diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/video_coding/frame_buffer2.cc | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/video_coding/frame_buffer2.cc')
-rw-r--r-- | third_party/libwebrtc/modules/video_coding/frame_buffer2.cc | 622 |
1 files changed, 622 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/video_coding/frame_buffer2.cc b/third_party/libwebrtc/modules/video_coding/frame_buffer2.cc new file mode 100644 index 0000000000..813ac69dd6 --- /dev/null +++ b/third_party/libwebrtc/modules/video_coding/frame_buffer2.cc @@ -0,0 +1,622 @@ +/* + * 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/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<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()); + + 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(); + } + + 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_); + + 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<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::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<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 |