Validates sending RTCP before RTP.

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 467 matching lines @@
 rtc::scoped_ptr<rtcp::RtcpPacket> RTCPSender::BuildSR(const RtcpContext& ctx) {
   for (int i = (RTCP_NUMBER_OF_SR - 2); i >= 0; i--) {
     // shift old
     last_send_report_[i + 1] = last_send_report_[i];
     last_rtcp_time_[i + 1] = last_rtcp_time_[i];
   }
 
   last_rtcp_time_[0] = Clock::NtpToMs(ctx.ntp_sec_, ctx.ntp_frac_);
   last_send_report_[0] = (ctx.ntp_sec_ << 16) + (ctx.ntp_frac_ >> 16);
 
+  // Timestamp shouldn't be esitmated before frame was received.
+  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_ +
       (clock_->TimeInMilliseconds() - last_frame_capture_time_ms_) *
           (ctx.feedback_state_.frequency_hz / 1000);
 
   rtcp::SenderReport* report = new rtcp::SenderReport();
@@ skipping 318 matching lines @@
     const uint16_t* nack_list,
     bool repeat,
     uint64_t pictureID) {
   PacketContainer container(transport_, event_log_);
   {
     CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
     if (method_ == RtcpMode::kOff) {
       LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
       return -1;
     }
+    // Add all flags as volatile. Non volatile entries will not be overwritten.
+    // All new volatile flags added will be consumed by the end of this call.
+    SetFlags(packet_types, true);
+
+    // Prevent sending streams to send SR before any media has been sent.
+    const bool can_calculate_rtp_timestamp = (last_frame_capture_time_ms_ >= 0);
+    if (sending_ && !can_calculate_rtp_timestamp) {
+      bool consumed_report_flag = ConsumeFlag(kRtcpReport);
+      bool consumed_sr_flag = ConsumeFlag(kRtcpSr);
+      bool sender_report = consumed_report_flag || consumed_sr_flag;
+      if (sender_report && AllVolatileFlagsConsumed()) {
+        // This call was for Sender Report and nothing else.
+        return 0;
+      }
+      if (method_ == RtcpMode::kCompound) {
+        // Not allowed to send any RTCP packet without sender report.
+        return -1;
+      }
+    }
+
+    if (packet_type_counter_.first_packet_time_ms == -1)
+      packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();
 
     // We need to send our NTP even if we haven't received any reports.
     uint32_t ntp_sec;
     uint32_t ntp_frac;
     clock_->CurrentNtp(ntp_sec, ntp_frac);
     RtcpContext context(feedback_state, nack_size, nack_list, repeat, pictureID,
                         ntp_sec, ntp_frac, &container);
 
-    PrepareReport(packet_types, feedback_state);
+    PrepareReport(feedback_state);
 
     auto it = report_flags_.begin();
     while (it != report_flags_.end()) {
       auto builder_it = builders_.find(it->type);
       RTC_DCHECK(builder_it != builders_.end());
       if (it->is_volatile) {
         report_flags_.erase(it++);
       } else {
         ++it;
       }
@@ skipping 10 matching lines @@
           remote_ssrc_, packet_type_counter_);
     }
 
     RTC_DCHECK(AllVolatileFlagsConsumed());
   }
 
   size_t bytes_sent = container.SendPackets();
   return bytes_sent == 0 ? -1 : 0;
 }
 
-void RTCPSender::PrepareReport(const std::set<RTCPPacketType>& packetTypes,
-                               const FeedbackState& feedback_state) {
-  // Add all flags as volatile. Non volatile entries will not be overwritten
-  // and all new volatile flags added will be consumed by the end of this call.
-  SetFlags(packetTypes, true);
-
-  if (packet_type_counter_.first_packet_time_ms == -1)
-    packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();
-
+void RTCPSender::PrepareReport(const FeedbackState& feedback_state) {
   bool generate_report;
   if (IsFlagPresent(kRtcpSr) || IsFlagPresent(kRtcpRr)) {
     // Report type already explicitly set, don't automatically populate.
     generate_report = true;
     RTC_DCHECK(ConsumeFlag(kRtcpReport) == false);
   } else {
     generate_report =
         (ConsumeFlag(kRtcpReport) && method_ == RtcpMode::kReducedSize) ||
         method_ == RtcpMode::kCompound;
     if (generate_report)
@@ skipping 195 matching lines @@
     // RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sender);
     // but we can't because of an incorrect warning (C4822) in MVS 2013.
   } sender(transport_, event_log_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc