Wire up send-side bandwidth estimation.

webrtc/modules/rtp_rtcp/source/rtcp_utility.h

 /*
 *  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.
 */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
 
 #include <stddef.h> // size_t, ptrdiff_t
 
+#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+namespace rtcp {
+class RtcpPacket;
+}
 namespace RTCPUtility {
 
 class NackStats {
  public:
   NackStats();
   ~NackStats();
 
   // Updates stats with requested sequence number.
   // This function should be called for each NACK request to calculate the
   // number of unique NACKed RTP packets.
@@ skipping 256 matching lines @@
 
   // RFC 3611
   kXrHeader,
   kXrReceiverReferenceTime,
   kXrDlrrReportBlock,
   kXrDlrrReportBlockItem,
   kXrVoipMetric,
 
   kApp,
   kAppItem,
+
+  // draft-holmer-rmcat-transport-wide-cc-extensions
+  kTransportFeedback,
 };
 
 struct RTCPRawPacket {
   const uint8_t* _ptrPacketBegin;
   const uint8_t* _ptrPacketEnd;
 };
 
 struct RTCPModRawPacket {
   uint8_t* _ptrPacketBegin;
   uint8_t* _ptrPacketEnd;
@@ skipping 33 matching lines @@
 class RTCPParserV2 {
  public:
   RTCPParserV2(
       const uint8_t* rtcpData,
       size_t rtcpDataLength,
       bool rtcpReducedSizeEnable);  // Set to true, to allow non-compound RTCP!
   ~RTCPParserV2();
 
   RTCPPacketTypes PacketType() const;
   const RTCPPacket& Packet() const;
+  rtcp::RtcpPacket* ReleaseRtcpPacket();

[stefan-webrtc, 2015/09/17 10:33:19]

How does this RtcpPacket differ from RTCPPacket? Is this part of a new
refactoring to use the new RtcpPacket class?

[sprang_webrtc, 2015/09/17 15:08:28]

This is the base class for the shiny new RtcpPacket classes asapersson@
introduced. Currently they're only used on the send side, but I think we should
work towards using them on the receive side as well..

   const RTCPRawPacket& RawPacket() const;
   ptrdiff_t LengthLeft() const;
 
   bool IsValid() const;
 
   RTCPPacketTypes Begin();
   RTCPPacketTypes Iterate();
 
  private:
   enum class ParseState {
@@ skipping 78 matching lines @@
 
   bool _validPacket;
   const uint8_t* _ptrRTCPData;
   const uint8_t* _ptrRTCPBlockEnd;
 
   ParseState _state;
   uint8_t _numberOfBlocks;
 
   RTCPPacketTypes _packetType;
   RTCPPacket _packet;
+  rtc::scoped_ptr<webrtc::rtcp::RtcpPacket> rtcp_packet_;
 };
 
 class RTCPPacketIterator {
  public:
   RTCPPacketIterator(uint8_t* rtcpData, size_t rtcpDataLength);
   ~RTCPPacketIterator();
 
   const RTCPCommonHeader* Begin();
   const RTCPCommonHeader* Iterate();
   const RTCPCommonHeader* Current();
 
  private:
   uint8_t* const _ptrBegin;
   uint8_t* const _ptrEnd;
 
   uint8_t* _ptrBlock;
 
   RTCPCommonHeader _header;
 };
 }  // RTCPUtility
 }  // namespace webrtc
 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_