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>> 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 mos...

Hello everyone, As discussed on last Monthly Meeting, Monty settled on recommending the extension .v+ogg for Theora files IF it would work on Linux. Probably many have tested it already, but I haven't seen anything said publically, so for the record, v+ogg works on both Windows and Linux. Hadn't a chance to test it on Mac OS X, but I'm sure it should work there too. And Plan 9 :)

...to do when it encounters a "sound only" type of file. We also need to keep in mind that, if this spec catches on, we'll probably get some third-party developers working on stuff independently of this group. I would have NO problem with a compromise (say .ogg.vorbis or .ogg-vorbis or .vogg or .v-ogg or whatever) but we do need to cater to Windows here. Linux is a good enough operating system that it can adapt to anything that works in Windows. Let's make sure that our weakest OS link is strong enough that it doesn't kill us. (Especially because it happens to be the most popul...

> 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...

...r was found; anyway the video output is generated correctly. Using ffmpeg2theora instead, the coding progess goes in loop and we have to stop the process. As you can see below the time [1.21.34] is probably the same in seconds [81.4] where ffmpeg found the error. F:\th15 -x320 -y256 -V256 -A45 -o vogg.ogg vi1.avi Input #0, avi, from 'vi1.avi': Duration: 00:08:04.8, start: 0.000000, bitrate: 30345 kb/s Stream #0.0: Video: dvvideo, yuv420p, 720x576, 25.00 fps Stream #0.1: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s Resize: 720x576 => 320x256 Resample: 48000Hz => 44100Hz...

Hi All, I am quite new to voice encoding and to Speex library. I have some programming questions that I am not able to answer even after having read the documentation (from the start to the end an from the end to the start) and by reading the sample code (provided in the documentation) and the speexdec.c and speexenc.c programs. I am currently working on an interface between Java code and