| /* |
| * 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 <map> |
| #include <memory> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/algorithm/container.h" |
| #include "absl/types/optional.h" |
| #include "api/adaptation/resource.h" |
| #include "api/call/bitrate_allocation.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/environment/environment.h" |
| #include "api/fec_controller.h" |
| #include "api/field_trials_view.h" |
| #include "api/metronome/metronome.h" |
| #include "api/rtp_parameters.h" |
| #include "api/rtp_sender_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/task_queue/task_queue_factory.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/time_delta.h" |
| #include "api/video/encoded_image.h" |
| #include "api/video/video_bitrate_allocation.h" |
| #include "api/video/video_codec_constants.h" |
| #include "api/video/video_codec_type.h" |
| #include "api/video/video_frame.h" |
| #include "api/video/video_frame_type.h" |
| #include "api/video/video_layers_allocation.h" |
| #include "api/video/video_source_interface.h" |
| #include "api/video/video_stream_encoder_settings.h" |
| #include "api/video_codecs/video_codec.h" |
| #include "api/video_codecs/video_encoder.h" |
| #include "api/video_codecs/video_encoder_factory.h" |
| #include "call/bitrate_allocator.h" |
| #include "call/rtp_config.h" |
| #include "call/rtp_transport_controller_send_interface.h" |
| #include "call/video_send_stream.h" |
| #include "media/base/media_constants.h" |
| #include "media/base/sdp_video_format_utils.h" |
| #include "modules/pacing/pacing_controller.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extension_size.h" |
| #include "modules/rtp_rtcp/source/rtp_sender.h" |
| #include "modules/video_coding/include/video_codec_interface.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/strings/string_builder.h" |
| #include "rtc_base/task_utils/repeating_task.h" |
| #include "rtc_base/trace_event.h" |
| #include "system_wrappers/include/clock.h" |
| #include "video/adaptation/overuse_frame_detector.h" |
| #include "video/config/video_encoder_config.h" |
| #include "video/encoder_rtcp_feedback.h" |
| #include "video/frame_cadence_adapter.h" |
| #include "video/send_delay_stats.h" |
| #include "video/send_statistics_proxy.h" |
| #include "video/video_stream_encoder.h" |
| #include "video/video_stream_encoder_interface.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); |
| |
| constexpr double kVideoHysteresis = 1.2; |
| constexpr double kScreenshareHysteresis = 1.35; |
| |
| constexpr int kMinDefaultAv1BitrateBps = |
| 15000; // This value acts as an absolute minimum AV1 bitrate limit. |
| |
| // When send-side BWE is used a stricter 1.1x pacing factor is used, rather than |
| // the 2.5x which is used with receive-side BWE. Provides a more careful |
| // bandwidth rampup with less risk of overshoots causing adverse effects like |
| // packet loss. Not used for receive side BWE, since there we lack the probing |
| // feature and so may result in too slow initial rampup. |
| static constexpr double kStrictPacingMultiplier = 1.1; |
| |
| 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 = |
| content_type == VideoEncoderConfig::ContentType::kScreen |
| ? kScreenshareHysteresis |
| : kVideoHysteresis; |
| 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; |
| } |
| |
| absl::optional<AlrExperimentSettings> GetAlrSettings( |
| const FieldTrialsView& field_trials, |
| VideoEncoderConfig::ContentType content_type) { |
| if (content_type == VideoEncoderConfig::ContentType::kScreen) { |
| return AlrExperimentSettings::CreateFromFieldTrial( |
| field_trials, |
| AlrExperimentSettings::kScreenshareProbingBweExperimentName); |
| } |
| return AlrExperimentSettings::CreateFromFieldTrial( |
| field_trials, |
| 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; |
| } |
| |
| // Returns an optional that has value iff TransportSeqNumExtensionConfigured |
| // is `true` for the given video send stream config. |
| absl::optional<float> GetConfiguredPacingFactor( |
| const VideoSendStream::Config& config, |
| VideoEncoderConfig::ContentType content_type, |
| const PacingConfig& default_pacing_config, |
| const FieldTrialsView& field_trials) { |
| if (!TransportSeqNumExtensionConfigured(config)) |
| return absl::nullopt; |
| |
| absl::optional<AlrExperimentSettings> alr_settings = |
| GetAlrSettings(field_trials, content_type); |
| if (alr_settings) |
| return alr_settings->pacing_factor; |
| |
| RateControlSettings rate_control_settings = |
| RateControlSettings::ParseFromKeyValueConfig(&field_trials); |
| return rate_control_settings.GetPacingFactor().value_or( |
| default_pacing_config.pacing_factor); |
| } |
| |
| int GetEncoderPriorityBitrate(std::string codec_name, |
| const FieldTrialsView& field_trials) { |
| int priority_bitrate = 0; |
| if (PayloadStringToCodecType(codec_name) == VideoCodecType::kVideoCodecAV1) { |
| webrtc::FieldTrialParameter<int> av1_priority_bitrate("bitrate", 0); |
| webrtc::ParseFieldTrial( |
| {&av1_priority_bitrate}, |
| field_trials.Lookup("WebRTC-AV1-OverridePriorityBitrate")); |
| priority_bitrate = av1_priority_bitrate; |
| } |
| return priority_bitrate; |
| } |
| |
| uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) { |
| if (initial_encoder_max_bitrate > 0) |
| return rtc::dchecked_cast<uint32_t>(initial_encoder_max_bitrate); |
| |
| // 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"; |
| return kFallbackMaxBitrateBps; |
| } |
| |
| int GetDefaultMinVideoBitrateBps(VideoCodecType codec_type) { |
| if (codec_type == VideoCodecType::kVideoCodecAV1) { |
| return kMinDefaultAv1BitrateBps; |
| } |
| return kDefaultMinVideoBitrateBps; |
| } |
| |
| size_t CalculateMaxHeaderSize(const RtpConfig& config) { |
| size_t header_size = kRtpHeaderSize; |
| size_t extensions_size = 0; |
| size_t fec_extensions_size = 0; |
| if (!config.extensions.empty()) { |
| RtpHeaderExtensionMap extensions_map(config.extensions); |
| extensions_size = RtpHeaderExtensionSize(RTPSender::VideoExtensionSizes(), |
| extensions_map); |
| fec_extensions_size = |
| RtpHeaderExtensionSize(RTPSender::FecExtensionSizes(), extensions_map); |
| } |
| header_size += extensions_size; |
| if (config.flexfec.payload_type >= 0) { |
| // All FEC extensions again plus maximum FlexFec overhead. |
| header_size += fec_extensions_size + 32; |
| } else { |
| if (config.ulpfec.ulpfec_payload_type >= 0) { |
| // Header with all the FEC extensions will be repeated plus maximum |
| // UlpFec overhead. |
| header_size += fec_extensions_size + 18; |
| } |
| if (config.ulpfec.red_payload_type >= 0) { |
| header_size += 1; // RED header. |
| } |
| } |
| // Additional room for Rtx. |
| if (config.rtx.payload_type >= 0) |
| header_size += kRtxHeaderSize; |
| return header_size; |
| } |
| |
| VideoStreamEncoder::BitrateAllocationCallbackType |
| GetBitrateAllocationCallbackType(const VideoSendStream::Config& config, |
| const FieldTrialsView& field_trials) { |
| if (webrtc::RtpExtension::FindHeaderExtensionByUri( |
| config.rtp.extensions, |
| webrtc::RtpExtension::kVideoLayersAllocationUri, |
| config.crypto_options.srtp.enable_encrypted_rtp_header_extensions |
| ? RtpExtension::Filter::kPreferEncryptedExtension |
| : RtpExtension::Filter::kDiscardEncryptedExtension)) { |
| return VideoStreamEncoder::BitrateAllocationCallbackType:: |
| kVideoLayersAllocation; |
| } |
| if (field_trials.IsEnabled("WebRTC-Target-Bitrate-Rtcp")) { |
| return VideoStreamEncoder::BitrateAllocationCallbackType:: |
| kVideoBitrateAllocation; |
| } |
| return VideoStreamEncoder::BitrateAllocationCallbackType:: |
| kVideoBitrateAllocationWhenScreenSharing; |
| } |
| |
| RtpSenderFrameEncryptionConfig CreateFrameEncryptionConfig( |
| const VideoSendStream::Config* config) { |
| RtpSenderFrameEncryptionConfig frame_encryption_config; |
| frame_encryption_config.frame_encryptor = config->frame_encryptor.get(); |
| frame_encryption_config.crypto_options = config->crypto_options; |
| return frame_encryption_config; |
| } |
| |
| RtpSenderObservers CreateObservers(RtcpRttStats* call_stats, |
| EncoderRtcpFeedback* encoder_feedback, |
| SendStatisticsProxy* stats_proxy, |
| SendPacketObserver* send_packet_observer) { |
| RtpSenderObservers observers; |
| observers.rtcp_rtt_stats = call_stats; |
| observers.intra_frame_callback = encoder_feedback; |
| observers.rtcp_loss_notification_observer = encoder_feedback; |
| 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_packet_observer = send_packet_observer; |
| return observers; |
| } |
| |
| std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder( |
| const Environment& env, |
| int num_cpu_cores, |
| SendStatisticsProxy* stats_proxy, |
| const VideoStreamEncoderSettings& encoder_settings, |
| VideoStreamEncoder::BitrateAllocationCallbackType |
| bitrate_allocation_callback_type, |
| Metronome* metronome, |
| webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) { |
| std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue = |
| env.task_queue_factory().CreateTaskQueue( |
| "EncoderQueue", TaskQueueFactory::Priority::NORMAL); |
| TaskQueueBase* encoder_queue_ptr = encoder_queue.get(); |
| return std::make_unique<VideoStreamEncoder>( |
| env, num_cpu_cores, stats_proxy, encoder_settings, |
| std::make_unique<OveruseFrameDetector>(env, stats_proxy), |
| FrameCadenceAdapterInterface::Create( |
| &env.clock(), encoder_queue_ptr, metronome, |
| /*worker_queue=*/TaskQueueBase::Current(), env.field_trials()), |
| std::move(encoder_queue), bitrate_allocation_callback_type, |
| encoder_selector); |
| } |
| |
| bool HasActiveEncodings(const VideoEncoderConfig& config) { |
| for (const VideoStream& stream : config.simulcast_layers) { |
| if (stream.active) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| } // namespace |
| |
| PacingConfig::PacingConfig(const FieldTrialsView& field_trials) |
| : pacing_factor("factor", kStrictPacingMultiplier), |
| max_pacing_delay("max_delay", PacingController::kMaxExpectedQueueLength) { |
| ParseFieldTrial({&pacing_factor, &max_pacing_delay}, |
| field_trials.Lookup("WebRTC-Video-Pacing")); |
| } |
| PacingConfig::PacingConfig(const PacingConfig&) = default; |
| PacingConfig::~PacingConfig() = default; |
| |
| VideoSendStreamImpl::VideoSendStreamImpl( |
| const Environment& env, |
| int num_cpu_cores, |
| RtcpRttStats* call_stats, |
| RtpTransportControllerSendInterface* transport, |
| Metronome* metronome, |
| BitrateAllocatorInterface* bitrate_allocator, |
| SendDelayStats* send_delay_stats, |
| VideoSendStream::Config config, |
| VideoEncoderConfig encoder_config, |
| const std::map<uint32_t, RtpState>& suspended_ssrcs, |
| const std::map<uint32_t, RtpPayloadState>& suspended_payload_states, |
| std::unique_ptr<FecController> fec_controller, |
| std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_for_test) |
| : env_(env), |
| transport_(transport), |
| stats_proxy_(&env_.clock(), |
| config, |
| encoder_config.content_type, |
| env_.field_trials()), |
| send_packet_observer_(&stats_proxy_, send_delay_stats), |
| config_(std::move(config)), |
| content_type_(encoder_config.content_type), |
| video_stream_encoder_( |
| video_stream_encoder_for_test |
| ? std::move(video_stream_encoder_for_test) |
| : CreateVideoStreamEncoder( |
| env_, |
| num_cpu_cores, |
| &stats_proxy_, |
| config_.encoder_settings, |
| GetBitrateAllocationCallbackType(config_, |
| env_.field_trials()), |
| metronome, |
| config_.encoder_selector)), |
| encoder_feedback_( |
| &env_.clock(), |
| SupportsPerLayerPictureLossIndication( |
| encoder_config.video_format.parameters), |
| config_.rtp.ssrcs, |
| video_stream_encoder_.get(), |
| [this](uint32_t ssrc, const std::vector<uint16_t>& seq_nums) { |
| return rtp_video_sender_->GetSentRtpPacketInfos(ssrc, seq_nums); |
| }), |
| 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_packet_observer_), |
| std::move(fec_controller), |
| CreateFrameEncryptionConfig(&config_), |
| config_.frame_transformer)), |
| has_alr_probing_( |
| config_.periodic_alr_bandwidth_probing || |
| GetAlrSettings(env_.field_trials(), encoder_config.content_type)), |
| pacing_config_(PacingConfig(env_.field_trials())), |
| worker_queue_(TaskQueueBase::Current()), |
| timed_out_(false), |
| bitrate_allocator_(bitrate_allocator), |
| has_active_encodings_(HasActiveEncodings(encoder_config)), |
| disable_padding_(true), |
| max_padding_bitrate_(0), |
| encoder_min_bitrate_bps_(0), |
| encoder_max_bitrate_bps_( |
| GetInitialEncoderMaxBitrate(encoder_config.max_bitrate_bps)), |
| encoder_target_rate_bps_(0), |
| encoder_bitrate_priority_(encoder_config.bitrate_priority), |
| encoder_av1_priority_bitrate_override_bps_( |
| GetEncoderPriorityBitrate(config_.rtp.payload_name, |
| env_.field_trials())), |
| configured_pacing_factor_( |
| GetConfiguredPacingFactor(config_, |
| content_type_, |
| pacing_config_, |
| env_.field_trials())) { |
| RTC_DCHECK_GE(config_.rtp.payload_type, 0); |
| RTC_DCHECK_LE(config_.rtp.payload_type, 127); |
| RTC_DCHECK(!config_.rtp.ssrcs.empty()); |
| RTC_DCHECK(transport_); |
| RTC_DCHECK_NE(encoder_max_bitrate_bps_, 0); |
| RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_.ToString(); |
| |
| RTC_CHECK( |
| AlrExperimentSettings::MaxOneFieldTrialEnabled(env_.field_trials())); |
| |
| absl::optional<bool> enable_alr_bw_probing; |
| |
| // If send-side BWE is enabled, check if we should apply updated probing and |
| // pacing settings. |
| if (configured_pacing_factor_) { |
| absl::optional<AlrExperimentSettings> alr_settings = |
| GetAlrSettings(env_.field_trials(), content_type_); |
| int queue_time_limit_ms; |
| if (alr_settings) { |
| enable_alr_bw_probing = true; |
| queue_time_limit_ms = alr_settings->max_paced_queue_time; |
| } else { |
| RateControlSettings rate_control_settings = |
| RateControlSettings::ParseFromKeyValueConfig(&env_.field_trials()); |
| enable_alr_bw_probing = rate_control_settings.UseAlrProbing(); |
| queue_time_limit_ms = pacing_config_.max_pacing_delay.Get().ms(); |
| } |
| |
| transport_->SetQueueTimeLimit(queue_time_limit_ms); |
| } |
| |
| if (config_.periodic_alr_bandwidth_probing) { |
| enable_alr_bw_probing = config_.periodic_alr_bandwidth_probing; |
| } |
| |
| if (enable_alr_bw_probing) { |
| transport->EnablePeriodicAlrProbing(*enable_alr_bw_probing); |
| } |
| |
| if (configured_pacing_factor_) |
| transport_->SetPacingFactor(*configured_pacing_factor_); |
| |
| // 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); |
| video_stream_encoder_->SetStartBitrate( |
| bitrate_allocator_->GetStartBitrate(this)); |
| video_stream_encoder_->SetFecControllerOverride(rtp_video_sender_); |
| ReconfigureVideoEncoder(std::move(encoder_config)); |
| } |
| |
| VideoSendStreamImpl::~VideoSendStreamImpl() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_LOG(LS_INFO) << "~VideoSendStreamImpl: " << config_.ToString(); |
| RTC_DCHECK(!started()); |
| RTC_DCHECK(!IsRunning()); |
| transport_->DestroyRtpVideoSender(rtp_video_sender_); |
| } |
| |
| void VideoSendStreamImpl::AddAdaptationResource( |
| rtc::scoped_refptr<Resource> resource) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->AddAdaptationResource(resource); |
| } |
| |
| std::vector<rtc::scoped_refptr<Resource>> |
| VideoSendStreamImpl::GetAdaptationResources() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return video_stream_encoder_->GetAdaptationResources(); |
| } |
| |
| void VideoSendStreamImpl::SetSource( |
| rtc::VideoSourceInterface<webrtc::VideoFrame>* source, |
| const DegradationPreference& degradation_preference) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->SetSource(source, degradation_preference); |
| } |
| |
| void VideoSendStreamImpl::ReconfigureVideoEncoder(VideoEncoderConfig config) { |
| ReconfigureVideoEncoder(std::move(config), nullptr); |
| } |
| |
| void VideoSendStreamImpl::ReconfigureVideoEncoder( |
| VideoEncoderConfig config, |
| SetParametersCallback callback) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK_EQ(content_type_, config.content_type); |
| RTC_LOG(LS_VERBOSE) << "Encoder config: " << config.ToString() |
| << " VideoSendStream config: " << config_.ToString(); |
| |
| has_active_encodings_ = HasActiveEncodings(config); |
| if (has_active_encodings_ && rtp_video_sender_->IsActive() && !IsRunning()) { |
| StartupVideoSendStream(); |
| } else if (!has_active_encodings_ && IsRunning()) { |
| StopVideoSendStream(); |
| } |
| video_stream_encoder_->ConfigureEncoder( |
| std::move(config), |
| config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp), |
| std::move(callback)); |
| } |
| |
| VideoSendStream::Stats VideoSendStreamImpl::GetStats() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return stats_proxy_.GetStats(); |
| } |
| |
| absl::optional<float> VideoSendStreamImpl::GetPacingFactorOverride() const { |
| return configured_pacing_factor_; |
| } |
| |
| void VideoSendStreamImpl::StopPermanentlyAndGetRtpStates( |
| VideoSendStreamImpl::RtpStateMap* rtp_state_map, |
| VideoSendStreamImpl::RtpPayloadStateMap* payload_state_map) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| video_stream_encoder_->Stop(); |
| |
| running_ = false; |
| // Always run these cleanup steps regardless of whether running_ was set |
| // or not. This will unregister callbacks before destruction. |
| // See `VideoSendStreamImpl::StopVideoSendStream` for more. |
| Stop(); |
| *rtp_state_map = GetRtpStates(); |
| *payload_state_map = GetRtpPayloadStates(); |
| } |
| |
| void VideoSendStreamImpl::GenerateKeyFrame( |
| const std::vector<std::string>& rids) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // Map rids to layers. If rids is empty, generate a keyframe for all layers. |
| std::vector<VideoFrameType> next_frames(config_.rtp.ssrcs.size(), |
| VideoFrameType::kVideoFrameKey); |
| if (!config_.rtp.rids.empty() && !rids.empty()) { |
| std::fill(next_frames.begin(), next_frames.end(), |
| VideoFrameType::kVideoFrameDelta); |
| for (const auto& rid : rids) { |
| for (size_t i = 0; i < config_.rtp.rids.size(); i++) { |
| if (config_.rtp.rids[i] == rid) { |
| next_frames[i] = VideoFrameType::kVideoFrameKey; |
| break; |
| } |
| } |
| } |
| } |
| if (video_stream_encoder_) { |
| video_stream_encoder_->SendKeyFrame(next_frames); |
| } |
| } |
| |
| void VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| rtp_video_sender_->DeliverRtcp(packet, length); |
| } |
| |
| bool VideoSendStreamImpl::started() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return rtp_video_sender_->IsActive(); |
| } |
| |
| void VideoSendStreamImpl::Start() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| // This sender is allowed to send RTP packets. Start monitoring and allocating |
| // a rate if there is also active encodings. (has_active_encodings_). |
| rtp_video_sender_->SetSending(true); |
| if (!IsRunning() && has_active_encodings_) { |
| StartupVideoSendStream(); |
| } |
| } |
| |
| bool VideoSendStreamImpl::IsRunning() const { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| return check_encoder_activity_task_.Running(); |
| } |
| |
| void VideoSendStreamImpl::StartupVideoSendStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| RTC_DCHECK(rtp_video_sender_->IsActive()); |
| RTC_DCHECK(has_active_encodings_); |
| |
| 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_, kEncoderTimeOut, [this] { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| 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(&thread_checker_); |
| RTC_LOG(LS_INFO) << "VideoSendStreamImpl::Stop"; |
| if (!rtp_video_sender_->IsActive()) |
| return; |
| |
| TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop", |
| TRACE_EVENT_SCOPE_GLOBAL); |
| rtp_video_sender_->SetSending(false); |
| if (IsRunning()) { |
| StopVideoSendStream(); |
| } |
| } |
| |
| void VideoSendStreamImpl::StopVideoSendStream() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| 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(&thread_checker_); |
| // 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) { |
| // OnBitrateAllocationUpdated is invoked from the encoder task queue or |
| // the worker_queue_. |
| auto task = [this, allocation] { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (encoder_target_rate_bps_ == 0) { |
| return; |
| } |
| int64_t now_ms = env_.clock().TimeInMilliseconds(); |
| 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); |
| }; |
| if (!worker_queue_->IsCurrent()) { |
| worker_queue_->PostTask( |
| SafeTask(worker_queue_safety_.flag(), std::move(task))); |
| } else { |
| task(); |
| } |
| } |
| |
| void VideoSendStreamImpl::OnVideoLayersAllocationUpdated( |
| VideoLayersAllocation allocation) { |
| // OnVideoLayersAllocationUpdated is handled on the encoder task queue in |
| // order to not race with OnEncodedImage callbacks. |
| rtp_video_sender_->OnVideoLayersAllocationUpdated(allocation); |
| } |
| |
| void VideoSendStreamImpl::SignalEncoderActive() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (IsRunning()) { |
| 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_), |
| encoder_av1_priority_bitrate_override_bps_, |
| !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) { |
| // Currently called on the encoder TQ |
| RTC_DCHECK(!worker_queue_->IsCurrent()); |
| auto closure = [this, streams = std::move(streams), is_svc, content_type, |
| min_transmit_bitrate_bps]() mutable { |
| RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size()); |
| TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged"); |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| |
| const VideoCodecType codec_type = |
| PayloadStringToCodecType(config_.rtp.payload_name); |
| |
| const absl::optional<DataRate> experimental_min_bitrate = |
| GetExperimentalMinVideoBitrate(env_.field_trials(), codec_type); |
| encoder_min_bitrate_bps_ = |
| experimental_min_bitrate |
| ? experimental_min_bitrate->bps() |
| : std::max(streams[0].min_bitrate_bps, |
| GetDefaultMinVideoBitrateBps(codec_type)); |
| |
| 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. |
| if (stream.active) { |
| encoder_max_bitrate_bps_ += stream.max_bitrate_bps; |
| } |
| 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 (IsRunning()) { |
| // The send stream is started already. Update the allocator with new |
| // bitrate limits. |
| bitrate_allocator_->AddObserver(this, GetAllocationConfig()); |
| } |
| }; |
| |
| worker_queue_->PostTask( |
| SafeTask(worker_queue_safety_.flag(), std::move(closure))); |
| } |
| |
| 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; |
| RTC_DCHECK(!worker_queue_->IsCurrent()); |
| |
| auto task_to_run_on_worker = [this]() { |
| RTC_DCHECK_RUN_ON(&thread_checker_); |
| if (disable_padding_) { |
| disable_padding_ = false; |
| // To ensure that padding bitrate is propagated to the bitrate allocator. |
| SignalEncoderActive(); |
| } |
| // Check if there's a throttled VideoBitrateAllocation that we should try |
| // sending. |
| auto& context = video_bitrate_allocation_context_; |
| if (context && context->throttled_allocation) { |
| OnBitrateAllocationUpdated(*context->throttled_allocation); |
| } |
| }; |
| worker_queue_->PostTask( |
| SafeTask(worker_queue_safety_.flag(), std::move(task_to_run_on_worker))); |
| |
| return rtp_video_sender_->OnEncodedImage(encoded_image, codec_specific_info); |
| } |
| |
| 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(&thread_checker_); |
| 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 |
