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/*
* Copyright (c) 2022 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 "video/video_stream_buffer_controller.h"
#include <algorithm>
#include <memory>
#include <utility>
#include "absl/base/attributes.h"
#include "absl/functional/bind_front.h"
#include "absl/types/optional.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/encoded_frame.h"
#include "api/video/frame_buffer.h"
#include "api/video/video_content_type.h"
#include "modules/video_coding/frame_helpers.h"
#include "modules/video_coding/timing/inter_frame_delay_variation_calculator.h"
#include "modules/video_coding/timing/jitter_estimator.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/thread_annotations.h"
#include "video/frame_decode_scheduler.h"
#include "video/frame_decode_timing.h"
#include "video/task_queue_frame_decode_scheduler.h"
#include "video/video_receive_stream_timeout_tracker.h"
namespace webrtc {
namespace {
// Max number of frames the buffer will hold.
static constexpr size_t kMaxFramesBuffered = 800;
// Max number of decoded frame info that will be saved.
static constexpr int kMaxFramesHistory = 1 << 13;
// Default value for the maximum decode queue size that is used when the
// low-latency renderer is used.
static constexpr size_t kZeroPlayoutDelayDefaultMaxDecodeQueueSize = 8;
struct FrameMetadata {
explicit FrameMetadata(const EncodedFrame& frame)
: is_last_spatial_layer(frame.is_last_spatial_layer),
is_keyframe(frame.is_keyframe()),
size(frame.size()),
contentType(frame.contentType()),
delayed_by_retransmission(frame.delayed_by_retransmission()),
rtp_timestamp(frame.RtpTimestamp()),
receive_time(frame.ReceivedTimestamp()) {}
const bool is_last_spatial_layer;
const bool is_keyframe;
const size_t size;
const VideoContentType contentType;
const bool delayed_by_retransmission;
const uint32_t rtp_timestamp;
const absl::optional<Timestamp> receive_time;
};
Timestamp MinReceiveTime(const EncodedFrame& frame) {
Timestamp first_recv_time = Timestamp::PlusInfinity();
for (const auto& packet_info : frame.PacketInfos()) {
if (packet_info.receive_time().IsFinite()) {
first_recv_time = std::min(first_recv_time, packet_info.receive_time());
}
}
return first_recv_time;
}
Timestamp ReceiveTime(const EncodedFrame& frame) {
absl::optional<Timestamp> ts = frame.ReceivedTimestamp();
RTC_DCHECK(ts.has_value()) << "Received frame must have a timestamp set!";
return *ts;
}
} // namespace
VideoStreamBufferController::VideoStreamBufferController(
Clock* clock,
TaskQueueBase* worker_queue,
VCMTiming* timing,
VideoStreamBufferControllerStatsObserver* stats_proxy,
FrameSchedulingReceiver* receiver,
TimeDelta max_wait_for_keyframe,
TimeDelta max_wait_for_frame,
std::unique_ptr<FrameDecodeScheduler> frame_decode_scheduler,
const FieldTrialsView& field_trials)
: field_trials_(field_trials),
clock_(clock),
stats_proxy_(stats_proxy),
receiver_(receiver),
timing_(timing),
frame_decode_scheduler_(std::move(frame_decode_scheduler)),
jitter_estimator_(clock_, field_trials),
buffer_(std::make_unique<FrameBuffer>(kMaxFramesBuffered,
kMaxFramesHistory,
field_trials)),
decode_timing_(clock_, timing_),
timeout_tracker_(
clock_,
worker_queue,
VideoReceiveStreamTimeoutTracker::Timeouts{
.max_wait_for_keyframe = max_wait_for_keyframe,
.max_wait_for_frame = max_wait_for_frame},
absl::bind_front(&VideoStreamBufferController::OnTimeout, this)),
zero_playout_delay_max_decode_queue_size_(
"max_decode_queue_size",
kZeroPlayoutDelayDefaultMaxDecodeQueueSize) {
RTC_DCHECK(stats_proxy_);
RTC_DCHECK(receiver_);
RTC_DCHECK(timing_);
RTC_DCHECK(clock_);
RTC_DCHECK(frame_decode_scheduler_);
ParseFieldTrial({&zero_playout_delay_max_decode_queue_size_},
field_trials.Lookup("WebRTC-ZeroPlayoutDelay"));
}
void VideoStreamBufferController::Stop() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
frame_decode_scheduler_->Stop();
timeout_tracker_.Stop();
decoder_ready_for_new_frame_ = false;
}
void VideoStreamBufferController::SetProtectionMode(
VCMVideoProtection protection_mode) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
protection_mode_ = protection_mode;
}
void VideoStreamBufferController::Clear() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
stats_proxy_->OnDroppedFrames(buffer_->CurrentSize());
buffer_ = std::make_unique<FrameBuffer>(kMaxFramesBuffered, kMaxFramesHistory,
field_trials_);
frame_decode_scheduler_->CancelOutstanding();
}
absl::optional<int64_t> VideoStreamBufferController::InsertFrame(
std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
FrameMetadata metadata(*frame);
int complete_units = buffer_->GetTotalNumberOfContinuousTemporalUnits();
if (buffer_->InsertFrame(std::move(frame))) {
RTC_DCHECK(metadata.receive_time) << "Frame receive time must be set!";
if (!metadata.delayed_by_retransmission && metadata.receive_time &&
(field_trials_.IsDisabled("WebRTC-IncomingTimestampOnMarkerBitOnly") ||
metadata.is_last_spatial_layer)) {
timing_->IncomingTimestamp(metadata.rtp_timestamp,
*metadata.receive_time);
}
if (complete_units < buffer_->GetTotalNumberOfContinuousTemporalUnits()) {
stats_proxy_->OnCompleteFrame(metadata.is_keyframe, metadata.size,
metadata.contentType);
MaybeScheduleFrameForRelease();
}
}
return buffer_->LastContinuousFrameId();
}
void VideoStreamBufferController::UpdateRtt(int64_t max_rtt_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
jitter_estimator_.UpdateRtt(TimeDelta::Millis(max_rtt_ms));
}
void VideoStreamBufferController::SetMaxWaits(TimeDelta max_wait_for_keyframe,
TimeDelta max_wait_for_frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
timeout_tracker_.SetTimeouts({.max_wait_for_keyframe = max_wait_for_keyframe,
.max_wait_for_frame = max_wait_for_frame});
}
void VideoStreamBufferController::StartNextDecode(bool keyframe_required) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
if (!timeout_tracker_.Running())
timeout_tracker_.Start(keyframe_required);
keyframe_required_ = keyframe_required;
if (keyframe_required_) {
timeout_tracker_.SetWaitingForKeyframe();
}
decoder_ready_for_new_frame_ = true;
MaybeScheduleFrameForRelease();
}
int VideoStreamBufferController::Size() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
return buffer_->CurrentSize();
}
void VideoStreamBufferController::OnFrameReady(
absl::InlinedVector<std::unique_ptr<EncodedFrame>, 4> frames,
Timestamp render_time) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
RTC_CHECK(!frames.empty())
<< "Callers must ensure there is at least one frame to decode.";
timeout_tracker_.OnEncodedFrameReleased();
Timestamp now = clock_->CurrentTime();
bool superframe_delayed_by_retransmission = false;
DataSize superframe_size = DataSize::Zero();
const EncodedFrame& first_frame = *frames.front();
Timestamp min_receive_time = MinReceiveTime(first_frame);
Timestamp max_receive_time = ReceiveTime(first_frame);
if (first_frame.is_keyframe())
keyframe_required_ = false;
// Gracefully handle bad RTP timestamps and render time issues.
if (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.RtpTimestamp();
jitter_estimator_.Reset();
timing_->Reset();
render_time = timing_->RenderTime(first_frame.RtpTimestamp(), now);
}
for (std::unique_ptr<EncodedFrame>& frame : frames) {
frame->SetRenderTime(render_time.ms());
superframe_delayed_by_retransmission |= frame->delayed_by_retransmission();
min_receive_time = std::min(min_receive_time, MinReceiveTime(*frame));
max_receive_time = std::max(max_receive_time, ReceiveTime(*frame));
superframe_size += DataSize::Bytes(frame->size());
}
if (!superframe_delayed_by_retransmission) {
absl::optional<TimeDelta> inter_frame_delay_variation =
ifdv_calculator_.Calculate(first_frame.RtpTimestamp(),
max_receive_time);
if (inter_frame_delay_variation) {
jitter_estimator_.UpdateEstimate(*inter_frame_delay_variation,
superframe_size);
}
static constexpr float kRttMult = 0.9f;
static constexpr TimeDelta kRttMultAddCap = TimeDelta::Millis(200);
timing_->SetJitterDelay(
jitter_estimator_.GetJitterEstimate(kRttMult, kRttMultAddCap));
timing_->UpdateCurrentDelay(render_time, now);
} else {
jitter_estimator_.FrameNacked();
}
// Update stats.
UpdateDroppedFrames();
UpdateFrameBufferTimings(min_receive_time, now);
UpdateTimingFrameInfo();
std::unique_ptr<EncodedFrame> frame =
CombineAndDeleteFrames(std::move(frames));
timing_->SetLastDecodeScheduledTimestamp(now);
decoder_ready_for_new_frame_ = false;
receiver_->OnEncodedFrame(std::move(frame));
}
void VideoStreamBufferController::OnTimeout(TimeDelta delay) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// Stop sending timeouts until receiver starts waiting for a new frame.
timeout_tracker_.Stop();
// If the stream is paused then ignore the timeout.
if (!decoder_ready_for_new_frame_) {
return;
}
decoder_ready_for_new_frame_ = false;
receiver_->OnDecodableFrameTimeout(delay);
}
void VideoStreamBufferController::FrameReadyForDecode(uint32_t rtp_timestamp,
Timestamp render_time) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// Check that the frame to decode is still valid before passing the frame for
// decoding.
auto decodable_tu_info = buffer_->DecodableTemporalUnitsInfo();
if (!decodable_tu_info) {
RTC_LOG(LS_ERROR)
<< "The frame buffer became undecodable during the wait "
"to decode frame with rtp-timestamp "
<< rtp_timestamp
<< ". Cancelling the decode of this frame, decoding "
"will resume when the frame buffers become decodable again.";
return;
}
RTC_DCHECK_EQ(rtp_timestamp, decodable_tu_info->next_rtp_timestamp)
<< "Frame buffer's next decodable frame was not the one sent for "
"extraction.";
auto frames = buffer_->ExtractNextDecodableTemporalUnit();
if (frames.empty()) {
RTC_LOG(LS_ERROR)
<< "The frame buffer should never return an empty temporal until list "
"when there is a decodable temporal unit.";
RTC_DCHECK_NOTREACHED();
return;
}
OnFrameReady(std::move(frames), render_time);
}
void VideoStreamBufferController::UpdateDroppedFrames()
RTC_RUN_ON(&worker_sequence_checker_) {
const int dropped_frames = buffer_->GetTotalNumberOfDroppedFrames() -
frames_dropped_before_last_new_frame_;
if (dropped_frames > 0)
stats_proxy_->OnDroppedFrames(dropped_frames);
frames_dropped_before_last_new_frame_ =
buffer_->GetTotalNumberOfDroppedFrames();
}
void VideoStreamBufferController::UpdateFrameBufferTimings(
Timestamp min_receive_time,
Timestamp now) {
// Update instantaneous delays.
auto timings = timing_->GetTimings();
if (timings.num_decoded_frames) {
stats_proxy_->OnFrameBufferTimingsUpdated(
timings.estimated_max_decode_time.ms(), timings.current_delay.ms(),
timings.target_delay.ms(), timings.minimum_delay.ms(),
timings.min_playout_delay.ms(), timings.render_delay.ms());
}
// The spec mandates that `jitterBufferDelay` is the "time the first
// packet is received by the jitter buffer (ingest timestamp) to the time it
// exits the jitter buffer (emit timestamp)". Since the "jitter buffer"
// is not a monolith in the webrtc.org implementation, we take the freedom to
// define "ingest timestamp" as "first packet received by
// RtpVideoStreamReceiver2" and "emit timestamp" as "decodable frame released
// by VideoStreamBufferController".
//
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
TimeDelta jitter_buffer_delay =
std::max(TimeDelta::Zero(), now - min_receive_time);
stats_proxy_->OnDecodableFrame(jitter_buffer_delay, timings.target_delay,
timings.minimum_delay);
}
void VideoStreamBufferController::UpdateTimingFrameInfo() {
absl::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
if (info)
stats_proxy_->OnTimingFrameInfoUpdated(*info);
}
bool VideoStreamBufferController::IsTooManyFramesQueued() const
RTC_RUN_ON(&worker_sequence_checker_) {
return buffer_->CurrentSize() > zero_playout_delay_max_decode_queue_size_;
}
void VideoStreamBufferController::ForceKeyFrameReleaseImmediately()
RTC_RUN_ON(&worker_sequence_checker_) {
RTC_DCHECK(keyframe_required_);
// Iterate through the frame buffer until there is a complete keyframe and
// release this right away.
while (buffer_->DecodableTemporalUnitsInfo()) {
auto next_frame = buffer_->ExtractNextDecodableTemporalUnit();
if (next_frame.empty()) {
RTC_DCHECK_NOTREACHED()
<< "Frame buffer should always return at least 1 frame.";
continue;
}
// Found keyframe - decode right away.
if (next_frame.front()->is_keyframe()) {
auto render_time = timing_->RenderTime(next_frame.front()->RtpTimestamp(),
clock_->CurrentTime());
OnFrameReady(std::move(next_frame), render_time);
return;
}
}
}
void VideoStreamBufferController::MaybeScheduleFrameForRelease()
RTC_RUN_ON(&worker_sequence_checker_) {
auto decodable_tu_info = buffer_->DecodableTemporalUnitsInfo();
if (!decoder_ready_for_new_frame_ || !decodable_tu_info) {
return;
}
if (keyframe_required_) {
return ForceKeyFrameReleaseImmediately();
}
// If already scheduled then abort.
if (frame_decode_scheduler_->ScheduledRtpTimestamp() ==
decodable_tu_info->next_rtp_timestamp) {
return;
}
TimeDelta max_wait = timeout_tracker_.TimeUntilTimeout();
// Ensures the frame is scheduled for decode before the stream times out.
// This is otherwise a race condition.
max_wait = std::max(max_wait - TimeDelta::Millis(1), TimeDelta::Zero());
absl::optional<FrameDecodeTiming::FrameSchedule> schedule;
while (decodable_tu_info) {
schedule = decode_timing_.OnFrameBufferUpdated(
decodable_tu_info->next_rtp_timestamp,
decodable_tu_info->last_rtp_timestamp, max_wait,
IsTooManyFramesQueued());
if (schedule) {
// Don't schedule if already waiting for the same frame.
if (frame_decode_scheduler_->ScheduledRtpTimestamp() !=
decodable_tu_info->next_rtp_timestamp) {
frame_decode_scheduler_->CancelOutstanding();
frame_decode_scheduler_->ScheduleFrame(
decodable_tu_info->next_rtp_timestamp, *schedule,
absl::bind_front(&VideoStreamBufferController::FrameReadyForDecode,
this));
}
return;
}
// If no schedule for current rtp, drop and try again.
buffer_->DropNextDecodableTemporalUnit();
decodable_tu_info = buffer_->DecodableTemporalUnitsInfo();
}
}
} // namespace webrtc