Revert of StartTimestamp generated randomly in RtpSender constructor

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 87 matching lines @@
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       rtt_ms_(0) {
   // Make sure that RTCP objects are aware of our SSRC.
   uint32_t SSRC = rtp_sender_.SSRC();
   rtcp_sender_.SetSSRC(SSRC);
   SetRtcpReceiverSsrcs(SSRC);
-
-  // Make sure rtcp sender use same timestamp offset as rtp sender.
-  rtcp_sender_.SetTimestampOffset(rtp_sender_.TimestampOffset());
-
-  // Set default packet size limit.
   SetMaxTransferUnit(IP_PACKET_SIZE);
 }
 
 // Returns the number of milliseconds until the module want a worker thread
 // to call Process.
 int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
   const int64_t now = clock_->TimeInMilliseconds();
   const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
   return kRtpRtcpMaxIdleTimeProcessMs - (now - last_process_time_);
 }
@@ skipping 137 matching lines @@
 
 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
   return rtp_sender_.DeRegisterSendPayload(payload_type);
 }
 
 int8_t ModuleRtpRtcpImpl::SendPayloadType() const {
   return rtp_sender_.SendPayloadType();
 }
 
 uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
-  return rtp_sender_.TimestampOffset();
+  return rtp_sender_.StartTimestamp();
 }
 
 // Configure start timestamp, default is a random number.
 void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
-  rtcp_sender_.SetTimestampOffset(timestamp);
-  rtp_sender_.SetTimestampOffset(timestamp);
+  rtcp_sender_.SetStartTimestamp(timestamp);
+  rtp_sender_.SetStartTimestamp(timestamp, true);
 }
 
 uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
   return rtp_sender_.SequenceNumber();
 }
 
 // Set SequenceNumber, default is a random number.
 void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
   rtp_sender_.SetSequenceNumber(seq_num);
 }
 
 void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
+  SetStartTimestamp(rtp_state.start_timestamp);
   rtp_sender_.SetRtpState(rtp_state);
-  rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
 }
 
 void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
   rtp_sender_.SetRtxRtpState(rtp_state);
 }
 
 RtpState ModuleRtpRtcpImpl::GetRtpState() const {
   return rtp_sender_.GetRtpState();
 }
 
@@ skipping 50 matching lines @@
 
 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false
     if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
       LOG(LS_WARNING) << "Failed to send RTCP BYE";
     }
 
     collision_detected_ = false;
 
+    // Generate a new time_stamp if true and not configured via API
     // Generate a new SSRC for the next "call" if false
     rtp_sender_.SetSendingStatus(sending);
+    if (sending) {
+      // Make sure the RTCP sender has the same timestamp offset.
+      rtcp_sender_.SetStartTimestamp(rtp_sender_.StartTimestamp());
+    }
 
     // Make sure that RTCP objects are aware of our SSRC (it could have changed
     // Due to collision)
     uint32_t SSRC = rtp_sender_.SSRC();
     rtcp_sender_.SetSSRC(SSRC);
     SetRtcpReceiverSsrcs(SSRC);
 
     return 0;
   }
   return 0;
@@ skipping 596 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc