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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/video/video_stream_buffer_controller.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/video/video_stream_buffer_controller.cc')
-rw-r--r-- | third_party/libwebrtc/video/video_stream_buffer_controller.cc | 422 |
1 files changed, 422 insertions, 0 deletions
diff --git a/third_party/libwebrtc/video/video_stream_buffer_controller.cc b/third_party/libwebrtc/video/video_stream_buffer_controller.cc new file mode 100644 index 0000000000..37724a8338 --- /dev/null +++ b/third_party/libwebrtc/video/video_stream_buffer_controller.cc @@ -0,0 +1,422 @@ +/* + * 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 <algorithm> +#include <memory> +#include <utility> + +#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<Timestamp> receive_time; +}; + +Timestamp ReceiveTime(const EncodedFrame& frame) { + absl::optional<Timestamp> 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<FrameDecodeScheduler> 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<FrameBuffer>(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<FrameBuffer>(kMaxFramesBuffered, kMaxFramesHistory, + field_trials_); + frame_decode_scheduler_->CancelOutstanding(); +} + +absl::optional<int64_t> VideoStreamBufferController::InsertFrame( + std::unique_ptr<EncodedFrame> 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<std::unique_ptr<EncodedFrame>, 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<EncodedFrame>& 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<TimeDelta> 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<EncodedFrame> 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<TimingFrameInfo> 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<FrameDecodeTiming::FrameSchedule> 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 |