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Commit 5f1f8447 authored by Jaroslav Kysela's avatar Jaroslav Kysela
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[ALSA] Add snd-ca0106 driver 

Documentation,PCI drivers,CA0106 driver
Added snd-ca0106 driver for SB Audigy LS / Live 24bit boards
by James Courtier-Dutton <James@superbug.demon.co.uk>.
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
parent af2c5dd3
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Documentation/sound/alsa/ALSA-Configuration.txt
View file @ 5f1f8447
......@@ -256,6 +256,14 @@ Module parameters
Module supports up to 8 cards.
Module snd-ca0106
-----------------
Module for Creative Audigy LS and SB Live 24bit
Module supports up to 8 cards.
Module snd-cmi8330
------------------
......
sound/pci/Kconfig
View file @ 5f1f8447
......@@ -184,6 +184,17 @@ config SND_EMU10K1X
To compile this driver as a module, choose M here: the module
will be called snd-emu10k1x.
config SND_CA0106
tristate "SB Audigy LS / Live 24bit"
depends on SND
select SND_AC97_CODEC
help
Say Y here to include support for the Sound Blaster Audigy LS
and Live 24bit.
To compile this driver as a module, choose M here: the module
will be called snd-ca0106.
config SND_KORG1212
tristate "Korg 1212 IO"
depends on SND
......
sound/pci/Makefile
View file @ 5f1f8447
......@@ -50,6 +50,7 @@ obj-$(CONFIG_SND) += \
ac97/ \
ali5451/ \
au88x0/ \
ca0106/ \
cs46xx/ \
emu10k1/ \
ice1712/ \
......
sound/pci/ca0106/Makefile 0 → 100644
View file @ 5f1f8447
snd-ca0106-objs := ca0106_main.o ca0106_proc.o ca0106_mixer.o
obj-$(CONFIG_SND_CA0106) += snd-ca0106.o
sound/pci/ca0106/ca0106.h 0 → 100644
View file @ 5f1f8447
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
* Version: 0.0.20
*
* FEATURES currently supported:
* See ca0106_main.c for features.
*
* Changelog:
* Support interrupts per period.
* Removed noise from Center/LFE channel when in Analog mode.
* Rename and remove mixer controls.
* 0.0.6
* Use separate card based DMA buffer for periods table list.
* 0.0.7
* Change remove and rename ctrls into lists.
* 0.0.8
* Try to fix capture sources.
* 0.0.9
* Fix AC3 output.
* Enable S32_LE format support.
* 0.0.10
* Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
* 0.0.11
* Add Model name recognition.
* 0.0.12
* Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
* Remove redundent "voice" handling.
* 0.0.13
* Single trigger call for multi channels.
* 0.0.14
* Set limits based on what the sound card hardware can do.
* playback periods_min=2, periods_max=8
* capture hw constraints require period_size = n * 64 bytes.
* playback hw constraints require period_size = n * 64 bytes.
* 0.0.15
* Separated ca0106.c into separate functional .c files.
* 0.0.16
* Implement 192000 sample rate.
* 0.0.17
* Add support for SB0410 and SB0413.
* 0.0.18
* Modified Copyright message.
* 0.0.19
* Added I2C and SPI registers. Filled in interrupt enable.
* 0.0.20
* Added GPIO info for SB Live 24bit.
*
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <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 = ? / 1 = ?
* bit 13 0 = ? / 1 = ?
* bit 14 0 = Mute / 1 = Analog out
* bit 15 0 = ? / 1 = ?
* Both bit 9 and bit 14 have to be set for analog sound to work on the SB Live 24bit.
*/
/* 8 general purpose programmable In/Out pins.
* GPI [8:0] Read only. Default 0.
* GPO [15:8] Default 0x9. (Default to SPDIF jack enabled for SPDIF)
* GPO Enable [23:16] Default 0x0f. Setting a bit to 1, causes the pin to be an output pin.
*/
#define AC97DATA 0x1c /* AC97 register set data register (16 bit) */
#define AC97ADDRESS 0x1e /* AC97 register set address register (8 bit) */
/********************************************************************************************************/
/* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */
/********************************************************************************************************/
/* Initally all registers from 0x00 to 0x3f have zero contents. */
#define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */
/* One list entry: 4 bytes for DMA address,
* 4 bytes for period_size << 16.
* One list entry is 8 bytes long.
* One list entry for each period in the buffer.
*/
/* ADDR[31:0], Default: 0x0 */
#define PLAYBACK_LIST_SIZE 0x01 /* Size of list in bytes << 16. E.g. 8 periods -> 0x00380000 */
/* SIZE[21:16], Default: 0x8 */
#define PLAYBACK_LIST_PTR 0x02 /* Pointer to the current period being played */
/* PTR[5:0], Default: 0x0 */
#define PLAYBACK_UNKNOWN3 0x03 /* Not used ?? */
#define PLAYBACK_DMA_ADDR 0x04 /* Playback DMA addresss */
/* DMA[31:0], Default: 0x0 */
#define PLAYBACK_PERIOD_SIZE 0x05 /* Playback period size. win2000 uses 0x04000000 */
/* SIZE[31:16], Default: 0x0 */
#define PLAYBACK_POINTER 0x06 /* Playback period pointer. Used with PLAYBACK_LIST_PTR to determine buffer position currently in DAC */
/* POINTER[15:0], Default: 0x0 */
#define PLAYBACK_PERIOD_END_ADDR 0x07 /* Playback fifo end address */
/* END_ADDR[15:0], FLAG[16] 0 = don't stop, 1 = stop */
#define PLAYBACK_FIFO_OFFSET_ADDRESS 0x08 /* Current fifo offset address [21:16] */
/* Cache size valid [5:0] */
#define PLAYBACK_UNKNOWN9 0x09 /* 0x9 to 0xf Unused */
#define CAPTURE_DMA_ADDR 0x10 /* Capture DMA address */
/* DMA[31:0], Default: 0x0 */
#define CAPTURE_BUFFER_SIZE 0x11 /* Capture buffer size */
/* SIZE[31:16], Default: 0x0 */
#define CAPTURE_POINTER 0x12 /* Capture buffer pointer. Sample currently in ADC */
/* POINTER[15:0], Default: 0x0 */
#define CAPTURE_FIFO_OFFSET_ADDRESS 0x13 /* Current fifo offset address [21:16] */
/* Cache size valid [5:0] */
#define PLAYBACK_LAST_SAMPLE 0x20 /* The sample currently being played */
/* 0x21 - 0x3f unused */
#define BASIC_INTERRUPT 0x40 /* Used by both playback and capture interrupt handler */
/* Playback (0x1<<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 -> Sheild on all three, 4 -> Red.
* So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card.
*/
/* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS
* The Rear SPDIF can be used for Stereo PCM and also AC3/DTS
* The Center/LFE SPDIF cannot be used for AC3/DTS, but can be used for Stereo PCM.
* Summary: For ALSA we use the Rear channel for SPDIF Digital AC3/DTS output
*/
/* A standard 2 pole mono mini-jack to RCA plug can be used for SPDIF Stereo PCM output from the Front channel.
* A standard 3 pole stereo mini-jack to 2 RCA plugs can be used for SPDIF AC3/DTS and Stereo PCM output utilising the Rear channel and just one of the RCA plugs.
*/
#define SPCS0 0x41 /* SPDIF output Channel Status 0 register. For Rear. default=0x02108004, non-audio=0x02108006 */
#define SPCS1 0x42 /* SPDIF output Channel Status 1 register. For Front */
#define SPCS2 0x43 /* SPDIF output Channel Status 2 register. For Center/LFE */
#define SPCS3 0x44 /* SPDIF output Channel Status 3 register. Unknown */
/* When Channel set to 0: */
#define SPCS_CLKACCYMASK 0x30000000 /* Clock accuracy */
#define SPCS_CLKACCY_1000PPM 0x00000000 /* 1000 parts per million */
#define SPCS_CLKACCY_50PPM 0x10000000 /* 50 parts per million */
#define SPCS_CLKACCY_VARIABLE 0x20000000 /* Variable accuracy */
#define SPCS_SAMPLERATEMASK 0x0f000000 /* Sample rate */
#define SPCS_SAMPLERATE_44 0x00000000 /* 44.1kHz sample rate */
#define SPCS_SAMPLERATE_48 0x02000000 /* 48kHz sample rate */
#define SPCS_SAMPLERATE_32 0x03000000 /* 32kHz sample rate */
#define SPCS_CHANNELNUMMASK 0x00f00000 /* Channel number */
#define SPCS_CHANNELNUM_UNSPEC 0x00000000 /* Unspecified channel number */
#define SPCS_CHANNELNUM_LEFT 0x00100000 /* Left channel */
#define SPCS_CHANNELNUM_RIGHT 0x00200000 /* Right channel */
#define SPCS_SOURCENUMMASK 0x000f0000 /* Source number */
#define SPCS_SOURCENUM_UNSPEC 0x00000000 /* Unspecified source number */
#define SPCS_GENERATIONSTATUS 0x00008000 /* Originality flag (see IEC-958 spec) */
#define SPCS_CATEGORYCODEMASK 0x00007f00 /* Category code (see IEC-958 spec) */
#define SPCS_MODEMASK 0x000000c0 /* Mode (see IEC-958 spec) */
#define SPCS_EMPHASISMASK 0x00000038 /* Emphasis */
#define SPCS_EMPHASIS_NONE 0x00000000 /* No emphasis */
#define SPCS_EMPHASIS_50_15 0x00000008 /* 50/15 usec 2 channel */
#define SPCS_COPYRIGHT 0x00000004 /* Copyright asserted flag -- do not modify */
#define SPCS_NOTAUDIODATA 0x00000002 /* 0 = Digital audio, 1 = not audio */
#define SPCS_PROFESSIONAL 0x00000001 /* 0 = Consumer (IEC-958), 1 = pro (AES3-1992) */
/* When Channel set to 1: */
#define SPCS_WORD_LENGTH_MASK 0x0000000f /* Word Length Mask */
#define SPCS_WORD_LENGTH_16 0x00000008 /* Word Length 16 bit */
#define SPCS_WORD_LENGTH_17 0x00000006 /* Word Length 17 bit */
#define SPCS_WORD_LENGTH_18 0x00000004 /* Word Length 18 bit */
#define SPCS_WORD_LENGTH_19 0x00000002 /* Word Length 19 bit */
#define SPCS_WORD_LENGTH_20A 0x0000000a /* Word Length 20 bit */
#define SPCS_WORD_LENGTH_20 0x00000009 /* Word Length 20 bit (both 0xa and 0x9 are 20 bit) */
#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */
#define SPCS_WORD_LENGTH_21 0x00000007 /* Word Length 21 bit */
#define SPCS_WORD_LENGTH_22 0x00000005 /* Word Length 22 bit */
#define SPCS_WORD_LENGTH_23 0x00000003 /* Word Length 23 bit */
#define SPCS_WORD_LENGTH_24 0x0000000b /* Word Length 24 bit */
#define SPCS_ORIGINAL_SAMPLE_RATE_MASK 0x000000f0 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_NONE 0x00000000 /* Original Sample rate not indicated */
#define SPCS_ORIGINAL_SAMPLE_RATE_16000 0x00000010 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_RES1 0x00000020 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_32000 0x00000030 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_12000 0x00000040 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_11025 0x00000050 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_8000 0x00000060 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_RES2 0x00000070 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_192000 0x00000080 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_24000 0x00000090 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_96000 0x000000a0 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_48000 0x000000b0 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_176400 0x000000c0 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_22050 0x000000d0 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_88200 0x000000e0 /* Original Sample rate */
#define SPCS_ORIGINAL_SAMPLE_RATE_44100 0x000000f0 /* Original Sample rate */
#define SPDIF_SELECT1 0x45 /* Enables SPDIF or Analogue outputs 0-SPDIF, 0xf00-Analogue */
/* 0x100 - Front, 0x800 - Rear, 0x200 - Center/LFE.
* But as the jack is shared, use 0xf00.
* The Windows2000 driver uses 0x0000000f for both digital and analog.
* 0xf00 introduces interesting noises onto the Center/LFE.
* If you turn the volume up, you hear computer noise,
* e.g. mouse moving, changing between app windows etc.
* So, I am going to set this to 0x0000000f all the time now,
* same as the windows driver does.
* Use register SPDIF_SELECT2(0x72) to switch between SPDIF and Analog.
*/
/* When Channel = 0:
* Wide SPDIF format [3:0] (one bit for each channel) (0=20bit, 1=24bit)
* Tristate SPDIF Output [11:8] (one bit for each channel) (0=Not tristate, 1=Tristate)
* SPDIF Bypass enable [19:16] (one bit for each channel) (0=Not bypass, 1=Bypass)
*/
/* When Channel = 1:
* SPDIF 0 User data [7:0]
* SPDIF 1 User data [15:8]
* SPDIF 0 User data [23:16]
* SPDIF 0 User data [31:24]
* User data can be sent by using the SPDIF output frame pending and SPDIF output user bit interrupts.
*/
#define WATERMARK 0x46 /* Test bit to indicate cache usage level */
#define SPDIF_INPUT_STATUS 0x49 /* SPDIF Input status register. Bits the same as SPCS.
* When Channel = 0: Bits the same as SPCS channel 0.
* When Channel = 1: Bits the same as SPCS channel 1.
* When Channel = 2:
* SPDIF Input User data [16:0]
* SPDIF Input Frame count [21:16]
*/
#define CAPTURE_CACHE_DATA 0x50 /* 0x50-0x5f Recorded samples. */
#define CAPTURE_SOURCE 0x60 /* Capture Source 0 = MIC */
#define CAPTURE_SOURCE_CHANNEL0 0xf0000000 /* Mask for selecting the Capture sources */
#define CAPTURE_SOURCE_CHANNEL1 0x0f000000 /* 0 - SPDIF mixer output. */
#define CAPTURE_SOURCE_CHANNEL2 0x00f00000 /* 1 - What you hear or . 2 - ?? */
#define CAPTURE_SOURCE_CHANNEL3 0x000f0000 /* 3 - Mic in, Line in, TAD in, Aux in. */
#define CAPTURE_SOURCE_RECORD_MAP 0x0000ffff /* Default 0x00e4 */
/* Record Map [7:0] (2 bits per channel) 0=mapped to channel 0, 1=mapped to channel 1, 2=mapped to channel2, 3=mapped to channel3
* Record source select for channel 0 [18:16]
* Record source select for channel 1 [22:20]
* Record source select for channel 2 [26:24]
* Record source select for channel 3 [30:28]
* 0 - SPDIF mixer output.
* 1 - i2s mixer output.
* 2 - SPDIF input.
* 3 - i2s input.
* 4 - AC97 capture.
* 5 - SRC output.
*/
#define CAPTURE_VOLUME1 0x61 /* Capture volume per channel 0-3 */
#define CAPTURE_VOLUME2 0x62 /* Capture volume per channel 4-7 */
#define PLAYBACK_ROUTING1 0x63 /* Playback routing of channels 0-7. Effects AC3 output. Default 0x32765410 */
#define ROUTING1_REAR 0x77000000 /* Channel_id 0 sends to 10, Channel_id 1 sends to 32 */
#define ROUTING1_NULL 0x00770000 /* Channel_id 2 sends to 54, Channel_id 3 sends to 76 */
#define ROUTING1_CENTER_LFE 0x00007700 /* 0x32765410 means, send Channel_id 0 to FRONT, Channel_id 1 to REAR */
#define ROUTING1_FRONT 0x00000077 /* Channel_id 2 to CENTER_LFE, Channel_id 3 to NULL. */
/* Channel_id's handle stereo channels. Channel X is a single mono channel */
/* Host is input from the PCI bus. */
/* Host channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
* Host channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
* Host channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
* Host channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
* Host channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
* Host channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
* Host channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
* Host channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
*/
#define PLAYBACK_ROUTING2 0x64 /* Playback Routing . Feeding Capture channels back into Playback. Effects AC3 output. Default 0x76767676 */
/* SRC is input from the capture inputs. */
/* SRC channel 0 [2:0] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 1 [6:4] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 2 [10:8] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 3 [14:12] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 4 [18:16] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 5 [22:20] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 6 [26:24] -> SPDIF Mixer/Router channel 0-7.
* SRC channel 7 [30:28] -> SPDIF Mixer/Router channel 0-7.
*/
#define PLAYBACK_MUTE 0x65 /* Unknown. While playing 0x0, while silent 0x00fc0000 */
/* SPDIF Mixer input control:
* Invert SRC to SPDIF Mixer [7-0] (One bit per channel)
* Invert Host to SPDIF Mixer [15:8] (One bit per channel)
* SRC to SPDIF Mixer disable [23:16] (One bit per channel)
* Host to SPDIF Mixer disable [31:24] (One bit per channel)
*/
#define PLAYBACK_VOLUME1 0x66 /* Playback SPDIF volume per channel. Set to the same PLAYBACK_VOLUME(0x6a) */
/* PLAYBACK_VOLUME1 must be set to 30303030 for SPDIF AC3 Playback */
/* SPDIF mixer input volume. 0=12dB, 0x30=0dB, 0xFE=-51.5dB, 0xff=Mute */
/* One register for each of the 4 stereo streams. */
/* SRC Right volume [7:0]
* SRC Left volume [15:8]
* Host Right volume [23:16]
* Host Left volume [31:24]
*/
#define CAPTURE_ROUTING1 0x67 /* Capture Routing. Default 0x32765410 */
/* Similar to register 0x63, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
#define CAPTURE_ROUTING2 0x68 /* Unknown Routing. Default 0x76767676 */
/* Similar to register 0x64, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
#define CAPTURE_MUTE 0x69 /* Unknown. While capturing 0x0, while silent 0x00fc0000 */
/* Similar to register 0x65, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
#define PLAYBACK_VOLUME2 0x6a /* Playback Analog volume per channel. Does not effect AC3 output */
/* Similar to register 0x66, except that the destination is the I2S mixer instead of the SPDIF mixer. I.E. Outputs to the Analog outputs instead of SPDIF. */
#define UNKNOWN6b 0x6b /* Unknown. Readonly. Default 00400000 00400000 00400000 00400000 */
#define UART_A_DATA 0x6c /* Uart, used in setting sample rates, bits per sample etc. */
#define UART_A_CMD 0x6d /* Uart, used in setting sample rates, bits per sample etc. */
#define UART_B_DATA 0x6e /* Uart, Unknown. */
#define UART_B_CMD 0x6f /* Uart, Unknown. */
#define SAMPLE_RATE_TRACKER_STATUS 0x70 /* Readonly. Default 00108000 00108000 00500000 00500000 */
/* Estimated sample rate [19:0] Relative to 48kHz. 0x8000 = 1.0
* Rate Locked [20]
* SPDIF Locked [21] For SPDIF channel only.
* Valid Audio [22] For SPDIF channel only.
*/
#define CAPTURE_CONTROL 0x71 /* Some sort of routing. default = 40c81000 30303030 30300000 00700000 */
/* Channel_id 0: 0x40c81000 must be changed to 0x40c80000 for SPDIF AC3 input or output. */
/* Channel_id 1: 0xffffffff(mute) 0x30303030(max) controls CAPTURE feedback into PLAYBACK. */
/* Sample rate output control register Channel=0
* Sample output rate [1:0] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
* Sample input rate [3:2] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
* SRC input source select [4] 0=Audio from digital mixer, 1=Audio from analog source.
* Record rate [9:8] (0=48kHz, 1=Not available, 2=96kHz, 3=192Khz)
* Record mixer output enable [12:10]
* I2S input rate master mode [15:14] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
* I2S output rate [17:16] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
* I2S output source select [18] (0=Audio from host, 1=Audio from SRC)
* Record mixer I2S enable [20:19] (enable/disable i2sin1 and i2sin0)
* I2S output master clock select [21] (0=256*I2S output rate, 1=512*I2S output rate.)
* I2S input master clock select [22] (0=256*I2S input rate, 1=512*I2S input rate.)
* I2S input mode [23] (0=Slave, 1=Master)
* SPDIF output rate [25:24] (0=48kHz, 1=44.1kHz, 2=96kHz, 3=192Khz)
* SPDIF output source select [26] (0=host, 1=SRC)
* Not used [27]
* Record Source 0 input [29:28] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
* Record Source 1 input [31:30] (0=SPDIF in, 1=I2S in, 2=AC97 Mic, 3=AC97 PCM)
*/
/* Sample rate output control register Channel=1
* I2S Input 0 volume Right [7:0]
* I2S Input 0 volume Left [15:8]
* I2S Input 1 volume Right [23:16]
* I2S Input 1 volume Left [31:24]
*/
/* Sample rate output control register Channel=2
* SPDIF Input volume Right [23:16]
* SPDIF Input volume Left [31:24]
*/
/* Sample rate output control register Channel=3
* No used
*/
#define SPDIF_SELECT2 0x72 /* Some sort of routing. Channel_id 0 only. default = 0x0f0f003f. Analog 0x000b0000, Digital 0x0b000000 */
#define ROUTING2_FRONT_MASK 0x00010000 /* Enable for Front speakers. */
#define ROUTING2_CENTER_LFE_MASK 0x00020000 /* Enable for Center/LFE speakers. */
#define ROUTING2_REAR_MASK 0x00080000 /* Enable for Rear speakers. */
/* Audio output control
* AC97 output enable [5:0]
* I2S output enable [19:16]
* SPDIF output enable [27:24]
*/
#define UNKNOWN73 0x73 /* Unknown. Readonly. Default 0x0 */
#define CHIP_VERSION 0x74 /* P17 Chip version. Channel_id 0 only. Default 00000071 */
#define EXTENDED_INT_MASK 0x75 /* Used by both playback and capture interrupt handler */
/* Sets which Interrupts are enabled. */
/* 0x00000001 = Half period. Playback.
* 0x00000010 = Full period. Playback.
* 0x00000100 = Half buffer. Playback.
* 0x00001000 = Full buffer. Playback.
* 0x00010000 = Half buffer. Capture.
* 0x00100000 = Full buffer. Capture.
* Capture can only do 2 periods.
* 0x01000000 = End audio. Playback.
* 0x40000000 = Half buffer Playback,Caputre xrun.
* 0x80000000 = Full buffer Playback,Caputre xrun.
*/
#define EXTENDED_INT 0x76 /* Used by both playback and capture interrupt handler */
/* Shows which interrupts are active at the moment. */
/* Same bit layout as EXTENDED_INT_MASK */
#define COUNTER77 0x77 /* Counter range 0 to 0x3fffff, 192000 counts per second. */
#define COUNTER78 0x78 /* Counter range 0 to 0x3fffff, 44100 counts per second. */
#define EXTENDED_INT_TIMER 0x79 /* Channel_id 0 only. Used by both playback and capture interrupt handler */
/* Causes interrupts based on timer intervals. */
#define SPI 0x7a /* SPI: Serial Interface Register */
#define I2C_A 0x7b /* I2C Address. 32 bit */
#define I2C_0 0x7c /* I2C Data Port 0. 32 bit */
#define I2C_1 0x7d /* I2C Data Port 1. 32 bit */
#define SET_CHANNEL 0 /* Testing channel outputs 0=Front, 1=Center/LFE, 2=Unknown, 3=Rear */
#define PCM_FRONT_CHANNEL 0
#define PCM_REAR_CHANNEL 1
#define PCM_CENTER_LFE_CHANNEL 2
#define PCM_UNKNOWN_CHANNEL 3
#define CONTROL_FRONT_CHANNEL 0
#define CONTROL_REAR_CHANNEL 3
#define CONTROL_CENTER_LFE_CHANNEL 1
#define CONTROL_UNKNOWN_CHANNEL 2
typedef struct snd_ca0106_channel ca0106_channel_t;
typedef struct snd_ca0106 ca0106_t;
typedef struct snd_ca0106_pcm ca0106_pcm_t;
struct snd_ca0106_channel {
ca0106_t *emu;
int number;
int use;
void (*interrupt)(ca0106_t *emu, ca0106_channel_t *channel);
ca0106_pcm_t *epcm;
};
struct snd_ca0106_pcm {
ca0106_t *emu;
snd_pcm_substream_t *substream;
int channel_id;
unsigned short running;
};
// definition of the chip-specific record
struct snd_ca0106 {
snd_card_t *card;
struct pci_dev *pci;
unsigned long port;
struct resource *res_port;
int irq;
unsigned int revision; /* chip revision */
unsigned int serial; /* serial number */
unsigned short model; /* subsystem id */
spinlock_t emu_lock;
ac97_t *ac97;
snd_pcm_t *pcm;
ca0106_channel_t playback_channels[4];
ca0106_channel_t capture_channels[4];
u32 spdif_bits[4]; /* s/pdif out setup */
int spdif_enable;
int capture_source;
struct snd_dma_buffer buffer;
};
int __devinit snd_ca0106_mixer(ca0106_t *emu);
int __devinit snd_ca0106_proc_init(ca0106_t * emu);
unsigned int snd_ca0106_ptr_read(ca0106_t * emu,
unsigned int reg,
unsigned int chn);
void snd_ca0106_ptr_write(ca0106_t *emu,
unsigned int reg,
unsigned int chn,
unsigned int data);
sound/pci/ca0106/ca0106_main.c 0 → 100644
View file @ 5f1f8447
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
* Version: 0.0.21
*
* 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.)
*
* BUGS:
* Some stability problems when unloading the snd-ca0106 kernel module.
* --
*
* TODO:
* 4 Capture channels, only one implemented so far.
* Other capture rates apart from 48khz not implemented.
* MIDI
* --
* GENERAL INFO:
* Model: SB0310
* P17 Chip: CA0106-DAT
* AC97 Codec: STAC 9721
* ADC: Philips 1361T (Stereo 24bit)
* DAC: WM8746EDS (6-channel, 24bit, 192Khz)
*
* GENERAL INFO:
* Model: SB0410
* P17 Chip: CA0106-DAT
* AC97 Codec: None
* ADC: WM8775EDS (4 Channel)
* DAC: CS4382 (114 dB, 24-Bit, 192 kHz, 8-Channel D/A Converter with DSD Support)
* SPDIF Out control switches between Mic in and SPDIF out.
* No sound out or mic input working yet.
*
* GENERAL INFO:
* Model: SB0413
* P17 Chip: CA0106-DAT
* AC97 Codec: None.
* ADC: Unknown
* DAC: Unknown
* Trying to handle it like the SB0410.
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <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 <sound/driver.h>
#include <linux/delay.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/slab.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>
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 int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
module_param_array(index, int, NULL, 0444);
MODULE_PARM_DESC(index, "Index value for the CA0106 soundcard.");
module_param_array(id, charp, NULL, 0444);
MODULE_PARM_DESC(id, "ID string for the CA0106 soundcard.");
module_param_array(enable, bool, NULL, 0444);
MODULE_PARM_DESC(enable, "Enable the CA0106 soundcard.");
#include "ca0106.h"
typedef struct {
u32 serial;
char * name;
} ca0106_names_t;
static ca0106_names_t ca0106_chip_names[] = {
{ 0x10021102, "AudigyLS [SB0310]"} ,
{ 0x10061102, "Live! 7.1 24bit [SB0410]"} , /* New Sound Blaster Live! 7.1 24bit. This does not have an AC97. 53SB041000001 */
{ 0x10071102, "Live! 7.1 24bit [SB0413]"} , /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */
{ 0, "AudigyLS [Unknown]" }
};
/* hardware definition */
static snd_pcm_hardware_t snd_ca0106_playback_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
.rates = SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
.rate_min = 48000,
.rate_max = 192000,
.channels_min = 2, //1,
.channels_max = 2, //6,
.buffer_bytes_max = (32*1024),
.period_bytes_min = 64,
.period_bytes_max = (16*1024),
.periods_min = 2,
.periods_max = 8,
.fifo_size = 0,
};
static snd_pcm_hardware_t snd_ca0106_capture_hw = {
.info = (SNDRV_PCM_INFO_MMAP |
SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.rates = SNDRV_PCM_RATE_48000,
.rate_min = 48000,
.rate_max = 48000,
.channels_min = 2,
.channels_max = 2,
.buffer_bytes_max = (32*1024),
.period_bytes_min = 64,
.period_bytes_max = (16*1024),
.periods_min = 2,
.periods_max = 2,
.fifo_size = 0,
};
unsigned int snd_ca0106_ptr_read(ca0106_t * emu,
unsigned int reg,
unsigned int chn)
{
unsigned long flags;
unsigned int regptr, val;
regptr = (reg << 16) | chn;
spin_lock_irqsave(&emu->emu_lock, flags);
outl(regptr, emu->port + PTR);
val = inl(emu->port + DATA);
spin_unlock_irqrestore(&emu->emu_lock, flags);
return val;
}
void snd_ca0106_ptr_write(ca0106_t *emu,
unsigned int reg,
unsigned int chn,
unsigned int data)
{
unsigned int regptr;
unsigned long flags;
regptr = (reg << 16) | chn;
spin_lock_irqsave(&emu->emu_lock, flags);
outl(regptr, emu->port + PTR);
outl(data, emu->port + DATA);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
static void snd_ca0106_intr_enable(ca0106_t *emu, unsigned int intrenb)
{
unsigned long flags;
unsigned int enable;
spin_lock_irqsave(&emu->emu_lock, flags);
enable = inl(emu->port + INTE) | intrenb;
outl(enable, emu->port + INTE);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
static void snd_ca0106_pcm_free_substream(snd_pcm_runtime_t *runtime)
{
ca0106_pcm_t *epcm = runtime->private_data;
if (epcm) {
kfree(epcm);
}
}
/* open_playback callback */
static int snd_ca0106_pcm_open_playback_channel(snd_pcm_substream_t *substream, int channel_id)
{
ca0106_t *chip = snd_pcm_substream_chip(substream);
ca0106_channel_t *channel = &(chip->playback_channels[channel_id]);
ca0106_pcm_t *epcm;
snd_pcm_runtime_t *runtime = substream->runtime;
int err;
epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL);
if (epcm == NULL)
return -ENOMEM;
epcm->emu = chip;
epcm->substream = substream;
epcm->channel_id=channel_id;
runtime->private_data = epcm;
runtime->private_free = snd_ca0106_pcm_free_substream;
runtime->hw = snd_ca0106_playback_hw;
channel->emu = chip;
channel->number = channel_id;
channel->use=1;
//printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
//channel->interrupt = snd_ca0106_pcm_channel_interrupt;
channel->epcm=epcm;
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
return err;
return 0;
}
/* close callback */
static int snd_ca0106_pcm_close_playback(snd_pcm_substream_t *substream)
{
ca0106_t *chip = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
chip->playback_channels[epcm->channel_id].use=0;
/* FIXME: maybe zero others */
return 0;
}
static int snd_ca0106_pcm_open_playback_front(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_playback_channel(substream, PCM_FRONT_CHANNEL);
}
static int snd_ca0106_pcm_open_playback_center_lfe(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_playback_channel(substream, PCM_CENTER_LFE_CHANNEL);
}
static int snd_ca0106_pcm_open_playback_unknown(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_playback_channel(substream, PCM_UNKNOWN_CHANNEL);
}
static int snd_ca0106_pcm_open_playback_rear(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_playback_channel(substream, PCM_REAR_CHANNEL);
}
/* open_capture callback */
static int snd_ca0106_pcm_open_capture_channel(snd_pcm_substream_t *substream, int channel_id)
{
ca0106_t *chip = snd_pcm_substream_chip(substream);
ca0106_channel_t *channel = &(chip->capture_channels[channel_id]);
ca0106_pcm_t *epcm;
snd_pcm_runtime_t *runtime = substream->runtime;
int err;
epcm = kcalloc(1, sizeof(*epcm), GFP_KERNEL);
if (epcm == NULL) {
snd_printk("open_capture_channel: failed epcm alloc\n");
return -ENOMEM;
}
epcm->emu = chip;
epcm->substream = substream;
epcm->channel_id=channel_id;
runtime->private_data = epcm;
runtime->private_free = snd_ca0106_pcm_free_substream;
runtime->hw = snd_ca0106_capture_hw;
channel->emu = chip;
channel->number = channel_id;
channel->use=1;
//printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
//channel->interrupt = snd_ca0106_pcm_channel_interrupt;
channel->epcm=epcm;
if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
return err;
//snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, &hw_constraints_capture_period_sizes);
if ((err = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64)) < 0)
return err;
return 0;
}
/* close callback */
static int snd_ca0106_pcm_close_capture(snd_pcm_substream_t *substream)
{
ca0106_t *chip = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
chip->capture_channels[epcm->channel_id].use=0;
/* FIXME: maybe zero others */
return 0;
}
static int snd_ca0106_pcm_open_0_capture(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_capture_channel(substream, 0);
}
static int snd_ca0106_pcm_open_1_capture(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_capture_channel(substream, 1);
}
static int snd_ca0106_pcm_open_2_capture(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_capture_channel(substream, 2);
}
static int snd_ca0106_pcm_open_3_capture(snd_pcm_substream_t *substream)
{
return snd_ca0106_pcm_open_capture_channel(substream, 3);
}
/* hw_params callback */
static int snd_ca0106_pcm_hw_params_playback(snd_pcm_substream_t *substream,
snd_pcm_hw_params_t * hw_params)
{
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
}
/* hw_free callback */
static int snd_ca0106_pcm_hw_free_playback(snd_pcm_substream_t *substream)
{
return snd_pcm_lib_free_pages(substream);
}
/* hw_params callback */
static int snd_ca0106_pcm_hw_params_capture(snd_pcm_substream_t *substream,
snd_pcm_hw_params_t * hw_params)
{
return snd_pcm_lib_malloc_pages(substream,
params_buffer_bytes(hw_params));
}
/* hw_free callback */
static int snd_ca0106_pcm_hw_free_capture(snd_pcm_substream_t *substream)
{
return snd_pcm_lib_free_pages(substream);
}
/* prepare playback callback */
static int snd_ca0106_pcm_prepare_playback(snd_pcm_substream_t *substream)
{
ca0106_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
int channel = epcm->channel_id;
u32 *table_base = (u32 *)(emu->buffer.area+(8*16*channel));
u32 period_size_bytes = frames_to_bytes(runtime, runtime->period_size);
u32 hcfg_mask = HCFG_PLAYBACK_S32_LE;
u32 hcfg_set = 0x00000000;
u32 hcfg;
u32 reg40_mask = 0x30000 << (channel<<1);
u32 reg40_set = 0;
u32 reg40;
/* FIXME: Depending on mixer selection of SPDIF out or not, select the spdif rate or the DAC rate. */
u32 reg71_mask = 0x03030000 ; /* Global. Set SPDIF rate. We only support 44100 to spdif, not to DAC. */
u32 reg71_set = 0;
u32 reg71;
int i;
//snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
//snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
/* Rate can be set per channel. */
/* reg40 control host to fifo */
/* reg71 controls DAC rate. */
switch (runtime->rate) {
case 44100:
reg40_set = 0x10000 << (channel<<1);
reg71_set = 0x01010000;
break;
case 48000:
reg40_set = 0;
reg71_set = 0;
break;
case 96000:
reg40_set = 0x20000 << (channel<<1);
reg71_set = 0x02020000;
break;
case 192000:
reg40_set = 0x30000 << (channel<<1);
reg71_set = 0x03030000;
break;
default:
reg40_set = 0;
reg71_set = 0;
break;
}
/* Format is a global setting */
/* FIXME: Only let the first channel accessed set this. */
switch (runtime->format) {
case SNDRV_PCM_FORMAT_S16_LE:
hcfg_set = 0;
break;
case SNDRV_PCM_FORMAT_S32_LE:
hcfg_set = HCFG_PLAYBACK_S32_LE;
break;
default:
hcfg_set = 0;
break;
}
hcfg = inl(emu->port + HCFG) ;
hcfg = (hcfg & ~hcfg_mask) | hcfg_set;
outl(hcfg, emu->port + HCFG);
reg40 = snd_ca0106_ptr_read(emu, 0x40, 0);
reg40 = (reg40 & ~reg40_mask) | reg40_set;
snd_ca0106_ptr_write(emu, 0x40, 0, reg40);
reg71 = snd_ca0106_ptr_read(emu, 0x71, 0);
reg71 = (reg71 & ~reg71_mask) | reg71_set;
snd_ca0106_ptr_write(emu, 0x71, 0, reg71);
/* FIXME: Check emu->buffer.size before actually writing to it. */
for(i=0; i < runtime->periods; i++) {
table_base[i*2]=runtime->dma_addr+(i*period_size_bytes);
table_base[(i*2)+1]=period_size_bytes<<16;
}
snd_ca0106_ptr_write(emu, PLAYBACK_LIST_ADDR, channel, emu->buffer.addr+(8*16*channel));
snd_ca0106_ptr_write(emu, PLAYBACK_LIST_SIZE, channel, (runtime->periods - 1) << 19);
snd_ca0106_ptr_write(emu, PLAYBACK_LIST_PTR, channel, 0);
snd_ca0106_ptr_write(emu, PLAYBACK_DMA_ADDR, channel, runtime->dma_addr);
snd_ca0106_ptr_write(emu, PLAYBACK_PERIOD_SIZE, channel, frames_to_bytes(runtime, runtime->period_size)<<16); // buffer size in bytes
snd_ca0106_ptr_write(emu, PLAYBACK_POINTER, channel, 0);
snd_ca0106_ptr_write(emu, 0x07, channel, 0x0);
snd_ca0106_ptr_write(emu, 0x08, channel, 0);
snd_ca0106_ptr_write(emu, PLAYBACK_MUTE, 0x0, 0x0); /* Unmute output */
#if 0
snd_ca0106_ptr_write(emu, SPCS0, 0,
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT );
}
#endif
return 0;
}
/* prepare capture callback */
static int snd_ca0106_pcm_prepare_capture(snd_pcm_substream_t *substream)
{
ca0106_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
int channel = epcm->channel_id;
//printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1));
snd_ca0106_ptr_write(emu, 0x13, channel, 0);
snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
snd_ca0106_ptr_write(emu, CAPTURE_POINTER, channel, 0);
return 0;
}
/* trigger_playback callback */
static int snd_ca0106_pcm_trigger_playback(snd_pcm_substream_t *substream,
int cmd)
{
ca0106_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime;
ca0106_pcm_t *epcm;
int channel;
int result = 0;
struct list_head *pos;
snd_pcm_substream_t *s;
u32 basic = 0;
u32 extended = 0;
int running=0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
running=1;
break;
case SNDRV_PCM_TRIGGER_STOP:
default:
running=0;
break;
}
snd_pcm_group_for_each(pos, substream) {
s = snd_pcm_group_substream_entry(pos);
runtime = s->runtime;
epcm = runtime->private_data;
channel = epcm->channel_id;
//snd_printk("channel=%d\n",channel);
epcm->running = running;
basic |= (0x1<<channel);
extended |= (0x10<<channel);
snd_pcm_trigger_done(s, substream);
}
//snd_printk("basic=0x%x, extended=0x%x\n",basic, extended);
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) | (extended));
snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0)|(basic));
break;
case SNDRV_PCM_TRIGGER_STOP:
snd_ca0106_ptr_write(emu, BASIC_INTERRUPT, 0, snd_ca0106_ptr_read(emu, BASIC_INTERRUPT, 0) & ~(basic));
snd_ca0106_ptr_write(emu, EXTENDED_INT_MASK, 0, snd_ca0106_ptr_read(emu, EXTENDED_INT_MASK, 0) & ~(extended));
break;
default:
result = -EINVAL;
break;
}
return result;
}
/* trigger_capture callback */
static int snd_ca0106_pcm_trigger_capture(snd_pcm_substream_t *substream,
int cmd)
{
ca0106_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
int channel = epcm->channel_id;
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(snd_pcm_substream_t *substream)
{
ca0106_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
snd_pcm_uframes_t ptr, ptr1, ptr2,ptr3,ptr4 = 0;
int channel = epcm->channel_id;
if (!epcm->running)
return 0;
ptr3 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel);
ptr4 = snd_ca0106_ptr_read(emu, PLAYBACK_LIST_PTR, channel);
if (ptr3 != ptr4) ptr1 = snd_ca0106_ptr_read(emu, PLAYBACK_POINTER, channel);
ptr2 = bytes_to_frames(runtime, ptr1);
ptr2+= (ptr4 >> 3) * runtime->period_size;
ptr=ptr2;
if (ptr >= runtime->buffer_size)
ptr -= runtime->buffer_size;
//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
return ptr;
}
/* pointer_capture callback */
static snd_pcm_uframes_t
snd_ca0106_pcm_pointer_capture(snd_pcm_substream_t *substream)
{
ca0106_t *emu = snd_pcm_substream_chip(substream);
snd_pcm_runtime_t *runtime = substream->runtime;
ca0106_pcm_t *epcm = runtime->private_data;
snd_pcm_uframes_t ptr, ptr1, ptr2 = 0;
int channel = channel=epcm->channel_id;
if (!epcm->running)
return 0;
ptr1 = snd_ca0106_ptr_read(emu, CAPTURE_POINTER, channel);
ptr2 = bytes_to_frames(runtime, ptr1);
ptr=ptr2;
if (ptr >= runtime->buffer_size)
ptr -= runtime->buffer_size;
//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
return ptr;
}
/* operators */
static snd_pcm_ops_t snd_ca0106_playback_front_ops = {
.open = snd_ca0106_pcm_open_playback_front,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_playback,
.hw_free = snd_ca0106_pcm_hw_free_playback,
.prepare = snd_ca0106_pcm_prepare_playback,
.trigger = snd_ca0106_pcm_trigger_playback,
.pointer = snd_ca0106_pcm_pointer_playback,
};
static snd_pcm_ops_t snd_ca0106_capture_0_ops = {
.open = snd_ca0106_pcm_open_0_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_capture,
.hw_free = snd_ca0106_pcm_hw_free_capture,
.prepare = snd_ca0106_pcm_prepare_capture,
.trigger = snd_ca0106_pcm_trigger_capture,
.pointer = snd_ca0106_pcm_pointer_capture,
};
static snd_pcm_ops_t snd_ca0106_capture_1_ops = {
.open = snd_ca0106_pcm_open_1_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_capture,
.hw_free = snd_ca0106_pcm_hw_free_capture,
.prepare = snd_ca0106_pcm_prepare_capture,
.trigger = snd_ca0106_pcm_trigger_capture,
.pointer = snd_ca0106_pcm_pointer_capture,
};
static snd_pcm_ops_t snd_ca0106_capture_2_ops = {
.open = snd_ca0106_pcm_open_2_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_capture,
.hw_free = snd_ca0106_pcm_hw_free_capture,
.prepare = snd_ca0106_pcm_prepare_capture,
.trigger = snd_ca0106_pcm_trigger_capture,
.pointer = snd_ca0106_pcm_pointer_capture,
};
static snd_pcm_ops_t snd_ca0106_capture_3_ops = {
.open = snd_ca0106_pcm_open_3_capture,
.close = snd_ca0106_pcm_close_capture,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_capture,
.hw_free = snd_ca0106_pcm_hw_free_capture,
.prepare = snd_ca0106_pcm_prepare_capture,
.trigger = snd_ca0106_pcm_trigger_capture,
.pointer = snd_ca0106_pcm_pointer_capture,
};
static snd_pcm_ops_t snd_ca0106_playback_center_lfe_ops = {
.open = snd_ca0106_pcm_open_playback_center_lfe,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_playback,
.hw_free = snd_ca0106_pcm_hw_free_playback,
.prepare = snd_ca0106_pcm_prepare_playback,
.trigger = snd_ca0106_pcm_trigger_playback,
.pointer = snd_ca0106_pcm_pointer_playback,
};
static snd_pcm_ops_t snd_ca0106_playback_unknown_ops = {
.open = snd_ca0106_pcm_open_playback_unknown,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_playback,
.hw_free = snd_ca0106_pcm_hw_free_playback,
.prepare = snd_ca0106_pcm_prepare_playback,
.trigger = snd_ca0106_pcm_trigger_playback,
.pointer = snd_ca0106_pcm_pointer_playback,
};
static snd_pcm_ops_t snd_ca0106_playback_rear_ops = {
.open = snd_ca0106_pcm_open_playback_rear,
.close = snd_ca0106_pcm_close_playback,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = snd_ca0106_pcm_hw_params_playback,
.hw_free = snd_ca0106_pcm_hw_free_playback,
.prepare = snd_ca0106_pcm_prepare_playback,
.trigger = snd_ca0106_pcm_trigger_playback,
.pointer = snd_ca0106_pcm_pointer_playback,
};
static unsigned short snd_ca0106_ac97_read(ac97_t *ac97,
unsigned short reg)
{
ca0106_t *emu = ac97->private_data;
unsigned long flags;
unsigned short val;
spin_lock_irqsave(&emu->emu_lock, flags);
outb(reg, emu->port + AC97ADDRESS);
val = inw(emu->port + AC97DATA);
spin_unlock_irqrestore(&emu->emu_lock, flags);
return val;
}
static void snd_ca0106_ac97_write(ac97_t *ac97,
unsigned short reg, unsigned short val)
{
ca0106_t *emu = ac97->private_data;
unsigned long flags;
spin_lock_irqsave(&emu->emu_lock, flags);
outb(reg, emu->port + AC97ADDRESS);
outw(val, emu->port + AC97DATA);
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
static int snd_ca0106_ac97(ca0106_t *chip)
{
ac97_bus_t *pbus;
ac97_template_t ac97;
int err;
static ac97_bus_ops_t ops = {
.write = snd_ca0106_ac97_write,
.read = snd_ca0106_ac97_read,
};
if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &pbus)) < 0)
return err;
memset(&ac97, 0, sizeof(ac97));
ac97.private_data = chip;
return snd_ac97_mixer(pbus, &ac97, &chip->ac97);
}
static int snd_ca0106_free(ca0106_t *chip)
{
if (chip->res_port != NULL) { /* avoid access to already used hardware */
// disable interrupts
snd_ca0106_ptr_write(chip, BASIC_INTERRUPT, 0, 0);
outl(0, chip->port + INTE);
snd_ca0106_ptr_write(chip, EXTENDED_INT_MASK, 0, 0);
udelay(1000);
// disable audio
//outl(HCFG_LOCKSOUNDCACHE, chip->port + HCFG);
outl(0, chip->port + HCFG);
/* FIXME: We need to stop and DMA transfers here.
* But as I am not sure how yet, we cannot from the dma pages.
* So we can fix: snd-malloc: Memory leak? pages not freed = 8
*/
}
// release the data
#if 1
if (chip->buffer.area)
snd_dma_free_pages(&chip->buffer);
#endif
// release the i/o port
if (chip->res_port) {
release_resource(chip->res_port);
kfree_nocheck(chip->res_port);
}
// release the irq
if (chip->irq >= 0)
free_irq(chip->irq, (void *)chip);
pci_disable_device(chip->pci);
kfree(chip);
return 0;
}
static int snd_ca0106_dev_free(snd_device_t *device)
{
ca0106_t *chip = device->device_data;
return snd_ca0106_free(chip);
}
static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id,
struct pt_regs *regs)
{
unsigned int status;
ca0106_t *chip = dev_id;
int i;
int mask;
unsigned int stat76;
ca0106_channel_t *pchannel;
spin_lock(&chip->emu_lock);
status = inl(chip->port + IPR);
// call updater, unlock before it
spin_unlock(&chip->emu_lock);
if (! status)
return IRQ_NONE;
stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
//snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76);
//snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
for(i = 0; i < 4; i++) {
pchannel = &(chip->playback_channels[i]);
if(stat76 & mask) {
/* FIXME: Select the correct substream for period elapsed */
if(pchannel->use) {
snd_pcm_period_elapsed(pchannel->epcm->substream);
//printk(KERN_INFO "interrupt [%d] used\n", i);
}
}
//printk(KERN_INFO "channel=%p\n",pchannel);
//printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
mask <<= 1;
}
mask = 0x110000; /* 0x1 for one half, 0x10 for the other half period. */
for(i = 0; i < 4; i++) {
pchannel = &(chip->capture_channels[i]);
if(stat76 & mask) {
/* FIXME: Select the correct substream for period elapsed */
if(pchannel->use) {
snd_pcm_period_elapsed(pchannel->epcm->substream);
//printk(KERN_INFO "interrupt [%d] used\n", i);
}
}
//printk(KERN_INFO "channel=%p\n",pchannel);
//printk(KERN_INFO "interrupt stat76[%d] = %08x, use=%d, channel=%d\n", i, stat76, pchannel->use, pchannel->number);
mask <<= 1;
}
snd_ca0106_ptr_write(chip, EXTENDED_INT, 0, stat76);
spin_lock(&chip->emu_lock);
// acknowledge the interrupt if necessary
outl(status, chip->port+IPR);
spin_unlock(&chip->emu_lock);
return IRQ_HANDLED;
}
static void snd_ca0106_pcm_free(snd_pcm_t *pcm)
{
ca0106_t *emu = pcm->private_data;
emu->pcm = NULL;
snd_pcm_lib_preallocate_free_for_all(pcm);
}
static int __devinit snd_ca0106_pcm(ca0106_t *emu, int device, snd_pcm_t **rpcm)
{
snd_pcm_t *pcm;
snd_pcm_substream_t *substream;
int err;
if (rpcm)
*rpcm = NULL;
if ((err = snd_pcm_new(emu->card, "ca0106", device, 1, 1, &pcm)) < 0)
return err;
pcm->private_data = emu;
pcm->private_free = snd_ca0106_pcm_free;
switch (device) {
case 0:
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_front_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_0_ops);
break;
case 1:
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_rear_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_1_ops);
break;
case 2:
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_center_lfe_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_2_ops);
break;
case 3:
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_ca0106_playback_unknown_ops);
snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_ca0106_capture_3_ops);
break;
}
pcm->info_flags = 0;
pcm->dev_subclass = SNDRV_PCM_SUBCLASS_GENERIC_MIX;
strcpy(pcm->name, "CA0106");
emu->pcm = pcm;
for(substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
substream;
substream = substream->next) {
if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
64*1024, 64*1024)) < 0) /* FIXME: 32*1024 for sound buffer, between 32and64 for Periods table. */
return err;
}
for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
substream;
substream = substream->next) {
if ((err = snd_pcm_lib_preallocate_pages(substream,
SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(emu->pci),
64*1024, 64*1024)) < 0)
return err;
}
if (rpcm)
*rpcm = pcm;
return 0;
}
static int __devinit snd_ca0106_create(snd_card_t *card,
struct pci_dev *pci,
ca0106_t **rchip)
{
ca0106_t *chip;
int err;
int ch;
static snd_device_ops_t ops = {
.dev_free = snd_ca0106_dev_free,
};
*rchip = NULL;
if ((err = pci_enable_device(pci)) < 0)
return err;
if (pci_set_dma_mask(pci, 0x0fffffff) < 0 ||
pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) {
printk(KERN_ERR "error to set 28bit mask DMA\n");
pci_disable_device(pci);
return -ENXIO;
}
chip = kcalloc(1, sizeof(*chip), GFP_KERNEL);
if (chip == NULL) {
pci_disable_device(pci);
return -ENOMEM;
}
chip->card = card;
chip->pci = pci;
chip->irq = -1;
spin_lock_init(&chip->emu_lock);
chip->port = pci_resource_start(pci, 0);
if ((chip->res_port = request_region(chip->port, 0x20,
"snd_ca0106")) == NULL) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot allocate the port\n");
return -EBUSY;
}
if (request_irq(pci->irq, snd_ca0106_interrupt,
SA_INTERRUPT|SA_SHIRQ, "snd_ca0106",
(void *)chip)) {
snd_ca0106_free(chip);
printk(KERN_ERR "cannot grab irq\n");
return -EBUSY;
}
chip->irq = pci->irq;
/* This stores the periods table. */
if(snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(pci), 1024, &chip->buffer) < 0) {
snd_ca0106_free(chip);
return -ENOMEM;
}
pci_set_master(pci);
/* read revision & serial */
pci_read_config_byte(pci, PCI_REVISION_ID, (char *)&chip->revision);
pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial);
pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model);
#if 1
printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model,
chip->revision, chip->serial);
#endif
outl(0, chip->port + INTE);
/*
* Init to 0x02109204 :
* Clock accuracy = 0 (1000ppm)
* Sample Rate = 2 (48kHz)
* Audio Channel = 1 (Left of 2)
* Source Number = 0 (Unspecified)
* Generation Status = 1 (Original for Cat Code 12)
* Cat Code = 12 (Digital Signal Mixer)
* Mode = 0 (Mode 0)
* Emphasis = 0 (None)
* CP = 1 (Copyright unasserted)
* AN = 0 (Audio data)
* P = 0 (Consumer)
*/
snd_ca0106_ptr_write(chip, SPCS0, 0,
chip->spdif_bits[0] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
/* Only SPCS1 has been tested */
snd_ca0106_ptr_write(chip, SPCS1, 0,
chip->spdif_bits[1] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
snd_ca0106_ptr_write(chip, SPCS2, 0,
chip->spdif_bits[2] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
snd_ca0106_ptr_write(chip, SPCS3, 0,
chip->spdif_bits[3] =
SPCS_CLKACCY_1000PPM | SPCS_SAMPLERATE_48 |
SPCS_CHANNELNUM_LEFT | SPCS_SOURCENUM_UNSPEC |
SPCS_GENERATIONSTATUS | 0x00001200 |
0x00000000 | SPCS_EMPHASIS_NONE | SPCS_COPYRIGHT);
snd_ca0106_ptr_write(chip, PLAYBACK_MUTE, 0, 0x00fc0000);
snd_ca0106_ptr_write(chip, CAPTURE_MUTE, 0, 0x00fc0000);
/* Write 0x8000 to AC97_REC_GAIN to mute it. */
outb(AC97_REC_GAIN, chip->port + AC97ADDRESS);
outw(0x8000, chip->port + AC97DATA);
#if 0
snd_ca0106_ptr_write(chip, SPCS0, 0, 0x2108006);
snd_ca0106_ptr_write(chip, 0x42, 0, 0x2108006);
snd_ca0106_ptr_write(chip, 0x43, 0, 0x2108006);
snd_ca0106_ptr_write(chip, 0x44, 0, 0x2108006);
#endif
//snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0xf0f003f); /* OSS drivers set this. */
/* Analog or Digital output */
snd_ca0106_ptr_write(chip, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(chip, SPDIF_SELECT2, 0, 0x000b0000); /* 0x0b000000 for digital, 0x000b0000 for analog, from win2000 drivers */
chip->spdif_enable = 0; /* Set digital SPDIF output off */
chip->capture_source = 3; /* Set CAPTURE_SOURCE */
//snd_ca0106_ptr_write(chip, 0x45, 0, 0); /* Analogue out */
//snd_ca0106_ptr_write(chip, 0x45, 0, 0xf00); /* Digital out */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 0, 0x40c81000); /* goes to 0x40c80000 when doing SPDIF IN/OUT */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 1, 0xffffffff); /* (Mute) CAPTURE feedback into PLAYBACK volume. Only lower 16 bits matter. */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 2, 0x30300000); /* SPDIF IN Volume */
snd_ca0106_ptr_write(chip, CAPTURE_CONTROL, 3, 0x00700000); /* SPDIF IN Volume, 0x70 = (vol & 0x3f) | 0x40 */
snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING1, 0, 0x32765410);
snd_ca0106_ptr_write(chip, PLAYBACK_ROUTING2, 0, 0x76767676);
snd_ca0106_ptr_write(chip, CAPTURE_ROUTING1, 0, 0x32765410);
snd_ca0106_ptr_write(chip, CAPTURE_ROUTING2, 0, 0x76767676);
for(ch = 0; ch < 4; ch++) {
snd_ca0106_ptr_write(chip, CAPTURE_VOLUME1, ch, 0x30303030); /* Only high 16 bits matter */
snd_ca0106_ptr_write(chip, CAPTURE_VOLUME2, ch, 0x30303030);
//snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0x40404040); /* Mute */
//snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0x40404040); /* Mute */
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME1, ch, 0xffffffff); /* Mute */
snd_ca0106_ptr_write(chip, PLAYBACK_VOLUME2, ch, 0xffffffff); /* Mute */
}
snd_ca0106_ptr_write(chip, CAPTURE_SOURCE, 0x0, 0x333300e4); /* Select MIC, Line in, TAD in, AUX in */
chip->capture_source = 3; /* Set CAPTURE_SOURCE */
if ((chip->serial == 0x10061102) || (chip->serial == 0x10071102) ) { /* The SB0410 and SB0413 use GPIO differently. */
/* FIXME: Still need to find out what the other GPIO bits do. E.g. For digital spdif out. */
outl(0x0, chip->port+GPIO);
//outl(0x00f0e000, chip->port+GPIO); /* Analog */
outl(0x005f4300, chip->port+GPIO); /* Analog */
} else {
outl(0x0, chip->port+GPIO);
outl(0x005f03a3, chip->port+GPIO); /* Analog */
//outl(0x005f02a2, chip->port+GPIO); /* SPDIF */
}
snd_ca0106_intr_enable(chip, 0x105); /* Win2000 uses 0x1e0 */
//outl(HCFG_LOCKSOUNDCACHE|HCFG_AUDIOENABLE, chip->port+HCFG);
//outl(0x00001409, chip->port+HCFG); /* 0x1000 causes AC3 to fails. Maybe it effects 24 bit output. */
//outl(0x00000009, chip->port+HCFG);
outl(HCFG_AC97 | HCFG_AUDIOENABLE, chip->port+HCFG); /* AC97 2.0, Enable outputs. */
if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
chip, &ops)) < 0) {
snd_ca0106_free(chip);
return err;
}
*rchip = chip;
return 0;
}
static int __devinit snd_ca0106_probe(struct pci_dev *pci,
const struct pci_device_id *pci_id)
{
static int dev;
snd_card_t *card;
ca0106_t *chip;
ca0106_names_t *c;
int err;
if (dev >= SNDRV_CARDS)
return -ENODEV;
if (!enable[dev]) {
dev++;
return -ENOENT;
}
card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
if (card == NULL)
return -ENOMEM;
if ((err = snd_ca0106_create(card, pci, &chip)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 0, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 1, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 2, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((err = snd_ca0106_pcm(chip, 3, NULL)) < 0) {
snd_card_free(card);
return err;
}
if ((chip->serial != 0x10061102) && (chip->serial != 0x10071102) ) { /* The SB0410 and SB0413 do not have an ac97 chip. */
if ((err = snd_ca0106_ac97(chip)) < 0) {
snd_card_free(card);
return err;
}
}
if ((err = snd_ca0106_mixer(chip)) < 0) {
snd_card_free(card);
return err;
}
snd_ca0106_proc_init(chip);
strcpy(card->driver, "CA0106");
strcpy(card->shortname, "CA0106");
for (c=ca0106_chip_names; c->serial; c++) {
if (c->serial == chip->serial) break;
}
sprintf(card->longname, "%s at 0x%lx irq %i",
c->name, chip->port, chip->irq);
if ((err = snd_card_register(card)) < 0) {
snd_card_free(card);
return err;
}
pci_set_drvdata(pci, card);
dev++;
return 0;
}
static void __devexit snd_ca0106_remove(struct pci_dev *pci)
{
snd_card_free(pci_get_drvdata(pci));
pci_set_drvdata(pci, NULL);
}
// PCI IDs
static struct pci_device_id snd_ca0106_ids[] = {
{ 0x1102, 0x0007, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* Audigy LS or Live 24bit */
{ 0, }
};
MODULE_DEVICE_TABLE(pci, snd_ca0106_ids);
// pci_driver definition
static struct pci_driver driver = {
.name = "CA0106",
.id_table = snd_ca0106_ids,
.probe = snd_ca0106_probe,
.remove = __devexit_p(snd_ca0106_remove),
};
// initialization of the module
static int __init alsa_card_ca0106_init(void)
{
int err;
if ((err = pci_module_init(&driver)) > 0)
return err;
return 0;
}
// clean up the module
static void __exit alsa_card_ca0106_exit(void)
{
pci_unregister_driver(&driver);
}
module_init(alsa_card_ca0106_init)
module_exit(alsa_card_ca0106_exit)
sound/pci/ca0106/ca0106_mixer.c 0 → 100644
View file @ 5f1f8447
#define __NO_VERSION__
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
* Version: 0.0.16
*
* FEATURES currently supported:
* See ca0106_main.c for features.
*
* Changelog:
* Support interrupts per period.
* Removed noise from Center/LFE channel when in Analog mode.
* Rename and remove mixer controls.
* 0.0.6
* Use separate card based DMA buffer for periods table list.
* 0.0.7
* Change remove and rename ctrls into lists.
* 0.0.8
* Try to fix capture sources.
* 0.0.9
* Fix AC3 output.
* Enable S32_LE format support.
* 0.0.10
* Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
* 0.0.11
* Add Model name recognition.
* 0.0.12
* Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
* Remove redundent "voice" handling.
* 0.0.13
* Single trigger call for multi channels.
* 0.0.14
* Set limits based on what the sound card hardware can do.
* playback periods_min=2, periods_max=8
* capture hw constraints require period_size = n * 64 bytes.
* playback hw constraints require period_size = n * 64 bytes.
* 0.0.15
* Separated ca0106.c into separate functional .c files.
* 0.0.16
* Modified Copyright message.
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <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 <sound/driver.h>
#include <linux/delay.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/slab.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 "ca0106.h"
static int snd_ca0106_shared_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int snd_ca0106_shared_spdif_get(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = emu->spdif_enable;
return 0;
}
static int snd_ca0106_shared_spdif_put(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
int change = 0;
u32 mask;
val = ucontrol->value.enumerated.item[0] ;
change = (emu->spdif_enable != val);
if (change) {
emu->spdif_enable = val;
if (val == 1) {
/* Digital */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x0b000000);
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) & ~0x1000);
mask = inl(emu->port + GPIO) & ~0x101;
outl(mask, emu->port + GPIO);
} else {
/* Analog */
snd_ca0106_ptr_write(emu, SPDIF_SELECT1, 0, 0xf);
snd_ca0106_ptr_write(emu, SPDIF_SELECT2, 0, 0x000b0000);
snd_ca0106_ptr_write(emu, CAPTURE_CONTROL, 0,
snd_ca0106_ptr_read(emu, CAPTURE_CONTROL, 0) | 0x1000);
mask = inl(emu->port + GPIO) | 0x101;
outl(mask, emu->port + GPIO);
}
}
return change;
}
static snd_kcontrol_new_t snd_ca0106_shared_spdif __devinitdata =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "SPDIF Out",
.info = snd_ca0106_shared_spdif_info,
.get = snd_ca0106_shared_spdif_get,
.put = snd_ca0106_shared_spdif_put
};
static int snd_ca0106_capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
static char *texts[6] = { "SPDIF out", "i2s mixer out", "SPDIF in", "i2s in", "AC97 in", "SRC out" };
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 6;
if (uinfo->value.enumerated.item > 5)
uinfo->value.enumerated.item = 5;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int snd_ca0106_capture_source_get(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = emu->capture_source;
return 0;
}
static int snd_ca0106_capture_source_put(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int val;
int change = 0;
u32 mask;
u32 source;
val = ucontrol->value.enumerated.item[0] ;
change = (emu->capture_source != val);
if (change) {
emu->capture_source = val;
source = (val << 28) | (val << 24) | (val << 20) | (val << 16);
mask = snd_ca0106_ptr_read(emu, CAPTURE_SOURCE, 0) & 0xffff;
snd_ca0106_ptr_write(emu, CAPTURE_SOURCE, 0, source | mask);
}
return change;
}
static snd_kcontrol_new_t snd_ca0106_capture_source __devinitdata =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Capture Source",
.info = snd_ca0106_capture_source_info,
.get = snd_ca0106_capture_source_get,
.put = snd_ca0106_capture_source_put
};
static int snd_ca0106_spdif_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
uinfo->count = 1;
return 0;
}
static int snd_ca0106_spdif_get(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
ucontrol->value.iec958.status[0] = (emu->spdif_bits[idx] >> 0) & 0xff;
ucontrol->value.iec958.status[1] = (emu->spdif_bits[idx] >> 8) & 0xff;
ucontrol->value.iec958.status[2] = (emu->spdif_bits[idx] >> 16) & 0xff;
ucontrol->value.iec958.status[3] = (emu->spdif_bits[idx] >> 24) & 0xff;
return 0;
}
static int snd_ca0106_spdif_get_mask(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ucontrol->value.iec958.status[0] = 0xff;
ucontrol->value.iec958.status[1] = 0xff;
ucontrol->value.iec958.status[2] = 0xff;
ucontrol->value.iec958.status[3] = 0xff;
return 0;
}
static int snd_ca0106_spdif_put(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
int change;
unsigned int val;
val = (ucontrol->value.iec958.status[0] << 0) |
(ucontrol->value.iec958.status[1] << 8) |
(ucontrol->value.iec958.status[2] << 16) |
(ucontrol->value.iec958.status[3] << 24);
change = val != emu->spdif_bits[idx];
if (change) {
snd_ca0106_ptr_write(emu, SPCS0 + idx, 0, val);
emu->spdif_bits[idx] = val;
}
return change;
}
static snd_kcontrol_new_t snd_ca0106_spdif_mask_control =
{
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,MASK),
.count = 4,
.info = snd_ca0106_spdif_info,
.get = snd_ca0106_spdif_get_mask
};
static snd_kcontrol_new_t snd_ca0106_spdif_control =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
.count = 4,
.info = snd_ca0106_spdif_info,
.get = snd_ca0106_spdif_get,
.put = snd_ca0106_spdif_put
};
static int snd_ca0106_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t * uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 255;
return 0;
}
static int snd_ca0106_volume_get(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int value;
value = snd_ca0106_ptr_read(emu, reg, channel_id);
ucontrol->value.integer.value[0] = 0xff - ((value >> 24) & 0xff); /* Left */
ucontrol->value.integer.value[1] = 0xff - ((value >> 16) & 0xff); /* Right */
return 0;
}
static int snd_ca0106_volume_get_spdif_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_FRONT_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_spdif_center_lfe(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_CENTER_LFE_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_spdif_unknown(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_UNKNOWN_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_spdif_rear(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_REAR_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_analog_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_FRONT_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_analog_center_lfe(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_CENTER_LFE_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_analog_unknown(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_UNKNOWN_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_analog_rear(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_REAR_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_get_feedback(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = 1;
int reg = CAPTURE_CONTROL;
return snd_ca0106_volume_get(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol, int reg, int channel_id)
{
ca0106_t *emu = snd_kcontrol_chip(kcontrol);
unsigned int value;
//value = snd_ca0106_ptr_read(emu, reg, channel_id);
//value = value & 0xffff;
value = ((0xff - ucontrol->value.integer.value[0]) << 24) | ((0xff - ucontrol->value.integer.value[1]) << 16);
value = value | ((0xff - ucontrol->value.integer.value[0]) << 8) | ((0xff - ucontrol->value.integer.value[1]) );
snd_ca0106_ptr_write(emu, reg, channel_id, value);
return 1;
}
static int snd_ca0106_volume_put_spdif_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_FRONT_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_spdif_center_lfe(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_CENTER_LFE_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_spdif_unknown(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_UNKNOWN_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_spdif_rear(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_REAR_CHANNEL;
int reg = PLAYBACK_VOLUME1;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_analog_front(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_FRONT_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_analog_center_lfe(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_CENTER_LFE_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_analog_unknown(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_UNKNOWN_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_analog_rear(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = CONTROL_REAR_CHANNEL;
int reg = PLAYBACK_VOLUME2;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static int snd_ca0106_volume_put_feedback(snd_kcontrol_t * kcontrol,
snd_ctl_elem_value_t * ucontrol)
{
int channel_id = 1;
int reg = CAPTURE_CONTROL;
return snd_ca0106_volume_put(kcontrol, ucontrol, reg, channel_id);
}
static snd_kcontrol_new_t snd_ca0106_volume_control_analog_front =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Analog Front Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_analog_front,
.put = snd_ca0106_volume_put_analog_front
};
static snd_kcontrol_new_t snd_ca0106_volume_control_analog_center_lfe =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Analog Center/LFE Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_analog_center_lfe,
.put = snd_ca0106_volume_put_analog_center_lfe
};
static snd_kcontrol_new_t snd_ca0106_volume_control_analog_unknown =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Analog Unknown Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_analog_unknown,
.put = snd_ca0106_volume_put_analog_unknown
};
static snd_kcontrol_new_t snd_ca0106_volume_control_analog_rear =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Analog Rear Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_analog_rear,
.put = snd_ca0106_volume_put_analog_rear
};
static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_front =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "SPDIF Front Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_spdif_front,
.put = snd_ca0106_volume_put_spdif_front
};
static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_center_lfe =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "SPDIF Center/LFE Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_spdif_center_lfe,
.put = snd_ca0106_volume_put_spdif_center_lfe
};
static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_unknown =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "SPDIF Unknown Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_spdif_unknown,
.put = snd_ca0106_volume_put_spdif_unknown
};
static snd_kcontrol_new_t snd_ca0106_volume_control_spdif_rear =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "SPDIF Rear Volume",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_spdif_rear,
.put = snd_ca0106_volume_put_spdif_rear
};
static snd_kcontrol_new_t snd_ca0106_volume_control_feedback =
{
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "CAPTURE feedback into PLAYBACK",
.info = snd_ca0106_volume_info,
.get = snd_ca0106_volume_get_feedback,
.put = snd_ca0106_volume_put_feedback
};
static int remove_ctl(snd_card_t *card, const char *name)
{
snd_ctl_elem_id_t id;
memset(&id, 0, sizeof(id));
strcpy(id.name, name);
id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
return snd_ctl_remove_id(card, &id);
}
static snd_kcontrol_t *ctl_find(snd_card_t *card, const char *name)
{
snd_ctl_elem_id_t sid;
memset(&sid, 0, sizeof(sid));
/* FIXME: strcpy is bad. */
strcpy(sid.name, name);
sid.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
return snd_ctl_find_id(card, &sid);
}
static int rename_ctl(snd_card_t *card, const char *src, const char *dst)
{
snd_kcontrol_t *kctl = ctl_find(card, src);
if (kctl) {
strcpy(kctl->id.name, dst);
return 0;
}
return -ENOENT;
}
int __devinit snd_ca0106_mixer(ca0106_t *emu)
{
int err;
snd_kcontrol_t *kctl;
snd_card_t *card = emu->card;
char **c;
static char *ca0106_remove_ctls[] = {
"Master Mono Playback Switch",
"Master Mono Playback Volume",
"3D Control - Switch",
"3D Control Sigmatel - Depth",
"PCM Playback Switch",
"PCM Playback Volume",
"CD Playback Switch",
"CD Playback Volume",
"Phone Playback Switch",
"Phone Playback Volume",
"Video Playback Switch",
"Video Playback Volume",
"PC Speaker Playback Switch",
"PC Speaker Playback Volume",
"Mono Output Select",
"Capture Source",
"Capture Switch",
"Capture Volume",
"External Amplifier",
"Sigmatel 4-Speaker Stereo Playback Switch",
"Sigmatel Surround Phase Inversion Playback ",
NULL
};
static char *ca0106_rename_ctls[] = {
"Master Playback Switch", "Capture Switch",
"Master Playback Volume", "Capture Volume",
"Line Playback Switch", "AC97 Line Capture Switch",
"Line Playback Volume", "AC97 Line Capture Volume",
"Aux Playback Switch", "AC97 Aux Capture Switch",
"Aux Playback Volume", "AC97 Aux Capture Volume",
"Mic Playback Switch", "AC97 Mic Capture Switch",
"Mic Playback Volume", "AC97 Mic Capture Volume",
"Mic Select", "AC97 Mic Select",
"Mic Boost (+20dB)", "AC97 Mic Boost (+20dB)",
NULL
};
#if 1
for (c=ca0106_remove_ctls; *c; c++)
remove_ctl(card, *c);
for (c=ca0106_rename_ctls; *c; c += 2)
rename_ctl(card, c[0], c[1]);
#endif
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_front, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_rear, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_center_lfe, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_analog_unknown, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_front, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_rear, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_center_lfe, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_spdif_unknown, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_volume_control_feedback, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_mask_control, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_shared_spdif, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = snd_ctl_new1(&snd_ca0106_capture_source, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
if ((kctl = ctl_find(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT))) != NULL) {
/* already defined by ac97, remove it */
/* FIXME: or do we need both controls? */
remove_ctl(card, SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT));
}
if ((kctl = snd_ctl_new1(&snd_ca0106_spdif_control, emu)) == NULL)
return -ENOMEM;
if ((err = snd_ctl_add(card, kctl)))
return err;
return 0;
}
sound/pci/ca0106/ca0106_proc.c 0 → 100644
View file @ 5f1f8447
#define __NO_VERSION__
/*
* Copyright (c) 2004 James Courtier-Dutton <James@superbug.demon.co.uk>
* Driver CA0106 chips. e.g. Sound Blaster Audigy LS and Live 24bit
* Version: 0.0.17
*
* FEATURES currently supported:
* See ca0106_main.c for features.
*
* Changelog:
* Support interrupts per period.
* Removed noise from Center/LFE channel when in Analog mode.
* Rename and remove mixer controls.
* 0.0.6
* Use separate card based DMA buffer for periods table list.
* 0.0.7
* Change remove and rename ctrls into lists.
* 0.0.8
* Try to fix capture sources.
* 0.0.9
* Fix AC3 output.
* Enable S32_LE format support.
* 0.0.10
* Enable playback 48000 and 96000 rates. (Rates other that these do not work, even with "plug:front".)
* 0.0.11
* Add Model name recognition.
* 0.0.12
* Correct interrupt timing. interrupt at end of period, instead of in the middle of a playback period.
* Remove redundent "voice" handling.
* 0.0.13
* Single trigger call for multi channels.
* 0.0.14
* Set limits based on what the sound card hardware can do.
* playback periods_min=2, periods_max=8
* capture hw constraints require period_size = n * 64 bytes.
* playback hw constraints require period_size = n * 64 bytes.
* 0.0.15
* Separate ca0106.c into separate functional .c files.
* 0.0.16
* Modified Copyright message.
* 0.0.17
* Add iec958 file in proc file system to show status of SPDIF in.
*
* This code was initally based on code from ALSA's emu10k1x.c which is:
* Copyright (c) by Francisco Moraes <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 <sound/driver.h>
#include <linux/delay.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/pci.h>
#include <linux/slab.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/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" },
};
void snd_ca0106_proc_dump_iec958( snd_info_buffer_t *buffer, u32 value)
{
int i;
u32 status[4];
status[0] = value & 0xff;
status[1] = (value >> 8) & 0xff;
status[2] = (value >> 16) & 0xff;
status[3] = (value >> 24) & 0xff;
if (! (status[0] & IEC958_AES0_PROFESSIONAL)) {
/* consumer */
snd_iprintf(buffer, "Mode: consumer\n");
snd_iprintf(buffer, "Data: ");
if (!(status[0] & IEC958_AES0_NONAUDIO)) {
snd_iprintf(buffer, "audio\n");
} else {
snd_iprintf(buffer, "non-audio\n");
}
snd_iprintf(buffer, "Rate: ");
switch (status[3] & IEC958_AES3_CON_FS) {
case IEC958_AES3_CON_FS_44100:
snd_iprintf(buffer, "44100 Hz\n");
break;
case IEC958_AES3_CON_FS_48000:
snd_iprintf(buffer, "48000 Hz\n");
break;
case IEC958_AES3_CON_FS_32000:
snd_iprintf(buffer, "32000 Hz\n");
break;
default:
snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Copyright: ");
if (status[0] & IEC958_AES0_CON_NOT_COPYRIGHT) {
snd_iprintf(buffer, "permitted\n");
} else {
snd_iprintf(buffer, "protected\n");
}
snd_iprintf(buffer, "Emphasis: ");
if ((status[0] & IEC958_AES0_CON_EMPHASIS) != IEC958_AES0_CON_EMPHASIS_5015) {
snd_iprintf(buffer, "none\n");
} else {
snd_iprintf(buffer, "50/15us\n");
}
snd_iprintf(buffer, "Category: ");
for (i = 0; i < ARRAY_SIZE(snd_ca0106_con_category); i++) {
if ((status[1] & IEC958_AES1_CON_CATEGORY) == snd_ca0106_con_category[i].val) {
snd_iprintf(buffer, "%s\n", snd_ca0106_con_category[i].name);
break;
}
}
if (i >= ARRAY_SIZE(snd_ca0106_con_category)) {
snd_iprintf(buffer, "unknown 0x%x\n", status[1] & IEC958_AES1_CON_CATEGORY);
}
snd_iprintf(buffer, "Original: ");
if (status[1] & IEC958_AES1_CON_ORIGINAL) {
snd_iprintf(buffer, "original\n");
} else {
snd_iprintf(buffer, "1st generation\n");
}
snd_iprintf(buffer, "Clock: ");
switch (status[3] & IEC958_AES3_CON_CLOCK) {
case IEC958_AES3_CON_CLOCK_1000PPM:
snd_iprintf(buffer, "1000 ppm\n");
break;
case IEC958_AES3_CON_CLOCK_50PPM:
snd_iprintf(buffer, "50 ppm\n");
break;
case IEC958_AES3_CON_CLOCK_VARIABLE:
snd_iprintf(buffer, "variable pitch\n");
break;
default:
snd_iprintf(buffer, "unknown\n");
break;
}
} else {
snd_iprintf(buffer, "Mode: professional\n");
snd_iprintf(buffer, "Data: ");
if (!(status[0] & IEC958_AES0_NONAUDIO)) {
snd_iprintf(buffer, "audio\n");
} else {
snd_iprintf(buffer, "non-audio\n");
}
snd_iprintf(buffer, "Rate: ");
switch (status[0] & IEC958_AES0_PRO_FS) {
case IEC958_AES0_PRO_FS_44100:
snd_iprintf(buffer, "44100 Hz\n");
break;
case IEC958_AES0_PRO_FS_48000:
snd_iprintf(buffer, "48000 Hz\n");
break;
case IEC958_AES0_PRO_FS_32000:
snd_iprintf(buffer, "32000 Hz\n");
break;
default:
snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Rate Locked: ");
if (status[0] & IEC958_AES0_PRO_FREQ_UNLOCKED)
snd_iprintf(buffer, "no\n");
else
snd_iprintf(buffer, "yes\n");
snd_iprintf(buffer, "Emphasis: ");
switch (status[0] & IEC958_AES0_PRO_EMPHASIS) {
case IEC958_AES0_PRO_EMPHASIS_CCITT:
snd_iprintf(buffer, "CCITT J.17\n");
break;
case IEC958_AES0_PRO_EMPHASIS_NONE:
snd_iprintf(buffer, "none\n");
break;
case IEC958_AES0_PRO_EMPHASIS_5015:
snd_iprintf(buffer, "50/15us\n");
break;
case IEC958_AES0_PRO_EMPHASIS_NOTID:
default:
snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Stereophonic: ");
if ((status[1] & IEC958_AES1_PRO_MODE) == IEC958_AES1_PRO_MODE_STEREOPHONIC) {
snd_iprintf(buffer, "stereo\n");
} else {
snd_iprintf(buffer, "not indicated\n");
}
snd_iprintf(buffer, "Userbits: ");
switch (status[1] & IEC958_AES1_PRO_USERBITS) {
case IEC958_AES1_PRO_USERBITS_192:
snd_iprintf(buffer, "192bit\n");
break;
case IEC958_AES1_PRO_USERBITS_UDEF:
snd_iprintf(buffer, "user-defined\n");
break;
default:
snd_iprintf(buffer, "unkown\n");
break;
}
snd_iprintf(buffer, "Sample Bits: ");
switch (status[2] & IEC958_AES2_PRO_SBITS) {
case IEC958_AES2_PRO_SBITS_20:
snd_iprintf(buffer, "20 bit\n");
break;
case IEC958_AES2_PRO_SBITS_24:
snd_iprintf(buffer, "24 bit\n");
break;
case IEC958_AES2_PRO_SBITS_UDEF:
snd_iprintf(buffer, "user defined\n");
break;
default:
snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Word Length: ");
switch (status[2] & IEC958_AES2_PRO_WORDLEN) {
case IEC958_AES2_PRO_WORDLEN_22_18:
snd_iprintf(buffer, "22 bit or 18 bit\n");
break;
case IEC958_AES2_PRO_WORDLEN_23_19:
snd_iprintf(buffer, "23 bit or 19 bit\n");
break;
case IEC958_AES2_PRO_WORDLEN_24_20:
snd_iprintf(buffer, "24 bit or 20 bit\n");
break;
case IEC958_AES2_PRO_WORDLEN_20_16:
snd_iprintf(buffer, "20 bit or 16 bit\n");
break;
default:
snd_iprintf(buffer, "unknown\n");
break;
}
}
}
static void snd_ca0106_proc_iec958(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
u32 value;
value = snd_ca0106_ptr_read(emu, SAMPLE_RATE_TRACKER_STATUS, 0);
snd_iprintf(buffer, "Status: %s, %s, %s\n",
(value & 0x100000) ? "Rate Locked" : "Not Rate Locked",
(value & 0x200000) ? "SPDIF Locked" : "No SPDIF Lock",
(value & 0x400000) ? "Audio Valid" : "No valid audio" );
snd_iprintf(buffer, "Estimated sample rate: %u\n",
((value & 0xfffff) * 48000) / 0x8000 );
if (value & 0x200000) {
snd_iprintf(buffer, "IEC958/SPDIF input status:\n");
value = snd_ca0106_ptr_read(emu, SPDIF_INPUT_STATUS, 0);
snd_ca0106_proc_dump_iec958(buffer, value);
}
snd_iprintf(buffer, "\n");
}
static void snd_ca0106_proc_reg_write32(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
unsigned long flags;
char line[64];
u32 reg, val;
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%x %x", &reg, &val) != 2)
continue;
if ((reg < 0x40) && (reg >=0) && (val <= 0xffffffff) ) {
spin_lock_irqsave(&emu->emu_lock, flags);
outl(val, emu->port + (reg & 0xfffffffc));
spin_unlock_irqrestore(&emu->emu_lock, flags);
}
}
}
static void snd_ca0106_proc_reg_read32(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
unsigned long value;
unsigned long flags;
int i;
snd_iprintf(buffer, "Registers:\n\n");
for(i = 0; i < 0x20; i+=4) {
spin_lock_irqsave(&emu->emu_lock, flags);
value = inl(emu->port + i);
spin_unlock_irqrestore(&emu->emu_lock, flags);
snd_iprintf(buffer, "Register %02X: %08lX\n", i, value);
}
}
static void snd_ca0106_proc_reg_read16(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
unsigned int value;
unsigned long flags;
int i;
snd_iprintf(buffer, "Registers:\n\n");
for(i = 0; i < 0x20; i+=2) {
spin_lock_irqsave(&emu->emu_lock, flags);
value = inw(emu->port + i);
spin_unlock_irqrestore(&emu->emu_lock, flags);
snd_iprintf(buffer, "Register %02X: %04X\n", i, value);
}
}
static void snd_ca0106_proc_reg_read8(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
unsigned int value;
unsigned long flags;
int i;
snd_iprintf(buffer, "Registers:\n\n");
for(i = 0; i < 0x20; i+=1) {
spin_lock_irqsave(&emu->emu_lock, flags);
value = inb(emu->port + i);
spin_unlock_irqrestore(&emu->emu_lock, flags);
snd_iprintf(buffer, "Register %02X: %02X\n", i, value);
}
}
static void snd_ca0106_proc_reg_read1(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
unsigned long value;
int i,j;
snd_iprintf(buffer, "Registers\n");
for(i = 0; i < 0x40; i++) {
snd_iprintf(buffer, "%02X: ",i);
for (j = 0; j < 4; j++) {
value = snd_ca0106_ptr_read(emu, i, j);
snd_iprintf(buffer, "%08lX ", value);
}
snd_iprintf(buffer, "\n");
}
}
static void snd_ca0106_proc_reg_read2(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
unsigned long value;
int i,j;
snd_iprintf(buffer, "Registers\n");
for(i = 0x40; i < 0x80; i++) {
snd_iprintf(buffer, "%02X: ",i);
for (j = 0; j < 4; j++) {
value = snd_ca0106_ptr_read(emu, i, j);
snd_iprintf(buffer, "%08lX ", value);
}
snd_iprintf(buffer, "\n");
}
}
static void snd_ca0106_proc_reg_write(snd_info_entry_t *entry,
snd_info_buffer_t * buffer)
{
ca0106_t *emu = entry->private_data;
char line[64];
unsigned int reg, channel_id , val;
while (!snd_info_get_line(buffer, line, sizeof(line))) {
if (sscanf(line, "%x %x %x", &reg, &channel_id, &val) != 3)
continue;
if ((reg < 0x80) && (reg >=0) && (val <= 0xffffffff) && (channel_id >=0) && (channel_id <= 3) )
snd_ca0106_ptr_write(emu, reg, channel_id, val);
}
}
int __devinit snd_ca0106_proc_init(ca0106_t * emu)
{
snd_info_entry_t *entry;
if(! snd_card_proc_new(emu->card, "iec958", &entry))
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_iec958);
if(! snd_card_proc_new(emu->card, "ca0106_reg32", &entry)) {
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read32);
entry->c.text.write_size = 64;
entry->c.text.write = snd_ca0106_proc_reg_write32;
}
if(! snd_card_proc_new(emu->card, "ca0106_reg16", &entry))
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read16);
if(! snd_card_proc_new(emu->card, "ca0106_reg8", &entry))
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read8);
if(! snd_card_proc_new(emu->card, "ca0106_regs1", &entry)) {
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read1);
entry->c.text.write_size = 64;
entry->c.text.write = snd_ca0106_proc_reg_write;
// entry->private_data = emu;
}
if(! snd_card_proc_new(emu->card, "ca0106_regs2", &entry))
snd_info_set_text_ops(entry, emu, 1024, snd_ca0106_proc_reg_read2);
return 0;
}
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