summaryrefslogtreecommitdiffstats
path: root/Documentation/sound/alsa/SB-Live-mixer.txt
diff options
context:
space:
mode:
Diffstat (limited to 'Documentation/sound/alsa/SB-Live-mixer.txt')
-rw-r--r--Documentation/sound/alsa/SB-Live-mixer.txt356
1 files changed, 356 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
new file mode 100644
index 0000000..651adaf
--- /dev/null
+++ b/Documentation/sound/alsa/SB-Live-mixer.txt
@@ -0,0 +1,356 @@
+
+ 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 acticle 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='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
+name='Music 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
OpenPOWER on IntegriCloud