ALSA: doc: ReSTize SB-Live-mixer document

Another simple conversion from a plain text file.
Put to cards subdirectory.

Signed-off-by: Takashi Iwai <tiwai@suse.de>

1 files changed, 0 insertions, 356 deletions

diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
deleted file mode 100644
index f4b5988f450c..000000000000
--- a/Documentation/sound/alsa/SB-Live-mixer.txt
+++ /dev/null
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-
-		Sound Blaster Live mixer / default DSP code
-		===========================================
-
-
-The EMU10K1 chips have a DSP part which can be programmed to support
-various ways of sample processing, which is described here.
-(This article does not deal with the overall functionality of the 
-EMU10K1 chips. See the manuals section for further details.)
-
-The ALSA driver programs this portion of chip by default code
-(can be altered later) which offers the following functionality:
-
-
-1) IEC958 (S/PDIF) raw PCM
---------------------------
-
-This PCM device (it's the 4th PCM device (index 3!) and first subdevice
-(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
-little endian streams without any modifications to the digital output
-(coaxial or optical). The universal interface allows the creation of up
-to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
-be easy to add support for multichannel devices to the current code,
-but the conversion routines exist only for stereo (2-channel streams)
-at the time. 
-
-Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
-
-
-2) Digital mixer controls
--------------------------
-
-These controls are built using the DSP instructions. They offer extended
-functionality. Only the default build-in code in the ALSA driver is described
-here. Note that the controls work as attenuators: the maximum value is the 
-neutral position leaving the signal unchanged. Note that if the  same destination 
-is mentioned in multiple controls, the signal is accumulated and can be wrapped 
-(set to maximal or minimal value without checking of overflow).
-
-
-Explanation of used abbreviations:
-
-DAC    - digital to analog converter
-ADC    - analog to digital converter
-I2S    - one-way three wire serial bus for digital sound by Philips Semiconductors
-         (this standard is used for connecting standalone DAC and ADC converters)
-LFE    - low frequency effects (subwoofer signal)
-AC97   - a chip containing an analog mixer, DAC and ADC converters
-IEC958 - S/PDIF
-FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
-         Each of the synthesizer voices can feed its output to these accumulators
-         and the DSP microcontroller can operate with the resulting sum.
-
-
-name='Wave Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
-The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Wave Surround Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
-The result samples are forwarded to the rear I2S DACs. These DACs operates
-separately (they are not inside the AC97 codec).
-
-name='Wave Center Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
-The result is mixed to mono signal (single channel) and forwarded to
-the ??rear?? right DAC PCM slot of the AC97 codec.
-
-name='Wave LFE Playback Volume',index=0
-
-This control is used to attenuate samples for left and right PCM FX-bus
-accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
-The result is mixed to mono signal (single channel) and forwarded to
-the ??rear?? left DAC PCM slot of the AC97 codec.
-
-name='Wave Capture Volume',index=0
-name='Wave Capture Switch',index=0
-
-These controls are used to attenuate samples for left and right PCM FX-bus
-accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Synth Playback Volume',index=0
-
-This control is used to attenuate samples for left and right MIDI FX-bus
-accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
-The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Synth Capture Volume',index=0
-name='Synth Capture Switch',index=0
-
-These controls are used to attenuate samples for left and right MIDI FX-bus
-accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Surround Playback Volume',index=0
-
-This control is used to attenuate samples for left and right rear PCM FX-bus
-accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
-The result samples are forwarded to the rear I2S DACs. These DACs operate
-separately (they are not inside the AC97 codec).
-
-name='Surround Capture Volume',index=0
-name='Surround Capture Switch',index=0
-
-These controls are used to attenuate samples for left and right rear PCM FX-bus
-accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
-The result is forwarded to the ADC capture FIFO (thus to the standard capture
-PCM device).
-
-name='Center Playback Volume',index=0
-
-This control is used to attenuate sample for center PCM FX-bus accumulator.
-ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
-to the ??rear?? right DAC PCM slot of the AC97 codec.
-
-name='LFE Playback Volume',index=0
-
-This control is used to attenuate sample for center PCM FX-bus accumulator.
-ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
-to the ??rear?? left DAC PCM slot of the AC97 codec.
-
-name='AC97 Playback Volume',index=0
-
-This control is used to attenuate samples for left and right front ADC PCM slots
-of the AC97 codec. The result samples are forwarded to the front DAC PCM
-slots of the AC97 codec.
-********************************************************************************
-*** Note: This control should be zero for the standard operations, otherwise ***
-*** a digital loopback is activated.                                         ***
-********************************************************************************
-
-name='AC97 Capture Volume',index=0
-
-This control is used to attenuate samples for left and right front ADC PCM slots
-of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
-the standard capture PCM device).
-********************************************************************************
-*** Note: This control should be 100 (maximal value), otherwise no analog    ***
-*** inputs of the AC97 codec can be captured (recorded).                     ***
-********************************************************************************
-
-name='IEC958 TTL Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 TTL
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='IEC958 TTL Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 TTL
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
-
-name='Zoom Video Playback Volume',index=0
-
-This control is used to attenuate samples from left and right zoom video
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the front DAC PCM slots of the AC97 codec.
-
-name='Zoom Video Capture Volume',index=0
-
-This control is used to attenuate samples from left and right zoom video
-digital inputs (usually used by a CDROM drive). The result samples are
-forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
-
-name='IEC958 LiveDrive Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 optical
-digital input. The result samples are forwarded to the front DAC PCM slots
-of the AC97 codec.
-
-name='IEC958 LiveDrive Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 optical
-digital inputs. The result samples are forwarded to the ADC capture FIFO
-(thus to the standard capture PCM device).
-
-name='IEC958 Coaxial Playback Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 coaxial
-digital inputs. The result samples are forwarded to the front DAC PCM slots
-of the AC97 codec.
-
-name='IEC958 Coaxial Capture Volume',index=0
-
-This control is used to attenuate samples from left and right IEC958 coaxial
-digital inputs. The result samples are forwarded to the ADC capture FIFO
-(thus to the standard capture PCM device).
-
-name='Line LiveDrive Playback Volume',index=0
-name='Line LiveDrive Playback Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the LiveDrive). The result samples are forwarded to the front
-DAC PCM slots of the AC97 codec.
-
-name='Line LiveDrive Capture Volume',index=1
-name='Line LiveDrive Capture Volume',index=1
-
-This control is used to attenuate samples from left and right I2S ADC
-inputs (on the LiveDrive). The result samples are forwarded to the ADC
-capture FIFO (thus to the standard capture PCM device).
-
-name='Tone Control - Switch',index=0
-
-This control turns the tone control on or off. The samples for front, rear
-and center / LFE outputs are affected.
-
-name='Tone Control - Bass',index=0
-
-This control sets the bass intensity. There is no neutral value!!
-When the tone control code is activated, the samples are always modified.
-The closest value to pure signal is 20.
-
-name='Tone Control - Treble',index=0
-
-This control sets the treble intensity. There is no neutral value!!
-When the tone control code is activated, the samples are always modified.
-The closest value to pure signal is 20.
-
-name='IEC958 Optical Raw Playback Switch',index=0
-
-If this switch is on, then the samples for the IEC958 (S/PDIF) digital
-output are taken only from the raw FX8010 PCM, otherwise standard front
-PCM samples are taken.
-
-name='Headphone Playback Volume',index=1
-
-This control attenuates the samples for the headphone output.
-
-name='Headphone Center Playback Switch',index=1
-
-If this switch is on, then the sample for the center PCM is put to the
-left headphone output (useful for SB Live cards without separate center/LFE
-output).
-
-name='Headphone LFE Playback Switch',index=1
-
-If this switch is on, then the sample for the center PCM is put to the
-right headphone output (useful for SB Live cards without separate center/LFE
-output).
-
-
-3) PCM stream related controls
-------------------------------
-
-name='EMU10K1 PCM Volume',index 0-31
-
-Channel volume attenuation in range 0-0xffff. The maximum value (no
-attenuation) is default. The channel mapping for three values is
-as follows:
-
-	0 - mono, default 0xffff (no attenuation)
-	1 - left, default 0xffff (no attenuation)
-	2 - right, default 0xffff (no attenuation)
-
-name='EMU10K1 PCM Send Routing',index 0-31
-
-This control specifies the destination - FX-bus accumulators. There are
-twelve values with this mapping:
-
-	 0 -  mono, A destination (FX-bus 0-15), default 0
-	 1 -  mono, B destination (FX-bus 0-15), default 1
-	 2 -  mono, C destination (FX-bus 0-15), default 2
-	 3 -  mono, D destination (FX-bus 0-15), default 3
-	 4 -  left, A destination (FX-bus 0-15), default 0
-	 5 -  left, B destination (FX-bus 0-15), default 1
-	 6 -  left, C destination (FX-bus 0-15), default 2
-	 7 -  left, D destination (FX-bus 0-15), default 3
-	 8 - right, A destination (FX-bus 0-15), default 0
-	 9 - right, B destination (FX-bus 0-15), default 1
-	10 - right, C destination (FX-bus 0-15), default 2
-	11 - right, D destination (FX-bus 0-15), default 3
-
-Don't forget that it's illegal to assign a channel to the same FX-bus accumulator 
-more than once (it means 0=0 && 1=0 is an invalid combination).
- 
-name='EMU10K1 PCM Send Volume',index 0-31
-
-It specifies the attenuation (amount) for given destination in range 0-255.
-The channel mapping is following:
-
-	 0 -  mono, A destination attn, default 255 (no attenuation)
-	 1 -  mono, B destination attn, default 255 (no attenuation)
-	 2 -  mono, C destination attn, default 0 (mute)
-	 3 -  mono, D destination attn, default 0 (mute)
-	 4 -  left, A destination attn, default 255 (no attenuation)
-	 5 -  left, B destination attn, default 0 (mute)
-	 6 -  left, C destination attn, default 0 (mute)
-	 7 -  left, D destination attn, default 0 (mute)
-	 8 - right, A destination attn, default 0 (mute)
-	 9 - right, B destination attn, default 255 (no attenuation)
-	10 - right, C destination attn, default 0 (mute)
-	11 - right, D destination attn, default 0 (mute)
-
-
-
-4) MANUALS/PATENTS:
--------------------
-
-ftp://opensource.creative.com/pub/doc
--------------------------------------
-
-        Files:
-        LM4545.pdf      AC97 Codec
-
-        m2049.pdf       The EMU10K1 Digital Audio Processor
-
-        hog63.ps        FX8010 - A DSP Chip Architecture for Audio Effects
-
-
-WIPO Patents
-------------
-        Patent numbers:
-        WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
-                        streams
-
-        WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
-
-        WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
-                        Execution and Audio Data Sequencing (Jan. 14, 1999)
-
-
-US Patents (http://www.uspto.gov/)
-----------------------------------
-
-        US 5925841      Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
-
-        US 5928342      Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
-                        with a multiport memory onto which multiple asynchronous
-                        digital sound samples can be concurrently loaded
-
-        US 5930158      Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
-
-        US 6032235      Memory initialization circuit (Tram) (Feb. 29, 2000)
-
-        US 6138207      Interpolation looping of audio samples in cache connected to    (Oct. 24, 2000)
-                        system bus with prioritization and modification of bus transfers
-                        in accordance with loop ends and minimum block sizes
-
-        US 6151670      Method for conserving memory storage using a (Nov. 21, 2000)
-                        pool of  short term memory registers
-
-        US 6195715      Interrupt control for multiple programs communicating with      (Feb. 27, 2001)
-                        a common interrupt by associating programs to GP registers,
-                        defining interrupt register, polling GP registers, and invoking
-                        callback routine associated with defined interrupt register