Hi, 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. thanking you, Regards, Devilal Jean-Marc Valin <jean-marc.valin@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 > think). > 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) > bitstream. > 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 that. Jean-Marc --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Phone Calls to the US (and 30+ countries) for 2?/min or less. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060327/951326cf/attachment.htm
> 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 idea)?> 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. Jean-Marc
Hi, I chose speex initially because i had some work in VQ on speex i.e. modifying split VQ to GMM based parametric VQ and I thought If I train the GMM based VQ codebooks with audio signal and then do audio coding with speex, I probably get a better(smaller) residual signal even with speex. But I couldnt get that. I was trying to get a lossless bitstream by MUXing the speex-bitstream and the entropy encoded error signal from speex. I still have 2 months time but I need to do some useful and fundamentally strong work(even if its a small improvement). I can work day and night, But want to come up with some results. Results could be anything which make sense. If you could suggest me some work in order to improve even a small part in lossless coding, I would be grateful. I am sorry for any act of not respecting you in the past. I consider you at the top in guiding me. Thanking you, Regards, Devilal Jean-Marc Valin <Jean-Marc.Valin@USherbrooke.ca> wrote: > 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 idea)?> 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. Jean-Marc *************************************************************************** Devi Lal Sharma Final Yr. Dual Degree(Communications) IIT MADRAS Mobile: +91-9986423985(I am currently in bangalore) email: devilal_sharma@yahoo.com, devilal@gmail.com *************************************************************************** --------------------------------- Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low rates. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20060327/b3cf2a88/attachment.html