| /* |
| * 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/rtp_video_stream_receiver.h" |
| |
| #include <algorithm> |
| #include <utility> |
| #include <vector> |
| |
| #include "webrtc/call/video_config.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/media/base/mediaconstants.h" |
| #include "webrtc/modules/pacing/packet_router.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "webrtc/modules/video_coding/frame_object.h" |
| #include "webrtc/modules/video_coding/h264_sprop_parameter_sets.h" |
| #include "webrtc/modules/video_coding/h264_sps_pps_tracker.h" |
| #include "webrtc/modules/video_coding/packet_buffer.h" |
| #include "webrtc/modules/video_coding/video_coding_impl.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/location.h" |
| #include "webrtc/rtc_base/logging.h" |
| #include "webrtc/system_wrappers/include/field_trial.h" |
| #include "webrtc/system_wrappers/include/metrics.h" |
| #include "webrtc/system_wrappers/include/timestamp_extrapolator.h" |
| #include "webrtc/video/receive_statistics_proxy.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| // TODO(philipel): Change kPacketBufferStartSize back to 32 in M63 see: |
| // crbug.com/752886 |
| constexpr int kPacketBufferStartSize = 512; |
| constexpr int kPacketBufferMaxSixe = 2048; |
| } |
| |
| std::unique_ptr<RtpRtcp> CreateRtpRtcpModule( |
| ReceiveStatistics* receive_statistics, |
| Transport* outgoing_transport, |
| RtcpRttStats* rtt_stats, |
| RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
| TransportSequenceNumberAllocator* transport_sequence_number_allocator) { |
| RtpRtcp::Configuration configuration; |
| configuration.audio = false; |
| configuration.receiver_only = true; |
| configuration.receive_statistics = receive_statistics; |
| configuration.outgoing_transport = outgoing_transport; |
| configuration.intra_frame_callback = nullptr; |
| configuration.rtt_stats = rtt_stats; |
| configuration.rtcp_packet_type_counter_observer = |
| rtcp_packet_type_counter_observer; |
| configuration.transport_sequence_number_allocator = |
| transport_sequence_number_allocator; |
| configuration.send_bitrate_observer = nullptr; |
| configuration.send_frame_count_observer = nullptr; |
| configuration.send_side_delay_observer = nullptr; |
| configuration.send_packet_observer = nullptr; |
| configuration.bandwidth_callback = nullptr; |
| configuration.transport_feedback_callback = nullptr; |
| |
| std::unique_ptr<RtpRtcp> rtp_rtcp(RtpRtcp::CreateRtpRtcp(configuration)); |
| rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
| |
| return rtp_rtcp; |
| } |
| |
| static const int kPacketLogIntervalMs = 10000; |
| |
| RtpVideoStreamReceiver::RtpVideoStreamReceiver( |
| Transport* transport, |
| RtcpRttStats* rtt_stats, |
| PacketRouter* packet_router, |
| const VideoReceiveStream::Config* config, |
| ReceiveStatisticsProxy* receive_stats_proxy, |
| ProcessThread* process_thread, |
| NackSender* nack_sender, |
| KeyFrameRequestSender* keyframe_request_sender, |
| video_coding::OnCompleteFrameCallback* complete_frame_callback, |
| VCMTiming* timing) |
| : clock_(Clock::GetRealTimeClock()), |
| config_(*config), |
| packet_router_(packet_router), |
| process_thread_(process_thread), |
| ntp_estimator_(clock_), |
| rtp_header_parser_(RtpHeaderParser::Create()), |
| rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_, |
| this, |
| this, |
| &rtp_payload_registry_)), |
| rtp_receive_statistics_(ReceiveStatistics::Create(clock_)), |
| ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)), |
| receiving_(false), |
| restored_packet_in_use_(false), |
| last_packet_log_ms_(-1), |
| rtp_rtcp_(CreateRtpRtcpModule(rtp_receive_statistics_.get(), |
| transport, |
| rtt_stats, |
| receive_stats_proxy, |
| packet_router)), |
| complete_frame_callback_(complete_frame_callback), |
| keyframe_request_sender_(keyframe_request_sender), |
| timing_(timing), |
| has_received_frame_(false) { |
| constexpr bool remb_candidate = true; |
| packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate); |
| rtp_receive_statistics_->RegisterRtpStatisticsCallback(receive_stats_proxy); |
| rtp_receive_statistics_->RegisterRtcpStatisticsCallback(receive_stats_proxy); |
| |
| RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) |
| << "A stream should not be configured with RTCP disabled. This value is " |
| "reserved for internal usage."; |
| RTC_DCHECK(config_.rtp.remote_ssrc != 0); |
| // TODO(pbos): What's an appropriate local_ssrc for receive-only streams? |
| RTC_DCHECK(config_.rtp.local_ssrc != 0); |
| RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc); |
| |
| rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode); |
| rtp_rtcp_->SetSSRC(config_.rtp.local_ssrc); |
| rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc); |
| rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); |
| |
| for (size_t i = 0; i < config_.rtp.extensions.size(); ++i) { |
| EnableReceiveRtpHeaderExtension(config_.rtp.extensions[i].uri, |
| config_.rtp.extensions[i].id); |
| } |
| |
| static const int kMaxPacketAgeToNack = 450; |
| const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0) |
| ? kMaxPacketAgeToNack |
| : kDefaultMaxReorderingThreshold; |
| rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold); |
| |
| if (config_.rtp.rtx_ssrc) { |
| rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc); |
| |
| for (const auto& kv : config_.rtp.rtx_associated_payload_types) { |
| RTC_DCHECK_NE(kv.first, 0); |
| rtp_payload_registry_.SetRtxPayloadType(kv.first, kv.second); |
| } |
| } |
| |
| if (IsUlpfecEnabled()) { |
| VideoCodec ulpfec_codec = {}; |
| ulpfec_codec.codecType = kVideoCodecULPFEC; |
| strncpy(ulpfec_codec.plName, "ulpfec", sizeof(ulpfec_codec.plName)); |
| ulpfec_codec.plType = config_.rtp.ulpfec.ulpfec_payload_type; |
| RTC_CHECK(AddReceiveCodec(ulpfec_codec)); |
| } |
| |
| if (IsRedEnabled()) { |
| VideoCodec red_codec = {}; |
| red_codec.codecType = kVideoCodecRED; |
| strncpy(red_codec.plName, "red", sizeof(red_codec.plName)); |
| red_codec.plType = config_.rtp.ulpfec.red_payload_type; |
| RTC_CHECK(AddReceiveCodec(red_codec)); |
| if (config_.rtp.ulpfec.red_rtx_payload_type != -1) { |
| rtp_payload_registry_.SetRtxPayloadType( |
| config_.rtp.ulpfec.red_rtx_payload_type, |
| config_.rtp.ulpfec.red_payload_type); |
| } |
| } |
| |
| if (config_.rtp.rtcp_xr.receiver_reference_time_report) |
| rtp_rtcp_->SetRtcpXrRrtrStatus(true); |
| |
| // Stats callback for CNAME changes. |
| rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy); |
| |
| process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); |
| |
| if (config_.rtp.nack.rtp_history_ms != 0) { |
| nack_module_.reset( |
| new NackModule(clock_, nack_sender, keyframe_request_sender)); |
| process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE); |
| } |
| |
| packet_buffer_ = video_coding::PacketBuffer::Create( |
| clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this); |
| reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this)); |
| } |
| |
| RtpVideoStreamReceiver::~RtpVideoStreamReceiver() { |
| RTC_DCHECK(secondary_sinks_.empty()); |
| |
| if (nack_module_) { |
| process_thread_->DeRegisterModule(nack_module_.get()); |
| } |
| |
| process_thread_->DeRegisterModule(rtp_rtcp_.get()); |
| |
| packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get()); |
| UpdateHistograms(); |
| } |
| |
| bool RtpVideoStreamReceiver::AddReceiveCodec( |
| const VideoCodec& video_codec, |
| const std::map<std::string, std::string>& codec_params) { |
| pt_codec_params_.insert(make_pair(video_codec.plType, codec_params)); |
| return AddReceiveCodec(video_codec); |
| } |
| |
| bool RtpVideoStreamReceiver::AddReceiveCodec(const VideoCodec& video_codec) { |
| int8_t old_pltype = -1; |
| if (rtp_payload_registry_.ReceivePayloadType(video_codec, &old_pltype) != |
| -1) { |
| rtp_payload_registry_.DeRegisterReceivePayload(old_pltype); |
| } |
| return rtp_payload_registry_.RegisterReceivePayload(video_codec) == 0; |
| } |
| |
| uint32_t RtpVideoStreamReceiver::GetRemoteSsrc() const { |
| return config_.rtp.remote_ssrc; |
| } |
| |
| int RtpVideoStreamReceiver::GetCsrcs(uint32_t* csrcs) const { |
| return rtp_receiver_->CSRCs(csrcs); |
| } |
| |
| RtpReceiver* RtpVideoStreamReceiver::GetRtpReceiver() const { |
| return rtp_receiver_.get(); |
| } |
| |
| int32_t RtpVideoStreamReceiver::OnReceivedPayloadData( |
| const uint8_t* payload_data, |
| 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); |
| VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp); |
| packet.timesNacked = |
| nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1; |
| packet.receive_time_ms = clock_->TimeInMilliseconds(); |
| |
| // In the case of a video stream without picture ids and no rtx the |
| // RtpFrameReferenceFinder will need to know about padding to |
| // correctly calculate frame references. |
| if (packet.sizeBytes == 0) { |
| reference_finder_->PaddingReceived(packet.seqNum); |
| packet_buffer_->PaddingReceived(packet.seqNum); |
| return 0; |
| } |
| |
| if (packet.codec == kVideoCodecH264) { |
| // Only when we start to receive packets will we know what payload type |
| // that will be used. When we know the payload type insert the correct |
| // sps/pps into the tracker. |
| if (packet.payloadType != last_payload_type_) { |
| last_payload_type_ = packet.payloadType; |
| InsertSpsPpsIntoTracker(packet.payloadType); |
| } |
| |
| switch (tracker_.CopyAndFixBitstream(&packet)) { |
| case video_coding::H264SpsPpsTracker::kRequestKeyframe: |
| keyframe_request_sender_->RequestKeyFrame(); |
| FALLTHROUGH(); |
| case video_coding::H264SpsPpsTracker::kDrop: |
| return 0; |
| case video_coding::H264SpsPpsTracker::kInsert: |
| break; |
| } |
| |
| } else { |
| uint8_t* data = new uint8_t[packet.sizeBytes]; |
| memcpy(data, packet.dataPtr, packet.sizeBytes); |
| packet.dataPtr = data; |
| } |
| |
| packet_buffer_->InsertPacket(&packet); |
| return 0; |
| } |
| |
| // TODO(nisse): Try to delete this method. Obstacles: It is used by |
| // ParseAndHandleEncapsulatingHeader, for handling Rtx packets, and |
| // for callbacks from |ulpfec_receiver_|. |
| void RtpVideoStreamReceiver::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; |
| } |
| header.payload_type_frequency = kVideoPayloadTypeFrequency; |
| bool in_order = IsPacketInOrder(header); |
| ReceivePacket(rtp_packet, rtp_packet_length, header, in_order); |
| } |
| |
| // TODO(pbos): Remove as soon as audio can handle a changing payload type |
| // without this callback. |
| int32_t RtpVideoStreamReceiver::OnInitializeDecoder( |
| const int8_t payload_type, |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| const int frequency, |
| const size_t channels, |
| const uint32_t rate) { |
| RTC_NOTREACHED(); |
| return 0; |
| } |
| |
| // This method handles both regular RTP packets and packets recovered |
| // via FlexFEC. |
| void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| |
| if (!receiving_) { |
| return; |
| } |
| |
| if (!packet.recovered()) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| |
| // Periodically log the RTP header of incoming packets. |
| if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) { |
| std::stringstream ss; |
| ss << "Packet received on SSRC: " << packet.Ssrc() |
| << " with payload type: " << static_cast<int>(packet.PayloadType()) |
| << ", timestamp: " << packet.Timestamp() |
| << ", sequence number: " << packet.SequenceNumber() |
| << ", arrival time: " << packet.arrival_time_ms(); |
| int32_t time_offset; |
| if (packet.GetExtension<TransmissionOffset>(&time_offset)) { |
| ss << ", toffset: " << time_offset; |
| } |
| uint32_t send_time; |
| if (packet.GetExtension<AbsoluteSendTime>(&send_time)) { |
| ss << ", abs send time: " << send_time; |
| } |
| LOG(LS_INFO) << ss.str(); |
| last_packet_log_ms_ = now_ms; |
| } |
| } |
| |
| // TODO(nisse): Delete use of GetHeader, but needs refactoring of |
| // ReceivePacket and IncomingPacket methods below. |
| RTPHeader header; |
| packet.GetHeader(&header); |
| |
| header.payload_type_frequency = kVideoPayloadTypeFrequency; |
| |
| bool in_order = IsPacketInOrder(header); |
| if (!packet.recovered()) { |
| // TODO(nisse): Why isn't this done for recovered packets? |
| rtp_payload_registry_.SetIncomingPayloadType(header); |
| } |
| ReceivePacket(packet.data(), packet.size(), header, in_order); |
| // 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). |
| if (!packet.recovered()) { |
| // TODO(nisse): We should pass a recovered flag to stats, to aid |
| // fixing bug bugs.webrtc.org/6339. |
| rtp_receive_statistics_->IncomingPacket( |
| header, packet.size(), IsPacketRetransmitted(header, in_order)); |
| } |
| |
| for (RtpPacketSinkInterface* secondary_sink : secondary_sinks_) { |
| secondary_sink->OnRtpPacket(packet); |
| } |
| } |
| |
| int32_t RtpVideoStreamReceiver::RequestKeyFrame() { |
| return rtp_rtcp_->RequestKeyFrame(); |
| } |
| |
| bool RtpVideoStreamReceiver::IsUlpfecEnabled() const { |
| return config_.rtp.ulpfec.ulpfec_payload_type != -1; |
| } |
| |
| bool RtpVideoStreamReceiver::IsRedEnabled() const { |
| return config_.rtp.ulpfec.red_payload_type != -1; |
| } |
| |
| bool RtpVideoStreamReceiver::IsRetransmissionsEnabled() const { |
| return config_.rtp.nack.rtp_history_ms > 0; |
| } |
| |
| void RtpVideoStreamReceiver::RequestPacketRetransmit( |
| const std::vector<uint16_t>& sequence_numbers) { |
| rtp_rtcp_->SendNack(sequence_numbers); |
| } |
| |
| int32_t RtpVideoStreamReceiver::ResendPackets(const uint16_t* sequence_numbers, |
| uint16_t length) { |
| return rtp_rtcp_->SendNACK(sequence_numbers, length); |
| } |
| |
| void RtpVideoStreamReceiver::OnReceivedFrame( |
| std::unique_ptr<video_coding::RtpFrameObject> frame) { |
| if (!has_received_frame_) { |
| has_received_frame_ = true; |
| if (frame->FrameType() != kVideoFrameKey) |
| keyframe_request_sender_->RequestKeyFrame(); |
| } |
| |
| if (!frame->delayed_by_retransmission()) |
| timing_->IncomingTimestamp(frame->timestamp, clock_->TimeInMilliseconds()); |
| reference_finder_->ManageFrame(std::move(frame)); |
| } |
| |
| void RtpVideoStreamReceiver::OnCompleteFrame( |
| std::unique_ptr<video_coding::FrameObject> frame) { |
| { |
| rtc::CritScope lock(&last_seq_num_cs_); |
| video_coding::RtpFrameObject* rtp_frame = |
| static_cast<video_coding::RtpFrameObject*>(frame.get()); |
| last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num(); |
| } |
| complete_frame_callback_->OnCompleteFrame(std::move(frame)); |
| } |
| |
| void RtpVideoStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms, |
| int64_t max_rtt_ms) { |
| if (nack_module_) |
| nack_module_->UpdateRtt(max_rtt_ms); |
| } |
| |
| rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedPacketMs() const { |
| return packet_buffer_->LastReceivedPacketMs(); |
| } |
| |
| rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs() |
| const { |
| return packet_buffer_->LastReceivedKeyframePacketMs(); |
| } |
| |
| void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| RTC_DCHECK(std::find(secondary_sinks_.cbegin(), secondary_sinks_.cend(), |
| sink) == secondary_sinks_.cend()); |
| secondary_sinks_.push_back(sink); |
| } |
| |
| void RtpVideoStreamReceiver::RemoveSecondarySink( |
| const RtpPacketSinkInterface* sink) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| auto it = std::find(secondary_sinks_.begin(), secondary_sinks_.end(), sink); |
| if (it == secondary_sinks_.end()) { |
| // We might be rolling-back a call whose setup failed mid-way. In such a |
| // case, it's simpler to remove "everything" rather than remember what |
| // has already been added. |
| LOG(LS_WARNING) << "Removal of unknown sink."; |
| return; |
| } |
| secondary_sinks_.erase(it); |
| } |
| |
| void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet, |
| size_t packet_length, |
| const RTPHeader& header, |
| bool in_order) { |
| if (rtp_payload_registry_.IsEncapsulated(header)) { |
| ParseAndHandleEncapsulatingHeader(packet, packet_length, header); |
| return; |
| } |
| 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; |
| } |
| rtp_receiver_->IncomingRtpPacket(header, payload, payload_length, |
| payload_specific, in_order); |
| } |
| |
| void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader( |
| const uint8_t* packet, size_t packet_length, const RTPHeader& header) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| 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 video_receiver about received FEC packets to avoid NACKing these |
| // packets. |
| NotifyReceiverOfFecPacket(header); |
| } |
| if (ulpfec_receiver_->AddReceivedRedPacket(header, packet, packet_length, |
| ulpfec_pt) != 0) { |
| return; |
| } |
| ulpfec_receiver_->ProcessReceivedFec(); |
| } 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; |
| } |
| // Remove the RTX header and parse the original RTP header. |
| if (packet_length < header.headerLength) |
| return; |
| if (packet_length > sizeof(restored_packet_)) |
| return; |
| if (restored_packet_in_use_) { |
| LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet."; |
| return; |
| } |
| if (!rtp_payload_registry_.RestoreOriginalPacket( |
| restored_packet_, packet, &packet_length, config_.rtp.remote_ssrc, |
| header)) { |
| LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: " |
| << header.ssrc << " payload type: " |
| << static_cast<int>(header.payloadType); |
| return; |
| } |
| restored_packet_in_use_ = true; |
| OnRecoveredPacket(restored_packet_, packet_length); |
| restored_packet_in_use_ = false; |
| } |
| } |
| |
| void RtpVideoStreamReceiver::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 = header.extension.videoRotation; |
| } |
| rtp_header.type.Video.content_type = VideoContentType::UNSPECIFIED; |
| if (header.extension.hasVideoContentType) { |
| rtp_header.type.Video.content_type = header.extension.videoContentType; |
| } |
| rtp_header.type.Video.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false}; |
| if (header.extension.has_video_timing) { |
| rtp_header.type.Video.video_timing = header.extension.video_timing; |
| } |
| rtp_header.type.Video.playout_delay = header.extension.playout_delay; |
| |
| OnReceivedPayloadData(nullptr, 0, &rtp_header); |
| } |
| |
| bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet, |
| size_t rtcp_packet_length) { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| |
| if (!receiving_) { |
| return false; |
| } |
| |
| rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
| |
| int64_t rtt = 0; |
| rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr); |
| if (rtt == 0) { |
| // Waiting for valid rtt. |
| return true; |
| } |
| uint32_t ntp_secs = 0; |
| uint32_t ntp_frac = 0; |
| uint32_t rtp_timestamp = 0; |
| if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr, |
| &rtp_timestamp) != 0) { |
| // Waiting for RTCP. |
| return true; |
| } |
| ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp); |
| |
| return true; |
| } |
| |
| void RtpVideoStreamReceiver::FrameContinuous(int64_t picture_id) { |
| if (!nack_module_) |
| return; |
| |
| int seq_num = -1; |
| { |
| rtc::CritScope lock(&last_seq_num_cs_); |
| auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id); |
| if (seq_num_it != last_seq_num_for_pic_id_.end()) |
| seq_num = seq_num_it->second; |
| } |
| if (seq_num != -1) |
| nack_module_->ClearUpTo(seq_num); |
| } |
| |
| void RtpVideoStreamReceiver::FrameDecoded(int64_t picture_id) { |
| int seq_num = -1; |
| { |
| rtc::CritScope lock(&last_seq_num_cs_); |
| auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id); |
| if (seq_num_it != last_seq_num_for_pic_id_.end()) { |
| seq_num = seq_num_it->second; |
| last_seq_num_for_pic_id_.erase(last_seq_num_for_pic_id_.begin(), |
| ++seq_num_it); |
| } |
| } |
| if (seq_num != -1) { |
| packet_buffer_->ClearTo(seq_num); |
| reference_finder_->ClearTo(seq_num); |
| } |
| } |
| |
| void RtpVideoStreamReceiver::SignalNetworkState(NetworkState state) { |
| rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode |
| : RtcpMode::kOff); |
| } |
| |
| void RtpVideoStreamReceiver::StartReceive() { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| receiving_ = true; |
| } |
| |
| void RtpVideoStreamReceiver::StopReceive() { |
| RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_); |
| receiving_ = false; |
| } |
| |
| bool RtpVideoStreamReceiver::IsPacketInOrder(const RTPHeader& header) const { |
| StreamStatistician* statistician = |
| rtp_receive_statistics_->GetStatistician(header.ssrc); |
| if (!statistician) |
| return false; |
| return statistician->IsPacketInOrder(header.sequenceNumber); |
| } |
| |
| bool RtpVideoStreamReceiver::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(config_.rtp.remote_ssrc, nullptr, nullptr, &min_rtt, nullptr); |
| return !in_order && |
| statistician->IsRetransmitOfOldPacket(header, min_rtt); |
| } |
| |
| void RtpVideoStreamReceiver::UpdateHistograms() { |
| FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter(); |
| if (counter.first_packet_time_ms == -1) |
| return; |
| |
| int64_t elapsed_sec = |
| (clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000; |
| if (elapsed_sec < metrics::kMinRunTimeInSeconds) |
| return; |
| |
| 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)); |
| } |
| } |
| |
| void RtpVideoStreamReceiver::EnableReceiveRtpHeaderExtension( |
| const std::string& extension, int id) { |
| // One-byte-extension local identifiers are in the range 1-14 inclusive. |
| RTC_DCHECK_GE(id, 1); |
| RTC_DCHECK_LE(id, 14); |
| RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); |
| RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension( |
| StringToRtpExtensionType(extension), id)); |
| } |
| |
| void RtpVideoStreamReceiver::InsertSpsPpsIntoTracker(uint8_t payload_type) { |
| auto codec_params_it = pt_codec_params_.find(payload_type); |
| if (codec_params_it == pt_codec_params_.end()) |
| return; |
| |
| LOG(LS_INFO) << "Found out of band supplied codec parameters for" |
| << " payload type: " << static_cast<int>(payload_type); |
| |
| H264SpropParameterSets sprop_decoder; |
| auto sprop_base64_it = |
| codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets); |
| |
| if (sprop_base64_it == codec_params_it->second.end()) |
| return; |
| |
| if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str())) |
| return; |
| |
| tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(), |
| sprop_decoder.pps_nalu()); |
| } |
| |
| } // namespace webrtc |