/* * Copyright (c) 2016 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 "modules/video_coding/frame_buffer2.h" #include #include #include #include #include #include #include #include "absl/container/inlined_vector.h" #include "api/units/data_size.h" #include "api/units/time_delta.h" #include "api/video/encoded_image.h" #include "api/video/video_timing.h" #include "modules/video_coding/frame_helpers.h" #include "modules/video_coding/include/video_coding_defines.h" #include "modules/video_coding/timing/jitter_estimator.h" #include "modules/video_coding/timing/timing.h" #include "rtc_base/checks.h" #include "rtc_base/experiments/rtt_mult_experiment.h" #include "rtc_base/logging.h" #include "rtc_base/numerics/sequence_number_util.h" #include "rtc_base/trace_event.h" #include "system_wrappers/include/clock.h" namespace webrtc { namespace video_coding { namespace { // Max number of frames the buffer will hold. constexpr size_t kMaxFramesBuffered = 800; // Default value for the maximum decode queue size that is used when the // low-latency renderer is used. constexpr size_t kZeroPlayoutDelayDefaultMaxDecodeQueueSize = 8; // Max number of decoded frame info that will be saved. constexpr int kMaxFramesHistory = 1 << 13; // The time it's allowed for a frame to be late to its rendering prediction and // still be rendered. constexpr int kMaxAllowedFrameDelayMs = 5; constexpr int64_t kLogNonDecodedIntervalMs = 5000; } // namespace FrameBuffer::FrameBuffer(Clock* clock, VCMTiming* timing, const FieldTrialsView& field_trials) : decoded_frames_history_(kMaxFramesHistory), clock_(clock), callback_queue_(nullptr), jitter_estimator_(clock, field_trials), timing_(timing), stopped_(false), protection_mode_(kProtectionNack), last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs), rtt_mult_settings_(RttMultExperiment::GetRttMultValue()), zero_playout_delay_max_decode_queue_size_( "max_decode_queue_size", kZeroPlayoutDelayDefaultMaxDecodeQueueSize) { ParseFieldTrial({&zero_playout_delay_max_decode_queue_size_}, field_trials.Lookup("WebRTC-ZeroPlayoutDelay")); callback_checker_.Detach(); } FrameBuffer::~FrameBuffer() { RTC_DCHECK_RUN_ON(&construction_checker_); } void FrameBuffer::NextFrame(int64_t max_wait_time_ms, bool keyframe_required, TaskQueueBase* callback_queue, NextFrameCallback handler) { RTC_DCHECK_RUN_ON(&callback_checker_); RTC_DCHECK(callback_queue->IsCurrent()); TRACE_EVENT0("webrtc", "FrameBuffer::NextFrame"); int64_t latest_return_time_ms = clock_->TimeInMilliseconds() + max_wait_time_ms; MutexLock lock(&mutex_); if (stopped_) { return; } latest_return_time_ms_ = latest_return_time_ms; keyframe_required_ = keyframe_required; frame_handler_ = handler; callback_queue_ = callback_queue; StartWaitForNextFrameOnQueue(); } void FrameBuffer::StartWaitForNextFrameOnQueue() { RTC_DCHECK(callback_queue_); RTC_DCHECK(!callback_task_.Running()); int64_t wait_ms = FindNextFrame(clock_->CurrentTime()); callback_task_ = RepeatingTaskHandle::DelayedStart( callback_queue_, TimeDelta::Millis(wait_ms), [this] { RTC_DCHECK_RUN_ON(&callback_checker_); // If this task has not been cancelled, we did not get any new frames // while waiting. Continue with frame delivery. std::unique_ptr frame; NextFrameCallback frame_handler; { MutexLock lock(&mutex_); if (!frames_to_decode_.empty()) { // We have frames, deliver! frame = GetNextFrame(); timing_->SetLastDecodeScheduledTimestamp(clock_->CurrentTime()); } else if (clock_->TimeInMilliseconds() < latest_return_time_ms_) { // If there's no frames to decode and there is still time left, it // means that the frame buffer was cleared between creation and // execution of this task. Continue waiting for the remaining time. int64_t wait_ms = FindNextFrame(clock_->CurrentTime()); return TimeDelta::Millis(wait_ms); } frame_handler = std::move(frame_handler_); CancelCallback(); } // Deliver frame, if any. Otherwise signal timeout. frame_handler(std::move(frame)); return TimeDelta::Zero(); // Ignored. }, TaskQueueBase::DelayPrecision::kHigh); } int64_t FrameBuffer::FindNextFrame(Timestamp now) { int64_t wait_ms = latest_return_time_ms_ - now.ms(); frames_to_decode_.clear(); // `last_continuous_frame_` may be empty below, but nullopt is smaller // than everything else and loop will immediately terminate as expected. for (auto frame_it = frames_.begin(); frame_it != frames_.end() && frame_it->first <= last_continuous_frame_; ++frame_it) { if (!frame_it->second.continuous || frame_it->second.num_missing_decodable > 0) { continue; } EncodedFrame* frame = frame_it->second.frame.get(); if (keyframe_required_ && !frame->is_keyframe()) continue; auto last_decoded_frame_timestamp = decoded_frames_history_.GetLastDecodedFrameTimestamp(); // TODO(https://bugs.webrtc.org/9974): consider removing this check // as it may make a stream undecodable after a very long delay between // frames. if (last_decoded_frame_timestamp && AheadOf(*last_decoded_frame_timestamp, frame->Timestamp())) { continue; } // Gather all remaining frames for the same superframe. std::vector current_superframe; current_superframe.push_back(frame_it); bool last_layer_completed = frame_it->second.frame->is_last_spatial_layer; FrameMap::iterator next_frame_it = frame_it; while (!last_layer_completed) { ++next_frame_it; if (next_frame_it == frames_.end() || !next_frame_it->second.frame) { break; } if (next_frame_it->second.frame->Timestamp() != frame->Timestamp() || !next_frame_it->second.continuous) { break; } if (next_frame_it->second.num_missing_decodable > 0) { bool has_inter_layer_dependency = false; for (size_t i = 0; i < EncodedFrame::kMaxFrameReferences && i < next_frame_it->second.frame->num_references; ++i) { if (next_frame_it->second.frame->references[i] >= frame_it->first) { has_inter_layer_dependency = true; break; } } // If the frame has an undecoded dependency that is not within the same // temporal unit then this frame is not yet ready to be decoded. If it // is within the same temporal unit then the not yet decoded dependency // is just a lower spatial frame, which is ok. if (!has_inter_layer_dependency || next_frame_it->second.num_missing_decodable > 1) { break; } } current_superframe.push_back(next_frame_it); last_layer_completed = next_frame_it->second.frame->is_last_spatial_layer; } // Check if the current superframe is complete. // TODO(bugs.webrtc.org/10064): consider returning all available to // decode frames even if the superframe is not complete yet. if (!last_layer_completed) { continue; } frames_to_decode_ = std::move(current_superframe); absl::optional render_time = frame->RenderTimestamp(); if (!render_time) { render_time = timing_->RenderTime(frame->Timestamp(), now); frame->SetRenderTime(render_time->ms()); } bool too_many_frames_queued = frames_.size() > zero_playout_delay_max_decode_queue_size_ ? true : false; wait_ms = timing_->MaxWaitingTime(*render_time, now, too_many_frames_queued).ms(); // This will cause the frame buffer to prefer high framerate rather // than high resolution in the case of the decoder not decoding fast // enough and the stream has multiple spatial and temporal layers. // For multiple temporal layers it may cause non-base layer frames to be // skipped if they are late. if (wait_ms < -kMaxAllowedFrameDelayMs) continue; break; } wait_ms = std::min(wait_ms, latest_return_time_ms_ - now.ms()); wait_ms = std::max(wait_ms, 0); return wait_ms; } std::unique_ptr FrameBuffer::GetNextFrame() { RTC_DCHECK_RUN_ON(&callback_checker_); Timestamp now = clock_->CurrentTime(); // TODO(ilnik): remove `frames_out` use frames_to_decode_ directly. std::vector> frames_out; RTC_DCHECK(!frames_to_decode_.empty()); bool superframe_delayed_by_retransmission = false; DataSize superframe_size = DataSize::Zero(); const EncodedFrame& first_frame = *frames_to_decode_[0]->second.frame; absl::optional render_time = first_frame.RenderTimestamp(); int64_t receive_time_ms = first_frame.ReceivedTime(); // Gracefully handle bad RTP timestamps and render time issues. if (!render_time || 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 (FrameMap::iterator& frame_it : frames_to_decode_) { RTC_DCHECK(frame_it != frames_.end()); std::unique_ptr frame = std::move(frame_it->second.frame); frame->SetRenderTime(render_time->ms()); superframe_delayed_by_retransmission |= frame->delayed_by_retransmission(); receive_time_ms = std::max(receive_time_ms, frame->ReceivedTime()); superframe_size += DataSize::Bytes(frame->size()); PropagateDecodability(frame_it->second); decoded_frames_history_.InsertDecoded(frame_it->first, frame->Timestamp()); frames_.erase(frames_.begin(), ++frame_it); frames_out.emplace_back(std::move(frame)); } if (!superframe_delayed_by_retransmission) { auto frame_delay = inter_frame_delay_.CalculateDelay( first_frame.Timestamp(), Timestamp::Millis(receive_time_ms)); 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(); } if (frames_out.size() == 1) { return std::move(frames_out[0]); } else { return CombineAndDeleteFrames(std::move(frames_out)); } } void FrameBuffer::SetProtectionMode(VCMVideoProtection mode) { TRACE_EVENT0("webrtc", "FrameBuffer::SetProtectionMode"); MutexLock lock(&mutex_); protection_mode_ = mode; } void FrameBuffer::Stop() { TRACE_EVENT0("webrtc", "FrameBuffer::Stop"); MutexLock lock(&mutex_); if (stopped_) return; stopped_ = true; CancelCallback(); } void FrameBuffer::Clear() { MutexLock lock(&mutex_); ClearFramesAndHistory(); } int FrameBuffer::Size() { MutexLock lock(&mutex_); return frames_.size(); } void FrameBuffer::UpdateRtt(int64_t rtt_ms) { MutexLock lock(&mutex_); jitter_estimator_.UpdateRtt(TimeDelta::Millis(rtt_ms)); } bool FrameBuffer::ValidReferences(const EncodedFrame& frame) const { for (size_t i = 0; i < frame.num_references; ++i) { if (frame.references[i] >= frame.Id()) return false; for (size_t j = i + 1; j < frame.num_references; ++j) { if (frame.references[i] == frame.references[j]) return false; } } return true; } void FrameBuffer::CancelCallback() { // Called from the callback queue or from within Stop(). frame_handler_ = {}; callback_task_.Stop(); callback_queue_ = nullptr; callback_checker_.Detach(); } int64_t FrameBuffer::InsertFrame(std::unique_ptr frame) { TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame"); RTC_DCHECK(frame); MutexLock lock(&mutex_); const auto& pis = frame->PacketInfos(); int64_t last_continuous_frame_id = last_continuous_frame_.value_or(-1); if (!ValidReferences(*frame)) { TRACE_EVENT2("webrtc", "FrameBuffer::InsertFrame Frame dropped (Invalid references)", "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id", frame->Id()); RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " has invalid frame references, dropping frame."; return last_continuous_frame_id; } if (frames_.size() >= kMaxFramesBuffered) { if (frame->is_keyframe()) { TRACE_EVENT2("webrtc", "FrameBuffer::InsertFrame Frames dropped (KF + Full buffer)", "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id", frame->Id()); RTC_LOG(LS_WARNING) << "Inserting keyframe " << frame->Id() << " but buffer is full, clearing" " buffer and inserting the frame."; ClearFramesAndHistory(); } else { TRACE_EVENT2("webrtc", "FrameBuffer::InsertFrame Frame dropped (Full buffer)", "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id", frame->Id()); RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " could not be inserted due to the frame " "buffer being full, dropping frame."; return last_continuous_frame_id; } } auto last_decoded_frame = decoded_frames_history_.GetLastDecodedFrameId(); auto last_decoded_frame_timestamp = decoded_frames_history_.GetLastDecodedFrameTimestamp(); if (last_decoded_frame && frame->Id() <= *last_decoded_frame) { if (AheadOf(frame->Timestamp(), *last_decoded_frame_timestamp) && frame->is_keyframe()) { // If this frame has a newer timestamp but an earlier frame id then we // assume there has been a jump in the frame id due to some encoder // reconfiguration or some other reason. Even though this is not according // to spec we can still continue to decode from this frame if it is a // keyframe. TRACE_EVENT2("webrtc", "FrameBuffer::InsertFrame Frames dropped (OOO + PicId jump)", "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id", frame->Id()); RTC_LOG(LS_WARNING) << "A jump in frame id was detected, clearing buffer."; ClearFramesAndHistory(); last_continuous_frame_id = -1; } else { TRACE_EVENT2("webrtc", "FrameBuffer::InsertFrame Frame dropped (Out of order)", "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id", frame->Id()); RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " inserted after frame " << *last_decoded_frame << " was handed off for decoding, dropping frame."; return last_continuous_frame_id; } } // Test if inserting this frame would cause the order of the frames to become // ambiguous (covering more than half the interval of 2^16). This can happen // when the frame id make large jumps mid stream. if (!frames_.empty() && frame->Id() < frames_.begin()->first && frames_.rbegin()->first < frame->Id()) { TRACE_EVENT2("webrtc", "FrameBuffer::InsertFrame Frames dropped (PicId big-jump)", "remote_ssrc", pis.empty() ? 0 : pis[0].ssrc(), "picture_id", frame->Id()); RTC_LOG(LS_WARNING) << "A jump in frame id was detected, clearing buffer."; ClearFramesAndHistory(); last_continuous_frame_id = -1; } auto info = frames_.emplace(frame->Id(), FrameInfo()).first; if (info->second.frame) { return last_continuous_frame_id; } if (!UpdateFrameInfoWithIncomingFrame(*frame, info)) return last_continuous_frame_id; // If ReceiveTime is negative then it is not a valid timestamp. if (!frame->delayed_by_retransmission() && frame->ReceivedTime() >= 0) timing_->IncomingTimestamp(frame->Timestamp(), Timestamp::Millis(frame->ReceivedTime())); // It can happen that a frame will be reported as fully received even if a // lower spatial layer frame is missing. info->second.frame = std::move(frame); if (info->second.num_missing_continuous == 0) { info->second.continuous = true; PropagateContinuity(info); last_continuous_frame_id = *last_continuous_frame_; // Since we now have new continuous frames there might be a better frame // to return from NextFrame. if (callback_queue_) { callback_queue_->PostTask([this] { MutexLock lock(&mutex_); if (!callback_task_.Running()) return; RTC_CHECK(frame_handler_); callback_task_.Stop(); StartWaitForNextFrameOnQueue(); }); } } return last_continuous_frame_id; } void FrameBuffer::PropagateContinuity(FrameMap::iterator start) { TRACE_EVENT0("webrtc", "FrameBuffer::PropagateContinuity"); RTC_DCHECK(start->second.continuous); std::queue continuous_frames; continuous_frames.push(start); // A simple BFS to traverse continuous frames. while (!continuous_frames.empty()) { auto frame = continuous_frames.front(); continuous_frames.pop(); if (!last_continuous_frame_ || *last_continuous_frame_ < frame->first) { last_continuous_frame_ = frame->first; } // Loop through all dependent frames, and if that frame no longer has // any unfulfilled dependencies then that frame is continuous as well. for (size_t d = 0; d < frame->second.dependent_frames.size(); ++d) { auto frame_ref = frames_.find(frame->second.dependent_frames[d]); RTC_DCHECK(frame_ref != frames_.end()); // TODO(philipel): Look into why we've seen this happen. if (frame_ref != frames_.end()) { --frame_ref->second.num_missing_continuous; if (frame_ref->second.num_missing_continuous == 0) { frame_ref->second.continuous = true; continuous_frames.push(frame_ref); } } } } } void FrameBuffer::PropagateDecodability(const FrameInfo& info) { TRACE_EVENT0("webrtc", "FrameBuffer::PropagateDecodability"); for (size_t d = 0; d < info.dependent_frames.size(); ++d) { auto ref_info = frames_.find(info.dependent_frames[d]); RTC_DCHECK(ref_info != frames_.end()); // TODO(philipel): Look into why we've seen this happen. if (ref_info != frames_.end()) { RTC_DCHECK_GT(ref_info->second.num_missing_decodable, 0U); --ref_info->second.num_missing_decodable; } } } bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame, FrameMap::iterator info) { TRACE_EVENT0("webrtc", "FrameBuffer::UpdateFrameInfoWithIncomingFrame"); auto last_decoded_frame = decoded_frames_history_.GetLastDecodedFrameId(); RTC_DCHECK(!last_decoded_frame || *last_decoded_frame < info->first); // In this function we determine how many missing dependencies this `frame` // has to become continuous/decodable. If a frame that this `frame` depend // on has already been decoded then we can ignore that dependency since it has // already been fulfilled. // // For all other frames we will register a backwards reference to this `frame` // so that `num_missing_continuous` and `num_missing_decodable` can be // decremented as frames become continuous/are decoded. struct Dependency { int64_t frame_id; bool continuous; }; std::vector not_yet_fulfilled_dependencies; // Find all dependencies that have not yet been fulfilled. for (size_t i = 0; i < frame.num_references; ++i) { // Does `frame` depend on a frame earlier than the last decoded one? if (last_decoded_frame && frame.references[i] <= *last_decoded_frame) { // Was that frame decoded? If not, this `frame` will never become // decodable. if (!decoded_frames_history_.WasDecoded(frame.references[i])) { int64_t now_ms = clock_->TimeInMilliseconds(); if (last_log_non_decoded_ms_ + kLogNonDecodedIntervalMs < now_ms) { RTC_LOG(LS_WARNING) << "Frame " << frame.Id() << " depends on a non-decoded frame more previous than the last " "decoded frame, dropping frame."; last_log_non_decoded_ms_ = now_ms; } return false; } } else { auto ref_info = frames_.find(frame.references[i]); bool ref_continuous = ref_info != frames_.end() && ref_info->second.continuous; not_yet_fulfilled_dependencies.push_back( {frame.references[i], ref_continuous}); } } info->second.num_missing_continuous = not_yet_fulfilled_dependencies.size(); info->second.num_missing_decodable = not_yet_fulfilled_dependencies.size(); for (const Dependency& dep : not_yet_fulfilled_dependencies) { if (dep.continuous) --info->second.num_missing_continuous; frames_[dep.frame_id].dependent_frames.push_back(frame.Id()); } return true; } void FrameBuffer::ClearFramesAndHistory() { TRACE_EVENT0("webrtc", "FrameBuffer::ClearFramesAndHistory"); frames_.clear(); last_continuous_frame_.reset(); frames_to_decode_.clear(); decoded_frames_history_.Clear(); } // TODO(philipel): Avoid the concatenation of frames here, by replacing // NextFrame and GetNextFrame with methods returning multiple frames. std::unique_ptr FrameBuffer::CombineAndDeleteFrames( std::vector> frames) const { RTC_DCHECK(!frames.empty()); absl::InlinedVector, 4> inlined; for (auto& frame : frames) { inlined.push_back(std::move(frame)); } return webrtc::CombineAndDeleteFrames(std::move(inlined)); } FrameBuffer::FrameInfo::FrameInfo() = default; FrameBuffer::FrameInfo::FrameInfo(FrameInfo&&) = default; FrameBuffer::FrameInfo::~FrameInfo() = default; } // namespace video_coding } // namespace webrtc