rtcp::Ij renamed to rtcp::ExtendedJitterReport and moved into own file

webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc

Index: webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
new file mode 100644
index 0000000000000000000000000000000000000000..dfdf232102eb6b1fb19567b47272f8f5a88b37c9
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.cc
@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+using webrtc::RTCPUtility::RtcpCommonHeader;
+
+namespace webrtc {
+namespace rtcp {
+
+// Transmission Time Offsets in RTP Streams (RFC 5450).
+//
+//      0                   1                   2                   3
+//      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+// hdr |V=2|P|    RC   |   PT=IJ=195   |             length            |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//     |                      inter-arrival jitter                     |
+//     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+//     .                                                               .
+//     .                                                               .
+//     .                                                               .
+//     |                      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.
+
+bool ExtendedJitterReport::Parse(const RtcpCommonHeader& header,
+                                 const uint8_t* payload) {
+  RTC_DCHECK(header.packet_type == kPacketType);
+
+  const uint8_t jitters_count = header.count_or_format;
+  const size_t kJitterSizeBytes = 4u;
+
+  if (header.payload_size_bytes < jitters_count * kJitterSizeBytes) {
+    LOG(LS_WARNING) << "Packet is too small to contain all the jitter.";
+    return false;
+  }
+
+  inter_arrival_jitters_.resize(jitters_count);
+  for (size_t index = 0; index < jitters_count; ++index) {
+    inter_arrival_jitters_[index] =
+        ByteReader<uint32_t>::ReadBigEndian(&payload[index * kJitterSizeBytes]);
+  }
+
+  return true;
+}
+
+bool ExtendedJitterReport::WithJitter(uint32_t jitter) {
+  if (inter_arrival_jitters_.size() >= kMaxNumberOfJitters) {
+    LOG(LS_WARNING) << "Max inter-arrival jitter items reached.";
+    return false;
+  }
+  inter_arrival_jitters_.push_back(jitter);
+  return true;
+}
+
+bool ExtendedJitterReport::Create(
+    uint8_t* packet,
+    size_t* index,
+    size_t max_length,
+    RtcpPacket::PacketReadyCallback* callback) const {
+  while (*index + BlockLength() > max_length) {
+    if (!OnBufferFull(packet, index, callback))
+      return false;
+  }
+  const size_t index_end = *index + BlockLength();
+  size_t length = inter_arrival_jitters_.size();
+  CreateHeader(length, kPacketType, length, packet, index);
+
+  for (uint32_t jitter : inter_arrival_jitters_) {
+    ByteWriter<uint32_t>::WriteBigEndian(packet + *index, jitter);
+    *index += sizeof(uint32_t);
+  }
+  // Sanity check.
+  RTC_DCHECK_EQ(index_end, *index);
+  return true;
+}
+
+}  // namespace rtcp
+}  // namespace webrtc