| /* |
| * Copyright 2018 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_send_stream_impl.h" |
| |
| #include <stdio.h> |
| |
| #include <algorithm> |
| #include <cstdint> |
| #include <string> |
| #include <utility> |
| |
| #include "absl/algorithm/container.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/rtp_parameters.h" |
| #include "api/scoped_refptr.h" |
| #include "api/video_codecs/video_codec.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "call/video_send_stream.h" |
| #include "modules/pacing/paced_sender.h" |
| #include "rtc_base/atomic_ops.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/experiments/alr_experiment.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/experiments/min_video_bitrate_experiment.h" |
| #include "rtc_base/experiments/rate_control_settings.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/synchronization/sequence_checker.h" |
| #include "rtc_base/thread_checker.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/clock.h" |
| #include "system_wrappers/include/field_trial.h" |
| |
| namespace webrtc { |
| namespace internal { |
| namespace { |
| |
| // Max positive size difference to treat allocations as "similar". |
| static constexpr int kMaxVbaSizeDifferencePercent = 10; |
| // Max time we will throttle similar video bitrate allocations. |
| static constexpr int64_t kMaxVbaThrottleTimeMs = 500; |
| |
| constexpr TimeDelta kEncoderTimeOut = TimeDelta::Seconds(2); |
| |
| bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) { |
| const std::vector<RtpExtension>& extensions = config.rtp.extensions; |
| return absl::c_any_of(extensions, [](const RtpExtension& ext) { |
| return ext.uri == RtpExtension::kTransportSequenceNumberUri; |
| }); |
| } |
| |
| // Calculate max padding bitrate for a multi layer codec. |
| int CalculateMaxPadBitrateBps(const std::vector<VideoStream>& streams, |
| bool is_svc, |
| VideoEncoderConfig::ContentType content_type, |
| int min_transmit_bitrate_bps, |
| bool pad_to_min_bitrate, |
| bool alr_probing) { |
| int pad_up_to_bitrate_bps = 0; |
| |
| RTC_DCHECK(!is_svc || streams.size() <= 1) << "Only one stream is allowed in " |
| "SVC mode."; |
| |
| // Filter out only the active streams; |
| std::vector<VideoStream> active_streams; |
| for (const VideoStream& stream : streams) { |
| if (stream.active) |
| active_streams.emplace_back(stream); |
| } |
| |
| if (active_streams.size() > 1 || (!active_streams.empty() && is_svc)) { |
| // Simulcast or SVC is used. |
| // if SVC is used, stream bitrates should already encode svc bitrates: |
| // min_bitrate = min bitrate of a lowest svc layer. |
| // target_bitrate = sum of target bitrates of lower layers + min bitrate |
| // of the last one (as used in the calculations below). |
| // max_bitrate = sum of all active layers' max_bitrate. |
| if (alr_probing) { |
| // With alr probing, just pad to the min bitrate of the lowest stream, |
| // probing will handle the rest of the rampup. |
| pad_up_to_bitrate_bps = active_streams[0].min_bitrate_bps; |
| } else { |
| // Without alr probing, pad up to start bitrate of the |
| // highest active stream. |
| const double hysteresis_factor = |
| RateControlSettings::ParseFromFieldTrials() |
| .GetSimulcastHysteresisFactor(content_type); |
| if (is_svc) { |
| // For SVC, since there is only one "stream", the padding bitrate |
| // needed to enable the top spatial layer is stored in the |
| // |target_bitrate_bps| field. |
| // TODO(sprang): This behavior needs to die. |
| pad_up_to_bitrate_bps = static_cast<int>( |
| hysteresis_factor * active_streams[0].target_bitrate_bps + 0.5); |
| } else { |
| const size_t top_active_stream_idx = active_streams.size() - 1; |
| pad_up_to_bitrate_bps = std::min( |
| static_cast<int>( |
| hysteresis_factor * |
| active_streams[top_active_stream_idx].min_bitrate_bps + |
| 0.5), |
| active_streams[top_active_stream_idx].target_bitrate_bps); |
| |
| // Add target_bitrate_bps of the lower active streams. |
| for (size_t i = 0; i < top_active_stream_idx; ++i) { |
| pad_up_to_bitrate_bps += active_streams[i].target_bitrate_bps; |
| } |
| } |
| } |
| } else if (!active_streams.empty() && pad_to_min_bitrate) { |
| pad_up_to_bitrate_bps = active_streams[0].min_bitrate_bps; |
| } |
| |
| pad_up_to_bitrate_bps = |
| std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps); |
| |
| return pad_up_to_bitrate_bps; |
| } |
| |
| RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig( |
| const VideoSendStream::Config* config) { |
| RtpSenderFrameEncryptionConfig frame_encryption_config; |
| frame_encryption_config.frame_encryptor = config->frame_encryptor; |
| frame_encryption_config.crypto_options = config->crypto_options; |
| return frame_encryption_config; |
| } |
| |
| RtpSenderObservers CreateObservers(RtcpRttStats* call_stats, |
| EncoderRtcpFeedback* encoder_feedback, |
| SendStatisticsProxy* stats_proxy, |
| SendDelayStats* send_delay_stats) { |
| RtpSenderObservers observers; |
| observers.rtcp_rtt_stats = call_stats; |
| observers.intra_frame_callback = encoder_feedback; |
| observers.rtcp_loss_notification_observer = encoder_feedback; |
| observers.rtcp_stats = stats_proxy; |
| observers.report_block_data_observer = stats_proxy; |
| observers.rtp_stats = stats_proxy; |
| observers.bitrate_observer = stats_proxy; |
| observers.frame_count_observer = stats_proxy; |
| observers.rtcp_type_observer = stats_proxy; |
| observers.send_delay_observer = stats_proxy; |
| observers.send_packet_observer = send_delay_stats; |
| return observers; |
| } |
| |
| absl::optional<AlrExperimentSettings> GetAlrSettings( |
| VideoEncoderConfig::ContentType content_type) { |
| if (content_type == VideoEncoderConfig::ContentType::kScreen) { |
| return AlrExperimentSettings::CreateFromFieldTrial( |
| AlrExperimentSettings::kScreenshareProbingBweExperimentName); |
| } |
| return AlrExperimentSettings::CreateFromFieldTrial( |
| AlrExperimentSettings::kStrictPacingAndProbingExperimentName); |
| } |
| |
| bool SameStreamsEnabled(const VideoBitrateAllocation& lhs, |
| const VideoBitrateAllocation& rhs) { |
| for (size_t si = 0; si < kMaxSpatialLayers; ++si) { |
| for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) { |
| if (lhs.HasBitrate(si, ti) != rhs.HasBitrate(si, ti)) { |
| return false; |
| } |
| } |
| } |
| return true; |
| } |
| } // namespace |
| |
| PacingConfig::PacingConfig() |
| : pacing_factor("factor", PacedSender::kDefaultPaceMultiplier), |
| max_pacing_delay("max_delay", |
| TimeDelta::Millis(PacedSender::kMaxQueueLengthMs)) { |
| ParseFieldTrial({&pacing_factor, &max_pacing_delay}, |
| field_trial::FindFullName("WebRTC-Video-Pacing")); |
| } |
| PacingConfig::PacingConfig(const PacingConfig&) = default; |
| PacingConfig::~PacingConfig() = default; |
| |
| VideoSendStreamImpl::VideoSendStreamImpl( |
| Clock* clock, |
| SendStatisticsProxy* stats_proxy, |
| rtc::TaskQueue* worker_queue, |
| RtcpRttStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| BitrateAllocatorInterface* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| VideoStreamEncoderInterface* video_stream_encoder, |
| RtcEventLog* event_log, |
| const VideoSendStream::Config* config, |
| int initial_encoder_max_bitrate, |
| double initial_encoder_bitrate_priority, |
| std::map<uint32_t, RtpState> suspended_ssrcs, |
| std::map<uint32_t, RtpPayloadState> suspended_payload_states, |
| VideoEncoderConfig::ContentType content_type, |
| std::unique_ptr<FecController> fec_controller) |
| : clock_(clock), |
| has_alr_probing_(config->periodic_alr_bandwidth_probing || |
| GetAlrSettings(content_type)), |
| pacing_config_(PacingConfig()), |
| stats_proxy_(stats_proxy), |
| config_(config), |
| worker_queue_(worker_queue), |
| timed_out_(false), |
| transport_(transport), |
| bitrate_allocator_(bitrate_allocator), |
| disable_padding_(true), |
| max_padding_bitrate_(0), |
| encoder_min_bitrate_bps_(0), |
| encoder_target_rate_bps_(0), |
| encoder_bitrate_priority_(initial_encoder_bitrate_priority), |
| has_packet_feedback_(false), |
| video_stream_encoder_(video_stream_encoder), |
| encoder_feedback_(clock, config_->rtp.ssrcs, video_stream_encoder), |
| bandwidth_observer_(transport->GetBandwidthObserver()), |
| rtp_video_sender_( |
| transport_->CreateRtpVideoSender(suspended_ssrcs, |
| suspended_payload_states, |
| config_->rtp, |
| config_->rtcp_report_interval_ms, |
| config_->send_transport, |
| CreateObservers(call_stats, |
| &encoder_feedback_, |
| stats_proxy_, |
| send_delay_stats), |
| event_log, |
| std::move(fec_controller), |
| CreateFrameEncryptionConfig(config_), |
| config->frame_transformer)), |
| weak_ptr_factory_(this) { |
| video_stream_encoder->SetFecControllerOverride(rtp_video_sender_); |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString(); |
| weak_ptr_ = weak_ptr_factory_.GetWeakPtr(); |
| |
| encoder_feedback_.SetRtpVideoSender(rtp_video_sender_); |
| |
| RTC_DCHECK(!config_->rtp.ssrcs.empty()); |
| RTC_DCHECK(transport_); |
| RTC_DCHECK_NE(initial_encoder_max_bitrate, 0); |
| |
| if (initial_encoder_max_bitrate > 0) { |
| encoder_max_bitrate_bps_ = |
| rtc::dchecked_cast<uint32_t>(initial_encoder_max_bitrate); |
| } else { |
| // TODO(srte): Make sure max bitrate is not set to negative values. We don't |
| // have any way to handle unset values in downstream code, such as the |
| // bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a |
| // behaviour that is not safe. Converting to 10 Mbps should be safe for |
| // reasonable use cases as it allows adding the max of multiple streams |
| // without wrappping around. |
| const int kFallbackMaxBitrateBps = 10000000; |
| RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = " |
| << initial_encoder_max_bitrate << " which is <= 0!"; |
| RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps"; |
| encoder_max_bitrate_bps_ = kFallbackMaxBitrateBps; |
| } |
| |
| RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled()); |
| // If send-side BWE is enabled, check if we should apply updated probing and |
| // pacing settings. |
| if (TransportSeqNumExtensionConfigured(*config_)) { |
| has_packet_feedback_ = true; |
| |
| absl::optional<AlrExperimentSettings> alr_settings = |
| GetAlrSettings(content_type); |
| if (alr_settings) { |
| transport->EnablePeriodicAlrProbing(true); |
| transport->SetPacingFactor(alr_settings->pacing_factor); |
| configured_pacing_factor_ = alr_settings->pacing_factor; |
| transport->SetQueueTimeLimit(alr_settings->max_paced_queue_time); |
| } else { |
| RateControlSettings rate_control_settings = |
| RateControlSettings::ParseFromFieldTrials(); |
| |
| transport->EnablePeriodicAlrProbing( |
| rate_control_settings.UseAlrProbing()); |
| const double pacing_factor = |
| rate_control_settings.GetPacingFactor().value_or( |
| pacing_config_.pacing_factor); |
| transport->SetPacingFactor(pacing_factor); |
| configured_pacing_factor_ = pacing_factor; |
| transport->SetQueueTimeLimit(pacing_config_.max_pacing_delay.Get().ms()); |
| } |
| } |
| |
| if (config_->periodic_alr_bandwidth_probing) { |
| transport->EnablePeriodicAlrProbing(true); |
| } |
| |
| RTC_DCHECK_GE(config_->rtp.payload_type, 0); |
| RTC_DCHECK_LE(config_->rtp.payload_type, 127); |
| |
| video_stream_encoder_->SetStartBitrate( |
| bitrate_allocator_->GetStartBitrate(this)); |
| } |
| |
| VideoSendStreamImpl::~VideoSendStreamImpl() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_DCHECK(!rtp_video_sender_->IsActive()) |
| << "VideoSendStreamImpl::Stop not called"; |
| RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString(); |
| transport_->DestroyRtpVideoSender(rtp_video_sender_); |
| } |
| |
| void VideoSendStreamImpl::RegisterProcessThread( |
| ProcessThread* module_process_thread) { |
| // Called on libjingle's worker thread (not worker_queue_), as part of the |
| // initialization steps. That's also the correct thread/queue for setting the |
| // state for |video_stream_encoder_|. |
| |
| // Only request rotation at the source when we positively know that the remote |
| // side doesn't support the rotation extension. This allows us to prepare the |
| // encoder in the expectation that rotation is supported - which is the common |
| // case. |
| bool rotation_applied = absl::c_none_of( |
| config_->rtp.extensions, [](const RtpExtension& extension) { |
| return extension.uri == RtpExtension::kVideoRotationUri; |
| }); |
| |
| video_stream_encoder_->SetSink(this, rotation_applied); |
| |
| rtp_video_sender_->RegisterProcessThread(module_process_thread); |
| } |
| |
| void VideoSendStreamImpl::DeRegisterProcessThread() { |
| rtp_video_sender_->DeRegisterProcessThread(); |
| } |
| |
| void VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { |
| // Runs on a network thread. |
| RTC_DCHECK(!worker_queue_->IsCurrent()); |
| rtp_video_sender_->DeliverRtcp(packet, length); |
| } |
| |
| void VideoSendStreamImpl::UpdateActiveSimulcastLayers( |
| const std::vector<bool> active_layers) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| bool previously_active = rtp_video_sender_->IsActive(); |
| rtp_video_sender_->SetActiveModules(active_layers); |
| if (!rtp_video_sender_->IsActive() && previously_active) { |
| // Payload router switched from active to inactive. |
| StopVideoSendStream(); |
| } else if (rtp_video_sender_->IsActive() && !previously_active) { |
| // Payload router switched from inactive to active. |
| StartupVideoSendStream(); |
| } |
| } |
| |
| void VideoSendStreamImpl::Start() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::Start"; |
| if (rtp_video_sender_->IsActive()) |
| return; |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start"); |
| rtp_video_sender_->SetActive(true); |
| StartupVideoSendStream(); |
| } |
| |
| void VideoSendStreamImpl::StartupVideoSendStream() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| bitrate_allocator_->AddObserver(this, GetAllocationConfig()); |
| // Start monitoring encoder activity. |
| { |
| RTC_DCHECK(!check_encoder_activity_task_.Running()); |
| |
| activity_ = false; |
| timed_out_ = false; |
| check_encoder_activity_task_ = RepeatingTaskHandle::DelayedStart( |
| worker_queue_->Get(), kEncoderTimeOut, [this] { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| if (!activity_) { |
| if (!timed_out_) { |
| SignalEncoderTimedOut(); |
| } |
| timed_out_ = true; |
| disable_padding_ = true; |
| } else if (timed_out_) { |
| SignalEncoderActive(); |
| timed_out_ = false; |
| } |
| activity_ = false; |
| return kEncoderTimeOut; |
| }); |
| } |
| |
| video_stream_encoder_->SendKeyFrame(); |
| } |
| |
| void VideoSendStreamImpl::Stop() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::Stop"; |
| if (!rtp_video_sender_->IsActive()) |
| return; |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop"); |
| rtp_video_sender_->SetActive(false); |
| StopVideoSendStream(); |
| } |
| |
| void VideoSendStreamImpl::StopVideoSendStream() { |
| bitrate_allocator_->RemoveObserver(this); |
| check_encoder_activity_task_.Stop(); |
| video_stream_encoder_->OnBitrateUpdated(DataRate::Zero(), DataRate::Zero(), |
| DataRate::Zero(), 0, 0, 0); |
| stats_proxy_->OnSetEncoderTargetRate(0); |
| } |
| |
| void VideoSendStreamImpl::SignalEncoderTimedOut() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| // If the encoder has not produced anything the last kEncoderTimeOut and it |
| // is supposed to, deregister as BitrateAllocatorObserver. This can happen |
| // if a camera stops producing frames. |
| if (encoder_target_rate_bps_ > 0) { |
| RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out."; |
| bitrate_allocator_->RemoveObserver(this); |
| } |
| } |
| |
| void VideoSendStreamImpl::OnBitrateAllocationUpdated( |
| const VideoBitrateAllocation& allocation) { |
| if (!worker_queue_->IsCurrent()) { |
| auto ptr = weak_ptr_; |
| worker_queue_->PostTask([=] { |
| if (!ptr.get()) |
| return; |
| ptr->OnBitrateAllocationUpdated(allocation); |
| }); |
| return; |
| } |
| |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| if (encoder_target_rate_bps_ != 0) { |
| if (video_bitrate_allocation_context_) { |
| // If new allocation is within kMaxVbaSizeDifferencePercent larger than |
| // the previously sent allocation and the same streams are still enabled, |
| // it is considered "similar". We do not want send similar allocations |
| // more once per kMaxVbaThrottleTimeMs. |
| const VideoBitrateAllocation& last = |
| video_bitrate_allocation_context_->last_sent_allocation; |
| const bool is_similar = |
| allocation.get_sum_bps() >= last.get_sum_bps() && |
| allocation.get_sum_bps() < |
| (last.get_sum_bps() * (100 + kMaxVbaSizeDifferencePercent)) / |
| 100 && |
| SameStreamsEnabled(allocation, last); |
| if (is_similar && |
| (now_ms - video_bitrate_allocation_context_->last_send_time_ms) < |
| kMaxVbaThrottleTimeMs) { |
| // This allocation is too similar, cache it and return. |
| video_bitrate_allocation_context_->throttled_allocation = allocation; |
| return; |
| } |
| } else { |
| video_bitrate_allocation_context_.emplace(); |
| } |
| |
| video_bitrate_allocation_context_->last_sent_allocation = allocation; |
| video_bitrate_allocation_context_->throttled_allocation.reset(); |
| video_bitrate_allocation_context_->last_send_time_ms = now_ms; |
| |
| // Send bitrate allocation metadata only if encoder is not paused. |
| rtp_video_sender_->OnBitrateAllocationUpdated(allocation); |
| } |
| } |
| |
| void VideoSendStreamImpl::SignalEncoderActive() { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| if (rtp_video_sender_->IsActive()) { |
| RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active."; |
| bitrate_allocator_->AddObserver(this, GetAllocationConfig()); |
| } |
| } |
| |
| MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const { |
| return MediaStreamAllocationConfig{ |
| static_cast<uint32_t>(encoder_min_bitrate_bps_), |
| encoder_max_bitrate_bps_, |
| static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_), |
| /* priority_bitrate */ 0, |
| !config_->suspend_below_min_bitrate, |
| encoder_bitrate_priority_}; |
| } |
| |
| void VideoSendStreamImpl::OnEncoderConfigurationChanged( |
| std::vector<VideoStream> streams, |
| bool is_svc, |
| VideoEncoderConfig::ContentType content_type, |
| int min_transmit_bitrate_bps) { |
| if (!worker_queue_->IsCurrent()) { |
| rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_; |
| worker_queue_->PostTask([send_stream, streams, is_svc, content_type, |
| min_transmit_bitrate_bps]() mutable { |
| if (send_stream) { |
| send_stream->OnEncoderConfigurationChanged( |
| std::move(streams), is_svc, content_type, min_transmit_bitrate_bps); |
| } |
| }); |
| return; |
| } |
| |
| RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size()); |
| TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged"); |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| |
| const VideoCodecType codec_type = |
| PayloadStringToCodecType(config_->rtp.payload_name); |
| |
| const absl::optional<DataRate> experimental_min_bitrate = |
| GetExperimentalMinVideoBitrate(codec_type); |
| encoder_min_bitrate_bps_ = |
| experimental_min_bitrate |
| ? experimental_min_bitrate->bps() |
| : std::max(streams[0].min_bitrate_bps, kDefaultMinVideoBitrateBps); |
| |
| encoder_max_bitrate_bps_ = 0; |
| double stream_bitrate_priority_sum = 0; |
| for (const auto& stream : streams) { |
| // We don't want to allocate more bitrate than needed to inactive streams. |
| encoder_max_bitrate_bps_ += stream.active ? stream.max_bitrate_bps : 0; |
| if (stream.bitrate_priority) { |
| RTC_DCHECK_GT(*stream.bitrate_priority, 0); |
| stream_bitrate_priority_sum += *stream.bitrate_priority; |
| } |
| } |
| RTC_DCHECK_GT(stream_bitrate_priority_sum, 0); |
| encoder_bitrate_priority_ = stream_bitrate_priority_sum; |
| encoder_max_bitrate_bps_ = |
| std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_), |
| encoder_max_bitrate_bps_); |
| |
| // TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead. |
| max_padding_bitrate_ = CalculateMaxPadBitrateBps( |
| streams, is_svc, content_type, min_transmit_bitrate_bps, |
| config_->suspend_below_min_bitrate, has_alr_probing_); |
| |
| // Clear stats for disabled layers. |
| for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) { |
| stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]); |
| } |
| |
| const size_t num_temporal_layers = |
| streams.back().num_temporal_layers.value_or(1); |
| |
| rtp_video_sender_->SetEncodingData(streams[0].width, streams[0].height, |
| num_temporal_layers); |
| |
| if (rtp_video_sender_->IsActive()) { |
| // The send stream is started already. Update the allocator with new bitrate |
| // limits. |
| bitrate_allocator_->AddObserver(this, GetAllocationConfig()); |
| } |
| } |
| |
| EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage( |
| const EncodedImage& encoded_image, |
| const CodecSpecificInfo* codec_specific_info) { |
| // Encoded is called on whatever thread the real encoder implementation run |
| // on. In the case of hardware encoders, there might be several encoders |
| // running in parallel on different threads. |
| |
| // Indicate that there still is activity going on. |
| activity_ = true; |
| |
| auto enable_padding_task = [this]() { |
| if (disable_padding_) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| disable_padding_ = false; |
| // To ensure that padding bitrate is propagated to the bitrate allocator. |
| SignalEncoderActive(); |
| } |
| }; |
| if (!worker_queue_->IsCurrent()) { |
| worker_queue_->PostTask(enable_padding_task); |
| } else { |
| enable_padding_task(); |
| } |
| |
| EncodedImageCallback::Result result(EncodedImageCallback::Result::OK); |
| result = |
| rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info); |
| // Check if there's a throttled VideoBitrateAllocation that we should try |
| // sending. |
| rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_; |
| auto update_task = [send_stream]() { |
| if (send_stream) { |
| RTC_DCHECK_RUN_ON(send_stream->worker_queue_); |
| auto& context = send_stream->video_bitrate_allocation_context_; |
| if (context && context->throttled_allocation) { |
| send_stream->OnBitrateAllocationUpdated(*context->throttled_allocation); |
| } |
| } |
| }; |
| if (!worker_queue_->IsCurrent()) { |
| worker_queue_->PostTask(update_task); |
| } else { |
| update_task(); |
| } |
| |
| return result; |
| } |
| |
| void VideoSendStreamImpl::OnDroppedFrame( |
| EncodedImageCallback::DropReason reason) { |
| activity_ = true; |
| } |
| |
| std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const { |
| return rtp_video_sender_->GetRtpStates(); |
| } |
| |
| std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates() |
| const { |
| return rtp_video_sender_->GetRtpPayloadStates(); |
| } |
| |
| uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_DCHECK(rtp_video_sender_->IsActive()) |
| << "VideoSendStream::Start has not been called."; |
| |
| // When the BWE algorithm doesn't pass a stable estimate, we'll use the |
| // unstable one instead. |
| if (update.stable_target_bitrate.IsZero()) { |
| update.stable_target_bitrate = update.target_bitrate; |
| } |
| |
| rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_->GetSendFrameRate()); |
| encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps(); |
| const uint32_t protection_bitrate_bps = |
| rtp_video_sender_->GetProtectionBitrateBps(); |
| DataRate link_allocation = DataRate::Zero(); |
| if (encoder_target_rate_bps_ > protection_bitrate_bps) { |
| link_allocation = |
| DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps); |
| } |
| DataRate overhead = |
| update.target_bitrate - DataRate::BitsPerSec(encoder_target_rate_bps_); |
| DataRate encoder_stable_target_rate = update.stable_target_bitrate; |
| if (encoder_stable_target_rate > overhead) { |
| encoder_stable_target_rate = encoder_stable_target_rate - overhead; |
| } else { |
| encoder_stable_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_); |
| } |
| |
| encoder_target_rate_bps_ = |
| std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); |
| |
| encoder_stable_target_rate = |
| std::min(DataRate::BitsPerSec(encoder_max_bitrate_bps_), |
| encoder_stable_target_rate); |
| |
| DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_); |
| link_allocation = std::max(encoder_target_rate, link_allocation); |
| video_stream_encoder_->OnBitrateUpdated( |
| encoder_target_rate, encoder_stable_target_rate, link_allocation, |
| rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256), |
| update.round_trip_time.ms(), update.cwnd_reduce_ratio); |
| stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_); |
| return protection_bitrate_bps; |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |
