similar to: Fwd: Re: Clarification on pshycho-acoustic in Vorbis (your non-MP3 guide)

Hi, 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 <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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the

(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

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'

> 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

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

> 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

>> 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

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,

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

> 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

Dear R help-list, I am trying to calculate Monmoniers algorithm. I have 3 data sets: coordinates (coord), distance matrix (comdist) and geographical distance matrix (geodist). I consistently get the error " dimnames does not match array extent". However, when I use the function "structure" I get the following results: > str(coord) 'data.frame': 114 obs. of 3

Hello there folks, i was having a look through bits of the code to see if there was anything to do with pitch control, or perhaps i should call it speed control. For instance, say you took a track off a DVD which is 48khz, forgot to run it through sox to downsample it to 44khz, yet encoded it anyway. It will sound slower (therefore, the pitch has dropped). Whilst dj-ing (though i am no

Hi I'm trying to run an admb model from R by using the system () command. The admb model runs fine when running it from the admb command line or when using emacs. However when I try it with system() then R crashes every time. And I tried using the R command line and RGui and in both it crashes. I also tried it from different computers. What I do is I first set the directory where I keep

On 01/19/2011 06:44 PM, LiMaoquan2000 wrote: > > Hi all, > > We have discussed so many about sampling rate asynchronous (or offset) > between rendering (D/A converter) and capturing (A/D converter) of > most PC soundcards. It seems all acoustic echo cancellers, include AEC > in speex, can not deal with this trouble, because it causes a drift of > echo path and also

I can only evaluate this with my subjective point of view. I had a special test scenario doing chat with cheap webcam microphones and loudspeakers. Fraunhofers solution was the only one that could eliminate the echo. In double talk the quality gets lower but is still very good. You might want to ask Fraunhofer for a demo version to test for yourself. I have no details on the algorithms being

Hello? My name is Hyeonwoo Oh. I?m developing video conference software using speex library. Can you give me some sample codes for acoustic echo canceller and noise reduction? I tried noise reduction API, but failed. Normal encode/decode API is working well. Thank you very much for your good job. I?m very pleased with speex. Best regards, Hyeonwoo Oh

>> There is also a IEEE paper, Adaptive Sampling Rate Correction for >> Acoustic Echo Control in Voice-Over-IP, which introduced a complex >> method to estimate the frequency offset and resynchronize the signals >> using arbitrary sampling rate conversion. I wonder if it can provide >> enough performance. Because I have also designed a sampling rate >>

I read the question: http://stackoverflow.com/questions/3403152/acoustic-echo-cancellation-in-java/3403198 But my problem is that my project is using Encoding and decoding mechanism using JSpeex and I am not sure whether I can integrate the above java code with my project. So can someone tell me how can I implement Acoustic Echo Cancellation (AEC) using JSpeex(Java Port of Speex).

Hi, i have a problem in assigning 2 conditions to grep() , my data look like this: DA 24 N7 Rad= 3.4 20 Sac= 0.93 Acc= 4.76 DA 24 N7 Rad= 3.4 14 Sac= 0.65 Acc= 3.33 DA 24 N7 Rad= 3.4 3 Sac= 0.14 Acc= 0.71 DA 24 N7 Rad= 3.4 11 Sac= 0.51 Acc= 2.62 DG 23 N7 Rad= 3.4 8 Sac= 0.37 Acc= 1.91 DG 23 N7 Rad= 3.4 5 Sac= 0.23 Acc= 1.19 DG 23 N7 Rad= 3.4 0 Sac= 0.00 Acc= 0.00 DG 23 N7 Rad= 3.4 3 Sac=

Hi All, I am a research associate in the Hong Kong Polytechnic University. One of my research interests is acoustic echo cancellation. Now I meet a big problem. When I was testing my own AEC module, I found that it was almost perfect in few computers but much worse in other computers. Then I tried AEC module in speex, it was almost the same result. Then I found the reason is that sound cards of