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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
67 lines
2.1 KiB
C++
67 lines
2.1 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
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#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
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#include "rtc_base/basictypes.h"
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namespace webrtc {
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namespace rtcp {
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class ReportBlock {
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public:
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static const size_t kLength = 24;
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ReportBlock();
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~ReportBlock() {}
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bool Parse(const uint8_t* buffer, size_t length);
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// Fills buffer with the ReportBlock.
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// Consumes ReportBlock::kLength bytes.
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void Create(uint8_t* buffer) const;
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void SetMediaSsrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
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void SetFractionLost(uint8_t fraction_lost) {
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fraction_lost_ = fraction_lost;
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}
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bool SetCumulativeLost(uint32_t cumulative_lost);
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void SetExtHighestSeqNum(uint32_t ext_highest_seq_num) {
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extended_high_seq_num_ = ext_highest_seq_num;
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}
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void SetJitter(uint32_t jitter) { jitter_ = jitter; }
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void SetLastSr(uint32_t last_sr) { last_sr_ = last_sr; }
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void SetDelayLastSr(uint32_t delay_last_sr) {
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delay_since_last_sr_ = delay_last_sr;
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}
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uint32_t source_ssrc() const { return source_ssrc_; }
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uint8_t fraction_lost() const { return fraction_lost_; }
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uint32_t cumulative_lost() const { return cumulative_lost_; }
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uint32_t extended_high_seq_num() const { return extended_high_seq_num_; }
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uint32_t jitter() const { return jitter_; }
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uint32_t last_sr() const { return last_sr_; }
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uint32_t delay_since_last_sr() const { return delay_since_last_sr_; }
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private:
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uint32_t source_ssrc_;
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uint8_t fraction_lost_;
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uint32_t cumulative_lost_;
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uint32_t extended_high_seq_num_;
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uint32_t jitter_;
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uint32_t last_sr_;
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uint32_t delay_since_last_sr_;
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};
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} // namespace rtcp
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
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