http://speex.sourceforge.net (http://speex.sourceforge.net)
Speex: Free Voice Coding
The Speex project aims to build an open-source (LGPL) patent-free voice codec. Unlike other codecs like MP3 and Ogg Vorbis , Speex is specially designed for compressing voice at low bit-rates for applications such as voice over IP (VoIP). In some sense, it is meant to be complementary to the Ogg Vorbis codec.
Download (source only):
http://sourceforge.net/project/showfiles.php?group_id=46651 (http://sourceforge.net/project/showfiles.php?group_id=46651)
"I'd like to let you know that Speex 0.1.0 (http://speex.sourceforge.net (http://speex.sourceforge.net)) has been released. Speex now uses the Ogg bitstream (using libogg). That means that there is now (limited) bitstream error recovery. Also, the narrowband bit-rate has been reduced from 15.7 kbps to 15.1 kbps and the wideband bit-rate has been reduced from 31.3 kbps to 27.7 kbps. The quality remains roughly the same for both narrowband and wideband. Once again, this breaks compatibility with previous versions.
I think the quality is now pretty good for both narrowband and wideband. That means that this could be considered as the highest-quality mode for Speex and from which we could "remove bits" for lower-quality modes.
However, I'd first like to have some feedback regarding quality. More precisely:
1) Are there any applications for which the current quality is not sufficient?
2) Are there any files that are badly affected by coding (clicks, bad distortion, ...)?
For those you would like to contribute, there is now a list of tasks that need to be done:
http://sourceforge.net/pm/task.php (http://sourceforge.net/pm/task.php)
Have fun, the download page is here:
http://sourceforge.net/project/showfiles.php?group_id=46651 (http://sourceforge.net/project/showfiles.php?group_id=46651)
Jean-Marc"
could anyone please compile and upload it anywhere ?
please can you compile speex because I really want to try it and I dont know how can I compile something (sorry for my bad english I m french)
thanks in advance
Originally posted by Benjamin Lebsanft
could anyone please compile and upload it anywhere ?
The program is still pretty much streamlined for Linux. Looking at the sources, it seems to me that the encoder/decoder would only work in that OS. (But I might be mistaken...)
Besides, it's still early. It lacks functionalities like Wav reading (It can only read RAW 16bit PCM)
Hopefully Win32 interface will be developped in future, as well as .wav reading support.
Regards;
Roberto.
Why couldn't a pure voice codec perform OK at bitrates ~8kbps (or even lower)? It's been done before (voxware + others). I'm not that impressed.
For VOIP, 15.1 kbps is pretty high. Double that figure for a single conversation in full duplex, and I wonder if a typical dialup modem could simultaneously send/receive constantly without stuttering.
Originally posted by fewtch
Why couldn't a pure voice codec perform OK at bitrates ~8kbps (or even lower)? It's been done before (voxware + others). I'm not that impressed.
The issue is if the methods those codecs use are patented or not.
--
GCP
Dunno if could be, but what you think to use it into .ogm for "voice chans" ?