Preserve RTP states for restarted VideoSendStreams.
A restarted VideoSendStream would previously be completely reset,
causing gaps in sequence numbers and potentially RTP timestamps as well.
This broke SRTP which requires fairly sequential sequence numbers.
Presumably, were this sent without SRTP, we'd still have problems on the
receiving end as the corresponding receiver is unaware of this reset.
Also adding annotation to RTPSender and addressing some unlocked
access to ssrc_, ssrc_rtx_ and rtx_.
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/20819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/video_engine/vie_rtp_rtcp_impl.h b/video_engine/vie_rtp_rtcp_impl.h
index 5eec0ef..4afe1c5 100644
--- a/video_engine/vie_rtp_rtcp_impl.h
+++ b/video_engine/vie_rtp_rtcp_impl.h
@@ -46,6 +46,11 @@
const uint8_t payload_type);
virtual int SetStartSequenceNumber(const int video_channel,
uint16_t sequence_number);
+ virtual void SetRtpStateForSsrc(int video_channel,
+ uint32_t ssrc,
+ const RtpState& rtp_state) OVERRIDE;
+ virtual RtpState GetRtpStateForSsrc(int video_channel,
+ uint32_t ssrc) OVERRIDE;
virtual int SetRTCPStatus(const int video_channel,
const ViERTCPMode rtcp_mode);
virtual int GetRTCPStatus(const int video_channel,