|  | /* | 
|  | *  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. | 
|  | * | 
|  | */ | 
|  |  | 
|  | #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_ | 
|  | #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_ | 
|  |  | 
|  | #include "webrtc/base/basictypes.h" | 
|  |  | 
|  | namespace webrtc { | 
|  | namespace rtcp { | 
|  |  | 
|  | class ReportBlock { | 
|  | public: | 
|  | static const size_t kLength = 24; | 
|  |  | 
|  | ReportBlock(); | 
|  | ~ReportBlock() {} | 
|  |  | 
|  | bool Parse(const uint8_t* buffer, size_t length); | 
|  |  | 
|  | // Fills buffer with the ReportBlock. | 
|  | // Consumes ReportBlock::kLength bytes. | 
|  | void Create(uint8_t* buffer) const; | 
|  |  | 
|  | void SetMediaSsrc(uint32_t ssrc) { source_ssrc_ = ssrc; } | 
|  | void SetFractionLost(uint8_t fraction_lost) { | 
|  | fraction_lost_ = fraction_lost; | 
|  | } | 
|  | bool SetCumulativeLost(uint32_t cumulative_lost); | 
|  | void SetExtHighestSeqNum(uint32_t ext_highest_seq_num) { | 
|  | extended_high_seq_num_ = ext_highest_seq_num; | 
|  | } | 
|  | void SetJitter(uint32_t jitter) { jitter_ = jitter; } | 
|  | void SetLastSr(uint32_t last_sr) { last_sr_ = last_sr; } | 
|  | void SetDelayLastSr(uint32_t delay_last_sr) { | 
|  | delay_since_last_sr_ = delay_last_sr; | 
|  | } | 
|  |  | 
|  | uint32_t source_ssrc() const { return source_ssrc_; } | 
|  | uint8_t fraction_lost() const { return fraction_lost_; } | 
|  | uint32_t cumulative_lost() const { return cumulative_lost_; } | 
|  | uint32_t extended_high_seq_num() const { return extended_high_seq_num_; } | 
|  | uint32_t jitter() const { return jitter_; } | 
|  | uint32_t last_sr() const { return last_sr_; } | 
|  | uint32_t delay_since_last_sr() const { return delay_since_last_sr_; } | 
|  |  | 
|  | private: | 
|  | uint32_t source_ssrc_; | 
|  | uint8_t fraction_lost_; | 
|  | uint32_t cumulative_lost_; | 
|  | uint32_t extended_high_seq_num_; | 
|  | uint32_t jitter_; | 
|  | uint32_t last_sr_; | 
|  | uint32_t delay_since_last_sr_; | 
|  | }; | 
|  |  | 
|  | }  // namespace rtcp | 
|  | }  // namespace webrtc | 
|  | #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_ |