| /* | |
| * 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 <algorithm> | |
| #include <cstdint> | |
| #include <cstdio> | |
| #include <map> | |
| #include <memory> | |
| #include <optional> | |
| #include <span> | |
| #include <string> | |
| #include <utility> | |
| #include <vector> | |
| #include "absl/algorithm/container.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/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/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". | |
| constexpr int kMaxVbaSizeDifferencePercent = 10; | |
| // Max time we will throttle similar video bitrate allocations. | |
| 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. | |
| 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; | |
| } | |
| std::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. | |
| std::optional<float> GetConfiguredPacingFactor( | |
| const VideoSendStream::Config& config, | |
| VideoEncoderConfig::ContentType content_type, | |
| const PacingConfig& default_pacing_config, | |
| const FieldTrialsView& field_trials) { | |
| if (!TransportSeqNumExtensionConfigured(config)) | |
| return std::nullopt; | |
| std::optional<AlrExperimentSettings> alr_settings = | |
| GetAlrSettings(field_trials, content_type); | |
| if (alr_settings) | |
| return alr_settings->pacing_factor; | |
| return RateControlSettings(field_trials) | |
| .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) { | |
| FieldTrialParameter<int> av1_priority_bitrate("bitrate", 0); | |
| 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 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; | |
| // Don't log an error for -1 since this is the default value that is used to | |
| // signal that the max bitrate is unset. | |
| if (initial_encoder_max_bitrate != -1) { | |
| 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 (RtpExtension::FindHeaderExtensionByUri( | |
| config.rtp.extensions, 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, | |
| VideoStreamEncoderSettings encoder_settings, | |
| VideoStreamEncoder::BitrateAllocationCallbackType | |
| bitrate_allocation_callback_type, | |
| Metronome* metronome, | |
| VideoEncoderFactory::EncoderSelectorInterface* encoder_selector, | |
| EncoderSwitchRequestCallback encoder_switch_request_callback) { | |
| std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue = | |
| env.task_queue_factory().CreateTaskQueue( | |
| "VideoEncoderQueue", | |
| env.field_trials().IsEnabled("WebRTC-MediaTaskQueuePriorities") | |
| ? TaskQueueFactory::Priority::kVideo | |
| : TaskQueueFactory::Priority::kNormal); | |
| TaskQueueBase* encoder_queue_ptr = encoder_queue.get(); | |
| return std::make_unique<VideoStreamEncoder>( | |
| env, num_cpu_cores, stats_proxy, std::move(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, std::move(encoder_switch_request_callback)); | |
| } | |
| 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, | |
| EncoderSwitchRequestCallback encoder_switch_request_callback, | |
| 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_, | |
| std::move(config_.encoder_settings), | |
| GetBitrateAllocationCallbackType(config_, | |
| env_.field_trials()), | |
| metronome, | |
| config_.encoder_selector, | |
| std::move(encoder_switch_request_callback))), | |
| encoder_feedback_( | |
| env_, | |
| 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())); | |
| std::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_) { | |
| std::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 { | |
| enable_alr_bw_probing = | |
| RateControlSettings(env_.field_trials()).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( | |
| scoped_refptr<Resource> resource) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| video_stream_encoder_->AddAdaptationResource(resource); | |
| } | |
| std::vector<scoped_refptr<Resource>> | |
| VideoSendStreamImpl::GetAdaptationResources() { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| return video_stream_encoder_->GetAdaptationResources(); | |
| } | |
| void VideoSendStreamImpl::SetSource( | |
| VideoSourceInterface<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_INFO) << "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(); | |
| } | |
| void VideoSendStreamImpl::SetStats(const Stats& stats) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| stats_proxy_.SetStats(stats); | |
| } | |
| void VideoSendStreamImpl::SetCsrcs(std::span<const uint32_t> csrcs) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| rtp_video_sender_->SetCsrcs(csrcs); | |
| } | |
| std::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(std::span<const uint8_t> packet) { | |
| RTC_DCHECK_RUN_ON(&thread_checker_); | |
| rtp_video_sender_->DeliverRtcp(packet); | |
| } | |
| 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(), 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{ | |
| .min_bitrate_bps = static_cast<uint32_t>(encoder_min_bitrate_bps_), | |
| .max_bitrate_bps = encoder_max_bitrate_bps_, | |
| .pad_up_bitrate_bps = | |
| static_cast<uint32_t>(disable_padding_ ? 0 : max_padding_bitrate_), | |
| .priority_bitrate_bps = encoder_av1_priority_bitrate_override_bps_, | |
| .enforce_min_bitrate = !config_.suspend_below_min_bitrate, | |
| .bitrate_priority = encoder_bitrate_priority_, | |
| .rate_elasticity = | |
| (content_type_ == VideoEncoderConfig::ContentType::kRealtimeVideo) | |
| ? std::optional(TrackRateElasticity::kCanConsumeExtraRate) | |
| : std::nullopt}; | |
| } | |
| 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 std::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)); | |
| double stream_bitrate_priority_sum = 0; | |
| uint32_t encoder_max_bitrate_bps = 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; | |
| if (encoder_max_bitrate_bps > 0) { | |
| 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; | |
| } | |
| void VideoSendStreamImpl::OnFrameDropped(uint32_t rtp_timestamp, | |
| int spatial_id, | |
| bool is_end_of_temporal_unit) { | |
| 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."; | |
| 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); | |
| } | |
| encoder_target_rate_bps_ = | |
| std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); | |
| 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, link_allocation, | |
| 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; | |
| } | |
| std::optional<DataRate> VideoSendStreamImpl::GetUsedRate() const { | |
| // This value is for real-time video. Screenshare may have unused bandwidth | |
| // that can be shared, and this needs to be changed to support that. | |
| return std::nullopt; | |
| } | |
| } // namespace internal | |
| } // namespace webrtc |