Use RtcpPacket to send FIR in RtcpSender

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 546 matching lines @@
   TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::PLI");
   ++packet_type_counter_.pli_packets;
   TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_PLICount",
                     ssrc_, packet_type_counter_.pli_packets);
 
   return BuildResult::kSuccess;
 }
 
 RTCPSender::BuildResult RTCPSender::BuildFIR(RtcpContext* ctx) {
-  // sanity
-  if (ctx->position + 20 >= IP_PACKET_SIZE)
+  uint8_t seq = sequence_number_fir_;
+  if (!ctx->repeat)
+    ++seq;  // Do not increase if repetition.
+
+  rtcp::Fir fir;
+  fir.From(ssrc_);
+  fir.To(remote_ssrc_);
+  fir.WithCommandSeqNum(seq);
+
+  PacketBuiltCallback callback(ctx);
+  if (!callback.BuildPacket(fir))
     return BuildResult::kTruncated;
 
-  if (!ctx->repeat)
-    sequence_number_fir_++;  // do not increase if repetition
-
-  // add full intra request indicator
-  uint8_t FMT = 4;
-  *ctx->AllocateData(1) = 0x80 + FMT;
-  *ctx->AllocateData(1) = 206;
-
-  //Length of 4
-  *ctx->AllocateData(1) = 0;
-  *ctx->AllocateData(1) = 4;
-
-  // Add our own SSRC
-  ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), ssrc_);
-
-  // RFC 5104     4.3.1.2.  Semantics
-  // SSRC of media source
-  ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), 0);
-
-  // Additional Feedback Control Information (FCI)
-  ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4), remote_ssrc_);
-
-  *ctx->AllocateData(1) = sequence_number_fir_;
-  *ctx->AllocateData(1) = 0;
-  *ctx->AllocateData(1) = 0;
-  *ctx->AllocateData(1) = 0;
+  sequence_number_fir_ = seq;  // Only update if we actually sent packet.

[åsapersson, 2015/08/04 12:07:27]

Will the caller know if the FIR request is not sent due to truncation? If not,
then a repeat for this request will correspond to the previous request.

[sprang_webrtc, 2015/08/04 12:59:38]

I updated the comment to the more clear "Update state only if we built the
packet."
That is, we don't want to update the state of the sequence number if we don't
even have space enough to build the packet. The behavior is the same as before,
we just need this temporary since the old code would return early in the
truncation case.

 
   TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTCPSender::FIR");
   ++packet_type_counter_.fir_packets;
   TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_FIRCount",
                     ssrc_, packet_type_counter_.fir_packets);
 
   return BuildResult::kSuccess;
 }
 
@@ skipping 881 matching lines @@
 
 bool RTCPSender::AllVolatileFlagsConsumed() const {
   for (const ReportFlag& flag : report_flags_) {
     if (flag.is_volatile)
       return false;
   }
   return true;
 }
 
 }  // namespace webrtc