[linux-audio-dev] Preliminary spec for OpenGraphics soundcard

[spikethehobbitmage at excite.com]

I have gained much valuable insight from reading the posts from last month on Linux Audio Dev and OpenGraphics.  I have tried to incorporate what wisdom I could.  Someone wiser than I could greatly improve on this, of course.  Hence this document.


Low to mid-range markets are saturated.  Buildable, but not likely saleable.  Top end market is pretty tight.  R&D budget for that level of gear is probably more than the video card project.  Semi-pro is more doable.

Best bet is an aggressive design to sell on performance and features.  Most advanced features can be implemented in software, so the hardware should be fairly straight-forward, but it still needs good throughput.

Target market
 Semi-pro audio (high end of target)
 High end gamers (mid range of target)
 Home theater (low end of target)
 Hobbyist
 -Hobbyist add-ons could greatly increase saleability to other markets
 US$300? price range for baseline system - this needs to be priced out
 19" rackmount boxes may cost more.  Mainly due to the sheer volume of good stuff that can be packed into them.  see below.


Features
 192kHz @ 24bit inputs and outputs (48 bit fixed point internal)
 on board resampling/mixing
  sample rate
  bit depth
# channels
  channel swap
 on board realtime compression/decompression/transcoding
 all analog I/O is in (shielded) breakout boxes
  more than one kind of box supported (including custom boxes)
  more than one box at a time supported
  breakout boxes may be up to 100m from host system
 MIDI
 5.1 optical
 ADAT
 sound samples/effects may be preloaded to card and played/mixed on demand
 full 3D audio processing
 custom codecs and effects generators may be loaded on demand
 on board speech synthesis - just another codec :)
 on board speech recognition - same :)
 IR remote control?
 watchdog timer?


Several people have voiced desire for 192kHz capability and have indicated media at this quality is available from some sources.

Native sample capabilities
 192kHz at 24 bit max - more than this would be overkill, and would hurt other capabilities without benefit
 96kHz, 48kHz, 44100Hz, 22050Hz, 11025Hz for good measure
 -there is no point resampling to a higher rate unless you are going to mix with a higher rate source. digital resampling does not increase sound quality.
 -resampling from 44100 (CD audio) and lower to 48k or higher creates artifacts (not an even multiple)
 -more processing power/bandwidth is available for effects/additional channels at lower sample speeds. halving the sample rate doubles the processing power available per sample.
 48 bit fixed point internal DSP for best accuracy
 all samples converted to 48 bit fixed point for internal processing
 -there should be no data loss from this conversion
 -if the hardware is efficient:
  -there should be negligible performance penalty for conversion
  -there should be no performance penalty for using the higher bit depth
 -simplified design if all DSP functions and codecs use the same format
 -downsampling from 48 to 24 bit for output is trivial if implemented in hardware


World clock
 world master capable
 may also use external world clock
 -card can repeat clock to host system and breakout boxes
 if no dedicated world clock wiring is desired or possible, soft world clock may be implemented
 -ie card and each peripheral generate their own clock and a periodic sync signal is provided over data channel. see below.
 -unit may not be able to run at full speed due to timing constraints/clock skew in this case. not acceptable for live work, but OK for lower markets.


Virtual back plane for maximum flexibility (any input may be mapped to any output with any filtering/effects)
 external effects box can therefore be patched in at any location, if desired
 volume controls are 10-12 bit analog inputs
  sample at 20Hz?
  volume control sampling does not need to be synchronized to any world clock
 mutes are 1 bit input arrays. same sampling as above.


I/O
 card is purely digital
 all analog and most digital I/O is in breakout boxes. see below.

Analog
 inputs and outputs are discrete from each other (no dual I/O jacks to toast a mic with)
 powered outputs are discrete from line outputs
 1/8", 1/4", RCA, balanced XLR as appropriate
 inputs
  digitally controlled analog gain or preamp for each channel
  analog prefilters to reduce digital aliasing
  digital post filter? (cropping excess digital bits is trivial)
 outputs
  digitally controlled output gain or postamp
  5th or 7th order Bessel filters standard

Digital
 optical S/PDIF in and out (5.1 audio)
 ADAT (lightpipe) in and out (8 channel, optical)
  does anybody have specs on this?
  how hard is it to get interface hardware?
  cost?
  licencing? (this is an open design, so this could be an issue)
 MIDI
 CD digital audio in/out (2 channel, 16 bit, 44100, twisted pair)
  this would be on-card, if provided
  could be used to link to other cards


Breakout Boxes
 more than one type of box should be supportable

  5 1/4" internal/external box?
   cheapest to build
   shielded for internal mounting
   line in/line out/mic/headphone/IR/MIDI
   5.1 line out?
   volume control input
   may only need 100baseT if bandwidth isn't a problem

  9" (approx) external box
   home theater/gamer
   5.1 analog and optical in/out
   5.1 analog speaker out (power amp)
   (5.1 analogs can be configured as 3 stereo pairs, or 6 mono)
   IR/MIDI/mic/headphone
   software controlled AM/FM radio tuner?
   LCD display?
   digital spectrum analyzer display?

  19" rack mount (someone with experience could probably give better layout)

   I/O patch board
    60-80 ports
     1/4", balanced XLR (with switchable phantom power), headphone
    1-2 volume controls per column
    ADAT in/out
    MIDI
    world clock in/out
    1000baseT

   Control box
    8-16 ports
     1/4", XLR (with switchable phantom power), headphone
    ADAT in/out
    MIDI
    world clock in/out
    32 volume control sliders with mutes (2 banks of 16)
    4 master volume sliders with mutes (same as above, just labelled differently?)
    knob switches? (to select option or backplane presets, etc)
     2 16 pos switches could be encoded in a byte or 4 4 pos switches, etc
    digital spectrum analyzer display? (also a programmable output)
    -this would suggest an equalizer somewhere would be in order too :)
    peak/RMS volume readouts? (which they are is a setting)
    1000baseT

 more than one box at a time should be supported
 master can command boxes to forward streams to each other directly?
 multi master?
  ie more than one card connected to a group of breakout boxes and each other
  more complex design
  allows cards to talk to each other directly or share I/O resources
 1000baseT connection
  NOT TCP/IP or IPv6 - should NOT be WAN routable
   don't need the extra overhead
   somebody is going to want to use the same switch their internet connection goes through....
 all analog audio channels are 24 bit
 volume control channels are 10-12 bit
 mute controls are bit arrays
 requires in-box microcontroller
  as much work as possible should be done in hardware
  minimal processing requirements due to simple protocol
  ECC encode/decode/fixup is a large part of workload
  no DSP capability, other than perhaps hardware filters
   any digital filtering should be done by Master unit
  simple remote commands
   box detection broadcast (true PnP)
   per channel input signal forwarding (either to output on same box, output on other box, or to Master)
   world clock mode selection
   set gain controls
   box identification
    manufacturer
    model
    serial number(optional)
    I/O count/descriptions
    read current settings
  main load is data stream
   latch and marshal input signals for transmit (controlled by world clock)
   unmarshal packets to output signal latches
    2 stage latch stepped by world clock
  lesser load is volume/mute update stream. same profile, just lower bandwidth


I/O bandwidth
 192kHz at 24bit is 576000 B/s
 48kHz at 24bit is 144000 B/s

max channels for 24 bit audio
                192kHz  48kHz
10baseT         1       6
100baseT        17      69
Firewire        69      277     (400MHz)
1000baseT       173     694

100MHz could barely handle 17 channels @ 192kHz
if you add ECC (usually a good idea) it would be even less
Firewire can handle up to 69 channels (less for ECC), but protocols/drivers are a mess
-how much does firewire hardware cost compared to 1000baseT? how to get?
-Firewire2 at 800MHz might be a possibility, but again, cost/availability?


Master Unit
 1/2 height 1/2 length card
 PCI or PCIe form factor
  PCIe
   should be main focus
    offers higher bandwidth/lower latency than PCI
    common on newer machines
    by the time this gets to market, this will be the dominant standard
   x1 should be fast enough
  PCI
   provided for legacy support
   large current install base
   limited lifetime
 1000baseT external connector
 100 or 1000baseT internal connector (depending on cost and whether the 5 1/4" box needs 1000baseT)
 world clock in/out (to connect to another card in the same computer)
 ADAT I/O?
 5.1 optical out?
 RAM
  32-64 MB (128?)
  DDR
  dual channel?
 CPU
  200MHz minimum
   can we do 300MHz, or even 400?
   overclockable with active cooling?
    3rd party fan/water cooling mounts
  dual core?
  RISC like instruction set
  48 bit fixed point DSP instructions or DSP farm
  8-32 bit integer math (48 bit integer math?)
  bitwise functions
  should be efficient for compression/decompression using various codecs
  32 bit address path so memory is upgradeable just by modifying the PCB
   I/O controls/registers should be separate bus, so memory map is linear
   minimal onboard boot loader, if required. all software is loaded from host system
  separate data and instruction L1 caches
  MMU so codecs run in own memory, don't direct have access to hardware
   necessary for buggy or untrusted codecs
  DMA to access host computer memory
  accelerated task switching
   multiple register files?
 OS
  embedded Linux to start with (at least until design is proofed)
  a true RTOS would be preferable
   multi-tasking
   SMP capable if CPU is dual core
   zero copy IO for performance
   will take a while to write :P
    maybe hack Linux? that is what it is for, after all
 on board watchdog timer?
  good if she locks up
  can be used as host computer watchdog timer as well
 hardware random # generator?
  can be used for white noise production


3 profiles of signal handling
 Live
  this is usually going from an analog input to an analog output
  highest priority in processing
  minimal buffering to reduce delay
   needs to be < 20ms, even if signal is passed through card more than once
    eg input -> card -> external effects box -> card -> output)
  this puts the highest load on the card
  critical for audiophiles
  may be important for gamers and hobbyists
  minimal importance for home theater
 Real-time
  this could be recording or playback
  precise timing is only necessary at source or output, respectively
  double buffering to improve efficiency is an asset
   even 500ms delay playing an MP3 doesn't matter. prebuffering several seconds wouldn't hurt anything either.
   video sync isn't affected if the timing signal back to the computer is generated AFTER the decompression codec output buffer
   double (or more) buffering during recording is critical to prevent data loss, especially if the audio is being compressed in real time
 Transcode
  lowest priority - this could be done entirely using time left over from the other operations
  buffering to improve efficiency is an must
  delay, clock skew, etc don't matter
  could be used with non-audio data with appropriate codecs (SETI at HOME, anyone?)


If I have missed or messed up anything, that is just me being me.  Please feel free to correct/mock/browbeat as appropriate. :)

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