webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00

80 lines
2.5 KiB
C++

/*
* 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_SENDER_REPORT_H_
#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SENDER_REPORT_H_
#include <vector>
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
namespace rtcp {
class CommonHeader;
class SenderReport : public RtcpPacket {
public:
static constexpr uint8_t kPacketType = 200;
static constexpr size_t kMaxNumberOfReportBlocks = 0x1f;
SenderReport();
~SenderReport() override;
// Parse assumes header is already parsed and validated.
bool Parse(const CommonHeader& packet);
void SetSenderSsrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
void SetNtp(NtpTime ntp) { ntp_ = ntp; }
void SetRtpTimestamp(uint32_t rtp_timestamp) {
rtp_timestamp_ = rtp_timestamp;
}
void SetPacketCount(uint32_t packet_count) {
sender_packet_count_ = packet_count;
}
void SetOctetCount(uint32_t octet_count) {
sender_octet_count_ = octet_count;
}
bool AddReportBlock(const ReportBlock& block);
bool SetReportBlocks(std::vector<ReportBlock> blocks);
void ClearReportBlocks() { report_blocks_.clear(); }
uint32_t sender_ssrc() const { return sender_ssrc_; }
NtpTime ntp() const { return ntp_; }
uint32_t rtp_timestamp() const { return rtp_timestamp_; }
uint32_t sender_packet_count() const { return sender_packet_count_; }
uint32_t sender_octet_count() const { return sender_octet_count_; }
const std::vector<ReportBlock>& report_blocks() const {
return report_blocks_;
}
size_t BlockLength() const override;
bool Create(uint8_t* packet,
size_t* index,
size_t max_length,
RtcpPacket::PacketReadyCallback* callback) const override;
private:
const size_t kSenderBaseLength = 24;
uint32_t sender_ssrc_;
NtpTime ntp_;
uint32_t rtp_timestamp_;
uint32_t sender_packet_count_;
uint32_t sender_octet_count_;
std::vector<ReportBlock> report_blocks_;
};
} // namespace rtcp
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_SENDER_REPORT_H_