From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 19:32:43 +0200 Subject: Adding upstream version 1:115.7.0. Signed-off-by: Daniel Baumann --- .../libwebrtc/video/video_send_stream_impl.cc | 625 +++++++++++++++++++++ 1 file changed, 625 insertions(+) create mode 100644 third_party/libwebrtc/video/video_send_stream_impl.cc (limited to 'third_party/libwebrtc/video/video_send_stream_impl.cc') diff --git a/third_party/libwebrtc/video/video_send_stream_impl.cc b/third_party/libwebrtc/video/video_send_stream_impl.cc new file mode 100644 index 0000000000..f34388e56a --- /dev/null +++ b/third_party/libwebrtc/video/video_send_stream_impl.cc @@ -0,0 +1,625 @@ +/* + * 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 + +#include +#include +#include +#include + +#include "absl/algorithm/container.h" +#include "api/crypto/crypto_options.h" +#include "api/rtp_parameters.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/video_codecs/video_codec.h" +#include "call/rtp_transport_controller_send_interface.h" +#include "call/video_send_stream.h" +#include "modules/pacing/pacing_controller.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/trace_event.h" +#include "system_wrappers/include/clock.h" +#include "system_wrappers/include/field_trial.h" + +namespace webrtc { +namespace internal { +namespace { + +// Max positive size difference to treat allocations as "similar". +static constexpr int kMaxVbaSizeDifferencePercent = 10; +// Max time we will throttle similar video bitrate allocations. +static constexpr int64_t kMaxVbaThrottleTimeMs = 500; + +constexpr TimeDelta kEncoderTimeOut = TimeDelta::Seconds(2); + +constexpr double kVideoHysteresis = 1.2; +constexpr double kScreenshareHysteresis = 1.35; + +// 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& 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& 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 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( + 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( + 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 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; +} + +// Returns an optional that has value iff TransportSeqNumExtensionConfigured +// is `true` for the given video send stream config. +absl::optional GetConfiguredPacingFactor( + const VideoSendStream::Config& config, + VideoEncoderConfig::ContentType content_type, + const PacingConfig& default_pacing_config) { + if (!TransportSeqNumExtensionConfigured(config)) + return absl::nullopt; + + absl::optional alr_settings = + GetAlrSettings(content_type); + if (alr_settings) + return alr_settings->pacing_factor; + + RateControlSettings rate_control_settings = + RateControlSettings::ParseFromFieldTrials(); + return rate_control_settings.GetPacingFactor().value_or( + default_pacing_config.pacing_factor); +} + +uint32_t GetInitialEncoderMaxBitrate(int initial_encoder_max_bitrate) { + if (initial_encoder_max_bitrate > 0) + return rtc::dchecked_cast(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; +} + +} // 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( + Clock* clock, + SendStatisticsProxy* stats_proxy, + RtpTransportControllerSendInterface* transport, + BitrateAllocatorInterface* bitrate_allocator, + VideoStreamEncoderInterface* video_stream_encoder, + const VideoSendStream::Config* config, + int initial_encoder_max_bitrate, + double initial_encoder_bitrate_priority, + VideoEncoderConfig::ContentType content_type, + RtpVideoSenderInterface* rtp_video_sender, + const FieldTrialsView& field_trials) + : clock_(clock), + has_alr_probing_(config->periodic_alr_bandwidth_probing || + GetAlrSettings(content_type)), + pacing_config_(PacingConfig(field_trials)), + stats_proxy_(stats_proxy), + config_(config), + rtp_transport_queue_(transport->GetWorkerQueue()), + timed_out_(false), + transport_(transport), + bitrate_allocator_(bitrate_allocator), + disable_padding_(true), + max_padding_bitrate_(0), + encoder_min_bitrate_bps_(0), + encoder_max_bitrate_bps_( + GetInitialEncoderMaxBitrate(initial_encoder_max_bitrate)), + encoder_target_rate_bps_(0), + encoder_bitrate_priority_(initial_encoder_bitrate_priority), + video_stream_encoder_(video_stream_encoder), + bandwidth_observer_(transport->GetBandwidthObserver()), + rtp_video_sender_(rtp_video_sender), + configured_pacing_factor_( + GetConfiguredPacingFactor(*config_, content_type, pacing_config_)) { + 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(initial_encoder_max_bitrate, 0); + RTC_LOG(LS_INFO) << "VideoSendStreamImpl: " << config_->ToString(); + + RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled()); + + // 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); + + absl::optional 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 alr_settings = + GetAlrSettings(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::ParseFromFieldTrials(); + 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); + } + + rtp_transport_queue_->RunOrPost(SafeTask(transport_queue_safety_, [this] { + if (configured_pacing_factor_) + transport_->SetPacingFactor(*configured_pacing_factor_); + + video_stream_encoder_->SetStartBitrate( + bitrate_allocator_->GetStartBitrate(this)); + })); +} + +VideoSendStreamImpl::~VideoSendStreamImpl() { + RTC_DCHECK_RUN_ON(&thread_checker_); + RTC_LOG(LS_INFO) << "~VideoSendStreamImpl: " << config_->ToString(); + // TODO(webrtc:14502): Change `transport_queue_safety_` to be of type + // ScopedTaskSafety if experiment WebRTC-SendPacketsOnWorkerThread succeed. + if (rtp_transport_queue_->IsCurrent()) { + transport_queue_safety_->SetNotAlive(); + } +} + +void VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) { + // Runs on a worker thread. + rtp_video_sender_->DeliverRtcp(packet, length); +} + +void VideoSendStreamImpl::StartPerRtpStream( + const std::vector active_layers) { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + bool previously_active = rtp_video_sender_->IsActive(); + rtp_video_sender_->SetActiveModules(active_layers); + if (!rtp_video_sender_->IsActive() && previously_active) { + StopVideoSendStream(); + } else if (rtp_video_sender_->IsActive() && !previously_active) { + StartupVideoSendStream(); + } +} + +void VideoSendStreamImpl::StartupVideoSendStream() { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + transport_queue_safety_->SetAlive(); + + 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( + rtp_transport_queue_->TaskQueueForDelayedTasks(), kEncoderTimeOut, + [this] { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + if (!activity_) { + if (!timed_out_) { + SignalEncoderTimedOut(); + } + timed_out_ = true; + disable_padding_ = true; + } else if (timed_out_) { + SignalEncoderActive(); + timed_out_ = false; + } + activity_ = false; + return kEncoderTimeOut; + }); + } + + video_stream_encoder_->SendKeyFrame(); +} + +void VideoSendStreamImpl::Stop() { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + RTC_LOG(LS_INFO) << "VideoSendStreamImpl::Stop"; + if (!rtp_video_sender_->IsActive()) + return; + + RTC_DCHECK(transport_queue_safety_->alive()); + TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop"); + rtp_video_sender_->Stop(); + StopVideoSendStream(); +} + +void VideoSendStreamImpl::StopVideoSendStream() { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + 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); + transport_queue_safety_->SetNotAlive(); +} + +void VideoSendStreamImpl::SignalEncoderTimedOut() { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + // If the encoder has not produced anything the last kEncoderTimeOut and it + // is supposed to, deregister as BitrateAllocatorObserver. This can happen + // if a camera stops producing frames. + if (encoder_target_rate_bps_ > 0) { + RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out."; + bitrate_allocator_->RemoveObserver(this); + } +} + +void VideoSendStreamImpl::OnBitrateAllocationUpdated( + const VideoBitrateAllocation& allocation) { + // OnBitrateAllocationUpdated is invoked from the encoder task queue or + // the rtp_transport_queue_. + auto task = [=] { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + if (encoder_target_rate_bps_ == 0) { + return; + } + int64_t now_ms = 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 (!rtp_transport_queue_->IsCurrent()) { + rtp_transport_queue_->TaskQueueForPost()->PostTask( + SafeTask(transport_queue_safety_, 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(rtp_transport_queue_); + if (rtp_video_sender_->IsActive()) { + RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active."; + bitrate_allocator_->AddObserver(this, GetAllocationConfig()); + } +} + +MediaStreamAllocationConfig VideoSendStreamImpl::GetAllocationConfig() const { + return MediaStreamAllocationConfig{ + static_cast(encoder_min_bitrate_bps_), + encoder_max_bitrate_bps_, + static_cast(disable_padding_ ? 0 : max_padding_bitrate_), + /* priority_bitrate */ 0, + !config_->suspend_below_min_bitrate, + encoder_bitrate_priority_}; +} + +void VideoSendStreamImpl::OnEncoderConfigurationChanged( + std::vector streams, + bool is_svc, + VideoEncoderConfig::ContentType content_type, + int min_transmit_bitrate_bps) { + // Currently called on the encoder TQ + RTC_DCHECK(!rtp_transport_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(rtp_transport_queue_); + + const VideoCodecType codec_type = + PayloadStringToCodecType(config_->rtp.payload_name); + + const absl::optional experimental_min_bitrate = + GetExperimentalMinVideoBitrate(codec_type); + encoder_min_bitrate_bps_ = + experimental_min_bitrate + ? experimental_min_bitrate->bps() + : std::max(streams[0].min_bitrate_bps, kDefaultMinVideoBitrateBps); + + encoder_max_bitrate_bps_ = 0; + double stream_bitrate_priority_sum = 0; + for (const auto& stream : streams) { + // We don't want to allocate more bitrate than needed to inactive streams. + encoder_max_bitrate_bps_ += stream.active ? stream.max_bitrate_bps : 0; + if (stream.bitrate_priority) { + RTC_DCHECK_GT(*stream.bitrate_priority, 0); + stream_bitrate_priority_sum += *stream.bitrate_priority; + } + } + RTC_DCHECK_GT(stream_bitrate_priority_sum, 0); + encoder_bitrate_priority_ = stream_bitrate_priority_sum; + encoder_max_bitrate_bps_ = + std::max(static_cast(encoder_min_bitrate_bps_), + encoder_max_bitrate_bps_); + + // TODO(bugs.webrtc.org/10266): Query the VideoBitrateAllocator instead. + max_padding_bitrate_ = CalculateMaxPadBitrateBps( + streams, is_svc, content_type, min_transmit_bitrate_bps, + config_->suspend_below_min_bitrate, has_alr_probing_); + + // Clear stats for disabled layers. + for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) { + stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]); + } + + const size_t num_temporal_layers = + streams.back().num_temporal_layers.value_or(1); + + rtp_video_sender_->SetEncodingData(streams[0].width, streams[0].height, + num_temporal_layers); + + if (rtp_video_sender_->IsActive()) { + // The send stream is started already. Update the allocator with new + // bitrate limits. + bitrate_allocator_->AddObserver(this, GetAllocationConfig()); + } + }; + + rtp_transport_queue_->TaskQueueForPost()->PostTask( + SafeTask(transport_queue_safety_, 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(!rtp_transport_queue_->IsCurrent()); + + auto task_to_run_on_worker = [this]() { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + 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); + } + }; + rtp_transport_queue_->TaskQueueForPost()->PostTask( + SafeTask(transport_queue_safety_, 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 VideoSendStreamImpl::GetRtpStates() const { + return rtp_video_sender_->GetRtpStates(); +} + +std::map VideoSendStreamImpl::GetRtpPayloadStates() + const { + return rtp_video_sender_->GetRtpPayloadStates(); +} + +uint32_t VideoSendStreamImpl::OnBitrateUpdated(BitrateAllocationUpdate update) { + RTC_DCHECK_RUN_ON(rtp_transport_queue_); + RTC_DCHECK(rtp_video_sender_->IsActive()) + << "VideoSendStream::Start has not been called."; + + // When the BWE algorithm doesn't pass a stable estimate, we'll use the + // unstable one instead. + if (update.stable_target_bitrate.IsZero()) { + update.stable_target_bitrate = update.target_bitrate; + } + + rtp_video_sender_->OnBitrateUpdated(update, stats_proxy_->GetSendFrameRate()); + encoder_target_rate_bps_ = rtp_video_sender_->GetPayloadBitrateBps(); + const uint32_t protection_bitrate_bps = + rtp_video_sender_->GetProtectionBitrateBps(); + DataRate link_allocation = DataRate::Zero(); + if (encoder_target_rate_bps_ > protection_bitrate_bps) { + link_allocation = + DataRate::BitsPerSec(encoder_target_rate_bps_ - protection_bitrate_bps); + } + DataRate overhead = + update.target_bitrate - DataRate::BitsPerSec(encoder_target_rate_bps_); + DataRate encoder_stable_target_rate = update.stable_target_bitrate; + if (encoder_stable_target_rate > overhead) { + encoder_stable_target_rate = encoder_stable_target_rate - overhead; + } else { + encoder_stable_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_); + } + + encoder_target_rate_bps_ = + std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_); + + encoder_stable_target_rate = + std::min(DataRate::BitsPerSec(encoder_max_bitrate_bps_), + encoder_stable_target_rate); + + DataRate encoder_target_rate = DataRate::BitsPerSec(encoder_target_rate_bps_); + link_allocation = std::max(encoder_target_rate, link_allocation); + video_stream_encoder_->OnBitrateUpdated( + encoder_target_rate, encoder_stable_target_rate, link_allocation, + rtc::dchecked_cast(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 -- cgit v1.2.3