[music-dsp] Low bitrate audio compression

[Theo]

This is probably a dumb idea but... 
For voice encoding,  would it be possible to detect the "base" frequency and the amplitude + frequency of the first 3 formants and if there is a noise component present.
Assuming the formants do not change too sudden you could do with less than 70 updates of the information per second and interpolate in-between.
The frequency information sits in a limited range and can be quite course, maybe partly be code it as offsets.
Say one byte for the "base" frequency, then one byte each for the frequencies of the formants.
Add an other 3 bytes for the volume of the formants and steal one bit for noise on/off.
That is 7 bytes per frame => @ 70 frames a second that is only 490 byte/second with room left to nibble off more.
For decoding a square/saw + noise oscillator, 3 parallel BP filters and a LP in series would do.
A overall volume level could be derivated from the formant volume data.
The result would probably sound quite "vocoded" (think ELO "mr Blue-sky" for reverence) but understand-able.

Theo

  ----- Original Message ----- 
  From: Ted David 
  To: music-dsp at aulos.calarts.edu 
  Sent: Wednesday, January 01, 2003 9:42 AM
  Subject: Re: [music-dsp] Low bitrate audio compression


  Yaakov, 
  Look at the website http://www.dvsinc.com/.  They use a technique Adaptive MultiBand Excitation (AMBE), which is a multiband vector quantization technique that achieves very low rates (e.g., 2.0, 2.4, ..., 4.8kbps -- selectable.)  Their technology is patented, and I don't know what the licensing fees are.  They sample 64kbps PCM voice and process it in 20ms frames.  Initial latency for the first frame is 60ms, but 20ms thereafter.  They have coded it for TI and Motorola DSP chips.  Their voice quality is good enough that the FAA has selected for use in the new Air Traffic Control digital communications acquisition.  Test have shown it to produce intelligible speech with BERs of up to 10%. 

  Ted David 

  Yaakov Stein wrote: 

    As usual, I only saw this thread after almost everything that can be said, has been. 
    Broadcom (a major cablemodem/DSL/VoIP chip supplier) 
    recently announced that they are now using a new compression technology 
    they call BroadVoice, mainly in order to get aroundIPR issues. 
    (They claim that they went back to techniques that 
    were patented inthe past, but all the patents have expired.) 
    It compresses music to 32Kbps and speech to 16Kbps, 
    and runs at about 10 MIPS with low delay (8 ms frame size). 
    I have ony seen one article on it in the open press, 
    and since I am more interested in the below 4 Kbps range 
    I haven't looked into it further. 

    Regarding variable speed playback, 
    when this is a design goal the best coders are those 
    based on sinusoidal transform (STC). 
    You could easily do a 8 Kbps STC, 
    but I am not sure about the IPR issues. 

    One technique that is frequently overlooked 
    is the sub-band coder, which was popular 
    for a brief period between ADPCM at 32Kbps 
    (which sounds terrible at 16Kbps) 
    and the advent of the CELP coders. 
    It will give you a nice 16Kbps, 
    but won't help with the variable speed playback 
    (although it is possibly low enough on MIPS 
    that you could afford to do a SOLA afterwards). 
    I do not know of IPR issues here - 
    this stuff has been around for years (over 17). 

    Last note - 16-20 Kbps is so easy to attain for voice 
    that you could probably make something up. 
    If the variable speed is an issue, try doing something 
    pitch synchronous, e.g. high order pitch sync LPC 
    with downsampled or ADPCM'ed residual. 

    Jonathan (Y) Stein   author at dspcsp.com 
      

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