| /* | |
| * 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 <algorithm> | |
| #include <cstdlib> | |
| #include <iterator> | |
| #include <memory> | |
| #include <queue> | |
| #include <utility> | |
| #include <vector> | |
| #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/jitter_estimator.h" | |
| #include "modules/video_coding/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" | |
| #include "system_wrappers/include/field_trial.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, | |
| VCMReceiveStatisticsCallback* stats_callback) | |
| : decoded_frames_history_(kMaxFramesHistory), | |
| clock_(clock), | |
| callback_queue_(nullptr), | |
| jitter_estimator_(clock), | |
| timing_(timing), | |
| stopped_(false), | |
| protection_mode_(kProtectionNack), | |
| stats_callback_(stats_callback), | |
| 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_trial::FindFullName("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, | |
| rtc::TaskQueue* 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_->Get(), 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<EncodedFrame> 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<FrameMap::iterator> 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<Timestamp> 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<int64_t>(wait_ms, latest_return_time_ms_ - now.ms()); | |
| wait_ms = std::max<int64_t>(wait_ms, 0); | |
| return wait_ms; | |
| } | |
| std::unique_ptr<EncodedFrame> FrameBuffer::GetNextFrame() { | |
| RTC_DCHECK_RUN_ON(&callback_checker_); | |
| Timestamp now = clock_->CurrentTime(); | |
| // TODO(ilnik): remove `frames_out` use frames_to_decode_ directly. | |
| std::vector<std::unique_ptr<EncodedFrame>> 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<Timestamp> 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, timing_->TargetVideoDelay())) { | |
| 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<EncodedFrame> 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()); | |
| // Remove decoded frame and all undecoded frames before it. | |
| if (stats_callback_) { | |
| unsigned int dropped_frames = | |
| std::count_if(frames_.begin(), frame_it, | |
| [](const std::pair<const int64_t, FrameInfo>& frame) { | |
| return frame.second.frame != nullptr; | |
| }); | |
| if (dropped_frames > 0) { | |
| stats_callback_->OnDroppedFrames(dropped_frames); | |
| } | |
| } | |
| 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<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(); | |
| } | |
| UpdateJitterDelay(); | |
| UpdateTimingFrameInfo(); | |
| 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<EncodedFrame> frame) { | |
| TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame"); | |
| RTC_DCHECK(frame); | |
| MutexLock lock(&mutex_); | |
| int64_t last_continuous_frame_id = last_continuous_frame_.value_or(-1); | |
| if (!ValidReferences(*frame)) { | |
| 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()) { | |
| RTC_LOG(LS_WARNING) << "Inserting keyframe " << frame->Id() | |
| << " but buffer is full, clearing" | |
| " buffer and inserting the frame."; | |
| ClearFramesAndHistory(); | |
| } else { | |
| 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. | |
| RTC_LOG(LS_WARNING) | |
| << "A jump in frame id was detected, clearing buffer."; | |
| ClearFramesAndHistory(); | |
| last_continuous_frame_id = -1; | |
| } else { | |
| 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()) { | |
| 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. | |
| if (stats_callback_ && frame->is_last_spatial_layer) { | |
| stats_callback_->OnCompleteFrame(frame->is_keyframe(), frame->size(), | |
| frame->contentType()); | |
| } | |
| 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<FrameMap::iterator> 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<Dependency> 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::UpdateJitterDelay() { | |
| TRACE_EVENT0("webrtc", "FrameBuffer::UpdateJitterDelay"); | |
| if (!stats_callback_) | |
| return; | |
| auto timings = timing_->GetTimings(); | |
| if (timings.num_decoded_frames > 0) { | |
| stats_callback_->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 FrameBuffer::UpdateTimingFrameInfo() { | |
| TRACE_EVENT0("webrtc", "FrameBuffer::UpdateTimingFrameInfo"); | |
| absl::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo(); | |
| if (info && stats_callback_) | |
| stats_callback_->OnTimingFrameInfoUpdated(*info); | |
| } | |
| void FrameBuffer::ClearFramesAndHistory() { | |
| TRACE_EVENT0("webrtc", "FrameBuffer::ClearFramesAndHistory"); | |
| if (stats_callback_) { | |
| unsigned int dropped_frames = | |
| std::count_if(frames_.begin(), frames_.end(), | |
| [](const std::pair<const int64_t, FrameInfo>& frame) { | |
| return frame.second.frame != nullptr; | |
| }); | |
| if (dropped_frames > 0) { | |
| stats_callback_->OnDroppedFrames(dropped_frames); | |
| } | |
| } | |
| 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<EncodedFrame> FrameBuffer::CombineAndDeleteFrames( | |
| std::vector<std::unique_ptr<EncodedFrame>> frames) const { | |
| RTC_DCHECK(!frames.empty()); | |
| absl::InlinedVector<std::unique_ptr<EncodedFrame>, 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 |