Removed extended jitter report from 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 565 matching lines @@
 //       .                                                               .
 //       |                      inter-arrival jitter                     |
 //       +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 //
 //  If present, this RTCP packet must be placed after a receiver report
 //  (inside a compound RTCP packet), and MUST have the same value for RC
 //  (reception report count) as the receiver report.
 
 RTCPSender::BuildResult RTCPSender::BuildExtendedJitterReport(
     RtcpContext* ctx) {
-  // sanity
-  if (ctx->position + 8 >= IP_PACKET_SIZE)
+  rtcp::Ij ij;
+  ij.WithJitterItem(ctx->jitter_transmission_offset);
+
+  PacketBuiltCallback callback(ctx);
+  if (!ij.BuildExternalBuffer(&ctx->buffer[ctx->position],
+                              ctx->buffer_size - ctx->position,
+                              &callback)) {
     return BuildResult::kTruncated;
+  }
 
-  // add picture loss indicator
-  uint8_t RC = 1;
-  *ctx->AllocateData(1) = 0x80 + RC;
-  *ctx->AllocateData(1) = 195;
-
-  // Used fixed length of 2
-  *ctx->AllocateData(1) = 0;
-  *ctx->AllocateData(1) = 1;
-
-  // Add inter-arrival jitter
-  ByteWriter<uint32_t>::WriteBigEndian(ctx->AllocateData(4),
-                                       ctx->jitter_transmission_offset);

[åsapersson, 2015/06/25 15:14:25]

Not related to this cl but it seems like ctx->jitter_transmission_offset is
always zero?

[sprang_webrtc, 2015/07/08 08:57:59]

Indeed it seems so. This has evidently been the case for a long time. OK to fix
that in a separate cl?

[stefan-webrtc, 2015/07/08 09:09:39]

I don't think it makes much sense to keep the transmission offset-based jitter.
We never use transmission offsets anyway... :)

[sprang_webrtc, 2015/07/23 13:50:40]

Acknowledged.

   return BuildResult::kSuccess;
 }
 
 RTCPSender::BuildResult RTCPSender::BuildPLI(RtcpContext* ctx) {
   // sanity
   if (ctx->position + 12 >= IP_PACKET_SIZE)
     return BuildResult::kTruncated;
 
   // add picture loss indicator
   uint8_t FMT = 1;
@@ skipping 946 matching lines @@
 
 bool RTCPSender::AllVolatileFlagsConsumed() const {
   for (const ReportFlag& flag : report_flags_) {
     if (flag.is_volatile)
       return false;
   }
   return true;
 }
 
 }  // namespace webrtc