Repo created

TMessagesProj/jni/voip/webrtc/video/video_send_stream_impl.cc

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+/*
+ *  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/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 "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(
+    Clock* clock,
+    int num_cpu_cores,
+    TaskQueueFactory* task_queue_factory,
+    SendStatisticsProxy* stats_proxy,
+    const VideoStreamEncoderSettings& encoder_settings,
+    VideoStreamEncoder::BitrateAllocationCallbackType
+        bitrate_allocation_callback_type,
+    const FieldTrialsView& field_trials,
+    Metronome* metronome,
+    webrtc::VideoEncoderFactory::EncoderSelectorInterface* encoder_selector) {
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> encoder_queue =
+      task_queue_factory->CreateTaskQueue("EncoderQueue",
+                                          TaskQueueFactory::Priority::NORMAL);
+  TaskQueueBase* encoder_queue_ptr = encoder_queue.get();
+  return std::make_unique<VideoStreamEncoder>(
+      clock, num_cpu_cores, stats_proxy, encoder_settings,
+      std::make_unique<OveruseFrameDetector>(stats_proxy),
+      FrameCadenceAdapterInterface::Create(
+          clock, encoder_queue_ptr, metronome,
+          /*worker_queue=*/TaskQueueBase::Current(), field_trials),
+      std::move(encoder_queue), bitrate_allocation_callback_type, field_trials,
+      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(
+    Clock* clock,
+    int num_cpu_cores,
+    TaskQueueFactory* task_queue_factory,
+    RtcpRttStats* call_stats,
+    RtpTransportControllerSendInterface* transport,
+    Metronome* metronome,
+    BitrateAllocatorInterface* bitrate_allocator,
+    SendDelayStats* send_delay_stats,
+    RtcEventLog* event_log,
+    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,
+    const FieldTrialsView& field_trials,
+    std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_for_test)
+    : transport_(transport),
+      stats_proxy_(clock, config, encoder_config.content_type, 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(
+                    clock,
+                    num_cpu_cores,
+                    task_queue_factory,
+                    &stats_proxy_,
+                    config_.encoder_settings,
+                    GetBitrateAllocationCallbackType(config_, field_trials),
+                    field_trials,
+                    metronome,
+                    config_.encoder_selector)),
+      encoder_feedback_(
+          clock,
+          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_),
+          event_log,
+          std::move(fec_controller),
+          CreateFrameEncryptionConfig(&config_),
+          config_.frame_transformer)),
+      clock_(clock),
+      has_alr_probing_(
+          config_.periodic_alr_bandwidth_probing ||
+          GetAlrSettings(field_trials, encoder_config.content_type)),
+      pacing_config_(PacingConfig(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, field_trials)),
+      configured_pacing_factor_(GetConfiguredPacingFactor(config_,
+                                                          content_type_,
+                                                          pacing_config_,
+                                                          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(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(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(&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");
+  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 = 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(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