webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
Yves Gerey 988cc0870b [Cleanup] Add missing #include. Remove useless ones.
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.

bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 11:32:56 +00:00

51 lines
1.2 KiB
C++

/*
* 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 MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
#define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
#include <stddef.h>
#include <stdint.h>
#include "system_wrappers/include/ntp_time.h"
namespace webrtc {
namespace rtcp {
class Rrtr {
public:
static const uint8_t kBlockType = 4;
static const uint16_t kBlockLength = 2;
static const size_t kLength = 4 * (kBlockLength + 1); // 12
Rrtr() {}
Rrtr(const Rrtr&) = default;
~Rrtr() {}
Rrtr& operator=(const Rrtr&) = default;
void Parse(const uint8_t* buffer);
// Fills buffer with the Rrtr.
// Consumes Rrtr::kLength bytes.
void Create(uint8_t* buffer) const;
void SetNtp(NtpTime ntp) { ntp_ = ntp; }
NtpTime ntp() const { return ntp_; }
private:
NtpTime ntp_;
};
} // namespace rtcp
} // namespace webrtc
#endif // MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_