Index: webrtc/video/rtp_stream_receiver.cc |
diff --git a/webrtc/video/rtp_stream_receiver.cc b/webrtc/video/rtp_stream_receiver.cc |
deleted file mode 100644 |
index 78057771d6e5705b65e37eebdbba59ef24fa8c88..0000000000000000000000000000000000000000 |
--- a/webrtc/video/rtp_stream_receiver.cc |
+++ /dev/null |
@@ -1,683 +0,0 @@ |
-/* |
- * 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_stream_receiver.h" |
- |
-#include <vector> |
-#include <utility> |
- |
-#include "webrtc/base/checks.h" |
-#include "webrtc/base/location.h" |
-#include "webrtc/base/logging.h" |
-#include "webrtc/common_types.h" |
-#include "webrtc/config.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/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 { |
-constexpr int kPacketBufferStartSize = 32; |
-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; |
- |
-RtpStreamReceiver::RtpStreamReceiver( |
- 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(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) { |
- packet_router_->AddReceiveRtpModule(rtp_rtcp_.get()); |
- 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_->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_payload_types) { |
- RTC_DCHECK(kv.second != 0); |
- rtp_payload_registry_.SetRtxPayloadType(kv.second, kv.first); |
- } |
- } |
- |
- 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)); |
-} |
- |
-RtpStreamReceiver::~RtpStreamReceiver() { |
- if (nack_module_) { |
- process_thread_->DeRegisterModule(nack_module_.get()); |
- } |
- |
- process_thread_->DeRegisterModule(rtp_rtcp_.get()); |
- |
- packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get()); |
- UpdateHistograms(); |
-} |
- |
-bool RtpStreamReceiver::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 RtpStreamReceiver::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 RtpStreamReceiver::GetRemoteSsrc() const { |
- return rtp_receiver_->SSRC(); |
-} |
- |
-int RtpStreamReceiver::GetCsrcs(uint32_t* csrcs) const { |
- return rtp_receiver_->CSRCs(csrcs); |
-} |
- |
-RtpReceiver* RtpStreamReceiver::GetRtpReceiver() const { |
- return rtp_receiver_.get(); |
-} |
- |
-int32_t RtpStreamReceiver::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; |
- |
- // 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); |
- 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 RtpStreamReceiver::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 RtpStreamReceiver::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; |
-} |
- |
-void RtpStreamReceiver::OnIncomingSSRCChanged(const uint32_t ssrc) { |
- rtp_rtcp_->SetRemoteSSRC(ssrc); |
-} |
- |
-// This method handles both regular RTP packets and packets recovered |
-// via FlexFEC. |
-void RtpStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) { |
- { |
- rtc::CritScope lock(&receive_cs_); |
- 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)); |
- } |
-} |
- |
-int32_t RtpStreamReceiver::RequestKeyFrame() { |
- return rtp_rtcp_->RequestKeyFrame(); |
-} |
- |
-bool RtpStreamReceiver::IsUlpfecEnabled() const { |
- return config_.rtp.ulpfec.ulpfec_payload_type != -1; |
-} |
- |
-bool RtpStreamReceiver::IsRedEnabled() const { |
- return config_.rtp.ulpfec.red_payload_type != -1; |
-} |
- |
-bool RtpStreamReceiver::IsRetransmissionsEnabled() const { |
- return config_.rtp.nack.rtp_history_ms > 0; |
-} |
- |
-void RtpStreamReceiver::RequestPacketRetransmit( |
- const std::vector<uint16_t>& sequence_numbers) { |
- rtp_rtcp_->SendNack(sequence_numbers); |
-} |
- |
-int32_t RtpStreamReceiver::ResendPackets(const uint16_t* sequence_numbers, |
- uint16_t length) { |
- return rtp_rtcp_->SendNACK(sequence_numbers, length); |
-} |
- |
-void RtpStreamReceiver::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 RtpStreamReceiver::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 RtpStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
- if (nack_module_) |
- nack_module_->UpdateRtt(max_rtt_ms); |
-} |
- |
-rtc::Optional<int64_t> RtpStreamReceiver::LastReceivedPacketMs() const { |
- return packet_buffer_->LastReceivedPacketMs(); |
-} |
- |
-rtc::Optional<int64_t> RtpStreamReceiver::LastReceivedKeyframePacketMs() const { |
- return packet_buffer_->LastReceivedKeyframePacketMs(); |
-} |
- |
-void RtpStreamReceiver::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 RtpStreamReceiver::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 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; |
- rtc::CritScope lock(&receive_cs_); |
- 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, rtp_receiver_->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 RtpStreamReceiver::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.playout_delay = header.extension.playout_delay; |
- |
- OnReceivedPayloadData(nullptr, 0, &rtp_header); |
-} |
- |
-bool RtpStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet, |
- size_t rtcp_packet_length) { |
- { |
- rtc::CritScope lock(&receive_cs_); |
- 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 RtpStreamReceiver::FrameContinuous(uint16_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 RtpStreamReceiver::FrameDecoded(uint16_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 RtpStreamReceiver::SignalNetworkState(NetworkState state) { |
- rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode |
- : RtcpMode::kOff); |
-} |
- |
-void RtpStreamReceiver::StartReceive() { |
- rtc::CritScope lock(&receive_cs_); |
- receiving_ = true; |
-} |
- |
-void RtpStreamReceiver::StopReceive() { |
- rtc::CritScope lock(&receive_cs_); |
- receiving_ = false; |
-} |
- |
-bool RtpStreamReceiver::IsPacketInOrder(const RTPHeader& header) const { |
- StreamStatistician* statistician = |
- rtp_receive_statistics_->GetStatistician(header.ssrc); |
- if (!statistician) |
- return false; |
- return statistician->IsPacketInOrder(header.sequenceNumber); |
-} |
- |
-bool RtpStreamReceiver::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(), nullptr, nullptr, &min_rtt, nullptr); |
- return !in_order && |
- statistician->IsRetransmitOfOldPacket(header, min_rtt); |
-} |
- |
-void RtpStreamReceiver::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 RtpStreamReceiver::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 RtpStreamReceiver::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 |