Add jitter logic to RtpAudioStream.
authorGerald Combs <gerald@wireshark.org>
Mon, 26 Oct 2015 15:34:58 +0000 (08:34 -0700)
committerGerald Combs <gerald@wireshark.org>
Tue, 27 Oct 2015 18:00:32 +0000 (18:00 +0000)
commit2ccb9d2d957a6c118c00a193d0fbe19712153208
tree42d36e350b002ecedea7c32fe198e1e3fa4875a8
parent25de4422c6c7bac10d0375a9ebdc2bd180dd733d
Add jitter logic to RtpAudioStream.

Copy the jitter logic from rtp_player.c to rtp_audio_stream.cpp. This
still isn't correct but the RTP player should now be complete enough to
start looking at the bug list at the top of rtp_player_dialog.cpp.

Disable timing and jitter controls while we're playing while we're here.
Fixes bug 11635.

Bug: 11635
Change-Id: Ie583ade522702cbe1bbcea4475a535caa1d74fa2
Reviewed-on: https://code.wireshark.org/review/11295
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
ui/qt/rtp_audio_stream.cpp
ui/qt/rtp_audio_stream.h
ui/qt/rtp_player_dialog.cpp
ui/qt/rtp_player_dialog.h
ui/qt/rtp_player_dialog.ui
ui/rtp_stream.h