Linux-2.6.12-rc2v2.6.12-rc2

Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.

Let it rip!

1 files changed, 345 insertions, 0 deletions

diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt
new file mode 100644
index 000000000000..5132fd95e074
--- /dev/null
+++ b/Documentation/sound/alsa/Audigy-mixer.txt
@@ -0,0 +1,345 @@
+
+                Sound Blaster Audigy mixer / default DSP code
+                ===========================================
+
+This is based on SB-Live-mixer.txt.
+
+The EMU10K2 chips have a DSP part which can be programmed to support 
+various ways of sample processing, which is described here.
+(This acticle does not deal with the overall functionality of the 
+EMU10K2 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) 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 EMU10K2 chip has an effect bus containing 64 accumulators.
+         Each of the synthesizer voices can feed its output to these accumulators
+         and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM 
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM 
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM 
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+
+This control is used to attenuate sample for LFE PCM FX-bus accumulator. 
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample 
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM 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 for
+stereo playback. The result samples are forwarded to the front DAC PCM slots 
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+
+This control is 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='Music 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='Music Capture Volume',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='Mic Playback Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to 
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD 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 Philips DAC.
+
+name='Audigy CD 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='IEC958 Optical 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 Philips DAC.
+
+name='IEC958 Optical 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='Line2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+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='Master Playback Volume',index=0
+
+This control is used to attenuate samples for front, surround, center and 
+LFE outputs.
+
+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.
+
+
+2) 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 24
+values with this mapping:
+
+         0 -  mono, A destination (FX-bus 0-63), default 0
+         1 -  mono, B destination (FX-bus 0-63), default 1
+         2 -  mono, C destination (FX-bus 0-63), default 2
+         3 -  mono, D destination (FX-bus 0-63), default 3
+         4 -  mono, E destination (FX-bus 0-63), default 0
+         5 -  mono, F destination (FX-bus 0-63), default 0
+         6 -  mono, G destination (FX-bus 0-63), default 0
+         7 -  mono, H destination (FX-bus 0-63), default 0
+         8 -  left, A destination (FX-bus 0-63), default 0
+         9 -  left, B destination (FX-bus 0-63), default 1
+        10 -  left, C destination (FX-bus 0-63), default 2
+        11 -  left, D destination (FX-bus 0-63), default 3
+        12 -  left, E destination (FX-bus 0-63), default 0
+        13 -  left, F destination (FX-bus 0-63), default 0
+        14 -  left, G destination (FX-bus 0-63), default 0
+        15 -  left, H destination (FX-bus 0-63), default 0
+        16 - right, A destination (FX-bus 0-63), default 0
+        17 - right, B destination (FX-bus 0-63), default 1
+        18 - right, C destination (FX-bus 0-63), default 2
+        19 - right, D destination (FX-bus 0-63), default 3
+        20 - right, E destination (FX-bus 0-63), default 0
+        21 - right, F destination (FX-bus 0-63), default 0
+        22 - right, G destination (FX-bus 0-63), default 0
+        23 - right, H destination (FX-bus 0-63), default 0
+
+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 -  mono, E destination attn, default 0 (mute)
+         5 -  mono, F destination attn, default 0 (mute)
+         6 -  mono, G destination attn, default 0 (mute)
+         7 -  mono, H destination attn, default 0 (mute)
+         8 -  left, A destination attn, default 255 (no attenuation)
+         9 -  left, B destination attn, default 0 (mute)
+        10 -  left, C destination attn, default 0 (mute)
+        11 -  left, D destination attn, default 0 (mute)
+        12 -  left, E destination attn, default 0 (mute)
+        13 -  left, F destination attn, default 0 (mute)
+        14 -  left, G destination attn, default 0 (mute)
+        15 -  left, H destination attn, default 0 (mute)
+        16 - right, A destination attn, default 0 (mute)
+        17 - right, B destination attn, default 255 (no attenuation)
+        18 - right, C destination attn, default 0 (mute)
+        19 - right, D destination attn, default 0 (mute)
+        20 - right, E destination attn, default 0 (mute)
+        21 - right, F destination attn, default 0 (mute)
+        22 - right, G destination attn, default 0 (mute)
+        23 - right, H 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