commit 09e8c47ee5c58e5e86b09dc1950f8d3f9f24cd9f
author xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed Jul 31 16:30:19 2013
committer xians@webrtc.org <xians@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Wed Jul 31 16:30:19 2013
tree 6af9b069aca31d5ed504522a631f7592bf89a8a7
parent 8fff1f065ea9d25970c3839294acdd606a5ddf22

Merge r4374 from stable to trunk.

r4374 was mistakenly committed to stable, so this is to re-merge back to trunk.

Store the sequence number in StopSend() and resume it in StartSend().

When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().Store the sequence number in StopSend() and resume it in StartSend().

When restarting the microphone device, we call StopSend() first, then
StartSend() later. Since we reset sequence number in StopSend(), it sometimes
causes libSRTP to complain about packets being replayed. Libjingle work around
it by caching the sequence number in WebRtcVoiceEngine.cc, and call
SetInitSequenceNumber() to resume the sequence number before StartSend().

This patch fixes this problem by storing the sequence number in StopSend(), and
resume it in StartSend(). So that we can remove the workaround in libjingle.

BUG=2102
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1922004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4451 4adac7df-926f-26a2-2b94-8c16560cd09d