webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.cc
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

49 lines
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.
*/
#include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace rtcp {
// Receiver Reference Time Report Block (RFC 3611).
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | BT=4 | reserved | block length = 2 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | NTP timestamp, most significant word |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | NTP timestamp, least significant word |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
void Rrtr::Parse(const uint8_t* buffer) {
RTC_DCHECK(buffer[0] == kBlockType);
// reserved = buffer[1];
RTC_DCHECK(ByteReader<uint16_t>::ReadBigEndian(&buffer[2]) == kBlockLength);
uint32_t seconds = ByteReader<uint32_t>::ReadBigEndian(&buffer[4]);
uint32_t fraction = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
ntp_.Set(seconds, fraction);
}
void Rrtr::Create(uint8_t* buffer) const {
const uint8_t kReserved = 0;
buffer[0] = kBlockType;
buffer[1] = kReserved;
ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockLength);
ByteWriter<uint32_t>::WriteBigEndian(&buffer[4], ntp_.seconds());
ByteWriter<uint32_t>::WriteBigEndian(&buffer[8], ntp_.fractions());
}
} // namespace rtcp
} // namespace webrtc