1 /*
2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/video_coding/frame_buffer2.h"
12
13 #include <algorithm>
14 #include <cstdlib>
15 #include <iterator>
16 #include <memory>
17 #include <queue>
18 #include <utility>
19 #include <vector>
20
21 #include "absl/container/inlined_vector.h"
22 #include "api/units/data_size.h"
23 #include "api/units/time_delta.h"
24 #include "api/video/encoded_image.h"
25 #include "api/video/video_timing.h"
26 #include "modules/video_coding/frame_helpers.h"
27 #include "modules/video_coding/include/video_coding_defines.h"
28 #include "modules/video_coding/timing/jitter_estimator.h"
29 #include "modules/video_coding/timing/timing.h"
30 #include "rtc_base/checks.h"
31 #include "rtc_base/experiments/rtt_mult_experiment.h"
32 #include "rtc_base/logging.h"
33 #include "rtc_base/numerics/sequence_number_util.h"
34 #include "rtc_base/trace_event.h"
35 #include "system_wrappers/include/clock.h"
36
37 namespace webrtc {
38 namespace video_coding {
39
40 namespace {
41 // Max number of frames the buffer will hold.
42 constexpr size_t kMaxFramesBuffered = 800;
43
44 // Default value for the maximum decode queue size that is used when the
45 // low-latency renderer is used.
46 constexpr size_t kZeroPlayoutDelayDefaultMaxDecodeQueueSize = 8;
47
48 // Max number of decoded frame info that will be saved.
49 constexpr int kMaxFramesHistory = 1 << 13;
50
51 // The time it's allowed for a frame to be late to its rendering prediction and
52 // still be rendered.
53 constexpr int kMaxAllowedFrameDelayMs = 5;
54
55 constexpr int64_t kLogNonDecodedIntervalMs = 5000;
56 } // namespace
57
FrameBuffer(Clock * clock,VCMTiming * timing,const FieldTrialsView & field_trials)58 FrameBuffer::FrameBuffer(Clock* clock,
59 VCMTiming* timing,
60 const FieldTrialsView& field_trials)
61 : decoded_frames_history_(kMaxFramesHistory),
62 clock_(clock),
63 callback_queue_(nullptr),
64 jitter_estimator_(clock, field_trials),
65 timing_(timing),
66 stopped_(false),
67 protection_mode_(kProtectionNack),
68 last_log_non_decoded_ms_(-kLogNonDecodedIntervalMs),
69 rtt_mult_settings_(RttMultExperiment::GetRttMultValue()),
70 zero_playout_delay_max_decode_queue_size_(
71 "max_decode_queue_size",
72 kZeroPlayoutDelayDefaultMaxDecodeQueueSize) {
73 ParseFieldTrial({&zero_playout_delay_max_decode_queue_size_},
74 field_trials.Lookup("WebRTC-ZeroPlayoutDelay"));
75 callback_checker_.Detach();
76 }
77
~FrameBuffer()78 FrameBuffer::~FrameBuffer() {
79 RTC_DCHECK_RUN_ON(&construction_checker_);
80 }
81
NextFrame(int64_t max_wait_time_ms,bool keyframe_required,TaskQueueBase * callback_queue,NextFrameCallback handler)82 void FrameBuffer::NextFrame(int64_t max_wait_time_ms,
83 bool keyframe_required,
84 TaskQueueBase* callback_queue,
85 NextFrameCallback handler) {
86 RTC_DCHECK_RUN_ON(&callback_checker_);
87 RTC_DCHECK(callback_queue->IsCurrent());
88 TRACE_EVENT0("webrtc", "FrameBuffer::NextFrame");
89 int64_t latest_return_time_ms =
90 clock_->TimeInMilliseconds() + max_wait_time_ms;
91
92 MutexLock lock(&mutex_);
93 if (stopped_) {
94 return;
95 }
96 latest_return_time_ms_ = latest_return_time_ms;
97 keyframe_required_ = keyframe_required;
98 frame_handler_ = handler;
99 callback_queue_ = callback_queue;
100 StartWaitForNextFrameOnQueue();
101 }
102
StartWaitForNextFrameOnQueue()103 void FrameBuffer::StartWaitForNextFrameOnQueue() {
104 RTC_DCHECK(callback_queue_);
105 RTC_DCHECK(!callback_task_.Running());
106 int64_t wait_ms = FindNextFrame(clock_->CurrentTime());
107 callback_task_ = RepeatingTaskHandle::DelayedStart(
108 callback_queue_, TimeDelta::Millis(wait_ms),
109 [this] {
110 RTC_DCHECK_RUN_ON(&callback_checker_);
111 // If this task has not been cancelled, we did not get any new frames
112 // while waiting. Continue with frame delivery.
113 std::unique_ptr<EncodedFrame> frame;
114 NextFrameCallback frame_handler;
115 {
116 MutexLock lock(&mutex_);
117 if (!frames_to_decode_.empty()) {
118 // We have frames, deliver!
119 frame = GetNextFrame();
120 timing_->SetLastDecodeScheduledTimestamp(clock_->CurrentTime());
121 } else if (clock_->TimeInMilliseconds() < latest_return_time_ms_) {
122 // If there's no frames to decode and there is still time left, it
123 // means that the frame buffer was cleared between creation and
124 // execution of this task. Continue waiting for the remaining time.
125 int64_t wait_ms = FindNextFrame(clock_->CurrentTime());
126 return TimeDelta::Millis(wait_ms);
127 }
128 frame_handler = std::move(frame_handler_);
129 CancelCallback();
130 }
131 // Deliver frame, if any. Otherwise signal timeout.
132 frame_handler(std::move(frame));
133 return TimeDelta::Zero(); // Ignored.
134 },
135 TaskQueueBase::DelayPrecision::kHigh);
136 }
137
FindNextFrame(Timestamp now)138 int64_t FrameBuffer::FindNextFrame(Timestamp now) {
139 int64_t wait_ms = latest_return_time_ms_ - now.ms();
140 frames_to_decode_.clear();
141
142 // `last_continuous_frame_` may be empty below, but nullopt is smaller
143 // than everything else and loop will immediately terminate as expected.
144 for (auto frame_it = frames_.begin();
145 frame_it != frames_.end() && frame_it->first <= last_continuous_frame_;
146 ++frame_it) {
147 if (!frame_it->second.continuous ||
148 frame_it->second.num_missing_decodable > 0) {
149 continue;
150 }
151
152 EncodedFrame* frame = frame_it->second.frame.get();
153
154 if (keyframe_required_ && !frame->is_keyframe())
155 continue;
156
157 auto last_decoded_frame_timestamp =
158 decoded_frames_history_.GetLastDecodedFrameTimestamp();
159
160 // TODO(https://bugs.webrtc.org/9974): consider removing this check
161 // as it may make a stream undecodable after a very long delay between
162 // frames.
163 if (last_decoded_frame_timestamp &&
164 AheadOf(*last_decoded_frame_timestamp, frame->Timestamp())) {
165 continue;
166 }
167
168 // Gather all remaining frames for the same superframe.
169 std::vector<FrameMap::iterator> current_superframe;
170 current_superframe.push_back(frame_it);
171 bool last_layer_completed = frame_it->second.frame->is_last_spatial_layer;
172 FrameMap::iterator next_frame_it = frame_it;
173 while (!last_layer_completed) {
174 ++next_frame_it;
175
176 if (next_frame_it == frames_.end() || !next_frame_it->second.frame) {
177 break;
178 }
179
180 if (next_frame_it->second.frame->Timestamp() != frame->Timestamp() ||
181 !next_frame_it->second.continuous) {
182 break;
183 }
184
185 if (next_frame_it->second.num_missing_decodable > 0) {
186 bool has_inter_layer_dependency = false;
187 for (size_t i = 0; i < EncodedFrame::kMaxFrameReferences &&
188 i < next_frame_it->second.frame->num_references;
189 ++i) {
190 if (next_frame_it->second.frame->references[i] >= frame_it->first) {
191 has_inter_layer_dependency = true;
192 break;
193 }
194 }
195
196 // If the frame has an undecoded dependency that is not within the same
197 // temporal unit then this frame is not yet ready to be decoded. If it
198 // is within the same temporal unit then the not yet decoded dependency
199 // is just a lower spatial frame, which is ok.
200 if (!has_inter_layer_dependency ||
201 next_frame_it->second.num_missing_decodable > 1) {
202 break;
203 }
204 }
205
206 current_superframe.push_back(next_frame_it);
207 last_layer_completed = next_frame_it->second.frame->is_last_spatial_layer;
208 }
209 // Check if the current superframe is complete.
210 // TODO(bugs.webrtc.org/10064): consider returning all available to
211 // decode frames even if the superframe is not complete yet.
212 if (!last_layer_completed) {
213 continue;
214 }
215
216 frames_to_decode_ = std::move(current_superframe);
217
218 absl::optional<Timestamp> render_time = frame->RenderTimestamp();
219 if (!render_time) {
220 render_time = timing_->RenderTime(frame->Timestamp(), now);
221 frame->SetRenderTime(render_time->ms());
222 }
223 bool too_many_frames_queued =
224 frames_.size() > zero_playout_delay_max_decode_queue_size_ ? true
225 : false;
226 wait_ms =
227 timing_->MaxWaitingTime(*render_time, now, too_many_frames_queued).ms();
228
229 // This will cause the frame buffer to prefer high framerate rather
230 // than high resolution in the case of the decoder not decoding fast
231 // enough and the stream has multiple spatial and temporal layers.
232 // For multiple temporal layers it may cause non-base layer frames to be
233 // skipped if they are late.
234 if (wait_ms < -kMaxAllowedFrameDelayMs)
235 continue;
236
237 break;
238 }
239 wait_ms = std::min<int64_t>(wait_ms, latest_return_time_ms_ - now.ms());
240 wait_ms = std::max<int64_t>(wait_ms, 0);
241 return wait_ms;
242 }
243
GetNextFrame()244 std::unique_ptr<EncodedFrame> FrameBuffer::GetNextFrame() {
245 RTC_DCHECK_RUN_ON(&callback_checker_);
246 Timestamp now = clock_->CurrentTime();
247 // TODO(ilnik): remove `frames_out` use frames_to_decode_ directly.
248 std::vector<std::unique_ptr<EncodedFrame>> frames_out;
249
250 RTC_DCHECK(!frames_to_decode_.empty());
251 bool superframe_delayed_by_retransmission = false;
252 DataSize superframe_size = DataSize::Zero();
253 const EncodedFrame& first_frame = *frames_to_decode_[0]->second.frame;
254 absl::optional<Timestamp> render_time = first_frame.RenderTimestamp();
255 int64_t receive_time_ms = first_frame.ReceivedTime();
256 // Gracefully handle bad RTP timestamps and render time issues.
257 if (!render_time || FrameHasBadRenderTiming(*render_time, now) ||
258 TargetVideoDelayIsTooLarge(timing_->TargetVideoDelay())) {
259 RTC_LOG(LS_WARNING) << "Resetting jitter estimator and timing module due "
260 "to bad render timing for rtp_timestamp="
261 << first_frame.Timestamp();
262 jitter_estimator_.Reset();
263 timing_->Reset();
264 render_time = timing_->RenderTime(first_frame.Timestamp(), now);
265 }
266
267 for (FrameMap::iterator& frame_it : frames_to_decode_) {
268 RTC_DCHECK(frame_it != frames_.end());
269 std::unique_ptr<EncodedFrame> frame = std::move(frame_it->second.frame);
270
271 frame->SetRenderTime(render_time->ms());
272
273 superframe_delayed_by_retransmission |= frame->delayed_by_retransmission();
274 receive_time_ms = std::max(receive_time_ms, frame->ReceivedTime());
275 superframe_size += DataSize::Bytes(frame->size());
276
277 PropagateDecodability(frame_it->second);
278 decoded_frames_history_.InsertDecoded(frame_it->first, frame->Timestamp());
279
280 frames_.erase(frames_.begin(), ++frame_it);
281
282 frames_out.emplace_back(std::move(frame));
283 }
284
285 if (!superframe_delayed_by_retransmission) {
286 auto frame_delay = inter_frame_delay_.CalculateDelay(
287 first_frame.Timestamp(), Timestamp::Millis(receive_time_ms));
288
289 if (frame_delay) {
290 jitter_estimator_.UpdateEstimate(*frame_delay, superframe_size);
291 }
292
293 float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
294 absl::optional<TimeDelta> rtt_mult_add_cap_ms = absl::nullopt;
295 if (rtt_mult_settings_.has_value()) {
296 rtt_mult = rtt_mult_settings_->rtt_mult_setting;
297 rtt_mult_add_cap_ms =
298 TimeDelta::Millis(rtt_mult_settings_->rtt_mult_add_cap_ms);
299 }
300 timing_->SetJitterDelay(
301 jitter_estimator_.GetJitterEstimate(rtt_mult, rtt_mult_add_cap_ms));
302 timing_->UpdateCurrentDelay(*render_time, now);
303 } else {
304 if (RttMultExperiment::RttMultEnabled())
305 jitter_estimator_.FrameNacked();
306 }
307
308 if (frames_out.size() == 1) {
309 return std::move(frames_out[0]);
310 } else {
311 return CombineAndDeleteFrames(std::move(frames_out));
312 }
313 }
314
SetProtectionMode(VCMVideoProtection mode)315 void FrameBuffer::SetProtectionMode(VCMVideoProtection mode) {
316 TRACE_EVENT0("webrtc", "FrameBuffer::SetProtectionMode");
317 MutexLock lock(&mutex_);
318 protection_mode_ = mode;
319 }
320
Stop()321 void FrameBuffer::Stop() {
322 TRACE_EVENT0("webrtc", "FrameBuffer::Stop");
323 MutexLock lock(&mutex_);
324 if (stopped_)
325 return;
326 stopped_ = true;
327
328 CancelCallback();
329 }
330
Clear()331 void FrameBuffer::Clear() {
332 MutexLock lock(&mutex_);
333 ClearFramesAndHistory();
334 }
335
Size()336 int FrameBuffer::Size() {
337 MutexLock lock(&mutex_);
338 return frames_.size();
339 }
340
UpdateRtt(int64_t rtt_ms)341 void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
342 MutexLock lock(&mutex_);
343 jitter_estimator_.UpdateRtt(TimeDelta::Millis(rtt_ms));
344 }
345
ValidReferences(const EncodedFrame & frame) const346 bool FrameBuffer::ValidReferences(const EncodedFrame& frame) const {
347 for (size_t i = 0; i < frame.num_references; ++i) {
348 if (frame.references[i] >= frame.Id())
349 return false;
350
351 for (size_t j = i + 1; j < frame.num_references; ++j) {
352 if (frame.references[i] == frame.references[j])
353 return false;
354 }
355 }
356
357 return true;
358 }
359
CancelCallback()360 void FrameBuffer::CancelCallback() {
361 // Called from the callback queue or from within Stop().
362 frame_handler_ = {};
363 callback_task_.Stop();
364 callback_queue_ = nullptr;
365 callback_checker_.Detach();
366 }
367
InsertFrame(std::unique_ptr<EncodedFrame> frame)368 int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
369 TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
370 RTC_DCHECK(frame);
371
372 MutexLock lock(&mutex_);
373
374 int64_t last_continuous_frame_id = last_continuous_frame_.value_or(-1);
375
376 if (!ValidReferences(*frame)) {
377 RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
378 << " has invalid frame references, dropping frame.";
379 return last_continuous_frame_id;
380 }
381
382 if (frames_.size() >= kMaxFramesBuffered) {
383 if (frame->is_keyframe()) {
384 RTC_LOG(LS_WARNING) << "Inserting keyframe " << frame->Id()
385 << " but buffer is full, clearing"
386 " buffer and inserting the frame.";
387 ClearFramesAndHistory();
388 } else {
389 RTC_LOG(LS_WARNING) << "Frame " << frame->Id()
390 << " could not be inserted due to the frame "
391 "buffer being full, dropping frame.";
392 return last_continuous_frame_id;
393 }
394 }
395
396 auto last_decoded_frame = decoded_frames_history_.GetLastDecodedFrameId();
397 auto last_decoded_frame_timestamp =
398 decoded_frames_history_.GetLastDecodedFrameTimestamp();
399 if (last_decoded_frame && frame->Id() <= *last_decoded_frame) {
400 if (AheadOf(frame->Timestamp(), *last_decoded_frame_timestamp) &&
401 frame->is_keyframe()) {
402 // If this frame has a newer timestamp but an earlier frame id then we
403 // assume there has been a jump in the frame id due to some encoder
404 // reconfiguration or some other reason. Even though this is not according
405 // to spec we can still continue to decode from this frame if it is a
406 // keyframe.
407 RTC_LOG(LS_WARNING)
408 << "A jump in frame id was detected, clearing buffer.";
409 ClearFramesAndHistory();
410 last_continuous_frame_id = -1;
411 } else {
412 RTC_LOG(LS_WARNING) << "Frame " << frame->Id() << " inserted after frame "
413 << *last_decoded_frame
414 << " was handed off for decoding, dropping frame.";
415 return last_continuous_frame_id;
416 }
417 }
418
419 // Test if inserting this frame would cause the order of the frames to become
420 // ambiguous (covering more than half the interval of 2^16). This can happen
421 // when the frame id make large jumps mid stream.
422 if (!frames_.empty() && frame->Id() < frames_.begin()->first &&
423 frames_.rbegin()->first < frame->Id()) {
424 RTC_LOG(LS_WARNING) << "A jump in frame id was detected, clearing buffer.";
425 ClearFramesAndHistory();
426 last_continuous_frame_id = -1;
427 }
428
429 auto info = frames_.emplace(frame->Id(), FrameInfo()).first;
430
431 if (info->second.frame) {
432 return last_continuous_frame_id;
433 }
434
435 if (!UpdateFrameInfoWithIncomingFrame(*frame, info))
436 return last_continuous_frame_id;
437
438 // If ReceiveTime is negative then it is not a valid timestamp.
439 if (!frame->delayed_by_retransmission() && frame->ReceivedTime() >= 0)
440 timing_->IncomingTimestamp(frame->Timestamp(),
441 Timestamp::Millis(frame->ReceivedTime()));
442
443 // It can happen that a frame will be reported as fully received even if a
444 // lower spatial layer frame is missing.
445 info->second.frame = std::move(frame);
446
447 if (info->second.num_missing_continuous == 0) {
448 info->second.continuous = true;
449 PropagateContinuity(info);
450 last_continuous_frame_id = *last_continuous_frame_;
451
452 // Since we now have new continuous frames there might be a better frame
453 // to return from NextFrame.
454 if (callback_queue_) {
455 callback_queue_->PostTask([this] {
456 MutexLock lock(&mutex_);
457 if (!callback_task_.Running())
458 return;
459 RTC_CHECK(frame_handler_);
460 callback_task_.Stop();
461 StartWaitForNextFrameOnQueue();
462 });
463 }
464 }
465
466 return last_continuous_frame_id;
467 }
468
PropagateContinuity(FrameMap::iterator start)469 void FrameBuffer::PropagateContinuity(FrameMap::iterator start) {
470 TRACE_EVENT0("webrtc", "FrameBuffer::PropagateContinuity");
471 RTC_DCHECK(start->second.continuous);
472
473 std::queue<FrameMap::iterator> continuous_frames;
474 continuous_frames.push(start);
475
476 // A simple BFS to traverse continuous frames.
477 while (!continuous_frames.empty()) {
478 auto frame = continuous_frames.front();
479 continuous_frames.pop();
480
481 if (!last_continuous_frame_ || *last_continuous_frame_ < frame->first) {
482 last_continuous_frame_ = frame->first;
483 }
484
485 // Loop through all dependent frames, and if that frame no longer has
486 // any unfulfilled dependencies then that frame is continuous as well.
487 for (size_t d = 0; d < frame->second.dependent_frames.size(); ++d) {
488 auto frame_ref = frames_.find(frame->second.dependent_frames[d]);
489 RTC_DCHECK(frame_ref != frames_.end());
490
491 // TODO(philipel): Look into why we've seen this happen.
492 if (frame_ref != frames_.end()) {
493 --frame_ref->second.num_missing_continuous;
494 if (frame_ref->second.num_missing_continuous == 0) {
495 frame_ref->second.continuous = true;
496 continuous_frames.push(frame_ref);
497 }
498 }
499 }
500 }
501 }
502
PropagateDecodability(const FrameInfo & info)503 void FrameBuffer::PropagateDecodability(const FrameInfo& info) {
504 TRACE_EVENT0("webrtc", "FrameBuffer::PropagateDecodability");
505 for (size_t d = 0; d < info.dependent_frames.size(); ++d) {
506 auto ref_info = frames_.find(info.dependent_frames[d]);
507 RTC_DCHECK(ref_info != frames_.end());
508 // TODO(philipel): Look into why we've seen this happen.
509 if (ref_info != frames_.end()) {
510 RTC_DCHECK_GT(ref_info->second.num_missing_decodable, 0U);
511 --ref_info->second.num_missing_decodable;
512 }
513 }
514 }
515
UpdateFrameInfoWithIncomingFrame(const EncodedFrame & frame,FrameMap::iterator info)516 bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
517 FrameMap::iterator info) {
518 TRACE_EVENT0("webrtc", "FrameBuffer::UpdateFrameInfoWithIncomingFrame");
519 auto last_decoded_frame = decoded_frames_history_.GetLastDecodedFrameId();
520 RTC_DCHECK(!last_decoded_frame || *last_decoded_frame < info->first);
521
522 // In this function we determine how many missing dependencies this `frame`
523 // has to become continuous/decodable. If a frame that this `frame` depend
524 // on has already been decoded then we can ignore that dependency since it has
525 // already been fulfilled.
526 //
527 // For all other frames we will register a backwards reference to this `frame`
528 // so that `num_missing_continuous` and `num_missing_decodable` can be
529 // decremented as frames become continuous/are decoded.
530 struct Dependency {
531 int64_t frame_id;
532 bool continuous;
533 };
534 std::vector<Dependency> not_yet_fulfilled_dependencies;
535
536 // Find all dependencies that have not yet been fulfilled.
537 for (size_t i = 0; i < frame.num_references; ++i) {
538 // Does `frame` depend on a frame earlier than the last decoded one?
539 if (last_decoded_frame && frame.references[i] <= *last_decoded_frame) {
540 // Was that frame decoded? If not, this `frame` will never become
541 // decodable.
542 if (!decoded_frames_history_.WasDecoded(frame.references[i])) {
543 int64_t now_ms = clock_->TimeInMilliseconds();
544 if (last_log_non_decoded_ms_ + kLogNonDecodedIntervalMs < now_ms) {
545 RTC_LOG(LS_WARNING)
546 << "Frame " << frame.Id()
547 << " depends on a non-decoded frame more previous than the last "
548 "decoded frame, dropping frame.";
549 last_log_non_decoded_ms_ = now_ms;
550 }
551 return false;
552 }
553 } else {
554 auto ref_info = frames_.find(frame.references[i]);
555 bool ref_continuous =
556 ref_info != frames_.end() && ref_info->second.continuous;
557 not_yet_fulfilled_dependencies.push_back(
558 {frame.references[i], ref_continuous});
559 }
560 }
561
562 info->second.num_missing_continuous = not_yet_fulfilled_dependencies.size();
563 info->second.num_missing_decodable = not_yet_fulfilled_dependencies.size();
564
565 for (const Dependency& dep : not_yet_fulfilled_dependencies) {
566 if (dep.continuous)
567 --info->second.num_missing_continuous;
568
569 frames_[dep.frame_id].dependent_frames.push_back(frame.Id());
570 }
571
572 return true;
573 }
574
ClearFramesAndHistory()575 void FrameBuffer::ClearFramesAndHistory() {
576 TRACE_EVENT0("webrtc", "FrameBuffer::ClearFramesAndHistory");
577 frames_.clear();
578 last_continuous_frame_.reset();
579 frames_to_decode_.clear();
580 decoded_frames_history_.Clear();
581 }
582
583 // TODO(philipel): Avoid the concatenation of frames here, by replacing
584 // NextFrame and GetNextFrame with methods returning multiple frames.
CombineAndDeleteFrames(std::vector<std::unique_ptr<EncodedFrame>> frames) const585 std::unique_ptr<EncodedFrame> FrameBuffer::CombineAndDeleteFrames(
586 std::vector<std::unique_ptr<EncodedFrame>> frames) const {
587 RTC_DCHECK(!frames.empty());
588 absl::InlinedVector<std::unique_ptr<EncodedFrame>, 4> inlined;
589 for (auto& frame : frames) {
590 inlined.push_back(std::move(frame));
591 }
592 return webrtc::CombineAndDeleteFrames(std::move(inlined));
593 }
594
595 FrameBuffer::FrameInfo::FrameInfo() = default;
596 FrameBuffer::FrameInfo::FrameInfo(FrameInfo&&) = default;
597 FrameBuffer::FrameInfo::~FrameInfo() = default;
598
599 } // namespace video_coding
600 } // namespace webrtc
601