| /* |
| * Copyright (c) 2013 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/video_receive_stream.h" |
| |
| #include <stdlib.h> |
| #include <string.h> |
| #include <algorithm> |
| #include <set> |
| #include <string> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/memory/memory.h" |
| #include "absl/types/optional.h" |
| #include "api/array_view.h" |
| #include "api/crypto/frame_decryptor_interface.h" |
| #include "api/video/encoded_image.h" |
| #include "api/video_codecs/sdp_video_format.h" |
| #include "api/video_codecs/video_codec.h" |
| #include "api/video_codecs/video_decoder_factory.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "call/rtp_stream_receiver_controller_interface.h" |
| #include "call/rtx_receive_stream.h" |
| #include "common_video/include/incoming_video_stream.h" |
| #include "media/base/h264_profile_level_id.h" |
| #include "modules/utility/include/process_thread.h" |
| #include "modules/video_coding/include/video_codec_interface.h" |
| #include "modules/video_coding/include/video_coding_defines.h" |
| #include "modules/video_coding/include/video_error_codes.h" |
| #include "modules/video_coding/timing.h" |
| #include "modules/video_coding/utility/vp8_header_parser.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/experiments/keyframe_interval_settings.h" |
| #include "rtc_base/location.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/platform_file.h" |
| #include "rtc_base/strings/string_builder.h" |
| #include "rtc_base/system/thread_registry.h" |
| #include "rtc_base/time_utils.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "video/call_stats.h" |
| #include "video/frame_dumping_decoder.h" |
| #include "video/receive_statistics_proxy.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| using video_coding::EncodedFrame; |
| using ReturnReason = video_coding::FrameBuffer::ReturnReason; |
| |
| constexpr int kMinBaseMinimumDelayMs = 0; |
| constexpr int kMaxBaseMinimumDelayMs = 10000; |
| |
| constexpr int kMaxWaitForKeyFrameMs = 200; |
| constexpr int kMaxWaitForFrameMs = 3000; |
| |
| VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) { |
| VideoCodec codec; |
| memset(&codec, 0, sizeof(codec)); |
| |
| codec.plType = decoder.payload_type; |
| codec.codecType = PayloadStringToCodecType(decoder.video_format.name); |
| |
| if (codec.codecType == kVideoCodecVP8) { |
| *(codec.VP8()) = VideoEncoder::GetDefaultVp8Settings(); |
| } else if (codec.codecType == kVideoCodecVP9) { |
| *(codec.VP9()) = VideoEncoder::GetDefaultVp9Settings(); |
| } else if (codec.codecType == kVideoCodecH264) { |
| *(codec.H264()) = VideoEncoder::GetDefaultH264Settings(); |
| } else if (codec.codecType == kVideoCodecMultiplex) { |
| VideoReceiveStream::Decoder associated_decoder = decoder; |
| associated_decoder.video_format = |
| SdpVideoFormat(CodecTypeToPayloadString(kVideoCodecVP9)); |
| VideoCodec associated_codec = CreateDecoderVideoCodec(associated_decoder); |
| associated_codec.codecType = kVideoCodecMultiplex; |
| return associated_codec; |
| } |
| |
| codec.width = 320; |
| codec.height = 180; |
| const int kDefaultStartBitrate = 300; |
| codec.startBitrate = codec.minBitrate = codec.maxBitrate = |
| kDefaultStartBitrate; |
| |
| return codec; |
| } |
| |
| // Video decoder class to be used for unknown codecs. Doesn't support decoding |
| // but logs messages to LS_ERROR. |
| class NullVideoDecoder : public webrtc::VideoDecoder { |
| public: |
| int32_t InitDecode(const webrtc::VideoCodec* codec_settings, |
| int32_t number_of_cores) override { |
| RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder."; |
| return WEBRTC_VIDEO_CODEC_OK; |
| } |
| |
| int32_t Decode(const webrtc::EncodedImage& input_image, |
| bool missing_frames, |
| int64_t render_time_ms) override { |
| RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding."; |
| return WEBRTC_VIDEO_CODEC_OK; |
| } |
| |
| int32_t RegisterDecodeCompleteCallback( |
| webrtc::DecodedImageCallback* callback) override { |
| RTC_LOG(LS_ERROR) |
| << "Can't register decode complete callback on NullVideoDecoder."; |
| return WEBRTC_VIDEO_CODEC_OK; |
| } |
| |
| int32_t Release() override { return WEBRTC_VIDEO_CODEC_OK; } |
| |
| const char* ImplementationName() const override { return "NullVideoDecoder"; } |
| }; |
| |
| // Inherit video_coding::EncodedFrame, which is the class used by |
| // video_coding::FrameBuffer and other components in the receive pipeline. It's |
| // a subclass of EncodedImage, and it always owns the buffer. |
| class EncodedFrameForMediaTransport : public video_coding::EncodedFrame { |
| public: |
| explicit EncodedFrameForMediaTransport( |
| MediaTransportEncodedVideoFrame frame) { |
| // TODO(nisse): This is ugly. We copy the EncodedImage (a base class of |
| // ours, in several steps), to get all the meta data. We should be using |
| // std::move in some way. Then we also need to handle the case of an unowned |
| // buffer, in which case we need to make an owned copy. |
| *static_cast<class EncodedImage*>(this) = frame.encoded_image(); |
| |
| // If we don't already own the buffer, make a copy. |
| Retain(); |
| |
| _payloadType = static_cast<uint8_t>(frame.payload_type()); |
| |
| // TODO(nisse): frame_id and picture_id are probably not the same thing. For |
| // a single layer, this should be good enough. |
| id.picture_id = frame.frame_id(); |
| id.spatial_layer = frame.encoded_image().SpatialIndex().value_or(0); |
| num_references = std::min(static_cast<size_t>(kMaxFrameReferences), |
| frame.referenced_frame_ids().size()); |
| for (size_t i = 0; i < num_references; i++) { |
| references[i] = frame.referenced_frame_ids()[i]; |
| } |
| } |
| |
| // TODO(nisse): Implement. Not sure how they are used. |
| int64_t ReceivedTime() const override { return 0; } |
| int64_t RenderTime() const override { return 0; } |
| }; |
| |
| // TODO(https://bugs.webrtc.org/9974): Consider removing this workaround. |
| // Maximum time between frames before resetting the FrameBuffer to avoid RTP |
| // timestamps wraparound to affect FrameBuffer. |
| constexpr int kInactiveStreamThresholdMs = 600000; // 10 minutes. |
| |
| } // namespace |
| |
| namespace internal { |
| |
| VideoReceiveStream::VideoReceiveStream( |
| TaskQueueFactory* task_queue_factory, |
| RtpStreamReceiverControllerInterface* receiver_controller, |
| int num_cpu_cores, |
| PacketRouter* packet_router, |
| VideoReceiveStream::Config config, |
| ProcessThread* process_thread, |
| CallStats* call_stats, |
| Clock* clock, |
| VCMTiming* timing) |
| : task_queue_factory_(task_queue_factory), |
| transport_adapter_(config.rtcp_send_transport), |
| config_(std::move(config)), |
| num_cpu_cores_(num_cpu_cores), |
| process_thread_(process_thread), |
| clock_(clock), |
| use_task_queue_( |
| !field_trial::IsDisabled("WebRTC-Video-DecodeOnTaskQueue")), |
| decode_thread_(&DecodeThreadFunction, |
| this, |
| "DecodingThread", |
| rtc::kHighestPriority), |
| call_stats_(call_stats), |
| stats_proxy_(&config_, clock_), |
| rtp_receive_statistics_( |
| ReceiveStatistics::Create(clock_, &stats_proxy_, &stats_proxy_)), |
| timing_(timing), |
| video_receiver_(clock_, timing_.get()), |
| rtp_video_stream_receiver_(clock_, |
| &transport_adapter_, |
| call_stats, |
| packet_router, |
| &config_, |
| rtp_receive_statistics_.get(), |
| &stats_proxy_, |
| process_thread_, |
| this, // NackSender |
| this, // OnCompleteFrameCallback |
| config_.frame_decryptor), |
| rtp_stream_sync_(this), |
| max_wait_for_keyframe_ms_(KeyframeIntervalSettings::ParseFromFieldTrials() |
| .MaxWaitForKeyframeMs() |
| .value_or(kMaxWaitForKeyFrameMs)), |
| max_wait_for_frame_ms_(KeyframeIntervalSettings::ParseFromFieldTrials() |
| .MaxWaitForFrameMs() |
| .value_or(kMaxWaitForFrameMs)), |
| decode_queue_(task_queue_factory_->CreateTaskQueue( |
| "DecodingQueue", |
| TaskQueueFactory::Priority::HIGH)) { |
| RTC_LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString(); |
| |
| RTC_DCHECK(config_.renderer); |
| RTC_DCHECK(process_thread_); |
| RTC_DCHECK(call_stats_); |
| |
| module_process_sequence_checker_.Detach(); |
| network_sequence_checker_.Detach(); |
| |
| RTC_DCHECK(!config_.decoders.empty()); |
| std::set<int> decoder_payload_types; |
| for (const Decoder& decoder : config_.decoders) { |
| RTC_CHECK(decoder.decoder_factory); |
| RTC_CHECK(decoder_payload_types.find(decoder.payload_type) == |
| decoder_payload_types.end()) |
| << "Duplicate payload type (" << decoder.payload_type |
| << ") for different decoders."; |
| decoder_payload_types.insert(decoder.payload_type); |
| } |
| |
| timing_->set_render_delay(config_.render_delay_ms); |
| |
| frame_buffer_.reset( |
| new video_coding::FrameBuffer(clock_, timing_.get(), &stats_proxy_)); |
| |
| process_thread_->RegisterModule(&rtp_stream_sync_, RTC_FROM_HERE); |
| if (config_.media_transport()) { |
| config_.media_transport()->SetReceiveVideoSink(this); |
| config_.media_transport()->AddRttObserver(this); |
| } else { |
| // Register with RtpStreamReceiverController. |
| media_receiver_ = receiver_controller->CreateReceiver( |
| config_.rtp.remote_ssrc, &rtp_video_stream_receiver_); |
| if (config_.rtp.rtx_ssrc) { |
| rtx_receive_stream_ = absl::make_unique<RtxReceiveStream>( |
| &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types, |
| config_.rtp.remote_ssrc, rtp_receive_statistics_.get()); |
| rtx_receiver_ = receiver_controller->CreateReceiver( |
| config_.rtp.rtx_ssrc, rtx_receive_stream_.get()); |
| } else { |
| rtp_receive_statistics_->EnableRetransmitDetection(config.rtp.remote_ssrc, |
| true); |
| } |
| } |
| } |
| |
| VideoReceiveStream::VideoReceiveStream( |
| TaskQueueFactory* task_queue_factory, |
| RtpStreamReceiverControllerInterface* receiver_controller, |
| int num_cpu_cores, |
| PacketRouter* packet_router, |
| VideoReceiveStream::Config config, |
| ProcessThread* process_thread, |
| CallStats* call_stats, |
| Clock* clock) |
| : VideoReceiveStream(task_queue_factory, |
| receiver_controller, |
| num_cpu_cores, |
| packet_router, |
| std::move(config), |
| process_thread, |
| call_stats, |
| clock, |
| new VCMTiming(clock)) {} |
| |
| VideoReceiveStream::~VideoReceiveStream() { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| RTC_LOG(LS_INFO) << "~VideoReceiveStream: " << config_.ToString(); |
| Stop(); |
| if (config_.media_transport()) { |
| config_.media_transport()->SetReceiveVideoSink(nullptr); |
| config_.media_transport()->RemoveRttObserver(this); |
| } |
| process_thread_->DeRegisterModule(&rtp_stream_sync_); |
| } |
| |
| void VideoReceiveStream::SignalNetworkState(NetworkState state) { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| rtp_video_stream_receiver_.SignalNetworkState(state); |
| } |
| |
| bool VideoReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) { |
| return rtp_video_stream_receiver_.DeliverRtcp(packet, length); |
| } |
| |
| void VideoReceiveStream::SetSync(Syncable* audio_syncable) { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| rtp_stream_sync_.ConfigureSync(audio_syncable); |
| } |
| |
| void VideoReceiveStream::Start() { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| |
| if (decoder_running_) { |
| return; |
| } |
| |
| const bool protected_by_fec = config_.rtp.protected_by_flexfec || |
| rtp_video_stream_receiver_.IsUlpfecEnabled(); |
| |
| frame_buffer_->Start(); |
| |
| if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() && |
| protected_by_fec) { |
| frame_buffer_->SetProtectionMode(kProtectionNackFEC); |
| } |
| |
| transport_adapter_.Enable(); |
| rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr; |
| if (config_.enable_prerenderer_smoothing) { |
| incoming_video_stream_.reset(new IncomingVideoStream( |
| task_queue_factory_, config_.render_delay_ms, this)); |
| renderer = incoming_video_stream_.get(); |
| } else { |
| renderer = this; |
| } |
| |
| for (const Decoder& decoder : config_.decoders) { |
| std::unique_ptr<VideoDecoder> video_decoder = |
| decoder.decoder_factory->LegacyCreateVideoDecoder(decoder.video_format, |
| config_.stream_id); |
| // If we still have no valid decoder, we have to create a "Null" decoder |
| // that ignores all calls. The reason we can get into this state is that the |
| // old decoder factory interface doesn't have a way to query supported |
| // codecs. |
| if (!video_decoder) { |
| video_decoder = absl::make_unique<NullVideoDecoder>(); |
| } |
| |
| std::string decoded_output_file = |
| field_trial::FindFullName("WebRTC-DecoderDataDumpDirectory"); |
| // Because '/' can't be used inside a field trial parameter, we use ':' |
| // instead. |
| absl::c_replace(decoded_output_file, ':', '/'); |
| if (!decoded_output_file.empty()) { |
| char filename_buffer[256]; |
| rtc::SimpleStringBuilder ssb(filename_buffer); |
| ssb << decoded_output_file << "/webrtc_receive_stream_" |
| << this->config_.rtp.remote_ssrc << "-" << rtc::TimeMicros() |
| << ".ivf"; |
| video_decoder = absl::make_unique<FrameDumpingDecoder>( |
| std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str())); |
| } |
| |
| video_decoders_.push_back(std::move(video_decoder)); |
| |
| video_receiver_.RegisterExternalDecoder(video_decoders_.back().get(), |
| decoder.payload_type); |
| VideoCodec codec = CreateDecoderVideoCodec(decoder); |
| |
| const bool raw_payload = |
| config_.rtp.raw_payload_types.count(codec.plType) > 0; |
| rtp_video_stream_receiver_.AddReceiveCodec( |
| codec, decoder.video_format.parameters, raw_payload); |
| RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec( |
| &codec, num_cpu_cores_, false)); |
| } |
| |
| RTC_DCHECK(renderer != nullptr); |
| video_stream_decoder_.reset( |
| new VideoStreamDecoder(&video_receiver_, &stats_proxy_, renderer)); |
| |
| // Make sure we register as a stats observer *after* we've prepared the |
| // |video_stream_decoder_|. |
| call_stats_->RegisterStatsObserver(this); |
| |
| // NOTE: *Not* registering video_receiver_ on process_thread_. Its Process |
| // method does nothing that is useful for us, since we no longer use the old |
| // jitter buffer. |
| |
| // Start the decode thread |
| video_receiver_.DecoderThreadStarting(); |
| stats_proxy_.DecoderThreadStarting(); |
| if (!use_task_queue_) { |
| decode_thread_.Start(); |
| } else { |
| decode_queue_.PostTask([this] { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| decoder_stopped_ = false; |
| StartNextDecode(); |
| }); |
| } |
| decoder_running_ = true; |
| rtp_video_stream_receiver_.StartReceive(); |
| } |
| |
| void VideoReceiveStream::Stop() { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| rtp_video_stream_receiver_.StopReceive(); |
| |
| stats_proxy_.OnUniqueFramesCounted( |
| rtp_video_stream_receiver_.GetUniqueFramesSeen()); |
| |
| if (!use_task_queue_) { |
| frame_buffer_->Stop(); |
| } else { |
| decode_queue_.PostTask([this] { frame_buffer_->Stop(); }); |
| } |
| call_stats_->DeregisterStatsObserver(this); |
| |
| if (decoder_running_) { |
| // TriggerDecoderShutdown will release any waiting decoder thread and make |
| // it stop immediately, instead of waiting for a timeout. Needs to be called |
| // before joining the decoder thread. |
| video_receiver_.TriggerDecoderShutdown(); |
| |
| if (!use_task_queue_) { |
| decode_thread_.Stop(); |
| } else { |
| rtc::Event done; |
| decode_queue_.PostTask([this, &done] { |
| RTC_DCHECK_RUN_ON(&decode_queue_); |
| decoder_stopped_ = true; |
| done.Set(); |
| }); |
| done.Wait(rtc::Event::kForever); |
| } |
| decoder_running_ = false; |
| video_receiver_.DecoderThreadStopped(); |
| stats_proxy_.DecoderThreadStopped(); |
| // Deregister external decoders so they are no longer running during |
| // destruction. This effectively stops the VCM since the decoder thread is |
| // stopped, the VCM is deregistered and no asynchronous decoder threads are |
| // running. |
| for (const Decoder& decoder : config_.decoders) |
| video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type); |
| } |
| |
| video_stream_decoder_.reset(); |
| incoming_video_stream_.reset(); |
| transport_adapter_.Disable(); |
| } |
| |
| VideoReceiveStream::Stats VideoReceiveStream::GetStats() const { |
| return stats_proxy_.GetStats(); |
| } |
| |
| void VideoReceiveStream::AddSecondarySink(RtpPacketSinkInterface* sink) { |
| rtp_video_stream_receiver_.AddSecondarySink(sink); |
| } |
| |
| void VideoReceiveStream::RemoveSecondarySink( |
| const RtpPacketSinkInterface* sink) { |
| rtp_video_stream_receiver_.RemoveSecondarySink(sink); |
| } |
| |
| bool VideoReceiveStream::SetBaseMinimumPlayoutDelayMs(int delay_ms) { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| if (delay_ms < kMinBaseMinimumDelayMs || delay_ms > kMaxBaseMinimumDelayMs) { |
| return false; |
| } |
| |
| rtc::CritScope cs(&playout_delay_lock_); |
| base_minimum_playout_delay_ms_ = delay_ms; |
| UpdatePlayoutDelays(); |
| return true; |
| } |
| |
| int VideoReceiveStream::GetBaseMinimumPlayoutDelayMs() const { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| |
| rtc::CritScope cs(&playout_delay_lock_); |
| return base_minimum_playout_delay_ms_; |
| } |
| |
| // TODO(tommi): This method grabs a lock 6 times. |
| void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) { |
| int64_t sync_offset_ms; |
| double estimated_freq_khz; |
| // TODO(tommi): GetStreamSyncOffsetInMs grabs three locks. One inside the |
| // function itself, another in GetChannel() and a third in |
| // GetPlayoutTimestamp. Seems excessive. Anyhow, I'm assuming the function |
| // succeeds most of the time, which leads to grabbing a fourth lock. |
| if (rtp_stream_sync_.GetStreamSyncOffsetInMs( |
| video_frame.timestamp(), video_frame.render_time_ms(), |
| &sync_offset_ms, &estimated_freq_khz)) { |
| // TODO(tommi): OnSyncOffsetUpdated grabs a lock. |
| stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms, estimated_freq_khz); |
| } |
| config_.renderer->OnFrame(video_frame); |
| |
| // TODO(tommi): OnRenderFrame grabs a lock too. |
| stats_proxy_.OnRenderedFrame(video_frame); |
| } |
| |
| void VideoReceiveStream::SetFrameDecryptor( |
| rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) { |
| rtp_video_stream_receiver_.SetFrameDecryptor(std::move(frame_decryptor)); |
| } |
| |
| void VideoReceiveStream::SendNack( |
| const std::vector<uint16_t>& sequence_numbers) { |
| rtp_video_stream_receiver_.RequestPacketRetransmit(sequence_numbers); |
| } |
| |
| void VideoReceiveStream::RequestKeyFrame() { |
| if (config_.media_transport()) { |
| config_.media_transport()->RequestKeyFrame(config_.rtp.remote_ssrc); |
| } else { |
| rtp_video_stream_receiver_.RequestKeyFrame(); |
| } |
| } |
| |
| void VideoReceiveStream::OnCompleteFrame( |
| std::unique_ptr<video_coding::EncodedFrame> frame) { |
| RTC_DCHECK_RUN_ON(&network_sequence_checker_); |
| // TODO(https://bugs.webrtc.org/9974): Consider removing this workaround. |
| int64_t time_now_ms = rtc::TimeMillis(); |
| if (last_complete_frame_time_ms_ > 0 && |
| time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) { |
| frame_buffer_->Clear(); |
| } |
| last_complete_frame_time_ms_ = time_now_ms; |
| |
| const PlayoutDelay& playout_delay = frame->EncodedImage().playout_delay_; |
| if (playout_delay.min_ms >= 0) { |
| rtc::CritScope cs(&playout_delay_lock_); |
| frame_minimum_playout_delay_ms_ = playout_delay.min_ms; |
| UpdatePlayoutDelays(); |
| } |
| |
| if (playout_delay.max_ms >= 0) { |
| rtc::CritScope cs(&playout_delay_lock_); |
| frame_maximum_playout_delay_ms_ = playout_delay.max_ms; |
| UpdatePlayoutDelays(); |
| } |
| |
| int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame)); |
| if (last_continuous_pid != -1) |
| rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid); |
| } |
| |
| void VideoReceiveStream::OnData(uint64_t channel_id, |
| MediaTransportEncodedVideoFrame frame) { |
| OnCompleteFrame( |
| absl::make_unique<EncodedFrameForMediaTransport>(std::move(frame))); |
| } |
| |
| void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
| RTC_DCHECK_RUN_ON(&module_process_sequence_checker_); |
| frame_buffer_->UpdateRtt(max_rtt_ms); |
| rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms); |
| } |
| |
| void VideoReceiveStream::OnRttUpdated(int64_t rtt_ms) { |
| frame_buffer_->UpdateRtt(rtt_ms); |
| } |
| |
| int VideoReceiveStream::id() const { |
| RTC_DCHECK_RUN_ON(&worker_sequence_checker_); |
| return config_.rtp.remote_ssrc; |
| } |
| |
| absl::optional<Syncable::Info> VideoReceiveStream::GetInfo() const { |
| RTC_DCHECK_RUN_ON(&module_process_sequence_checker_); |
| absl::optional<Syncable::Info> info = |
| rtp_video_stream_receiver_.GetSyncInfo(); |
| |
| if (!info) |
| return absl::nullopt; |
| |
| info->current_delay_ms = timing_->TargetVideoDelay(); |
| return info; |
| } |
| |
| uint32_t VideoReceiveStream::GetPlayoutTimestamp() const { |
| RTC_NOTREACHED(); |
| return 0; |
| } |
| |
| void VideoReceiveStream::SetMinimumPlayoutDelay(int delay_ms) { |
| RTC_DCHECK_RUN_ON(&module_process_sequence_checker_); |
| rtc::CritScope cs(&playout_delay_lock_); |
| syncable_minimum_playout_delay_ms_ = delay_ms; |
| UpdatePlayoutDelays(); |
| } |
| |
| int64_t VideoReceiveStream::GetWaitMs() const { |
| return keyframe_required_ ? max_wait_for_keyframe_ms_ |
| : max_wait_for_frame_ms_; |
| } |
| |
| void VideoReceiveStream::StartNextDecode() { |
| RTC_DCHECK(use_task_queue_); |
| TRACE_EVENT0("webrtc", "VideoReceiveStream::StartNextDecode"); |
| |
| struct DecodeTask { |
| void operator()() { |
| RTC_DCHECK_RUN_ON(&stream->decode_queue_); |
| if (stream->decoder_stopped_) |
| return; |
| if (frame) { |
| stream->HandleEncodedFrame(std::move(frame)); |
| } else { |
| stream->HandleFrameBufferTimeout(); |
| } |
| stream->StartNextDecode(); |
| } |
| VideoReceiveStream* stream; |
| std::unique_ptr<EncodedFrame> frame; |
| }; |
| |
| frame_buffer_->NextFrame( |
| GetWaitMs(), keyframe_required_, &decode_queue_, |
| [this](std::unique_ptr<EncodedFrame> frame, ReturnReason res) { |
| RTC_DCHECK_EQ(frame == nullptr, res == ReturnReason::kTimeout); |
| RTC_DCHECK_EQ(frame != nullptr, res == ReturnReason::kFrameFound); |
| decode_queue_.PostTask(DecodeTask{this, std::move(frame)}); |
| }); |
| } |
| |
| void VideoReceiveStream::DecodeThreadFunction(void* ptr) { |
| ScopedRegisterThreadForDebugging thread_dbg(RTC_FROM_HERE); |
| while (static_cast<VideoReceiveStream*>(ptr)->Decode()) { |
| } |
| } |
| |
| bool VideoReceiveStream::Decode() { |
| RTC_DCHECK(!use_task_queue_); |
| TRACE_EVENT0("webrtc", "VideoReceiveStream::Decode"); |
| |
| std::unique_ptr<video_coding::EncodedFrame> frame; |
| video_coding::FrameBuffer::ReturnReason res = |
| frame_buffer_->NextFrame(GetWaitMs(), &frame, keyframe_required_); |
| |
| if (res == ReturnReason::kStopped) { |
| return false; |
| } |
| |
| if (frame) { |
| RTC_DCHECK_EQ(res, ReturnReason::kFrameFound); |
| HandleEncodedFrame(std::move(frame)); |
| } else { |
| RTC_DCHECK_EQ(res, ReturnReason::kTimeout); |
| HandleFrameBufferTimeout(); |
| } |
| return true; |
| } |
| |
| void VideoReceiveStream::HandleEncodedFrame( |
| std::unique_ptr<EncodedFrame> frame) { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| |
| // Current OnPreDecode only cares about QP for VP8. |
| int qp = -1; |
| if (frame->CodecSpecific()->codecType == kVideoCodecVP8) { |
| if (!vp8::GetQp(frame->data(), frame->size(), &qp)) { |
| RTC_LOG(LS_WARNING) << "Failed to extract QP from VP8 video frame"; |
| } |
| } |
| stats_proxy_.OnPreDecode(frame->CodecSpecific()->codecType, qp); |
| |
| int decode_result = video_receiver_.Decode(frame.get()); |
| if (decode_result == WEBRTC_VIDEO_CODEC_OK || |
| decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) { |
| keyframe_required_ = false; |
| frame_decoded_ = true; |
| rtp_video_stream_receiver_.FrameDecoded(frame->id.picture_id); |
| |
| if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) |
| RequestKeyFrame(); |
| } else if (!frame_decoded_ || !keyframe_required_ || |
| (last_keyframe_request_ms_ + max_wait_for_keyframe_ms_ < now_ms)) { |
| keyframe_required_ = true; |
| // TODO(philipel): Remove this keyframe request when downstream project |
| // has been fixed. |
| RequestKeyFrame(); |
| last_keyframe_request_ms_ = now_ms; |
| } |
| } |
| |
| void VideoReceiveStream::HandleFrameBufferTimeout() { |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| absl::optional<int64_t> last_packet_ms = |
| rtp_video_stream_receiver_.LastReceivedPacketMs(); |
| absl::optional<int64_t> last_keyframe_packet_ms = |
| rtp_video_stream_receiver_.LastReceivedKeyframePacketMs(); |
| |
| // To avoid spamming keyframe requests for a stream that is not active we |
| // check if we have received a packet within the last 5 seconds. |
| bool stream_is_active = last_packet_ms && now_ms - *last_packet_ms < 5000; |
| if (!stream_is_active) |
| stats_proxy_.OnStreamInactive(); |
| |
| // If we recently have been receiving packets belonging to a keyframe then |
| // we assume a keyframe is currently being received. |
| bool receiving_keyframe = |
| last_keyframe_packet_ms && |
| now_ms - *last_keyframe_packet_ms < max_wait_for_keyframe_ms_; |
| |
| if (stream_is_active && !receiving_keyframe && |
| (!config_.crypto_options.sframe.require_frame_encryption || |
| rtp_video_stream_receiver_.IsDecryptable())) { |
| RTC_LOG(LS_WARNING) << "No decodable frame in " << GetWaitMs() |
| << " ms, requesting keyframe."; |
| RequestKeyFrame(); |
| } |
| } |
| |
| void VideoReceiveStream::UpdatePlayoutDelays() const { |
| const int minimum_delay_ms = |
| std::max({frame_minimum_playout_delay_ms_, base_minimum_playout_delay_ms_, |
| syncable_minimum_playout_delay_ms_}); |
| if (minimum_delay_ms >= 0) { |
| timing_->set_min_playout_delay(minimum_delay_ms); |
| } |
| |
| const int maximum_delay_ms = frame_maximum_playout_delay_ms_; |
| if (maximum_delay_ms >= 0) { |
| timing_->set_max_playout_delay(maximum_delay_ms); |
| } |
| } |
| |
| std::vector<webrtc::RtpSource> VideoReceiveStream::GetSources() const { |
| return rtp_video_stream_receiver_.GetSources(); |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |
