[Speex-dev] Speex for sampling freq >48KHz

Jean-Marc Valin Jean-Marc.Valin at USherbrooke.ca
Mon Mar 27 04:48:27 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. 

Speex is also a lossy codec.

> And I dont think it ca regenerate a better matching waveform than
> speex. 

At bit-rates above 32 kbps, Vorbis tends to produce better results than
Speex, even for speech. The only advantages of Speex over Vorbis at
these high rates is the lower latency and lower encoding complexity.

> 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.

FLAC is entirely different. You need to choose between perfect quality
and low bit-rate. FLAC isn't better or worse than Speex. If you can't
decide between the two, then you've obviously have no idea what your
after in the first place.

> 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.

You apply entropy coding to the Speex bit-stream or you encode the Speex
error to make a lossless codec (which I already mentioned is a stupid

> Please suggest me clearly as I have very small time left to wrap up my
> work to submit.

To be honest, that's the least of my problem. If you were clear in the
first place and listened to advice I already gave, you might have been
better off already.


More information about the Speex-dev mailing list