search for: subbands

Hello, I have improved quality for -l1->-l4 settings.I have properly set up the wavelet coeff threshold of the dead-zone according to the different wavelet subbands, I have then saved another 1Ko per .nhw files without noticeable difference and so I have raised again the quantization.The reconstructed images have now a better precision and still a good neatness. Update at: http://nhwcodec.blogspot.com/ Else I realize that JPEG XR (2009) and WebP (2011) are q...

http://www.subband.com server down? I want to regist MacAmp... and now, Does anybody know, free-vorbis decoder for macintosh. Kaz Shioda shioda@bgm.ne.jp --- >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-request@xiph.org' containing only the word 'unsubscribe' in

>I maintain such a page here: > >http://thebasement.org/sound/vorbis_info > >It's probably woefully incomplete. Well, there are only a few more, AFAIK: - The Quinntissential Player [Windows] http://www.quinnware.com/ - Zinf [Windows & Linux] http://www.zinf.org/ - MacAmp [Mac] http://www.subband.com/macamp/ - Audion 3 [Mac] http://www.panic.com/audion/ --- >8 ----

I write a work over Vorbis. ? ?Now I have a few questions to the function. ? ?For what is the MDCT used? ? ?For what is the FFT used? ? ?How is the expiration with the encodieren? ?In the reference to the MDCT,FFT and the hearing model . ? ?Is there a volume fragmentation in Subband as MP3? ? ?If it is possible, the answers in the detail ?Thanks ? -------------- next

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 ...

Hi again. Since mostly stupid people gave me stupid answers last time, I will try again. MP3 and Ogg Vorbis use different models in order to remove most of the bits. What is the names of these models and how do they differ? <p><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

Well, i don't know if we speak of the same ... The quality is ok, but the encoding of a PCM frame on the tested machine took 2ms, which (if the PCM frame is 7ms in length) means the machine can only encode three streams in realtime ... <p><p>Am Sa, 2003-11-22 um 03.30 schrieb Devendra Jalihal: > CELP based coders perform poorly when you mix streams. Perhaps, we need > a

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

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

Hi I have gone through the document available in the net regarding the Vorbis encoder /Decoder. Based on that i have prepared a understanding document on the encoder/decoder block. I would like to know whether my understanding of the coder is OK. If there are any other additional block /information pl. provide me with the same. Thanks and regards S.Padmashri <HR NOSHADE> <UL>

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

OK, into the nitty-gritty, albeit a high-level version. If it sounds like I'm glossing important details, you're right. This discusses only the basic DSP; precise coding, framing, sync, etc, will be in another mail. Vorbis is a hybrid transform domain general purpose audio encoder, like MPEG in some respects (it is rooted in much of the same basic theory). For the most part, the

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

> 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

On Fri, Aug 08, 2003 at 09:20:09PM -0400, Chris G (Moguta) wrote: > >I maintain such a page here: > > > >http://thebasement.org/sound/vorbis_info > > > >It's probably woefully incomplete. > > Well, there are only a few more, AFAIK: > - MacAmp [Mac] > http://www.subband.com/macamp/ > > - Audion 3 [Mac] > http://www.panic.com/audion/

On Sun, Aug 10, 2003 at 02:31:40PM -0400, jared jennings wrote: > On Fri, Aug 08, 2003 at 09:20:09PM -0400, Chris G (Moguta) wrote: > > >I maintain such a page here: > > > > > >http://thebasement.org/sound/vorbis_info > > > > > >It's probably woefully incomplete. > > > > Well, there are only a few more, AFAIK: > > - MacAmp

Hello vorbis developer! I'm not very experienced in informatics, cause i'm only an audio engineer and sorry for my broken english but i have an idea... After the filterbank the maximum aplitude in each band decides about the nessesary bits, then the psychoacoustic model decides how much of the signal can be truncated without hearable noise. What do you think about quantising the

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'

Some thoughts for the new year: 1) MDCT is good for image coding 2) image coding and audio coding are two very different things 3) combine 1 and 2 4) if a psycho model is good, after leaving out what it tells you you can without hurting quality, applying the same model should yield the same results as you got before 5) from 4: decode -> encode -> decode should result in (almost) the