similar to: 48000 Hz in vorbis rc3?

Hello :-) Recently I have switched to Creative Audigy 2 ZS [SB0350] PCI audio card from SoundBlaster Live! The new audio device although using the same kernel driver have some problems with audio/video streams - sound does not resemble original at all and the video player hangs (until audio buffer is flushed I guess). Did anyone enountered similar problem? I guess there is something wrong with

Hi Julien, Quote from : http://dspguru.com/dsp/faqs/multirate/resampling "The problem is that for resampling factors close to 1.0, the interpolation factor can be quite large. For example, in the case described above of changing from the sampling rate from 48 kHz to 44.1 kHz, the ratio is only 0.91875, yet the interpolation factor is 147!" My guess is that Opus would perform similar to

On Fri, Nov 14, 2008 at 3:57 AM, Vincent Burel <vincent.burel at vb-audio.com>wrote: > > Speech compression algorithms always are tunned to particular freq, > > else they would take tons of time. That's because they use knowledge > > that speech pitch (and other params) lies in well specified regions. > > Thus if you feed algorithm with wrong freq, you'll

Ok, this is slightly offtopic, but relates to the quality of input for speex :) I'm working on echo cancellation by means of sampling the wave mix of the sound card as well as the microphone. I originally had two sound cards, which had some synchronization problems (now solved, more or less), but I have also discovered a much better solution using ASIO 2.0, which enables me to sample

hi,All, I want to know whether Opus has AEC features like Speex? Thanks 2012/10/17 <opus-request at xiph.org> > Send opus mailing list submissions to > opus at xiph.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.xiph.org/mailman/listinfo/opus > or, via email, send a message with subject or body 'help' to

Hi all, The conclusion of the discussion is that most sound cards indeed have different capture and playing frequencies for the unknown reasons. But we all know the adaptive filter of the AEC relies on the synchronization of the far-end and near-end sampling rates. Then Has anybody tried to use speex AEC in Windows system? How do you solve this problem? (I have tested speex AEC. In most

Hi, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

From: Steve Underwood <steveu at coppice.org> > It seems some cards use a PLL for their ADC, so they can lock to an > incoming SPDIF signal, but always use a local crystal clock source for > their DAC. These cards do not have their ADC and DAC synchronised. Do common on-board or PCI sound card lock to some incoming signal? Yes, there is a crystal oscillator and a PLL or divider to

The following is mostly for Nick, but perhaps there are others who might be interested in one part or another of the following... Nick, I think we're basically on the same page here. I'll first try to cover some of my basic ideas in general, then respond to your specific comments inline. For one thing, I have long believed that it is smart to keep analog audio circuits

Hi, For transforming stereo to mono, averaging is fine and that's what everybody does. For sampling rate conversion, it's another matter (too long for this email) and you should read a bit about it a perhaps grab a library that does that. As for echo cancellation, it will be less complex (and as good) on a (cleanly) down-sampled signal (and certainly not on stereo). Jean-Marc Le mardi

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

Le ven 10/01/2003 à 14:39, John Hayes a écrit : > I've been playing with speex for use in a VoIP application between PC's. One > thing I've found (correlating to the documentation) it that speex runs much > faster and produced much better output when it's fed a 32kHz signal instead > of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate > and feed

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

Hi , I have read that it is posible to encode higher sample rates like 96 khz or 192khz? and the output is 48 khz, the resample is internally.? http://wiki.xiph.org/OpusFAQ But it is possible to encode? 44.1khz. It is resampled to 48khz or I have to make the resample by myself and then encode it with opus. thnx, arctor -------------- next part -------------- An HTML attachment was scrubbed...

Hi, A question: does vorbis rc3 support 22050 Hz sample rate? rc2 used to support it, but I can't seem to make it work with rc3 :(( <p>Akos <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-request@xiph.org' containing only the word 'unsubscribe'

The encoder and decoder can handle, 8, 12, 16, 24 and 48 kHz input/output. If doesn't matter what it gets encoded to/decoded from. you can initialize a decoder at 8 kHz and it'll still decode 48 kHz audio fine (you just won't get the high frequencies obviously). For sampling rates other than 8/12/16/24/48, then you'll have to do resampling. Have a look at the speexdsp resampler if

hi, I have a small and quick question. When speex divide the UWB speech signal into High and low bands, JMV said to me in HA that : "For ultra-wideband (0-16 kHz, 32 kHz sampling), I first split the band into wideband (0-8 kHz) and "very high band" (8-16 kHz). Then, the wideband itself is split into low (0-4 kHz) and high (4-8 kHz) band. So there's a total of 3 bands

>> you are better off using the vogg orbis codec. speex is meant >> specifically for telephonic voice. it takes a single human voice and >> compresses it well. it cannot handle muliple voices or music very well. >That part is true, so of course it depends on the application. I guess I >should have added that for most applications, 16 kHz is recommended >instead of 44.1

On Sat, Jun 04, 2005 at 08:00:45AM -0700, Ralph Giles wrote: > The replay gain code has dos line endings in CVS, which causes problems > for the Sun compiler, among others. Attached is a patch for the lazy, > but it's probably easier to fix locally and commit. Now with actual patch... -r -------------- next part -------------- Index:

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to