Speex, the open source VoIP codec, close to a 1.2 final version
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The open source community has been developing its codec for speech: Speex. Now it comes close to the final release of the 1.2 version. The Speex project is part of the Xiph Foundation, the same that already challenged the MP3 audio format with its (Ogg) Vorbis format — better, according to us.
Just like for the audio, Speex also relies on the Ogg format to store its bitstreams as files on a computer. However, as VoIP doesn’t require file storage capability, Speex doesn’t need the Vorbis audio compression. Ekiga, formely GnomeMeeting, one of the most popular VoIP client on Linux platform, is among the software that integrates this codec.
Xiph claims that Speex “has a number of features that aren’t in other codecs such as intensity stereo encoding, integration of multiple sampling rates in the same bitstream, and a variable bit rate mode”.
And in the 1.2 release beta1 of Speex, there should be a lot of improvements, say the team, such as:
- The general quality, both at the encoder level and the decoder level.
- Input/output high-pass filters improvement,
- Minor regressions in previous 1.1.x releases now fixed.
- A strange and rare instability problem with pure sinusoids has also been fixed.
- Memory use has been greatly reduced, especially for fixed-point and narrowband.
- The fixed-point narrowband encoder+decoder memory use has been cut by more than half, making it possible to fit both in less than 6 kB of RAM.
The final version sounds promising.
Oct 3, 2006 | By Nuno
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