| /* |
| * 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/base/logging.h" |
| #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_cvo.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/video_coding/main/interface/video_coding.h" |
| #include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
| #include "webrtc/system_wrappers/include/metrics.h" |
| #include "webrtc/system_wrappers/include/tick_util.h" |
| #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" |
| #include "webrtc/system_wrappers/include/trace.h" |
| |
| namespace webrtc { |
| |
| static const int kPacketLogIntervalMs = 10000; |
| |
| ViEReceiver::ViEReceiver(VideoCodingModule* module_vcm, |
| RemoteBitrateEstimator* remote_bitrate_estimator, |
| RtpFeedback* rtp_feedback) |
| : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()), |
| clock_(Clock::GetRealTimeClock()), |
| rtp_header_parser_(RtpHeaderParser::Create()), |
| rtp_payload_registry_( |
| new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))), |
| rtp_receiver_( |
| RtpReceiver::CreateVideoReceiver(clock_, |
| this, |
| rtp_feedback, |
| rtp_payload_registry_.get())), |
| rtp_receive_statistics_(ReceiveStatistics::Create(clock_)), |
| fec_receiver_(FecReceiver::Create(this)), |
| rtp_rtcp_(NULL), |
| vcm_(module_vcm), |
| remote_bitrate_estimator_(remote_bitrate_estimator), |
| ntp_estimator_(new RemoteNtpTimeEstimator(clock_)), |
| receiving_(false), |
| restored_packet_in_use_(false), |
| receiving_ast_enabled_(false), |
| receiving_cvo_enabled_(false), |
| receiving_tsn_enabled_(false), |
| last_packet_log_ms_(-1) { |
| assert(remote_bitrate_estimator); |
| } |
| |
| ViEReceiver::~ViEReceiver() { |
| UpdateHistograms(); |
| } |
| |
| void ViEReceiver::UpdateHistograms() { |
| FecPacketCounter counter = fec_receiver_->GetPacketCounter(); |
| if (counter.num_packets > 0) { |
| RTC_HISTOGRAM_PERCENTAGE( |
| "WebRTC.Video.ReceivedFecPacketsInPercent", |
| static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets)); |
| } |
| if (counter.num_fec_packets > 0) { |
| RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec", |
| static_cast<int>(counter.num_recovered_packets * |
| 100 / counter.num_fec_packets)); |
| } |
| } |
| |
| 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::SetRtxPayloadType(int payload_type, |
| int associated_payload_type) { |
| rtp_payload_registry_->SetRtxPayloadType(payload_type, |
| associated_payload_type); |
| } |
| |
| void ViEReceiver::SetUseRtxPayloadMappingOnRestore(bool val) { |
| rtp_payload_registry_->set_use_rtx_payload_mapping_on_restore(val); |
| } |
| |
| void ViEReceiver::SetRtxSsrc(uint32_t ssrc) { |
| rtp_payload_registry_->SetRtxSsrc(ssrc); |
| } |
| |
| bool ViEReceiver::GetRtxSsrc(uint32_t* ssrc) const { |
| return rtp_payload_registry_->GetRtxSsrc(ssrc); |
| } |
| |
| bool ViEReceiver::IsFecEnabled() const { |
| return rtp_payload_registry_->ulpfec_payload_type() > -1; |
| } |
| |
| 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::RegisterRtpRtcpModules( |
| const std::vector<RtpRtcp*>& rtp_modules) { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| // Only change the "simulcast" modules, the base module can be accessed |
| // without a lock whereas the simulcast modules require locking as they can be |
| // changed in runtime. |
| rtp_rtcp_simulcast_ = |
| std::vector<RtpRtcp*>(rtp_modules.begin() + 1, 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); |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveVideoRotationStatus(bool enable, int id) { |
| if (enable) { |
| if (rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionVideoRotation, id)) { |
| receiving_cvo_enabled_ = true; |
| return true; |
| } else { |
| return false; |
| } |
| } else { |
| receiving_cvo_enabled_ = false; |
| return rtp_header_parser_->DeregisterRtpHeaderExtension( |
| kRtpExtensionVideoRotation); |
| } |
| } |
| |
| bool ViEReceiver::SetReceiveTransportSequenceNumber(bool enable, int id) { |
| if (enable) { |
| if (rtp_header_parser_->RegisterRtpHeaderExtension( |
| kRtpExtensionTransportSequenceNumber, id)) { |
| receiving_tsn_enabled_ = true; |
| return true; |
| } else { |
| return false; |
| } |
| } else { |
| receiving_tsn_enabled_ = false; |
| return rtp_header_parser_->DeregisterRtpHeaderExtension( |
| kRtpExtensionTransportSequenceNumber); |
| } |
| } |
| |
| int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet, |
| size_t 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, |
| size_t 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 size_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, |
| size_t rtp_packet_length) { |
| RTPHeader header; |
| if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, &header)) { |
| return false; |
| } |
| header.payload_type_frequency = kVideoPayloadTypeFrequency; |
| bool in_order = IsPacketInOrder(header); |
| return ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); |
| } |
| |
| int ViEReceiver::InsertRTPPacket(const uint8_t* rtp_packet, |
| size_t rtp_packet_length, |
| const PacketTime& packet_time) { |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (!receiving_) { |
| return -1; |
| } |
| } |
| |
| RTPHeader header; |
| if (!rtp_header_parser_->Parse(rtp_packet, rtp_packet_length, |
| &header)) { |
| return -1; |
| } |
| size_t payload_length = rtp_packet_length - header.headerLength; |
| int64_t arrival_time_ms; |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| if (packet_time.timestamp != -1) |
| arrival_time_ms = (packet_time.timestamp + 500) / 1000; |
| else |
| arrival_time_ms = now_ms; |
| |
| { |
| // Periodically log the RTP header of incoming packets. |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) { |
| std::stringstream ss; |
| ss << "Packet received on SSRC: " << header.ssrc << " with payload type: " |
| << static_cast<int>(header.payloadType) << ", timestamp: " |
| << header.timestamp << ", sequence number: " << header.sequenceNumber |
| << ", arrival time: " << arrival_time_ms; |
| if (header.extension.hasTransmissionTimeOffset) |
| ss << ", toffset: " << header.extension.transmissionTimeOffset; |
| if (header.extension.hasAbsoluteSendTime) |
| ss << ", abs send time: " << header.extension.absoluteSendTime; |
| LOG(LS_INFO) << ss.str(); |
| last_packet_log_ms_ = now_ms; |
| } |
| } |
| |
| remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_length, |
| header, true); |
| 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, |
| size_t 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; |
| assert(packet_length >= header.headerLength); |
| size_t payload_length = packet_length - header.headerLength; |
| 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, |
| size_t 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, packet_length); |
| // Notify vcm about received FEC packets to avoid NACKing these packets. |
| NotifyReceiverOfFecPacket(header); |
| } |
| 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 > 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; |
| } |
| if (!rtp_payload_registry_->RestoreOriginalPacket( |
| restored_packet_, 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_, packet_length); |
| restored_packet_in_use_ = false; |
| return ret; |
| } |
| return false; |
| } |
| |
| void ViEReceiver::NotifyReceiverOfFecPacket(const RTPHeader& header) { |
| int8_t last_media_payload_type = |
| rtp_payload_registry_->last_received_media_payload_type(); |
| if (last_media_payload_type < 0) { |
| LOG(LS_WARNING) << "Failed to get last media payload type."; |
| return; |
| } |
| // Fake an empty media packet. |
| WebRtcRTPHeader rtp_header = {}; |
| rtp_header.header = header; |
| rtp_header.header.payloadType = last_media_payload_type; |
| rtp_header.header.paddingLength = 0; |
| PayloadUnion payload_specific; |
| if (!rtp_payload_registry_->GetPayloadSpecifics(last_media_payload_type, |
| &payload_specific)) { |
| LOG(LS_WARNING) << "Failed to get payload specifics."; |
| return; |
| } |
| rtp_header.type.Video.codec = payload_specific.Video.videoCodecType; |
| rtp_header.type.Video.rotation = kVideoRotation_0; |
| if (header.extension.hasVideoRotation) { |
| rtp_header.type.Video.rotation = |
| ConvertCVOByteToVideoRotation(header.extension.videoRotation); |
| } |
| OnReceivedPayloadData(NULL, 0, &rtp_header); |
| } |
| |
| int ViEReceiver::InsertRTCPPacket(const uint8_t* rtcp_packet, |
| size_t rtcp_packet_length) { |
| { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| if (!receiving_) { |
| return -1; |
| } |
| |
| for (RtpRtcp* rtp_rtcp : rtp_rtcp_simulcast_) |
| 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; |
| } |
| |
| int64_t rtt = 0; |
| rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, NULL, NULL, NULL); |
| if (rtt == 0) { |
| // Waiting for valid rtt. |
| return 0; |
| } |
| uint32_t ntp_secs = 0; |
| uint32_t ntp_frac = 0; |
| uint32_t rtp_timestamp = 0; |
| if (0 != rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL, |
| &rtp_timestamp)) { |
| // Waiting for RTCP. |
| return 0; |
| } |
| ntp_estimator_->UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); |
| |
| return 0; |
| } |
| |
| void ViEReceiver::StartReceive() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| receiving_ = true; |
| } |
| |
| void ViEReceiver::StopReceive() { |
| CriticalSectionScoped cs(receive_cs_.get()); |
| receiving_ = false; |
| } |
| |
| 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. |
| int64_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 |
