| /* |
| * 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 "api/crypto/cryptooptions.h" |
| #include "api/rtpparameters.h" |
| #include "api/video_codecs/video_codec.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "call/video_send_stream.h" |
| #include "common_types.h" // NOLINT(build/include) |
| #include "modules/pacing/paced_sender.h" |
| #include "rtc_base/atomicops.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/experiments/alr_experiment.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/scoped_ref_ptr.h" |
| #include "rtc_base/sequenced_task_checker.h" |
| #include "rtc_base/thread_checker.h" |
| #include "rtc_base/timeutils.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; |
| |
| bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) { |
| const std::vector<RtpExtension>& extensions = config.rtp.extensions; |
| return std::find_if( |
| extensions.begin(), extensions.end(), [](const RtpExtension& ext) { |
| return ext.uri == RtpExtension::kTransportSequenceNumberUri; |
| }) != extensions.end(); |
| } |
| |
| const char kForcedFallbackFieldTrial[] = |
| "WebRTC-VP8-Forced-Fallback-Encoder-v2"; |
| |
| absl::optional<int> GetFallbackMinBpsFromFieldTrial() { |
| if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial)) |
| return absl::nullopt; |
| |
| std::string group = |
| webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial); |
| if (group.empty()) |
| return absl::nullopt; |
| |
| int min_pixels; |
| int max_pixels; |
| int min_bps; |
| if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels, |
| &min_bps) != 3) { |
| return absl::nullopt; |
| } |
| |
| if (min_bps <= 0) |
| return absl::nullopt; |
| |
| return min_bps; |
| } |
| |
| int GetEncoderMinBitrateBps() { |
| const int kDefaultEncoderMinBitrateBps = 30000; |
| return GetFallbackMinBpsFromFieldTrial().value_or( |
| kDefaultEncoderMinBitrateBps); |
| } |
| |
| // Calculate max padding bitrate for a multi layer codec. |
| int CalculateMaxPadBitrateBps(const std::vector<VideoStream>& streams, |
| int min_transmit_bitrate_bps, |
| bool pad_to_min_bitrate, |
| bool alr_probing) { |
| int pad_up_to_bitrate_bps = 0; |
| |
| // 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) { |
| 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 { |
| // Pad to min bitrate of the highest layer. |
| pad_up_to_bitrate_bps = |
| active_streams[active_streams.size() - 1].min_bitrate_bps; |
| // Add target_bitrate_bps of the lower layers. |
| for (size_t i = 0; i < active_streams.size() - 1; ++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(CallStats* 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_stats = 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::ms(PacedSender::kMaxQueueLengthMs)) { |
| ParseFieldTrial({&pacing_factor, &max_pacing_delay}, |
| field_trial::FindFullName("WebRTC-Video-Pacing")); |
| } |
| PacingConfig::PacingConfig(const PacingConfig&) = default; |
| PacingConfig::~PacingConfig() = default; |
| |
| // CheckEncoderActivityTask is used for tracking when the encoder last produced |
| // and encoded video frame. If the encoder has not produced anything the last |
| // kEncoderTimeOutMs we also want to stop sending padding. |
| class VideoSendStreamImpl::CheckEncoderActivityTask : public rtc::QueuedTask { |
| public: |
| static const int kEncoderTimeOutMs = 2000; |
| explicit CheckEncoderActivityTask( |
| const rtc::WeakPtr<VideoSendStreamImpl>& send_stream) |
| : activity_(0), send_stream_(std::move(send_stream)), timed_out_(false) {} |
| |
| void Stop() { |
| RTC_CHECK(task_checker_.CalledSequentially()); |
| send_stream_.reset(); |
| } |
| |
| void UpdateEncoderActivity() { |
| // UpdateEncoderActivity is called from VideoSendStreamImpl::Encoded 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. |
| rtc::AtomicOps::ReleaseStore(&activity_, 1); |
| } |
| |
| private: |
| bool Run() override { |
| RTC_CHECK(task_checker_.CalledSequentially()); |
| if (!send_stream_) |
| return true; |
| if (!rtc::AtomicOps::AcquireLoad(&activity_)) { |
| if (!timed_out_) { |
| send_stream_->SignalEncoderTimedOut(); |
| } |
| timed_out_ = true; |
| } else if (timed_out_) { |
| send_stream_->SignalEncoderActive(); |
| timed_out_ = false; |
| } |
| rtc::AtomicOps::ReleaseStore(&activity_, 0); |
| |
| rtc::TaskQueue::Current()->PostDelayedTask( |
| std::unique_ptr<rtc::QueuedTask>(this), kEncoderTimeOutMs); |
| // Return false to prevent this task from being deleted. Ownership has been |
| // transferred to the task queue when PostDelayedTask was called. |
| return false; |
| } |
| volatile int activity_; |
| |
| rtc::SequencedTaskChecker task_checker_; |
| rtc::WeakPtr<VideoSendStreamImpl> send_stream_; |
| bool timed_out_; |
| }; |
| |
| VideoSendStreamImpl::VideoSendStreamImpl( |
| SendStatisticsProxy* stats_proxy, |
| rtc::TaskQueue* worker_queue, |
| CallStats* 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, |
| MediaTransportInterface* media_transport) |
| : has_alr_probing_(config->periodic_alr_bandwidth_probing || |
| GetAlrSettings(content_type)), |
| pacing_config_(PacingConfig()), |
| stats_proxy_(stats_proxy), |
| config_(config), |
| worker_queue_(worker_queue), |
| check_encoder_activity_task_(nullptr), |
| call_stats_(call_stats), |
| transport_(transport), |
| bitrate_allocator_(bitrate_allocator), |
| 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::GetRealTimeClock(), |
| 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_))), |
| weak_ptr_factory_(this), |
| media_transport_(media_transport) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString(); |
| weak_ptr_ = weak_ptr_factory_.GetWeakPtr(); |
| |
| if (media_transport_) { |
| // The configured ssrc is interpreted as a channel id, so there must be |
| // exactly one. |
| RTC_DCHECK_EQ(config_->rtp.ssrcs.size(), 1); |
| } else { |
| RTC_DCHECK(!config_->rtp.ssrcs.empty()); |
| } |
| RTC_DCHECK(call_stats_); |
| 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 { |
| transport->EnablePeriodicAlrProbing(false); |
| transport->SetPacingFactor(pacing_config_.pacing_factor); |
| configured_pacing_factor_ = pacing_config_.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)); |
| |
| // 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 = |
| std::find_if(config_->rtp.extensions.begin(), |
| config_->rtp.extensions.end(), |
| [](const RtpExtension& extension) { |
| return extension.uri == RtpExtension::kVideoRotationUri; |
| }) == config_->rtp.extensions.end(); |
| |
| video_stream_encoder_->SetSink(this, rotation_applied); |
| } |
| |
| 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) { |
| rtp_video_sender_->RegisterProcessThread(module_process_thread); |
| } |
| |
| void VideoSendStreamImpl::DeRegisterProcessThread() { |
| rtp_video_sender_->DeRegisterProcessThread(); |
| } |
| |
| bool 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); |
| return true; |
| } |
| |
| void VideoSendStreamImpl::UpdateActiveSimulcastLayers( |
| const std::vector<bool> active_layers) { |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| RTC_LOG(LS_INFO) << "VideoSendStream::UpdateActiveSimulcastLayers"; |
| 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, |
| MediaStreamAllocationConfig{ |
| static_cast<uint32_t>(encoder_min_bitrate_bps_), |
| encoder_max_bitrate_bps_, static_cast<uint32_t>(max_padding_bitrate_), |
| !config_->suspend_below_min_bitrate, config_->track_id, |
| encoder_bitrate_priority_, has_packet_feedback_}); |
| // Start monitoring encoder activity. |
| { |
| rtc::CritScope lock(&encoder_activity_crit_sect_); |
| RTC_DCHECK(!check_encoder_activity_task_); |
| check_encoder_activity_task_ = new CheckEncoderActivityTask(weak_ptr_); |
| worker_queue_->PostDelayedTask( |
| std::unique_ptr<rtc::QueuedTask>(check_encoder_activity_task_), |
| CheckEncoderActivityTask::kEncoderTimeOutMs); |
| } |
| |
| 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); |
| { |
| rtc::CritScope lock(&encoder_activity_crit_sect_); |
| check_encoder_activity_task_->Stop(); |
| check_encoder_activity_task_ = nullptr; |
| } |
| video_stream_encoder_->OnBitrateUpdated(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 kEncoderTimeOutMs 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 = rtc::TimeMillis(); |
| 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_); |
| RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active."; |
| bitrate_allocator_->AddObserver( |
| this, |
| MediaStreamAllocationConfig{ |
| static_cast<uint32_t>(encoder_min_bitrate_bps_), |
| encoder_max_bitrate_bps_, static_cast<uint32_t>(max_padding_bitrate_), |
| !config_->suspend_below_min_bitrate, config_->track_id, |
| encoder_bitrate_priority_, has_packet_feedback_}); |
| } |
| |
| void VideoSendStreamImpl::OnEncoderConfigurationChanged( |
| std::vector<VideoStream> streams, |
| int min_transmit_bitrate_bps) { |
| if (!worker_queue_->IsCurrent()) { |
| rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_; |
| worker_queue_->PostTask([send_stream, streams, min_transmit_bitrate_bps]() { |
| if (send_stream) |
| send_stream->OnEncoderConfigurationChanged(std::move(streams), |
| min_transmit_bitrate_bps); |
| }); |
| return; |
| } |
| RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size()); |
| TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged"); |
| RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size()); |
| RTC_DCHECK_RUN_ON(worker_queue_); |
| |
| encoder_min_bitrate_bps_ = |
| std::max(streams[0].min_bitrate_bps, GetEncoderMinBitrateBps()); |
| 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_); |
| |
| const VideoCodecType codec_type = |
| PayloadStringToCodecType(config_->rtp.payload_name); |
| if (codec_type == kVideoCodecVP9) { |
| max_padding_bitrate_ = has_alr_probing_ ? streams[0].min_bitrate_bps |
| : streams[0].target_bitrate_bps; |
| } else { |
| max_padding_bitrate_ = CalculateMaxPadBitrateBps( |
| streams, 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, MediaStreamAllocationConfig{ |
| static_cast<uint32_t>(encoder_min_bitrate_bps_), |
| encoder_max_bitrate_bps_, |
| static_cast<uint32_t>(max_padding_bitrate_), |
| !config_->suspend_below_min_bitrate, config_->track_id, |
| encoder_bitrate_priority_, has_packet_feedback_}); |
| } |
| } |
| |
| EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage( |
| const EncodedImage& encoded_image, |
| const CodecSpecificInfo* codec_specific_info, |
| const RTPFragmentationHeader* fragmentation) { |
| // 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. |
| { |
| rtc::CritScope lock(&encoder_activity_crit_sect_); |
| if (check_encoder_activity_task_) |
| check_encoder_activity_task_->UpdateEncoderActivity(); |
| } |
| |
| EncodedImageCallback::Result result(EncodedImageCallback::Result::OK); |
| if (media_transport_) { |
| int64_t frame_id; |
| { |
| // TODO(nisse): Responsibility for allocation of frame ids should move to |
| // VideoStreamEncoder. |
| rtc::CritScope cs(&media_transport_id_lock_); |
| frame_id = media_transport_frame_id_++; |
| } |
| // TODO(nisse): Responsibility for reference meta data should be moved |
| // upstream, ideally close to the encoders, but probably VideoStreamEncoder |
| // will need to do some translation to produce reference info using frame |
| // ids. |
| std::vector<int64_t> referenced_frame_ids; |
| if (encoded_image._frameType != kVideoFrameKey) { |
| RTC_DCHECK_GT(frame_id, 0); |
| referenced_frame_ids.push_back(frame_id - 1); |
| } |
| media_transport_->SendVideoFrame( |
| config_->rtp.ssrcs[0], webrtc::MediaTransportEncodedVideoFrame( |
| frame_id, referenced_frame_ids, |
| config_->rtp.payload_type, encoded_image)); |
| } else { |
| result = rtp_video_sender_->OnEncodedImage( |
| encoded_image, codec_specific_info, fragmentation); |
| } |
| // 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; |
| } |
| |
| 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."; |
| |
| rtp_video_sender_->OnBitrateUpdated( |
| update.target_bitrate.bps(), |
| rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256), |
| update.round_trip_time.ms(), stats_proxy_->GetSendFrameRate()); |
| encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps(); |
| encoder_target_rate_bps_ = |
| std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); |
| video_stream_encoder_->OnBitrateUpdated( |
| encoder_target_rate_bps_, |
| rtc::dchecked_cast<uint8_t>(update.packet_loss_ratio * 256), |
| update.round_trip_time.ms()); |
| stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_); |
| return rtp_video_sender_->GetProtectionBitrateBps(); |
| } |
| |
| } // namespace internal |
| } // namespace webrtc |
