da161d795c
git cl format --full BUG=webrtc:5565 NOTRY=true Review-Url: https://codereview.webrtc.org/2259213002 Cr-Commit-Position: refs/heads/master@{#13832}
138 lines
4.1 KiB
C++
138 lines
4.1 KiB
C++
/*
|
|
* Copyright (c) 2012 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 "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
|
|
|
|
#include <assert.h> // assert
|
|
#include <string.h> // memset
|
|
|
|
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
|
|
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
|
|
|
|
namespace webrtc {
|
|
namespace RTCPHelp {
|
|
|
|
RTCPPacketInformation::RTCPPacketInformation()
|
|
: rtcpPacketTypeFlags(0),
|
|
remoteSSRC(0),
|
|
nackSequenceNumbers(),
|
|
applicationSubType(0),
|
|
applicationName(0),
|
|
applicationData(),
|
|
applicationLength(0),
|
|
rtt(0),
|
|
interArrivalJitter(0),
|
|
sliPictureId(0),
|
|
rpsiPictureId(0),
|
|
receiverEstimatedMaxBitrate(0),
|
|
ntp_secs(0),
|
|
ntp_frac(0),
|
|
rtp_timestamp(0),
|
|
xr_originator_ssrc(0),
|
|
xr_dlrr_item(false),
|
|
VoIPMetric(nullptr) {}
|
|
|
|
RTCPPacketInformation::~RTCPPacketInformation() {
|
|
delete[] applicationData;
|
|
}
|
|
|
|
void RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric) {
|
|
VoIPMetric.reset(new RTCPVoIPMetric());
|
|
memcpy(VoIPMetric.get(), metric, sizeof(RTCPVoIPMetric));
|
|
}
|
|
|
|
void RTCPPacketInformation::AddApplicationData(const uint8_t* data,
|
|
const uint16_t size) {
|
|
uint8_t* oldData = applicationData;
|
|
uint16_t oldLength = applicationLength;
|
|
|
|
// Don't copy more than kRtcpAppCode_DATA_SIZE bytes.
|
|
uint16_t copySize = size;
|
|
if (size > kRtcpAppCode_DATA_SIZE) {
|
|
copySize = kRtcpAppCode_DATA_SIZE;
|
|
}
|
|
|
|
applicationLength += copySize;
|
|
applicationData = new uint8_t[applicationLength];
|
|
|
|
if (oldData) {
|
|
memcpy(applicationData, oldData, oldLength);
|
|
memcpy(applicationData + oldLength, data, copySize);
|
|
delete[] oldData;
|
|
} else {
|
|
memcpy(applicationData, data, copySize);
|
|
}
|
|
}
|
|
|
|
void RTCPPacketInformation::ResetNACKPacketIdArray() {
|
|
nackSequenceNumbers.clear();
|
|
}
|
|
|
|
void RTCPPacketInformation::AddNACKPacket(const uint16_t packetID) {
|
|
if (nackSequenceNumbers.size() >= kSendSideNackListSizeSanity) {
|
|
return;
|
|
}
|
|
nackSequenceNumbers.push_back(packetID);
|
|
}
|
|
|
|
void RTCPPacketInformation::AddReportInfo(
|
|
const RTCPReportBlockInformation& report_block_info) {
|
|
this->rtt = report_block_info.RTT;
|
|
report_blocks.push_back(report_block_info.remoteReceiveBlock);
|
|
}
|
|
|
|
RTCPReportBlockInformation::RTCPReportBlockInformation()
|
|
: remoteReceiveBlock(),
|
|
remoteMaxJitter(0),
|
|
RTT(0),
|
|
minRTT(0),
|
|
maxRTT(0),
|
|
avgRTT(0),
|
|
numAverageCalcs(0) {
|
|
memset(&remoteReceiveBlock, 0, sizeof(remoteReceiveBlock));
|
|
}
|
|
|
|
RTCPReportBlockInformation::~RTCPReportBlockInformation() {}
|
|
|
|
RTCPReceiveInformation::RTCPReceiveInformation() = default;
|
|
RTCPReceiveInformation::~RTCPReceiveInformation() = default;
|
|
|
|
void RTCPReceiveInformation::InsertTmmbrItem(uint32_t sender_ssrc,
|
|
const rtcp::TmmbItem& tmmbr_item,
|
|
int64_t current_time_ms) {
|
|
TimedTmmbrItem* entry = &tmmbr_[sender_ssrc];
|
|
entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, tmmbr_item.bitrate_bps(),
|
|
tmmbr_item.packet_overhead());
|
|
entry->last_updated_ms = current_time_ms;
|
|
}
|
|
|
|
void RTCPReceiveInformation::GetTmmbrSet(
|
|
int64_t current_time_ms,
|
|
std::vector<rtcp::TmmbItem>* candidates) {
|
|
// Use audio define since we don't know what interval the remote peer use.
|
|
int64_t timeouted_ms = current_time_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
|
|
for (auto it = tmmbr_.begin(); it != tmmbr_.end();) {
|
|
if (it->second.last_updated_ms < timeouted_ms) {
|
|
// Erase timeout entries.
|
|
it = tmmbr_.erase(it);
|
|
} else {
|
|
candidates->push_back(it->second.tmmbr_item);
|
|
++it;
|
|
}
|
|
}
|
|
}
|
|
|
|
void RTCPReceiveInformation::ClearTmmbr() {
|
|
tmmbr_.clear();
|
|
}
|
|
|
|
} // namespace RTCPHelp
|
|
} // namespace webrtc
|