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webrtc / src / 15b2ca7e772d7181f1ee3e426f1c341e89c9bbc2 / . / video / video_receive_stream2.cc
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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/pending_task_safety_flag.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/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()),
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, // 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_.Get(), 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::SetRtcpMode(RtcpMode mode) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<RtcpMode&>(config_.rtp.rtcp_mode) = mode;
rtp_video_stream_receiver_.SetRtcpMode(mode);
}
void VideoReceiveStream2::SetFlexFecProtection(
RtpPacketSinkInterface* flexfec_sink) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtp_video_stream_receiver_.SetPacketSink(flexfec_sink);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<RtpPacketSinkInterface*&>(config_.rtp.packet_sink_) = flexfec_sink;
const_cast<bool&>(config_.rtp.protected_by_flexfec) =
(flexfec_sink != nullptr);
}
void VideoReceiveStream2::SetLossNotificationEnabled(bool enabled) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
// TODO(tommi): Stop using the config struct for the internal state.
const_cast<bool&>(config_.rtp.lntf.enabled) = enabled;
rtp_video_stream_receiver_.SetLossNotificationEnabled(enabled);
}
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(
SafeTask(task_safety_.flag(), [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::RequestKeyFrame(Timestamp now) {
RTC_DCHECK_RUN_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) {
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) {
RTC_DCHECK_RUN_ON(&decode_queue_);
Timestamp now = clock_->CurrentTime();
// TODO(bugs.webrtc.org/11993): PostTask to the network thread.
call_->worker_thread()->PostTask(
SafeTask(task_safety_.flag(), [this, wait_time, now] {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
HandleFrameBufferTimeout(now, 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) {
RTC_DCHECK_RUN_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(SafeTask(
task_safety_.flag(),
[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) {
RTC_DCHECK_RUN_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;
}
void VideoReceiveStream2::HandleKeyFrameGeneration(
bool received_frame_is_keyframe,
Timestamp now,
bool always_request_key_frame,
bool keyframe_request_is_due) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
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);
}
}
void VideoReceiveStream2::HandleFrameBufferTimeout(Timestamp now,
TimeDelta wait) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
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);
}
}
bool VideoReceiveStream2::IsReceivingKeyFrame(Timestamp now) const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
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 {
RTC_DCHECK_RUN_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