| /* |
| * 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/video_engine/vie_receiver.h" |
| |
| #include <vector> |
| |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/utility/interface/rtp_dump.h" |
| #include "webrtc/modules/video_coding/main/interface/video_coding.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/interface/logging.h" |
| #include "webrtc/system_wrappers/interface/tick_util.h" |
| #include "webrtc/system_wrappers/interface/timestamp_extrapolator.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| namespace webrtc { |
| |
| ViEReceiver::ViEReceiver(const int32_t channel_id, |
| VideoCodingModule* module_vcm, |
| RemoteBitrateEstimator* remote_bitrate_estimator, |
| RtpFeedback* rtp_feedback) |
| : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
| rtp_header_parser_(RtpHeaderParser::Create()), |
| rtp_payload_registry_(new RTPPayloadRegistry( |
| RTPPayloadStrategy::CreateStrategy(false))), |
| rtp_receiver_(RtpReceiver::CreateVideoReceiver( |
| channel_id, Clock::GetRealTimeClock(), this, rtp_feedback, |
| rtp_payload_registry_.get())), |
| rtp_receive_statistics_(ReceiveStatistics::Create( |
| Clock::GetRealTimeClock())), |
| fec_receiver_(FecReceiver::Create(this)), |
| rtp_rtcp_(NULL), |
| vcm_(module_vcm), |
| remote_bitrate_estimator_(remote_bitrate_estimator), |
| ntp_estimator_(new RemoteNtpTimeEstimator(Clock::GetRealTimeClock())), |
| rtp_dump_(NULL), |
| receiving_(false), |
| restored_packet_in_use_(false), |
| receiving_ast_enabled_(false) { |
| assert(remote_bitrate_estimator); |
| } |
| |
| ViEReceiver::~ViEReceiver() { |
| if (rtp_dump_) { |
| rtp_dump_->Stop(); |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveCodec(const VideoCodec& video_codec) { |
| int8_t old_pltype = -1; |
| if (rtp_payload_registry_->ReceivePayloadType(video_codec.plName, |
| kVideoPayloadTypeFrequency, |
| 0, |
| video_codec.maxBitrate, |
| &old_pltype) != -1) { |
| rtp_payload_registry_->DeRegisterReceivePayload(old_pltype); |
| } |
| |
| return RegisterPayload(video_codec); |
| } |
| |
| bool ViEReceiver::RegisterPayload(const VideoCodec& video_codec) { |
| return rtp_receiver_->RegisterReceivePayload(video_codec.plName, |
| video_codec.plType, |
| kVideoPayloadTypeFrequency, |
| 0, |
| video_codec.maxBitrate) == 0; |
| } |
| |
| void ViEReceiver::SetNackStatus(bool enable, |
| int max_nack_reordering_threshold) { |
| if (!enable) { |
| // Reset the threshold back to the lower default threshold when NACK is |
| // disabled since we no longer will be receiving retransmissions. |
| max_nack_reordering_threshold = kDefaultMaxReorderingThreshold; |
| } |
| rtp_receive_statistics_->SetMaxReorderingThreshold( |
| max_nack_reordering_threshold); |
| rtp_receiver_->SetNACKStatus(enable ? kNackRtcp : kNackOff); |
| } |
| |
| void ViEReceiver::SetRtxStatus(bool enable, uint32_t ssrc) { |
| rtp_payload_registry_->SetRtxStatus(enable, ssrc); |
| } |
| |
| void ViEReceiver::SetRtxPayloadType(uint32_t payload_type) { |
| rtp_payload_registry_->SetRtxPayloadType(payload_type); |
| } |
| |
| uint32_t ViEReceiver::GetRemoteSsrc() const { |
| return rtp_receiver_->SSRC(); |
| } |
| |
| int ViEReceiver::GetCsrcs(uint32_t* csrcs) const { |
| return rtp_receiver_->CSRCs(csrcs); |
| } |
| |
| void ViEReceiver::SetRtpRtcpModule(RtpRtcp* module) { |
| rtp_rtcp_ = module; |
| } |
| |
| RtpReceiver* ViEReceiver::GetRtpReceiver() const { |
| return rtp_receiver_.get(); |
| } |
| |
| void ViEReceiver::RegisterSimulcastRtpRtcpModules( |
| const std::list<RtpRtcp*>& rtp_modules) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| rtp_rtcp_simulcast_.clear(); |
| |
| if (!rtp_modules.empty()) { |
| rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(), |
| rtp_modules.begin(), |
| rtp_modules.end()); |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveTimestampOffsetStatus(bool enable, int id) { |
| if (enable) { |
| return rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset, id); |
| } else { |
| return rtp_header_parser_->DeregisterRtpHeaderExtension( |
| kRtpExtensionTransmissionTimeOffset); |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) { |
| if (enable) { |
| if (rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime, id)) { |
| receiving_ast_enabled_ = true; |
| return true; |
| } else { |
| return false; |
| } |
| } else { |
| receiving_ast_enabled_ = false; |
| return rtp_header_parser_->DeregisterRtpHeaderExtension( |
| kRtpExtensionAbsoluteSendTime); |
| } |
| } |
| |
| int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, |
| int rtp_packet_length, |
| const PacketTime& packet_time) { |
| return InsertRTPPacket(static_cast<const uint8_t*>(rtp_packet), |
| rtp_packet_length, packet_time); |
| } |
| |
| int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet, |
| int rtcp_packet_length) { |
| return InsertRTCPPacket(static_cast<const uint8_t*>(rtcp_packet), |
| rtcp_packet_length); |
| } |
| |
| int32_t ViEReceiver::OnReceivedPayloadData( |
| const uint8_t* payload_data, const uint16_t payload_size, |
| const WebRtcRTPHeader* rtp_header) { |
| WebRtcRTPHeader rtp_header_with_ntp = *rtp_header; |
| rtp_header_with_ntp.ntp_time_ms = |
| ntp_estimator_->Estimate(rtp_header->header.timestamp); |
| if (vcm_->IncomingPacket(payload_data, |
| payload_size, |
| rtp_header_with_ntp) != 0) { |
| // Check this... |
| return -1; |
| } |
| return 0; |
| } |
| |
| bool ViEReceiver::OnRecoveredPacket(const uint8_t* rtp_packet, |
| int rtp_packet_length) { |
| RTPHeader header; |
| if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { |
| return false; |
| } |
| header.payload_type_frequency = kVideoPayloadTypeFrequency; |
| return ReceivePacket(rtp_packet, rtp_packet_length, header, false); |
| } |
| |
| void ViEReceiver::ReceivedBWEPacket( |
| int64_t arrival_time_ms, int payload_size, const RTPHeader& header) { |
| // Only forward if the incoming packet *and* the channel are both configured |
| // to receive absolute sender time. RTP time stamps may have different rates |
| // for audio and video and shouldn't be mixed. |
| if (header.extension.hasAbsoluteSendTime && receiving_ast_enabled_) { |
| remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size, |
| header); |
| } |
| } |
| |
| int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet, |
| int rtp_packet_length, |
| const PacketTime& packet_time) { |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (!receiving_) { |
| return -1; |
| } |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket(rtp_packet, |
| static_cast<uint16_t>(rtp_packet_length)); |
| } |
| } |
| |
| RTPHeader header; |
| if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, |
| &header)) { |
| return -1; |
| } |
| int payload_length = rtp_packet_length - header.headerLength; |
| int64_t arrival_time_ms; |
| if (packet_time.timestamp != -1) |
| arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
| else |
| arrival_time_ms = TickTime::MillisecondTimestamp(); |
| |
| remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, |
| payload_length, header); |
| header.payload_type_frequency = kVideoPayloadTypeFrequency; |
| |
| bool in_order = IsPacketInOrder(header); |
| rtp_payload_registry_->SetIncomingPayloadType(header); |
| int ret = ReceivePacket(rtp_packet, rtp_packet_length, header, in_order) |
| ? 0 |
| : -1; |
| // Update receive statistics after ReceivePacket. |
| // Receive statistics will be reset if the payload type changes (make sure |
| // that the first packet is included in the stats). |
| rtp_receive_statistics_->IncomingPacket( |
| header, rtp_packet_length, IsPacketRetransmitted(header, in_order)); |
| return ret; |
| } |
| |
| bool ViEReceiver::ReceivePacket(const uint8_t* packet, |
| int packet_length, |
| const RTPHeader& header, |
| bool in_order) { |
| if (rtp_payload_registry_->IsEncapsulated(header)) { |
| return ParseAndHandleEncapsulatingHeader(packet, packet_length, header); |
| } |
| const uint8_t* payload = packet + header.headerLength; |
| int payload_length = packet_length - header.headerLength; |
| assert(payload_length >= 0); |
| PayloadUnion payload_specific; |
| if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType, |
| &payload_specific)) { |
| return false; |
| } |
| return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length, |
| payload_specific, in_order); |
| } |
| |
| bool ViEReceiver::ParseAndHandleEncapsulatingHeader(const uint8_t* packet, |
| int packet_length, |
| const RTPHeader& header) { |
| if (rtp_payload_registry_->IsRed(header)) { |
| int8_t ulpfec_pt = rtp_payload_registry_->ulpfec_payload_type(); |
| if (packet[header.headerLength] == ulpfec_pt) |
| rtp_receive_statistics_->FecPacketReceived(header.ssrc); |
| if (fec_receiver_->AddReceivedRedPacket( |
| header, packet, packet_length, ulpfec_pt) != 0) { |
| return false; |
| } |
| return fec_receiver_->ProcessReceivedFec() == 0; |
| } else if (rtp_payload_registry_->IsRtx(header)) { |
| if (header.headerLength + header.paddingLength == packet_length) { |
| // This is an empty packet and should be silently dropped before trying to |
| // parse the RTX header. |
| return true; |
| } |
| // Remove the RTX header and parse the original RTP header. |
| if (packet_length < header.headerLength) |
| return false; |
| if (packet_length > static_cast<int>(sizeof(restored_packet_))) |
| return false; |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (restored_packet_in_use_) { |
| LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet."; |
| return false; |
| } |
| uint8_t* restored_packet_ptr = restored_packet_; |
| if (!rtp_payload_registry_->RestoreOriginalPacket( |
| &restored_packet_ptr, packet, &packet_length, rtp_receiver_->SSRC(), |
| header)) { |
| LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header"; |
| return false; |
| } |
| restored_packet_in_use_ = true; |
| bool ret = OnRecoveredPacket(restored_packet_ptr, packet_length); |
| restored_packet_in_use_ = false; |
| return ret; |
| } |
| return false; |
| } |
| |
| int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet, |
| int rtcp_packet_length) { |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (!receiving_) { |
| return -1; |
| } |
| |
| if (rtp_dump_) { |
| rtp_dump_->DumpPacket( |
| rtcp_packet, static_cast<uint16_t>(rtcp_packet_length)); |
| } |
| |
| std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin(); |
| while (it != rtp_rtcp_simulcast_.end()) { |
| RtpRtcp* rtp_rtcp = *it++; |
| rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
| } |
| } |
| assert(rtp_rtcp_); // Should be set by owner at construction time. |
| int ret = rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
| if (ret != 0) { |
| return ret; |
| } |
| |
| ntp_estimator_->UpdateRtcpTimestamp(rtp_receiver_->SSRC(), rtp_rtcp_); |
| |
| return 0; |
| } |
| |
| void ViEReceiver::StartReceive() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| receiving_ = true; |
| } |
| |
| void ViEReceiver::StopReceive() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| receiving_ = false; |
| } |
| |
| int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (rtp_dump_) { |
| // Restart it if it already exists and is started |
| rtp_dump_->Stop(); |
| } else { |
| rtp_dump_ = RtpDump::CreateRtpDump(); |
| if (rtp_dump_ == NULL) { |
| return -1; |
| } |
| } |
| if (rtp_dump_->Start(file_nameUTF8) != 0) { |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| return -1; |
| } |
| return 0; |
| } |
| |
| int ViEReceiver::StopRTPDump() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (rtp_dump_) { |
| if (rtp_dump_->IsActive()) { |
| rtp_dump_->Stop(); |
| } |
| RtpDump::DestroyRtpDump(rtp_dump_); |
| rtp_dump_ = NULL; |
| } else { |
| return -1; |
| } |
| return 0; |
| } |
| |
| void ViEReceiver::GetReceiveBandwidthEstimatorStats( |
| ReceiveBandwidthEstimatorStats* output) const { |
| remote_bitrate_estimator_->GetStats(output); |
| } |
| |
| ReceiveStatistics* ViEReceiver::GetReceiveStatistics() const { |
| return rtp_receive_statistics_.get(); |
| } |
| |
| bool ViEReceiver::IsPacketInOrder(const RTPHeader& header) const { |
| StreamStatistician* statistician = |
| rtp_receive_statistics_->GetStatistician(header.ssrc); |
| if (!statistician) |
| return false; |
| return statistician->IsPacketInOrder(header.sequenceNumber); |
| } |
| |
| bool ViEReceiver::IsPacketRetransmitted(const RTPHeader& header, |
| bool in_order) const { |
| // Retransmissions are handled separately if RTX is enabled. |
| if (rtp_payload_registry_->RtxEnabled()) |
| return false; |
| StreamStatistician* statistician = |
| rtp_receive_statistics_->GetStatistician(header.ssrc); |
| if (!statistician) |
| return false; |
| // Check if this is a retransmission. |
| uint16_t min_rtt = 0; |
| rtp_rtcp_->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL); |
| return !in_order && |
| statistician->IsRetransmitOfOldPacket(header, min_rtt); |
| } |
| } // namespace webrtc |
