[rtp_rtcp] Fix potentional time difference between rtp and rtcp packets.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1854 matching lines @@
 StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
   CriticalSectionScoped cs(statistics_crit_.get());
   return rtp_stats_callback_;
 }
 
 uint32_t RTPSender::BitrateSent() const {
   return total_bitrate_sent_.BitrateLast();
 }
 
 void RTPSender::SetRtpState(const RtpState& rtp_state) {
-  SetStartTimestamp(rtp_state.start_timestamp, true);
   CriticalSectionScoped lock(send_critsect_.get());
   sequence_number_ = rtp_state.sequence_number;
   sequence_number_forced_ = true;
   timestamp_ = rtp_state.timestamp;
   capture_time_ms_ = rtp_state.capture_time_ms;
   last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
   media_has_been_sent_ = rtp_state.media_has_been_sent;
 }
 
 RtpState RTPSender::GetRtpState() const {
@@ skipping 19 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc