本文整理汇总了C++中RtcpHeader类的典型用法代码示例。如果您正苦于以下问题:C++ RtcpHeader类的具体用法?C++ RtcpHeader怎么用?C++ RtcpHeader使用的例子?那么, 这里精选的类代码示例或许可以为您提供帮助。
在下文中一共展示了RtcpHeader类的15个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于系统推荐出更棒的C++代码示例。
示例1: if
void MediaStream::onTransportData(std::shared_ptr<DataPacket> incoming_packet, Transport *transport) {
if ((audio_sink_ == nullptr && video_sink_ == nullptr && fb_sink_ == nullptr)) {
return;
}
std::shared_ptr<DataPacket> packet = std::make_shared<DataPacket>(*incoming_packet);
if (transport->mediaType == AUDIO_TYPE) {
packet->type = AUDIO_PACKET;
} else if (transport->mediaType == VIDEO_TYPE) {
packet->type = VIDEO_PACKET;
}
auto stream_ptr = shared_from_this();
worker_->task([stream_ptr, packet]{
if (!stream_ptr->pipeline_initialized_) {
ELOG_DEBUG("%s message: Pipeline not initialized yet.", stream_ptr->toLog());
return;
}
char* buf = packet->data;
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
if (!chead->isRtcp()) {
uint32_t recvSSRC = head->getSSRC();
if (stream_ptr->isVideoSourceSSRC(recvSSRC)) {
packet->type = VIDEO_PACKET;
} else if (stream_ptr->isAudioSourceSSRC(recvSSRC)) {
packet->type = AUDIO_PACKET;
}
}
stream_ptr->pipeline_->read(std::move(packet));
});
}
示例2: ELOG_DEBUG
void WebRtcConnection::writeSsrc(char* buf, int len, unsigned int ssrc) {
ELOG_DEBUG("LEN %d", len);
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
//if it is RTCP we check it it is a compound packet
if (chead->isRtcp()) {
char* movingBuf = buf;
int rtcpLength = 0;
int totalLength = 0;
do{
movingBuf+=rtcpLength;
RtcpHeader *chead= reinterpret_cast<RtcpHeader*>(movingBuf);
rtcpLength= (ntohs(chead->length)+1)*4;
totalLength+= rtcpLength;
ELOG_DEBUG("Is RTCP, prev SSRC %u, new %u, len %d ", chead->getSSRC(), ssrc, rtcpLength);
chead->ssrc=htonl(ssrc);
if (chead->packettype == RTCP_PS_Feedback_PT){
FirHeader *thefir = reinterpret_cast<FirHeader*>(movingBuf);
if (thefir->fmt == 4){ // It is a FIR Packet, we generate it
this->sendPLI();
}
}
} while(totalLength<len);
} else {
head->setSSRC(ssrc);
}
}
示例3: handleSR
void RRGenerationHandler::handleSR(std::shared_ptr<dataPacket> packet) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data);
auto rr_packet_pair = rr_info_map_.find(chead->getSSRC());
if (rr_packet_pair == rr_info_map_.end()) {
ELOG_DEBUG("%s message: handleRtpPacket ssrc not found, ssrc: %u", connection_->toLog(), chead->getSSRC());
return;
}
std::shared_ptr<RRPackets> selected_packet_info = rr_packet_pair->second;
selected_packet_info->last_sr_mid_ntp = chead->get32MiddleNtp();
selected_packet_info->last_sr_ts = packet->received_time_ms;
uint32_t expected = selected_packet_info->extended_seq - selected_packet_info->base_seq + 1;
selected_packet_info->lost = expected - selected_packet_info->packets_received;
uint8_t fraction = 0;
uint32_t expected_interval = expected - selected_packet_info->expected_prior;
selected_packet_info->expected_prior = expected;
uint32_t received_interval = selected_packet_info->packets_received - selected_packet_info->received_prior;
selected_packet_info->received_prior = selected_packet_info->packets_received;
uint32_t lost_interval = expected_interval - received_interval;
if (expected_interval != 0 && lost_interval > 0) {
fraction = (lost_interval << 8) / expected_interval;
}
selected_packet_info->frac_lost = fraction;
if (!use_timing_) {
sendRR(selected_packet_info);
}
}
示例4: memcpy
void SdesTransport::onNiceData(unsigned int component_id, char* data, int len, NiceConnection* nice) {
//boost::mutex::scoped_lock lock(readMutex_);
int length = len;
SrtpChannel *srtp = srtp_;
if (this->getTransportState() == TRANSPORT_READY) {
memcpy(unprotectBuf_, data, len);
if (component_id == 2) {
srtp = srtcp_;
}
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (unprotectBuf_);
if (chead->isRtcp()){
if(srtp->unprotectRtcp(unprotectBuf_, &length)<0)
return;
} else {
if(srtp->unprotectRtp(unprotectBuf_, &length)<0)
return;
}
if (length <= 0)
return;
getTransportListener()->onTransportData(unprotectBuf_, length, this);
}
}
示例5: read
void RRGenerationHandler::read(Context *ctx, std::shared_ptr<dataPacket> packet) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data);
if (!chead->isRtcp() && enabled_) {
handleRtpPacket(packet);
} else if (chead->packettype == RTCP_Sender_PT && enabled_) {
handleSR(packet);
}
ctx->fireRead(packet);
}
示例6: processRtcpHeaders
void WebRtcConnection::writeSsrc(char* buf, int len, unsigned int ssrc) {
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
//if it is RTCP we check it it is a compound packet
if (chead->isRtcp()) {
processRtcpHeaders(buf,len,ssrc);
} else {
head->ssrc=htonl(ssrc);
}
}
示例7: write
void SenderBandwidthEstimationHandler::write(Context *ctx, std::shared_ptr<dataPacket> packet) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data);
if (!chead->isRtcp() && packet->type == VIDEO_PACKET) {
period_packets_sent_++;
} else if (chead->getPacketType() == RTCP_Sender_PT &&
chead->getSSRC() == connection_->getVideoSinkSSRC()) {
analyzeSr(chead);
}
ctx->fireWrite(packet);
}
示例8: handleSr
void RtcpRrGenerator::handleSr(std::shared_ptr<dataPacket> packet) {
RtcpHeader* chead = reinterpret_cast<RtcpHeader*>(packet->data);
if (ssrc_ != chead->getSSRC()) {
ELOG_DEBUG("message: handleRtpPacket ssrc not found, ssrc: %u", chead->getSSRC());
return;
}
rr_info_.last_sr_mid_ntp = chead->get32MiddleNtp();
rr_info_.last_sr_ts = packet->received_time_ms;
}
示例9: read
void MediaStream::read(std::shared_ptr<DataPacket> packet) {
char* buf = packet->data;
int len = packet->length;
// PROCESS RTCP
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
uint32_t recvSSRC = 0;
if (!chead->isRtcp()) {
recvSSRC = head->getSSRC();
} else if (chead->packettype == RTCP_Sender_PT) { // Sender Report
recvSSRC = chead->getSSRC();
}
// DELIVER FEEDBACK (RR, FEEDBACK PACKETS)
if (chead->isFeedback()) {
if (fb_sink_ != nullptr && should_send_feedback_) {
fb_sink_->deliverFeedback(std::move(packet));
}
} else {
// RTP or RTCP Sender Report
if (bundle_) {
// Check incoming SSRC
// Deliver data
if (isVideoSourceSSRC(recvSSRC)) {
parseIncomingPayloadType(buf, len, VIDEO_PACKET);
video_sink_->deliverVideoData(std::move(packet));
} else if (isAudioSourceSSRC(recvSSRC)) {
parseIncomingPayloadType(buf, len, AUDIO_PACKET);
audio_sink_->deliverAudioData(std::move(packet));
} else {
ELOG_DEBUG("%s read video unknownSSRC: %u, localVideoSSRC: %u, localAudioSSRC: %u",
toLog(), recvSSRC, this->getVideoSourceSSRC(), this->getAudioSourceSSRC());
}
} else {
if (packet->type == AUDIO_PACKET && audio_sink_ != nullptr) {
parseIncomingPayloadType(buf, len, AUDIO_PACKET);
// Firefox does not send SSRC in SDP
if (getAudioSourceSSRC() == 0) {
ELOG_DEBUG("%s discoveredAudioSourceSSRC:%u", toLog(), recvSSRC);
this->setAudioSourceSSRC(recvSSRC);
}
audio_sink_->deliverAudioData(std::move(packet));
} else if (packet->type == VIDEO_PACKET && video_sink_ != nullptr) {
parseIncomingPayloadType(buf, len, VIDEO_PACKET);
// Firefox does not send SSRC in SDP
if (getVideoSourceSSRC() == 0) {
ELOG_DEBUG("%s discoveredVideoSourceSSRC:%u", toLog(), recvSSRC);
this->setVideoSourceSSRC(recvSSRC);
}
// change ssrc for RTP packets, don't touch here if RTCP
video_sink_->deliverVideoData(std::move(packet));
}
} // if not bundle
} // if not Feedback
}
示例10: write
void RtcpProcessorHandler::write(Context *ctx, std::shared_ptr<dataPacket> packet) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data);
if (chead->isFeedback()) {
int length = processor_->analyzeFeedback(packet->data, packet->length);
if (length) {
ctx->fireWrite(packet);
}
return;
}
ctx->fireWrite(packet);
}
示例11: write
void SRPacketHandler::write(Context *ctx, std::shared_ptr<dataPacket> packet) {
if (initialized_ && enabled_) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data);
if (!chead->isRtcp() && enabled_) {
handleRtpPacket(packet);
} else if (chead->packettype == RTCP_Sender_PT && enabled_) {
handleSR(packet);
}
}
ctx->fireWrite(packet);
}
示例12: ELOG_DEBUG
void DtlsTransport::onNiceData(packetPtr packet) {
int len = packet->length;
char *data = packet->data;
unsigned int component_id = packet->comp;
int length = len;
SrtpChannel *srtp = srtp_.get();
if (DtlsTransport::isDtlsPacket(data, len)) {
ELOG_DEBUG("%s message: Received DTLS message, transportName: %s, componentId: %u",
toLog(), transport_name.c_str(), component_id);
if (component_id == 1) {
if (rtp_resender_.get() != NULL) {
rtp_resender_->cancel();
}
dtlsRtp->read(reinterpret_cast<unsigned char*>(data), len);
} else {
if (rtcp_resender_.get() != NULL) {
rtcp_resender_->cancel();
}
dtlsRtcp->read(reinterpret_cast<unsigned char*>(data), len);
}
return;
} else if (this->getTransportState() == TRANSPORT_READY) {
unprotect_packet_->length = len;
unprotect_packet_->received_time_ms = packet->received_time_ms;
memcpy(unprotect_packet_->data, data, len);
if (dtlsRtcp != NULL && component_id == 2) {
srtp = srtcp_.get();
}
if (srtp != NULL) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(unprotect_packet_->data);
if (chead->isRtcp()) {
if (srtp->unprotectRtcp(unprotect_packet_->data, &unprotect_packet_->length) < 0) {
return;
}
} else {
if (srtp->unprotectRtp(unprotect_packet_->data, &unprotect_packet_->length) < 0) {
return;
}
}
} else {
return;
}
if (length <= 0) {
return;
}
getTransportListener()->onTransportData(unprotect_packet_, this);
}
}
示例13: read
void RtcpProcessorHandler::read(Context *ctx, std::shared_ptr<dataPacket> packet) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (packet->data);
if (chead->isRtcp()) {
if (chead->packettype == RTCP_Sender_PT) { // Sender Report
processor_->analyzeSr(chead);
}
} else {
if (stats_->getNode()["total"].hasChild("bitrateCalculated")) {
processor_->setPublisherBW(stats_->getNode()["total"]["bitrateCalculated"].value());
}
}
processor_->checkRtcpFb();
ctx->fireRead(packet);
}
示例14: read
void PacketCodecParser::read(Context *ctx, std::shared_ptr<DataPacket> packet) {
RtcpHeader *chead = reinterpret_cast<RtcpHeader*>(packet->data);
if (!chead->isRtcp() && enabled_) {
RtpHeader *rtp_header = reinterpret_cast<RtpHeader*>(packet->data);
RtpMap *codec =
stream_->getRemoteSdpInfo()->getCodecByExternalPayloadType(
rtp_header->getPayloadType());
if (codec) {
packet->codec = codec->encoding_name;
packet->clock_rate = codec->clock_rate;
ELOG_DEBUG("Reading codec: %s, clock: %u", packet->codec.c_str(), packet->clock_rate);
}
}
ctx->fireRead(std::move(packet));
}
示例15: deliverFeedback_
int WebRtcConnection::deliverFeedback_(char* buf, int len){
// Check where to send the feedback
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
// ELOG_DEBUG("received Feedback type %u ssrc %u, sourcessrc %u", chead->packettype, chead->getSSRC(), chead->getSourceSSRC());
if (chead->getSourceSSRC() == this->getAudioSourceSSRC()) {
writeSsrc(buf,len,this->getAudioSinkSSRC());
} else {
writeSsrc(buf,len,this->getVideoSinkSSRC());
}
if (videoTransport_ != NULL) {
this->queueData(0, buf, len, videoTransport_);
}
return len;
}