diff options
Diffstat (limited to 'sound/pci/ca0106')
| -rw-r--r-- | sound/pci/ca0106/Makefile | 4 | ||||
| -rw-r--r-- | sound/pci/ca0106/ca0106.h | 742 | ||||
| -rw-r--r-- | sound/pci/ca0106/ca0106_main.c | 1971 | ||||
| -rw-r--r-- | sound/pci/ca0106/ca0106_mixer.c | 932 | ||||
| -rw-r--r-- | sound/pci/ca0106/ca0106_proc.c | 453 | ||||
| -rw-r--r-- | sound/pci/ca0106/ca_midi.c | 316 | ||||
| -rw-r--r-- | sound/pci/ca0106/ca_midi.h | 66 |
7 files changed, 4484 insertions, 0 deletions
diff --git a/sound/pci/ca0106/Makefile b/sound/pci/ca0106/Makefile new file mode 100644 index 000000000..c1455fc55 --- /dev/null +++ b/sound/pci/ca0106/Makefile @@ -0,0 +1,4 @@ +snd-ca0106-objs := ca0106_main.o ca0106_mixer.o ca_midi.o +snd-ca0106-$(CONFIG_SND_PROC_FS) += ca0106_proc.o + +obj-$(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 000000000..9847b669c --- /dev/null +++ b/sound/pci/ca0106/ca0106.h @@ -0,0 +1,742 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.22 + * + * 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. + * 0.0.21 + * Implement support for Line-in capture on SB Live 24bit. + * 0.0.22 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * + * + * This code was initially based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/************************************************************************************************/ +/* PCI function 0 registers, address = <val> + 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 = 48 Khz / 1 = 96 Khz Analog out on SB Live 24bit. + * 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 */ +/********************************************************************************************************/ + +/* Initially all registers from 0x00 to 0x3f have zero contents. */ +#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ + /* One list entry: 4 bytes for DMA address, + * 4 bytes for period_size << 16. + * One list entry is 8 bytes long. + * One list entry for each period in the buffer. + */ + /* 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 address */ + /* 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<<channel_id) */ + /* Capture (0x100<<channel_id) */ + /* Playback sample rate 96000 = 0x20000 */ + /* Start Playback [3:0] (one bit per channel) + * Start Capture [11:8] (one bit per channel) + * Playback rate [23:16] (2 bits per channel) (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * Playback mixer in enable [27:24] (one bit per channel) + * Playback mixer out enable [31:28] (one bit per channel) + */ +/* The Digital out jack is shared with the Center/LFE Analogue output. + * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 + * For Analogue: 1 -> 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 -> Shield 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_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 MIDI_UART_A_DATA 0x6c /* Midi Uart A Data */ +#define MIDI_UART_A_CMD 0x6d /* Midi Uart A Command/Status */ +#define MIDI_UART_B_DATA 0x6e /* Midi Uart B Data (currently unused) */ +#define MIDI_UART_B_CMD 0x6f /* Midi Uart B Command/Status (currently unused) */ + +/* unique channel identifier for midi->channel */ + +#define CA0106_MIDI_CHAN_A 0x1 +#define CA0106_MIDI_CHAN_B 0x2 + +/* from mpu401 */ + +#define CA0106_MIDI_INPUT_AVAIL 0x80 +#define CA0106_MIDI_OUTPUT_READY 0x40 +#define CA0106_MPU401_RESET 0xff +#define CA0106_MPU401_ENTER_UART 0x3f +#define CA0106_MPU401_ACK 0xfe + +#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */ + /* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0 + * Rate Locked [20] + * SPDIF Locked [21] For SPDIF channel only. + * Valid Audio [22] For SPDIF channel only. + */ +#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */ + /* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */ + /* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */ + /* Sample rate output control register Channel=0 + * Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) + * SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source. + * Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz) + * Record mixer output enable [12:10] + * I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * I2S output source select [18] (0=Audio from host, 1=Audio from SRC) + * Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0) + * I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.) + * I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.) + * I2S input mode [23] (0=Slave, 1=Master) + * SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz) + * SPDIF output source select [26] (0=host, 1=SRC) + * Not used [27] + * Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) + * Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM) + */ + /* Sample rate output control register Channel=1 + * I2S Input 0 volume Right [7:0] + * I2S Input 0 volume Left [15:8] + * I2S Input 1 volume Right [23:16] + * I2S Input 1 volume Left [31:24] + */ + /* Sample rate output control register Channel=2 + * SPDIF Input volume Right [23:16] + * SPDIF Input volume Left [31:24] + */ + /* Sample rate output control register Channel=3 + * No used + */ +#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */ +#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */ +#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */ +#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */ + /* Audio output control + * AC97 output enable [5:0] + * I2S output enable [19:16] + * SPDIF output enable [27:24] + */ +#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */ +#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */ +#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */ + /* Sets which Interrupts are enabled. */ + /* 0x00000001 = Half period. Playback. + * 0x00000010 = Full period. Playback. + * 0x00000100 = Half buffer. Playback. + * 0x00001000 = Full buffer. Playback. + * 0x00010000 = Half buffer. Capture. + * 0x00100000 = Full buffer. Capture. + * Capture can only do 2 periods. + * 0x01000000 = End audio. Playback. + * 0x40000000 = Half buffer Playback,Caputre xrun. + * 0x80000000 = Full buffer Playback,Caputre xrun. + */ +#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */ + /* Shows which interrupts are active at the moment. */ + /* Same bit layout as EXTENDED_INT_MASK */ +#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */ +#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */ +#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */ + /* Causes interrupts based on timer intervals. */ +#define SPI 0x7a /* SPI: Serial Interface Register */ +#define I2C_A 0x7b /* I2C Address. 32 bit */ +#define I2C_D0 0x7c /* I2C Data Port 0. 32 bit */ +#define I2C_D1 0x7d /* I2C Data Port 1. 32 bit */ +//I2C values +#define I2C_A_ADC_ADD_MASK 0x000000fe //The address is a 7 bit address +#define I2C_A_ADC_RW_MASK 0x00000001 //bit mask for R/W +#define I2C_A_ADC_TRANS_MASK 0x00000010 //Bit mask for I2c address DAC value +#define I2C_A_ADC_ABORT_MASK 0x00000020 //Bit mask for I2C transaction abort flag +#define I2C_A_ADC_LAST_MASK 0x00000040 //Bit mask for Last word transaction +#define I2C_A_ADC_BYTE_MASK 0x00000080 //Bit mask for Byte Mode + +#define I2C_A_ADC_ADD 0x00000034 //This is the Device address for ADC +#define I2C_A_ADC_READ 0x00000001 //To perform a read operation +#define I2C_A_ADC_START 0x00000100 //Start I2C transaction +#define I2C_A_ADC_ABORT 0x00000200 //I2C transaction abort +#define I2C_A_ADC_LAST 0x00000400 //I2C last transaction +#define I2C_A_ADC_BYTE 0x00000800 //I2C one byte mode + +#define I2C_D_ADC_REG_MASK 0xfe000000 //ADC address register +#define I2C_D_ADC_DAT_MASK 0x01ff0000 //ADC data register + +#define ADC_TIMEOUT 0x00000007 //ADC Timeout Clock Disable +#define ADC_IFC_CTRL 0x0000000b //ADC Interface Control +#define ADC_MASTER 0x0000000c //ADC Master Mode Control +#define ADC_POWER 0x0000000d //ADC PowerDown Control +#define ADC_ATTEN_ADCL 0x0000000e //ADC Attenuation ADCL +#define ADC_ATTEN_ADCR 0x0000000f //ADC Attenuation ADCR +#define ADC_ALC_CTRL1 0x00000010 //ADC ALC Control 1 +#define ADC_ALC_CTRL2 0x00000011 //ADC ALC Control 2 +#define ADC_ALC_CTRL3 0x00000012 //ADC ALC Control 3 +#define ADC_NOISE_CTRL 0x00000013 //ADC Noise Gate Control +#define ADC_LIMIT_CTRL 0x00000014 //ADC Limiter Control +#define ADC_MUX 0x00000015 //ADC Mux offset + +#if 0 +/* FIXME: Not tested yet. */ +#define ADC_GAIN_MASK 0x000000ff //Mask for ADC Gain +#define ADC_ZERODB 0x000000cf //Value to set ADC to 0dB +#define ADC_MUTE_MASK 0x000000c0 //Mask for ADC mute +#define ADC_MUTE 0x000000c0 //Value to mute ADC +#define ADC_OSR 0x00000008 //Mask for ADC oversample rate select +#define ADC_TIMEOUT_DISABLE 0x00000008 //Value and mask to disable Timeout clock +#define ADC_HPF_DISABLE 0x00000100 //Value and mask to disable High pass filter +#define ADC_TRANWIN_MASK 0x00000070 //Mask for Length of Transient Window +#endif + +#define ADC_MUX_MASK 0x0000000f //Mask for ADC Mux +#define ADC_MUX_PHONE 0x00000001 //Value to select TAD at ADC Mux (Not used) +#define ADC_MUX_MIC 0x00000002 //Value to select Mic at ADC Mux +#define ADC_MUX_LINEIN 0x00000004 //Value to select LineIn at ADC Mux +#define ADC_MUX_AUX 0x00000008 //Value to select Aux at ADC Mux + +#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */ +#define PCM_FRONT_CHANNEL 0 +#define PCM_REAR_CHANNEL 1 +#define PCM_CENTER_LFE_CHANNEL 2 +#define PCM_UNKNOWN_CHANNEL 3 +#define CONTROL_FRONT_CHANNEL 0 +#define CONTROL_REAR_CHANNEL 3 +#define CONTROL_CENTER_LFE_CHANNEL 1 +#define CONTROL_UNKNOWN_CHANNEL 2 + + +/* Based on WM8768 Datasheet Rev 4.2 page 32 */ +#define SPI_REG_MASK 0x1ff /* 16-bit SPI writes have a 7-bit address */ +#define SPI_REG_SHIFT 9 /* followed by 9 bits of data */ + +#define SPI_LDA1_REG 0 /* digital attenuation */ +#define SPI_RDA1_REG 1 +#define SPI_LDA2_REG 4 +#define SPI_RDA2_REG 5 +#define SPI_LDA3_REG 6 +#define SPI_RDA3_REG 7 +#define SPI_LDA4_REG 13 +#define SPI_RDA4_REG 14 +#define SPI_MASTDA_REG 8 + +#define SPI_DA_BIT_UPDATE (1<<8) /* update attenuation values */ +#define SPI_DA_BIT_0dB 0xff /* 0 dB */ +#define SPI_DA_BIT_infdB 0x00 /* inf dB attenuation (mute) */ + +#define SPI_PL_REG 2 +#define SPI_PL_BIT_L_M (0<<5) /* left channel = mute */ +#define SPI_PL_BIT_L_L (1<<5) /* left channel = left */ +#define SPI_PL_BIT_L_R (2<<5) /* left channel = right */ +#define SPI_PL_BIT_L_C (3<<5) /* left channel = (L+R)/2 */ +#define SPI_PL_BIT_R_M (0<<7) /* right channel = mute */ +#define SPI_PL_BIT_R_L (1<<7) /* right channel = left */ +#define SPI_PL_BIT_R_R (2<<7) /* right channel = right */ +#define SPI_PL_BIT_R_C (3<<7) /* right channel = (L+R)/2 */ +#define SPI_IZD_REG 2 +#define SPI_IZD_BIT (0<<4) /* infinite zero detect */ + +#define SPI_FMT_REG 3 +#define SPI_FMT_BIT_RJ (0<<0) /* right justified mode */ +#define SPI_FMT_BIT_LJ (1<<0) /* left justified mode */ +#define SPI_FMT_BIT_I2S (2<<0) /* I2S mode */ +#define SPI_FMT_BIT_DSP (3<<0) /* DSP Modes A or B */ +#define SPI_LRP_REG 3 +#define SPI_LRP_BIT (1<<2) /* invert LRCLK polarity */ +#define SPI_BCP_REG 3 +#define SPI_BCP_BIT (1<<3) /* invert BCLK polarity */ +#define SPI_IWL_REG 3 +#define SPI_IWL_BIT_16 (0<<4) /* 16-bit world length */ +#define SPI_IWL_BIT_20 (1<<4) /* 20-bit world length */ +#define SPI_IWL_BIT_24 (2<<4) /* 24-bit world length */ +#define SPI_IWL_BIT_32 (3<<4) /* 32-bit world length */ + +#define SPI_MS_REG 10 +#define SPI_MS_BIT (1<<5) /* master mode */ +#define SPI_RATE_REG 10 /* only applies in master mode */ +#define SPI_RATE_BIT_128 (0<<6) /* MCLK = LRCLK * 128 */ +#define SPI_RATE_BIT_192 (1<<6) +#define SPI_RATE_BIT_256 (2<<6) +#define SPI_RATE_BIT_384 (3<<6) +#define SPI_RATE_BIT_512 (4<<6) +#define SPI_RATE_BIT_768 (5<<6) + +/* They really do label the bit for the 4th channel "4" and not "3" */ +#define SPI_DMUTE0_REG 9 +#define SPI_DMUTE1_REG 9 +#define SPI_DMUTE2_REG 9 +#define SPI_DMUTE4_REG 15 +#define SPI_DMUTE0_BIT (1<<3) +#define SPI_DMUTE1_BIT (1<<4) +#define SPI_DMUTE2_BIT (1<<5) +#define SPI_DMUTE4_BIT (1<<2) + +#define SPI_PHASE0_REG 3 +#define SPI_PHASE1_REG 3 +#define SPI_PHASE2_REG 3 +#define SPI_PHASE4_REG 15 +#define SPI_PHASE0_BIT (1<<6) +#define SPI_PHASE1_BIT (1<<7) +#define SPI_PHASE2_BIT (1<<8) +#define SPI_PHASE4_BIT (1<<3) + +#define SPI_PDWN_REG 2 /* power down all DACs */ +#define SPI_PDWN_BIT (1<<2) +#define SPI_DACD0_REG 10 /* power down individual DACs */ +#define SPI_DACD1_REG 10 +#define SPI_DACD2_REG 10 +#define SPI_DACD4_REG 15 +#define SPI_DACD0_BIT (1<<1) +#define SPI_DACD1_BIT (1<<2) +#define SPI_DACD2_BIT (1<<3) +#define SPI_DACD4_BIT (1<<0) /* datasheet error says it's 1 */ + +#define SPI_PWRDNALL_REG 10 /* power down everything */ +#define SPI_PWRDNALL_BIT (1<<4) + +#include "ca_midi.h" + +struct snd_ca0106; + +struct snd_ca0106_channel { + struct snd_ca0106 *emu; + int number; + int use; + void (*interrupt)(struct snd_ca0106 *emu, struct snd_ca0106_channel *channel); + struct snd_ca0106_pcm *epcm; +}; + +struct snd_ca0106_pcm { + struct snd_ca0106 *emu; + struct snd_pcm_substream *substream; + int channel_id; + unsigned short running; +}; + +struct snd_ca0106_details { + u32 serial; + char * name; + int ac97; /* ac97 = 0 -> Select MIC, Line in, TAD in, AUX in. + ac97 = 1 -> Default to AC97 in. */ + int gpio_type; /* gpio_type = 1 -> shared mic-in/line-in + gpio_type = 2 -> shared side-out/line-in. */ + int i2c_adc; /* with i2c_adc=1, the driver adds some capture volume + controls, phone, mic, line-in and aux. */ + u16 spi_dac; /* spi_dac = 0 -> no spi interface for DACs + spi_dac = 0x<front><rear><center-lfe><side> + -> specifies DAC id for each channel pair. */ +}; + +// definition of the chip-specific record +struct snd_ca0106 { + struct snd_card *card; + struct snd_ca0106_details *details; + struct pci_dev *pci; + + unsigned long port; + struct resource *res_port; + int irq; + + unsigned int serial; /* serial number */ + unsigned short model; /* subsystem id */ + + spinlock_t emu_lock; + + struct snd_ac97 *ac97; + struct snd_pcm *pcm[4]; + + struct snd_ca0106_channel playback_channels[4]; + struct snd_ca0106_channel capture_channels[4]; + u32 spdif_bits[4]; /* s/pdif out default setup */ + u32 spdif_str_bits[4]; /* s/pdif out per-stream setup */ + int spdif_enable; + int capture_source; + int i2c_capture_source; + u8 i2c_capture_volume[4][2]; + int capture_mic_line_in; + + struct snd_dma_buffer buffer; + + struct snd_ca_midi midi; + struct snd_ca_midi midi2; + + u16 spi_dac_reg[16]; + +#ifdef CONFIG_PM_SLEEP +#define NUM_SAVED_VOLUMES 9 + unsigned int saved_vol[NUM_SAVED_VOLUMES]; +#endif +}; + +int snd_ca0106_mixer(struct snd_ca0106 *emu); +int snd_ca0106_proc_init(struct snd_ca0106 * emu); + +unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * emu, + unsigned int reg, + unsigned int chn); + +void snd_ca0106_ptr_write(struct snd_ca0106 *emu, + unsigned int reg, + unsigned int chn, + unsigned int data); + +int snd_ca0106_i2c_write(struct snd_ca0106 *emu, u32 reg, u32 value); + +int snd_ca0106_spi_write(struct snd_ca0106 * emu, + unsigned int data); + +#ifdef CONFIG_PM_SLEEP +void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip); +void snd_ca0106_mixer_resume(struct snd_ca0106 *chip); +#else +#define snd_ca0106_mixer_suspend(chip) do { } while (0) +#define snd_ca0106_mixer_resume(chip) do { } while (0) +#endif diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c new file mode 100644 index 000000000..675b812e9 --- /dev/null +++ b/sound/pci/ca0106/ca0106_main.c @@ -0,0 +1,1971 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.25 + * + * 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 + * 0.0.23 + * Implement support for Line-in capture on SB Live 24bit. + * 0.0.24 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * 0.0.25 + * Powerdown SPI DAC channels when not in use + * + * 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 initially based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/info.h> + +MODULE_AUTHOR("James Courtier-Dutton <James@superbug.demon.co.uk>"); +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 bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static uint subsystem[SNDRV_CARDS]; /* Force card subsystem model */ + +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."); +module_param_array(subsystem, uint, NULL, 0444); +MODULE_PARM_DESC(subsystem, "Force card subsystem model."); + +#include "ca0106.h" + +static struct snd_ca0106_details ca0106_chip_details[] = { + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* Tested: + * See ALSA bug#3251 + */ + { .serial = 0x10131102, + .name = "X-Fi Extreme Audio [SBxxxx]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* + * CTRL:CA0111-WTLF + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + /* Tested: + * Playback on front, rear, center/lfe speakers + * Capture from Mic in. + * Not-Tested: + * Capture from Line in. + * Playback to digital out. + */ + { .serial = 0x10121102, + .name = "X-Fi Extreme Audio [SB0790]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ + /* AudigyLS[SB0310] */ + { .serial = 0x10021102, + .name = "AudigyLS [SB0310]", + .ac97 = 1 } , + /* Unknown AudigyLS that also says SB0310 on it */ + { .serial = 0x10051102, + .name = "AudigyLS [SB0310b]", + .ac97 = 1 } , + /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */ + { .serial = 0x10061102, + .name = "Live! 7.1 24bit [SB0410]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ + { .serial = 0x10071102, + .name = "Live! 7.1 24bit [SB0413]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Audigy SE. Has a different DAC. */ + /* SB0570: + * CTRL:CA0106-DAT + * ADC: WM8775EDS + * DAC: WM8768GEDS + */ + { .serial = 0x100a1102, + .name = "Audigy SE [SB0570]", + .gpio_type = 1, + .i2c_adc = 1, + .spi_dac = 0x4021 } , + /* New Audigy LS. Has a different DAC. */ + /* SB0570: + * CTRL:CA0106-DAT + * ADC: WM8775EDS + * DAC: WM8768GEDS + */ + { .serial = 0x10111102, + .name = "Audigy SE OEM [SB0570a]", + .gpio_type = 1, + .i2c_adc = 1, + .spi_dac = 0x4021 } , + /* Sound Blaster 5.1vx + * Tested: Playback on front, rear, center/lfe speakers + * Not-Tested: Capture + */ + { .serial = 0x10041102, + .name = "Sound Blaster 5.1vx [SB1070]", + .gpio_type = 1, + .i2c_adc = 0, + .spi_dac = 0x0124 + } , + /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ + /* SB0438 + * CTRL:CA0106-DAT + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + { .serial = 0x10091462, + .name = "MSI K8N Diamond MB [SB0438]", + .gpio_type = 2, + .i2c_adc = 1 } , + /* MSI K8N Diamond PLUS MB */ + { .serial = 0x10091102, + .name = "MSI K8N Diamond MB", + .gpio_type = 2, + .i2c_adc = 1, + .spi_dac = 0x4021 } , + /* Giga-byte GA-G1975X mobo + * Novell bnc#395807 + */ + /* FIXME: the GPIO and I2C setting aren't tested well */ + { .serial = 0x1458a006, + .name = "Giga-byte GA-G1975X", + .gpio_type = 1, + .i2c_adc = 1 }, + /* Shuttle XPC SD31P which has an onboard Creative Labs + * Sound Blaster Live! 24-bit EAX + * high-definition 7.1 audio processor". + * Added using info from andrewvegan in alsa bug #1298 + */ + { .serial = 0x30381297, + .name = "Shuttle XPC SD31P [SD31P]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* Shuttle XPC SD11G5 which has an onboard Creative Labs + * Sound Blaster Live! 24-bit EAX + * high-definition 7.1 audio processor". + * Fixes ALSA bug#1600 + */ + { .serial = 0x30411297, + .name = "Shuttle XPC SD11G5 [SD11G5]", + .gpio_type = 1, + .i2c_adc = 1 } , + { .serial = 0, + .name = "AudigyLS [Unknown]" } +}; + +/* hardware definition */ +static const struct snd_pcm_hardware snd_ca0106_playback_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, + .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 const struct snd_pcm_hardware 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 | SNDRV_PCM_FMTBIT_S32_LE, +#if 0 /* FIXME: looks like 44.1kHz capture causes noisy output on 48kHz */ + .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 44100, +#else + .rates = (SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000), + .rate_min = 48000, +#endif /* FIXME */ + .rate_max = 192000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 65536 - 128, + .period_bytes_min = 64, + .period_bytes_max = 32768 - 64, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 0, +}; + +unsigned int snd_ca0106_ptr_read(struct snd_ca0106 * 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(struct snd_ca0106 *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); +} + +int snd_ca0106_spi_write(struct snd_ca0106 * emu, + unsigned int data) +{ + unsigned int reset, set; + unsigned int reg, tmp; + int n, result; + reg = SPI; + if (data > 0xffff) /* Only 16bit values allowed */ + return 1; + tmp = snd_ca0106_ptr_read(emu, reg, 0); + reset = (tmp & ~0x3ffff) | 0x20000; /* Set xxx20000 */ + set = reset | 0x10000; /* Set xxx1xxxx */ + snd_ca0106_ptr_write(emu, reg, 0, reset | data); + tmp = snd_ca0106_ptr_read(emu, reg, 0); /* write post */ + snd_ca0106_ptr_write(emu, reg, 0, set | data); + result = 1; + /* Wait for status bit to return to 0 */ + for (n = 0; n < 100; n++) { + udelay(10); + tmp = snd_ca0106_ptr_read(emu, reg, 0); + if (!(tmp & 0x10000)) { + result = 0; + break; + } + } + if (result) /* Timed out */ + return 1; + snd_ca0106_ptr_write(emu, reg, 0, reset | data); + tmp = snd_ca0106_ptr_read(emu, reg, 0); /* Write post */ + return 0; +} + +/* The ADC does not support i2c read, so only write is implemented */ +int snd_ca0106_i2c_write(struct snd_ca0106 *emu, + u32 reg, + u32 value) +{ + u32 tmp; + int timeout = 0; + int status; + int retry; + if ((reg > 0x7f) || (value > 0x1ff)) { + dev_err(emu->card->dev, "i2c_write: invalid values.\n"); + return -EINVAL; + } + + tmp = reg << 25 | value << 16; + /* + dev_dbg(emu->card->dev, "I2C-write:reg=0x%x, value=0x%x\n", reg, value); + */ + /* Not sure what this I2C channel controls. */ + /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ + + /* This controls the I2C connected to the WM8775 ADC Codec */ + snd_ca0106_ptr_write(emu, I2C_D1, 0, tmp); + + for (retry = 0; retry < 10; retry++) { + /* Send the data to i2c */ + //tmp = snd_ca0106_ptr_read(emu, I2C_A, 0); + //tmp = tmp & ~(I2C_A_ADC_READ|I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD_MASK); + tmp = 0; + tmp = tmp | (I2C_A_ADC_LAST|I2C_A_ADC_START|I2C_A_ADC_ADD); + snd_ca0106_ptr_write(emu, I2C_A, 0, tmp); + + /* Wait till the transaction ends */ + while (1) { + status = snd_ca0106_ptr_read(emu, I2C_A, 0); + /*dev_dbg(emu->card->dev, "I2C:status=0x%x\n", status);*/ + timeout++; + if ((status & I2C_A_ADC_START) == 0) + break; + + if (timeout > 1000) + break; + } + //Read back and see if the transaction is successful + if ((status & I2C_A_ADC_ABORT) == 0) + break; + } + + if (retry == 10) { + dev_err(emu->card->dev, "Writing to ADC failed!\n"); + return -EINVAL; + } + + return 0; +} + + +static void snd_ca0106_intr_enable(struct snd_ca0106 *emu, unsigned int intrenb) +{ + unsigned long flags; + unsigned int intr_enable; + + spin_lock_irqsave(&emu->emu_lock, flags); + intr_enable = inl(emu->port + INTE) | intrenb; + outl(intr_enable, emu->port + INTE); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + +static void snd_ca0106_intr_disable(struct snd_ca0106 *emu, unsigned int intrenb) +{ + unsigned long flags; + unsigned int intr_enable; + + spin_lock_irqsave(&emu->emu_lock, flags); + intr_enable = inl(emu->port + INTE) & ~intrenb; + outl(intr_enable, emu->port + INTE); + spin_unlock_irqrestore(&emu->emu_lock, flags); +} + + +static void snd_ca0106_pcm_free_substream(struct snd_pcm_runtime *runtime) +{ + kfree(runtime->private_data); +} + +static const int spi_dacd_reg[] = { + SPI_DACD0_REG, + SPI_DACD1_REG, + SPI_DACD2_REG, + 0, + SPI_DACD4_REG, +}; +static const int spi_dacd_bit[] = { + SPI_DACD0_BIT, + SPI_DACD1_BIT, + SPI_DACD2_BIT, + 0, + SPI_DACD4_BIT, +}; + +static void restore_spdif_bits(struct snd_ca0106 *chip, int idx) +{ + if (chip->spdif_str_bits[idx] != chip->spdif_bits[idx]) { + chip->spdif_str_bits[idx] = chip->spdif_bits[idx]; + snd_ca0106_ptr_write(chip, SPCS0 + idx, 0, + chip->spdif_str_bits[idx]); + } +} + +static int snd_ca0106_channel_dac(struct snd_ca0106 *chip, + struct snd_ca0106_details *details, + int channel_id) +{ + switch (channel_id) { + case PCM_FRONT_CHANNEL: + return (details->spi_dac & 0xf000) >> (4 * 3); + case PCM_REAR_CHANNEL: + return (details->spi_dac & 0x0f00) >> (4 * 2); + case PCM_CENTER_LFE_CHANNEL: + return (details->spi_dac & 0x00f0) >> (4 * 1); + case PCM_UNKNOWN_CHANNEL: + return (details->spi_dac & 0x000f) >> (4 * 0); + default: + dev_dbg(chip->card->dev, "ca0106: unknown channel_id %d\n", + channel_id); + } + return 0; +} + +static int snd_ca0106_pcm_power_dac(struct snd_ca0106 *chip, int channel_id, + int power) +{ + if (chip->details->spi_dac) { + const int dac = snd_ca0106_channel_dac(chip, chip->details, + channel_id); + const int reg = spi_dacd_reg[dac]; + const int bit = spi_dacd_bit[dac]; + + if (power) + /* Power up */ + chip->spi_dac_reg[reg] &= ~bit; + else + /* Power down */ + chip->spi_dac_reg[reg] |= bit; + if (snd_ca0106_spi_write(chip, chip->spi_dac_reg[reg]) != 0) + return -ENXIO; + } + return 0; +} + +/* open_playback callback */ +static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substream, + int channel_id) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_ca0106_channel *channel = &(chip->playback_channels[channel_id]); + struct snd_ca0106_pcm *epcm; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + epcm = kzalloc(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; + /* + dev_dbg(chip->card->dev, "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; + snd_pcm_set_sync(substream); + + /* Front channel dac should already be on */ + if (channel_id != PCM_FRONT_CHANNEL) { + err = snd_ca0106_pcm_power_dac(chip, channel_id, 1); + if (err < 0) + return err; + } + + restore_spdif_bits(chip, channel_id); + + return 0; +} + +/* close callback */ +static int snd_ca0106_pcm_close_playback(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + chip->playback_channels[epcm->channel_id].use = 0; + + restore_spdif_bits(chip, epcm->channel_id); + + /* Front channel dac should stay on */ + if (epcm->channel_id != PCM_FRONT_CHANNEL) { + int err; + err = snd_ca0106_pcm_power_dac(chip, epcm->channel_id, 0); + if (err < 0) + return err; + } + + /* FIXME: maybe zero others */ + return 0; +} + +static int snd_ca0106_pcm_open_playback_front(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_center_lfe(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_unknown(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL); +} + +static int snd_ca0106_pcm_open_playback_rear(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL); +} + +/* open_capture callback */ +static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substream, + int channel_id) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_ca0106_channel *channel = &(chip->capture_channels[channel_id]); + struct snd_ca0106_pcm *epcm; + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); + if (!epcm) + 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; + /* + dev_dbg(chip->card->dev, "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(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *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(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 0); +} + +static int snd_ca0106_pcm_open_1_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 1); +} + +static int snd_ca0106_pcm_open_2_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 2); +} + +static int snd_ca0106_pcm_open_3_capture(struct snd_pcm_substream *substream) +{ + return snd_ca0106_pcm_open_capture_channel(substream, 3); +} + +/* hw_params callback */ +static int snd_ca0106_pcm_hw_params_playback(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *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(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* hw_params callback */ +static int snd_ca0106_pcm_hw_params_capture(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *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(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* prepare playback callback */ +static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *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; + +#if 0 /* debug */ + dev_dbg(emu->card->dev, + "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)); + dev_dbg(emu->card->dev, + "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + dev_dbg(emu->card->dev, + "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ + /* 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(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + int channel = epcm->channel_id; + u32 hcfg_mask = HCFG_CAPTURE_S32_LE; + u32 hcfg_set = 0x00000000; + u32 hcfg; + u32 over_sampling=0x2; + u32 reg71_mask = 0x0000c000 ; /* Global. Set ADC rate. */ + u32 reg71_set = 0; + u32 reg71; + +#if 0 /* debug */ + dev_dbg(emu->card->dev, + "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)); + dev_dbg(emu->card->dev, + "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + dev_dbg(emu->card->dev, + "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ + /* reg71 controls ADC rate. */ + switch (runtime->rate) { + case 44100: + reg71_set = 0x00004000; + break; + case 48000: + reg71_set = 0; + break; + case 96000: + reg71_set = 0x00008000; + over_sampling=0xa; + break; + case 192000: + reg71_set = 0x0000c000; + over_sampling=0xa; + break; + default: + 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_CAPTURE_S32_LE; + break; + default: + hcfg_set = 0; + break; + } + hcfg = inl(emu->port + HCFG) ; + hcfg = (hcfg & ~hcfg_mask) | hcfg_set; + outl(hcfg, emu->port + HCFG); + reg71 = snd_ca0106_ptr_read(emu, 0x71, 0); + reg71 = (reg71 & ~reg71_mask) | reg71_set; + snd_ca0106_ptr_write(emu, 0x71, 0, reg71); + if (emu->details->i2c_adc == 1) { /* The SB0410 and SB0413 use I2C to control ADC. */ + snd_ca0106_i2c_write(emu, ADC_MASTER, over_sampling); /* Adjust the over sampler to better suit the capture rate. */ + } + + + /* + dev_dbg(emu->card->dev, + "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(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime; + struct snd_ca0106_pcm *epcm; + int channel; + int result = 0; + struct snd_pcm_substream *s; + u32 basic = 0; + u32 extended = 0; + u32 bits; + int running = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + running = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + default: + running = 0; + break; + } + snd_pcm_group_for_each_entry(s, substream) { + if (snd_pcm_substream_chip(s) != emu || + s->stream != SNDRV_PCM_STREAM_PLAYBACK) + continue; + runtime = s->runtime; + epcm = runtime->private_data; + channel = epcm->channel_id; + /* dev_dbg(emu->card->dev, "channel=%d\n", channel); */ + epcm->running = running; + basic |= (0x1 << channel); + extended |= (0x10 << channel); + snd_pcm_trigger_done(s, substream); + } + /* dev_dbg(emu->card->dev, "basic=0x%x, extended=0x%x\n",basic, extended); */ + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0); + bits |= extended; + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits); + bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0); + bits |= basic; + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + bits = snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0); + bits &= ~basic; + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, bits); + bits = snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0); + bits &= ~extended; + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, bits); + break; + default: + result = -EINVAL; + break; + } + return result; +} + +/* trigger_capture callback */ +static int snd_ca0106_pcm_trigger_capture(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *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<<channel)); + snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(0x100<<channel)); + epcm->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<<channel)); + snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(0x110000<<channel)); + epcm->running = 0; + break; + default: + result = -EINVAL; + break; + } + return result; +} + +/* pointer_playback callback */ +static snd_pcm_uframes_t +snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + unsigned int ptr, prev_ptr; + int channel = epcm->channel_id; + int timeout = 10; + + if (!epcm->running) + return 0; + + prev_ptr = -1; + do { + ptr = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel); + ptr = (ptr >> 3) * runtime->period_size; + ptr += bytes_to_frames(runtime, + snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel)); + if (ptr >= runtime->buffer_size) + ptr -= runtime->buffer_size; + if (prev_ptr == ptr) + return ptr; + prev_ptr = ptr; + } while (--timeout); + dev_warn(emu->card->dev, "ca0106: unstable DMA pointer!\n"); + return 0; +} + +/* pointer_capture callback */ +static snd_pcm_uframes_t +snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) +{ + struct snd_ca0106 *emu = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_ca0106_pcm *epcm = runtime->private_data; + snd_pcm_uframes_t ptr, ptr1, ptr2 = 0; + int 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; + /* + dev_dbg(emu->card->dev, "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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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 const struct snd_pcm_ops 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(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct snd_ca0106 *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(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + struct snd_ca0106 *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(struct snd_ca0106 *chip) +{ + struct snd_ac97_bus *pbus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops 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; + ac97.scaps = AC97_SCAP_NO_SPDIF; + return snd_ac97_mixer(pbus, &ac97, &chip->ac97); +} + +static void ca0106_stop_chip(struct snd_ca0106 *chip); + +static int snd_ca0106_free(struct snd_ca0106 *chip) +{ + if (chip->res_port != NULL) { + /* avoid access to already used hardware */ + ca0106_stop_chip(chip); + } + if (chip->irq >= 0) + free_irq(chip->irq, chip); + // release the data +#if 1 + if (chip->buffer.area) + snd_dma_free_pages(&chip->buffer); +#endif + + // release the i/o port + release_and_free_resource(chip->res_port); + + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_ca0106_dev_free(struct snd_device *device) +{ + struct snd_ca0106 *chip = device->device_data; + return snd_ca0106_free(chip); +} + +static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) +{ + unsigned int status; + + struct snd_ca0106 *chip = dev_id; + int i; + int mask; + unsigned int stat76; + struct snd_ca0106_channel *pchannel; + + status = inl(chip->port + IPR); + if (! status) + return IRQ_NONE; + + stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); + /* + dev_dbg(emu->card->dev, "interrupt status = 0x%08x, stat76=0x%08x\n", + status, stat76); + dev_dbg(emu->card->dev, "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); + /* dev_dbg(emu->card->dev, "interrupt [%d] used\n", i); */ + } + } + /* + dev_dbg(emu->card->dev, "channel=%p\n", pchannel); + dev_dbg(emu->card->dev, "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); + /* dev_dbg(emu->card->dev, "interrupt [%d] used\n", i); */ + } + } + /* + dev_dbg(emu->card->dev, "channel=%p\n", pchannel); + dev_dbg(emu->card->dev, "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); + + if (chip->midi.dev_id && + (status & (chip->midi.ipr_tx|chip->midi.ipr_rx))) { + if (chip->midi.interrupt) + chip->midi.interrupt(&chip->midi, status); + else + chip->midi.interrupt_disable(&chip->midi, chip->midi.tx_enable | chip->midi.rx_enable); + } + + // acknowledge the interrupt if necessary + outl(status, chip->port+IPR); + + return IRQ_HANDLED; +} + +static const struct snd_pcm_chmap_elem surround_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_RL, SNDRV_CHMAP_RR } }, + { } +}; + +static const struct snd_pcm_chmap_elem clfe_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_FC, SNDRV_CHMAP_LFE } }, + { } +}; + +static const struct snd_pcm_chmap_elem side_map[] = { + { .channels = 2, + .map = { SNDRV_CHMAP_SL, SNDRV_CHMAP_SR } }, + { } +}; + +static int snd_ca0106_pcm(struct snd_ca0106 *emu, int device) +{ + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + const struct snd_pcm_chmap_elem *map = NULL; + int err; + + err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm); + if (err < 0) + return err; + + pcm->private_data = emu; + + 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); + map = snd_pcm_std_chmaps; + 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); + map = surround_map; + 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); + map = clfe_map; + 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); + map = side_map; + break; + } + + pcm->info_flags = 0; + strcpy(pcm->name, "CA0106"); + + 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; + } + + err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK, map, 2, + 1 << 2, NULL); + if (err < 0) + return err; + + emu->pcm[device] = pcm; + + return 0; +} + +#define SPI_REG(reg, value) (((reg) << SPI_REG_SHIFT) | (value)) +static unsigned int spi_dac_init[] = { + SPI_REG(SPI_LDA1_REG, SPI_DA_BIT_0dB), /* 0dB dig. attenuation */ + SPI_REG(SPI_RDA1_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_PL_REG, SPI_PL_BIT_L_L | SPI_PL_BIT_R_R | SPI_IZD_BIT), + SPI_REG(SPI_FMT_REG, SPI_FMT_BIT_I2S | SPI_IWL_BIT_24), + SPI_REG(SPI_LDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA2_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_LDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA3_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_MASTDA_REG, SPI_DA_BIT_0dB), + SPI_REG(9, 0x00), + SPI_REG(SPI_MS_REG, SPI_DACD0_BIT | SPI_DACD1_BIT | SPI_DACD2_BIT), + SPI_REG(12, 0x00), + SPI_REG(SPI_LDA4_REG, SPI_DA_BIT_0dB), + SPI_REG(SPI_RDA4_REG, SPI_DA_BIT_0dB | SPI_DA_BIT_UPDATE), + SPI_REG(SPI_DACD4_REG, SPI_DACD4_BIT), +}; + +static unsigned int i2c_adc_init[][2] = { + { 0x17, 0x00 }, /* Reset */ + { 0x07, 0x00 }, /* Timeout */ + { 0x0b, 0x22 }, /* Interface control */ + { 0x0c, 0x22 }, /* Master mode control */ + { 0x0d, 0x08 }, /* Powerdown control */ + { 0x0e, 0xcf }, /* Attenuation Left 0x01 = -103dB, 0xff = 24dB */ + { 0x0f, 0xcf }, /* Attenuation Right 0.5dB steps */ + { 0x10, 0x7b }, /* ALC Control 1 */ + { 0x11, 0x00 }, /* ALC Control 2 */ + { 0x12, 0x32 }, /* ALC Control 3 */ + { 0x13, 0x00 }, /* Noise gate control */ + { 0x14, 0xa6 }, /* Limiter control */ + { 0x15, ADC_MUX_LINEIN }, /* ADC Mixer control */ +}; + +static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) +{ + int ch; + unsigned int def_bits; + + 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) + */ + def_bits = + SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 | + SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC | + SPCS_GENERATIONSTATUS | 0x00001200 | + 0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT; + if (!resume) { + chip->spdif_str_bits[0] = chip->spdif_bits[0] = def_bits; + chip->spdif_str_bits[1] = chip->spdif_bits[1] = def_bits; + chip->spdif_str_bits[2] = chip->spdif_bits[2] = def_bits; + chip->spdif_str_bits[3] = chip->spdif_bits[3] = def_bits; + } + /* Only SPCS1 has been tested */ + snd_ca0106_ptr_write(chip, SPCS1, 0, chip->spdif_str_bits[1]); + snd_ca0106_ptr_write(chip, SPCS0, 0, chip->spdif_str_bits[0]); + snd_ca0106_ptr_write(chip, SPCS2, 0, chip->spdif_str_bits[2]); + snd_ca0106_ptr_write(chip, SPCS3, 0, chip->spdif_str_bits[3]); + + 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 /* FIXME: what are these? */ + 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 + + /* OSS drivers set this. */ + /* snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); */ + + /* Analog or Digital output */ + snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf); + /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers. + * Use 0x000f0000 for surround71 + */ + snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000f0000); + + chip->spdif_enable = 0; /* Set digital SPDIF output off */ + /*snd_ca0106_ptr_write(chip, 0x45, 0, 0);*/ /* Analogue out */ + /*snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00);*/ /* Digital out */ + + /* goes to 0x40c80000 when doing SPDIF IN/OUT */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); + /* (Mute) CAPTURE feedback into PLAYBACK volume. + * Only lower 16 bits matter. + */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); + /* SPDIF IN Volume */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); + /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */ + snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); + + 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++) { + /* Only high 16 bits matter */ + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); + snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030); +#if 0 /* Mute */ + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); + snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); +#endif + } + if (chip->details->i2c_adc == 1) { + /* Select MIC, Line in, TAD in, AUX in */ + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); + /* Default to CAPTURE_SOURCE to i2s in */ + if (!resume) + chip->capture_source = 3; + } else if (chip->details->ac97 == 1) { + /* Default to AC97 in */ + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x444400e4); + /* Default to CAPTURE_SOURCE to AC97 in */ + if (!resume) + chip->capture_source = 4; + } else { + /* Select MIC, Line in, TAD in, AUX in */ + snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); + /* Default to Set CAPTURE_SOURCE to i2s in */ + if (!resume) + chip->capture_source = 3; + } + + if (chip->details->gpio_type == 2) { + /* The SB0438 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(0x005f5301, chip->port+GPIO); /* Analog */ + } else if (chip->details->gpio_type == 1) { + /* 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(0x005f5301, 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); */ + /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */ + /* outl(0x00001409, chip->port+HCFG); */ + /* outl(0x00000009, chip->port+HCFG); */ + /* AC97 2.0, Enable outputs. */ + outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); + + if (chip->details->i2c_adc == 1) { + /* The SB0410 and SB0413 use I2C to control ADC. */ + int size, n; + + size = ARRAY_SIZE(i2c_adc_init); + /* dev_dbg(emu->card->dev, "I2C:array size=0x%x\n", size); */ + for (n = 0; n < size; n++) + snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], + i2c_adc_init[n][1]); + for (n = 0; n < 4; n++) { + chip->i2c_capture_volume[n][0] = 0xcf; + chip->i2c_capture_volume[n][1] = 0xcf; + } + chip->i2c_capture_source = 2; /* Line in */ + /* Enable Line-in capture. MIC in currently untested. */ + /* snd_ca0106_i2c_write(chip, ADC_MUX, ADC_MUX_LINEIN); */ + } + + if (chip->details->spi_dac) { + /* The SB0570 use SPI to control DAC. */ + int size, n; + + size = ARRAY_SIZE(spi_dac_init); + for (n = 0; n < size; n++) { + int reg = spi_dac_init[n] >> SPI_REG_SHIFT; + + snd_ca0106_spi_write(chip, spi_dac_init[n]); + if (reg < ARRAY_SIZE(chip->spi_dac_reg)) + chip->spi_dac_reg[reg] = spi_dac_init[n]; + } + + /* Enable front dac only */ + snd_ca0106_pcm_power_dac(chip, PCM_FRONT_CHANNEL, 1); + } +} + +static void ca0106_stop_chip(struct snd_ca0106 *chip) +{ + /* 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 + */ +} + +static int snd_ca0106_create(int dev, struct snd_card *card, + struct pci_dev *pci, + struct snd_ca0106 **rchip) +{ + struct snd_ca0106 *chip; + struct snd_ca0106_details *c; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_ca0106_dev_free, + }; + + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + if (dma_set_mask(&pci->dev, DMA_BIT_MASK(32)) < 0 || + dma_set_coherent_mask(&pci->dev, DMA_BIT_MASK(32)) < 0) { + dev_err(card->dev, "error to set 32bit mask DMA\n"); + pci_disable_device(pci); + return -ENXIO; + } + + chip = kzalloc(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); + chip->res_port = request_region(chip->port, 0x20, "snd_ca0106"); + if (!chip->res_port) { + snd_ca0106_free(chip); + dev_err(card->dev, "cannot allocate the port\n"); + return -EBUSY; + } + + if (request_irq(pci->irq, snd_ca0106_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + snd_ca0106_free(chip); + dev_err(card->dev, "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 serial */ + pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); + pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); + dev_info(card->dev, "Model %04x Rev %08x Serial %08x\n", + chip->model, pci->revision, chip->serial); + strcpy(card->driver, "CA0106"); + strcpy(card->shortname, "CA0106"); + + for (c = ca0106_chip_details; c->serial; c++) { + if (subsystem[dev]) { + if (c->serial == subsystem[dev]) + break; + } else if (c->serial == chip->serial) + break; + } + chip->details = c; + if (subsystem[dev]) { + dev_info(card->dev, "Sound card name=%s, " + "subsystem=0x%x. Forced to subsystem=0x%x\n", + c->name, chip->serial, subsystem[dev]); + } + + sprintf(card->longname, "%s at 0x%lx irq %i", + c->name, chip->port, chip->irq); + + ca0106_init_chip(chip, 0); + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_ca0106_free(chip); + return err; + } + *rchip = chip; + return 0; +} + + +static void ca0106_midi_interrupt_enable(struct snd_ca_midi *midi, int intr) +{ + snd_ca0106_intr_enable((struct snd_ca0106 *)(midi->dev_id), intr); +} + +static void ca0106_midi_interrupt_disable(struct snd_ca_midi *midi, int intr) +{ + snd_ca0106_intr_disable((struct snd_ca0106 *)(midi->dev_id), intr); +} + +static unsigned char ca0106_midi_read(struct snd_ca_midi *midi, int idx) +{ + return (unsigned char)snd_ca0106_ptr_read((struct snd_ca0106 *)(midi->dev_id), + midi->port + idx, 0); +} + +static void ca0106_midi_write(struct snd_ca_midi *midi, int data, int idx) +{ + snd_ca0106_ptr_write((struct snd_ca0106 *)(midi->dev_id), midi->port + idx, 0, data); +} + +static struct snd_card *ca0106_dev_id_card(void *dev_id) +{ + return ((struct snd_ca0106 *)dev_id)->card; +} + +static int ca0106_dev_id_port(void *dev_id) +{ + return ((struct snd_ca0106 *)dev_id)->port; +} + +static int snd_ca0106_midi(struct snd_ca0106 *chip, unsigned int channel) +{ + struct snd_ca_midi *midi; + char *name; + int err; + + if (channel == CA0106_MIDI_CHAN_B) { + name = "CA0106 MPU-401 (UART) B"; + midi = &chip->midi2; + midi->tx_enable = INTE_MIDI_TX_B; + midi->rx_enable = INTE_MIDI_RX_B; + midi->ipr_tx = IPR_MIDI_TX_B; + midi->ipr_rx = IPR_MIDI_RX_B; + midi->port = MIDI_UART_B_DATA; + } else { + name = "CA0106 MPU-401 (UART)"; + midi = &chip->midi; + midi->tx_enable = INTE_MIDI_TX_A; + midi->rx_enable = INTE_MIDI_TX_B; + midi->ipr_tx = IPR_MIDI_TX_A; + midi->ipr_rx = IPR_MIDI_RX_A; + midi->port = MIDI_UART_A_DATA; + } + + midi->reset = CA0106_MPU401_RESET; + midi->enter_uart = CA0106_MPU401_ENTER_UART; + midi->ack = CA0106_MPU401_ACK; + + midi->input_avail = CA0106_MIDI_INPUT_AVAIL; + midi->output_ready = CA0106_MIDI_OUTPUT_READY; + + midi->channel = channel; + + midi->interrupt_enable = ca0106_midi_interrupt_enable; + midi->interrupt_disable = ca0106_midi_interrupt_disable; + + midi->read = ca0106_midi_read; + midi->write = ca0106_midi_write; + + midi->get_dev_id_card = ca0106_dev_id_card; + midi->get_dev_id_port = ca0106_dev_id_port; + + midi->dev_id = chip; + + if ((err = ca_midi_init(chip, midi, 0, name)) < 0) + return err; + + return 0; +} + + +static int snd_ca0106_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_ca0106 *chip; + int i, err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + err = snd_card_new(&pci->dev, index[dev], id[dev], THIS_MODULE, + 0, &card); + if (err < 0) + return err; + + err = snd_ca0106_create(dev, card, pci, &chip); + if (err < 0) + goto error; + card->private_data = chip; + + for (i = 0; i < 4; i++) { + err = snd_ca0106_pcm(chip, i); + if (err < 0) + goto error; + } + + if (chip->details->ac97 == 1) { + /* The SB0410 and SB0413 do not have an AC97 chip. */ + err = snd_ca0106_ac97(chip); + if (err < 0) + goto error; + } + err = snd_ca0106_mixer(chip); + if (err < 0) + goto error; + + dev_dbg(card->dev, "probe for MIDI channel A ..."); + err = snd_ca0106_midi(chip, CA0106_MIDI_CHAN_A); + if (err < 0) + goto error; + dev_dbg(card->dev, " done.\n"); + +#ifdef CONFIG_SND_PROC_FS + snd_ca0106_proc_init(chip); +#endif + + err = snd_card_register(card); + if (err < 0) + goto error; + + pci_set_drvdata(pci, card); + dev++; + return 0; + + error: + snd_card_free(card); + return err; +} + +static void snd_ca0106_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); +} + +#ifdef CONFIG_PM_SLEEP +static int snd_ca0106_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct snd_ca0106 *chip = card->private_data; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + for (i = 0; i < 4; i++) + snd_pcm_suspend_all(chip->pcm[i]); + if (chip->details->ac97) + snd_ac97_suspend(chip->ac97); + snd_ca0106_mixer_suspend(chip); + + ca0106_stop_chip(chip); + return 0; +} + +static int snd_ca0106_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct snd_ca0106 *chip = card->private_data; + int i; + + ca0106_init_chip(chip, 1); + + if (chip->details->ac97) + snd_ac97_resume(chip->ac97); + snd_ca0106_mixer_resume(chip); + if (chip->details->spi_dac) { + for (i = 0; i < ARRAY_SIZE(chip->spi_dac_reg); i++) + snd_ca0106_spi_write(chip, chip->spi_dac_reg[i]); + } + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static SIMPLE_DEV_PM_OPS(snd_ca0106_pm, snd_ca0106_suspend, snd_ca0106_resume); +#define SND_CA0106_PM_OPS &snd_ca0106_pm +#else +#define SND_CA0106_PM_OPS NULL +#endif + +// PCI IDs +static const struct pci_device_id snd_ca0106_ids[] = { + { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ + { 0, } +}; +MODULE_DEVICE_TABLE(pci, snd_ca0106_ids); + +// pci_driver definition +static struct pci_driver ca0106_driver = { + .name = KBUILD_MODNAME, + .id_table = snd_ca0106_ids, + .probe = snd_ca0106_probe, + .remove = snd_ca0106_remove, + .driver = { + .pm = SND_CA0106_PM_OPS, + }, +}; + +module_pci_driver(ca0106_driver); diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c new file mode 100644 index 000000000..b4d341533 --- /dev/null +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -0,0 +1,932 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.18 + * + * 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. + * 0.0.17 + * Implement Mic and Line in Capture. + * 0.0.18 + * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) + * + * This code was initially based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/moduleparam.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/info.h> +#include <sound/tlv.h> +#include <linux/io.h> + +#include "ca0106.h" + +static void ca0106_spdif_enable(struct snd_ca0106 *emu) +{ + unsigned int val; + + if (emu->spdif_enable) { + /* Digital */ + snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000); + val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000; + snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val); + val = inl(emu->port + GPIO) & ~0x101; + outl(val, emu->port + GPIO); + + } else { + /* Analog */ + snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf); + snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000f0000); + val = snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000; + snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0, val); + val = inl(emu->port + GPIO) | 0x101; + outl(val, emu->port + GPIO); + } +} + +static void ca0106_set_capture_source(struct snd_ca0106 *emu) +{ + unsigned int val = emu->capture_source; + unsigned int source, mask; + 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); +} + +static void ca0106_set_i2c_capture_source(struct snd_ca0106 *emu, + unsigned int val, int force) +{ + unsigned int ngain, ogain; + u32 source; + + snd_ca0106_i2c_write(emu, ADC_MUX, 0); /* Mute input */ + ngain = emu->i2c_capture_volume[val][0]; /* Left */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][0]; /* Left */ + if (force || ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ngain & 0xff); + ngain = emu->i2c_capture_volume[val][1]; /* Right */ + ogain = emu->i2c_capture_volume[emu->i2c_capture_source][1]; /* Right */ + if (force || ngain != ogain) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ngain & 0xff); + source = 1 << val; + snd_ca0106_i2c_write(emu, ADC_MUX, source); /* Set source */ + emu->i2c_capture_source = val; +} + +static void ca0106_set_capture_mic_line_in(struct snd_ca0106 *emu) +{ + u32 tmp; + + if (emu->capture_mic_line_in) { + /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */ + tmp = inl(emu->port+GPIO) & ~0x400; + tmp = tmp | 0x400; + outl(tmp, emu->port+GPIO); + /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_MIC); */ + } else { + /* snd_ca0106_i2c_write(emu, ADC_MUX, 0); */ /* Mute input */ + tmp = inl(emu->port+GPIO) & ~0x400; + outl(tmp, emu->port+GPIO); + /* snd_ca0106_i2c_write(emu, ADC_MUX, ADC_MUX_LINEIN); */ + } +} + +static void ca0106_set_spdif_bits(struct snd_ca0106 *emu, int idx) +{ + snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, emu->spdif_str_bits[idx]); +} + +/* + */ +static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale1, -5175, 25, 1); +static const DECLARE_TLV_DB_SCALE(snd_ca0106_db_scale2, -10350, 50, 1); + +#define snd_ca0106_shared_spdif_info snd_ctl_boolean_mono_info + +static int snd_ca0106_shared_spdif_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = emu->spdif_enable; + return 0; +} + +static int snd_ca0106_shared_spdif_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + + val = !!ucontrol->value.integer.value[0]; + change = (emu->spdif_enable != val); + if (change) { + emu->spdif_enable = val; + ca0106_spdif_enable(emu); + } + return change; +} + +static int snd_ca0106_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[6] = { + "IEC958 out", "i2s mixer out", "IEC958 in", "i2s in", "AC97 in", "SRC out" + }; + + return snd_ctl_enum_info(uinfo, 1, 6, texts); +} + +static int snd_ca0106_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->capture_source; + return 0; +} + +static int snd_ca0106_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + + val = ucontrol->value.enumerated.item[0] ; + if (val >= 6) + return -EINVAL; + change = (emu->capture_source != val); + if (change) { + emu->capture_source = val; + ca0106_set_capture_source(emu); + } + return change; +} + +static int snd_ca0106_i2c_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[4] = { + "Phone", "Mic", "Line in", "Aux" + }; + + return snd_ctl_enum_info(uinfo, 1, 4, texts); +} + +static int snd_ca0106_i2c_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->i2c_capture_source; + return 0; +} + +static int snd_ca0106_i2c_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int source_id; + int change = 0; + /* If the capture source has changed, + * update the capture volume from the cached value + * for the particular source. + */ + source_id = ucontrol->value.enumerated.item[0] ; + if (source_id >= 4) + return -EINVAL; + change = (emu->i2c_capture_source != source_id); + if (change) { + ca0106_set_i2c_capture_source(emu, source_id, 0); + } + return change; +} + +static int snd_ca0106_capture_line_in_side_out_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Side out", "Line in" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_ca0106_capture_mic_line_in_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[2] = { "Line in", "Mic in" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int snd_ca0106_capture_mic_line_in_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + + ucontrol->value.enumerated.item[0] = emu->capture_mic_line_in; + return 0; +} + +static int snd_ca0106_capture_mic_line_in_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int val; + int change = 0; + + val = ucontrol->value.enumerated.item[0] ; + if (val > 1) + return -EINVAL; + change = (emu->capture_mic_line_in != val); + if (change) { + emu->capture_mic_line_in = val; + ca0106_set_capture_mic_line_in(emu); + } + return change; +} + +static const struct snd_kcontrol_new snd_ca0106_capture_mic_line_in = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Shared Mic/Line in Capture Switch", + .info = snd_ca0106_capture_mic_line_in_info, + .get = snd_ca0106_capture_mic_line_in_get, + .put = snd_ca0106_capture_mic_line_in_put +}; + +static const struct snd_kcontrol_new snd_ca0106_capture_line_in_side_out = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Shared Line in/Side out Capture Switch", + .info = snd_ca0106_capture_line_in_side_out_info, + .get = snd_ca0106_capture_mic_line_in_get, + .put = snd_ca0106_capture_mic_line_in_put +}; + + +static int snd_ca0106_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static void decode_spdif_bits(unsigned char *status, unsigned int bits) +{ + status[0] = (bits >> 0) & 0xff; + status[1] = (bits >> 8) & 0xff; + status[2] = (bits >> 16) & 0xff; + status[3] = (bits >> 24) & 0xff; +} + +static int snd_ca0106_spdif_get_default(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + decode_spdif_bits(ucontrol->value.iec958.status, + emu->spdif_bits[idx]); + return 0; +} + +static int snd_ca0106_spdif_get_stream(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + decode_spdif_bits(ucontrol->value.iec958.status, + emu->spdif_str_bits[idx]); + return 0; +} + +static int snd_ca0106_spdif_get_mask(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *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 unsigned int encode_spdif_bits(unsigned char *status) +{ + return ((unsigned int)status[0] << 0) | + ((unsigned int)status[1] << 8) | + ((unsigned int)status[2] << 16) | + ((unsigned int)status[3] << 24); +} + +static int snd_ca0106_spdif_put_default(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + unsigned int val; + + val = encode_spdif_bits(ucontrol->value.iec958.status); + if (val != emu->spdif_bits[idx]) { + emu->spdif_bits[idx] = val; + /* FIXME: this isn't safe, but needed to keep the compatibility + * with older alsa-lib config + */ + emu->spdif_str_bits[idx] = val; + ca0106_set_spdif_bits(emu, idx); + return 1; + } + return 0; +} + +static int snd_ca0106_spdif_put_stream(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + unsigned int val; + + val = encode_spdif_bits(ucontrol->value.iec958.status); + if (val != emu->spdif_str_bits[idx]) { + emu->spdif_str_bits[idx] = val; + ca0106_set_spdif_bits(emu, idx); + return 1; + } + return 0; +} + +static int snd_ca0106_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *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(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int value; + int channel_id, reg; + + channel_id = (kcontrol->private_value >> 8) & 0xff; + reg = kcontrol->private_value & 0xff; + + 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_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int oval, nval; + int channel_id, reg; + + channel_id = (kcontrol->private_value >> 8) & 0xff; + reg = kcontrol->private_value & 0xff; + + oval = snd_ca0106_ptr_read(emu, reg, channel_id); + nval = ((0xff - ucontrol->value.integer.value[0]) << 24) | + ((0xff - ucontrol->value.integer.value[1]) << 16); + nval |= ((0xff - ucontrol->value.integer.value[0]) << 8) | + ((0xff - ucontrol->value.integer.value[1]) ); + if (oval == nval) + return 0; + snd_ca0106_ptr_write(emu, reg, channel_id, nval); + return 1; +} + +static int snd_ca0106_i2c_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *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_i2c_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + int source_id; + + source_id = kcontrol->private_value; + + ucontrol->value.integer.value[0] = emu->i2c_capture_volume[source_id][0]; + ucontrol->value.integer.value[1] = emu->i2c_capture_volume[source_id][1]; + return 0; +} + +static int snd_ca0106_i2c_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int ogain; + unsigned int ngain; + int source_id; + int change = 0; + + source_id = kcontrol->private_value; + ogain = emu->i2c_capture_volume[source_id][0]; /* Left */ + ngain = ucontrol->value.integer.value[0]; + if (ngain > 0xff) + return -EINVAL; + if (ogain != ngain) { + if (emu->i2c_capture_source == source_id) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCL, ((ngain) & 0xff) ); + emu->i2c_capture_volume[source_id][0] = ucontrol->value.integer.value[0]; + change = 1; + } + ogain = emu->i2c_capture_volume[source_id][1]; /* Right */ + ngain = ucontrol->value.integer.value[1]; + if (ngain > 0xff) + return -EINVAL; + if (ogain != ngain) { + if (emu->i2c_capture_source == source_id) + snd_ca0106_i2c_write(emu, ADC_ATTEN_ADCR, ((ngain) & 0xff)); + emu->i2c_capture_volume[source_id][1] = ucontrol->value.integer.value[1]; + change = 1; + } + + return change; +} + +#define spi_mute_info snd_ctl_boolean_mono_info + +static int spi_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT; + unsigned int bit = kcontrol->private_value & SPI_REG_MASK; + + ucontrol->value.integer.value[0] = !(emu->spi_dac_reg[reg] & bit); + return 0; +} + +static int spi_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_ca0106 *emu = snd_kcontrol_chip(kcontrol); + unsigned int reg = kcontrol->private_value >> SPI_REG_SHIFT; + unsigned int bit = kcontrol->private_value & SPI_REG_MASK; + int ret; + + ret = emu->spi_dac_reg[reg] & bit; + if (ucontrol->value.integer.value[0]) { + if (!ret) /* bit already cleared, do nothing */ + return 0; + emu->spi_dac_reg[reg] &= ~bit; + } else { + if (ret) /* bit already set, do nothing */ + return 0; + emu->spi_dac_reg[reg] |= bit; + } + + ret = snd_ca0106_spi_write(emu, emu->spi_dac_reg[reg]); + return ret ? -EINVAL : 1; +} + +#define CA_VOLUME(xname,chid,reg) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_ca0106_volume_info, \ + .get = snd_ca0106_volume_get, \ + .put = snd_ca0106_volume_put, \ + .tlv = { .p = snd_ca0106_db_scale1 }, \ + .private_value = ((chid) << 8) | (reg) \ +} + +static struct snd_kcontrol_new snd_ca0106_volume_ctls[] = { + CA_VOLUME("Analog Front Playback Volume", + CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2), + CA_VOLUME("Analog Rear Playback Volume", + CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2), + CA_VOLUME("Analog Center/LFE Playback Volume", + CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2), + CA_VOLUME("Analog Side Playback Volume", + CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2), + + CA_VOLUME("IEC958 Front Playback Volume", + CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1), + CA_VOLUME("IEC958 Rear Playback Volume", + CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1), + CA_VOLUME("IEC958 Center/LFE Playback Volume", + CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1), + CA_VOLUME("IEC958 Unknown Playback Volume", + CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1), + + CA_VOLUME("CAPTURE feedback Playback Volume", + 1, CAPTURE_CONTROL), + + { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK), + .count = 4, + .info = snd_ca0106_spdif_info, + .get = snd_ca0106_spdif_get_mask + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Playback Switch", + .info = snd_ca0106_shared_spdif_info, + .get = snd_ca0106_shared_spdif_get, + .put = snd_ca0106_shared_spdif_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Source Capture Enum", + .info = snd_ca0106_capture_source_info, + .get = snd_ca0106_capture_source_get, + .put = snd_ca0106_capture_source_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Source Capture Enum", + .info = snd_ca0106_i2c_capture_source_info, + .get = snd_ca0106_i2c_capture_source_get, + .put = snd_ca0106_i2c_capture_source_put + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .count = 4, + .info = snd_ca0106_spdif_info, + .get = snd_ca0106_spdif_get_default, + .put = snd_ca0106_spdif_put_default + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PCM_STREAM), + .count = 4, + .info = snd_ca0106_spdif_info, + .get = snd_ca0106_spdif_get_stream, + .put = snd_ca0106_spdif_put_stream + }, +}; + +#define I2C_VOLUME(xname,chid) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = snd_ca0106_i2c_volume_info, \ + .get = snd_ca0106_i2c_volume_get, \ + .put = snd_ca0106_i2c_volume_put, \ + .tlv = { .p = snd_ca0106_db_scale2 }, \ + .private_value = chid \ +} + +static struct snd_kcontrol_new snd_ca0106_volume_i2c_adc_ctls[] = { + I2C_VOLUME("Phone Capture Volume", 0), + I2C_VOLUME("Mic Capture Volume", 1), + I2C_VOLUME("Line in Capture Volume", 2), + I2C_VOLUME("Aux Capture Volume", 3), +}; + +static const int spi_dmute_reg[] = { + SPI_DMUTE0_REG, + SPI_DMUTE1_REG, + SPI_DMUTE2_REG, + 0, + SPI_DMUTE4_REG, +}; +static const int spi_dmute_bit[] = { + SPI_DMUTE0_BIT, + SPI_DMUTE1_BIT, + SPI_DMUTE2_BIT, + 0, + SPI_DMUTE4_BIT, +}; + +static struct snd_kcontrol_new +snd_ca0106_volume_spi_dac_ctl(struct snd_ca0106_details *details, + int channel_id) +{ + struct snd_kcontrol_new spi_switch = {0}; + int reg, bit; + int dac_id; + + spi_switch.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + spi_switch.access = SNDRV_CTL_ELEM_ACCESS_READWRITE; + spi_switch.info = spi_mute_info; + spi_switch.get = spi_mute_get; + spi_switch.put = spi_mute_put; + + switch (channel_id) { + case PCM_FRONT_CHANNEL: + spi_switch.name = "Analog Front Playback Switch"; + dac_id = (details->spi_dac & 0xf000) >> (4 * 3); + break; + case PCM_REAR_CHANNEL: + spi_switch.name = "Analog Rear Playback Switch"; + dac_id = (details->spi_dac & 0x0f00) >> (4 * 2); + break; + case PCM_CENTER_LFE_CHANNEL: + spi_switch.name = "Analog Center/LFE Playback Switch"; + dac_id = (details->spi_dac & 0x00f0) >> (4 * 1); + break; + case PCM_UNKNOWN_CHANNEL: + spi_switch.name = "Analog Side Playback Switch"; + dac_id = (details->spi_dac & 0x000f) >> (4 * 0); + break; + default: + /* Unused channel */ + spi_switch.name = NULL; + dac_id = 0; + } + reg = spi_dmute_reg[dac_id]; + bit = spi_dmute_bit[dac_id]; + + spi_switch.private_value = (reg << SPI_REG_SHIFT) | bit; + + return spi_switch; +} + +static int remove_ctl(struct snd_card *card, const char *name) +{ + struct snd_ctl_elem_id 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 struct snd_kcontrol *ctl_find(struct snd_card *card, const char *name) +{ + struct snd_ctl_elem_id 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(struct snd_card *card, const char *src, const char *dst) +{ + struct snd_kcontrol *kctl = ctl_find(card, src); + if (kctl) { + strcpy(kctl->id.name, dst); + return 0; + } + return -ENOENT; +} + +#define ADD_CTLS(emu, ctls) \ + do { \ + int i, _err; \ + for (i = 0; i < ARRAY_SIZE(ctls); i++) { \ + _err = snd_ctl_add(card, snd_ctl_new1(&ctls[i], emu)); \ + if (_err < 0) \ + return _err; \ + } \ + } while (0) + +static +DECLARE_TLV_DB_SCALE(snd_ca0106_master_db_scale, -6375, 25, 1); + +static char *slave_vols[] = { + "Analog Front Playback Volume", + "Analog Rear Playback Volume", + "Analog Center/LFE Playback Volume", + "Analog Side Playback Volume", + "IEC958 Front Playback Volume", + "IEC958 Rear Playback Volume", + "IEC958 Center/LFE Playback Volume", + "IEC958 Unknown Playback Volume", + "CAPTURE feedback Playback Volume", + NULL +}; + +static char *slave_sws[] = { + "Analog Front Playback Switch", + "Analog Rear Playback Switch", + "Analog Center/LFE Playback Switch", + "Analog Side Playback Switch", + "IEC958 Playback Switch", + NULL +}; + +static void add_slaves(struct snd_card *card, + struct snd_kcontrol *master, char **list) +{ + for (; *list; list++) { + struct snd_kcontrol *slave = ctl_find(card, *list); + if (slave) + snd_ctl_add_slave(master, slave); + } +} + +int snd_ca0106_mixer(struct snd_ca0106 *emu) +{ + int err; + struct snd_card *card = emu->card; + char **c; + struct snd_kcontrol *vmaster; + 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", + "Beep Playback Switch", + "Beep Playback Volume", + "Mono Output Select", + "Capture Source", + "Capture Switch", + "Capture Volume", + "External Amplifier", + "Sigmatel 4-Speaker Stereo Playback Switch", + "Surround Phase Inversion Playback Switch", + 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 + + ADD_CTLS(emu, snd_ca0106_volume_ctls); + if (emu->details->i2c_adc == 1) { + ADD_CTLS(emu, snd_ca0106_volume_i2c_adc_ctls); + if (emu->details->gpio_type == 1) + err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_mic_line_in, emu)); + else /* gpio_type == 2 */ + err = snd_ctl_add(card, snd_ctl_new1(&snd_ca0106_capture_line_in_side_out, emu)); + if (err < 0) + return err; + } + if (emu->details->spi_dac) { + int i; + for (i = 0;; i++) { + struct snd_kcontrol_new ctl; + ctl = snd_ca0106_volume_spi_dac_ctl(emu->details, i); + if (!ctl.name) + break; + err = snd_ctl_add(card, snd_ctl_new1(&ctl, emu)); + if (err < 0) + return err; + } + } + + /* Create virtual master controls */ + vmaster = snd_ctl_make_virtual_master("Master Playback Volume", + snd_ca0106_master_db_scale); + if (!vmaster) + return -ENOMEM; + err = snd_ctl_add(card, vmaster); + if (err < 0) + return err; + add_slaves(card, vmaster, slave_vols); + + if (emu->details->spi_dac) { + vmaster = snd_ctl_make_virtual_master("Master Playback Switch", + NULL); + if (!vmaster) + return -ENOMEM; + err = snd_ctl_add(card, vmaster); + if (err < 0) + return err; + add_slaves(card, vmaster, slave_sws); + } + + strcpy(card->mixername, "CA0106"); + return 0; +} + +#ifdef CONFIG_PM_SLEEP +struct ca0106_vol_tbl { + unsigned int channel_id; + unsigned int reg; +}; + +static struct ca0106_vol_tbl saved_volumes[NUM_SAVED_VOLUMES] = { + { CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME2 }, + { CONTROL_FRONT_CHANNEL, PLAYBACK_VOLUME1 }, + { CONTROL_REAR_CHANNEL, PLAYBACK_VOLUME1 }, + { CONTROL_CENTER_LFE_CHANNEL, PLAYBACK_VOLUME1 }, + { CONTROL_UNKNOWN_CHANNEL, PLAYBACK_VOLUME1 }, + { 1, CAPTURE_CONTROL }, +}; + +void snd_ca0106_mixer_suspend(struct snd_ca0106 *chip) +{ + int i; + + /* save volumes */ + for (i = 0; i < NUM_SAVED_VOLUMES; i++) + chip->saved_vol[i] = + snd_ca0106_ptr_read(chip, saved_volumes[i].reg, + saved_volumes[i].channel_id); +} + +void snd_ca0106_mixer_resume(struct snd_ca0106 *chip) +{ + int i; + + for (i = 0; i < NUM_SAVED_VOLUMES; i++) + snd_ca0106_ptr_write(chip, saved_volumes[i].reg, + saved_volumes[i].channel_id, + chip->saved_vol[i]); + + ca0106_spdif_enable(chip); + ca0106_set_capture_source(chip); + ca0106_set_i2c_capture_source(chip, chip->i2c_capture_source, 1); + for (i = 0; i < 4; i++) + ca0106_set_spdif_bits(chip, i); + if (chip->details->i2c_adc) + ca0106_set_capture_mic_line_in(chip); +} +#endif /* CONFIG_PM_SLEEP */ diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c new file mode 100644 index 000000000..a2c85cc37 --- /dev/null +++ b/sound/pci/ca0106/ca0106_proc.c @@ -0,0 +1,453 @@ +/* + * Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk> + * Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit + * Version: 0.0.18 + * + * 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. + * 0.0.18 + * Implement support for Line-in capture on SB Live 24bit. + * + * This code was initially based on code from ALSA's emu10k1x.c which is: + * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/moduleparam.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/info.h> +#include <sound/asoundef.h> + +#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" }, +}; + + +static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *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, "unknown\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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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 && 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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *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 && val <= 0xffffffff && channel_id <= 3) + snd_ca0106_ptr_write(emu, reg, channel_id, val); + } +} + +static void snd_ca0106_proc_i2c_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_ca0106 *emu = entry->private_data; + char line[64]; + unsigned int reg, val; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "%x %x", ®, &val) != 2) + continue; + if ((reg <= 0x7f) || (val <= 0x1ff)) { + snd_ca0106_i2c_write(emu, reg, val); + } + } +} + +int snd_ca0106_proc_init(struct snd_ca0106 *emu) +{ + struct snd_info_entry *entry; + + if(! snd_card_proc_new(emu->card, "iec958", &entry)) + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_iec958); + if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) { + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read32); + entry->c.text.write = snd_ca0106_proc_reg_write32; + entry->mode |= 0200; + } + if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry)) + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read16); + if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry)) + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read8); + if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) { + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read1); + entry->c.text.write = snd_ca0106_proc_reg_write; + entry->mode |= 0200; + } + if(! snd_card_proc_new(emu->card, "ca0106_i2c", &entry)) { + entry->c.text.write = snd_ca0106_proc_i2c_write; + entry->private_data = emu; + entry->mode |= 0200; + } + if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry)) + snd_info_set_text_ops(entry, emu, snd_ca0106_proc_reg_read2); + return 0; +} diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c new file mode 100644 index 000000000..4d4d38520 --- /dev/null +++ b/sound/pci/ca0106/ca_midi.c @@ -0,0 +1,316 @@ +/* + * Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de> + * Creative Audio MIDI, for the CA0106 Driver + * Version: 0.0.1 + * + * Changelog: + * Implementation is based on mpu401 and emu10k1x and + * tested with ca0106. + * mpu401: Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * emu10k1x: Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + * + */ + +#include <linux/spinlock.h> +#include <sound/core.h> +#include <sound/rawmidi.h> + +#include "ca_midi.h" + +#define ca_midi_write_data(midi, data) midi->write(midi, data, 0) +#define ca_midi_write_cmd(midi, data) midi->write(midi, data, 1) +#define ca_midi_read_data(midi) midi->read(midi, 0) +#define ca_midi_read_stat(midi) midi->read(midi, 1) +#define ca_midi_input_avail(midi) (!(ca_midi_read_stat(midi) & midi->input_avail)) +#define ca_midi_output_ready(midi) (!(ca_midi_read_stat(midi) & midi->output_ready)) + +static void ca_midi_clear_rx(struct snd_ca_midi *midi) +{ + int timeout = 100000; + for (; timeout > 0 && ca_midi_input_avail(midi); timeout--) + ca_midi_read_data(midi); +#ifdef CONFIG_SND_DEBUG + if (timeout <= 0) + pr_err("ca_midi_clear_rx: timeout (status = 0x%x)\n", + ca_midi_read_stat(midi)); +#endif +} + +static void ca_midi_interrupt(struct snd_ca_midi *midi, unsigned int status) +{ + unsigned char byte; + + if (midi->rmidi == NULL) { + midi->interrupt_disable(midi,midi->tx_enable | midi->rx_enable); + return; + } + + spin_lock(&midi->input_lock); + if ((status & midi->ipr_rx) && ca_midi_input_avail(midi)) { + if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) { + ca_midi_clear_rx(midi); + } else { + byte = ca_midi_read_data(midi); + if(midi->substream_input) + snd_rawmidi_receive(midi->substream_input, &byte, 1); + + + } + } + spin_unlock(&midi->input_lock); + + spin_lock(&midi->output_lock); + if ((status & midi->ipr_tx) && ca_midi_output_ready(midi)) { + if (midi->substream_output && + snd_rawmidi_transmit(midi->substream_output, &byte, 1) == 1) { + ca_midi_write_data(midi, byte); + } else { + midi->interrupt_disable(midi,midi->tx_enable); + } + } + spin_unlock(&midi->output_lock); + +} + +static void ca_midi_cmd(struct snd_ca_midi *midi, unsigned char cmd, int ack) +{ + unsigned long flags; + int timeout, ok; + + spin_lock_irqsave(&midi->input_lock, flags); + ca_midi_write_data(midi, 0x00); + /* ca_midi_clear_rx(midi); */ + + ca_midi_write_cmd(midi, cmd); + if (ack) { + ok = 0; + timeout = 10000; + while (!ok && timeout-- > 0) { + if (ca_midi_input_avail(midi)) { + if (ca_midi_read_data(midi) == midi->ack) + ok = 1; + } + } + if (!ok && ca_midi_read_data(midi) == midi->ack) + ok = 1; + } else { + ok = 1; + } + spin_unlock_irqrestore(&midi->input_lock, flags); + if (!ok) + pr_err("ca_midi_cmd: 0x%x failed at 0x%x (status = 0x%x, data = 0x%x)!!!\n", + cmd, + midi->get_dev_id_port(midi->dev_id), + ca_midi_read_stat(midi), + ca_midi_read_data(midi)); +} + +static int ca_midi_input_open(struct snd_rawmidi_substream *substream) +{ + struct snd_ca_midi *midi = substream->rmidi->private_data; + unsigned long flags; + + if (snd_BUG_ON(!midi->dev_id)) + return -ENXIO; + spin_lock_irqsave(&midi->open_lock, flags); + midi->midi_mode |= CA_MIDI_MODE_INPUT; + midi->substream_input = substream; + if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) { + spin_unlock_irqrestore(&midi->open_lock, flags); + ca_midi_cmd(midi, midi->reset, 1); + ca_midi_cmd(midi, midi->enter_uart, 1); + } else { + spin_unlock_irqrestore(&midi->open_lock, flags); + } + return 0; +} + +static int ca_midi_output_open(struct snd_rawmidi_substream *substream) +{ + struct snd_ca_midi *midi = substream->rmidi->private_data; + unsigned long flags; + + if (snd_BUG_ON(!midi->dev_id)) + return -ENXIO; + spin_lock_irqsave(&midi->open_lock, flags); + midi->midi_mode |= CA_MIDI_MODE_OUTPUT; + midi->substream_output = substream; + if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) { + spin_unlock_irqrestore(&midi->open_lock, flags); + ca_midi_cmd(midi, midi->reset, 1); + ca_midi_cmd(midi, midi->enter_uart, 1); + } else { + spin_unlock_irqrestore(&midi->open_lock, flags); + } + return 0; +} + +static int ca_midi_input_close(struct snd_rawmidi_substream *substream) +{ + struct snd_ca_midi *midi = substream->rmidi->private_data; + unsigned long flags; + + if (snd_BUG_ON(!midi->dev_id)) + return -ENXIO; + spin_lock_irqsave(&midi->open_lock, flags); + midi->interrupt_disable(midi,midi->rx_enable); + midi->midi_mode &= ~CA_MIDI_MODE_INPUT; + midi->substream_input = NULL; + if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT)) { + spin_unlock_irqrestore(&midi->open_lock, flags); + ca_midi_cmd(midi, midi->reset, 0); + } else { + spin_unlock_irqrestore(&midi->open_lock, flags); + } + return 0; +} + +static int ca_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct snd_ca_midi *midi = substream->rmidi->private_data; + unsigned long flags; + + if (snd_BUG_ON(!midi->dev_id)) + return -ENXIO; + + spin_lock_irqsave(&midi->open_lock, flags); + + midi->interrupt_disable(midi,midi->tx_enable); + midi->midi_mode &= ~CA_MIDI_MODE_OUTPUT; + midi->substream_output = NULL; + + if (!(midi->midi_mode & CA_MIDI_MODE_INPUT)) { + spin_unlock_irqrestore(&midi->open_lock, flags); + ca_midi_cmd(midi, midi->reset, 0); + } else { + spin_unlock_irqrestore(&midi->open_lock, flags); + } + return 0; +} + +static void ca_midi_input_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct snd_ca_midi *midi = substream->rmidi->private_data; + + if (snd_BUG_ON(!midi->dev_id)) + return; + + if (up) { + midi->interrupt_enable(midi,midi->rx_enable); + } else { + midi->interrupt_disable(midi, midi->rx_enable); + } +} + +static void ca_midi_output_trigger(struct snd_rawmidi_substream *substream, int up) +{ + struct snd_ca_midi *midi = substream->rmidi->private_data; + unsigned long flags; + + if (snd_BUG_ON(!midi->dev_id)) + return; + + if (up) { + int max = 4; + unsigned char byte; + + spin_lock_irqsave(&midi->output_lock, flags); + + /* try to send some amount of bytes here before interrupts */ + while (max > 0) { + if (ca_midi_output_ready(midi)) { + if (!(midi->midi_mode & CA_MIDI_MODE_OUTPUT) || + snd_rawmidi_transmit(substream, &byte, 1) != 1) { + /* no more data */ + spin_unlock_irqrestore(&midi->output_lock, flags); + return; + } + ca_midi_write_data(midi, byte); + max--; + } else { + break; + } + } + + spin_unlock_irqrestore(&midi->output_lock, flags); + midi->interrupt_enable(midi,midi->tx_enable); + + } else { + midi->interrupt_disable(midi,midi->tx_enable); + } +} + +static const struct snd_rawmidi_ops ca_midi_output = +{ + .open = ca_midi_output_open, + .close = ca_midi_output_close, + .trigger = ca_midi_output_trigger, +}; + +static const struct snd_rawmidi_ops ca_midi_input = +{ + .open = ca_midi_input_open, + .close = ca_midi_input_close, + .trigger = ca_midi_input_trigger, +}; + +static void ca_midi_free(struct snd_ca_midi *midi) +{ + midi->interrupt = NULL; + midi->interrupt_enable = NULL; + midi->interrupt_disable = NULL; + midi->read = NULL; + midi->write = NULL; + midi->get_dev_id_card = NULL; + midi->get_dev_id_port = NULL; + midi->rmidi = NULL; +} + +static void ca_rmidi_free(struct snd_rawmidi *rmidi) +{ + ca_midi_free(rmidi->private_data); +} + +int ca_midi_init(void *dev_id, struct snd_ca_midi *midi, int device, char *name) +{ + struct snd_rawmidi *rmidi; + int err; + + if ((err = snd_rawmidi_new(midi->get_dev_id_card(midi->dev_id), name, device, 1, 1, &rmidi)) < 0) + return err; + + midi->dev_id = dev_id; + midi->interrupt = ca_midi_interrupt; + + spin_lock_init(&midi->open_lock); + spin_lock_init(&midi->input_lock); + spin_lock_init(&midi->output_lock); + + strcpy(rmidi->name, name); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &ca_midi_output); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &ca_midi_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT | + SNDRV_RAWMIDI_INFO_INPUT | + SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = midi; + rmidi->private_free = ca_rmidi_free; + + midi->rmidi = rmidi; + return 0; +} + diff --git a/sound/pci/ca0106/ca_midi.h b/sound/pci/ca0106/ca_midi.h new file mode 100644 index 000000000..922ed3e37 --- /dev/null +++ b/sound/pci/ca0106/ca_midi.h @@ -0,0 +1,66 @@ +/* + * Copyright 10/16/2005 Tilman Kranz <tilde@tk-sls.de> + * Creative Audio MIDI, for the CA0106 Driver + * Version: 0.0.1 + * + * Changelog: + * See ca_midi.c + * + * 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 <linux/spinlock.h> +#include <sound/rawmidi.h> +#include <sound/mpu401.h> + +#define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT +#define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT + +struct snd_ca_midi { + + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *substream_input; + struct snd_rawmidi_substream *substream_output; + + void *dev_id; + + spinlock_t input_lock; + spinlock_t output_lock; + spinlock_t open_lock; + + unsigned int channel; + + unsigned int midi_mode; + int port; + int tx_enable, rx_enable; + int ipr_tx, ipr_rx; + + int input_avail, output_ready; + int ack, reset, enter_uart; + + void (*interrupt)(struct snd_ca_midi *midi, unsigned int status); + void (*interrupt_enable)(struct snd_ca_midi *midi, int intr); + void (*interrupt_disable)(struct snd_ca_midi *midi, int intr); + + unsigned char (*read)(struct snd_ca_midi *midi, int idx); + void (*write)(struct snd_ca_midi *midi, int data, int idx); + + /* get info from dev_id */ + struct snd_card *(*get_dev_id_card)(void *dev_id); + int (*get_dev_id_port)(void *dev_id); +}; + +int ca_midi_init(void *card, struct snd_ca_midi *midi, int device, char *name); |