| /* | 
 |  *  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 MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_ | 
 | #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_ | 
 |  | 
 | #include <vector> | 
 |  | 
 | #include "modules/rtp_rtcp/source/rtcp_packet/psfb.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace rtcp { | 
 | class CommonHeader; | 
 |  | 
 | // Receiver Estimated Max Bitrate (REMB) (draft-alvestrand-rmcat-remb). | 
 | class Remb : public Psfb { | 
 |  public: | 
 |   static constexpr size_t kMaxNumberOfSsrcs = 0xff; | 
 |  | 
 |   Remb(); | 
 |   Remb(const Remb&); | 
 |   ~Remb() override; | 
 |  | 
 |   // Parse assumes header is already parsed and validated. | 
 |   bool Parse(const CommonHeader& packet); | 
 |  | 
 |   bool SetSsrcs(std::vector<uint32_t> ssrcs); | 
 |   void SetBitrateBps(int64_t bitrate_bps) { bitrate_bps_ = bitrate_bps; } | 
 |  | 
 |   int64_t bitrate_bps() const { return bitrate_bps_; } | 
 |   const std::vector<uint32_t>& ssrcs() const { return ssrcs_; } | 
 |  | 
 |   size_t BlockLength() const override; | 
 |  | 
 |   bool Create(uint8_t* packet, | 
 |               size_t* index, | 
 |               size_t max_length, | 
 |               PacketReadyCallback callback) const override; | 
 |  | 
 |  private: | 
 |   static constexpr uint32_t kUniqueIdentifier = 0x52454D42;  // 'R' 'E' 'M' 'B'. | 
 |  | 
 |   // Media ssrc is unused, shadow base class setter and getter. | 
 |   void SetMediaSsrc(uint32_t); | 
 |   uint32_t media_ssrc() const; | 
 |  | 
 |   int64_t bitrate_bps_; | 
 |   std::vector<uint32_t> ssrcs_; | 
 | }; | 
 | }  // namespace rtcp | 
 | }  // namespace webrtc | 
 | #endif  // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REMB_H_ |