| /* |
| * 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 "video/rtp_video_stream_receiver2.h" |
| |
| #include <algorithm> |
| #include <limits> |
| #include <memory> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/memory/memory.h" |
| #include "absl/types/optional.h" |
| #include "api/video/video_codec_type.h" |
| #include "media/base/media_constants.h" |
| #include "modules/pacing/packet_router.h" |
| #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "modules/rtp_rtcp/include/receive_statistics.h" |
| #include "modules/rtp_rtcp/include/rtp_cvo.h" |
| #include "modules/rtp_rtcp/include/ulpfec_receiver.h" |
| #include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h" |
| #include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h" |
| #include "modules/rtp_rtcp/source/rtp_format.h" |
| #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h" |
| #include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "modules/rtp_rtcp/source/rtp_rtcp_config.h" |
| #include "modules/rtp_rtcp/source/video_rtp_depacketizer.h" |
| #include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h" |
| #include "modules/video_coding/frame_object.h" |
| #include "modules/video_coding/h264_sprop_parameter_sets.h" |
| #include "modules/video_coding/h264_sps_pps_tracker.h" |
| #include "modules/video_coding/nack_requester.h" |
| #include "modules/video_coding/packet_buffer.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "system_wrappers/include/metrics.h" |
| #include "system_wrappers/include/ntp_time.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| // TODO(philipel): Change kPacketBufferStartSize back to 32 in M63 see: |
| // crbug.com/752886 |
| constexpr int kPacketBufferStartSize = 512; |
| constexpr int kPacketBufferMaxSize = 2048; |
| |
| int PacketBufferMaxSize(const FieldTrialsView& field_trials) { |
| // The group here must be a positive power of 2, in which case that is used as |
| // size. All other values shall result in the default value being used. |
| const std::string group_name = |
| field_trials.Lookup("WebRTC-PacketBufferMaxSize"); |
| int packet_buffer_max_size = kPacketBufferMaxSize; |
| if (!group_name.empty() && |
| (sscanf(group_name.c_str(), "%d", &packet_buffer_max_size) != 1 || |
| packet_buffer_max_size <= 0 || |
| // Verify that the number is a positive power of 2. |
| (packet_buffer_max_size & (packet_buffer_max_size - 1)) != 0)) { |
| RTC_LOG(LS_WARNING) << "Invalid packet buffer max size: " << group_name; |
| packet_buffer_max_size = kPacketBufferMaxSize; |
| } |
| return packet_buffer_max_size; |
| } |
| |
| std::unique_ptr<ModuleRtpRtcpImpl2> CreateRtpRtcpModule( |
| Clock* clock, |
| ReceiveStatistics* receive_statistics, |
| Transport* outgoing_transport, |
| RtcpRttStats* rtt_stats, |
| RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
| RtcpCnameCallback* rtcp_cname_callback, |
| bool non_sender_rtt_measurement, |
| uint32_t local_ssrc) { |
| RtpRtcpInterface::Configuration configuration; |
| configuration.clock = clock; |
| configuration.audio = false; |
| configuration.receiver_only = true; |
| configuration.receive_statistics = receive_statistics; |
| configuration.outgoing_transport = outgoing_transport; |
| configuration.rtt_stats = rtt_stats; |
| configuration.rtcp_packet_type_counter_observer = |
| rtcp_packet_type_counter_observer; |
| configuration.rtcp_cname_callback = rtcp_cname_callback; |
| configuration.local_media_ssrc = local_ssrc; |
| configuration.non_sender_rtt_measurement = non_sender_rtt_measurement; |
| |
| std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp = |
| ModuleRtpRtcpImpl2::Create(configuration); |
| rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
| |
| return rtp_rtcp; |
| } |
| |
| std::unique_ptr<NackRequester> MaybeConstructNackModule( |
| TaskQueueBase* current_queue, |
| NackPeriodicProcessor* nack_periodic_processor, |
| const VideoReceiveStreamInterface::Config& config, |
| Clock* clock, |
| NackSender* nack_sender, |
| KeyFrameRequestSender* keyframe_request_sender, |
| const FieldTrialsView& field_trials) { |
| if (config.rtp.nack.rtp_history_ms == 0) |
| return nullptr; |
| |
| // TODO(bugs.webrtc.org/12420): pass rtp_history_ms to the nack module. |
| return std::make_unique<NackRequester>(current_queue, nack_periodic_processor, |
| clock, nack_sender, |
| keyframe_request_sender, field_trials); |
| } |
| |
| static const int kPacketLogIntervalMs = 10000; |
| |
| } // namespace |
| |
| RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RtcpFeedbackBuffer( |
| KeyFrameRequestSender* key_frame_request_sender, |
| NackSender* nack_sender, |
| LossNotificationSender* loss_notification_sender) |
| : key_frame_request_sender_(key_frame_request_sender), |
| nack_sender_(nack_sender), |
| loss_notification_sender_(loss_notification_sender), |
| request_key_frame_(false) { |
| RTC_DCHECK(key_frame_request_sender_); |
| RTC_DCHECK(nack_sender_); |
| RTC_DCHECK(loss_notification_sender_); |
| packet_sequence_checker_.Detach(); |
| } |
| |
| void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RequestKeyFrame() { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| request_key_frame_ = true; |
| } |
| |
| void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendNack( |
| const std::vector<uint16_t>& sequence_numbers, |
| bool buffering_allowed) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK(!sequence_numbers.empty()); |
| nack_sequence_numbers_.insert(nack_sequence_numbers_.end(), |
| sequence_numbers.cbegin(), |
| sequence_numbers.cend()); |
| if (!buffering_allowed) { |
| // Note that while *buffering* is not allowed, *batching* is, meaning that |
| // previously buffered messages may be sent along with the current message. |
| SendBufferedRtcpFeedback(); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendLossNotification( |
| uint16_t last_decoded_seq_num, |
| uint16_t last_received_seq_num, |
| bool decodability_flag, |
| bool buffering_allowed) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK(buffering_allowed); |
| RTC_DCHECK(!lntf_state_) |
| << "SendLossNotification() called twice in a row with no call to " |
| "SendBufferedRtcpFeedback() in between."; |
| lntf_state_ = absl::make_optional<LossNotificationState>( |
| last_decoded_seq_num, last_received_seq_num, decodability_flag); |
| } |
| |
| void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendBufferedRtcpFeedback() { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| |
| bool request_key_frame = false; |
| std::vector<uint16_t> nack_sequence_numbers; |
| absl::optional<LossNotificationState> lntf_state; |
| |
| std::swap(request_key_frame, request_key_frame_); |
| std::swap(nack_sequence_numbers, nack_sequence_numbers_); |
| std::swap(lntf_state, lntf_state_); |
| |
| if (lntf_state) { |
| // If either a NACK or a key frame request is sent, we should buffer |
| // the LNTF and wait for them (NACK or key frame request) to trigger |
| // the compound feedback message. |
| // Otherwise, the LNTF should be sent out immediately. |
| const bool buffering_allowed = |
| request_key_frame || !nack_sequence_numbers.empty(); |
| |
| loss_notification_sender_->SendLossNotification( |
| lntf_state->last_decoded_seq_num, lntf_state->last_received_seq_num, |
| lntf_state->decodability_flag, buffering_allowed); |
| } |
| |
| if (request_key_frame) { |
| key_frame_request_sender_->RequestKeyFrame(); |
| } else if (!nack_sequence_numbers.empty()) { |
| nack_sender_->SendNack(nack_sequence_numbers, true); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::ClearLossNotificationState() { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| lntf_state_.reset(); |
| } |
| |
| RtpVideoStreamReceiver2::RtpVideoStreamReceiver2( |
| TaskQueueBase* current_queue, |
| Clock* clock, |
| Transport* transport, |
| RtcpRttStats* rtt_stats, |
| PacketRouter* packet_router, |
| const VideoReceiveStreamInterface::Config* config, |
| ReceiveStatistics* rtp_receive_statistics, |
| RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
| RtcpCnameCallback* rtcp_cname_callback, |
| NackPeriodicProcessor* nack_periodic_processor, |
| OnCompleteFrameCallback* complete_frame_callback, |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor, |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer, |
| const FieldTrialsView& field_trials) |
| : field_trials_(field_trials), |
| clock_(clock), |
| config_(*config), |
| packet_router_(packet_router), |
| ntp_estimator_(clock), |
| rtp_header_extensions_(config_.rtp.extensions), |
| forced_playout_delay_max_ms_("max_ms", absl::nullopt), |
| forced_playout_delay_min_ms_("min_ms", absl::nullopt), |
| rtp_receive_statistics_(rtp_receive_statistics), |
| ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, |
| this, |
| config->rtp.extensions)), |
| packet_sink_(config->rtp.packet_sink_), |
| receiving_(false), |
| last_packet_log_ms_(-1), |
| rtp_rtcp_(CreateRtpRtcpModule( |
| clock, |
| rtp_receive_statistics_, |
| transport, |
| rtt_stats, |
| rtcp_packet_type_counter_observer, |
| rtcp_cname_callback, |
| config_.rtp.rtcp_xr.receiver_reference_time_report, |
| config_.rtp.local_ssrc)), |
| complete_frame_callback_(complete_frame_callback), |
| keyframe_request_method_(config_.rtp.keyframe_method), |
| // TODO(bugs.webrtc.org/10336): Let `rtcp_feedback_buffer_` communicate |
| // directly with `rtp_rtcp_`. |
| rtcp_feedback_buffer_(this, this, this), |
| nack_module_(MaybeConstructNackModule(current_queue, |
| nack_periodic_processor, |
| config_, |
| clock_, |
| &rtcp_feedback_buffer_, |
| &rtcp_feedback_buffer_, |
| field_trials_)), |
| packet_buffer_(kPacketBufferStartSize, |
| PacketBufferMaxSize(field_trials_)), |
| reference_finder_(std::make_unique<RtpFrameReferenceFinder>()), |
| has_received_frame_(false), |
| frames_decryptable_(false), |
| absolute_capture_time_interpolator_(clock) { |
| packet_sequence_checker_.Detach(); |
| constexpr bool remb_candidate = true; |
| if (packet_router_) |
| packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate); |
| |
| RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff) |
| << "A stream should not be configured with RTCP disabled. This value is " |
| "reserved for internal usage."; |
| // 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_->SetRemoteSSRC(config_.rtp.remote_ssrc); |
| |
| if (config_.rtp.nack.rtp_history_ms > 0) { |
| static constexpr int kMaxPacketAgeToNack = 450; |
| rtp_receive_statistics_->SetMaxReorderingThreshold(config_.rtp.remote_ssrc, |
| kMaxPacketAgeToNack); |
| } |
| ParseFieldTrial( |
| {&forced_playout_delay_max_ms_, &forced_playout_delay_min_ms_}, |
| field_trials_.Lookup("WebRTC-ForcePlayoutDelay")); |
| |
| if (config_.rtp.lntf.enabled) { |
| loss_notification_controller_ = |
| std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_, |
| &rtcp_feedback_buffer_); |
| } |
| |
| // Only construct the encrypted receiver if frame encryption is enabled. |
| if (config_.crypto_options.sframe.require_frame_encryption) { |
| buffered_frame_decryptor_ = |
| std::make_unique<BufferedFrameDecryptor>(this, this, field_trials_); |
| if (frame_decryptor != nullptr) { |
| buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor)); |
| } |
| } |
| |
| if (frame_transformer) { |
| frame_transformer_delegate_ = |
| rtc::make_ref_counted<RtpVideoStreamReceiverFrameTransformerDelegate>( |
| this, std::move(frame_transformer), rtc::Thread::Current(), |
| config_.rtp.remote_ssrc); |
| frame_transformer_delegate_->Init(); |
| } |
| } |
| |
| RtpVideoStreamReceiver2::~RtpVideoStreamReceiver2() { |
| if (packet_router_) |
| packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get()); |
| UpdateHistograms(); |
| if (frame_transformer_delegate_) |
| frame_transformer_delegate_->Reset(); |
| } |
| |
| void RtpVideoStreamReceiver2::AddReceiveCodec( |
| uint8_t payload_type, |
| VideoCodecType video_codec, |
| const std::map<std::string, std::string>& codec_params, |
| bool raw_payload) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (codec_params.count(cricket::kH264FmtpSpsPpsIdrInKeyframe) || |
| field_trials_.IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) { |
| packet_buffer_.ForceSpsPpsIdrIsH264Keyframe(); |
| } |
| payload_type_map_.emplace( |
| payload_type, raw_payload ? std::make_unique<VideoRtpDepacketizerRaw>() |
| : CreateVideoRtpDepacketizer(video_codec)); |
| pt_codec_params_.emplace(payload_type, codec_params); |
| } |
| |
| absl::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| Syncable::Info info; |
| if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs, |
| &info.capture_time_ntp_frac, |
| /*rtcp_arrival_time_secs=*/nullptr, |
| /*rtcp_arrival_time_frac=*/nullptr, |
| &info.capture_time_source_clock) != 0) { |
| return absl::nullopt; |
| } |
| |
| if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_) { |
| return absl::nullopt; |
| } |
| info.latest_received_capture_timestamp = *last_received_rtp_timestamp_; |
| info.latest_receive_time_ms = last_received_rtp_system_time_->ms(); |
| |
| // Leaves info.current_delay_ms uninitialized. |
| return info; |
| } |
| |
| RtpVideoStreamReceiver2::ParseGenericDependenciesResult |
| RtpVideoStreamReceiver2::ParseGenericDependenciesExtension( |
| const RtpPacketReceived& rtp_packet, |
| RTPVideoHeader* video_header) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (rtp_packet.HasExtension<RtpDependencyDescriptorExtension>()) { |
| webrtc::DependencyDescriptor dependency_descriptor; |
| if (!rtp_packet.GetExtension<RtpDependencyDescriptorExtension>( |
| video_structure_.get(), &dependency_descriptor)) { |
| // Descriptor is there, but failed to parse. Either it is invalid, |
| // or too old packet (after relevant video_structure_ changed), |
| // or too new packet (before relevant video_structure_ arrived). |
| // Drop such packet to be on the safe side. |
| // TODO(bugs.webrtc.org/10342): Stash too new packet. |
| RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() |
| << " Failed to parse dependency descriptor."; |
| return kDropPacket; |
| } |
| if (dependency_descriptor.attached_structure != nullptr && |
| !dependency_descriptor.first_packet_in_frame) { |
| RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc() |
| << "Invalid dependency descriptor: structure " |
| "attached to non first packet of a frame."; |
| return kDropPacket; |
| } |
| video_header->is_first_packet_in_frame = |
| dependency_descriptor.first_packet_in_frame; |
| video_header->is_last_packet_in_frame = |
| dependency_descriptor.last_packet_in_frame; |
| |
| int64_t frame_id = |
| frame_id_unwrapper_.Unwrap(dependency_descriptor.frame_number); |
| auto& generic_descriptor_info = video_header->generic.emplace(); |
| generic_descriptor_info.frame_id = frame_id; |
| generic_descriptor_info.spatial_index = |
| dependency_descriptor.frame_dependencies.spatial_id; |
| generic_descriptor_info.temporal_index = |
| dependency_descriptor.frame_dependencies.temporal_id; |
| for (int fdiff : dependency_descriptor.frame_dependencies.frame_diffs) { |
| generic_descriptor_info.dependencies.push_back(frame_id - fdiff); |
| } |
| generic_descriptor_info.decode_target_indications = |
| dependency_descriptor.frame_dependencies.decode_target_indications; |
| if (dependency_descriptor.resolution) { |
| video_header->width = dependency_descriptor.resolution->Width(); |
| video_header->height = dependency_descriptor.resolution->Height(); |
| } |
| |
| // FrameDependencyStructure is sent in dependency descriptor of the first |
| // packet of a key frame and required for parsed dependency descriptor in |
| // all the following packets until next key frame. |
| // Save it if there is a (potentially) new structure. |
| if (dependency_descriptor.attached_structure) { |
| RTC_DCHECK(dependency_descriptor.first_packet_in_frame); |
| if (video_structure_frame_id_ > frame_id) { |
| RTC_LOG(LS_WARNING) |
| << "Arrived key frame with id " << frame_id << " and structure id " |
| << dependency_descriptor.attached_structure->structure_id |
| << " is older than the latest received key frame with id " |
| << *video_structure_frame_id_ << " and structure id " |
| << video_structure_->structure_id; |
| return kDropPacket; |
| } |
| video_structure_ = std::move(dependency_descriptor.attached_structure); |
| video_structure_frame_id_ = frame_id; |
| video_header->frame_type = VideoFrameType::kVideoFrameKey; |
| } else { |
| video_header->frame_type = VideoFrameType::kVideoFrameDelta; |
| } |
| return kHasGenericDescriptor; |
| } |
| |
| RtpGenericFrameDescriptor generic_frame_descriptor; |
| if (!rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>( |
| &generic_frame_descriptor)) { |
| return kNoGenericDescriptor; |
| } |
| |
| video_header->is_first_packet_in_frame = |
| generic_frame_descriptor.FirstPacketInSubFrame(); |
| video_header->is_last_packet_in_frame = |
| generic_frame_descriptor.LastPacketInSubFrame(); |
| |
| if (generic_frame_descriptor.FirstPacketInSubFrame()) { |
| video_header->frame_type = |
| generic_frame_descriptor.FrameDependenciesDiffs().empty() |
| ? VideoFrameType::kVideoFrameKey |
| : VideoFrameType::kVideoFrameDelta; |
| |
| auto& generic_descriptor_info = video_header->generic.emplace(); |
| int64_t frame_id = |
| frame_id_unwrapper_.Unwrap(generic_frame_descriptor.FrameId()); |
| generic_descriptor_info.frame_id = frame_id; |
| generic_descriptor_info.spatial_index = |
| generic_frame_descriptor.SpatialLayer(); |
| generic_descriptor_info.temporal_index = |
| generic_frame_descriptor.TemporalLayer(); |
| for (uint16_t fdiff : generic_frame_descriptor.FrameDependenciesDiffs()) { |
| generic_descriptor_info.dependencies.push_back(frame_id - fdiff); |
| } |
| } |
| video_header->width = generic_frame_descriptor.Width(); |
| video_header->height = generic_frame_descriptor.Height(); |
| return kHasGenericDescriptor; |
| } |
| |
| void RtpVideoStreamReceiver2::OnReceivedPayloadData( |
| rtc::CopyOnWriteBuffer codec_payload, |
| const RtpPacketReceived& rtp_packet, |
| const RTPVideoHeader& video) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| |
| auto packet = |
| std::make_unique<video_coding::PacketBuffer::Packet>(rtp_packet, video); |
| |
| int64_t unwrapped_rtp_seq_num = |
| rtp_seq_num_unwrapper_.Unwrap(rtp_packet.SequenceNumber()); |
| auto& packet_info = |
| packet_infos_ |
| .emplace( |
| unwrapped_rtp_seq_num, |
| RtpPacketInfo( |
| rtp_packet.Ssrc(), rtp_packet.Csrcs(), rtp_packet.Timestamp(), |
| /*audio_level=*/absl::nullopt, |
| rtp_packet.GetExtension<AbsoluteCaptureTimeExtension>(), |
| /*receive_time_ms=*/clock_->CurrentTime())) |
| .first->second; |
| |
| // Try to extrapolate absolute capture time if it is missing. |
| packet_info.set_absolute_capture_time( |
| absolute_capture_time_interpolator_.OnReceivePacket( |
| AbsoluteCaptureTimeInterpolator::GetSource(packet_info.ssrc(), |
| packet_info.csrcs()), |
| packet_info.rtp_timestamp(), |
| // Assume frequency is the same one for all video frames. |
| kVideoPayloadTypeFrequency, packet_info.absolute_capture_time())); |
| |
| RTPVideoHeader& video_header = packet->video_header; |
| video_header.rotation = kVideoRotation_0; |
| video_header.content_type = VideoContentType::UNSPECIFIED; |
| video_header.video_timing.flags = VideoSendTiming::kInvalid; |
| video_header.is_last_packet_in_frame |= rtp_packet.Marker(); |
| |
| rtp_packet.GetExtension<VideoOrientation>(&video_header.rotation); |
| rtp_packet.GetExtension<VideoContentTypeExtension>( |
| &video_header.content_type); |
| rtp_packet.GetExtension<VideoTimingExtension>(&video_header.video_timing); |
| if (forced_playout_delay_max_ms_ && forced_playout_delay_min_ms_) { |
| video_header.playout_delay.max_ms = *forced_playout_delay_max_ms_; |
| video_header.playout_delay.min_ms = *forced_playout_delay_min_ms_; |
| } else { |
| rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay); |
| } |
| |
| ParseGenericDependenciesResult generic_descriptor_state = |
| ParseGenericDependenciesExtension(rtp_packet, &video_header); |
| |
| if (!rtp_packet.recovered()) { |
| UpdatePacketReceiveTimestamps( |
| rtp_packet, video_header.frame_type == VideoFrameType::kVideoFrameKey); |
| } |
| |
| if (generic_descriptor_state == kDropPacket) { |
| Timestamp now = clock_->CurrentTime(); |
| if (video_structure_ == nullptr && |
| next_keyframe_request_for_missing_video_structure_ < now) { |
| // No video structure received yet, most likely part of the initial |
| // keyframe was lost. |
| RequestKeyFrame(); |
| next_keyframe_request_for_missing_video_structure_ = |
| now + TimeDelta::Seconds(1); |
| } |
| return; |
| } |
| |
| // Color space should only be transmitted in the last packet of a frame, |
| // therefore, neglect it otherwise so that last_color_space_ is not reset by |
| // mistake. |
| if (video_header.is_last_packet_in_frame) { |
| video_header.color_space = rtp_packet.GetExtension<ColorSpaceExtension>(); |
| if (video_header.color_space || |
| video_header.frame_type == VideoFrameType::kVideoFrameKey) { |
| // Store color space since it's only transmitted when changed or for key |
| // frames. Color space will be cleared if a key frame is transmitted |
| // without color space information. |
| last_color_space_ = video_header.color_space; |
| } else if (last_color_space_) { |
| video_header.color_space = last_color_space_; |
| } |
| } |
| video_header.video_frame_tracking_id = |
| rtp_packet.GetExtension<VideoFrameTrackingIdExtension>(); |
| |
| if (loss_notification_controller_) { |
| if (rtp_packet.recovered()) { |
| // TODO(bugs.webrtc.org/10336): Implement support for reordering. |
| RTC_LOG(LS_INFO) |
| << "LossNotificationController does not support reordering."; |
| } else if (generic_descriptor_state == kNoGenericDescriptor) { |
| RTC_LOG(LS_WARNING) << "LossNotificationController requires generic " |
| "frame descriptor, but it is missing."; |
| } else { |
| if (video_header.is_first_packet_in_frame) { |
| RTC_DCHECK(video_header.generic); |
| LossNotificationController::FrameDetails frame; |
| frame.is_keyframe = |
| video_header.frame_type == VideoFrameType::kVideoFrameKey; |
| frame.frame_id = video_header.generic->frame_id; |
| frame.frame_dependencies = video_header.generic->dependencies; |
| loss_notification_controller_->OnReceivedPacket( |
| rtp_packet.SequenceNumber(), &frame); |
| } else { |
| loss_notification_controller_->OnReceivedPacket( |
| rtp_packet.SequenceNumber(), nullptr); |
| } |
| } |
| } |
| |
| if (nack_module_) { |
| const bool is_keyframe = |
| video_header.is_first_packet_in_frame && |
| video_header.frame_type == VideoFrameType::kVideoFrameKey; |
| |
| packet->times_nacked = nack_module_->OnReceivedPacket( |
| rtp_packet.SequenceNumber(), is_keyframe, rtp_packet.recovered()); |
| } else { |
| packet->times_nacked = -1; |
| } |
| |
| if (codec_payload.size() == 0) { |
| NotifyReceiverOfEmptyPacket(packet->seq_num); |
| rtcp_feedback_buffer_.SendBufferedRtcpFeedback(); |
| return; |
| } |
| |
| 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->payload_type != last_payload_type_) { |
| last_payload_type_ = packet->payload_type; |
| InsertSpsPpsIntoTracker(packet->payload_type); |
| } |
| |
| video_coding::H264SpsPpsTracker::FixedBitstream fixed = |
| tracker_.CopyAndFixBitstream( |
| rtc::MakeArrayView(codec_payload.cdata(), codec_payload.size()), |
| &packet->video_header); |
| |
| switch (fixed.action) { |
| case video_coding::H264SpsPpsTracker::kRequestKeyframe: |
| rtcp_feedback_buffer_.RequestKeyFrame(); |
| rtcp_feedback_buffer_.SendBufferedRtcpFeedback(); |
| [[fallthrough]]; |
| case video_coding::H264SpsPpsTracker::kDrop: |
| return; |
| case video_coding::H264SpsPpsTracker::kInsert: |
| packet->video_payload = std::move(fixed.bitstream); |
| break; |
| } |
| |
| } else { |
| packet->video_payload = std::move(codec_payload); |
| } |
| |
| rtcp_feedback_buffer_.SendBufferedRtcpFeedback(); |
| frame_counter_.Add(packet->timestamp); |
| OnInsertedPacket(packet_buffer_.InsertPacket(std::move(packet))); |
| } |
| |
| void RtpVideoStreamReceiver2::OnRecoveredPacket(const uint8_t* rtp_packet, |
| size_t rtp_packet_length) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| |
| RtpPacketReceived packet; |
| if (!packet.Parse(rtp_packet, rtp_packet_length)) |
| return; |
| if (packet.PayloadType() == config_.rtp.red_payload_type) { |
| RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation"; |
| return; |
| } |
| |
| packet.IdentifyExtensions(rtp_header_extensions_); |
| packet.set_payload_type_frequency(kVideoPayloadTypeFrequency); |
| // TODO(bugs.webrtc.org/7135): UlpfecReceiverImpl::ProcessReceivedFec passes |
| // both original (decapsulated) media packets and recovered packets to this |
| // callback. We need a way to distinguish, for setting packet.recovered() |
| // correctly. Ideally, move RED decapsulation out of the Ulpfec |
| // implementation. |
| |
| ReceivePacket(packet); |
| } |
| |
| // This method handles both regular RTP packets and packets recovered |
| // via FlexFEC. |
| void RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived& packet) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| |
| if (!receiving_) |
| return; |
| |
| ReceivePacket(packet); |
| |
| // 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()) { |
| rtp_receive_statistics_->OnRtpPacket(packet); |
| } |
| |
| if (packet_sink_) { |
| packet_sink_->OnRtpPacket(packet); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::RequestKeyFrame() { |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| // TODO(bugs.webrtc.org/10336): Allow the sender to ignore key frame requests |
| // issued by anything other than the LossNotificationController if it (the |
| // sender) is relying on LNTF alone. |
| if (keyframe_request_method_ == KeyFrameReqMethod::kPliRtcp) { |
| rtp_rtcp_->SendPictureLossIndication(); |
| } else if (keyframe_request_method_ == KeyFrameReqMethod::kFirRtcp) { |
| rtp_rtcp_->SendFullIntraRequest(); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::SendNack( |
| const std::vector<uint16_t>& sequence_numbers, |
| bool /*buffering_allowed*/) { |
| rtp_rtcp_->SendNack(sequence_numbers); |
| } |
| |
| void RtpVideoStreamReceiver2::SendLossNotification( |
| uint16_t last_decoded_seq_num, |
| uint16_t last_received_seq_num, |
| bool decodability_flag, |
| bool buffering_allowed) { |
| RTC_DCHECK(config_.rtp.lntf.enabled); |
| rtp_rtcp_->SendLossNotification(last_decoded_seq_num, last_received_seq_num, |
| decodability_flag, buffering_allowed); |
| } |
| |
| bool RtpVideoStreamReceiver2::IsUlpfecEnabled() const { |
| return config_.rtp.ulpfec_payload_type != -1; |
| } |
| |
| bool RtpVideoStreamReceiver2::IsRetransmissionsEnabled() const { |
| return config_.rtp.nack.rtp_history_ms > 0; |
| } |
| |
| bool RtpVideoStreamReceiver2::IsDecryptable() const { |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| return frames_decryptable_; |
| } |
| |
| void RtpVideoStreamReceiver2::OnInsertedPacket( |
| video_coding::PacketBuffer::InsertResult result) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| video_coding::PacketBuffer::Packet* first_packet = nullptr; |
| int max_nack_count; |
| int64_t min_recv_time; |
| int64_t max_recv_time; |
| std::vector<rtc::ArrayView<const uint8_t>> payloads; |
| RtpPacketInfos::vector_type packet_infos; |
| |
| bool frame_boundary = true; |
| for (auto& packet : result.packets) { |
| // PacketBuffer promisses frame boundaries are correctly set on each |
| // packet. Document that assumption with the DCHECKs. |
| RTC_DCHECK_EQ(frame_boundary, packet->is_first_packet_in_frame()); |
| int64_t unwrapped_rtp_seq_num = |
| rtp_seq_num_unwrapper_.Unwrap(packet->seq_num); |
| RTC_DCHECK(packet_infos_.count(unwrapped_rtp_seq_num) > 0); |
| RtpPacketInfo& packet_info = packet_infos_[unwrapped_rtp_seq_num]; |
| if (packet->is_first_packet_in_frame()) { |
| first_packet = packet.get(); |
| max_nack_count = packet->times_nacked; |
| min_recv_time = packet_info.receive_time().ms(); |
| max_recv_time = packet_info.receive_time().ms(); |
| } else { |
| max_nack_count = std::max(max_nack_count, packet->times_nacked); |
| min_recv_time = std::min(min_recv_time, packet_info.receive_time().ms()); |
| max_recv_time = std::max(max_recv_time, packet_info.receive_time().ms()); |
| } |
| payloads.emplace_back(packet->video_payload); |
| packet_infos.push_back(packet_info); |
| |
| frame_boundary = packet->is_last_packet_in_frame(); |
| if (packet->is_last_packet_in_frame()) { |
| auto depacketizer_it = payload_type_map_.find(first_packet->payload_type); |
| RTC_CHECK(depacketizer_it != payload_type_map_.end()); |
| |
| rtc::scoped_refptr<EncodedImageBuffer> bitstream = |
| depacketizer_it->second->AssembleFrame(payloads); |
| if (!bitstream) { |
| // Failed to assemble a frame. Discard and continue. |
| continue; |
| } |
| |
| const video_coding::PacketBuffer::Packet& last_packet = *packet; |
| OnAssembledFrame(std::make_unique<RtpFrameObject>( |
| first_packet->seq_num, // |
| last_packet.seq_num, // |
| last_packet.marker_bit, // |
| max_nack_count, // |
| min_recv_time, // |
| max_recv_time, // |
| first_packet->timestamp, // |
| ntp_estimator_.Estimate(first_packet->timestamp), // |
| last_packet.video_header.video_timing, // |
| first_packet->payload_type, // |
| first_packet->codec(), // |
| last_packet.video_header.rotation, // |
| last_packet.video_header.content_type, // |
| first_packet->video_header, // |
| last_packet.video_header.color_space, // |
| RtpPacketInfos(std::move(packet_infos)), // |
| std::move(bitstream))); |
| payloads.clear(); |
| packet_infos.clear(); |
| } |
| } |
| RTC_DCHECK(frame_boundary); |
| if (result.buffer_cleared) { |
| last_received_rtp_system_time_.reset(); |
| last_received_keyframe_rtp_system_time_.reset(); |
| last_received_keyframe_rtp_timestamp_.reset(); |
| packet_infos_.clear(); |
| RequestKeyFrame(); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::OnAssembledFrame( |
| std::unique_ptr<RtpFrameObject> frame) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK(frame); |
| |
| const absl::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor = |
| frame->GetRtpVideoHeader().generic; |
| |
| if (loss_notification_controller_ && descriptor) { |
| loss_notification_controller_->OnAssembledFrame( |
| frame->first_seq_num(), descriptor->frame_id, |
| absl::c_linear_search(descriptor->decode_target_indications, |
| DecodeTargetIndication::kDiscardable), |
| descriptor->dependencies); |
| } |
| |
| // If frames arrive before a key frame, they would not be decodable. |
| // In that case, request a key frame ASAP. |
| if (!has_received_frame_) { |
| if (frame->FrameType() != VideoFrameType::kVideoFrameKey) { |
| // `loss_notification_controller_`, if present, would have already |
| // requested a key frame when the first packet for the non-key frame |
| // had arrived, so no need to replicate the request. |
| if (!loss_notification_controller_) { |
| RequestKeyFrame(); |
| } |
| } |
| has_received_frame_ = true; |
| } |
| |
| // Reset `reference_finder_` if `frame` is new and the codec have changed. |
| if (current_codec_) { |
| bool frame_is_newer = |
| AheadOf(frame->Timestamp(), last_assembled_frame_rtp_timestamp_); |
| |
| if (frame->codec_type() != current_codec_) { |
| if (frame_is_newer) { |
| // When we reset the `reference_finder_` we don't want new picture ids |
| // to overlap with old picture ids. To ensure that doesn't happen we |
| // start from the `last_completed_picture_id_` and add an offset in case |
| // of reordering. |
| reference_finder_ = std::make_unique<RtpFrameReferenceFinder>( |
| last_completed_picture_id_ + std::numeric_limits<uint16_t>::max()); |
| current_codec_ = frame->codec_type(); |
| } else { |
| // Old frame from before the codec switch, discard it. |
| return; |
| } |
| } |
| |
| if (frame_is_newer) { |
| last_assembled_frame_rtp_timestamp_ = frame->Timestamp(); |
| } |
| } else { |
| current_codec_ = frame->codec_type(); |
| last_assembled_frame_rtp_timestamp_ = frame->Timestamp(); |
| } |
| |
| if (buffered_frame_decryptor_ != nullptr) { |
| buffered_frame_decryptor_->ManageEncryptedFrame(std::move(frame)); |
| } else if (frame_transformer_delegate_) { |
| frame_transformer_delegate_->TransformFrame(std::move(frame)); |
| } else { |
| OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame))); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::OnCompleteFrames( |
| RtpFrameReferenceFinder::ReturnVector frames) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| for (auto& frame : frames) { |
| last_seq_num_for_pic_id_[frame->Id()] = frame->last_seq_num(); |
| |
| last_completed_picture_id_ = |
| std::max(last_completed_picture_id_, frame->Id()); |
| complete_frame_callback_->OnCompleteFrame(std::move(frame)); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::OnDecryptedFrame( |
| std::unique_ptr<RtpFrameObject> frame) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame))); |
| } |
| |
| void RtpVideoStreamReceiver2::OnDecryptionStatusChange( |
| FrameDecryptorInterface::Status status) { |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| // Called from BufferedFrameDecryptor::DecryptFrame. |
| frames_decryptable_ = |
| (status == FrameDecryptorInterface::Status::kOk) || |
| (status == FrameDecryptorInterface::Status::kRecoverable); |
| } |
| |
| void RtpVideoStreamReceiver2::SetFrameDecryptor( |
| rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) { |
| // TODO(bugs.webrtc.org/11993): Update callers or post the operation over to |
| // the network thread. |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (buffered_frame_decryptor_ == nullptr) { |
| buffered_frame_decryptor_ = |
| std::make_unique<BufferedFrameDecryptor>(this, this, field_trials_); |
| } |
| buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor)); |
| } |
| |
| void RtpVideoStreamReceiver2::SetDepacketizerToDecoderFrameTransformer( |
| rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) { |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| frame_transformer_delegate_ = |
| rtc::make_ref_counted<RtpVideoStreamReceiverFrameTransformerDelegate>( |
| this, std::move(frame_transformer), rtc::Thread::Current(), |
| config_.rtp.remote_ssrc); |
| frame_transformer_delegate_->Init(); |
| } |
| |
| void RtpVideoStreamReceiver2::SetRtpExtensions( |
| const std::vector<RtpExtension>& extensions) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| rtp_header_extensions_.Reset(extensions); |
| } |
| |
| const RtpHeaderExtensionMap& RtpVideoStreamReceiver2::GetRtpExtensions() const { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| return rtp_header_extensions_; |
| } |
| |
| void RtpVideoStreamReceiver2::UpdateRtt(int64_t max_rtt_ms) { |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| if (nack_module_) |
| nack_module_->UpdateRtt(max_rtt_ms); |
| } |
| |
| void RtpVideoStreamReceiver2::OnLocalSsrcChange(uint32_t local_ssrc) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| rtp_rtcp_->SetLocalSsrc(local_ssrc); |
| } |
| |
| void RtpVideoStreamReceiver2::SetRtcpMode(RtcpMode mode) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| rtp_rtcp_->SetRTCPStatus(mode); |
| } |
| |
| void RtpVideoStreamReceiver2::SetPacketSink( |
| RtpPacketSinkInterface* packet_sink) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| packet_sink_ = packet_sink; |
| } |
| |
| void RtpVideoStreamReceiver2::SetLossNotificationEnabled(bool enabled) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (enabled && !loss_notification_controller_) { |
| loss_notification_controller_ = |
| std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_, |
| &rtcp_feedback_buffer_); |
| } else if (!enabled && loss_notification_controller_) { |
| loss_notification_controller_.reset(); |
| rtcp_feedback_buffer_.ClearLossNotificationState(); |
| } |
| } |
| |
| absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (last_received_rtp_system_time_) { |
| return absl::optional<int64_t>(last_received_rtp_system_time_->ms()); |
| } |
| return absl::nullopt; |
| } |
| |
| absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs() |
| const { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (last_received_keyframe_rtp_system_time_) { |
| return absl::optional<int64_t>( |
| last_received_keyframe_rtp_system_time_->ms()); |
| } |
| return absl::nullopt; |
| } |
| |
| void RtpVideoStreamReceiver2::ManageFrame( |
| std::unique_ptr<RtpFrameObject> frame) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| OnCompleteFrames(reference_finder_->ManageFrame(std::move(frame))); |
| } |
| |
| void RtpVideoStreamReceiver2::ReceivePacket(const RtpPacketReceived& packet) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| if (packet.payload_size() == 0) { |
| // Padding or keep-alive packet. |
| // TODO(nisse): Could drop empty packets earlier, but need to figure out how |
| // they should be counted in stats. |
| NotifyReceiverOfEmptyPacket(packet.SequenceNumber()); |
| return; |
| } |
| if (packet.PayloadType() == config_.rtp.red_payload_type) { |
| ParseAndHandleEncapsulatingHeader(packet); |
| return; |
| } |
| |
| const auto type_it = payload_type_map_.find(packet.PayloadType()); |
| if (type_it == payload_type_map_.end()) { |
| return; |
| } |
| absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload = |
| type_it->second->Parse(packet.PayloadBuffer()); |
| if (parsed_payload == absl::nullopt) { |
| RTC_LOG(LS_WARNING) << "Failed parsing payload."; |
| return; |
| } |
| |
| OnReceivedPayloadData(std::move(parsed_payload->video_payload), packet, |
| parsed_payload->video_header); |
| } |
| |
| void RtpVideoStreamReceiver2::ParseAndHandleEncapsulatingHeader( |
| const RtpPacketReceived& packet) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (packet.PayloadType() == config_.rtp.red_payload_type && |
| packet.payload_size() > 0) { |
| if (packet.payload()[0] == config_.rtp.ulpfec_payload_type) { |
| // Notify video_receiver about received FEC packets to avoid NACKing these |
| // packets. |
| NotifyReceiverOfEmptyPacket(packet.SequenceNumber()); |
| } |
| if (!ulpfec_receiver_->AddReceivedRedPacket( |
| packet, config_.rtp.ulpfec_payload_type)) { |
| return; |
| } |
| ulpfec_receiver_->ProcessReceivedFec(); |
| } |
| } |
| |
| // 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. |
| void RtpVideoStreamReceiver2::NotifyReceiverOfEmptyPacket(uint16_t seq_num) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| |
| OnCompleteFrames(reference_finder_->PaddingReceived(seq_num)); |
| |
| OnInsertedPacket(packet_buffer_.InsertPadding(seq_num)); |
| if (nack_module_) { |
| nack_module_->OnReceivedPacket(seq_num, /* is_keyframe = */ false, |
| /* is _recovered = */ false); |
| } |
| if (loss_notification_controller_) { |
| // TODO(bugs.webrtc.org/10336): Handle empty packets. |
| RTC_LOG(LS_WARNING) |
| << "LossNotificationController does not expect empty packets."; |
| } |
| } |
| |
| bool RtpVideoStreamReceiver2::DeliverRtcp(const uint8_t* rtcp_packet, |
| size_t rtcp_packet_length) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| |
| if (!receiving_) { |
| return false; |
| } |
| |
| rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length); |
| |
| int64_t rtt = 0; |
| rtp_rtcp_->RTT(config_.rtp.remote_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; |
| uint32_t received_ntp_secs = 0; |
| uint32_t received_ntp_frac = 0; |
| if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &received_ntp_secs, |
| &received_ntp_frac, &rtp_timestamp) != 0) { |
| // Waiting for RTCP. |
| return true; |
| } |
| NtpTime received_ntp(received_ntp_secs, received_ntp_frac); |
| int64_t time_since_received = |
| clock_->CurrentNtpInMilliseconds() - received_ntp.ToMs(); |
| // Don't use old SRs to estimate time. |
| if (time_since_received <= 1) { |
| ntp_estimator_.UpdateRtcpTimestamp( |
| TimeDelta::Millis(rtt), NtpTime(ntp_secs, ntp_frac), rtp_timestamp); |
| absl::optional<int64_t> remote_to_local_clock_offset = |
| ntp_estimator_.EstimateRemoteToLocalClockOffset(); |
| if (remote_to_local_clock_offset.has_value()) { |
| capture_clock_offset_updater_.SetRemoteToLocalClockOffset( |
| *remote_to_local_clock_offset); |
| } |
| } |
| |
| return true; |
| } |
| |
| void RtpVideoStreamReceiver2::FrameContinuous(int64_t picture_id) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| if (!nack_module_) |
| return; |
| |
| int seq_num = -1; |
| 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 RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| int seq_num = -1; |
| 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) { |
| int64_t unwrapped_rtp_seq_num = rtp_seq_num_unwrapper_.Unwrap(seq_num); |
| packet_infos_.erase(packet_infos_.begin(), |
| packet_infos_.upper_bound(unwrapped_rtp_seq_num)); |
| packet_buffer_.ClearTo(seq_num); |
| reference_finder_->ClearTo(seq_num); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode |
| : RtcpMode::kOff); |
| } |
| |
| void RtpVideoStreamReceiver2::StartReceive() { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| receiving_ = true; |
| } |
| |
| void RtpVideoStreamReceiver2::StopReceive() { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| receiving_ = false; |
| } |
| |
| void RtpVideoStreamReceiver2::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)); |
| } |
| if (config_.rtp.ulpfec_payload_type != -1) { |
| RTC_HISTOGRAM_COUNTS_10000( |
| "WebRTC.Video.FecBitrateReceivedInKbps", |
| static_cast<int>(counter.num_bytes * 8 / elapsed_sec / 1000)); |
| } |
| } |
| |
| void RtpVideoStreamReceiver2::InsertSpsPpsIntoTracker(uint8_t payload_type) { |
| RTC_DCHECK_RUN_ON(&packet_sequence_checker_); |
| RTC_DCHECK_RUN_ON(&worker_task_checker_); |
| |
| auto codec_params_it = pt_codec_params_.find(payload_type); |
| if (codec_params_it == pt_codec_params_.end()) |
| return; |
| |
| RTC_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()); |
| } |
| |
| void RtpVideoStreamReceiver2::UpdatePacketReceiveTimestamps( |
| const RtpPacketReceived& packet, |
| bool is_keyframe) { |
| Timestamp now = clock_->CurrentTime(); |
| if (is_keyframe || |
| last_received_keyframe_rtp_timestamp_ == packet.Timestamp()) { |
| last_received_keyframe_rtp_timestamp_ = packet.Timestamp(); |
| last_received_keyframe_rtp_system_time_ = now; |
| } |
| last_received_rtp_system_time_ = now; |
| last_received_rtp_timestamp_ = packet.Timestamp(); |
| |
| // Periodically log the RTP header of incoming packets. |
| if (now.ms() - last_packet_log_ms_ > kPacketLogIntervalMs) { |
| rtc::StringBuilder 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: " << ToString(packet.arrival_time()); |
| 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; |
| } |
| RTC_LOG(LS_INFO) << ss.str(); |
| last_packet_log_ms_ = now.ms(); |
| } |
| } |
| |
| } // namespace webrtc |
