mini-jack to two RCA plugs cable. + * Plug one of the RCA plugs into the Coax input of the external decoder/receiver.) + * ( In theory one could output 3 different AC3 streams at once, to 3 different SPDIF outputs. ) + * Notes on how to capture sound: + * The AC97 is used in the PLAYBACK direction. + * The output from the AC97 chip, instead of reaching the speakers, is fed into the Philips 1361T ADC. + * So, to record from the MIC, set the MIC Playback volume to max, + * unmute the MIC and turn up the MASTER Playback volume. + * So, to prevent feedback when capturing, minimise the "Capture feedback into Playback" volume. + * + * The only playback controls that currently do anything are: - + * Analog Front + * Analog Rear + * Analog Center/LFE + * SPDIF Front + * SPDIF Rear + * SPDIF Center/LFE + * + * For capture from Mic in or Line in. + * Digital/Analog ( switch must be in Analog mode for CAPTURE. ) + * + * CAPTURE feedback into PLAYBACK + * + * Changelog: + * Support interrupts per period. + * Removed noise from Center/LFE channel when in Analog mode. + * Rename and remove mixer controls. + * 0.0.6 + * Use separate card based DMA buffer for periods table list. + * 0.0.7 + * Change remove and rename ctrls into lists. + * 0.0.8 + * Try to fix capture sources. + * 0.0.9 + * Fix AC3 output. + * Enable S32_LE format support. + * 0.0.10 + * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) + * 0.0.11 + * Add Model name recognition. + * 0.0.12 + * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. + * Remove redundent "voice" handling. + * 0.0.13 + * Single trigger call for multi channels. + * 0.0.14 + * Set limits based on what the sound card hardware can do. + * playback periods_min=2, periods_max=8 + * capture hw constraints require period_size = n * 64 bytes. + * playback hw constraints require period_size = n * 64 bytes. + * 0.0.15 + * Minor updates. + * 0.0.16 + * Implement 192000 sample rate. + * 0.0.17 + * Add support for SB0410 and SB0413. + * 0.0.18 + * Modified Copyright message. + * 0.0.19 + * Finally fix support for SB Live 24 bit. SB0410 and SB0413. + * The output codec needs resetting, otherwise all output is muted. + * 0.0.20 + * Merge "pci_disable_device(pci);" fixes. + * 0.0.21 + * Add 4 capture channels. (SPDIF only comes in on channel 0. ) + * Add SPDIF capture using optional digital I/O module for SB Live 24bit. (Analog capture does not yet work.) + * 0.0.22 + * Add support for MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97. From kiksen, bug #901 + * + * BUGS: + * Some stability problems when unloading the snd-ca0106 kernel module. + * -- + * + * TODO: + * 4 Capture channels, only one implemented so far. + * Other capture rates apart from 48khz not implemented. + * MIDI + * -- + * GENERAL INFO: + * Model: SB0310 + * P17 Chip: CA0106-DAT + * AC97 Codec: STAC 9721 + * ADC: Philips 1361T (Stereo 24bit) + * DAC: WM8746EDS (6-channel, 24bit, 192Khz) + * + * GENERAL INFO: + * Model: SB0410 + * P17 Chip: CA0106-DAT + * AC97 Codec: None + * ADC: WM8775EDS (4 Channel) + * DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support) + * SPDIF Out control switches between Mic in and SPDIF out. + * No sound out or mic input working yet. + * + * GENERAL INFO: + * Model: SB0413 + * P17 Chip: CA0106-DAT + * AC97 Codec: None. + * ADC: Unknown + * DAC: Unknown + * Trying to handle it like the SB0410. + * + * This code was initally based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes + * + * 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 + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +MODULE_AUTHOR("James Courtier-Dutton "); +MODULE_DESCRIPTION("CA0106"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Creative,SB CA0106 chip}}"); + +// module parameters (see "Module Parameters") +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard."); + +#include "ca0106.h" + +typedef struct { + u32 serial; + char * name; +} ca0106_names_t; + +static ca0106_names_t ca0106_chip_names[] = { + { 0x10021102, "AudigyLS [SB0310]"} , + { 0x10051102, "AudigyLS [SB0310b]"} , /* Unknown AudigyLS that also says SB0310 on it */ + { 0x10061102, "Live! 7.1 24bit [SB0410]"} , /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */ + { 0x10071102, "Live! 7.1 24bit [SB0413]"} , /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ + { 0x10091462, "MSI K8N Diamond MB [SB0438]"}, /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ + { 0, "AudigyLS [Unknown]" } +}; + +/* hardware definition */ +static snd_pcm_hardware_t snd_ca0106_playback_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000, + .rate_min = 48000, + .rate_max = 192000, + .channels_min = 2, //1, + .channels_max = 2, //6, + .buffer_bytes_max = ((65536 - 64) * 8), + .period_bytes_min = 64, + .period_bytes_max = (65536 - 64), + .periods_min = 2, + .periods_max = 8, + .fifo_size = 0, +}; + +static snd_pcm_hardware_t snd_ca0106_capture_hw = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = ((65536 - 64) * 8), + .period_bytes_min = 64, + .period_bytes_max = (65536 - 64), + .periods_min = 2, + .periods_max = 2, + .fifo_size = 0, +}; + +unsigned int snd_ca0106_ptr_read(ca0106_t * emu, + unsigned int reg, + unsigned int chn) +{ + unsigned long flags; + unsigned int regptr, val; + + regptr = (reg << 16) | chn; + + spin_lock_irqsave(&emu->emu_lock, flags); + outl(regptr, emu->port + PTR); + val = inl(emu->port + DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); + return val; +} + +void snd_ca0106_ptr_write(ca0106_t *emu, + unsigned int reg, + unsigned int chn, + unsigned int data) +{ + unsigned int regptr; + unsigned long flags; + + regptr = (reg << 16) | chn; + + spin_lock_irqsave(&emu->emu_lock, flags); + outl(regptr, emu->port + PTR); + outl(data, emu->port + DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +static void snd_ca0106_intr_enable(ca0106_t *emu, unsigned int intrenb) +{ + unsigned long flags; + unsigned int enable; + + spin_lock_irqsave(&emu->emu_lock, flags); + enable = inl(emu->port + INTE) | intrenb; + outl(enable, emu->port + INTE); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +static void snd_ca0106_pcm_free_substream(snd_pcm_runtime_t *runtime) +{ + ca0106_pcm_t *epcm = runtime->private_data; + + if (epcm) { + kfree(epcm); + } +} + +/* open_playback callback */ +static int snd_ca0106_pcm_open_playback_channel(snd_pcm_substream_t *substream, int channel_id) +{ + ca0106_t *chip = snd_pcm_substream_chip(substream); + ca0106_channel_t *channel = &(chip->playback_channels[channel_id]); + ca0106_pcm_t *epcm; + snd_pcm_runtime_t *runtime = substream->runtime; + int err; + + epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL); + + if (epcm == NULL) + return -ENOMEM; + epcm->emu = chip; + epcm->substream = substream; + epcm->channel_id=channel_id; + + runtime->private_data = epcm; + runtime->private_free = snd_ca0106_pcm_free_substream; + + runtime->hw = snd_ca0106_playback_hw; + + channel->emu = chip; + channel->number = channel_id; + + channel->use=1; + //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + //channel->interrupt = snd_ca0106_pcm_channel_interrupt; + channel->epcm=epcm; + if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + return err; + if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + return err; + return 0; +} + +/* close callback */ +static int snd_ca0106_pcm_close_playback(snd_pcm_substream_t *substream) +{ + ca0106_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + ca0106_pcm_t *epcm = runtime->private_data; + chip->playback_channels[epcm->channel_id].use=0; +/* FIXME: maybe zero others */ + return 0; +} + +static int snd_ca0106_pcm_open_playback_front(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_center_lfe(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_unknown(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_rear(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL); +} + +/* open_capture callback */ +static int snd_ca0106_pcm_open_capture_channel(snd_pcm_substream_t *substream, int channel_id) +{ + ca0106_t *chip = snd_pcm_substream_chip(substream); + ca0106_channel_t *channel = &(chip->capture_channels[channel_id]); + ca0106_pcm_t *epcm; + snd_pcm_runtime_t *runtime = substream->runtime; + int err; + + epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL); + if (epcm == NULL) { + snd_printk("open_capture_channel: failed epcm alloc\n"); + return -ENOMEM; + } + epcm->emu = chip; + epcm->substream = substream; + epcm->channel_id=channel_id; + + runtime->private_data = epcm; + runtime->private_free = snd_ca0106_pcm_free_substream; + + runtime->hw = snd_ca0106_capture_hw; + + channel->emu = chip; + channel->number = channel_id; + + channel->use=1; + //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + //channel->interrupt = snd_ca0106_pcm_channel_interrupt; + channel->epcm=epcm; + if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) + return err; + //snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes); + if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0) + return err; + return 0; +} + +/* close callback */ +static int snd_ca0106_pcm_close_capture(snd_pcm_substream_t *substream) +{ + ca0106_t *chip = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + ca0106_pcm_t *epcm = runtime->private_data; + chip->capture_channels[epcm->channel_id].use=0; +/* FIXME: maybe zero others */ + return 0; +} + +static int snd_ca0106_pcm_open_0_capture(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 0); +} + +static int snd_ca0106_pcm_open_1_capture(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 1); +} + +static int snd_ca0106_pcm_open_2_capture(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 2); +} + +static int snd_ca0106_pcm_open_3_capture(snd_pcm_substream_t *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 3); +} + +/* hw_params callback */ +static int snd_ca0106_pcm_hw_params_playback(snd_pcm_substream_t *substream, + snd_pcm_hw_params_t * hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_ca0106_pcm_hw_free_playback(snd_pcm_substream_t *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* hw_params callback */ +static int snd_ca0106_pcm_hw_params_capture(snd_pcm_substream_t *substream, + snd_pcm_hw_params_t * hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +/* hw_free callback */ +static int snd_ca0106_pcm_hw_free_capture(snd_pcm_substream_t *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare playback callback */ +static int snd_ca0106_pcm_prepare_playback(snd_pcm_substream_t *substream) +{ + ca0106_t *emu = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + ca0106_pcm_t *epcm = runtime->private_data; + int channel = epcm->channel_id; + u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel)); + u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size); + u32 hcfg_mask = HCFG_PLAYBACK_S32_LE; + u32 hcfg_set = 0x00000000; + u32 hcfg; + u32 reg40_mask = 0x30000 << (channel<<1); + u32 reg40_set = 0; + u32 reg40; + /* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */ + u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */ + u32 reg71_set = 0; + u32 reg71; + int i; + + //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); + //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); + //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); + /* Rate can be set per channel. */ + /* reg40 control host to fifo */ + /* reg71 controls DAC rate. */ + switch (runtime->rate) { + case 44100: + reg40_set = 0x10000 << (channel<<1); + reg71_set = 0x01010000; + break; + case 48000: + reg40_set = 0; + reg71_set = 0; + break; + case 96000: + reg40_set = 0x20000 << (channel<<1); + reg71_set = 0x02020000; + break; + case 192000: + reg40_set = 0x30000 << (channel<<1); + reg71_set = 0x03030000; + break; + default: + reg40_set = 0; + reg71_set = 0; + break; + } + /* Format is a global setting */ + /* FIXME: Only let the first channel accessed set this. */ + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + hcfg_set = 0; + break; + case SNDRV_PCM_FORMAT_S32_LE: + hcfg_set = HCFG_PLAYBACK_S32_LE; + break; + default: + hcfg_set = 0; + break; + } + hcfg = inl(emu->port + HCFG) ; + hcfg = (hcfg & ~hcfg_mask) | hcfg_set; + outl(hcfg, emu->port + HCFG); + reg40 = snd_ca0106_ptr_read(emu, 0x40, 0); + reg40 = (reg40 & ~reg40_mask) | reg40_set; + snd_ca0106_ptr_write(emu, 0x40, 0, reg40); + reg71 = snd_ca0106_ptr_read(emu, 0x71, 0); + reg71 = (reg71 & ~reg71_mask) | reg71_set; + snd_ca0106_ptr_write(emu, 0x71, 0, reg71); + + /* FIXME: Check emu->buffer.size before actually writing to it. */ + for(i=0; i < runtime->periods; i++) { + table_base[i*2]=runtime->dma_addr+(i*period_size_bytes); + table_base[(i*2)+1]=period_size_bytes<<16; + } + + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel)); + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19); + snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0); + snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr); + snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes + /* FIXME test what 0 bytes does. */ + snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, 0); // buffer size in bytes + snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0); + snd_ca0106_ptr_write(emu, 0x07, channel, 0x0); + snd_ca0106_ptr_write(emu, 0x08, channel, 0); + snd_ca0106_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */ +#if 0 + snd_ca0106_ptr_write(emu, SPCS0, 0, + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT ); + } +#endif + + return 0; +} + +/* prepare capture callback */ +static int snd_ca0106_pcm_prepare_capture(snd_pcm_substream_t *substream) +{ + ca0106_t *emu = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + ca0106_pcm_t *epcm = runtime->private_data; + int channel = epcm->channel_id; + //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + snd_ca0106_ptr_write(emu, 0x13, channel, 0); + snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); + snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes + snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0); + + return 0; +} + +/* trigger_playback callback */ +static int snd_ca0106_pcm_trigger_playback(snd_pcm_substream_t *substream, + int cmd) +{ + ca0106_t *emu = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime; + ca0106_pcm_t *epcm; + int channel; + int result = 0; + struct list_head *pos; + snd_pcm_substream_t *s; + u32 basic = 0; + u32 extended = 0; + int running=0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + running=1; + break; + case SNDRV_PCM_TRIGGER_STOP: + default: + running=0; + break; + } + snd_pcm_group_for_each(pos, substream) { + s = snd_pcm_group_substream_entry(pos); + runtime = s->runtime; + epcm = runtime->private_data; + channel = epcm->channel_id; + //snd_printk("channel=%d\n",channel); + epcm->running = running; + basic |= (0x1<runtime; + ca0106_pcm_t *epcm = runtime->private_data; + int channel = epcm->channel_id; + int result = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) | (0x110000<running = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(0x100<running = 0; + break; + default: + result = -EINVAL; + break; + } + return result; +} + +/* pointer_playback callback */ +static snd_pcm_uframes_t +snd_ca0106_pcm_pointer_playback(snd_pcm_substream_t *substream) +{ + ca0106_t *emu = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + ca0106_pcm_t *epcm = runtime->private_data; + snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0; + int channel = epcm->channel_id; + + if (!epcm->running) + return 0; + + ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); + ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel); + ptr2 = bytes_to_frames(runtime, ptr1); + ptr2+= (ptr4 >> 3) * runtime->period_size; + ptr=ptr2; + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + + return ptr; +} + +/* pointer_capture callback */ +static snd_pcm_uframes_t +snd_ca0106_pcm_pointer_capture(snd_pcm_substream_t *substream) +{ + ca0106_t *emu = snd_pcm_substream_chip(substream); + snd_pcm_runtime_t *runtime = substream->runtime; + ca0106_pcm_t *epcm = runtime->private_data; + snd_pcm_uframes_t ptr, ptr1, ptr2 = 0; + int channel = channel=epcm->channel_id; + + if (!epcm->running) + return 0; + + ptr1 = snd_ca0106_ptr_read(emu, CAPTURE_POINTER, channel); + ptr2 = bytes_to_frames(runtime, ptr1); + ptr=ptr2; + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); + + return ptr; +} + +/* operators */ +static snd_pcm_ops_t snd_ca0106_playback_front_ops = { + .open = snd_ca0106_pcm_open_playback_front, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + +static snd_pcm_ops_t snd_ca0106_capture_0_ops = { + .open = snd_ca0106_pcm_open_0_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static snd_pcm_ops_t snd_ca0106_capture_1_ops = { + .open = snd_ca0106_pcm_open_1_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static snd_pcm_ops_t snd_ca0106_capture_2_ops = { + .open = snd_ca0106_pcm_open_2_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static snd_pcm_ops_t snd_ca0106_capture_3_ops = { + .open = snd_ca0106_pcm_open_3_capture, + .close = snd_ca0106_pcm_close_capture, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_capture, + .hw_free = snd_ca0106_pcm_hw_free_capture, + .prepare = snd_ca0106_pcm_prepare_capture, + .trigger = snd_ca0106_pcm_trigger_capture, + .pointer = snd_ca0106_pcm_pointer_capture, +}; + +static snd_pcm_ops_t snd_ca0106_playback_center_lfe_ops = { + .open = snd_ca0106_pcm_open_playback_center_lfe, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + +static snd_pcm_ops_t snd_ca0106_playback_unknown_ops = { + .open = snd_ca0106_pcm_open_playback_unknown, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + +static snd_pcm_ops_t snd_ca0106_playback_rear_ops = { + .open = snd_ca0106_pcm_open_playback_rear, + .close = snd_ca0106_pcm_close_playback, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ca0106_pcm_hw_params_playback, + .hw_free = snd_ca0106_pcm_hw_free_playback, + .prepare = snd_ca0106_pcm_prepare_playback, + .trigger = snd_ca0106_pcm_trigger_playback, + .pointer = snd_ca0106_pcm_pointer_playback, +}; + + +static unsigned short snd_ca0106_ac97_read(ac97_t *ac97, + unsigned short reg) +{ + ca0106_t *emu = ac97->private_data; + unsigned long flags; + unsigned short val; + + spin_lock_irqsave(&emu->emu_lock, flags); + outb(reg, emu->port + AC97ADDRESS); + val = inw(emu->port + AC97DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); + return val; +} + +static void snd_ca0106_ac97_write(ac97_t *ac97, + unsigned short reg, unsigned short val) +{ + ca0106_t *emu = ac97->private_data; + unsigned long flags; + + spin_lock_irqsave(&emu->emu_lock, flags); + outb(reg, emu->port + AC97ADDRESS); + outw(val, emu->port + AC97DATA); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +static int snd_ca0106_ac97(ca0106_t *chip) +{ + ac97_bus_t *pbus; + ac97_template_t ac97; + int err; + static ac97_bus_ops_t ops = { + .write = snd_ca0106_ac97_write, + .read = snd_ca0106_ac97_read, + }; + + if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0) + return err; + pbus->no_vra = 1; /* we don't need VRA */ + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = chip; + return snd_ac97_mixer(pbus, &ac97, &chip->ac97); +} + +static int snd_ca0106_free(ca0106_t *chip) +{ + if (chip->res_port != NULL) { /* avoid access to already used hardware */ + // disable interrupts + snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0); + outl(0, chip->port + INTE); + snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0); + udelay(1000); + // disable audio + //outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG); + outl(0, chip->port + HCFG); + /* FIXME: We need to stop and DMA transfers here. + * But as I am not sure how yet, we cannot from the dma pages. + * So we can fix: snd-malloc: Memory leak? pages not freed = 8 + */ + } + // release the data +#if 1 + if (chip->buffer.area) + snd_dma_free_pages(&chip->buffer); +#endif + + // release the i/o port + if (chip->res_port) { + release_resource(chip->res_port); + kfree_nocheck(chip->res_port); + } + // release the irq + if (chip->irq >= 0) + free_irq(chip->irq, (void *)chip); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_ca0106_dev_free(snd_device_t *device) +{ + ca0106_t *chip = device->device_data; + return snd_ca0106_free(chip); +} + +static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id, + struct pt_regs *regs) +{ + unsigned int status; + + ca0106_t *chip = dev_id; + int i; + int mask; + unsigned int stat76; + ca0106_channel_t *pchannel; + + spin_lock(&chip->emu_lock); + + status = inl(chip->port + IPR); + + // call updater, unlock before it + spin_unlock(&chip->emu_lock); + + if (! status) + return IRQ_NONE; + + stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); + //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); + //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ + for(i = 0; i < 4; i++) { + pchannel = &(chip->playback_channels[i]); + if(stat76 & mask) { +/* FIXME: Select the correct substream for period elapsed */ + if(pchannel->use) { + snd_pcm_period_elapsed(pchannel->epcm->substream); + //printk(KERN_INFO "interrupt [%d] used\n", i); + } + } + //printk(KERN_INFO "channel=%p\n",pchannel); + //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); + mask <<= 1; + } + mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */ + for(i = 0; i < 4; i++) { + pchannel = &(chip->capture_channels[i]); + if(stat76 & mask) { +/* FIXME: Select the correct substream for period elapsed */ + if(pchannel->use) { + snd_pcm_period_elapsed(pchannel->epcm->substream); + //printk(KERN_INFO "interrupt [%d] used\n", i); + } + } + //printk(KERN_INFO "channel=%p\n",pchannel); + //printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number); + mask <<= 1; + } + + snd_ca0106_ptr_write(chip, EXTENDED_INT, 0, stat76); + spin_lock(&chip->emu_lock); + // acknowledge the interrupt if necessary + outl(status, chip->port+IPR); + + spin_unlock(&chip->emu_lock); + + return IRQ_HANDLED; +} + +static void snd_ca0106_pcm_free(snd_pcm_t *pcm) +{ + ca0106_t *emu = pcm->private_data; + emu->pcm = NULL; + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int __devinit snd_ca0106_pcm(ca0106_t *emu, int device, snd_pcm_t **rpcm) +{ + snd_pcm_t *pcm; + snd_pcm_substream_t *substream; + int err; + + if (rpcm) + *rpcm = NULL; + if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0) + return err; + + pcm->private_data = emu; + pcm->private_free = snd_ca0106_pcm_free; + + switch (device) { + case 0: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops); + break; + case 1: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops); + break; + case 2: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops); + break; + case 3: + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops); + break; + } + + pcm->info_flags = 0; + pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX; + strcpy(pcm->name, "CA0106"); + emu->pcm = pcm; + + for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + substream; + substream = substream->next) { + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + 64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */ + return err; + } + + for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + substream; + substream = substream->next) { + if ((err = snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(emu->pci), + 64*1024, 64*1024)) < 0) + return err; + } + + if (rpcm) + *rpcm = pcm; + + return 0; +} + +static int __devinit snd_ca0106_create(snd_card_t *card, + struct pci_dev *pci, + ca0106_t **rchip) +{ + ca0106_t *chip; + int err; + int ch; + static snd_device_ops_t ops = { + .dev_free = snd_ca0106_dev_free, + }; + + *rchip = NULL; + + if ((err = pci_enable_device(pci)) < 0) + return err; + if (pci_set_dma_mask(pci, 0xffffffffUL) < 0 || + pci_set_consistent_dma_mask(pci, 0xffffffffUL) < 0) { + printk(KERN_ERR "error to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kcalloc(1, sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + chip->irq = -1; + + spin_lock_init(&chip->emu_lock); + + chip->port = pci_resource_start(pci, 0); + if ((chip->res_port = request_region(chip->port, 0x20, + "snd_ca0106")) == NULL) { + snd_ca0106_free(chip); + printk(KERN_ERR "cannot allocate the port\n"); + return -EBUSY; + } + + if (request_irq(pci->irq, snd_ca0106_interrupt, + SA_INTERRUPT|SA_SHIRQ, "snd_ca0106", + (void *)chip)) { + snd_ca0106_free(chip); + printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + + /* This stores the periods table. */ + if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) { + snd_ca0106_free(chip); + return -ENOMEM; + } + + pci_set_master(pci); + /* read revision & serial */ + pci_read_config_byte(pci, PCI_REVISION_ID, (char *)&chip->revision); + pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); +#if 1 + printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model, + chip->revision, chip->serial); +#endif + + outl(0, chip->port + INTE); + + /* + * Init to 0x02109204 : + * Clock accuracy = 0 (1000ppm) + * Sample Rate = 2 (48kHz) + * Audio Channel = 1 (Left of 2) + * Source Number = 0 (Unspecified) + * Generation Status = 1 (Original for Cat Code 12) + * Cat Code = 12 (Digital Signal Mixer) + * Mode = 0 (Mode 0) + * Emphasis = 0 (None) + * CP = 1 (Copyright unasserted) + * AN = 0 (Audio data) + * P = 0 (Consumer) + */ + snd_ca0106_ptr_write(chip, SPCS0, 0, + chip->spdif_bits[0] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + /* Only SPCS1 has been tested */ + snd_ca0106_ptr_write(chip, SPCS1, 0, + chip->spdif_bits[1] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + snd_ca0106_ptr_write(chip, SPCS2, 0, + chip->spdif_bits[2] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + snd_ca0106_ptr_write(chip, SPCS3, 0, + chip->spdif_bits[3] = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT); + + snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000); + snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000); + + /* Write 0x8000 to AC97_REC_GAIN to mute it. */ + outb(AC97_REC_GAIN, chip->port + AC97ADDRESS); + outw(0x8000, chip->port + AC97DATA); +#if 0 + snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006); + snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006); + snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006); + snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006); +#endif + + //snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */ + /* Analog or Digital output */ + snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000b0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers */ + chip->spdif_enable = 0; /* Set digital SPDIF output off */ + chip->capture_source = 3; /* Set CAPTURE_SOURCE */ + //snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */ + //snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */ + + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */ + snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410); + snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676); + snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410); + snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676); + for(ch = 0; ch < 4; ch++) { + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */ + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030); + //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */ + //snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */ + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */ + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */ + } + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */ + chip->capture_source = 3; /* Set CAPTURE_SOURCE */ + + if ((chip->serial == 0x10061102) || + (chip->serial == 0x10071102) || + (chip->serial == 0x10091462)) { /* The SB0410 and SB0413 use GPIO differently. */ + /* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */ + outl(0x0, chip->port+GPIO); + //outl(0x00f0e000, chip->port+GPIO); /* Analog */ + outl(0x005f4300, chip->port+GPIO); /* Analog */ + } else { + outl(0x0, chip->port+GPIO); + outl(0x005f03a3, chip->port+GPIO); /* Analog */ + //outl(0x005f02a2, chip->port+GPIO); /* SPDIF */ + } + snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */ + + //outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG); + //outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ + //outl(0x00000009, chip->port+HCFG); + outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */ + + if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, + chip, &ops)) < 0) { + snd_ca0106_free(chip); + return err; + } + *rchip = chip; + return 0; +} + +static int __devinit snd_ca0106_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + snd_card_t *card; + ca0106_t *chip; + ca0106_names_t *c; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + if ((err = snd_ca0106_create(card, pci, &chip)) < 0) { + snd_card_free(card); + return err; + } + + if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) { + snd_card_free(card); + return err; + } + if ((chip->serial != 0x10061102) && + (chip->serial != 0x10071102) && + (chip->serial != 0x10091462) ) { /* The SB0410 and SB0413 do not have an ac97 chip. */ + if ((err = snd_ca0106_ac97(chip)) < 0) { + snd_card_free(card); + return err; + } + } + if ((err = snd_ca0106_mixer(chip)) < 0) { + snd_card_free(card); + return err; + } + + snd_ca0106_proc_init(chip); + + strcpy(card->driver, "CA0106"); + strcpy(card->shortname, "CA0106"); + + for (c=ca0106_chip_names; c->serial; c++) { + if (c->serial == chip->serial) break; + } + sprintf(card->longname, "%s at 0x%lx irq %i", + c->name, chip->port, chip->irq); + + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static void __devexit snd_ca0106_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +// PCI IDs +static struct pci_device_id snd_ca0106_ids[] = { + { 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */ + { 0, } +}; +MODULE_DEVICE_TABLE(pci, snd_ca0106_ids); + +// pci_driver definition +static struct pci_driver driver = { + .name = "CA0106", + .id_table = snd_ca0106_ids, + .probe = snd_ca0106_probe, + .remove = __devexit_p(snd_ca0106_remove), +}; + +// initialization of the module +static int __init alsa_card_ca0106_init(void) +{ + int err; + + if ((err = pci_module_init(&driver)) > 0) + return err; + + return 0; +} + +// clean up the module +static void __exit alsa_card_ca0106_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_ca0106_init) +module_exit(alsa_card_ca0106_exit) diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c new file mode 100644 index 000000000000..97bed1b0899d --- /dev/null +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -0,0 +1,634 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.16 + * + * FEATURES currently supported: + * See ca0106_main.c for features. + * + * Changelog: + * Support interrupts per period. + * Removed noise from Center/LFE channel when in Analog mode. + * Rename and remove mixer controls. + * 0.0.6 + * Use separate card based DMA buffer for periods table list. + * 0.0.7 + * Change remove and rename ctrls into lists. + * 0.0.8 + * Try to fix capture sources. + * 0.0.9 + * Fix AC3 output. + * Enable S32_LE format support. + * 0.0.10 + * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) + * 0.0.11 + * Add Model name recognition. + * 0.0.12 + * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. + * Remove redundent "voice" handling. + * 0.0.13 + * Single trigger call for multi channels. + * 0.0.14 + * Set limits based on what the sound card hardware can do. + * playback periods_min=2, periods_max=8 + * capture hw constraints require period_size = n * 64 bytes. + * playback hw constraints require period_size = n * 64 bytes. + * 0.0.15 + * Separated ca0106.c into separate functional .c files. + * 0.0.16 + * Modified Copyright message. + * + * This code was initally based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes + * + * 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 + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ca0106.h" + +static int snd_ca0106_shared_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; +} + +static int snd_ca0106_shared_spdif_get(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->spdif_enable; + return 0; +} + +static int snd_ca0106_shared_spdif_put(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + u32 mask; + + val = ucontrol->value.enumerated.item[0] ; + change = (emu->spdif_enable != val); + if (change) { + emu->spdif_enable = val; + if (val == 1) { + /* Digital */ + snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000); + snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, + snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000); + mask = inl(emu->port + GPIO) & ~0x101; + outl(mask, emu->port + GPIO); + + } else { + /* Analog */ + snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000b0000); + snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, + snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000); + mask = inl(emu->port + GPIO) | 0x101; + outl(mask, emu->port + GPIO); + } + } + return change; +} + +static snd_kcontrol_new_t snd_ca0106_shared_spdif __devinitdata = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Out", + .info = snd_ca0106_shared_spdif_info, + .get = snd_ca0106_shared_spdif_get, + .put = snd_ca0106_shared_spdif_put +}; + +static int snd_ca0106_capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo) +{ + static char *texts[6] = { "SPDIF out", "i2s mixer out", "SPDIF in", "i2s in", "AC97 in", "SRC out" }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 6; + if (uinfo->value.enumerated.item > 5) + uinfo->value.enumerated.item = 5; + strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_ca0106_capture_source_get(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->capture_source; + return 0; +} + +static int snd_ca0106_capture_source_put(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + u32 mask; + u32 source; + + val = ucontrol->value.enumerated.item[0] ; + change = (emu->capture_source != val); + if (change) { + emu->capture_source = val; + source = (val << 28) | (val << 24) | (val << 20) | (val << 16); + mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff; + snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask); + } + return change; +} + +static snd_kcontrol_new_t snd_ca0106_capture_source __devinitdata = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_ca0106_capture_source_info, + .get = snd_ca0106_capture_source_get, + .put = snd_ca0106_capture_source_put +}; + +static int snd_ca0106_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int snd_ca0106_spdif_get(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff; + ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff; + ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff; + ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff; + return 0; +} + +static int snd_ca0106_spdif_get_mask(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ucontrol->value.iec958.status[0] = 0xff; + ucontrol->value.iec958.status[1] = 0xff; + ucontrol->value.iec958.status[2] = 0xff; + ucontrol->value.iec958.status[3] = 0xff; + return 0; +} + +static int snd_ca0106_spdif_put(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int change; + unsigned int val; + + val = (ucontrol->value.iec958.status[0] << 0) | + (ucontrol->value.iec958.status[1] << 8) | + (ucontrol->value.iec958.status[2] << 16) | + (ucontrol->value.iec958.status[3] << 24); + change = val != emu->spdif_bits[idx]; + if (change) { + snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val); + emu->spdif_bits[idx] = val; + } + return change; +} + +static snd_kcontrol_new_t snd_ca0106_spdif_mask_control = +{ + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), + .count = 4, + .info = snd_ca0106_spdif_info, + .get = snd_ca0106_spdif_get_mask +}; + +static snd_kcontrol_new_t snd_ca0106_spdif_control = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .count = 4, + .info = snd_ca0106_spdif_info, + .get = snd_ca0106_spdif_get, + .put = snd_ca0106_spdif_put +}; + +static int snd_ca0106_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 255; + return 0; +} + +static int snd_ca0106_volume_get(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol, int reg, int channel_id) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + unsigned int value; + + value = snd_ca0106_ptr_read(emu, reg, channel_id); + ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */ + ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */ + return 0; +} + +static int snd_ca0106_volume_get_spdif_front(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_FRONT_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} + +static int snd_ca0106_volume_get_spdif_center_lfe(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_CENTER_LFE_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_get_spdif_unknown(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_UNKNOWN_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_get_spdif_rear(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_REAR_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_get_analog_front(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_FRONT_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} + +static int snd_ca0106_volume_get_analog_center_lfe(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_CENTER_LFE_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_get_analog_unknown(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_UNKNOWN_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_get_analog_rear(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_REAR_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} + +static int snd_ca0106_volume_get_feedback(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = 1; + int reg = CAPTURE_CONTROL; + return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id); +} + +static int snd_ca0106_volume_put(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol, int reg, int channel_id) +{ + ca0106_t *emu = snd_kcontrol_chip(kcontrol); + unsigned int value; + //value = snd_ca0106_ptr_read(emu, reg, channel_id); + //value = value & 0xffff; + value = ((0xff - ucontrol->value.integer.value[0]) << 24) | ((0xff - ucontrol->value.integer.value[1]) << 16); + value = value | ((0xff - ucontrol->value.integer.value[0]) << 8) | ((0xff - ucontrol->value.integer.value[1]) ); + snd_ca0106_ptr_write(emu, reg, channel_id, value); + return 1; +} +static int snd_ca0106_volume_put_spdif_front(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_FRONT_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_spdif_center_lfe(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_CENTER_LFE_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_spdif_unknown(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_UNKNOWN_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_spdif_rear(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_REAR_CHANNEL; + int reg = PLAYBACK_VOLUME1; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_analog_front(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_FRONT_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_analog_center_lfe(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_CENTER_LFE_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_analog_unknown(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_UNKNOWN_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} +static int snd_ca0106_volume_put_analog_rear(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = CONTROL_REAR_CHANNEL; + int reg = PLAYBACK_VOLUME2; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} + +static int snd_ca0106_volume_put_feedback(snd_kcontrol_t * kcontrol, + snd_ctl_elem_value_t * ucontrol) +{ + int channel_id = 1; + int reg = CAPTURE_CONTROL; + return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id); +} + +static snd_kcontrol_new_t snd_ca0106_volume_control_analog_front = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Front Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_analog_front, + .put = snd_ca0106_volume_put_analog_front +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_analog_center_lfe = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Center/LFE Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_analog_center_lfe, + .put = snd_ca0106_volume_put_analog_center_lfe +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_analog_unknown = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Unknown Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_analog_unknown, + .put = snd_ca0106_volume_put_analog_unknown +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_analog_rear = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Rear Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_analog_rear, + .put = snd_ca0106_volume_put_analog_rear +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_front = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Front Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_spdif_front, + .put = snd_ca0106_volume_put_spdif_front +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_center_lfe = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Center/LFE Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_spdif_center_lfe, + .put = snd_ca0106_volume_put_spdif_center_lfe +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_unknown = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Unknown Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_spdif_unknown, + .put = snd_ca0106_volume_put_spdif_unknown +}; +static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_rear = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "SPDIF Rear Volume", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_spdif_rear, + .put = snd_ca0106_volume_put_spdif_rear +}; + +static snd_kcontrol_new_t snd_ca0106_volume_control_feedback = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "CAPTURE feedback into PLAYBACK", + .info = snd_ca0106_volume_info, + .get = snd_ca0106_volume_get_feedback, + .put = snd_ca0106_volume_put_feedback +}; + + +static int remove_ctl(snd_card_t *card, const char *name) +{ + snd_ctl_elem_id_t id; + memset(&id, 0, sizeof(id)); + strcpy(id.name, name); + id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_remove_id(card, &id); +} + +static snd_kcontrol_t *ctl_find(snd_card_t *card, const char *name) +{ + snd_ctl_elem_id_t sid; + memset(&sid, 0, sizeof(sid)); + /* FIXME: strcpy is bad. */ + strcpy(sid.name, name); + sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + return snd_ctl_find_id(card, &sid); +} + +static int rename_ctl(snd_card_t *card, const char *src, const char *dst) +{ + snd_kcontrol_t *kctl = ctl_find(card, src); + if (kctl) { + strcpy(kctl->id.name, dst); + return 0; + } + return -ENOENT; +} + +int __devinit snd_ca0106_mixer(ca0106_t *emu) +{ + int err; + snd_kcontrol_t *kctl; + snd_card_t *card = emu->card; + char **c; + static char *ca0106_remove_ctls[] = { + "Master Mono Playback Switch", + "Master Mono Playback Volume", + "3D Control - Switch", + "3D Control Sigmatel - Depth", + "PCM Playback Switch", + "PCM Playback Volume", + "CD Playback Switch", + "CD Playback Volume", + "Phone Playback Switch", + "Phone Playback Volume", + "Video Playback Switch", + "Video Playback Volume", + "PC Speaker Playback Switch", + "PC Speaker Playback Volume", + "Mono Output Select", + "Capture Source", + "Capture Switch", + "Capture Volume", + "External Amplifier", + "Sigmatel 4-Speaker Stereo Playback Switch", + "Sigmatel Surround Phase Inversion Playback ", + NULL + }; + static char *ca0106_rename_ctls[] = { + "Master Playback Switch", "Capture Switch", + "Master Playback Volume", "Capture Volume", + "Line Playback Switch", "AC97 Line Capture Switch", + "Line Playback Volume", "AC97 Line Capture Volume", + "Aux Playback Switch", "AC97 Aux Capture Switch", + "Aux Playback Volume", "AC97 Aux Capture Volume", + "Mic Playback Switch", "AC97 Mic Capture Switch", + "Mic Playback Volume", "AC97 Mic Capture Volume", + "Mic Select", "AC97 Mic Select", + "Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)", + NULL + }; +#if 1 + for (c=ca0106_remove_ctls; *c; c++) + remove_ctl(card, *c); + for (c=ca0106_rename_ctls; *c; c += 2) + rename_ctl(card, c[0], c[1]); +#endif + + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_front, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_rear, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_center_lfe, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_unknown, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_front, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_rear, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_center_lfe, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_unknown, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_feedback, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_mask_control, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_shared_spdif, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = snd_ctl_new1(&snd_ca0106_capture_source, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + if ((kctl = ctl_find(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT))) != NULL) { + /* already defined by ac97, remove it */ + /* FIXME: or do we need both controls? */ + remove_ctl(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT)); + } + if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_control, emu)) == NULL) + return -ENOMEM; + if ((err = snd_ctl_add(card, kctl))) + return err; + return 0; +} + diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c new file mode 100644 index 000000000000..afb711421e47 --- /dev/null +++ b/sound/pci/ca0106/ca0106_proc.c @@ -0,0 +1,436 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.17 + * + * FEATURES currently supported: + * See ca0106_main.c for features. + * + * Changelog: + * Support interrupts per period. + * Removed noise from Center/LFE channel when in Analog mode. + * Rename and remove mixer controls. + * 0.0.6 + * Use separate card based DMA buffer for periods table list. + * 0.0.7 + * Change remove and rename ctrls into lists. + * 0.0.8 + * Try to fix capture sources. + * 0.0.9 + * Fix AC3 output. + * Enable S32_LE format support. + * 0.0.10 + * Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".) + * 0.0.11 + * Add Model name recognition. + * 0.0.12 + * Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period. + * Remove redundent "voice" handling. + * 0.0.13 + * Single trigger call for multi channels. + * 0.0.14 + * Set limits based on what the sound card hardware can do. + * playback periods_min=2, periods_max=8 + * capture hw constraints require period_size = n * 64 bytes. + * playback hw constraints require period_size = n * 64 bytes. + * 0.0.15 + * Separate ca0106.c into separate functional .c files. + * 0.0.16 + * Modified Copyright message. + * 0.0.17 + * Add iec958 file in proc file system to show status of SPDIF in. + * + * This code was initally based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes + * + * 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 + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ca0106.h" + + +struct snd_ca0106_category_str { + int val; + const char *name; +}; + +static struct snd_ca0106_category_str snd_ca0106_con_category[] = { + { IEC958_AES1_CON_DAT, "DAT" }, + { IEC958_AES1_CON_VCR, "VCR" }, + { IEC958_AES1_CON_MICROPHONE, "microphone" }, + { IEC958_AES1_CON_SYNTHESIZER, "synthesizer" }, + { IEC958_AES1_CON_RATE_CONVERTER, "rate converter" }, + { IEC958_AES1_CON_MIXER, "mixer" }, + { IEC958_AES1_CON_SAMPLER, "sampler" }, + { IEC958_AES1_CON_PCM_CODER, "PCM coder" }, + { IEC958_AES1_CON_IEC908_CD, "CD" }, + { IEC958_AES1_CON_NON_IEC908_CD, "non-IEC908 CD" }, + { IEC958_AES1_CON_GENERAL, "general" }, +}; + + +void snd_ca0106_proc_dump_iec958( snd_info_buffer_t *buffer, u32 value) +{ + int i; + u32 status[4]; + status[0] = value & 0xff; + status[1] = (value >> 8) & 0xff; + status[2] = (value >> 16) & 0xff; + status[3] = (value >> 24) & 0xff; + + if (! (status[0] & IEC958_AES0_PROFESSIONAL)) { + /* consumer */ + snd_iprintf(buffer, "Mode: consumer\n"); + snd_iprintf(buffer, "Data: "); + if (!(status[0] & IEC958_AES0_NONAUDIO)) { + snd_iprintf(buffer, "audio\n"); + } else { + snd_iprintf(buffer, "non-audio\n"); + } + snd_iprintf(buffer, "Rate: "); + switch (status[3] & IEC958_AES3_CON_FS) { + case IEC958_AES3_CON_FS_44100: + snd_iprintf(buffer, "44100 Hz\n"); + break; + case IEC958_AES3_CON_FS_48000: + snd_iprintf(buffer, "48000 Hz\n"); + break; + case IEC958_AES3_CON_FS_32000: + snd_iprintf(buffer, "32000 Hz\n"); + break; + default: + snd_iprintf(buffer, "unknown\n"); + break; + } + snd_iprintf(buffer, "Copyright: "); + if (status[0] & IEC958_AES0_CON_NOT_COPYRIGHT) { + snd_iprintf(buffer, "permitted\n"); + } else { + snd_iprintf(buffer, "protected\n"); + } + snd_iprintf(buffer, "Emphasis: "); + if ((status[0] & IEC958_AES0_CON_EMPHASIS) != IEC958_AES0_CON_EMPHASIS_5015) { + snd_iprintf(buffer, "none\n"); + } else { + snd_iprintf(buffer, "50/15us\n"); + } + snd_iprintf(buffer, "Category: "); + for (i = 0; i < ARRAY_SIZE(snd_ca0106_con_category); i++) { + if ((status[1] & IEC958_AES1_CON_CATEGORY) == snd_ca0106_con_category[i].val) { + snd_iprintf(buffer, "%s\n", snd_ca0106_con_category[i].name); + break; + } + } + if (i >= ARRAY_SIZE(snd_ca0106_con_category)) { + snd_iprintf(buffer, "unknown 0x%x\n", status[1] & IEC958_AES1_CON_CATEGORY); + } + snd_iprintf(buffer, "Original: "); + if (status[1] & IEC958_AES1_CON_ORIGINAL) { + snd_iprintf(buffer, "original\n"); + } else { + snd_iprintf(buffer, "1st generation\n"); + } + snd_iprintf(buffer, "Clock: "); + switch (status[3] & IEC958_AES3_CON_CLOCK) { + case IEC958_AES3_CON_CLOCK_1000PPM: + snd_iprintf(buffer, "1000 ppm\n"); + break; + case IEC958_AES3_CON_CLOCK_50PPM: + snd_iprintf(buffer, "50 ppm\n"); + break; + case IEC958_AES3_CON_CLOCK_VARIABLE: + snd_iprintf(buffer, "variable pitch\n"); + break; + default: + snd_iprintf(buffer, "unknown\n"); + break; + } + } else { + snd_iprintf(buffer, "Mode: professional\n"); + snd_iprintf(buffer, "Data: "); + if (!(status[0] & IEC958_AES0_NONAUDIO)) { + snd_iprintf(buffer, "audio\n"); + } else { + snd_iprintf(buffer, "non-audio\n"); + } + snd_iprintf(buffer, "Rate: "); + switch (status[0] & IEC958_AES0_PRO_FS) { + case IEC958_AES0_PRO_FS_44100: + snd_iprintf(buffer, "44100 Hz\n"); + break; + case IEC958_AES0_PRO_FS_48000: + snd_iprintf(buffer, "48000 Hz\n"); + break; + case IEC958_AES0_PRO_FS_32000: + snd_iprintf(buffer, "32000 Hz\n"); + break; + default: + snd_iprintf(buffer, "unknown\n"); + break; + } + snd_iprintf(buffer, "Rate Locked: "); + if (status[0] & IEC958_AES0_PRO_FREQ_UNLOCKED) + snd_iprintf(buffer, "no\n"); + else + snd_iprintf(buffer, "yes\n"); + snd_iprintf(buffer, "Emphasis: "); + switch (status[0] & IEC958_AES0_PRO_EMPHASIS) { + case IEC958_AES0_PRO_EMPHASIS_CCITT: + snd_iprintf(buffer, "CCITT J.17\n"); + break; + case IEC958_AES0_PRO_EMPHASIS_NONE: + snd_iprintf(buffer, "none\n"); + break; + case IEC958_AES0_PRO_EMPHASIS_5015: + snd_iprintf(buffer, "50/15us\n"); + break; + case IEC958_AES0_PRO_EMPHASIS_NOTID: + default: + snd_iprintf(buffer, "unknown\n"); + break; + } + snd_iprintf(buffer, "Stereophonic: "); + if ((status[1] & IEC958_AES1_PRO_MODE) == IEC958_AES1_PRO_MODE_STEREOPHONIC) { + snd_iprintf(buffer, "stereo\n"); + } else { + snd_iprintf(buffer, "not indicated\n"); + } + snd_iprintf(buffer, "Userbits: "); + switch (status[1] & IEC958_AES1_PRO_USERBITS) { + case IEC958_AES1_PRO_USERBITS_192: + snd_iprintf(buffer, "192bit\n"); + break; + case IEC958_AES1_PRO_USERBITS_UDEF: + snd_iprintf(buffer, "user-defined\n"); + break; + default: + snd_iprintf(buffer, "unkown\n"); + break; + } + snd_iprintf(buffer, "Sample Bits: "); + switch (status[2] & IEC958_AES2_PRO_SBITS) { + case IEC958_AES2_PRO_SBITS_20: + snd_iprintf(buffer, "20 bit\n"); + break; + case IEC958_AES2_PRO_SBITS_24: + snd_iprintf(buffer, "24 bit\n"); + break; + case IEC958_AES2_PRO_SBITS_UDEF: + snd_iprintf(buffer, "user defined\n"); + break; + default: + snd_iprintf(buffer, "unknown\n"); + break; + } + snd_iprintf(buffer, "Word Length: "); + switch (status[2] & IEC958_AES2_PRO_WORDLEN) { + case IEC958_AES2_PRO_WORDLEN_22_18: + snd_iprintf(buffer, "22 bit or 18 bit\n"); + break; + case IEC958_AES2_PRO_WORDLEN_23_19: + snd_iprintf(buffer, "23 bit or 19 bit\n"); + break; + case IEC958_AES2_PRO_WORDLEN_24_20: + snd_iprintf(buffer, "24 bit or 20 bit\n"); + break; + case IEC958_AES2_PRO_WORDLEN_20_16: + snd_iprintf(buffer, "20 bit or 16 bit\n"); + break; + default: + snd_iprintf(buffer, "unknown\n"); + break; + } + } +} + +static void snd_ca0106_proc_iec958(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + u32 value; + + value = snd_ca0106_ptr_read(emu, SAMPLE_RATE_TRACKER_STATUS, 0); + snd_iprintf(buffer, "Status: %s, %s, %s\n", + (value & 0x100000) ? "Rate Locked" : "Not Rate Locked", + (value & 0x200000) ? "SPDIF Locked" : "No SPDIF Lock", + (value & 0x400000) ? "Audio Valid" : "No valid audio" ); + snd_iprintf(buffer, "Estimated sample rate: %u\n", + ((value & 0xfffff) * 48000) / 0x8000 ); + if (value & 0x200000) { + snd_iprintf(buffer, "IEC958/SPDIF input status:\n"); + value = snd_ca0106_ptr_read(emu, SPDIF_INPUT_STATUS, 0); + snd_ca0106_proc_dump_iec958(buffer, value); + } + + snd_iprintf(buffer, "\n"); +} + +static void snd_ca0106_proc_reg_write32(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + unsigned long flags; + char line[64]; + u32 reg, val; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%x %x", ®, &val) != 2) + continue; + if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) { + spin_lock_irqsave(&emu->emu_lock, flags); + outl(val, emu->port + (reg & 0xfffffffc)); + spin_unlock_irqrestore(&emu->emu_lock, flags); + } + } +} + +static void snd_ca0106_proc_reg_read32(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + unsigned long value; + unsigned long flags; + int i; + snd_iprintf(buffer, "Registers:\n\n"); + for(i = 0; i < 0x20; i+=4) { + spin_lock_irqsave(&emu->emu_lock, flags); + value = inl(emu->port + i); + spin_unlock_irqrestore(&emu->emu_lock, flags); + snd_iprintf(buffer, "Register %02X: %08lX\n", i, value); + } +} + +static void snd_ca0106_proc_reg_read16(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + unsigned int value; + unsigned long flags; + int i; + snd_iprintf(buffer, "Registers:\n\n"); + for(i = 0; i < 0x20; i+=2) { + spin_lock_irqsave(&emu->emu_lock, flags); + value = inw(emu->port + i); + spin_unlock_irqrestore(&emu->emu_lock, flags); + snd_iprintf(buffer, "Register %02X: %04X\n", i, value); + } +} + +static void snd_ca0106_proc_reg_read8(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + unsigned int value; + unsigned long flags; + int i; + snd_iprintf(buffer, "Registers:\n\n"); + for(i = 0; i < 0x20; i+=1) { + spin_lock_irqsave(&emu->emu_lock, flags); + value = inb(emu->port + i); + spin_unlock_irqrestore(&emu->emu_lock, flags); + snd_iprintf(buffer, "Register %02X: %02X\n", i, value); + } +} + +static void snd_ca0106_proc_reg_read1(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + unsigned long value; + int i,j; + + snd_iprintf(buffer, "Registers\n"); + for(i = 0; i < 0x40; i++) { + snd_iprintf(buffer, "%02X: ",i); + for (j = 0; j < 4; j++) { + value = snd_ca0106_ptr_read(emu, i, j); + snd_iprintf(buffer, "%08lX ", value); + } + snd_iprintf(buffer, "\n"); + } +} + +static void snd_ca0106_proc_reg_read2(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + unsigned long value; + int i,j; + + snd_iprintf(buffer, "Registers\n"); + for(i = 0x40; i < 0x80; i++) { + snd_iprintf(buffer, "%02X: ",i); + for (j = 0; j < 4; j++) { + value = snd_ca0106_ptr_read(emu, i, j); + snd_iprintf(buffer, "%08lX ", value); + } + snd_iprintf(buffer, "\n"); + } +} + +static void snd_ca0106_proc_reg_write(snd_info_entry_t *entry, + snd_info_buffer_t * buffer) +{ + ca0106_t *emu = entry->private_data; + char line[64]; + unsigned int reg, channel_id , val; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%x %x %x", ®, &channel_id, &val) != 3) + continue; + if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) ) + snd_ca0106_ptr_write(emu, reg, channel_id, val); + } +} + + +int __devinit snd_ca0106_proc_init(ca0106_t * emu) +{ + snd_info_entry_t *entry; + + if(! snd_card_proc_new(emu->card, "iec958", &entry)) + snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_iec958); + if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { + snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32); + entry->c.text.write_size = 64; + entry->c.text.write = snd_ca0106_proc_reg_write32; + } + if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) + snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16); + if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry)) + snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read8); + if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { + snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1); + entry->c.text.write_size = 64; + entry->c.text.write = snd_ca0106_proc_reg_write; +// entry->private_data = emu; + } + if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) + snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read2); + return 0; +} + -- cgit v1.2.3-59-g8ed1b