[vorbis] Ogg Voxpop

Michael Smith msmith at xn.com.au
Mon Feb 12 19:16:24 PST 2001



> My current thought is to filter the input down to a bandwidth
> of 7KHz or 4KHz (traditional values for high and low quality
> speech), decimate the samples so that the sound is sampled
> at, say, 44/3=14.6KHz or 44/5=8.8KHz, then run it through
> the standard Vorbis encoder.  Vorbis then sees an ordinary
> 20KHz bandwidth stream that sounds like a tape recording
> running at 3 to 5 times normal speed and encodes it as
> usual.

No, vorbis would see a normal stream of data, at a sampling rate you
specify. Vorbis does NOT require that input be 44.1kHz, though that is
what is has been tuned for. Notably, pre-echo will be particularly bad
with lower sampling rates.

> 
> I checked the mailing list archives, and found an old thread
> about low bit-rate encoding that quickly degenerated into
> a highly bogus discussion of the proper way to decimate
> the sample sequence.  We don't need to do that again, so
> assume the filtering and decimation is done properly,
> is there any reason this scheme could not work?  Are
> there hooks in the Vorbis stream format to tell the decoder
> that this as been done so that it will know to play back
> slower than normal?  Do I have to write all this myself,
> or is it already in there if I just know the parameter to set?

There is no 'slower than normal'. The stream format specifies the
sampling rate, vorbis will encode and play back at that rate (well,
playback is really for the frontend to care about - on the decode side I
don't think vorbis itself needs to worry about the sampling rate at
all).

Michael

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