The 0.2 release includes dependency to the Speex (libspeex1 and the corresponding gst plugin in gstreamer0.10-plugins-good-kulve). The Speex is compiled from SVN trunk.

I made some light (just looked the numbers, no playback) performance tests. The two tables below show how much time in seconds it took to encode a 60 second audio clip (8000Hz, mono, 16bit). The first column shows the complexity used (scale is 0-10) and the rest of the columns show the encoding time with different qualities (0-10).

no VBR, no DTX, no VAD:

c/q      1       2       3       5       7
0       12       8      10      14      11
1       12       9      10      15      11
2       13       8      12      16      16
3       13      10      15      22      19
4       13      10      16      23      20

VBR, DTX, VAD:

c/q      1       2       3       5       7
0       10      10      11      14      16
1        9      10      13      14      16
2       10      12      13      15      18
3       11      13      15      19      22
4       11      13      17      20      24

The next table shows the difference between fixed point and floating point versions.

c        fixed     float
0        11        17
1        11        18
2        13        20
3        19        29

So, use the fixed point Speex API, if possible (the gst doesn’t use it). If using the floating point API, the libspeex will make the conversion.

For VoIP a good quality setting could be quality 7, complexity 2 and VBR, VAD, and DTX on.