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webrtc_m130/video/video_receive_stream2.cc
Evan Shrubsole 3fa9a66f22 Cap max decode delay for FrameBuffer3 
When a large queue of frames builds up due to a lost frame, the decode
delay can sometimes become quite large. In this case the stream may
signal as timed out when in fact it is not. Instead, the delay should
be capped at the timeout limit.

Bug: webrtc:14168
Change-Id: I5b4e8851b2c6d7d27a698627dc1633931d7fc00e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265404
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37199}
2022-06-13 14:52:46 +00:00

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/*
* Copyright (c) 2020 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_receive_stream2.h"
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include <memory>
#include <set>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/container/inlined_vector.h"
#include "absl/functional/bind_front.h"
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/frequency.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/encoded_image.h"
#include "api/video/frame_buffer.h"
#include "api/video_codecs/h264_profile_level_id.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/rtp_stream_receiver_controller_interface.h"
#include "call/rtx_receive_stream.h"
#include "common_video/include/incoming_video_stream.h"
#include "modules/video_coding/frame_buffer2.h"
#include "modules/video_coding/frame_helpers.h"
#include "modules/video_coding/include/video_codec_interface.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "modules/video_coding/include/video_error_codes.h"
#include "modules/video_coding/timing/timing.h"
#include "modules/video_coding/utility/vp8_header_parser.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/rtt_mult_experiment.h"
#include "rtc_base/location.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/pending_task_safety_flag.h"
#include "rtc_base/task_utils/to_queued_task.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "video/call_stats2.h"
#include "video/frame_decode_scheduler.h"
#include "video/frame_dumping_decoder.h"
#include "video/receive_statistics_proxy2.h"
#include "video/video_receive_stream_timeout_tracker.h"
namespace webrtc {
namespace internal {
namespace {
// The default delay before re-requesting a key frame to be sent.
constexpr TimeDelta kMinBaseMinimumDelay = TimeDelta::Zero();
constexpr TimeDelta kMaxBaseMinimumDelay = TimeDelta::Seconds(10);
// Create a decoder for the preferred codec before the stream starts and any
// other decoder lazily on demand.
constexpr int kDefaultMaximumPreStreamDecoders = 1;
// Concrete instance of RecordableEncodedFrame wrapping needed content
// from EncodedFrame.
class WebRtcRecordableEncodedFrame : public RecordableEncodedFrame {
public:
explicit WebRtcRecordableEncodedFrame(
const EncodedFrame& frame,
RecordableEncodedFrame::EncodedResolution resolution)
: buffer_(frame.GetEncodedData()),
render_time_ms_(frame.RenderTime()),
codec_(frame.CodecSpecific()->codecType),
is_key_frame_(frame.FrameType() == VideoFrameType::kVideoFrameKey),
resolution_(resolution) {
if (frame.ColorSpace()) {
color_space_ = *frame.ColorSpace();
}
}
// VideoEncodedSinkInterface::FrameBuffer
rtc::scoped_refptr<const EncodedImageBufferInterface> encoded_buffer()
const override {
return buffer_;
}
absl::optional<webrtc::ColorSpace> color_space() const override {
return color_space_;
}
VideoCodecType codec() const override { return codec_; }
bool is_key_frame() const override { return is_key_frame_; }
EncodedResolution resolution() const override { return resolution_; }
Timestamp render_time() const override {
return Timestamp::Millis(render_time_ms_);
}
private:
rtc::scoped_refptr<EncodedImageBufferInterface> buffer_;
int64_t render_time_ms_;
VideoCodecType codec_;
bool is_key_frame_;
EncodedResolution resolution_;
absl::optional<webrtc::ColorSpace> color_space_;
};
RenderResolution InitialDecoderResolution(const FieldTrialsView& field_trials) {
FieldTrialOptional<int> width("w");
FieldTrialOptional<int> height("h");
ParseFieldTrial({&width, &height},
field_trials.Lookup("WebRTC-Video-InitialDecoderResolution"));
if (width && height) {
return RenderResolution(width.Value(), height.Value());
}
return RenderResolution(320, 180);
}
// Video decoder class to be used for unknown codecs. Doesn't support decoding
// but logs messages to LS_ERROR.
class NullVideoDecoder : public webrtc::VideoDecoder {
public:
bool Configure(const Settings& settings) override {
RTC_LOG(LS_ERROR) << "Can't initialize NullVideoDecoder.";
return true;
}
int32_t Decode(const webrtc::EncodedImage& input_image,
bool missing_frames,
int64_t render_time_ms) override {
RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding.";
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t RegisterDecodeCompleteCallback(
webrtc::DecodedImageCallback* callback) override {
RTC_LOG(LS_ERROR)
<< "Can't register decode complete callback on NullVideoDecoder.";
return WEBRTC_VIDEO_CODEC_OK;
}
int32_t Release() override { return WEBRTC_VIDEO_CODEC_OK; }
const char* ImplementationName() const override { return "NullVideoDecoder"; }
};
bool IsKeyFrameAndUnspecifiedResolution(const EncodedFrame& frame) {
return frame.FrameType() == VideoFrameType::kVideoFrameKey &&
frame.EncodedImage()._encodedWidth == 0 &&
frame.EncodedImage()._encodedHeight == 0;
}
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// Maximum time between frames before resetting the FrameBuffer to avoid RTP
// timestamps wraparound to affect FrameBuffer.
constexpr TimeDelta kInactiveStreamThreshold = TimeDelta::Minutes(10);
std::string OptionalDelayToLogString(const absl::optional<TimeDelta> opt) {
return opt.has_value() ? ToLogString(*opt) : "<unset>";
}
} // namespace
TimeDelta DetermineMaxWaitForFrame(
const VideoReceiveStreamInterface::Config& config,
bool is_keyframe) {
// A (arbitrary) conversion factor between the remotely signalled NACK buffer
// time (if not present defaults to 1000ms) and the maximum time we wait for a
// remote frame. Chosen to not change existing defaults when using not
// rtx-time.
const int conversion_factor = 3;
const TimeDelta rtp_history =
TimeDelta::Millis(config.rtp.nack.rtp_history_ms);
if (rtp_history > TimeDelta::Zero() &&
conversion_factor * rtp_history < kMaxWaitForFrame) {
return is_keyframe ? rtp_history : conversion_factor * rtp_history;
}
return is_keyframe ? kMaxWaitForKeyFrame : kMaxWaitForFrame;
}
VideoReceiveStream2::VideoReceiveStream2(
TaskQueueFactory* task_queue_factory,
Call* call,
int num_cpu_cores,
PacketRouter* packet_router,
VideoReceiveStreamInterface::Config config,
CallStats* call_stats,
Clock* clock,
std::unique_ptr<VCMTiming> timing,
NackPeriodicProcessor* nack_periodic_processor,
DecodeSynchronizer* decode_sync)
: task_queue_factory_(task_queue_factory),
transport_adapter_(config.rtcp_send_transport),
config_(std::move(config)),
num_cpu_cores_(num_cpu_cores),
call_(call),
clock_(clock),
call_stats_(call_stats),
source_tracker_(clock_),
stats_proxy_(remote_ssrc(),
clock_,
call->worker_thread(),
call->trials()),
rtp_receive_statistics_(ReceiveStatistics::Create(clock_)),
timing_(std::move(timing)),
video_receiver_(clock_, timing_.get(), call->trials()),
rtp_video_stream_receiver_(call->worker_thread(),
clock_,
&transport_adapter_,
call_stats->AsRtcpRttStats(),
packet_router,
&config_,
rtp_receive_statistics_.get(),
&stats_proxy_,
&stats_proxy_,
nack_periodic_processor,
this, // NackSender
nullptr, // Use default KeyFrameRequestSender
this, // OnCompleteFrameCallback
std::move(config_.frame_decryptor),
std::move(config_.frame_transformer),
call->trials()),
rtp_stream_sync_(call->worker_thread(), this),
max_wait_for_keyframe_(DetermineMaxWaitForFrame(config_, true)),
max_wait_for_frame_(DetermineMaxWaitForFrame(config_, false)),
maximum_pre_stream_decoders_("max", kDefaultMaximumPreStreamDecoders),
decode_sync_(decode_sync),
decode_queue_(task_queue_factory_->CreateTaskQueue(
"DecodingQueue",
TaskQueueFactory::Priority::HIGH)) {
RTC_LOG(LS_INFO) << "VideoReceiveStream2: " << config_.ToString();
RTC_DCHECK(call_->worker_thread());
RTC_DCHECK(config_.renderer);
RTC_DCHECK(call_stats_);
packet_sequence_checker_.Detach();
RTC_DCHECK(!config_.decoders.empty());
RTC_CHECK(config_.decoder_factory);
std::set<int> decoder_payload_types;
for (const Decoder& decoder : config_.decoders) {
RTC_CHECK(decoder_payload_types.find(decoder.payload_type) ==
decoder_payload_types.end())
<< "Duplicate payload type (" << decoder.payload_type
<< ") for different decoders.";
decoder_payload_types.insert(decoder.payload_type);
}
timing_->set_render_delay(TimeDelta::Millis(config_.render_delay_ms));
frame_buffer_ = FrameBufferProxy::CreateFromFieldTrial(
clock_, call_->worker_thread(), timing_.get(), &stats_proxy_,
&decode_queue_, this, max_wait_for_keyframe_, max_wait_for_frame_,
decode_sync_, call_->trials());
if (rtx_ssrc()) {
rtx_receive_stream_ = std::make_unique<RtxReceiveStream>(
&rtp_video_stream_receiver_, config_.rtp.rtx_associated_payload_types,
remote_ssrc(), rtp_receive_statistics_.get());
} else {
rtp_receive_statistics_->EnableRetransmitDetection(remote_ssrc(), true);
}
ParseFieldTrial(
{
&maximum_pre_stream_decoders_,
},
call_->trials().Lookup("WebRTC-PreStreamDecoders"));
}
VideoReceiveStream2::~VideoReceiveStream2() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
RTC_LOG(LS_INFO) << "~VideoReceiveStream2: " << config_.ToString();
RTC_DCHECK(!media_receiver_);
RTC_DCHECK(!rtx_receiver_);
Stop();
}
void VideoReceiveStream2::RegisterWithTransport(
RtpStreamReceiverControllerInterface* receiver_controller) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK(!media_receiver_);
RTC_DCHECK(!rtx_receiver_);
// Register with RtpStreamReceiverController.
media_receiver_ = receiver_controller->CreateReceiver(
remote_ssrc(), &rtp_video_stream_receiver_);
if (rtx_ssrc()) {
RTC_DCHECK(rtx_receive_stream_);
rtx_receiver_ = receiver_controller->CreateReceiver(
rtx_ssrc(), rtx_receive_stream_.get());
}
}
void VideoReceiveStream2::UnregisterFromTransport() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
media_receiver_.reset();
rtx_receiver_.reset();
}
const std::string& VideoReceiveStream2::sync_group() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return config_.sync_group;
}
void VideoReceiveStream2::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtp_video_stream_receiver_.SignalNetworkState(state);
}
bool VideoReceiveStream2::DeliverRtcp(const uint8_t* packet, size_t length) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return rtp_video_stream_receiver_.DeliverRtcp(packet, length);
}
void VideoReceiveStream2::SetSync(Syncable* audio_syncable) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_stream_sync_.ConfigureSync(audio_syncable);
}
void VideoReceiveStream2::SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (config_.rtp.local_ssrc == local_ssrc)
return;
// TODO(tommi): Make sure we don't rely on local_ssrc via the config struct.
const_cast<uint32_t&>(config_.rtp.local_ssrc) = local_ssrc;
rtp_video_stream_receiver_.OnLocalSsrcChange(local_ssrc);
}
void VideoReceiveStream2::Start() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
if (decoder_running_) {
return;
}
const bool protected_by_fec = config_.rtp.protected_by_flexfec ||
rtp_video_stream_receiver_.IsUlpfecEnabled();
if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
protected_by_fec) {
frame_buffer_->SetProtectionMode(kProtectionNackFEC);
}
transport_adapter_.Enable();
rtc::VideoSinkInterface<VideoFrame>* renderer = nullptr;
if (config_.enable_prerenderer_smoothing) {
incoming_video_stream_.reset(new IncomingVideoStream(
task_queue_factory_, config_.render_delay_ms, this));
renderer = incoming_video_stream_.get();
} else {
renderer = this;
}
int decoders_count = 0;
for (const Decoder& decoder : config_.decoders) {
// Create up to maximum_pre_stream_decoders_ up front, wait the the other
// decoders until they are requested (i.e., we receive the corresponding
// payload).
if (decoders_count < maximum_pre_stream_decoders_) {
CreateAndRegisterExternalDecoder(decoder);
++decoders_count;
}
VideoDecoder::Settings settings;
settings.set_codec_type(
PayloadStringToCodecType(decoder.video_format.name));
settings.set_max_render_resolution(
InitialDecoderResolution(call_->trials()));
settings.set_number_of_cores(num_cpu_cores_);
const bool raw_payload =
config_.rtp.raw_payload_types.count(decoder.payload_type) > 0;
{
// TODO(bugs.webrtc.org/11993): Make this call on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.AddReceiveCodec(
decoder.payload_type, settings.codec_type(),
decoder.video_format.parameters, raw_payload);
}
video_receiver_.RegisterReceiveCodec(decoder.payload_type, settings);
}
RTC_DCHECK(renderer != nullptr);
video_stream_decoder_.reset(
new VideoStreamDecoder(&video_receiver_, &stats_proxy_, renderer));
// Make sure we register as a stats observer *after* we've prepared the
// `video_stream_decoder_`.
call_stats_->RegisterStatsObserver(this);
// Start decoding on task queue.
video_receiver_.DecoderThreadStarting();
stats_proxy_.DecoderThreadStarting();
decode_queue_.PostTask([this] {
RTC_DCHECK_RUN_ON(&decode_queue_);
decoder_stopped_ = false;
});
frame_buffer_->StartNextDecode(true);
decoder_running_ = true;
{
// TODO(bugs.webrtc.org/11993): Make this call on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.StartReceive();
}
}
void VideoReceiveStream2::Stop() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
{
// TODO(bugs.webrtc.org/11993): Make this call on the network thread.
// Also call `GetUniqueFramesSeen()` at the same time (since it's a counter
// that's updated on the network thread).
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.StopReceive();
}
stats_proxy_.OnUniqueFramesCounted(
rtp_video_stream_receiver_.GetUniqueFramesSeen());
frame_buffer_->StopOnWorker();
call_stats_->DeregisterStatsObserver(this);
if (decoder_running_) {
rtc::Event done;
decode_queue_.PostTask([this, &done] {
RTC_DCHECK_RUN_ON(&decode_queue_);
decoder_stopped_ = true;
done.Set();
});
done.Wait(rtc::Event::kForever);
decoder_running_ = false;
video_receiver_.DecoderThreadStopped();
stats_proxy_.DecoderThreadStopped();
// Deregister external decoders so they are no longer running during
// destruction. This effectively stops the VCM since the decoder thread is
// stopped, the VCM is deregistered and no asynchronous decoder threads are
// running.
for (const Decoder& decoder : config_.decoders)
video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
UpdateHistograms();
}
video_stream_decoder_.reset();
incoming_video_stream_.reset();
transport_adapter_.Disable();
}
void VideoReceiveStream2::SetRtpExtensions(
std::vector<RtpExtension> extensions) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.SetRtpExtensions(extensions);
// TODO(tommi): We don't use the `c.rtp.extensions` member in the
// VideoReceiveStream2 class, so this const_cast<> is a temporary hack to keep
// things consistent between VideoReceiveStream2 and RtpVideoStreamReceiver2
// for debugging purposes. The `packet_sequence_checker_` gives us assurances
// that from a threading perspective, this is still safe. The accessors that
// give read access to this state, run behind the same check.
// The alternative to the const_cast<> would be to make `config_` non-const
// and guarded by `packet_sequence_checker_`. However the scope of that state
// is huge (the whole Config struct), and would require all methods that touch
// the struct to abide the needs of the `extensions` member.
const_cast<std::vector<RtpExtension>&>(config_.rtp.extensions) =
std::move(extensions);
}
RtpHeaderExtensionMap VideoReceiveStream2::GetRtpExtensionMap() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return rtp_video_stream_receiver_.GetRtpExtensions();
}
bool VideoReceiveStream2::transport_cc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
return config_.rtp.transport_cc;
}
void VideoReceiveStream2::SetTransportCc(bool transport_cc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<bool&>(config_.rtp.transport_cc) = transport_cc;
}
void VideoReceiveStream2::CreateAndRegisterExternalDecoder(
const Decoder& decoder) {
TRACE_EVENT0("webrtc",
"VideoReceiveStream2::CreateAndRegisterExternalDecoder");
std::unique_ptr<VideoDecoder> video_decoder =
config_.decoder_factory->CreateVideoDecoder(decoder.video_format);
// If we still have no valid decoder, we have to create a "Null" decoder
// that ignores all calls. The reason we can get into this state is that the
// old decoder factory interface doesn't have a way to query supported
// codecs.
if (!video_decoder) {
video_decoder = std::make_unique<NullVideoDecoder>();
}
std::string decoded_output_file =
call_->trials().Lookup("WebRTC-DecoderDataDumpDirectory");
// Because '/' can't be used inside a field trial parameter, we use ';'
// instead.
// This is only relevant to WebRTC-DecoderDataDumpDirectory
// field trial. ';' is chosen arbitrary. Even though it's a legal character
// in some file systems, we can sacrifice ability to use it in the path to
// dumped video, since it's developers-only feature for debugging.
absl::c_replace(decoded_output_file, ';', '/');
if (!decoded_output_file.empty()) {
char filename_buffer[256];
rtc::SimpleStringBuilder ssb(filename_buffer);
ssb << decoded_output_file << "/webrtc_receive_stream_" << remote_ssrc()
<< "-" << rtc::TimeMicros() << ".ivf";
video_decoder = CreateFrameDumpingDecoderWrapper(
std::move(video_decoder), FileWrapper::OpenWriteOnly(ssb.str()));
}
video_decoders_.push_back(std::move(video_decoder));
video_receiver_.RegisterExternalDecoder(video_decoders_.back().get(),
decoder.payload_type);
}
VideoReceiveStreamInterface::Stats VideoReceiveStream2::GetStats() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
VideoReceiveStream2::Stats stats = stats_proxy_.GetStats();
stats.total_bitrate_bps = 0;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(stats.ssrc);
if (statistician) {
stats.rtp_stats = statistician->GetStats();
stats.total_bitrate_bps = statistician->BitrateReceived();
}
if (rtx_ssrc()) {
StreamStatistician* rtx_statistician =
rtp_receive_statistics_->GetStatistician(rtx_ssrc());
if (rtx_statistician)
stats.total_bitrate_bps += rtx_statistician->BitrateReceived();
}
return stats;
}
void VideoReceiveStream2::UpdateHistograms() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
absl::optional<int> fraction_lost;
StreamDataCounters rtp_stats;
StreamStatistician* statistician =
rtp_receive_statistics_->GetStatistician(remote_ssrc());
if (statistician) {
fraction_lost = statistician->GetFractionLostInPercent();
rtp_stats = statistician->GetReceiveStreamDataCounters();
}
if (rtx_ssrc()) {
StreamStatistician* rtx_statistician =
rtp_receive_statistics_->GetStatistician(rtx_ssrc());
if (rtx_statistician) {
StreamDataCounters rtx_stats =
rtx_statistician->GetReceiveStreamDataCounters();
stats_proxy_.UpdateHistograms(fraction_lost, rtp_stats, &rtx_stats);
return;
}
}
stats_proxy_.UpdateHistograms(fraction_lost, rtp_stats, nullptr);
}
bool VideoReceiveStream2::SetBaseMinimumPlayoutDelayMs(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
TimeDelta delay = TimeDelta::Millis(delay_ms);
if (delay < kMinBaseMinimumDelay || delay > kMaxBaseMinimumDelay) {
return false;
}
base_minimum_playout_delay_ = delay;
UpdatePlayoutDelays();
return true;
}
int VideoReceiveStream2::GetBaseMinimumPlayoutDelayMs() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
constexpr TimeDelta kDefaultBaseMinPlayoutDelay = TimeDelta::Millis(-1);
// Unset must be -1.
static_assert(-1 == kDefaultBaseMinPlayoutDelay.ms(), "");
return base_minimum_playout_delay_.value_or(kDefaultBaseMinPlayoutDelay).ms();
}
void VideoReceiveStream2::OnFrame(const VideoFrame& video_frame) {
VideoFrameMetaData frame_meta(video_frame, clock_->CurrentTime());
// TODO(bugs.webrtc.org/10739): we should set local capture clock offset for
// `video_frame.packet_infos`. But VideoFrame is const qualified here.
call_->worker_thread()->PostTask(
ToQueuedTask(task_safety_, [frame_meta, this]() {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
int64_t video_playout_ntp_ms;
int64_t sync_offset_ms;
double estimated_freq_khz;
if (rtp_stream_sync_.GetStreamSyncOffsetInMs(
frame_meta.rtp_timestamp, frame_meta.render_time_ms(),
&video_playout_ntp_ms, &sync_offset_ms, &estimated_freq_khz)) {
stats_proxy_.OnSyncOffsetUpdated(video_playout_ntp_ms, sync_offset_ms,
estimated_freq_khz);
}
stats_proxy_.OnRenderedFrame(frame_meta);
}));
source_tracker_.OnFrameDelivered(video_frame.packet_infos());
config_.renderer->OnFrame(video_frame);
webrtc::MutexLock lock(&pending_resolution_mutex_);
if (pending_resolution_.has_value()) {
if (!pending_resolution_->empty() &&
(video_frame.width() != static_cast<int>(pending_resolution_->width) ||
video_frame.height() !=
static_cast<int>(pending_resolution_->height))) {
RTC_LOG(LS_WARNING)
<< "Recordable encoded frame stream resolution was reported as "
<< pending_resolution_->width << "x" << pending_resolution_->height
<< " but the stream is now " << video_frame.width()
<< video_frame.height();
}
pending_resolution_ = RecordableEncodedFrame::EncodedResolution{
static_cast<unsigned>(video_frame.width()),
static_cast<unsigned>(video_frame.height())};
}
}
void VideoReceiveStream2::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
rtp_video_stream_receiver_.SetFrameDecryptor(std::move(frame_decryptor));
}
void VideoReceiveStream2::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
rtp_video_stream_receiver_.SetDepacketizerToDecoderFrameTransformer(
std::move(frame_transformer));
}
void VideoReceiveStream2::SendNack(
const std::vector<uint16_t>& sequence_numbers,
bool buffering_allowed) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
RTC_DCHECK(buffering_allowed);
rtp_video_stream_receiver_.RequestPacketRetransmit(sequence_numbers);
}
void VideoReceiveStream2::RequestKeyFrame(Timestamp now) {
// Running on worker_sequence_checker_.
// Called from RtpVideoStreamReceiver (rtp_video_stream_receiver_ is
// ultimately responsible).
rtp_video_stream_receiver_.RequestKeyFrame();
decode_queue_.PostTask([this, now]() {
RTC_DCHECK_RUN_ON(&decode_queue_);
last_keyframe_request_ = now;
});
}
void VideoReceiveStream2::OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// TODO(https://bugs.webrtc.org/13343): Remove this check when FrameBuffer3 is
// deployed. With FrameBuffer3, this case is properly handled and tested in
// the FrameBufferProxyTest.PausedStream unit test.
Timestamp time_now = clock_->CurrentTime();
if (last_complete_frame_time_ &&
time_now - *last_complete_frame_time_ > kInactiveStreamThreshold) {
frame_buffer_->Clear();
}
last_complete_frame_time_ = time_now;
const VideoPlayoutDelay& playout_delay = frame->EncodedImage().playout_delay_;
if (playout_delay.min_ms >= 0) {
frame_minimum_playout_delay_ = TimeDelta::Millis(playout_delay.min_ms);
UpdatePlayoutDelays();
}
if (playout_delay.max_ms >= 0) {
frame_maximum_playout_delay_ = TimeDelta::Millis(playout_delay.max_ms);
UpdatePlayoutDelays();
}
auto last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
if (last_continuous_pid.has_value()) {
{
// TODO(bugs.webrtc.org/11993): Call on the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.FrameContinuous(*last_continuous_pid);
}
}
}
void VideoReceiveStream2::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
// TODO(bugs.webrtc.org/13757): Replace with TimeDelta.
frame_buffer_->UpdateRtt(max_rtt_ms);
rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms);
stats_proxy_.OnRttUpdate(avg_rtt_ms);
}
uint32_t VideoReceiveStream2::id() const {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
return remote_ssrc();
}
absl::optional<Syncable::Info> VideoReceiveStream2::GetInfo() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
absl::optional<Syncable::Info> info =
rtp_video_stream_receiver_.GetSyncInfo();
if (!info)
return absl::nullopt;
info->current_delay_ms = timing_->TargetVideoDelay().ms();
return info;
}
bool VideoReceiveStream2::GetPlayoutRtpTimestamp(uint32_t* rtp_timestamp,
int64_t* time_ms) const {
RTC_DCHECK_NOTREACHED();
return false;
}
void VideoReceiveStream2::SetEstimatedPlayoutNtpTimestampMs(
int64_t ntp_timestamp_ms,
int64_t time_ms) {
RTC_DCHECK_NOTREACHED();
}
bool VideoReceiveStream2::SetMinimumPlayoutDelay(int delay_ms) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
syncable_minimum_playout_delay_ = TimeDelta::Millis(delay_ms);
UpdatePlayoutDelays();
return true;
}
TimeDelta VideoReceiveStream2::GetMaxWait() const {
return keyframe_required_ ? max_wait_for_keyframe_ : max_wait_for_frame_;
}
void VideoReceiveStream2::OnEncodedFrame(std::unique_ptr<EncodedFrame> frame) {
if (!decode_queue_.IsCurrent()) {
decode_queue_.PostTask([this, frame = std::move(frame)]() mutable {
OnEncodedFrame(std::move(frame));
});
return;
}
RTC_DCHECK_RUN_ON(&decode_queue_);
if (decoder_stopped_)
return;
HandleEncodedFrame(std::move(frame));
frame_buffer_->StartNextDecode(keyframe_required_);
}
void VideoReceiveStream2::OnDecodableFrameTimeout(TimeDelta wait_time) {
if (!call_->worker_thread()->IsCurrent()) {
call_->worker_thread()->PostTask(ToQueuedTask(
task_safety_,
[this, wait_time] { OnDecodableFrameTimeout(wait_time); }));
return;
}
// TODO(bugs.webrtc.org/11993): PostTask to the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
HandleFrameBufferTimeout(clock_->CurrentTime(), wait_time);
decode_queue_.PostTask([this] {
RTC_DCHECK_RUN_ON(&decode_queue_);
frame_buffer_->StartNextDecode(keyframe_required_);
});
}
void VideoReceiveStream2::HandleEncodedFrame(
std::unique_ptr<EncodedFrame> frame) {
// Running on `decode_queue_`.
Timestamp now = clock_->CurrentTime();
// Current OnPreDecode only cares about QP for VP8.
int qp = -1;
if (frame->CodecSpecific()->codecType == kVideoCodecVP8) {
if (!vp8::GetQp(frame->data(), frame->size(), &qp)) {
RTC_LOG(LS_WARNING) << "Failed to extract QP from VP8 video frame";
}
}
stats_proxy_.OnPreDecode(frame->CodecSpecific()->codecType, qp);
bool force_request_key_frame = false;
int64_t decoded_frame_picture_id = -1;
const bool keyframe_request_is_due =
!last_keyframe_request_ ||
now >= (*last_keyframe_request_ + max_wait_for_keyframe_);
if (!video_receiver_.IsExternalDecoderRegistered(frame->PayloadType())) {
// Look for the decoder with this payload type.
for (const Decoder& decoder : config_.decoders) {
if (decoder.payload_type == frame->PayloadType()) {
CreateAndRegisterExternalDecoder(decoder);
break;
}
}
}
int64_t frame_id = frame->Id();
bool received_frame_is_keyframe =
frame->FrameType() == VideoFrameType::kVideoFrameKey;
int decode_result = DecodeAndMaybeDispatchEncodedFrame(std::move(frame));
if (decode_result == WEBRTC_VIDEO_CODEC_OK ||
decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) {
keyframe_required_ = false;
frame_decoded_ = true;
decoded_frame_picture_id = frame_id;
if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)
force_request_key_frame = true;
} else if (!frame_decoded_ || !keyframe_required_ ||
keyframe_request_is_due) {
keyframe_required_ = true;
// TODO(philipel): Remove this keyframe request when downstream project
// has been fixed.
force_request_key_frame = true;
}
{
// TODO(bugs.webrtc.org/11993): Make this PostTask to the network thread.
call_->worker_thread()->PostTask(ToQueuedTask(
task_safety_,
[this, now, received_frame_is_keyframe, force_request_key_frame,
decoded_frame_picture_id, keyframe_request_is_due]() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
if (decoded_frame_picture_id != -1)
rtp_video_stream_receiver_.FrameDecoded(decoded_frame_picture_id);
HandleKeyFrameGeneration(received_frame_is_keyframe, now,
force_request_key_frame,
keyframe_request_is_due);
}));
}
}
int VideoReceiveStream2::DecodeAndMaybeDispatchEncodedFrame(
std::unique_ptr<EncodedFrame> frame) {
// Running on decode_queue_.
// If `buffered_encoded_frames_` grows out of control (=60 queued frames),
// maybe due to a stuck decoder, we just halt the process here and log the
// error.
const bool encoded_frame_output_enabled =
encoded_frame_buffer_function_ != nullptr &&
buffered_encoded_frames_.size() < kBufferedEncodedFramesMaxSize;
EncodedFrame* frame_ptr = frame.get();
if (encoded_frame_output_enabled) {
// If we receive a key frame with unset resolution, hold on dispatching the
// frame and following ones until we know a resolution of the stream.
// NOTE: The code below has a race where it can report the wrong
// resolution for keyframes after an initial keyframe of other resolution.
// However, the only known consumer of this information is the W3C
// MediaRecorder and it will only use the resolution in the first encoded
// keyframe from WebRTC, so misreporting is fine.
buffered_encoded_frames_.push_back(std::move(frame));
if (buffered_encoded_frames_.size() == kBufferedEncodedFramesMaxSize)
RTC_LOG(LS_ERROR) << "About to halt recordable encoded frame output due "
"to too many buffered frames.";
webrtc::MutexLock lock(&pending_resolution_mutex_);
if (IsKeyFrameAndUnspecifiedResolution(*frame_ptr) &&
!pending_resolution_.has_value())
pending_resolution_.emplace();
}
int decode_result = video_receiver_.Decode(frame_ptr);
if (encoded_frame_output_enabled) {
absl::optional<RecordableEncodedFrame::EncodedResolution>
pending_resolution;
{
// Fish out `pending_resolution_` to avoid taking the mutex on every lap
// or dispatching under the mutex in the flush loop.
webrtc::MutexLock lock(&pending_resolution_mutex_);
if (pending_resolution_.has_value())
pending_resolution = *pending_resolution_;
}
if (!pending_resolution.has_value() || !pending_resolution->empty()) {
// Flush the buffered frames.
for (const auto& frame : buffered_encoded_frames_) {
RecordableEncodedFrame::EncodedResolution resolution{
frame->EncodedImage()._encodedWidth,
frame->EncodedImage()._encodedHeight};
if (IsKeyFrameAndUnspecifiedResolution(*frame)) {
RTC_DCHECK(!pending_resolution->empty());
resolution = *pending_resolution;
}
encoded_frame_buffer_function_(
WebRtcRecordableEncodedFrame(*frame, resolution));
}
buffered_encoded_frames_.clear();
}
}
return decode_result;
}
// RTC_RUN_ON(packet_sequence_checker_)
void VideoReceiveStream2::HandleKeyFrameGeneration(
bool received_frame_is_keyframe,
Timestamp now,
bool always_request_key_frame,
bool keyframe_request_is_due) {
bool request_key_frame = always_request_key_frame;
// Repeat sending keyframe requests if we've requested a keyframe.
if (keyframe_generation_requested_) {
if (received_frame_is_keyframe) {
keyframe_generation_requested_ = false;
} else if (keyframe_request_is_due) {
if (!IsReceivingKeyFrame(now)) {
request_key_frame = true;
}
} else {
// It hasn't been long enough since the last keyframe request, do nothing.
}
}
if (request_key_frame) {
// HandleKeyFrameGeneration is initiated from the decode thread -
// RequestKeyFrame() triggers a call back to the decode thread.
// Perhaps there's a way to avoid that.
RequestKeyFrame(now);
}
}
// RTC_RUN_ON(packet_sequence_checker_)
void VideoReceiveStream2::HandleFrameBufferTimeout(Timestamp now,
TimeDelta wait) {
absl::optional<int64_t> last_packet_ms =
rtp_video_stream_receiver_.LastReceivedPacketMs();
// To avoid spamming keyframe requests for a stream that is not active we
// check if we have received a packet within the last 5 seconds.
constexpr TimeDelta kInactiveDuraction = TimeDelta::Seconds(5);
const bool stream_is_active =
last_packet_ms &&
now - Timestamp::Millis(*last_packet_ms) < kInactiveDuraction;
if (!stream_is_active)
stats_proxy_.OnStreamInactive();
if (stream_is_active && !IsReceivingKeyFrame(now) &&
(!config_.crypto_options.sframe.require_frame_encryption ||
rtp_video_stream_receiver_.IsDecryptable())) {
RTC_LOG(LS_WARNING) << "No decodable frame in " << wait
<< ", requesting keyframe.";
RequestKeyFrame(now);
}
}
// RTC_RUN_ON(packet_sequence_checker_)
bool VideoReceiveStream2::IsReceivingKeyFrame(Timestamp now) const {
absl::optional<int64_t> last_keyframe_packet_ms =
rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();
// If we recently have been receiving packets belonging to a keyframe then
// we assume a keyframe is currently being received.
bool receiving_keyframe = last_keyframe_packet_ms &&
now - Timestamp::Millis(*last_keyframe_packet_ms) <
max_wait_for_keyframe_;
return receiving_keyframe;
}
void VideoReceiveStream2::UpdatePlayoutDelays() const {
// Running on worker_sequence_checker_.
const std::initializer_list<absl::optional<TimeDelta>> min_delays = {
frame_minimum_playout_delay_, base_minimum_playout_delay_,
syncable_minimum_playout_delay_};
// Since nullopt < anything, this will return the largest of the minumum
// delays, or nullopt if all are nullopt.
absl::optional<TimeDelta> minimum_delay = std::max(min_delays);
if (minimum_delay) {
auto num_playout_delays_set =
absl::c_count_if(min_delays, [](auto opt) { return opt.has_value(); });
if (num_playout_delays_set > 1 &&
timing_->min_playout_delay() != minimum_delay) {
RTC_LOG(LS_WARNING)
<< "Multiple playout delays set. Actual delay value set to "
<< *minimum_delay << " frame min delay="
<< OptionalDelayToLogString(frame_maximum_playout_delay_)
<< " base min delay="
<< OptionalDelayToLogString(base_minimum_playout_delay_)
<< " sync min delay="
<< OptionalDelayToLogString(syncable_minimum_playout_delay_);
}
timing_->set_min_playout_delay(*minimum_delay);
if (frame_minimum_playout_delay_ == TimeDelta::Zero() &&
frame_maximum_playout_delay_ > TimeDelta::Zero()) {
// TODO(kron): Estimate frame rate from video stream.
constexpr Frequency kFrameRate = Frequency::Hertz(60);
// Convert playout delay in ms to number of frames.
int max_composition_delay_in_frames =
std::lrint(*frame_maximum_playout_delay_ * kFrameRate);
// Subtract frames in buffer.
max_composition_delay_in_frames =
std::max(max_composition_delay_in_frames - frame_buffer_->Size(), 0);
timing_->SetMaxCompositionDelayInFrames(max_composition_delay_in_frames);
}
}
if (frame_maximum_playout_delay_) {
timing_->set_max_playout_delay(*frame_maximum_playout_delay_);
}
}
std::vector<webrtc::RtpSource> VideoReceiveStream2::GetSources() const {
return source_tracker_.GetSources();
}
VideoReceiveStream2::RecordingState
VideoReceiveStream2::SetAndGetRecordingState(RecordingState state,
bool generate_key_frame) {
RTC_DCHECK_RUN_ON(&worker_sequence_checker_);
rtc::Event event;
// Save old state, set the new state.
RecordingState old_state;
decode_queue_.PostTask(
[this, &event, &old_state, callback = std::move(state.callback),
generate_key_frame,
last_keyframe_request =
Timestamp::Millis(state.last_keyframe_request_ms.value_or(0))] {
RTC_DCHECK_RUN_ON(&decode_queue_);
old_state.callback = std::move(encoded_frame_buffer_function_);
encoded_frame_buffer_function_ = std::move(callback);
old_state.last_keyframe_request_ms =
last_keyframe_request_.value_or(Timestamp::Zero()).ms();
last_keyframe_request_ =
generate_key_frame ? clock_->CurrentTime() : last_keyframe_request;
event.Set();
});
if (generate_key_frame) {
rtp_video_stream_receiver_.RequestKeyFrame();
{
// TODO(bugs.webrtc.org/11993): Post this to the network thread.
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
keyframe_generation_requested_ = true;
}
}
event.Wait(rtc::Event::kForever);
return old_state;
}
void VideoReceiveStream2::GenerateKeyFrame() {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RequestKeyFrame(clock_->CurrentTime());
keyframe_generation_requested_ = true;
}
} // namespace internal
} // namespace webrtc
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