/* * Copyright (c) 2022 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_stream_buffer_controller.h" #include #include #include #include "absl/base/attributes.h" #include "absl/functional/bind_front.h" #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "api/units/data_size.h" #include "api/video/encoded_frame.h" #include "api/video/frame_buffer.h" #include "api/video/video_content_type.h" #include "modules/video_coding/frame_helpers.h" #include "modules/video_coding/timing/inter_frame_delay.h" #include "modules/video_coding/timing/jitter_estimator.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/thread_annotations.h" #include "rtc_base/trace_event.h" #include "video/frame_decode_scheduler.h" #include "video/frame_decode_timing.h" #include "video/task_queue_frame_decode_scheduler.h" #include "video/video_receive_stream_timeout_tracker.h" namespace webrtc { namespace { // Max number of frames the buffer will hold. static constexpr size_t kMaxFramesBuffered = 800; // Max number of decoded frame info that will be saved. static constexpr int kMaxFramesHistory = 1 << 13; // Default value for the maximum decode queue size that is used when the // low-latency renderer is used. static constexpr size_t kZeroPlayoutDelayDefaultMaxDecodeQueueSize = 8; struct FrameMetadata { explicit FrameMetadata(const EncodedFrame& frame) : is_last_spatial_layer(frame.is_last_spatial_layer), is_keyframe(frame.is_keyframe()), size(frame.size()), contentType(frame.contentType()), delayed_by_retransmission(frame.delayed_by_retransmission()), rtp_timestamp(frame.Timestamp()), receive_time(frame.ReceivedTimestamp()) {} const bool is_last_spatial_layer; const bool is_keyframe; const size_t size; const VideoContentType contentType; const bool delayed_by_retransmission; const uint32_t rtp_timestamp; const absl::optional receive_time; }; Timestamp ReceiveTime(const EncodedFrame& frame) { absl::optional ts = frame.ReceivedTimestamp(); RTC_DCHECK(ts.has_value()) << "Received frame must have a timestamp set!"; return *ts; } } // namespace VideoStreamBufferController::VideoStreamBufferController( Clock* clock, TaskQueueBase* worker_queue, VCMTiming* timing, VCMReceiveStatisticsCallback* stats_proxy, FrameSchedulingReceiver* receiver, TimeDelta max_wait_for_keyframe, TimeDelta max_wait_for_frame, std::unique_ptr frame_decode_scheduler, const FieldTrialsView& field_trials) : field_trials_(field_trials), clock_(clock), stats_proxy_(stats_proxy), receiver_(receiver), timing_(timing), frame_decode_scheduler_(std::move(frame_decode_scheduler)), jitter_estimator_(clock_, field_trials), buffer_(std::make_unique(kMaxFramesBuffered, kMaxFramesHistory, field_trials)), decode_timing_(clock_, timing_), timeout_tracker_( clock_, worker_queue, VideoReceiveStreamTimeoutTracker::Timeouts{ .max_wait_for_keyframe = max_wait_for_keyframe, .max_wait_for_frame = max_wait_for_frame}, absl::bind_front(&VideoStreamBufferController::OnTimeout, this)), zero_playout_delay_max_decode_queue_size_( "max_decode_queue_size", kZeroPlayoutDelayDefaultMaxDecodeQueueSize) { RTC_DCHECK(stats_proxy_); RTC_DCHECK(receiver_); RTC_DCHECK(timing_); RTC_DCHECK(clock_); RTC_DCHECK(frame_decode_scheduler_); ParseFieldTrial({&zero_playout_delay_max_decode_queue_size_}, field_trials.Lookup("WebRTC-ZeroPlayoutDelay")); } void VideoStreamBufferController::Stop() { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); frame_decode_scheduler_->Stop(); timeout_tracker_.Stop(); decoder_ready_for_new_frame_ = false; } void VideoStreamBufferController::SetProtectionMode( VCMVideoProtection protection_mode) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); protection_mode_ = protection_mode; } void VideoStreamBufferController::Clear() { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); stats_proxy_->OnDroppedFrames(buffer_->CurrentSize()); buffer_ = std::make_unique(kMaxFramesBuffered, kMaxFramesHistory, field_trials_); frame_decode_scheduler_->CancelOutstanding(); } absl::optional VideoStreamBufferController::InsertFrame( std::unique_ptr frame) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); FrameMetadata metadata(*frame); const uint32_t ssrc = frame->PacketInfos().empty() ? 0 : frame->PacketInfos()[0].ssrc(); const int64_t frameId = frame->Id(); int complete_units = buffer_->GetTotalNumberOfContinuousTemporalUnits(); if (buffer_->InsertFrame(std::move(frame))) { RTC_DCHECK(metadata.receive_time) << "Frame receive time must be set!"; if (!metadata.delayed_by_retransmission && metadata.receive_time && (field_trials_.IsDisabled("WebRTC-IncomingTimestampOnMarkerBitOnly") || metadata.is_last_spatial_layer)) { timing_->IncomingTimestamp(metadata.rtp_timestamp, *metadata.receive_time); } if (complete_units < buffer_->GetTotalNumberOfContinuousTemporalUnits()) { TRACE_EVENT2("webrtc", "VideoStreamBufferController::InsertFrame Frame Complete", "remote_ssrc", ssrc, "frame_id", frameId); stats_proxy_->OnCompleteFrame(metadata.is_keyframe, metadata.size, metadata.contentType); MaybeScheduleFrameForRelease(); } } return buffer_->LastContinuousFrameId(); } void VideoStreamBufferController::UpdateRtt(int64_t max_rtt_ms) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); jitter_estimator_.UpdateRtt(TimeDelta::Millis(max_rtt_ms)); } void VideoStreamBufferController::SetMaxWaits(TimeDelta max_wait_for_keyframe, TimeDelta max_wait_for_frame) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); timeout_tracker_.SetTimeouts({.max_wait_for_keyframe = max_wait_for_keyframe, .max_wait_for_frame = max_wait_for_frame}); } void VideoStreamBufferController::StartNextDecode(bool keyframe_required) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); if (!timeout_tracker_.Running()) timeout_tracker_.Start(keyframe_required); keyframe_required_ = keyframe_required; if (keyframe_required_) { timeout_tracker_.SetWaitingForKeyframe(); } decoder_ready_for_new_frame_ = true; MaybeScheduleFrameForRelease(); } int VideoStreamBufferController::Size() { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); return buffer_->CurrentSize(); } void VideoStreamBufferController::OnFrameReady( absl::InlinedVector, 4> frames, Timestamp render_time) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); RTC_CHECK(!frames.empty()) << "Callers must ensure there is at least one frame to decode."; timeout_tracker_.OnEncodedFrameReleased(); Timestamp now = clock_->CurrentTime(); bool superframe_delayed_by_retransmission = false; DataSize superframe_size = DataSize::Zero(); const EncodedFrame& first_frame = *frames.front(); Timestamp receive_time = ReceiveTime(first_frame); if (first_frame.is_keyframe()) keyframe_required_ = false; // Gracefully handle bad RTP timestamps and render time issues. if (FrameHasBadRenderTiming(render_time, now) || TargetVideoDelayIsTooLarge(timing_->TargetVideoDelay())) { RTC_LOG(LS_WARNING) << "Resetting jitter estimator and timing module due " "to bad render timing for rtp_timestamp=" << first_frame.Timestamp(); jitter_estimator_.Reset(); timing_->Reset(); render_time = timing_->RenderTime(first_frame.Timestamp(), now); } for (std::unique_ptr& frame : frames) { frame->SetRenderTime(render_time.ms()); superframe_delayed_by_retransmission |= frame->delayed_by_retransmission(); receive_time = std::max(receive_time, ReceiveTime(*frame)); superframe_size += DataSize::Bytes(frame->size()); } if (!superframe_delayed_by_retransmission) { auto frame_delay = inter_frame_delay_.CalculateDelay( first_frame.Timestamp(), receive_time); if (frame_delay) { jitter_estimator_.UpdateEstimate(*frame_delay, superframe_size); } float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0; absl::optional rtt_mult_add_cap_ms = absl::nullopt; if (rtt_mult_settings_.has_value()) { rtt_mult = rtt_mult_settings_->rtt_mult_setting; rtt_mult_add_cap_ms = TimeDelta::Millis(rtt_mult_settings_->rtt_mult_add_cap_ms); } timing_->SetJitterDelay( jitter_estimator_.GetJitterEstimate(rtt_mult, rtt_mult_add_cap_ms)); timing_->UpdateCurrentDelay(render_time, now); } else if (RttMultExperiment::RttMultEnabled()) { jitter_estimator_.FrameNacked(); } // Update stats. UpdateDroppedFrames(); UpdateDiscardedPackets(); UpdateJitterDelay(); UpdateTimingFrameInfo(); std::unique_ptr frame = CombineAndDeleteFrames(std::move(frames)); timing_->SetLastDecodeScheduledTimestamp(now); decoder_ready_for_new_frame_ = false; receiver_->OnEncodedFrame(std::move(frame)); } void VideoStreamBufferController::OnTimeout(TimeDelta delay) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); // Stop sending timeouts until receiver starts waiting for a new frame. timeout_tracker_.Stop(); // If the stream is paused then ignore the timeout. if (!decoder_ready_for_new_frame_) { return; } decoder_ready_for_new_frame_ = false; receiver_->OnDecodableFrameTimeout(delay); } void VideoStreamBufferController::FrameReadyForDecode(uint32_t rtp_timestamp, Timestamp render_time) { RTC_DCHECK_RUN_ON(&worker_sequence_checker_); // Check that the frame to decode is still valid before passing the frame for // decoding. auto decodable_tu_info = buffer_->DecodableTemporalUnitsInfo(); if (!decodable_tu_info) { RTC_LOG(LS_ERROR) << "The frame buffer became undecodable during the wait " "to decode frame with rtp-timestamp " << rtp_timestamp << ". Cancelling the decode of this frame, decoding " "will resume when the frame buffers become decodable again."; return; } RTC_DCHECK_EQ(rtp_timestamp, decodable_tu_info->next_rtp_timestamp) << "Frame buffer's next decodable frame was not the one sent for " "extraction."; auto frames = buffer_->ExtractNextDecodableTemporalUnit(); if (frames.empty()) { RTC_LOG(LS_ERROR) << "The frame buffer should never return an empty temporal until list " "when there is a decodable temporal unit."; RTC_DCHECK_NOTREACHED(); return; } OnFrameReady(std::move(frames), render_time); } void VideoStreamBufferController::UpdateDroppedFrames() RTC_RUN_ON(&worker_sequence_checker_) { const int dropped_frames = buffer_->GetTotalNumberOfDroppedFrames() - frames_dropped_before_last_new_frame_; if (dropped_frames > 0) stats_proxy_->OnDroppedFrames(dropped_frames); frames_dropped_before_last_new_frame_ = buffer_->GetTotalNumberOfDroppedFrames(); } void VideoStreamBufferController::UpdateDiscardedPackets() RTC_RUN_ON(&worker_sequence_checker_) { const int discarded_packets = buffer_->GetTotalNumberOfDiscardedPackets() - packets_discarded_before_last_new_frame_; if (discarded_packets > 0) { stats_proxy_->OnDiscardedPackets(discarded_packets); } packets_discarded_before_last_new_frame_ = buffer_->GetTotalNumberOfDiscardedPackets(); } void VideoStreamBufferController::UpdateJitterDelay() { auto timings = timing_->GetTimings(); if (timings.num_decoded_frames) { stats_proxy_->OnFrameBufferTimingsUpdated( timings.max_decode_duration.ms(), timings.current_delay.ms(), timings.target_delay.ms(), timings.jitter_buffer_delay.ms(), timings.min_playout_delay.ms(), timings.render_delay.ms()); } } void VideoStreamBufferController::UpdateTimingFrameInfo() { absl::optional info = timing_->GetTimingFrameInfo(); if (info) stats_proxy_->OnTimingFrameInfoUpdated(*info); } bool VideoStreamBufferController::IsTooManyFramesQueued() const RTC_RUN_ON(&worker_sequence_checker_) { return buffer_->CurrentSize() > zero_playout_delay_max_decode_queue_size_; } void VideoStreamBufferController::ForceKeyFrameReleaseImmediately() RTC_RUN_ON(&worker_sequence_checker_) { RTC_DCHECK(keyframe_required_); // Iterate through the frame buffer until there is a complete keyframe and // release this right away. while (buffer_->DecodableTemporalUnitsInfo()) { auto next_frame = buffer_->ExtractNextDecodableTemporalUnit(); if (next_frame.empty()) { RTC_DCHECK_NOTREACHED() << "Frame buffer should always return at least 1 frame."; continue; } // Found keyframe - decode right away. if (next_frame.front()->is_keyframe()) { auto render_time = timing_->RenderTime(next_frame.front()->Timestamp(), clock_->CurrentTime()); OnFrameReady(std::move(next_frame), render_time); return; } } } void VideoStreamBufferController::MaybeScheduleFrameForRelease() RTC_RUN_ON(&worker_sequence_checker_) { auto decodable_tu_info = buffer_->DecodableTemporalUnitsInfo(); if (!decoder_ready_for_new_frame_ || !decodable_tu_info) { return; } if (keyframe_required_) { return ForceKeyFrameReleaseImmediately(); } // If already scheduled then abort. if (frame_decode_scheduler_->ScheduledRtpTimestamp() == decodable_tu_info->next_rtp_timestamp) { return; } TimeDelta max_wait = timeout_tracker_.TimeUntilTimeout(); // Ensures the frame is scheduled for decode before the stream times out. // This is otherwise a race condition. max_wait = std::max(max_wait - TimeDelta::Millis(1), TimeDelta::Zero()); absl::optional schedule; while (decodable_tu_info) { schedule = decode_timing_.OnFrameBufferUpdated( decodable_tu_info->next_rtp_timestamp, decodable_tu_info->last_rtp_timestamp, max_wait, IsTooManyFramesQueued()); if (schedule) { // Don't schedule if already waiting for the same frame. if (frame_decode_scheduler_->ScheduledRtpTimestamp() != decodable_tu_info->next_rtp_timestamp) { frame_decode_scheduler_->CancelOutstanding(); frame_decode_scheduler_->ScheduleFrame( decodable_tu_info->next_rtp_timestamp, *schedule, absl::bind_front(&VideoStreamBufferController::FrameReadyForDecode, this)); } return; } // If no schedule for current rtp, drop and try again. buffer_->DropNextDecodableTemporalUnit(); decodable_tu_info = buffer_->DecodableTemporalUnitsInfo(); } } } // namespace webrtc