similar to: Optimizing speex for 44.1kHz

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:

> The cost of down-sampling, if done efficiently, is probably less then > the cost difference between 32 kHz and 44.1 kHz so it's probably worth > it. If you don't care about standard sampling rate, you could even to a > 2/3 conversion which would get you 29.4 kHz... I'm curious why not just sample at a lower rate if it's just VoIP anyway? My opinion is that 44kHz

----- Original Message ----- From: "Conrad Parker" <conrad at metadecks.org> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca>; <speex-dev at xiph.org> Sent: Thursday, November 13, 2008 1:18 AM Subject: Re: [Speex-dev] SPEEX on iPhone ? > 2008/11/13 Vincent Burel

Hi all, I've got a 400MHz XScale-PXA255 board, and I want to stream voice from it over a network connection at 28.8baud. This calls for a capable voice encoder which can encode at about 24kbps. I was damn happy when I found Speex and said goodbye to MP3 :) However, i'm still a long way from realtime encoding using speexenc, is this possible? Using the fixed point math option in

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

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

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, 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

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

[Just curious, and seizing the opportunity to communicate with other folks who are doing the same kind of thing I am...] How are you measuring the latency? I tried measuring it with my program (also Win32-based, also using DirectSound[Capture]) and came up with around 130ms. To measure it, I placed the mic near a speaker to get feedback going, had my program connect to itself (local

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

This is a really good point, and definitely a recurring theme on this mailing list. :) I wonder, what are some better options for handling this issue, other than to keep saying "just use 8/16/32kHz"? - Extend Speex to support other sample rates (seems unlikely..?) - Integrate a resampling algorithm into libspeex - Maintain a list of recommended resampling libraries that work well

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

>> 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

Happy New Year from the Xiphophorus Team. It took longer than we thought, but I think everyone will agree this is our best release yet. With drastic quality improvements and new bitrate management features, this release brings us one step closer to 1.0. Along with all the lovely VBR modes you are used to, you now have millions more. oggenc's quality settings are 0-10 in increments of

Hi: I've been experimenting with streaming vorbis across my network from the windows box to an icecast2 server on my linux machine (the server I usually use is having some network issues at present). I've tested both ostream 0.7.1 and oddcast beta26. It seems that neither can encode on the fly on my P2-266 at 44100 fast enough to be able to stream it. Ostream looked the more promising

Hi, I have ported the speex codec to PocketPC. However there are some issues coming. Firstly, i am trying to use a sampling rate of 44.1KHz and 14KHz after changing some of the code. When i try to decode and listen , there is some break in sound which occurs only at the starting of the sample file. On the first run, it works fine but the break increases upto a certain level on subsequent runs .

I know the focus for Opus is low delay, but I've been watching its development with interest because of the potential for audiobook/podcast use, where latency is practically irrelevant. I hear the upcoming USAC codec will give good results for this niche (though listening test results don't seem to be available to the public yet), but I also hear it'll be extremely patent