similar to: Psycho Acoustic models i Speech Coding

(This is almost out of topic but anyway...) It is surprising how little research effort have been put into psy-acou models for CELP. The basic problem lies in that it is not easy to alter the LP model without distroying the minimum-phase property (ie. the stability of the predictor). That leaves us with psy-acou modelling of the noise-part only. However, my own research is in constrained

> Does anyone have an idea about the possibility to apply psychoacoustic > models as the ones in mp3 or AAC to a CELP coder? Thanks! > /Pontus This is (sort of) done in the decoder with the optional perceptual filtering. Speex tries to shape the noise so that it sounds more pleasant, I believe. To really use perceptual coding, one would require fine granular control over quantization

I wonder what psycho-acoustic model is used in ogg vorbis. Did they create their own model (this requires big listening tests) ? Or is it something that was already there, like gpsycho ? <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org'

Hi everyone, For those who like to play with experimental stuff, I've got something new to test. If you compile with --enable-vorbis-psy , Speex uses a variant of the Vorbis psycho-acoustic model to shape the coding noise. So far, I've obtained a slight increase in quality, but I'm interested in feedback from others. This still needs a lot of tuning and has received only a minimal

Happy new year all! I have had a few issues with this sort of stuff- you could adjust NB_ENC_STACK or SB_ENC_STACK (which ever you may be fiddling with) and see if the problem goes away- could be the allocation for the vorbis psy tables takes up more space than the current stack size allows (not sure how allocated? Could be your settings for supporting alloca/etc. are set different than jm,

> When enabling and compiling this with mingw on Win32, my debugger > complains about heap overruns when calling speex_encoder_destroy(). This > could be a mingw issue though, as I also found a stackalignment bug which > prevents me from _USING_SSE (apparantly -mpreferred-stack-size is just > "prefered" and therefore ignored... *Sigh*). Just so I understand, the

> Hi everyone, > > For those who like to play with experimental stuff, I've got something > new to test. If you compile with --enable-vorbis-psy , Speex uses a > variant of the Vorbis psycho-acoustic model to shape the coding noise. > So far, I've obtained a slight increase in quality, but I'm interested > in feedback from others. This still needs a lot of tuning

Hi Tom, None of the new code depends on the stack allocator, but I just fixed an off-by-one overflow on the heap. Maybe that was the error and your debugger is just reporting a wrong offset. Could you try what's in the current svn (you were using svn already, right)? Jean-Marc > I have had a few issues with this sort of stuff- you could adjust NB_ENC_STACK > or SB_ENC_STACK (which

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

>> When enabling and compiling this with mingw on Win32, my debugger >> complains about heap overruns when calling speex_encoder_destroy(). This >> could be a mingw issue though, as I also found a stackalignment bug which >> prevents me from _USING_SSE (apparantly -mpreferred-stack-size is just >> "prefered" and therefore ignored... *Sigh*). > > Just

CELP based coders perform poorly when you mix streams. Perhaps, we need a transform code or a subband code at somewhat higher rate 12-16 kbps that will retain the quality after transcoding and mixing. Is the speex ultra-wideband based on subband coding? -Devendra Jalihal <p>On Fri, 21 Nov 2003, Ulrich B. Staudinger wrote: > Hi, > been silent in this discussion for quite long ...

There's no perfect solution to the multiple client problem. Each approach has advantages and drawbacks: 1) Mixing at the server - Allows a constant bandwidth for every client - Allows compatibility with regular VoIP prones - Requires transcoding, even when only on person is talking - Higher bit-rate required for the general case (one speaker is talking) 2) Sending multiple streams - Possible

Hi all, This should please all those who want to know more about Speex or Vorbis. Monty and I have just finished writing this paper: "Improved Noise Weighting in CELP Coding of Speech - Applying the Vorbis Psychoacoustic Model To Speex" which you can get at: http://people.xiph.org/~jm/papers/aes120_speex_vorbis.pdf It's probably the best description of the Speex encoder to data and

Hi all, This should please all those who want to know more about Speex or Vorbis. Monty and I have just finished writing this paper: "Improved Noise Weighting in CELP Coding of Speech - Applying the Vorbis Psychoacoustic Model To Speex" which you can get at: http://people.xiph.org/~jm/papers/aes120_speex_vorbis.pdf It's probably the best description of the Speex encoder to data and

Hello, the voice-communication TeamSpeak (www.teamspeak.org) is currently testing a version that supports speex codecs. The quality for high bitrates is quite good. BUT, the low-bandwidth speex codecs that are currently used arent very good. What I did to find this out: I comprared a speex AVB with 6.3 KBit/sec (total, overhead for packets and stuff included) and the 6.3 Kbit/sec Celp Codec

Hi, We would like to announce the first release of the Speex project. Speex (http://speex.sourceforge.net) is an open-source (LGPL), patent-free compression format allowing an alternative to expensive proprietary codecs. Unlike Ogg Vorbis which compresses general audio, Speex is designed especially for speech. For that reason, Speex is meant to be a complement to Vorbis. Since it is specialized

https://bugs.freedesktop.org/show_bug.cgi?id=62835 Priority: medium Bug ID: 62835 Assignee: nouveau at lists.freedesktop.org Summary: Screen freezes after resume from s2ram QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: pontus.fuchs at gmail.com

hi all, thanks Jean for the codec. I am new to speech coding. Am interested to implement speex on a TI dsp, say C54. just downloaded speex v 1.1.6. when I removed optimisation flags for win32 in libspeex.dsp, I get a slightly different output for decoding using speexdec.exe (debug version). no prob for speexenc.exe except for some differences in the ogg header. for anyone interested, i

Hi. I'm an undergraduate linguistics major and computer science minor at the University of Colorado in Boulder, and am taking a couple classes this semester which give me the opportunity to do a research project - one on introductory acoustics in the physics department, and one in the linguistics department on phonetics and phonology. I've got an idea, but I'd like to hear from anyone

https://bugs.freedesktop.org/show_bug.cgi?id=59057 Priority: medium Bug ID: 59057 Assignee: nouveau at lists.freedesktop.org Summary: Black screen after resume from s2ram QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: pontus.fuchs at gmail.com