[Speex-dev] Speex for sampling freq >48KHz
devilal_sharma at yahoo.com
Mon Mar 27 04:12:49 PST 2006
I have one doubt again, that is Vorbis use DCT/MDCT based algorithm and also use psychoacoustic model so this is lossy codec. And I dont think it ca regenerate a better matching waveform than speex.
Then there comes FLAC which is the perfect answer to my question, I suppose. But my concern is this that FLAC use simple prediction algorithm and doesnt use any CELP based algo which could have model the waveform coding by having a large codebook and comparing the residual signal and selecting the codebook index.
For this, shall I start understanding and modifying FLAC itself in case I need to do something for lossless coding or I can try on Speex and than apply entropy coding.
I am getting quite good(comparable) results for audio signal(44.1KHz) if use speex and separate entropy coding.
Please suggest me clearly as I have very small time left to wrap up my work to submit.
Jean-Marc Valin <jean-marc.valin at usherbrooke.ca> wrote: > I was just trying to use speex for sampling frequency >48KHz. In the
> original Speex-1.0.5 its restricted only upto 48KHz. I tired to modify
> it by changing the boundary conditions( the error conditions, i.e. if
> sampling freq >48KHz, it gives error) in /src/speexenc.c and then it
> atleast doesnt give the error, there is flow in decoding or encoding(i
> I suspect there are other constraints in the encoder and decoder to
> modify other than this to make it work properly.
> At this point, I am not getting the proper decoded(reconstructed)
> Can you please suggest me the places where I should make changes to
> incorporate this requirement.
OK, I'm not sure what exactly you're trying to achieve, but I can't see
any sane reason to use Speex at > 48 kHz sampling. I already think 44.1
kHz is a bad idea. If you want high-fidelity, use Vorbis, not Speex. If
you want decent quality speech, use Speex at 16 kHz or something like
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