StartTimestamp generated randomly in RtpSender constructor

webrtc/modules/rtp_rtcp/source/rtcp_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 143 matching lines @@
     : audio_(audio),
       clock_(clock),
       random_(clock_->TimeInMicroseconds()),
       method_(RtcpMode::kOff),
       event_log_(event_log),
       transport_(outgoing_transport),
       using_nack_(false),
       sending_(false),
       remb_enabled_(false),
       next_time_to_send_rtcp_(0),
-      start_timestamp_(0),
+      timestamp_offset_(0),
       last_rtp_timestamp_(0),
       last_frame_capture_time_ms_(-1),
       ssrc_(0),
       remote_ssrc_(0),
       receive_statistics_(receive_statistics),
 
       sequence_number_fir_(0),
 
       remb_bitrate_(0),
 
@@ skipping 106 matching lines @@
     SetFlag(RTCPPacketType::kRtcpTmmbr, false);
   } else {
     ConsumeFlag(RTCPPacketType::kRtcpTmmbr, true);
   }
 }
 
 void RTCPSender::SetMaxPayloadLength(size_t max_payload_length) {
   max_payload_length_ = max_payload_length;
 }
 
-void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
+void RTCPSender::SetTimestampOffset(uint32_t timestamp_offset) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
-  start_timestamp_ = start_timestamp;
+  timestamp_offset_ = timestamp_offset;
 }
 
 void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
                                 int64_t capture_time_ms) {
   rtc::CritScope lock(&critical_section_rtcp_sender_);
   last_rtp_timestamp_ = rtp_timestamp;
   if (capture_time_ms < 0) {
     // We don't currently get a capture time from VoiceEngine.
     last_frame_capture_time_ms_ = clock_->TimeInMilliseconds();
   } else {
@@ skipping 133 matching lines @@
 }
 
 std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
   // Timestamp shouldn't be estimated before first media frame.
   RTC_DCHECK_GE(last_frame_capture_time_ms_, 0);
   // The timestamp of this RTCP packet should be estimated as the timestamp of
   // the frame being captured at this moment. We are calculating that
   // timestamp as the last frame's timestamp + the time since the last frame
   // was captured.
   uint32_t rtp_timestamp =
-      start_timestamp_ + last_rtp_timestamp_ +
+      timestamp_offset_ + last_rtp_timestamp_ +
       (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) *
           (ctx.feedback_state_.frequency_hz / 1000);
 
   rtcp::SenderReport* report = new rtcp::SenderReport();
   report->From(ssrc_);
   report->WithNtp(NtpTime(ctx.ntp_sec_, ctx.ntp_frac_));
   report->WithRtpTimestamp(rtp_timestamp);
   report->WithPacketCount(ctx.feedback_state_.packets_sent);
   report->WithOctetCount(ctx.feedback_state_.media_bytes_sent);
 
@@ skipping 594 matching lines @@
     // but we can't because of an incorrect warning (C4822) in MVS 2013.
   } sender(transport_, event_log_);
 
   RTC_DCHECK_LE(max_payload_length_, static_cast<size_t>(IP_PACKET_SIZE));
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, max_payload_length_, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc