Steven Critchfield wrote:
As far as I know there is no code patch to be applied. The question comes basically when the recordings go off the PSTN. In the uncompressed WAV format, the audio is downshifted for playback. This means it is played at the same volume as any of the compressed formats. The question then I guess is maybe we need to be able to specify a gain setting and deal with it before we route the audio to the audio writers. But then again, my memory about the internals is fuzzy and not up to date right now as to whether that is doable. Maybe the gain setting in the uncompressed wav format should be removed.
My information that sox was involved came from Digium support which I now see isn't involved at all. I'm now digging through the Asterisk source code a bit to see if I can spot anything.
One thing I noticed is that the voicemail function uses the ast_play_and_record(...) in app.c and that the record application in
apps/app_record.c use similar but not common code. Unless I'm missing
something (very possible), if the low volume problem exists in one of
these functions it should happen in both (they both do eventually call
ast_writestream in file.c).
I haven't tried it yet - can anyone confirm (or deny) that the low volume problem not only exists with voicemail, but the record app as well?
I don't see any special gain normalization code in the record app done as a post process that would account for the results being different.
Steve
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