[Asterisk-code-review] app mp3: use correct buffer size for streams (asterisk[master])

[Michael Kuron]

Michael Kuron has posted comments on this change.

Change subject: app_mp3: use correct buffer size for streams
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Patch Set 1:

8kHz is currently hard-coded in app_mp3 (that's the -r 8000 flag). I guess that code was written before Asterisk supported wide-band audio. I could try to have it automatically use the native sampling rate of the bridged channel -- if you have any suggestions what function to call to get the sampling rate of the other channel and what function to call to set the sampling rate of the current channel, that would be appreciated; if not, I'll try to figure it out myself.

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Gerrit-MessageType: comment
Gerrit-Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Michael Kuron <m.kuron at gmx.de>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Michael Kuron <m.kuron at gmx.de>
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