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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 09:22:09 +0000
commit43a97878ce14b72f0981164f87f2e35e14151312 (patch)
tree620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/video_coding/frame_buffer2.cc
parentInitial commit. (diff)
downloadfirefox-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.cc622
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