[rtp_rtcp] rtcp::Xr moved into own file and renamed to ExtendedReports on the way

webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h

+/*
+ *  Copyright (c) 2016 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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_
+
+#include <vector>
+
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+
+namespace webrtc {
+namespace rtcp {
+
+// From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
+class ExtendedReports : public RtcpPacket {
+ public:
+  typedef std::vector<RTCPUtility::RTCPPacketXRDLRRReportBlockItem> DlrrBlock;
+  ExtendedReports() : RtcpPacket() {
+    memset(&xr_header_, 0, sizeof(xr_header_));
+  }
+
+  virtual ~ExtendedReports() {}
+
+  void From(uint32_t ssrc) {
+    xr_header_.OriginatorSSRC = ssrc;
+  }
+
+  // Max 50 items of each of {Rrtr, Dlrr, VoipMetric} allowed per Xr.
+  bool WithRrtr(Rrtr* rrtr);
+  bool WithDlrr(Dlrr* dlrr);
+  bool WithVoipMetric(VoipMetric* voip_metric);
+
+ protected:
+  bool Create(uint8_t* packet,
+              size_t* index,
+              size_t max_length,
+              RtcpPacket::PacketReadyCallback* callback) const override;
+
+ private:
+  static const int kMaxNumberOfRrtrBlocks = 50;
+  static const int kMaxNumberOfDlrrBlocks = 50;
+  static const int kMaxNumberOfVoipMetricBlocks = 50;
+
+  size_t BlockLength() const {
+    const size_t kXrHeaderLength = 8;
+    return kXrHeaderLength + RrtrLength() + DlrrLength() + VoipMetricLength();
+  }
+
+  size_t RrtrLength() const { return Rrtr::kLength * rrtr_blocks_.size(); }
+
+  size_t DlrrLength() const;
+
+  size_t VoipMetricLength() const {
+    return VoipMetric::kLength * voip_metric_blocks_.size();
+  }
+
+  RTCPUtility::RTCPPacketXR xr_header_;
+  std::vector<Rrtr> rrtr_blocks_;
+  std::vector<Dlrr> dlrr_blocks_;
+  std::vector<VoipMetric> voip_metric_blocks_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(ExtendedReports);
+};
+
+}  // namespace rtcp
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_EXTENDED_REPORTS_H_