Consolidate the three different RTP stream structs in ui/rtp_stream.h,
ui/gtk/rtp_player.c, and ui/voip_calls.c into one. Make the member names
a bit more consistent. Document what each GList contains. Use nstime_t
for timestamps since that's what we get from the frame data. Use g_new0
to initialize our structs.
Change-Id: I2b3f8f2051394a6a98a5c7bc49c117f07161d031
Reviewed-on: https://code.wireshark.org/review/5843
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Add Telephony menu items for VoIP Calls and SIP Flows. Put VoIP Calls at
the top, since that seems to be the primary item.
Add configure-time checks for QtMultimediaWidgets in anticipation of
adding a VoIP playback dialog.
Add an icon for the playback button. (Yes, I've been avoiding
GNOME-level gratuitous icons so far but this is one of the rare
occiasions where it makes sense.)
Add a help link define for the VoIP calls dialog.
Change-Id: I5d0799685c598ad9af76fe9667f8ea7d14b66050
Reviewed-on: https://code.wireshark.org/review/5674
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
We don't need to call the VoIP tap reset and draw callbacks repeately.
Do so only once from the RTP tap. Packet callbacks should return a
gboolean.
Clean up some function names and make some static.
Change-Id: I5c934ce8ce7f279861e8cc73235bbfc27d7fe622
Reviewed-on: https://code.wireshark.org/review/5396
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Gerald Combs