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android / platform / external / webrtc / 4591fbd09f9cb6e83433c49a12dd8524c2806502 / . / webrtc / modules / rtp_rtcp / source / rtcp_sender.cc
blob: 9991992bc3bbfa96646462d668099d612ac09d18 [file] [log] [blame]
/*
* 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_sender.h"
#include <assert.h> // assert
#include <stdlib.h> // rand
#include <string.h> // memcpy
#include <algorithm> // min
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
using RTCPUtility::RTCPCnameInformation;
NACKStringBuilder::NACKStringBuilder() :
_stream(""), _count(0), _consecutive(false)
{
// Empty.
}
NACKStringBuilder::~NACKStringBuilder() {}
void NACKStringBuilder::PushNACK(uint16_t nack)
{
if (_count == 0)
{
_stream << nack;
} else if (nack == _prevNack + 1)
{
_consecutive = true;
} else
{
if (_consecutive)
{
_stream << "-" << _prevNack;
_consecutive = false;
}
_stream << "," << nack;
}
_count++;
_prevNack = nack;
}
std::string NACKStringBuilder::GetResult()
{
if (_consecutive)
{
_stream << "-" << _prevNack;
_consecutive = false;
}
return _stream.str();
}
RTCPSender::FeedbackState::FeedbackState()
: send_payload_type(0),
frequency_hz(0),
packets_sent(0),
media_bytes_sent(0),
send_bitrate(0),
last_rr_ntp_secs(0),
last_rr_ntp_frac(0),
remote_sr(0),
has_last_xr_rr(false) {}
RTCPSender::RTCPSender(const int32_t id,
const bool audio,
Clock* clock,
ReceiveStatistics* receive_statistics) :
_id(id),
_audio(audio),
_clock(clock),
_method(kRtcpOff),
_criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()),
_cbTransport(NULL),
_criticalSectionRTCPSender(CriticalSectionWrapper::CreateCriticalSection()),
_usingNack(false),
_sending(false),
_sendTMMBN(false),
_REMB(false),
_sendREMB(false),
_TMMBR(false),
_IJ(false),
_nextTimeToSendRTCP(0),
start_timestamp_(0),
last_rtp_timestamp_(0),
last_frame_capture_time_ms_(-1),
_SSRC(0),
_remoteSSRC(0),
_CNAME(),
receive_statistics_(receive_statistics),
internal_report_blocks_(),
external_report_blocks_(),
_csrcCNAMEs(),
_cameraDelayMS(0),
_lastSendReport(),
_lastRTCPTime(),
last_xr_rr_(),
_CSRCs(0),
_CSRC(),
_includeCSRCs(true),
_sequenceNumberFIR(0),
_rembBitrate(0),
_tmmbrHelp(),
_tmmbr_Send(0),
_packetOH_Send(0),
_appSend(false),
_appSubType(0),
_appName(),
_appData(NULL),
_appLength(0),
xrSendReceiverReferenceTimeEnabled_(false),
_xrSendVoIPMetric(false),
_xrVoIPMetric()
{
memset(_CNAME, 0, sizeof(_CNAME));
memset(_lastSendReport, 0, sizeof(_lastSendReport));
memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
}
RTCPSender::~RTCPSender() {
delete [] _appData;
while (!internal_report_blocks_.empty()) {
delete internal_report_blocks_.begin()->second;
internal_report_blocks_.erase(internal_report_blocks_.begin());
}
while (!external_report_blocks_.empty()) {
std::map<uint32_t, RTCPReportBlock*>::iterator it =
external_report_blocks_.begin();
delete it->second;
external_report_blocks_.erase(it);
}
while (!_csrcCNAMEs.empty()) {
std::map<uint32_t, RTCPCnameInformation*>::iterator it =
_csrcCNAMEs.begin();
delete it->second;
_csrcCNAMEs.erase(it);
}
delete _criticalSectionTransport;
delete _criticalSectionRTCPSender;
}
int32_t
RTCPSender::RegisterSendTransport(Transport* outgoingTransport)
{
CriticalSectionScoped lock(_criticalSectionTransport);
_cbTransport = outgoingTransport;
return 0;
}
RTCPMethod
RTCPSender::Status() const
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _method;
}
int32_t
RTCPSender::SetRTCPStatus(const RTCPMethod method)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(method != kRtcpOff)
{
if(_audio)
{
_nextTimeToSendRTCP = _clock->TimeInMilliseconds() +
(RTCP_INTERVAL_AUDIO_MS/2);
} else
{
_nextTimeToSendRTCP = _clock->TimeInMilliseconds() +
(RTCP_INTERVAL_VIDEO_MS/2);
}
}
_method = method;
return 0;
}
bool
RTCPSender::Sending() const
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _sending;
}
int32_t
RTCPSender::SetSendingStatus(const FeedbackState& feedback_state, bool sending)
{
bool sendRTCPBye = false;
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_method != kRtcpOff)
{
if(sending == false && _sending == true)
{
// Trigger RTCP bye
sendRTCPBye = true;
}
}
_sending = sending;
}
if(sendRTCPBye)
{
return SendRTCP(feedback_state, kRtcpBye);
}
return 0;
}
bool
RTCPSender::REMB() const
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _REMB;
}
int32_t
RTCPSender::SetREMBStatus(const bool enable)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_REMB = enable;
return 0;
}
int32_t
RTCPSender::SetREMBData(const uint32_t bitrate,
const std::vector<uint32_t>& ssrcs)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_rembBitrate = bitrate;
remb_ssrcs_ = ssrcs;
_sendREMB = true;
// Send a REMB immediately if we have a new REMB. The frequency of REMBs is
// throttled by the caller.
_nextTimeToSendRTCP = _clock->TimeInMilliseconds();
return 0;
}
bool
RTCPSender::TMMBR() const
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _TMMBR;
}
int32_t
RTCPSender::SetTMMBRStatus(const bool enable)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_TMMBR = enable;
return 0;
}
bool
RTCPSender::IJ() const
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return _IJ;
}
int32_t
RTCPSender::SetIJStatus(const bool enable)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_IJ = enable;
return 0;
}
void RTCPSender::SetStartTimestamp(uint32_t start_timestamp) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
start_timestamp_ = start_timestamp;
}
void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
int64_t capture_time_ms) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
last_rtp_timestamp_ = rtp_timestamp;
if (capture_time_ms < 0) {
// We don't currently get a capture time from VoiceEngine.
last_frame_capture_time_ms_ = _clock->TimeInMilliseconds();
} else {
last_frame_capture_time_ms_ = capture_time_ms;
}
}
void
RTCPSender::SetSSRC( const uint32_t ssrc)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_SSRC != 0)
{
// not first SetSSRC, probably due to a collision
// schedule a new RTCP report
// make sure that we send a RTP packet
_nextTimeToSendRTCP = _clock->TimeInMilliseconds() + 100;
}
_SSRC = ssrc;
}
void RTCPSender::SetRemoteSSRC(uint32_t ssrc)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_remoteSSRC = ssrc;
}
int32_t
RTCPSender::SetCameraDelay(const int32_t delayMS)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(delayMS > 1000 || delayMS < -1000)
{
LOG(LS_WARNING) << "Delay can't be larger than 1 second: "
<< delayMS << " ms";
return -1;
}
_cameraDelayMS = delayMS;
return 0;
}
int32_t RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
if (!cName)
return -1;
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_CNAME[RTCP_CNAME_SIZE - 1] = 0;
strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1);
return 0;
}
int32_t RTCPSender::AddMixedCNAME(const uint32_t SSRC,
const char cName[RTCP_CNAME_SIZE]) {
assert(cName);
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if (_csrcCNAMEs.size() >= kRtpCsrcSize) {
return -1;
}
RTCPCnameInformation* ptr = new RTCPCnameInformation();
ptr->name[RTCP_CNAME_SIZE - 1] = 0;
strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1);
_csrcCNAMEs[SSRC] = ptr;
return 0;
}
int32_t RTCPSender::RemoveMixedCNAME(const uint32_t SSRC) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
std::map<uint32_t, RTCPCnameInformation*>::iterator it =
_csrcCNAMEs.find(SSRC);
if (it == _csrcCNAMEs.end()) {
return -1;
}
delete it->second;
_csrcCNAMEs.erase(it);
return 0;
}
bool
RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const
{
/*
For audio we use a fix 5 sec interval
For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
that should be extremely rare
From RFC 3550
MAX RTCP BW is 5% if the session BW
A send report is approximately 65 bytes inc CNAME
A receiver report is approximately 28 bytes
The RECOMMENDED value for the reduced minimum in seconds is 360
divided by the session bandwidth in kilobits/second. This minimum
is smaller than 5 seconds for bandwidths greater than 72 kb/s.
If the participant has not yet sent an RTCP packet (the variable
initial is true), the constant Tmin is set to 2.5 seconds, else it
is set to 5 seconds.
The interval between RTCP packets is varied randomly over the
range [0.5,1.5] times the calculated interval to avoid unintended
synchronization of all participants
if we send
If the participant is a sender (we_sent true), the constant C is
set to the average RTCP packet size (avg_rtcp_size) divided by 25%
of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
number of senders.
if we receive only
If we_sent is not true, the constant C is set
to the average RTCP packet size divided by 75% of the RTCP
bandwidth. The constant n is set to the number of receivers
(members - senders). If the number of senders is greater than
25%, senders and receivers are treated together.
reconsideration NOT required for peer-to-peer
"timer reconsideration" is
employed. This algorithm implements a simple back-off mechanism
which causes users to hold back RTCP packet transmission if the
group sizes are increasing.
n = number of members
C = avg_size/(rtcpBW/4)
3. The deterministic calculated interval Td is set to max(Tmin, n*C).
4. The calculated interval T is set to a number uniformly distributed
between 0.5 and 1.5 times the deterministic calculated interval.
5. The resulting value of T is divided by e-3/2=1.21828 to compensate
for the fact that the timer reconsideration algorithm converges to
a value of the RTCP bandwidth below the intended average
*/
int64_t now = _clock->TimeInMilliseconds();
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_method == kRtcpOff)
{
return false;
}
if(!_audio && sendKeyframeBeforeRTP)
{
// for video key-frames we want to send the RTCP before the large key-frame
// if we have a 100 ms margin
now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
}
if(now >= _nextTimeToSendRTCP)
{
return true;
} else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin
{
// wrap
return true;
}
return false;
}
uint32_t
RTCPSender::LastSendReport( uint32_t& lastRTCPTime)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
lastRTCPTime = _lastRTCPTime[0];
return _lastSendReport[0];
}
uint32_t
RTCPSender::SendTimeOfSendReport(const uint32_t sendReport)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
// This is only saved when we are the sender
if((_lastSendReport[0] == 0) || (sendReport == 0))
{
return 0; // will be ignored
} else
{
for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i)
{
if( _lastSendReport[i] == sendReport)
{
return _lastRTCPTime[i];
}
}
}
return 0;
}
bool RTCPSender::SendTimeOfXrRrReport(uint32_t mid_ntp,
int64_t* time_ms) const {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if (last_xr_rr_.empty()) {
return false;
}
std::map<uint32_t, int64_t>::const_iterator it = last_xr_rr_.find(mid_ntp);
if (it == last_xr_rr_.end()) {
return false;
}
*time_ms = it->second;
return true;
}
void RTCPSender::GetPacketTypeCounter(
RtcpPacketTypeCounter* packet_counter) const {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
*packet_counter = packet_type_counter_;
}
int32_t RTCPSender::AddExternalReportBlock(
uint32_t SSRC,
const RTCPReportBlock* reportBlock) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return AddReportBlock(SSRC, &external_report_blocks_, reportBlock);
}
int32_t RTCPSender::AddReportBlock(
uint32_t SSRC,
std::map<uint32_t, RTCPReportBlock*>* report_blocks,
const RTCPReportBlock* reportBlock) {
assert(reportBlock);
if (report_blocks->size() >= RTCP_MAX_REPORT_BLOCKS) {
LOG(LS_WARNING) << "Too many report blocks.";
return -1;
}
std::map<uint32_t, RTCPReportBlock*>::iterator it =
report_blocks->find(SSRC);
if (it != report_blocks->end()) {
delete it->second;
report_blocks->erase(it);
}
RTCPReportBlock* copyReportBlock = new RTCPReportBlock();
memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock));
(*report_blocks)[SSRC] = copyReportBlock;
return 0;
}
int32_t RTCPSender::RemoveExternalReportBlock(uint32_t SSRC) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
std::map<uint32_t, RTCPReportBlock*>::iterator it =
external_report_blocks_.find(SSRC);
if (it == external_report_blocks_.end()) {
return -1;
}
delete it->second;
external_report_blocks_.erase(it);
return 0;
}
int32_t RTCPSender::BuildSR(const FeedbackState& feedback_state,
uint8_t* rtcpbuffer,
int& pos,
uint32_t NTPsec,
uint32_t NTPfrac)
{
// sanity
if(pos + 52 >= IP_PACKET_SIZE)
{
LOG(LS_WARNING) << "Failed to build Sender Report.";
return -2;
}
uint32_t RTPtime;
uint32_t posNumberOfReportBlocks = pos;
rtcpbuffer[pos++]=(uint8_t)0x80;
// Sender report
rtcpbuffer[pos++]=(uint8_t)200;
for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--)
{
// shift old
_lastSendReport[i+1] = _lastSendReport[i];
_lastRTCPTime[i+1] =_lastRTCPTime[i];
}
_lastRTCPTime[0] = Clock::NtpToMs(NTPsec, NTPfrac);
_lastSendReport[0] = (NTPsec << 16) + (NTPfrac >> 16);
// The timestamp of this RTCP packet should be estimated as the timestamp of
// the frame being captured at this moment. We are calculating that
// timestamp as the last frame's timestamp + the time since the last frame
// was captured.
RTPtime = start_timestamp_ + last_rtp_timestamp_ +
(_clock->TimeInMilliseconds() - last_frame_capture_time_ms_) *
(feedback_state.frequency_hz / 1000);
// Add sender data
// Save for our length field
pos++;
pos++;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// NTP
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPsec);
pos += 4;
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, NTPfrac);
pos += 4;
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, RTPtime);
pos += 4;
//sender's packet count
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos,
feedback_state.packets_sent);
pos += 4;
//sender's octet count
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos,
feedback_state.media_bytes_sent);
pos += 4;
uint8_t numberOfReportBlocks = 0;
int32_t retVal = WriteAllReportBlocksToBuffer(rtcpbuffer, pos,
numberOfReportBlocks,
NTPsec, NTPfrac);
if(retVal < 0)
{
//
return retVal ;
}
pos = retVal;
rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
uint16_t len = uint16_t((pos/4) -1);
RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len);
return 0;
}
int32_t RTCPSender::BuildSDEC(uint8_t* rtcpbuffer, int& pos) {
size_t lengthCname = strlen(_CNAME);
assert(lengthCname < RTCP_CNAME_SIZE);
// sanity
if(pos + 12 + lengthCname >= IP_PACKET_SIZE) {
LOG(LS_WARNING) << "Failed to build SDEC.";
return -2;
}
// SDEC Source Description
// We always need to add SDES CNAME
rtcpbuffer[pos++] = static_cast<uint8_t>(0x80 + 1 + _csrcCNAMEs.size());
rtcpbuffer[pos++] = static_cast<uint8_t>(202);
// handle SDES length later on
uint32_t SDESLengthPos = pos;
pos++;
pos++;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// CNAME = 1
rtcpbuffer[pos++] = static_cast<uint8_t>(1);
//
rtcpbuffer[pos++] = static_cast<uint8_t>(lengthCname);
uint16_t SDESLength = 10;
memcpy(&rtcpbuffer[pos], _CNAME, lengthCname);
pos += lengthCname;
SDESLength += (uint16_t)lengthCname;
uint16_t padding = 0;
// We must have a zero field even if we have an even multiple of 4 bytes
if ((pos % 4) == 0) {
padding++;
rtcpbuffer[pos++]=0;
}
while ((pos % 4) != 0) {
padding++;
rtcpbuffer[pos++]=0;
}
SDESLength += padding;
std::map<uint32_t, RTCPUtility::RTCPCnameInformation*>::iterator it =
_csrcCNAMEs.begin();
for(; it != _csrcCNAMEs.end(); it++) {
RTCPCnameInformation* cname = it->second;
uint32_t SSRC = it->first;
// Add SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, SSRC);
pos += 4;
// CNAME = 1
rtcpbuffer[pos++] = static_cast<uint8_t>(1);
size_t length = strlen(cname->name);
assert(length < RTCP_CNAME_SIZE);
rtcpbuffer[pos++]= static_cast<uint8_t>(length);
SDESLength += 6;
memcpy(&rtcpbuffer[pos],cname->name, length);
pos += length;
SDESLength += length;
uint16_t padding = 0;
// We must have a zero field even if we have an even multiple of 4 bytes
if((pos % 4) == 0){
padding++;
rtcpbuffer[pos++]=0;
}
while((pos % 4) != 0){
padding++;
rtcpbuffer[pos++] = 0;
}
SDESLength += padding;
}
// in 32-bit words minus one and we don't count the header
uint16_t buffer_length = (SDESLength / 4) - 1;
RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, buffer_length);
return 0;
}
int32_t
RTCPSender::BuildRR(uint8_t* rtcpbuffer,
int& pos,
const uint32_t NTPsec,
const uint32_t NTPfrac)
{
// sanity one block
if(pos + 32 >= IP_PACKET_SIZE)
{
return -2;
}
uint32_t posNumberOfReportBlocks = pos;
rtcpbuffer[pos++]=(uint8_t)0x80;
rtcpbuffer[pos++]=(uint8_t)201;
// Save for our length field
pos++;
pos++;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
uint8_t numberOfReportBlocks = 0;
int retVal = WriteAllReportBlocksToBuffer(rtcpbuffer, pos,
numberOfReportBlocks,
NTPsec, NTPfrac);
if(retVal < 0)
{
return pos;
}
pos = retVal;
rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
uint16_t len = uint16_t((pos)/4 -1);
RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + 2, len);
return 0;
}
// From RFC 5450: Transmission Time Offsets in RTP Streams.
// 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
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// hdr |V=2|P| RC | PT=IJ=195 | length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | inter-arrival jitter |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// . .
// . .
// . .
// | inter-arrival jitter |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// If present, this RTCP packet must be placed after a receiver report
// (inside a compound RTCP packet), and MUST have the same value for RC
// (reception report count) as the receiver report.
int32_t
RTCPSender::BuildExtendedJitterReport(
uint8_t* rtcpbuffer,
int& pos,
const uint32_t jitterTransmissionTimeOffset)
{
if (external_report_blocks_.size() > 0)
{
// TODO(andresp): Remove external report blocks since they are not
// supported.
LOG(LS_ERROR) << "Handling of external report blocks not implemented.";
return 0;
}
// sanity
if(pos + 8 >= IP_PACKET_SIZE)
{
return -2;
}
// add picture loss indicator
uint8_t RC = 1;
rtcpbuffer[pos++]=(uint8_t)0x80 + RC;
rtcpbuffer[pos++]=(uint8_t)195;
// Used fixed length of 2
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)(1);
// Add inter-arrival jitter
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos,
jitterTransmissionTimeOffset);
pos += 4;
return 0;
}
int32_t
RTCPSender::BuildPLI(uint8_t* rtcpbuffer, int& pos)
{
// sanity
if(pos + 12 >= IP_PACKET_SIZE)
{
return -2;
}
// add picture loss indicator
uint8_t FMT = 1;
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)206;
//Used fixed length of 2
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)(2);
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Add the remote SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
return 0;
}
int32_t RTCPSender::BuildFIR(uint8_t* rtcpbuffer,
int& pos,
bool repeat) {
// sanity
if(pos + 20 >= IP_PACKET_SIZE) {
return -2;
}
if (!repeat) {
_sequenceNumberFIR++; // do not increase if repetition
}
// add full intra request indicator
uint8_t FMT = 4;
rtcpbuffer[pos++] = (uint8_t)0x80 + FMT;
rtcpbuffer[pos++] = (uint8_t)206;
//Length of 4
rtcpbuffer[pos++] = (uint8_t)0;
rtcpbuffer[pos++] = (uint8_t)(4);
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// RFC 5104 4.3.1.2. Semantics
// SSRC of media source
rtcpbuffer[pos++] = (uint8_t)0;
rtcpbuffer[pos++] = (uint8_t)0;
rtcpbuffer[pos++] = (uint8_t)0;
rtcpbuffer[pos++] = (uint8_t)0;
// Additional Feedback Control Information (FCI)
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
rtcpbuffer[pos++] = (uint8_t)(_sequenceNumberFIR);
rtcpbuffer[pos++] = (uint8_t)0;
rtcpbuffer[pos++] = (uint8_t)0;
rtcpbuffer[pos++] = (uint8_t)0;
return 0;
}
/*
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First | Number | PictureID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
int32_t
RTCPSender::BuildSLI(uint8_t* rtcpbuffer, int& pos, const uint8_t pictureID)
{
// sanity
if(pos + 16 >= IP_PACKET_SIZE)
{
return -2;
}
// add slice loss indicator
uint8_t FMT = 2;
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)206;
//Used fixed length of 3
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)(3);
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Add the remote SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
// Add first, number & picture ID 6 bits
// first = 0, 13 - bits
// number = 0x1fff, 13 - bits only ones for now
uint32_t sliField = (0x1fff << 6)+ (0x3f & pictureID);
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, sliField);
pos += 4;
return 0;
}
/*
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
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PB |0| Payload Type| Native RPSI bit string |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| defined per codec ... | Padding (0) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
/*
* Note: not generic made for VP8
*/
int32_t
RTCPSender::BuildRPSI(uint8_t* rtcpbuffer,
int& pos,
const uint64_t pictureID,
const uint8_t payloadType)
{
// sanity
if(pos + 24 >= IP_PACKET_SIZE)
{
return -2;
}
// add Reference Picture Selection Indication
uint8_t FMT = 3;
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)206;
// calc length
uint32_t bitsRequired = 7;
uint8_t bytesRequired = 1;
while((pictureID>>bitsRequired) > 0)
{
bitsRequired += 7;
bytesRequired++;
}
uint8_t size = 3;
if(bytesRequired > 6)
{
size = 5;
} else if(bytesRequired > 2)
{
size = 4;
}
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=size;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Add the remote SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
// calc padding length
uint8_t paddingBytes = 4-((2+bytesRequired)%4);
if(paddingBytes == 4)
{
paddingBytes = 0;
}
// add padding length in bits
rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24
pos++;
// add payload type
rtcpbuffer[pos] = payloadType;
pos++;
// add picture ID
for(int i = bytesRequired-1; i > 0; i--)
{
rtcpbuffer[pos] = 0x80 | uint8_t(pictureID >> (i*7));
pos++;
}
// add last byte of picture ID
rtcpbuffer[pos] = uint8_t(pictureID & 0x7f);
pos++;
// add padding
for(int j = 0; j <paddingBytes; j++)
{
rtcpbuffer[pos] = 0;
pos++;
}
return 0;
}
int32_t
RTCPSender::BuildREMB(uint8_t* rtcpbuffer, int& pos)
{
// sanity
if(pos + 20 + 4 * remb_ssrcs_.size() >= IP_PACKET_SIZE)
{
return -2;
}
// add application layer feedback
uint8_t FMT = 15;
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)206;
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=remb_ssrcs_.size() + 4;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Remote SSRC must be 0
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, 0);
pos += 4;
rtcpbuffer[pos++]='R';
rtcpbuffer[pos++]='E';
rtcpbuffer[pos++]='M';
rtcpbuffer[pos++]='B';
rtcpbuffer[pos++] = remb_ssrcs_.size();
// 6 bit Exp
// 18 bit mantissa
uint8_t brExp = 0;
for(uint32_t i=0; i<64; i++)
{
if(_rembBitrate <= ((uint32_t)262143 << i))
{
brExp = i;
break;
}
}
const uint32_t brMantissa = (_rembBitrate >> brExp);
rtcpbuffer[pos++]=(uint8_t)((brExp << 2) + ((brMantissa >> 16) & 0x03));
rtcpbuffer[pos++]=(uint8_t)(brMantissa >> 8);
rtcpbuffer[pos++]=(uint8_t)(brMantissa);
for (size_t i = 0; i < remb_ssrcs_.size(); i++)
{
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, remb_ssrcs_[i]);
pos += 4;
}
return 0;
}
void
RTCPSender::SetTargetBitrate(unsigned int target_bitrate)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_tmmbr_Send = target_bitrate / 1000;
}
int32_t RTCPSender::BuildTMMBR(ModuleRtpRtcpImpl* rtp_rtcp_module,
uint8_t* rtcpbuffer,
int& pos) {
if (rtp_rtcp_module == NULL)
return -1;
// Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate
// If the sender is an owner of the TMMBN -> send TMMBR
// If not an owner but the TMMBR would enter the TMMBN -> send TMMBR
// get current bounding set from RTCP receiver
bool tmmbrOwner = false;
// store in candidateSet, allocates one extra slot
TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet();
// holding _criticalSectionRTCPSender while calling RTCPreceiver which
// will accuire _criticalSectionRTCPReceiver is a potental deadlock but
// since RTCPreceiver is not doing the reverse we should be fine
int32_t lengthOfBoundingSet =
rtp_rtcp_module->BoundingSet(tmmbrOwner, candidateSet);
if(lengthOfBoundingSet > 0)
{
for (int32_t i = 0; i < lengthOfBoundingSet; i++)
{
if( candidateSet->Tmmbr(i) == _tmmbr_Send &&
candidateSet->PacketOH(i) == _packetOH_Send)
{
// do not send the same tuple
return 0;
}
}
if(!tmmbrOwner)
{
// use received bounding set as candidate set
// add current tuple
candidateSet->SetEntry(lengthOfBoundingSet,
_tmmbr_Send,
_packetOH_Send,
_SSRC);
int numCandidates = lengthOfBoundingSet+ 1;
// find bounding set
TMMBRSet* boundingSet = NULL;
int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet);
if(numBoundingSet > 0 || numBoundingSet <= numCandidates)
{
tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet);
}
if(!tmmbrOwner)
{
// did not enter bounding set, no meaning to send this request
return 0;
}
}
}
if(_tmmbr_Send)
{
// sanity
if(pos + 20 >= IP_PACKET_SIZE)
{
return -2;
}
// add TMMBR indicator
uint8_t FMT = 3;
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)205;
//Length of 4
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)(4);
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// RFC 5104 4.2.1.2. Semantics
// SSRC of media source
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)0;
// Additional Feedback Control Information (FCI)
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
uint32_t bitRate = _tmmbr_Send*1000;
uint32_t mmbrExp = 0;
for(uint32_t i=0;i<64;i++)
{
if(bitRate <= ((uint32_t)131071 << i))
{
mmbrExp = i;
break;
}
}
uint32_t mmbrMantissa = (bitRate >> mmbrExp);
rtcpbuffer[pos++]=(uint8_t)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
rtcpbuffer[pos++]=(uint8_t)(mmbrMantissa >> 7);
rtcpbuffer[pos++]=(uint8_t)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01));
rtcpbuffer[pos++]=(uint8_t)(_packetOH_Send);
}
return 0;
}
int32_t
RTCPSender::BuildTMMBN(uint8_t* rtcpbuffer, int& pos)
{
TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend();
if(boundingSet == NULL)
{
return -1;
}
// sanity
if(pos + 12 + boundingSet->lengthOfSet()*8 >= IP_PACKET_SIZE)
{
LOG(LS_WARNING) << "Failed to build TMMBN.";
return -2;
}
uint8_t FMT = 4;
// add TMMBN indicator
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)205;
//Add length later
int posLength = pos;
pos++;
pos++;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// RFC 5104 4.2.2.2. Semantics
// SSRC of media source
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)0;
// Additional Feedback Control Information (FCI)
int numBoundingSet = 0;
for(uint32_t n=0; n< boundingSet->lengthOfSet(); n++)
{
if (boundingSet->Tmmbr(n) > 0)
{
uint32_t tmmbrSSRC = boundingSet->Ssrc(n);
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, tmmbrSSRC);
pos += 4;
uint32_t bitRate = boundingSet->Tmmbr(n) * 1000;
uint32_t mmbrExp = 0;
for(int i=0; i<64; i++)
{
if(bitRate <= ((uint32_t)131071 << i))
{
mmbrExp = i;
break;
}
}
uint32_t mmbrMantissa = (bitRate >> mmbrExp);
uint32_t measuredOH = boundingSet->PacketOH(n);
rtcpbuffer[pos++]=(uint8_t)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
rtcpbuffer[pos++]=(uint8_t)(mmbrMantissa >> 7);
rtcpbuffer[pos++]=(uint8_t)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01));
rtcpbuffer[pos++]=(uint8_t)(measuredOH);
numBoundingSet++;
}
}
uint16_t length= (uint16_t)(2+2*numBoundingSet);
rtcpbuffer[posLength++]=(uint8_t)(length>>8);
rtcpbuffer[posLength]=(uint8_t)(length);
return 0;
}
int32_t
RTCPSender::BuildAPP(uint8_t* rtcpbuffer, int& pos)
{
// sanity
if(_appData == NULL)
{
LOG(LS_WARNING) << "Failed to build app specific.";
return -1;
}
if(pos + 12 + _appLength >= IP_PACKET_SIZE)
{
LOG(LS_WARNING) << "Failed to build app specific.";
return -2;
}
rtcpbuffer[pos++]=(uint8_t)0x80 + _appSubType;
// Add APP ID
rtcpbuffer[pos++]=(uint8_t)204;
uint16_t length = (_appLength>>2) + 2; // include SSRC and name
rtcpbuffer[pos++]=(uint8_t)(length>>8);
rtcpbuffer[pos++]=(uint8_t)(length);
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Add our application name
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _appName);
pos += 4;
// Add the data
memcpy(rtcpbuffer +pos, _appData,_appLength);
pos += _appLength;
return 0;
}
int32_t
RTCPSender::BuildNACK(uint8_t* rtcpbuffer,
int& pos,
const int32_t nackSize,
const uint16_t* nackList,
std::string* nackString)
{
// sanity
if(pos + 16 >= IP_PACKET_SIZE)
{
LOG(LS_WARNING) << "Failed to build NACK.";
return -2;
}
// int size, uint16_t* nackList
// add nack list
uint8_t FMT = 1;
rtcpbuffer[pos++]=(uint8_t)0x80 + FMT;
rtcpbuffer[pos++]=(uint8_t)205;
rtcpbuffer[pos++]=(uint8_t) 0;
int nackSizePos = pos;
rtcpbuffer[pos++]=(uint8_t)(3); //setting it to one kNACK signal as default
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Add the remote SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
// Build NACK bitmasks and write them to the RTCP message.
// The nack list should be sorted and not contain duplicates if one
// wants to build the smallest rtcp nack packet.
int numOfNackFields = 0;
int maxNackFields = std::min<int>(kRtcpMaxNackFields,
(IP_PACKET_SIZE - pos) / 4);
int i = 0;
while (i < nackSize && numOfNackFields < maxNackFields) {
uint16_t nack = nackList[i++];
uint16_t bitmask = 0;
while (i < nackSize) {
int shift = static_cast<uint16_t>(nackList[i] - nack) - 1;
if (shift >= 0 && shift <= 15) {
bitmask |= (1 << shift);
++i;
} else {
break;
}
}
// Write the sequence number and the bitmask to the packet.
assert(pos + 4 < IP_PACKET_SIZE);
RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, nack);
pos += 2;
RtpUtility::AssignUWord16ToBuffer(rtcpbuffer + pos, bitmask);
pos += 2;
numOfNackFields++;
}
rtcpbuffer[nackSizePos] = static_cast<uint8_t>(2 + numOfNackFields);
if (i != nackSize) {
LOG(LS_WARNING) << "Nack list too large for one packet.";
}
// Report stats.
NACKStringBuilder stringBuilder;
for (int idx = 0; idx < i; ++idx) {
stringBuilder.PushNACK(nackList[idx]);
nack_stats_.ReportRequest(nackList[idx]);
}
*nackString = stringBuilder.GetResult();
packet_type_counter_.nack_requests = nack_stats_.requests();
packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
return 0;
}
int32_t
RTCPSender::BuildBYE(uint8_t* rtcpbuffer, int& pos)
{
// sanity
if(pos + 8 >= IP_PACKET_SIZE)
{
return -2;
}
if(_includeCSRCs)
{
// Add a bye packet
rtcpbuffer[pos++]=(uint8_t)0x80 + 1 + _CSRCs; // number of SSRC+CSRCs
rtcpbuffer[pos++]=(uint8_t)203;
// length
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)(1 + _CSRCs);
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// add CSRCs
for(int i = 0; i < _CSRCs; i++)
{
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _CSRC[i]);
pos += 4;
}
} else
{
// Add a bye packet
rtcpbuffer[pos++]=(uint8_t)0x80 + 1; // number of SSRC+CSRCs
rtcpbuffer[pos++]=(uint8_t)203;
// length
rtcpbuffer[pos++]=(uint8_t)0;
rtcpbuffer[pos++]=(uint8_t)1;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
}
return 0;
}
int32_t RTCPSender::BuildReceiverReferenceTime(uint8_t* buffer,
int& pos,
uint32_t ntp_sec,
uint32_t ntp_frac) {
const int kRrTimeBlockLength = 20;
if (pos + kRrTimeBlockLength >= IP_PACKET_SIZE) {
return -2;
}
if (last_xr_rr_.size() >= RTCP_NUMBER_OF_SR) {
last_xr_rr_.erase(last_xr_rr_.begin());
}
last_xr_rr_.insert(std::pair<uint32_t, int64_t>(
RTCPUtility::MidNtp(ntp_sec, ntp_frac),
Clock::NtpToMs(ntp_sec, ntp_frac)));
// Add XR header.
buffer[pos++] = 0x80;
buffer[pos++] = 207;
buffer[pos++] = 0; // XR packet length.
buffer[pos++] = 4; // XR packet length.
// Add our own SSRC.
RtpUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC);
pos += 4;
// 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 |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Add Receiver Reference Time Report block.
buffer[pos++] = 4; // BT.
buffer[pos++] = 0; // Reserved.
buffer[pos++] = 0; // Block length.
buffer[pos++] = 2; // Block length.
// NTP timestamp.
RtpUtility::AssignUWord32ToBuffer(buffer + pos, ntp_sec);
pos += 4;
RtpUtility::AssignUWord32ToBuffer(buffer + pos, ntp_frac);
pos += 4;
return 0;
}
int32_t RTCPSender::BuildDlrr(uint8_t* buffer,
int& pos,
const RtcpReceiveTimeInfo& info) {
const int kDlrrBlockLength = 24;
if (pos + kDlrrBlockLength >= IP_PACKET_SIZE) {
return -2;
}
// Add XR header.
buffer[pos++] = 0x80;
buffer[pos++] = 207;
buffer[pos++] = 0; // XR packet length.
buffer[pos++] = 5; // XR packet length.
// Add our own SSRC.
RtpUtility::AssignUWord32ToBuffer(buffer + pos, _SSRC);
pos += 4;
// 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=5 | reserved | block length |
// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
// | SSRC_1 (SSRC of first receiver) | sub-
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
// | last RR (LRR) | 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | delay since last RR (DLRR) |
// +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
// | SSRC_2 (SSRC of second receiver) | sub-
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
// : ... : 2
// Add DLRR sub block.
buffer[pos++] = 5; // BT.
buffer[pos++] = 0; // Reserved.
buffer[pos++] = 0; // Block length.
buffer[pos++] = 3; // Block length.
// NTP timestamp.
RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.sourceSSRC);
pos += 4;
RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.lastRR);
pos += 4;
RtpUtility::AssignUWord32ToBuffer(buffer + pos, info.delaySinceLastRR);
pos += 4;
return 0;
}
int32_t
RTCPSender::BuildVoIPMetric(uint8_t* rtcpbuffer, int& pos)
{
// sanity
if(pos + 44 >= IP_PACKET_SIZE)
{
return -2;
}
// Add XR header
rtcpbuffer[pos++]=(uint8_t)0x80;
rtcpbuffer[pos++]=(uint8_t)207;
uint32_t XRLengthPos = pos;
// handle length later on
pos++;
pos++;
// Add our own SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
pos += 4;
// Add a VoIP metrics block
rtcpbuffer[pos++]=7;
rtcpbuffer[pos++]=0;
rtcpbuffer[pos++]=0;
rtcpbuffer[pos++]=8;
// Add the remote SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
pos += 4;
rtcpbuffer[pos++] = _xrVoIPMetric.lossRate;
rtcpbuffer[pos++] = _xrVoIPMetric.discardRate;
rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity;
rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity;
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.burstDuration >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.burstDuration);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.gapDuration >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.gapDuration);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.roundTripDelay >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.roundTripDelay);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.endSystemDelay >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.endSystemDelay);
rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel;
rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel;
rtcpbuffer[pos++] = _xrVoIPMetric.RERL;
rtcpbuffer[pos++] = _xrVoIPMetric.Gmin;
rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor;
rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor;
rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ;
rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ;
rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig;
rtcpbuffer[pos++] = 0; // reserved
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBnominal >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBnominal);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBmax >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBmax);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBabsMax >> 8);
rtcpbuffer[pos++] = (uint8_t)(_xrVoIPMetric.JBabsMax);
rtcpbuffer[XRLengthPos]=(uint8_t)(0);
rtcpbuffer[XRLengthPos+1]=(uint8_t)(10);
return 0;
}
int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
uint32_t packetTypeFlags,
int32_t nackSize,
const uint16_t* nackList,
bool repeat,
uint64_t pictureID) {
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_method == kRtcpOff)
{
LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
return -1;
}
}
uint8_t rtcp_buffer[IP_PACKET_SIZE];
int rtcp_length = PrepareRTCP(feedback_state,
packetTypeFlags,
nackSize,
nackList,
repeat,
pictureID,
rtcp_buffer,
IP_PACKET_SIZE);
if (rtcp_length < 0) {
return -1;
}
// Sanity don't send empty packets.
if (rtcp_length == 0)
{
return -1;
}
return SendToNetwork(rtcp_buffer, static_cast<size_t>(rtcp_length));
}
int RTCPSender::PrepareRTCP(const FeedbackState& feedback_state,
uint32_t packetTypeFlags,
int32_t nackSize,
const uint16_t* nackList,
bool repeat,
uint64_t pictureID,
uint8_t* rtcp_buffer,
int buffer_size) {
uint32_t rtcpPacketTypeFlags = packetTypeFlags;
// Collect the received information.
uint32_t NTPsec = 0;
uint32_t NTPfrac = 0;
uint32_t jitterTransmissionOffset = 0;
int position = 0;
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_TMMBR ) // Attach TMMBR to send and receive reports.
{
rtcpPacketTypeFlags |= kRtcpTmmbr;
}
if(_appSend)
{
rtcpPacketTypeFlags |= kRtcpApp;
_appSend = false;
}
if(_REMB && _sendREMB)
{
// Always attach REMB to SR if that is configured. Note that REMB is
// only sent on one of the RTP modules in the REMB group.
rtcpPacketTypeFlags |= kRtcpRemb;
}
if(_xrSendVoIPMetric)
{
rtcpPacketTypeFlags |= kRtcpXrVoipMetric;
_xrSendVoIPMetric = false;
}
if(_sendTMMBN) // Set when having received a TMMBR.
{
rtcpPacketTypeFlags |= kRtcpTmmbn;
_sendTMMBN = false;
}
if (rtcpPacketTypeFlags & kRtcpReport)
{
if (xrSendReceiverReferenceTimeEnabled_ && !_sending)
{
rtcpPacketTypeFlags |= kRtcpXrReceiverReferenceTime;
}
if (feedback_state.has_last_xr_rr)
{
rtcpPacketTypeFlags |= kRtcpXrDlrrReportBlock;
}
}
if(_method == kRtcpCompound)
{
if(_sending)
{
rtcpPacketTypeFlags |= kRtcpSr;
} else
{
rtcpPacketTypeFlags |= kRtcpRr;
}
} else if(_method == kRtcpNonCompound)
{
if(rtcpPacketTypeFlags & kRtcpReport)
{
if(_sending)
{
rtcpPacketTypeFlags |= kRtcpSr;
} else
{
rtcpPacketTypeFlags |= kRtcpRr;
}
}
}
if( rtcpPacketTypeFlags & kRtcpRr ||
rtcpPacketTypeFlags & kRtcpSr)
{
// generate next time to send a RTCP report
// seeded from RTP constructor
int32_t random = rand() % 1000;
int32_t timeToNext = RTCP_INTERVAL_AUDIO_MS;
if(_audio)
{
timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) +
(RTCP_INTERVAL_AUDIO_MS*random/1000);
}else
{
uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
if(_sending)
{
// Calculate bandwidth for video; 360 / send bandwidth in kbit/s.
uint32_t send_bitrate_kbit = feedback_state.send_bitrate / 1000;
if (send_bitrate_kbit != 0)
minIntervalMs = 360000 / send_bitrate_kbit;
}
if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
{
minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
}
timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000);
}
_nextTimeToSendRTCP = _clock->TimeInMilliseconds() + timeToNext;
}
// If the data does not fit in the packet we fill it as much as possible.
int32_t buildVal = 0;
// We need to send our NTP even if we haven't received any reports.
_clock->CurrentNtp(NTPsec, NTPfrac);
if (ShouldSendReportBlocks(rtcpPacketTypeFlags)) {
StatisticianMap statisticians =
receive_statistics_->GetActiveStatisticians();
if (!statisticians.empty()) {
StatisticianMap::const_iterator it;
int i;
for (it = statisticians.begin(), i = 0; it != statisticians.end();
++it, ++i) {
RTCPReportBlock report_block;
if (PrepareReport(
feedback_state, it->second, &report_block, &NTPsec, &NTPfrac))
AddReportBlock(it->first, &internal_report_blocks_, &report_block);
}
if (_IJ && !statisticians.empty()) {
rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
}
}
}
if(rtcpPacketTypeFlags & kRtcpSr)
{
buildVal = BuildSR(feedback_state, rtcp_buffer, position, NTPsec, NTPfrac);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
buildVal = BuildSDEC(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}else if(rtcpPacketTypeFlags & kRtcpRr)
{
buildVal = BuildRR(rtcp_buffer, position, NTPsec, NTPfrac);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
// only of set
if(_CNAME[0] != 0)
{
buildVal = BuildSDEC(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
}
}
}
if(rtcpPacketTypeFlags & kRtcpTransmissionTimeOffset)
{
// If present, this RTCP packet must be placed after a
// receiver report.
buildVal = BuildExtendedJitterReport(rtcp_buffer,
position,
jitterTransmissionOffset);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpPli)
{
buildVal = BuildPLI(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::PLI");
++packet_type_counter_.pli_packets;
TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_PLICount", _SSRC,
packet_type_counter_.pli_packets);
}
if(rtcpPacketTypeFlags & kRtcpFir)
{
buildVal = BuildFIR(rtcp_buffer, position, repeat);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::FIR");
++packet_type_counter_.fir_packets;
TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_FIRCount", _SSRC,
packet_type_counter_.fir_packets);
}
if(rtcpPacketTypeFlags & kRtcpSli)
{
buildVal = BuildSLI(rtcp_buffer, position, (uint8_t)pictureID);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpRpsi)
{
const int8_t payloadType = feedback_state.send_payload_type;
if (payloadType == -1) {
return -1;
}
buildVal = BuildRPSI(rtcp_buffer, position, pictureID,
(uint8_t)payloadType);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpRemb)
{
buildVal = BuildREMB(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
TRACE_EVENT_INSTANT0("webrtc_rtp", "RTCPSender::REMB");
}
if(rtcpPacketTypeFlags & kRtcpBye)
{
buildVal = BuildBYE(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpApp)
{
buildVal = BuildAPP(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpTmmbr)
{
buildVal = BuildTMMBR(feedback_state.module, rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpTmmbn)
{
buildVal = BuildTMMBN(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if(rtcpPacketTypeFlags & kRtcpNack)
{
std::string nackString;
buildVal = BuildNACK(rtcp_buffer, position, nackSize, nackList,
&nackString);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
TRACE_EVENT_INSTANT1("webrtc_rtp", "RTCPSender::NACK",
"nacks", TRACE_STR_COPY(nackString.c_str()));
++packet_type_counter_.nack_packets;
TRACE_COUNTER_ID1("webrtc_rtp", "RTCP_NACKCount", _SSRC,
packet_type_counter_.nack_packets);
}
if(rtcpPacketTypeFlags & kRtcpXrVoipMetric)
{
buildVal = BuildVoIPMetric(rtcp_buffer, position);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if (rtcpPacketTypeFlags & kRtcpXrReceiverReferenceTime)
{
buildVal = BuildReceiverReferenceTime(rtcp_buffer,
position,
NTPsec,
NTPfrac);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
if (rtcpPacketTypeFlags & kRtcpXrDlrrReportBlock)
{
buildVal = BuildDlrr(rtcp_buffer, position, feedback_state.last_xr_rr);
if (buildVal == -1) {
return -1;
} else if (buildVal == -2) {
return position;
}
}
return position;
}
bool RTCPSender::ShouldSendReportBlocks(uint32_t rtcp_packet_type) const {
return Status() == kRtcpCompound ||
(rtcp_packet_type & kRtcpReport) ||
(rtcp_packet_type & kRtcpSr) ||
(rtcp_packet_type & kRtcpRr);
}
bool RTCPSender::PrepareReport(const FeedbackState& feedback_state,
StreamStatistician* statistician,
RTCPReportBlock* report_block,
uint32_t* ntp_secs, uint32_t* ntp_frac) {
// Do we have receive statistics to send?
RtcpStatistics stats;
if (!statistician->GetStatistics(&stats, true))
return false;
report_block->fractionLost = stats.fraction_lost;
report_block->cumulativeLost = stats.cumulative_lost;
report_block->extendedHighSeqNum =
stats.extended_max_sequence_number;
report_block->jitter = stats.jitter;
// get our NTP as late as possible to avoid a race
_clock->CurrentNtp(*ntp_secs, *ntp_frac);
// Delay since last received report
uint32_t delaySinceLastReceivedSR = 0;
if ((feedback_state.last_rr_ntp_secs != 0) ||
(feedback_state.last_rr_ntp_frac != 0)) {
// get the 16 lowest bits of seconds and the 16 higest bits of fractions
uint32_t now=*ntp_secs&0x0000FFFF;
now <<=16;
now += (*ntp_frac&0xffff0000)>>16;
uint32_t receiveTime = feedback_state.last_rr_ntp_secs&0x0000FFFF;
receiveTime <<=16;
receiveTime += (feedback_state.last_rr_ntp_frac&0xffff0000)>>16;
delaySinceLastReceivedSR = now-receiveTime;
}
report_block->delaySinceLastSR = delaySinceLastReceivedSR;
report_block->lastSR = feedback_state.remote_sr;
return true;
}
int32_t
RTCPSender::SendToNetwork(const uint8_t* dataBuffer,
const size_t length)
{
CriticalSectionScoped lock(_criticalSectionTransport);
if(_cbTransport)
{
if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0)
{
return 0;
}
}
return -1;
}
int32_t
RTCPSender::SetCSRCStatus(const bool include)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
_includeCSRCs = include;
return 0;
}
int32_t
RTCPSender::SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize],
const uint8_t arrLength)
{
assert(arrLength <= kRtpCsrcSize);
CriticalSectionScoped lock(_criticalSectionRTCPSender);
for(int i = 0; i < arrLength;i++)
{
_CSRC[i] = arrOfCSRC[i];
}
_CSRCs = arrLength;
return 0;
}
int32_t
RTCPSender::SetApplicationSpecificData(const uint8_t subType,
const uint32_t name,
const uint8_t* data,
const uint16_t length)
{
if(length %4 != 0)
{
LOG(LS_ERROR) << "Failed to SetApplicationSpecificData.";
return -1;
}
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if(_appData)
{
delete [] _appData;
}
_appSend = true;
_appSubType = subType;
_appName = name;
_appData = new uint8_t[length];
_appLength = length;
memcpy(_appData, data, length);
return 0;
}
int32_t
RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric));
_xrSendVoIPMetric = true;
return 0;
}
void RTCPSender::SendRtcpXrReceiverReferenceTime(bool enable) {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
xrSendReceiverReferenceTimeEnabled_ = enable;
}
bool RTCPSender::RtcpXrReceiverReferenceTime() const {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return xrSendReceiverReferenceTimeEnabled_;
}
// called under critsect _criticalSectionRTCPSender
int32_t RTCPSender::WriteAllReportBlocksToBuffer(
uint8_t* rtcpbuffer,
int pos,
uint8_t& numberOfReportBlocks,
const uint32_t NTPsec,
const uint32_t NTPfrac) {
numberOfReportBlocks = external_report_blocks_.size();
numberOfReportBlocks += internal_report_blocks_.size();
if ((pos + numberOfReportBlocks * 24) >= IP_PACKET_SIZE) {
LOG(LS_WARNING) << "Can't fit all report blocks.";
return -1;
}
pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, internal_report_blocks_);
while (!internal_report_blocks_.empty()) {
delete internal_report_blocks_.begin()->second;
internal_report_blocks_.erase(internal_report_blocks_.begin());
}
pos = WriteReportBlocksToBuffer(rtcpbuffer, pos, external_report_blocks_);
return pos;
}
int32_t RTCPSender::WriteReportBlocksToBuffer(
uint8_t* rtcpbuffer,
int32_t position,
const std::map<uint32_t, RTCPReportBlock*>& report_blocks) {
std::map<uint32_t, RTCPReportBlock*>::const_iterator it =
report_blocks.begin();
for (; it != report_blocks.end(); it++) {
uint32_t remoteSSRC = it->first;
RTCPReportBlock* reportBlock = it->second;
if (reportBlock) {
// Remote SSRC
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position, remoteSSRC);
position += 4;
// fraction lost
rtcpbuffer[position++] = reportBlock->fractionLost;
// cumulative loss
RtpUtility::AssignUWord24ToBuffer(rtcpbuffer + position,
reportBlock->cumulativeLost);
position += 3;
// extended highest seq_no, contain the highest sequence number received
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position,
reportBlock->extendedHighSeqNum);
position += 4;
// Jitter
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position,
reportBlock->jitter);
position += 4;
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position,
reportBlock->lastSR);
position += 4;
RtpUtility::AssignUWord32ToBuffer(rtcpbuffer + position,
reportBlock->delaySinceLastSR);
position += 4;
}
}
return position;
}
// no callbacks allowed inside this function
int32_t
RTCPSender::SetTMMBN(const TMMBRSet* boundingSet,
const uint32_t maxBitrateKbit)
{
CriticalSectionScoped lock(_criticalSectionRTCPSender);
if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit))
{
_sendTMMBN = true;
return 0;
}
return -1;
}
} // namespace webrtc