Wireshark-commits: [Wireshark-commits] rev 18588: /trunk/plugins/listen_rtp/ /trunk/plugins/listen_
http://anonsvn.wireshark.org/viewvc/viewvc.cgi?view=rev&revision=18588
User: etxrab
Date: 2006/06/27 09:44 PM
Log:
From Alejandro Vaquero:
The "listen_rtp" plugin that allows to listen audio RTP
conversations.
The plugin is integrated to the "Voip Calls" feature. There is a new
"listen" button in the "Voip Calls" that once the calls are selected and
the "listen" is clicked, a new window will open. In this window you can
change the simulated jitter buffer to be used for decoding the RTP
packets. In this first implementation, only a static jitter buffer can
be simulated. Then pressing "decode" will decode all the RTP and
generate the graphical view of the audio channels. From there you can
select up to two channels at the same time (to be played in the left and
right channels) and then play, pause, stop,etc.....ok, the attached
screenshot it is self explanatory (hopefully).
The only codecs available from now are G711u and G711a law.
The PortAudio (www.portaudio.com) library is used to play audio.
This is an open source cross-platform Audio library. I have tested this
on Windows XP and in a linux FC4.
NOTE: only the plugin soure files checked in at this time.
Directory: /trunk/plugins/listen_rtp/codecs/G711a/
Changes Path Action
+18 -0 G711adecode.c Added
+3 -0 G711adecode.h Added
+36 -0 G711atable.h Added
Directory: /trunk/plugins/listen_rtp/codecs/G711u/
Changes Path Action
+18 -0 G711udecode.c Added
+3 -0 G711udecode.h Added
+36 -0 G711utable.h Added
Directory: /trunk/plugins/listen_rtp/
Changes Path Action
+5 -0 AUTHORS Added
+54 -0 Makefile.am Added
+53 -0 Makefile.nmake Added
+1 -0 NEWS Added
+49 -0 listen_rtp_plugin.c Added
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