s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine

webrtc/media/engine/webrtcvideoengine2.cc

-/*
- *  Copyright (c) 2014 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 "webrtc/media/engine/webrtcvideoengine2.h"
-
-#include <stdio.h>
-#include <algorithm>
-#include <set>
-#include <string>
-#include <utility>
-
-#include "webrtc/api/video/i420_buffer.h"
-#include "webrtc/api/video_codecs/video_decoder.h"
-#include "webrtc/api/video_codecs/video_encoder.h"
-#include "webrtc/base/copyonwritebuffer.h"
-#include "webrtc/base/logging.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/timeutils.h"
-#include "webrtc/base/trace_event.h"
-#include "webrtc/call/call.h"
-#include "webrtc/common_video/h264/profile_level_id.h"
-#include "webrtc/media/engine/constants.h"
-#include "webrtc/media/engine/internalencoderfactory.h"
-#include "webrtc/media/engine/internaldecoderfactory.h"
-#include "webrtc/media/engine/simulcast.h"
-#include "webrtc/media/engine/videoencodersoftwarefallbackwrapper.h"
-#include "webrtc/media/engine/videodecodersoftwarefallbackwrapper.h"
-#include "webrtc/media/engine/webrtcmediaengine.h"
-#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
-#include "webrtc/media/engine/webrtcvoiceengine.h"
-#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
-
-using DegradationPreference = webrtc::VideoSendStream::DegradationPreference;
-
-namespace cricket {
-namespace {
-// If this field trial is enabled, we will enable sending FlexFEC and disable
-// sending ULPFEC whenever the former has been negotiated in the SDPs.
-bool IsFlexfecFieldTrialEnabled() {
-  return webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03");
-}
-
-// If this field trial is enabled, the "flexfec-03" codec may have been
-// advertised as being supported in the local SDP. That means that we must be
-// ready to receive FlexFEC packets. See internalencoderfactory.cc.
-bool IsFlexfecAdvertisedFieldTrialEnabled() {
-  return webrtc::field_trial::IsEnabled("WebRTC-FlexFEC-03-Advertised");
-}
-
-// If this field trial is enabled, we will report VideoContentType RTP extension
-// in capabilities (thus, it will end up in the default SDP and extension will
-// be sent for all key-frames).
-bool IsVideoContentTypeExtensionFieldTrialEnabled() {
-  return webrtc::field_trial::IsEnabled("WebRTC-VideoContentTypeExtension");
-}
-
-// Wrap cricket::WebRtcVideoEncoderFactory as a webrtc::VideoEncoderFactory.
-class EncoderFactoryAdapter : public webrtc::VideoEncoderFactory {
- public:
-  // EncoderFactoryAdapter doesn't take ownership of |factory|, which is owned
-  // by e.g. PeerConnectionFactory.
-  explicit EncoderFactoryAdapter(cricket::WebRtcVideoEncoderFactory* factory)
-      : factory_(factory) {}
-  virtual ~EncoderFactoryAdapter() {}
-
-  // Implement webrtc::VideoEncoderFactory.
-  webrtc::VideoEncoder* Create() override {
-    return factory_->CreateVideoEncoder(VideoCodec(kVp8CodecName));
-  }
-
-  void Destroy(webrtc::VideoEncoder* encoder) override {
-    return factory_->DestroyVideoEncoder(encoder);
-  }
-
- private:
-  cricket::WebRtcVideoEncoderFactory* const factory_;
-};
-
-// An encoder factory that wraps Create requests for simulcastable codec types
-// with a webrtc::SimulcastEncoderAdapter. Non simulcastable codec type
-// requests are just passed through to the contained encoder factory.
-class WebRtcSimulcastEncoderFactory
-    : public cricket::WebRtcVideoEncoderFactory {
- public:
-  // WebRtcSimulcastEncoderFactory doesn't take ownership of |factory|, which is
-  // owned by e.g. PeerConnectionFactory.
-  explicit WebRtcSimulcastEncoderFactory(
-      cricket::WebRtcVideoEncoderFactory* factory)
-      : factory_(factory) {}
-
-  static bool UseSimulcastEncoderFactory(
-      const std::vector<cricket::VideoCodec>& codecs) {
-    // If any codec is VP8, use the simulcast factory. If asked to create a
-    // non-VP8 codec, we'll just return a contained factory encoder directly.
-    for (const auto& codec : codecs) {
-      if (CodecNamesEq(codec.name.c_str(), kVp8CodecName)) {
-        return true;
-      }
-    }
-    return false;
-  }
-
-  webrtc::VideoEncoder* CreateVideoEncoder(
-      const cricket::VideoCodec& codec) override {
-    RTC_DCHECK(factory_ != NULL);
-    // If it's a codec type we can simulcast, create a wrapped encoder.
-    if (CodecNamesEq(codec.name.c_str(), kVp8CodecName)) {
-      return new webrtc::SimulcastEncoderAdapter(
-          new EncoderFactoryAdapter(factory_));
-    }
-    webrtc::VideoEncoder* encoder = factory_->CreateVideoEncoder(codec);
-    if (encoder) {
-      non_simulcast_encoders_.push_back(encoder);
-    }
-    return encoder;
-  }
-
-  const std::vector<cricket::VideoCodec>& supported_codecs() const override {
-    return factory_->supported_codecs();
-  }
-
-  bool EncoderTypeHasInternalSource(
-      webrtc::VideoCodecType type) const override {
-    return factory_->EncoderTypeHasInternalSource(type);
-  }
-
-  void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override {
-    // Check first to see if the encoder wasn't wrapped in a
-    // SimulcastEncoderAdapter. In that case, ask the factory to destroy it.
-    if (std::remove(non_simulcast_encoders_.begin(),
-                    non_simulcast_encoders_.end(),
-                    encoder) != non_simulcast_encoders_.end()) {
-      factory_->DestroyVideoEncoder(encoder);
-      return;
-    }
-
-    // Otherwise, SimulcastEncoderAdapter can be deleted directly, and will call
-    // DestroyVideoEncoder on the factory for individual encoder instances.
-    delete encoder;
-  }
-
- private:
-  cricket::WebRtcVideoEncoderFactory* factory_;
-  // A list of encoders that were created without being wrapped in a
-  // SimulcastEncoderAdapter.
-  std::vector<webrtc::VideoEncoder*> non_simulcast_encoders_;
-};
-
-void AddDefaultFeedbackParams(VideoCodec* codec) {
-  codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamCcm, kRtcpFbCcmParamFir));
-  codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kParamValueEmpty));
-  codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
-  codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
-  codec->AddFeedbackParam(
-      FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
-}
-
-static std::string CodecVectorToString(const std::vector<VideoCodec>& codecs) {
-  std::stringstream out;
-  out << '{';
-  for (size_t i = 0; i < codecs.size(); ++i) {
-    out << codecs[i].ToString();
-    if (i != codecs.size() - 1) {
-      out << ", ";
-    }
-  }
-  out << '}';
-  return out.str();
-}
-
-static bool ValidateCodecFormats(const std::vector<VideoCodec>& codecs) {
-  bool has_video = false;
-  for (size_t i = 0; i < codecs.size(); ++i) {
-    if (!codecs[i].ValidateCodecFormat()) {
-      return false;
-    }
-    if (codecs[i].GetCodecType() == VideoCodec::CODEC_VIDEO) {
-      has_video = true;
-    }
-  }
-  if (!has_video) {
-    LOG(LS_ERROR) << "Setting codecs without a video codec is invalid: "
-                  << CodecVectorToString(codecs);
-    return false;
-  }
-  return true;
-}
-
-static bool ValidateStreamParams(const StreamParams& sp) {
-  if (sp.ssrcs.empty()) {
-    LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
-    return false;
-  }
-
-  std::vector<uint32_t> primary_ssrcs;
-  sp.GetPrimarySsrcs(&primary_ssrcs);
-  std::vector<uint32_t> rtx_ssrcs;
-  sp.GetFidSsrcs(primary_ssrcs, &rtx_ssrcs);
-  for (uint32_t rtx_ssrc : rtx_ssrcs) {
-    bool rtx_ssrc_present = false;
-    for (uint32_t sp_ssrc : sp.ssrcs) {
-      if (sp_ssrc == rtx_ssrc) {
-        rtx_ssrc_present = true;
-        break;
-      }
-    }
-    if (!rtx_ssrc_present) {
-      LOG(LS_ERROR) << "RTX SSRC '" << rtx_ssrc
-                    << "' missing from StreamParams ssrcs: " << sp.ToString();
-      return false;
-    }
-  }
-  if (!rtx_ssrcs.empty() && primary_ssrcs.size() != rtx_ssrcs.size()) {
-    LOG(LS_ERROR)
-        << "RTX SSRCs exist, but don't cover all SSRCs (unsupported): "
-        << sp.ToString();
-    return false;
-  }
-
-  return true;
-}
-
-// Returns true if the given codec is disallowed from doing simulcast.
-bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
-  return CodecNamesEq(codec_name, kH264CodecName) ||
-         CodecNamesEq(codec_name, kVp9CodecName);
-}
-
-// The selected thresholds for QVGA and VGA corresponded to a QP around 10.
-// The change in QP declined above the selected bitrates.
-static int GetMaxDefaultVideoBitrateKbps(int width, int height) {
-  if (width * height <= 320 * 240) {
-    return 600;
-  } else if (width * height <= 640 * 480) {
-    return 1700;
-  } else if (width * height <= 960 * 540) {
-    return 2000;
-  } else {
-    return 2500;
-  }
-}
-
-bool GetVp9LayersFromFieldTrialGroup(int* num_spatial_layers,
-                                     int* num_temporal_layers) {
-  std::string group = webrtc::field_trial::FindFullName("WebRTC-SupportVP9SVC");
-  if (group.empty())
-    return false;
-
-  if (sscanf(group.c_str(), "EnabledByFlag_%dSL%dTL", num_spatial_layers,
-             num_temporal_layers) != 2) {
-    return false;
-  }
-  const int kMaxSpatialLayers = 2;
-  if (*num_spatial_layers > kMaxSpatialLayers || *num_spatial_layers < 1)
-    return false;
-
-  const int kMaxTemporalLayers = 3;
-  if (*num_temporal_layers > kMaxTemporalLayers || *num_temporal_layers < 1)
-    return false;
-
-  return true;
-}
-
-int GetDefaultVp9SpatialLayers() {
-  int num_sl;
-  int num_tl;
-  if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
-    return num_sl;
-  }
-  return 1;
-}
-
-int GetDefaultVp9TemporalLayers() {
-  int num_sl;
-  int num_tl;
-  if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
-    return num_tl;
-  }
-  return 1;
-}
-
-class EncoderStreamFactory
-    : public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
- public:
-  EncoderStreamFactory(std::string codec_name,
-                       int max_qp,
-                       int max_framerate,
-                       bool is_screencast,
-                       bool conference_mode)
-      : codec_name_(codec_name),
-        max_qp_(max_qp),
-        max_framerate_(max_framerate),
-        is_screencast_(is_screencast),
-        conference_mode_(conference_mode) {}
-
- private:
-  std::vector<webrtc::VideoStream> CreateEncoderStreams(
-      int width,
-      int height,
-      const webrtc::VideoEncoderConfig& encoder_config) override {
-    if (is_screencast_ &&
-        (!conference_mode_ || !cricket::UseSimulcastScreenshare())) {
-      RTC_DCHECK_EQ(1, encoder_config.number_of_streams);
-    }
-    if (encoder_config.number_of_streams > 1 ||
-        (CodecNamesEq(codec_name_, kVp8CodecName) && is_screencast_ &&
-         conference_mode_)) {
-      return GetSimulcastConfig(encoder_config.number_of_streams, width, height,
-                                encoder_config.max_bitrate_bps, max_qp_,
-                                max_framerate_, is_screencast_);
-    }
-
-    // For unset max bitrates set default bitrate for non-simulcast.
-    int max_bitrate_bps =
-        (encoder_config.max_bitrate_bps > 0)
-            ? encoder_config.max_bitrate_bps
-            : GetMaxDefaultVideoBitrateKbps(width, height) * 1000;
-
-    webrtc::VideoStream stream;
-    stream.width = width;
-    stream.height = height;
-    stream.max_framerate = max_framerate_;
-    stream.min_bitrate_bps = kMinVideoBitrateKbps * 1000;
-    stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps;
-    stream.max_qp = max_qp_;
-
-    if (CodecNamesEq(codec_name_, kVp9CodecName) && !is_screencast_) {
-      stream.temporal_layer_thresholds_bps.resize(
-          GetDefaultVp9TemporalLayers() - 1);
-    }
-
-    std::vector<webrtc::VideoStream> streams;
-    streams.push_back(stream);
-    return streams;
-  }
-
-  const std::string codec_name_;
-  const int max_qp_;
-  const int max_framerate_;
-  const bool is_screencast_;
-  const bool conference_mode_;
-};
-
-}  // namespace
-
-// Constants defined in webrtc/media/engine/constants.h
-// TODO(pbos): Move these to a separate constants.cc file.
-const int kMinVideoBitrateKbps = 30;
-
-const int kVideoMtu = 1200;
-const int kVideoRtpBufferSize = 65536;
-
-// This constant is really an on/off, lower-level configurable NACK history
-// duration hasn't been implemented.
-static const int kNackHistoryMs = 1000;
-
-static const int kDefaultQpMax = 56;
-
-static const int kDefaultRtcpReceiverReportSsrc = 1;
-
-// Minimum time interval for logging stats.
-static const int64_t kStatsLogIntervalMs = 10000;
-
-static std::vector<VideoCodec> GetSupportedCodecs(
-    const WebRtcVideoEncoderFactory* external_encoder_factory);
-
-rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
-WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
-    const VideoCodec& codec) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  bool is_screencast = parameters_.options.is_screencast.value_or(false);
-  // No automatic resizing when using simulcast or screencast.
-  bool automatic_resize =
-      !is_screencast && parameters_.config.rtp.ssrcs.size() == 1;
-  bool frame_dropping = !is_screencast;
-  bool denoising;
-  bool codec_default_denoising = false;
-  if (is_screencast) {
-    denoising = false;
-  } else {
-    // Use codec default if video_noise_reduction is unset.
-    codec_default_denoising = !parameters_.options.video_noise_reduction;
-    denoising = parameters_.options.video_noise_reduction.value_or(false);
-  }
-
-  if (CodecNamesEq(codec.name, kH264CodecName)) {
-    webrtc::VideoCodecH264 h264_settings =
-        webrtc::VideoEncoder::GetDefaultH264Settings();
-    h264_settings.frameDroppingOn = frame_dropping;
-    return new rtc::RefCountedObject<
-        webrtc::VideoEncoderConfig::H264EncoderSpecificSettings>(h264_settings);
-  }
-  if (CodecNamesEq(codec.name, kVp8CodecName)) {
-    webrtc::VideoCodecVP8 vp8_settings =
-        webrtc::VideoEncoder::GetDefaultVp8Settings();
-    vp8_settings.automaticResizeOn = automatic_resize;
-    // VP8 denoising is enabled by default.
-    vp8_settings.denoisingOn = codec_default_denoising ? true : denoising;
-    vp8_settings.frameDroppingOn = frame_dropping;
-    return new rtc::RefCountedObject<
-        webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
-  }
-  if (CodecNamesEq(codec.name, kVp9CodecName)) {
-    webrtc::VideoCodecVP9 vp9_settings =
-        webrtc::VideoEncoder::GetDefaultVp9Settings();
-    if (is_screencast) {
-      // TODO(asapersson): Set to 2 for now since there is a DCHECK in
-      // VideoSendStream::ReconfigureVideoEncoder.
-      vp9_settings.numberOfSpatialLayers = 2;
-    } else {
-      vp9_settings.numberOfSpatialLayers = GetDefaultVp9SpatialLayers();
-    }
-    // VP9 denoising is disabled by default.
-    vp9_settings.denoisingOn = codec_default_denoising ? true : denoising;
-    vp9_settings.frameDroppingOn = frame_dropping;
-    vp9_settings.automaticResizeOn = automatic_resize;
-    return new rtc::RefCountedObject<
-        webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
-  }
-  return nullptr;
-}
-
-DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
-    : default_sink_(nullptr) {}
-
-UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
-    WebRtcVideoChannel2* channel,
-    uint32_t ssrc) {
-  rtc::Optional<uint32_t> default_recv_ssrc =
-      channel->GetDefaultReceiveStreamSsrc();
-
-  if (default_recv_ssrc) {
-    LOG(LS_INFO) << "Destroying old default receive stream for SSRC=" << ssrc
-                 << ".";
-    channel->RemoveRecvStream(*default_recv_ssrc);
-  }
-
-  StreamParams sp;
-  sp.ssrcs.push_back(ssrc);
-  LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
-  if (!channel->AddRecvStream(sp, true)) {
-    LOG(LS_WARNING) << "Could not create default receive stream.";
-  }
-
-  channel->SetSink(ssrc, default_sink_);
-  return kDeliverPacket;
-}
-
-rtc::VideoSinkInterface<webrtc::VideoFrame>*
-DefaultUnsignalledSsrcHandler::GetDefaultSink() const {
-  return default_sink_;
-}
-
-void DefaultUnsignalledSsrcHandler::SetDefaultSink(
-    WebRtcVideoChannel2* channel,
-    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
-  default_sink_ = sink;
-  rtc::Optional<uint32_t> default_recv_ssrc =
-      channel->GetDefaultReceiveStreamSsrc();
-  if (default_recv_ssrc) {
-    channel->SetSink(*default_recv_ssrc, default_sink_);
-  }
-}
-
-WebRtcVideoEngine2::WebRtcVideoEngine2()
-    : initialized_(false),
-      external_decoder_factory_(NULL),
-      external_encoder_factory_(NULL) {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
-}
-
-WebRtcVideoEngine2::~WebRtcVideoEngine2() {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
-}
-
-void WebRtcVideoEngine2::Init() {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
-  initialized_ = true;
-}
-
-WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
-    webrtc::Call* call,
-    const MediaConfig& config,
-    const VideoOptions& options) {
-  RTC_DCHECK(initialized_);
-  LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString();
-  return new WebRtcVideoChannel2(call, config, options,
-                                 external_encoder_factory_,
-                                 external_decoder_factory_);
-}
-
-std::vector<VideoCodec> WebRtcVideoEngine2::codecs() const {
-  return GetSupportedCodecs(external_encoder_factory_);
-}
-
-RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
-  RtpCapabilities capabilities;
-  capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kTimestampOffsetUri,
-                           webrtc::RtpExtension::kTimestampOffsetDefaultId));
-  capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
-                           webrtc::RtpExtension::kAbsSendTimeDefaultId));
-  capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kVideoRotationUri,
-                           webrtc::RtpExtension::kVideoRotationDefaultId));
-  capabilities.header_extensions.push_back(webrtc::RtpExtension(
-      webrtc::RtpExtension::kTransportSequenceNumberUri,
-      webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
-  capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kPlayoutDelayUri,
-                           webrtc::RtpExtension::kPlayoutDelayDefaultId));
-  if (IsVideoContentTypeExtensionFieldTrialEnabled()) {
-    capabilities.header_extensions.push_back(
-        webrtc::RtpExtension(webrtc::RtpExtension::kVideoContentTypeUri,
-                             webrtc::RtpExtension::kVideoContentTypeDefaultId));
-  }
-  return capabilities;
-}
-
-void WebRtcVideoEngine2::SetExternalDecoderFactory(
-    WebRtcVideoDecoderFactory* decoder_factory) {
-  RTC_DCHECK(!initialized_);
-  external_decoder_factory_ = decoder_factory;
-}
-
-void WebRtcVideoEngine2::SetExternalEncoderFactory(
-    WebRtcVideoEncoderFactory* encoder_factory) {
-  RTC_DCHECK(!initialized_);
-  if (external_encoder_factory_ == encoder_factory)
-    return;
-
-  // No matter what happens we shouldn't hold on to a stale
-  // WebRtcSimulcastEncoderFactory.
-  simulcast_encoder_factory_.reset();
-
-  if (encoder_factory &&
-      WebRtcSimulcastEncoderFactory::UseSimulcastEncoderFactory(
-          encoder_factory->supported_codecs())) {
-    simulcast_encoder_factory_.reset(
-        new WebRtcSimulcastEncoderFactory(encoder_factory));
-    encoder_factory = simulcast_encoder_factory_.get();
-  }
-  external_encoder_factory_ = encoder_factory;
-}
-
-// This is a helper function for AppendVideoCodecs below. It will return the
-// first unused dynamic payload type (in the range [96, 127]), or nothing if no
-// payload type is unused.
-static rtc::Optional<int> NextFreePayloadType(
-    const std::vector<VideoCodec>& codecs) {
-  static const int kFirstDynamicPayloadType = 96;
-  static const int kLastDynamicPayloadType = 127;
-  bool is_payload_used[1 + kLastDynamicPayloadType - kFirstDynamicPayloadType] =
-      {false};
-  for (const VideoCodec& codec : codecs) {
-    if (kFirstDynamicPayloadType <= codec.id &&
-        codec.id <= kLastDynamicPayloadType) {
-      is_payload_used[codec.id - kFirstDynamicPayloadType] = true;
-    }
-  }
-  for (int i = kFirstDynamicPayloadType; i <= kLastDynamicPayloadType; ++i) {
-    if (!is_payload_used[i - kFirstDynamicPayloadType])
-      return rtc::Optional<int>(i);
-  }
-  // No free payload type.
-  return rtc::Optional<int>();
-}
-
-// This is a helper function for GetSupportedCodecs below. It will append new
-// unique codecs from |input_codecs| to |unified_codecs|. It will add default
-// feedback params to the codecs and will also add an associated RTX codec for
-// recognized codecs (VP8, VP9, H264, and RED).
-static void AppendVideoCodecs(const std::vector<VideoCodec>& input_codecs,
-                              std::vector<VideoCodec>* unified_codecs) {
-  for (VideoCodec codec : input_codecs) {
-    const rtc::Optional<int> payload_type =
-        NextFreePayloadType(*unified_codecs);
-    if (!payload_type)
-      return;
-    codec.id = *payload_type;
-    // TODO(magjed): Move the responsibility of setting these parameters to the
-    // encoder factories instead.
-    if (codec.name != kRedCodecName && codec.name != kUlpfecCodecName &&
-        codec.name != kFlexfecCodecName)
-      AddDefaultFeedbackParams(&codec);
-    // Don't add same codec twice.
-    if (FindMatchingCodec(*unified_codecs, codec))
-      continue;
-
-    unified_codecs->push_back(codec);
-
-    // Add associated RTX codec for recognized codecs.
-    // TODO(deadbeef): Should we add RTX codecs for external codecs whose names
-    // we don't recognize?
-    if (CodecNamesEq(codec.name, kVp8CodecName) ||
-        CodecNamesEq(codec.name, kVp9CodecName) ||
-        CodecNamesEq(codec.name, kH264CodecName) ||
-        CodecNamesEq(codec.name, kRedCodecName)) {
-      const rtc::Optional<int> rtx_payload_type =
-          NextFreePayloadType(*unified_codecs);
-      if (!rtx_payload_type)
-        return;
-      unified_codecs->push_back(
-          VideoCodec::CreateRtxCodec(*rtx_payload_type, codec.id));
-    }
-  }
-}
-
-static std::vector<VideoCodec> GetSupportedCodecs(
-    const WebRtcVideoEncoderFactory* external_encoder_factory) {
-  const std::vector<VideoCodec> internal_codecs =
-      InternalEncoderFactory().supported_codecs();
-  LOG(LS_INFO) << "Internally supported codecs: "
-               << CodecVectorToString(internal_codecs);
-
-  std::vector<VideoCodec> unified_codecs;
-  AppendVideoCodecs(internal_codecs, &unified_codecs);
-
-  if (external_encoder_factory != nullptr) {
-    const std::vector<VideoCodec>& external_codecs =
-        external_encoder_factory->supported_codecs();
-    AppendVideoCodecs(external_codecs, &unified_codecs);
-    LOG(LS_INFO) << "Codecs supported by the external encoder factory: "
-                 << CodecVectorToString(external_codecs);
-  }
-
-  return unified_codecs;
-}
-
-WebRtcVideoChannel2::WebRtcVideoChannel2(
-    webrtc::Call* call,
-    const MediaConfig& config,
-    const VideoOptions& options,
-    WebRtcVideoEncoderFactory* external_encoder_factory,
-    WebRtcVideoDecoderFactory* external_decoder_factory)
-    : VideoMediaChannel(config),
-      call_(call),
-      unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
-      video_config_(config.video),
-      external_encoder_factory_(external_encoder_factory),
-      external_decoder_factory_(external_decoder_factory),
-      default_send_options_(options),
-      last_stats_log_ms_(-1) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-
-  rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
-  sending_ = false;
-  recv_codecs_ = MapCodecs(GetSupportedCodecs(external_encoder_factory));
-  recv_flexfec_payload_type_ = recv_codecs_.front().flexfec_payload_type;
-}
-
-WebRtcVideoChannel2::~WebRtcVideoChannel2() {
-  for (auto& kv : send_streams_)
-    delete kv.second;
-  for (auto& kv : receive_streams_)
-    delete kv.second;
-}
-
-rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>
-WebRtcVideoChannel2::SelectSendVideoCodec(
-    const std::vector<VideoCodecSettings>& remote_mapped_codecs) const {
-  const std::vector<VideoCodec> local_supported_codecs =
-      GetSupportedCodecs(external_encoder_factory_);
-  // Select the first remote codec that is supported locally.
-  for (const VideoCodecSettings& remote_mapped_codec : remote_mapped_codecs) {
-    // For H264, we will limit the encode level to the remote offered level
-    // regardless if level asymmetry is allowed or not. This is strictly not
-    // following the spec in https://tools.ietf.org/html/rfc6184#section-8.2.2
-    // since we should limit the encode level to the lower of local and remote
-    // level when level asymmetry is not allowed.
-    if (FindMatchingCodec(local_supported_codecs, remote_mapped_codec.codec))
-      return rtc::Optional<VideoCodecSettings>(remote_mapped_codec);
-  }
-  // No remote codec was supported.
-  return rtc::Optional<VideoCodecSettings>();
-}
-
-bool WebRtcVideoChannel2::NonFlexfecReceiveCodecsHaveChanged(
-    std::vector<VideoCodecSettings> before,
-    std::vector<VideoCodecSettings> after) {
-  if (before.size() != after.size()) {
-    return true;
-  }
-
-  // The receive codec order doesn't matter, so we sort the codecs before
-  // comparing. This is necessary because currently the
-  // only way to change the send codec is to munge SDP, which causes
-  // the receive codec list to change order, which causes the streams
-  // to be recreates which causes a "blink" of black video.  In order
-  // to support munging the SDP in this way without recreating receive
-  // streams, we ignore the order of the received codecs so that
-  // changing the order doesn't cause this "blink".
-  auto comparison =
-      [](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) {
-        return codec1.codec.id > codec2.codec.id;
-      };
-  std::sort(before.begin(), before.end(), comparison);
-  std::sort(after.begin(), after.end(), comparison);
-
-  // Changes in FlexFEC payload type are handled separately in
-  // WebRtcVideoChannel2::GetChangedRecvParameters, so disregard FlexFEC in the
-  // comparison here.
-  return !std::equal(before.begin(), before.end(), after.begin(),
-                     VideoCodecSettings::EqualsDisregardingFlexfec);
-}
-
-bool WebRtcVideoChannel2::GetChangedSendParameters(
-    const VideoSendParameters& params,
-    ChangedSendParameters* changed_params) const {
-  if (!ValidateCodecFormats(params.codecs) ||
-      !ValidateRtpExtensions(params.extensions)) {
-    return false;
-  }
-
-  // Select one of the remote codecs that will be used as send codec.
-  rtc::Optional<VideoCodecSettings> selected_send_codec =
-      SelectSendVideoCodec(MapCodecs(params.codecs));
-
-  if (!selected_send_codec) {
-    LOG(LS_ERROR) << "No video codecs supported.";
-    return false;
-  }
-
-  // Never enable sending FlexFEC, unless we are in the experiment.
-  if (!IsFlexfecFieldTrialEnabled()) {
-    if (selected_send_codec->flexfec_payload_type != -1) {
-      LOG(LS_INFO) << "Remote supports flexfec-03, but we will not send since "
-                   << "WebRTC-FlexFEC-03 field trial is not enabled.";
-    }
-    selected_send_codec->flexfec_payload_type = -1;
-  }
-
-  if (!send_codec_ || *selected_send_codec != *send_codec_)
-    changed_params->codec = selected_send_codec;
-
-  // Handle RTP header extensions.
-  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
-      params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
-  if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
-    changed_params->rtp_header_extensions =
-        rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
-  }
-
-  // Handle max bitrate.
-  if (params.max_bandwidth_bps != send_params_.max_bandwidth_bps &&
-      params.max_bandwidth_bps >= -1) {
-    // 0 or -1 uncaps max bitrate.
-    // TODO(pbos): Reconsider how 0 should be treated. It is not mentioned as a
-    // special value and might very well be used for stopping sending.
-    changed_params->max_bandwidth_bps = rtc::Optional<int>(
-        params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
-  }
-
-  // Handle conference mode.
-  if (params.conference_mode != send_params_.conference_mode) {
-    changed_params->conference_mode =
-        rtc::Optional<bool>(params.conference_mode);
-  }
-
-  // Handle RTCP mode.
-  if (params.rtcp.reduced_size != send_params_.rtcp.reduced_size) {
-    changed_params->rtcp_mode = rtc::Optional<webrtc::RtcpMode>(
-        params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
-                                 : webrtc::RtcpMode::kCompound);
-  }
-
-  return true;
-}
-
-rtc::DiffServCodePoint WebRtcVideoChannel2::PreferredDscp() const {
-  return rtc::DSCP_AF41;
-}
-
-bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
-  LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
-  ChangedSendParameters changed_params;
-  if (!GetChangedSendParameters(params, &changed_params)) {
-    return false;
-  }
-
-  if (changed_params.codec) {
-    const VideoCodecSettings& codec_settings = *changed_params.codec;
-    send_codec_ = rtc::Optional<VideoCodecSettings>(codec_settings);
-    LOG(LS_INFO) << "Using codec: " << codec_settings.codec.ToString();
-  }
-
-  if (changed_params.rtp_header_extensions) {
-    send_rtp_extensions_ = changed_params.rtp_header_extensions;
-  }
-
-  if (changed_params.codec || changed_params.max_bandwidth_bps) {
-    if (params.max_bandwidth_bps == -1) {
-      // Unset the global max bitrate (max_bitrate_bps) if max_bandwidth_bps is
-      // -1, which corresponds to no "b=AS" attribute in SDP. Note that the
-      // global max bitrate may be set below in GetBitrateConfigForCodec, from
-      // the codec max bitrate.
-      // TODO(pbos): This should be reconsidered (codec max bitrate should
-      // probably not affect global call max bitrate).
-      bitrate_config_.max_bitrate_bps = -1;
-    }
-    if (send_codec_) {
-      // TODO(holmer): Changing the codec parameters shouldn't necessarily mean
-      // that we change the min/max of bandwidth estimation. Reevaluate this.
-      bitrate_config_ = GetBitrateConfigForCodec(send_codec_->codec);
-      if (!changed_params.codec) {
-        // If the codec isn't changing, set the start bitrate to -1 which means
-        // "unchanged" so that BWE isn't affected.
-        bitrate_config_.start_bitrate_bps = -1;
-      }
-    }
-    if (params.max_bandwidth_bps >= 0) {
-      // Note that max_bandwidth_bps intentionally takes priority over the
-      // bitrate config for the codec. This allows FEC to be applied above the
-      // codec target bitrate.
-      // TODO(pbos): Figure out whether b=AS means max bitrate for this
-      // WebRtcVideoChannel2 (in which case we're good), or per sender (SSRC),
-      // in which case this should not set a Call::BitrateConfig but rather
-      // reconfigure all senders.
-      bitrate_config_.max_bitrate_bps =
-          params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
-    }
-    call_->SetBitrateConfig(bitrate_config_);
-  }
-
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    for (auto& kv : send_streams_) {
-      kv.second->SetSendParameters(changed_params);
-    }
-    if (changed_params.codec || changed_params.rtcp_mode) {
-      // Update receive feedback parameters from new codec or RTCP mode.
-      LOG(LS_INFO)
-          << "SetFeedbackOptions on all the receive streams because the send "
-             "codec or RTCP mode has changed.";
-      for (auto& kv : receive_streams_) {
-        RTC_DCHECK(kv.second != nullptr);
-        kv.second->SetFeedbackParameters(
-            HasNack(send_codec_->codec), HasRemb(send_codec_->codec),
-            HasTransportCc(send_codec_->codec),
-            params.rtcp.reduced_size ? webrtc::RtcpMode::kReducedSize
-                                     : webrtc::RtcpMode::kCompound);
-      }
-    }
-  }
-  send_params_ = params;
-  return true;
-}
-
-webrtc::RtpParameters WebRtcVideoChannel2::GetRtpSendParameters(
-    uint32_t ssrc) const {
-  rtc::CritScope stream_lock(&stream_crit_);
-  auto it = send_streams_.find(ssrc);
-  if (it == send_streams_.end()) {
-    LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
-                    << "with ssrc " << ssrc << " which doesn't exist.";
-    return webrtc::RtpParameters();
-  }
-
-  webrtc::RtpParameters rtp_params = it->second->GetRtpParameters();
-  // Need to add the common list of codecs to the send stream-specific
-  // RTP parameters.
-  for (const VideoCodec& codec : send_params_.codecs) {
-    rtp_params.codecs.push_back(codec.ToCodecParameters());
-  }
-  return rtp_params;
-}
-
-bool WebRtcVideoChannel2::SetRtpSendParameters(
-    uint32_t ssrc,
-    const webrtc::RtpParameters& parameters) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpSendParameters");
-  rtc::CritScope stream_lock(&stream_crit_);
-  auto it = send_streams_.find(ssrc);
-  if (it == send_streams_.end()) {
-    LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream "
-                  << "with ssrc " << ssrc << " which doesn't exist.";
-    return false;
-  }
-
-  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
-  // different order (which should change the send codec).
-  webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
-  if (current_parameters.codecs != parameters.codecs) {
-    LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
-                  << "is not currently supported.";
-    return false;
-  }
-
-  return it->second->SetRtpParameters(parameters);
-}
-
-webrtc::RtpParameters WebRtcVideoChannel2::GetRtpReceiveParameters(
-    uint32_t ssrc) const {
-  webrtc::RtpParameters rtp_params;
-  rtc::CritScope stream_lock(&stream_crit_);
-  // SSRC of 0 represents an unsignaled receive stream.
-  if (ssrc == 0) {
-    if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) {
-      LOG(LS_WARNING) << "Attempting to get RTP parameters for the default, "
-                         "unsignaled video receive stream, but not yet "
-                         "configured to receive such a stream.";
-      return rtp_params;
-    }
-    rtp_params.encodings.emplace_back();
-  } else {
-    auto it = receive_streams_.find(ssrc);
-    if (it == receive_streams_.end()) {
-      LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
-                      << "with SSRC " << ssrc << " which doesn't exist.";
-      return webrtc::RtpParameters();
-    }
-    // TODO(deadbeef): Return stream-specific parameters, beyond just SSRC.
-    rtp_params.encodings.emplace_back();
-    rtp_params.encodings[0].ssrc = it->second->GetFirstPrimarySsrc();
-  }
-
-  // Add codecs, which any stream is prepared to receive.
-  for (const VideoCodec& codec : recv_params_.codecs) {
-    rtp_params.codecs.push_back(codec.ToCodecParameters());
-  }
-  return rtp_params;
-}
-
-bool WebRtcVideoChannel2::SetRtpReceiveParameters(
-    uint32_t ssrc,
-    const webrtc::RtpParameters& parameters) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRtpReceiveParameters");
-  rtc::CritScope stream_lock(&stream_crit_);
-
-  // SSRC of 0 represents an unsignaled receive stream.
-  if (ssrc == 0) {
-    if (!default_unsignalled_ssrc_handler_.GetDefaultSink()) {
-      LOG(LS_WARNING) << "Attempting to set RTP parameters for the default, "
-                         "unsignaled video receive stream, but not yet "
-                         "configured to receive such a stream.";
-      return false;
-    }
-  } else {
-    auto it = receive_streams_.find(ssrc);
-    if (it == receive_streams_.end()) {
-      LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
-                      << "with SSRC " << ssrc << " which doesn't exist.";
-      return false;
-    }
-  }
-
-  webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
-  if (current_parameters != parameters) {
-    LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
-                  << "unsupported.";
-    return false;
-  }
-  return true;
-}
-
-bool WebRtcVideoChannel2::GetChangedRecvParameters(
-    const VideoRecvParameters& params,
-    ChangedRecvParameters* changed_params) const {
-  if (!ValidateCodecFormats(params.codecs) ||
-      !ValidateRtpExtensions(params.extensions)) {
-    return false;
-  }
-
-  // Handle receive codecs.
-  const std::vector<VideoCodecSettings> mapped_codecs =
-      MapCodecs(params.codecs);
-  if (mapped_codecs.empty()) {
-    LOG(LS_ERROR) << "SetRecvParameters called without any video codecs.";
-    return false;
-  }
-
-  // Verify that every mapped codec is supported locally.
-  const std::vector<VideoCodec> local_supported_codecs =
-      GetSupportedCodecs(external_encoder_factory_);
-  for (const VideoCodecSettings& mapped_codec : mapped_codecs) {
-    if (!FindMatchingCodec(local_supported_codecs, mapped_codec.codec)) {
-      LOG(LS_ERROR) << "SetRecvParameters called with unsupported video codec: "
-                    << mapped_codec.codec.ToString();
-      return false;
-    }
-  }
-
-  if (NonFlexfecReceiveCodecsHaveChanged(recv_codecs_, mapped_codecs)) {
-    changed_params->codec_settings =
-        rtc::Optional<std::vector<VideoCodecSettings>>(mapped_codecs);
-  }
-
-  // Handle RTP header extensions.
-  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
-      params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
-  if (filtered_extensions != recv_rtp_extensions_) {
-    changed_params->rtp_header_extensions =
-        rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
-  }
-
-  int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type;
-  if (flexfec_payload_type != recv_flexfec_payload_type_) {
-    changed_params->flexfec_payload_type =
-        rtc::Optional<int>(flexfec_payload_type);
-  }
-
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvParameters");
-  LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
-  ChangedRecvParameters changed_params;
-  if (!GetChangedRecvParameters(params, &changed_params)) {
-    return false;
-  }
-  if (changed_params.flexfec_payload_type) {
-    LOG(LS_INFO) << "Changing FlexFEC payload type (recv) from "
-                 << recv_flexfec_payload_type_ << " to "
-                 << *changed_params.flexfec_payload_type;
-    recv_flexfec_payload_type_ = *changed_params.flexfec_payload_type;
-  }
-  if (changed_params.rtp_header_extensions) {
-    recv_rtp_extensions_ = *changed_params.rtp_header_extensions;
-  }
-  if (changed_params.codec_settings) {
-    LOG(LS_INFO) << "Changing recv codecs from "
-                 << CodecSettingsVectorToString(recv_codecs_) << " to "
-                 << CodecSettingsVectorToString(*changed_params.codec_settings);
-    recv_codecs_ = *changed_params.codec_settings;
-  }
-
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    for (auto& kv : receive_streams_) {
-      kv.second->SetRecvParameters(changed_params);
-    }
-  }
-  recv_params_ = params;
-  return true;
-}
-
-std::string WebRtcVideoChannel2::CodecSettingsVectorToString(
-    const std::vector<VideoCodecSettings>& codecs) {
-  std::stringstream out;
-  out << '{';
-  for (size_t i = 0; i < codecs.size(); ++i) {
-    out << codecs[i].codec.ToString();
-    if (i != codecs.size() - 1) {
-      out << ", ";
-    }
-  }
-  out << '}';
-  return out.str();
-}
-
-bool WebRtcVideoChannel2::GetSendCodec(VideoCodec* codec) {
-  if (!send_codec_) {
-    LOG(LS_VERBOSE) << "GetSendCodec: No send codec set.";
-    return false;
-  }
-  *codec = send_codec_->codec;
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetSend(bool send) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSend");
-  LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
-  if (send && !send_codec_) {
-    LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
-    return false;
-  }
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    for (const auto& kv : send_streams_) {
-      kv.second->SetSend(send);
-    }
-  }
-  sending_ = send;
-  return true;
-}
-
-// TODO(nisse): The enable argument was used for mute logic which has
-// been moved to VideoBroadcaster. So remove the argument from this
-// method.
-bool WebRtcVideoChannel2::SetVideoSend(
-    uint32_t ssrc,
-    bool enable,
-    const VideoOptions* options,
-    rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
-  TRACE_EVENT0("webrtc", "SetVideoSend");
-  RTC_DCHECK(ssrc != 0);
-  LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable
-               << ", options: " << (options ? options->ToString() : "nullptr")
-               << ", source = " << (source ? "(source)" : "nullptr") << ")";
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  const auto& kv = send_streams_.find(ssrc);
-  if (kv == send_streams_.end()) {
-    // Allow unknown ssrc only if source is null.
-    RTC_CHECK(source == nullptr);
-    LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-    return false;
-  }
-
-  return kv->second->SetVideoSend(enable, options, source);
-}
-
-bool WebRtcVideoChannel2::ValidateSendSsrcAvailability(
-    const StreamParams& sp) const {
-  for (uint32_t ssrc : sp.ssrcs) {
-    if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) {
-      LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists.";
-      return false;
-    }
-  }
-  return true;
-}
-
-bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability(
-    const StreamParams& sp) const {
-  for (uint32_t ssrc : sp.ssrcs) {
-    if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) {
-      LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc
-                    << "' already exists.";
-      return false;
-    }
-  }
-  return true;
-}
-
-bool WebRtcVideoChannel2::AddSendStream(const StreamParams& sp) {
-  LOG(LS_INFO) << "AddSendStream: " << sp.ToString();
-  if (!ValidateStreamParams(sp))
-    return false;
-
-  rtc::CritScope stream_lock(&stream_crit_);
-
-  if (!ValidateSendSsrcAvailability(sp))
-    return false;
-
-  for (uint32_t used_ssrc : sp.ssrcs)
-    send_ssrcs_.insert(used_ssrc);
-
-  webrtc::VideoSendStream::Config config(this);
-  config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate;
-  config.periodic_alr_bandwidth_probing =
-      video_config_.periodic_alr_bandwidth_probing;
-  WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
-      call_, sp, std::move(config), default_send_options_,
-      external_encoder_factory_, video_config_.enable_cpu_overuse_detection,
-      bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
-      send_params_);
-
-  uint32_t ssrc = sp.first_ssrc();
-  RTC_DCHECK(ssrc != 0);
-  send_streams_[ssrc] = stream;
-
-  if (rtcp_receiver_report_ssrc_ == kDefaultRtcpReceiverReportSsrc) {
-    rtcp_receiver_report_ssrc_ = ssrc;
-    LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because we added "
-                    "a send stream.";
-    for (auto& kv : receive_streams_)
-      kv.second->SetLocalSsrc(ssrc);
-  }
-  if (sending_) {
-    stream->SetSend(true);
-  }
-
-  return true;
-}
-
-bool WebRtcVideoChannel2::RemoveSendStream(uint32_t ssrc) {
-  LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
-
-  WebRtcVideoSendStream* removed_stream;
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-        send_streams_.find(ssrc);
-    if (it == send_streams_.end()) {
-      return false;
-    }
-
-    for (uint32_t old_ssrc : it->second->GetSsrcs())
-      send_ssrcs_.erase(old_ssrc);
-
-    removed_stream = it->second;
-    send_streams_.erase(it);
-
-    // Switch receiver report SSRCs, the one in use is no longer valid.
-    if (rtcp_receiver_report_ssrc_ == ssrc) {
-      rtcp_receiver_report_ssrc_ = send_streams_.empty()
-                                       ? kDefaultRtcpReceiverReportSsrc
-                                       : send_streams_.begin()->first;
-      LOG(LS_INFO) << "SetLocalSsrc on all the receive streams because the "
-                      "previous local SSRC was removed.";
-
-      for (auto& kv : receive_streams_) {
-        kv.second->SetLocalSsrc(rtcp_receiver_report_ssrc_);
-      }
-    }
-  }
-
-  delete removed_stream;
-
-  return true;
-}
-
-void WebRtcVideoChannel2::DeleteReceiveStream(
-    WebRtcVideoChannel2::WebRtcVideoReceiveStream* stream) {
-  for (uint32_t old_ssrc : stream->GetSsrcs())
-    receive_ssrcs_.erase(old_ssrc);
-  delete stream;
-}
-
-bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp) {
-  return AddRecvStream(sp, false);
-}
-
-bool WebRtcVideoChannel2::AddRecvStream(const StreamParams& sp,
-                                        bool default_stream) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-
-  LOG(LS_INFO) << "AddRecvStream" << (default_stream ? " (default stream)" : "")
-               << ": " << sp.ToString();
-  if (!ValidateStreamParams(sp))
-    return false;
-
-  uint32_t ssrc = sp.first_ssrc();
-  RTC_DCHECK(ssrc != 0);  // TODO(pbos): Is this ever valid?
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  // Remove running stream if this was a default stream.
-  const auto& prev_stream = receive_streams_.find(ssrc);
-  if (prev_stream != receive_streams_.end()) {
-    if (default_stream || !prev_stream->second->IsDefaultStream()) {
-      LOG(LS_ERROR) << "Receive stream for SSRC '" << ssrc
-                    << "' already exists.";
-      return false;
-    }
-    DeleteReceiveStream(prev_stream->second);
-    receive_streams_.erase(prev_stream);
-  }
-
-  if (!ValidateReceiveSsrcAvailability(sp))
-    return false;
-
-  for (uint32_t used_ssrc : sp.ssrcs)
-    receive_ssrcs_.insert(used_ssrc);
-
-  webrtc::VideoReceiveStream::Config config(this);
-  webrtc::FlexfecReceiveStream::Config flexfec_config(this);
-  ConfigureReceiverRtp(&config, &flexfec_config, sp);
-
-  config.disable_prerenderer_smoothing =
-      video_config_.disable_prerenderer_smoothing;
-  config.sync_group = sp.sync_label;
-
-  receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
-      call_, sp, std::move(config), external_decoder_factory_, default_stream,
-      recv_codecs_, flexfec_config);
-
-  return true;
-}
-
-void WebRtcVideoChannel2::ConfigureReceiverRtp(
-    webrtc::VideoReceiveStream::Config* config,
-    webrtc::FlexfecReceiveStream::Config* flexfec_config,
-    const StreamParams& sp) const {
-  uint32_t ssrc = sp.first_ssrc();
-
-  config->rtp.remote_ssrc = ssrc;
-  config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
-
-  // TODO(pbos): This protection is against setting the same local ssrc as
-  // remote which is not permitted by the lower-level API. RTCP requires a
-  // corresponding sender SSRC. Figure out what to do when we don't have
-  // (receive-only) or know a good local SSRC.
-  if (config->rtp.remote_ssrc == config->rtp.local_ssrc) {
-    if (config->rtp.local_ssrc != kDefaultRtcpReceiverReportSsrc) {
-      config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc;
-    } else {
-      config->rtp.local_ssrc = kDefaultRtcpReceiverReportSsrc + 1;
-    }
-  }
-
-  // Whether or not the receive stream sends reduced size RTCP is determined
-  // by the send params.
-  // TODO(deadbeef): Once we change "send_params" to "sender_params" and
-  // "recv_params" to "receiver_params", we should get this out of
-  // receiver_params_.
-  config->rtp.rtcp_mode = send_params_.rtcp.reduced_size
-                              ? webrtc::RtcpMode::kReducedSize
-                              : webrtc::RtcpMode::kCompound;
-
-  config->rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
-  config->rtp.transport_cc =
-      send_codec_ ? HasTransportCc(send_codec_->codec) : false;
-
-  sp.GetFidSsrc(ssrc, &config->rtp.rtx_ssrc);
-
-  config->rtp.extensions = recv_rtp_extensions_;
-
-  // TODO(brandtr): Generalize when we add support for multistream protection.
-  flexfec_config->payload_type = recv_flexfec_payload_type_;
-  if (IsFlexfecAdvertisedFieldTrialEnabled() &&
-      sp.GetFecFrSsrc(ssrc, &flexfec_config->remote_ssrc)) {
-    flexfec_config->protected_media_ssrcs = {ssrc};
-    flexfec_config->local_ssrc = config->rtp.local_ssrc;
-    flexfec_config->rtcp_mode = config->rtp.rtcp_mode;
-    // TODO(brandtr): We should be spec-compliant and set |transport_cc| here
-    // based on the rtcp-fb for the FlexFEC codec, not the media codec.
-    flexfec_config->transport_cc = config->rtp.transport_cc;
-    flexfec_config->rtp_header_extensions = config->rtp.extensions;
-  }
-}
-
-bool WebRtcVideoChannel2::RemoveRecvStream(uint32_t ssrc) {
-  LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
-  if (ssrc == 0) {
-    LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
-    return false;
-  }
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator stream =
-      receive_streams_.find(ssrc);
-  if (stream == receive_streams_.end()) {
-    LOG(LS_ERROR) << "Stream not found for ssrc: " << ssrc;
-    return false;
-  }
-  DeleteReceiveStream(stream->second);
-  receive_streams_.erase(stream);
-
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetSink(
-    uint32_t ssrc,
-    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
-  LOG(LS_INFO) << "SetSink: ssrc:" << ssrc << " "
-               << (sink ? "(ptr)" : "nullptr");
-  if (ssrc == 0) {
-    // Do not hold |stream_crit_| here, since SetDefaultSink will call
-    // WebRtcVideoChannel2::GetDefaultReceiveStreamSsrc().
-    default_unsignalled_ssrc_handler_.SetDefaultSink(this, sink);
-    return true;
-  }
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
-      receive_streams_.find(ssrc);
-  if (it == receive_streams_.end()) {
-    return false;
-  }
-
-  it->second->SetSink(sink);
-  return true;
-}
-
-bool WebRtcVideoChannel2::GetStats(VideoMediaInfo* info) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::GetStats");
-
-  // Log stats periodically.
-  bool log_stats = false;
-  int64_t now_ms = rtc::TimeMillis();
-  if (last_stats_log_ms_ == -1 ||
-      now_ms - last_stats_log_ms_ > kStatsLogIntervalMs) {
-    last_stats_log_ms_ = now_ms;
-    log_stats = true;
-  }
-
-  info->Clear();
-  FillSenderStats(info, log_stats);
-  FillReceiverStats(info, log_stats);
-  FillSendAndReceiveCodecStats(info);
-  // TODO(holmer): We should either have rtt available as a metric on
-  // VideoSend/ReceiveStreams, or we should remove rtt from VideoSenderInfo.
-  webrtc::Call::Stats stats = call_->GetStats();
-  if (stats.rtt_ms != -1) {
-    for (size_t i = 0; i < info->senders.size(); ++i) {
-      info->senders[i].rtt_ms = stats.rtt_ms;
-    }
-  }
-
-  if (log_stats)
-    LOG(LS_INFO) << stats.ToString(now_ms);
-
-  return true;
-}
-
-void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info,
-                                          bool log_stats) {
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-           send_streams_.begin();
-       it != send_streams_.end(); ++it) {
-    video_media_info->senders.push_back(
-        it->second->GetVideoSenderInfo(log_stats));
-  }
-}
-
-void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info,
-                                            bool log_stats) {
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
-           receive_streams_.begin();
-       it != receive_streams_.end(); ++it) {
-    video_media_info->receivers.push_back(
-        it->second->GetVideoReceiverInfo(log_stats));
-  }
-}
-
-void WebRtcVideoChannel2::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream =
-           send_streams_.begin();
-       stream != send_streams_.end(); ++stream) {
-    stream->second->FillBitrateInfo(bwe_info);
-  }
-}
-
-void WebRtcVideoChannel2::FillSendAndReceiveCodecStats(
-    VideoMediaInfo* video_media_info) {
-  for (const VideoCodec& codec : send_params_.codecs) {
-    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
-    video_media_info->send_codecs.insert(
-        std::make_pair(codec_params.payload_type, std::move(codec_params)));
-  }
-  for (const VideoCodec& codec : recv_params_.codecs) {
-    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
-    video_media_info->receive_codecs.insert(
-        std::make_pair(codec_params.payload_type, std::move(codec_params)));
-  }
-}
-
-void WebRtcVideoChannel2::OnPacketReceived(
-    rtc::CopyOnWriteBuffer* packet,
-    const rtc::PacketTime& packet_time) {
-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
-                                              packet_time.not_before);
-  const webrtc::PacketReceiver::DeliveryStatus delivery_result =
-      call_->Receiver()->DeliverPacket(
-          webrtc::MediaType::VIDEO,
-          packet->cdata(), packet->size(),
-          webrtc_packet_time);
-  switch (delivery_result) {
-    case webrtc::PacketReceiver::DELIVERY_OK:
-      return;
-    case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
-      return;
-    case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
-      break;
-  }
-
-  uint32_t ssrc = 0;
-  if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
-    return;
-  }
-
-  int payload_type = 0;
-  if (!GetRtpPayloadType(packet->cdata(), packet->size(), &payload_type)) {
-    return;
-  }
-
-  // See if this payload_type is registered as one that usually gets its own
-  // SSRC (RTX) or at least is safe to drop either way (FEC). If it is, and
-  // it wasn't handled above by DeliverPacket, that means we don't know what
-  // stream it associates with, and we shouldn't ever create an implicit channel
-  // for these.
-  for (auto& codec : recv_codecs_) {
-    if (payload_type == codec.rtx_payload_type ||
-        payload_type == codec.ulpfec.red_rtx_payload_type ||
-        payload_type == codec.ulpfec.ulpfec_payload_type) {
-      return;
-    }
-  }
-  if (payload_type == recv_flexfec_payload_type_) {
-    return;
-  }
-
-  switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
-    case UnsignalledSsrcHandler::kDropPacket:
-      return;
-    case UnsignalledSsrcHandler::kDeliverPacket:
-      break;
-  }
-
-  if (call_->Receiver()->DeliverPacket(
-          webrtc::MediaType::VIDEO,
-          packet->cdata(), packet->size(),
-          webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
-    LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
-    return;
-  }
-}
-
-void WebRtcVideoChannel2::OnRtcpReceived(
-    rtc::CopyOnWriteBuffer* packet,
-    const rtc::PacketTime& packet_time) {
-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
-                                              packet_time.not_before);
-  // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
-  // for both audio and video on the same path. Since BundleFilter doesn't
-  // filter RTCP anymore incoming RTCP packets could've been going to audio (so
-  // logging failures spam the log).
-  call_->Receiver()->DeliverPacket(
-      webrtc::MediaType::VIDEO,
-      packet->cdata(), packet->size(),
-      webrtc_packet_time);
-}
-
-void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
-  LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
-  call_->SignalChannelNetworkState(
-      webrtc::MediaType::VIDEO,
-      ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
-}
-
-void WebRtcVideoChannel2::OnNetworkRouteChanged(
-    const std::string& transport_name,
-    const rtc::NetworkRoute& network_route) {
-  call_->OnNetworkRouteChanged(transport_name, network_route);
-}
-
-void WebRtcVideoChannel2::OnTransportOverheadChanged(
-    int transport_overhead_per_packet) {
-  call_->OnTransportOverheadChanged(webrtc::MediaType::VIDEO,
-                                    transport_overhead_per_packet);
-}
-
-void WebRtcVideoChannel2::SetInterface(NetworkInterface* iface) {
-  MediaChannel::SetInterface(iface);
-  // Set the RTP recv/send buffer to a bigger size
-  MediaChannel::SetOption(NetworkInterface::ST_RTP,
-                          rtc::Socket::OPT_RCVBUF,
-                          kVideoRtpBufferSize);
-
-  // Speculative change to increase the outbound socket buffer size.
-  // In b/15152257, we are seeing a significant number of packets discarded
-  // due to lack of socket buffer space, although it's not yet clear what the
-  // ideal value should be.
-  MediaChannel::SetOption(NetworkInterface::ST_RTP,
-                          rtc::Socket::OPT_SNDBUF,
-                          kVideoRtpBufferSize);
-}
-
-rtc::Optional<uint32_t> WebRtcVideoChannel2::GetDefaultReceiveStreamSsrc() {
-  rtc::CritScope stream_lock(&stream_crit_);
-  rtc::Optional<uint32_t> ssrc;
-  for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) {
-    if (it->second->IsDefaultStream()) {
-      ssrc.emplace(it->first);
-      break;
-    }
-  }
-  return ssrc;
-}
-
-bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
-                                  size_t len,
-                                  const webrtc::PacketOptions& options) {
-  rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
-  rtc::PacketOptions rtc_options;
-  rtc_options.packet_id = options.packet_id;
-  return MediaChannel::SendPacket(&packet, rtc_options);
-}
-
-bool WebRtcVideoChannel2::SendRtcp(const uint8_t* data, size_t len) {
-  rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
-  return MediaChannel::SendRtcp(&packet, rtc::PacketOptions());
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
-    VideoSendStreamParameters(
-        webrtc::VideoSendStream::Config config,
-        const VideoOptions& options,
-        int max_bitrate_bps,
-        const rtc::Optional<VideoCodecSettings>& codec_settings)
-    : config(std::move(config)),
-      options(options),
-      max_bitrate_bps(max_bitrate_bps),
-      conference_mode(false),
-      codec_settings(codec_settings) {}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder(
-    webrtc::VideoEncoder* encoder,
-    const cricket::VideoCodec& codec,
-    bool external)
-    : encoder(encoder),
-      external_encoder(nullptr),
-      codec(codec),
-      external(external) {
-  if (external) {
-    external_encoder = encoder;
-    this->encoder =
-        new webrtc::VideoEncoderSoftwareFallbackWrapper(codec, encoder);
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
-    webrtc::Call* call,
-    const StreamParams& sp,
-    webrtc::VideoSendStream::Config config,
-    const VideoOptions& options,
-    WebRtcVideoEncoderFactory* external_encoder_factory,
-    bool enable_cpu_overuse_detection,
-    int max_bitrate_bps,
-    const rtc::Optional<VideoCodecSettings>& codec_settings,
-    const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
-    // TODO(deadbeef): Don't duplicate information between send_params,
-    // rtp_extensions, options, etc.
-    const VideoSendParameters& send_params)
-    : worker_thread_(rtc::Thread::Current()),
-      ssrcs_(sp.ssrcs),
-      ssrc_groups_(sp.ssrc_groups),
-      call_(call),
-      enable_cpu_overuse_detection_(enable_cpu_overuse_detection),
-      source_(nullptr),
-      external_encoder_factory_(external_encoder_factory),
-      internal_encoder_factory_(new InternalEncoderFactory()),
-      stream_(nullptr),
-      encoder_sink_(nullptr),
-      parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
-      rtp_parameters_(CreateRtpParametersWithOneEncoding()),
-      allocated_encoder_(nullptr, cricket::VideoCodec(), false),
-      sending_(false) {
-  parameters_.config.rtp.max_packet_size = kVideoMtu;
-  parameters_.conference_mode = send_params.conference_mode;
-
-  sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
-
-  // ValidateStreamParams should prevent this from happening.
-  RTC_CHECK(!parameters_.config.rtp.ssrcs.empty());
-  rtp_parameters_.encodings[0].ssrc =
-      rtc::Optional<uint32_t>(parameters_.config.rtp.ssrcs[0]);
-
-  // RTX.
-  sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
-                 &parameters_.config.rtp.rtx.ssrcs);
-
-  // FlexFEC SSRCs.
-  // TODO(brandtr): This code needs to be generalized when we add support for
-  // multistream protection.
-  if (IsFlexfecFieldTrialEnabled()) {
-    uint32_t flexfec_ssrc;
-    bool flexfec_enabled = false;
-    for (uint32_t primary_ssrc : parameters_.config.rtp.ssrcs) {
-      if (sp.GetFecFrSsrc(primary_ssrc, &flexfec_ssrc)) {
-        if (flexfec_enabled) {
-          LOG(LS_INFO) << "Multiple FlexFEC streams in local SDP, but "
-                          "our implementation only supports a single FlexFEC "
-                          "stream. Will not enable FlexFEC for proposed "
-                          "stream with SSRC: "
-                       << flexfec_ssrc << ".";
-          continue;
-        }
-
-        flexfec_enabled = true;
-        parameters_.config.rtp.flexfec.ssrc = flexfec_ssrc;
-        parameters_.config.rtp.flexfec.protected_media_ssrcs = {primary_ssrc};
-      }
-    }
-  }
-
-  parameters_.config.rtp.c_name = sp.cname;
-  if (rtp_extensions) {
-    parameters_.config.rtp.extensions = *rtp_extensions;
-  }
-  parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
-                                         ? webrtc::RtcpMode::kReducedSize
-                                         : webrtc::RtcpMode::kCompound;
-  if (codec_settings) {
-    bool force_encoder_allocation = false;
-    SetCodec(*codec_settings, force_encoder_allocation);
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
-  if (stream_ != NULL) {
-    call_->DestroyVideoSendStream(stream_);
-  }
-  DestroyVideoEncoder(&allocated_encoder_);
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoSend(
-    bool enable,
-    const VideoOptions* options,
-    rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend");
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-
-  // Ignore |options| pointer if |enable| is false.
-  bool options_present = enable && options;
-
-  if (options_present) {
-    VideoOptions old_options = parameters_.options;
-    parameters_.options.SetAll(*options);
-    if (parameters_.options.is_screencast.value_or(false) !=
-            old_options.is_screencast.value_or(false) &&
-        parameters_.codec_settings) {
-      // If screen content settings change, we may need to recreate the codec
-      // instance so that the correct type is used.
-
-      bool force_encoder_allocation = true;
-      SetCodec(*parameters_.codec_settings, force_encoder_allocation);
-      // Mark screenshare parameter as being updated, then test for any other
-      // changes that may require codec reconfiguration.
-      old_options.is_screencast = options->is_screencast;
-    }
-    if (parameters_.options != old_options) {
-      ReconfigureEncoder();
-    }
-  }
-
-  if (source_ && stream_) {
-    stream_->SetSource(nullptr, DegradationPreference::kDegradationDisabled);
-  }
-  // Switch to the new source.
-  source_ = source;
-  if (source && stream_) {
-    stream_->SetSource(this, GetDegradationPreference());
-  }
-  return true;
-}
-
-webrtc::VideoSendStream::DegradationPreference
-WebRtcVideoChannel2::WebRtcVideoSendStream::GetDegradationPreference() const {
-  // Do not adapt resolution for screen content as this will likely
-  // result in blurry and unreadable text.
-  // |this| acts like a VideoSource to make sure SinkWants are handled on the
-  // correct thread.
-  DegradationPreference degradation_preference;
-  if (!enable_cpu_overuse_detection_) {
-    degradation_preference = DegradationPreference::kDegradationDisabled;
-  } else {
-    if (parameters_.options.is_screencast.value_or(false)) {
-      degradation_preference = DegradationPreference::kMaintainResolution;
-    } else {
-      degradation_preference = DegradationPreference::kMaintainFramerate;
-    }
-  }
-  return degradation_preference;
-}
-
-const std::vector<uint32_t>&
-WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
-  return ssrcs_;
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder
-WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
-    const VideoCodec& codec,
-    bool force_encoder_allocation) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  // Do not re-create encoders of the same type.
-  if (!force_encoder_allocation && codec == allocated_encoder_.codec &&
-      allocated_encoder_.encoder != nullptr) {
-    return allocated_encoder_;
-  }
-
-  // Try creating external encoder.
-  if (external_encoder_factory_ != nullptr &&
-      FindMatchingCodec(external_encoder_factory_->supported_codecs(), codec)) {
-    webrtc::VideoEncoder* encoder =
-        external_encoder_factory_->CreateVideoEncoder(codec);
-    if (encoder != nullptr)
-      return AllocatedEncoder(encoder, codec, true /* is_external */);
-  }
-
-  // Try creating internal encoder.
-  if (FindMatchingCodec(internal_encoder_factory_->supported_codecs(), codec)) {
-    if (parameters_.encoder_config.content_type ==
-            webrtc::VideoEncoderConfig::ContentType::kScreen &&
-        parameters_.conference_mode && UseSimulcastScreenshare()) {
-      // TODO(sprang): Remove this adapter once libvpx supports simulcast with
-      // same-resolution substreams.
-      WebRtcSimulcastEncoderFactory adapter_factory(
-          internal_encoder_factory_.get());
-      return AllocatedEncoder(adapter_factory.CreateVideoEncoder(codec), codec,
-                              false /* is_external */);
-    }
-    return AllocatedEncoder(
-        internal_encoder_factory_->CreateVideoEncoder(codec), codec,
-        false /* is_external */);
-  }
-
-  // This shouldn't happen, we should not be trying to create something we don't
-  // support.
-  RTC_NOTREACHED();
-  return AllocatedEncoder(NULL, cricket::VideoCodec(), false);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
-    AllocatedEncoder* encoder) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (encoder->external) {
-    external_encoder_factory_->DestroyVideoEncoder(encoder->external_encoder);
-  }
-  delete encoder->encoder;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
-    const VideoCodecSettings& codec_settings,
-    bool force_encoder_allocation) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec);
-  RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0);
-
-  AllocatedEncoder new_encoder =
-      CreateVideoEncoder(codec_settings.codec, force_encoder_allocation);
-  parameters_.config.encoder_settings.encoder = new_encoder.encoder;
-  parameters_.config.encoder_settings.full_overuse_time = new_encoder.external;
-  parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
-  parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
-  if (new_encoder.external) {
-    webrtc::VideoCodecType type =
-        webrtc::PayloadNameToCodecType(codec_settings.codec.name)
-            .value_or(webrtc::kVideoCodecUnknown);
-    parameters_.config.encoder_settings.internal_source =
-        external_encoder_factory_->EncoderTypeHasInternalSource(type);
-  } else {
-    parameters_.config.encoder_settings.internal_source = false;
-  }
-  parameters_.config.rtp.ulpfec = codec_settings.ulpfec;
-  parameters_.config.rtp.flexfec.payload_type =
-      codec_settings.flexfec_payload_type;
-
-  // Set RTX payload type if RTX is enabled.
-  if (!parameters_.config.rtp.rtx.ssrcs.empty()) {
-    if (codec_settings.rtx_payload_type == -1) {
-      LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
-                         "payload type. Ignoring.";
-      parameters_.config.rtp.rtx.ssrcs.clear();
-    } else {
-      parameters_.config.rtp.rtx.payload_type = codec_settings.rtx_payload_type;
-    }
-  }
-
-  parameters_.config.rtp.nack.rtp_history_ms =
-      HasNack(codec_settings.codec) ? kNackHistoryMs : 0;
-
-  parameters_.codec_settings =
-      rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings);
-
-  LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec.";
-  RecreateWebRtcStream();
-  if (allocated_encoder_.encoder != new_encoder.encoder) {
-    DestroyVideoEncoder(&allocated_encoder_);
-    allocated_encoder_ = new_encoder;
-  }
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
-    const ChangedSendParameters& params) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  // |recreate_stream| means construction-time parameters have changed and the
-  // sending stream needs to be reset with the new config.
-  bool recreate_stream = false;
-  if (params.rtcp_mode) {
-    parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
-    recreate_stream = true;
-  }
-  if (params.rtp_header_extensions) {
-    parameters_.config.rtp.extensions = *params.rtp_header_extensions;
-    recreate_stream = true;
-  }
-  if (params.max_bandwidth_bps) {
-    parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
-    ReconfigureEncoder();
-  }
-  if (params.conference_mode) {
-    parameters_.conference_mode = *params.conference_mode;
-  }
-
-  // Set codecs and options.
-  if (params.codec) {
-    bool force_encoder_allocation = false;
-    SetCodec(*params.codec, force_encoder_allocation);
-    recreate_stream = false;  // SetCodec has already recreated the stream.
-  } else if (params.conference_mode && parameters_.codec_settings) {
-    bool force_encoder_allocation = false;
-    SetCodec(*parameters_.codec_settings, force_encoder_allocation);
-    recreate_stream = false;  // SetCodec has already recreated the stream.
-  }
-  if (recreate_stream) {
-    LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
-    RecreateWebRtcStream();
-  }
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpParameters(
-    const webrtc::RtpParameters& new_parameters) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (!ValidateRtpParameters(new_parameters)) {
-    return false;
-  }
-
-  bool reconfigure_encoder = new_parameters.encodings[0].max_bitrate_bps !=
-                             rtp_parameters_.encodings[0].max_bitrate_bps;
-  rtp_parameters_ = new_parameters;
-  // Codecs are currently handled at the WebRtcVideoChannel2 level.
-  rtp_parameters_.codecs.clear();
-  if (reconfigure_encoder) {
-    ReconfigureEncoder();
-  }
-  // Encoding may have been activated/deactivated.
-  UpdateSendState();
-  return true;
-}
-
-webrtc::RtpParameters
-WebRtcVideoChannel2::WebRtcVideoSendStream::GetRtpParameters() const {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  return rtp_parameters_;
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::ValidateRtpParameters(
-    const webrtc::RtpParameters& rtp_parameters) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (rtp_parameters.encodings.size() != 1) {
-    LOG(LS_ERROR)
-        << "Attempted to set RtpParameters without exactly one encoding";
-    return false;
-  }
-  if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
-    LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
-    return false;
-  }
-  return true;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::UpdateSendState() {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  // TODO(deadbeef): Need to handle more than one encoding in the future.
-  RTC_DCHECK(rtp_parameters_.encodings.size() == 1u);
-  if (sending_ && rtp_parameters_.encodings[0].active) {
-    RTC_DCHECK(stream_ != nullptr);
-    stream_->Start();
-  } else {
-    if (stream_ != nullptr) {
-      stream_->Stop();
-    }
-  }
-}
-
-webrtc::VideoEncoderConfig
-WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
-    const VideoCodec& codec) const {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  webrtc::VideoEncoderConfig encoder_config;
-  bool is_screencast = parameters_.options.is_screencast.value_or(false);
-  if (is_screencast) {
-    encoder_config.min_transmit_bitrate_bps =
-        1000 * parameters_.options.screencast_min_bitrate_kbps.value_or(0);
-    encoder_config.content_type =
-        webrtc::VideoEncoderConfig::ContentType::kScreen;
-  } else {
-    encoder_config.min_transmit_bitrate_bps = 0;
-    encoder_config.content_type =
-        webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
-  }
-
-  // By default, the stream count for the codec configuration should match the
-  // number of negotiated ssrcs. But if the codec is blacklisted for simulcast
-  // or a screencast (and not in simulcast screenshare experiment), only
-  // configure a single stream.
-  encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();
-  if (IsCodecBlacklistedForSimulcast(codec.name) ||
-      (is_screencast &&
-       (!UseSimulcastScreenshare() || !parameters_.conference_mode))) {
-    encoder_config.number_of_streams = 1;
-  }
-
-  int stream_max_bitrate = parameters_.max_bitrate_bps;
-  if (rtp_parameters_.encodings[0].max_bitrate_bps) {
-    stream_max_bitrate =
-        MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),
-                    parameters_.max_bitrate_bps);
-  }
-
-  int codec_max_bitrate_kbps;
-  if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) {
-    stream_max_bitrate = codec_max_bitrate_kbps * 1000;
-  }
-  encoder_config.max_bitrate_bps = stream_max_bitrate;
-
-  int max_qp = kDefaultQpMax;
-  codec.GetParam(kCodecParamMaxQuantization, &max_qp);
-  encoder_config.video_stream_factory =
-      new rtc::RefCountedObject<EncoderStreamFactory>(
-          codec.name, max_qp, kDefaultVideoMaxFramerate, is_screencast,
-          parameters_.conference_mode);
-  return encoder_config;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::ReconfigureEncoder() {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (!stream_) {
-    // The webrtc::VideoSendStream |stream_| has not yet been created but other
-    // parameters has changed.
-    return;
-  }
-
-  RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0);
-
-  RTC_CHECK(parameters_.codec_settings);
-  VideoCodecSettings codec_settings = *parameters_.codec_settings;
-
-  webrtc::VideoEncoderConfig encoder_config =
-      CreateVideoEncoderConfig(codec_settings.codec);
-
-  encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
-      codec_settings.codec);
-
-  stream_->ReconfigureVideoEncoder(encoder_config.Copy());
-
-  encoder_config.encoder_specific_settings = NULL;
-
-  parameters_.encoder_config = std::move(encoder_config);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSend(bool send) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  sending_ = send;
-  UpdateSendState();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::RemoveSink(
-    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  RTC_DCHECK(encoder_sink_ == sink);
-  encoder_sink_ = nullptr;
-  source_->RemoveSink(sink);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::AddOrUpdateSink(
-    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
-    const rtc::VideoSinkWants& wants) {
-  if (worker_thread_ == rtc::Thread::Current()) {
-    // AddOrUpdateSink is called on |worker_thread_| if this is the first
-    // registration of |sink|.
-    RTC_DCHECK_RUN_ON(&thread_checker_);
-    encoder_sink_ = sink;
-    source_->AddOrUpdateSink(encoder_sink_, wants);
-  } else {
-    // Subsequent calls to AddOrUpdateSink will happen on the encoder task
-    // queue.
-    invoker_.AsyncInvoke<void>(
-        RTC_FROM_HERE, worker_thread_, [this, sink, wants] {
-          RTC_DCHECK_RUN_ON(&thread_checker_);
-          // |sink| may be invalidated after this task was posted since
-          // RemoveSink is called on the worker thread.
-          bool encoder_sink_valid = (sink == encoder_sink_);
-          if (source_ && encoder_sink_valid) {
-            source_->AddOrUpdateSink(encoder_sink_, wants);
-          }
-        });
-  }
-}
-
-VideoSenderInfo WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo(
-    bool log_stats) {
-  VideoSenderInfo info;
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
-    info.add_ssrc(ssrc);
-
-  if (parameters_.codec_settings) {
-    info.codec_name = parameters_.codec_settings->codec.name;
-    info.codec_payload_type = rtc::Optional<int>(
-        parameters_.codec_settings->codec.id);
-  }
-
-  if (stream_ == NULL)
-    return info;
-
-  webrtc::VideoSendStream::Stats stats = stream_->GetStats();
-
-  if (log_stats)
-    LOG(LS_INFO) << stats.ToString(rtc::TimeMillis());
-
-  info.adapt_changes = stats.number_of_cpu_adapt_changes;
-  info.adapt_reason =
-      stats.cpu_limited_resolution ? ADAPTREASON_CPU : ADAPTREASON_NONE;
-
-  // Get bandwidth limitation info from stream_->GetStats().
-  // Input resolution (output from video_adapter) can be further scaled down or
-  // higher video layer(s) can be dropped due to bitrate constraints.
-  // Note, adapt_changes only include changes from the video_adapter.
-  if (stats.bw_limited_resolution)
-    info.adapt_reason |= ADAPTREASON_BANDWIDTH;
-
-  info.encoder_implementation_name = stats.encoder_implementation_name;
-  info.ssrc_groups = ssrc_groups_;
-  info.framerate_input = stats.input_frame_rate;
-  info.framerate_sent = stats.encode_frame_rate;
-  info.avg_encode_ms = stats.avg_encode_time_ms;
-  info.encode_usage_percent = stats.encode_usage_percent;
-  info.frames_encoded = stats.frames_encoded;
-  info.qp_sum = stats.qp_sum;
-
-  info.nominal_bitrate = stats.media_bitrate_bps;
-  info.preferred_bitrate = stats.preferred_media_bitrate_bps;
-
-  info.send_frame_width = 0;
-  info.send_frame_height = 0;
-  for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
-           stats.substreams.begin();
-       it != stats.substreams.end(); ++it) {
-    // TODO(pbos): Wire up additional stats, such as padding bytes.
-    webrtc::VideoSendStream::StreamStats stream_stats = it->second;
-    info.bytes_sent += stream_stats.rtp_stats.transmitted.payload_bytes +
-                       stream_stats.rtp_stats.transmitted.header_bytes +
-                       stream_stats.rtp_stats.transmitted.padding_bytes;
-    info.packets_sent += stream_stats.rtp_stats.transmitted.packets;
-    info.packets_lost += stream_stats.rtcp_stats.cumulative_lost;
-    if (stream_stats.width > info.send_frame_width)
-      info.send_frame_width = stream_stats.width;
-    if (stream_stats.height > info.send_frame_height)
-      info.send_frame_height = stream_stats.height;
-    info.firs_rcvd += stream_stats.rtcp_packet_type_counts.fir_packets;
-    info.nacks_rcvd += stream_stats.rtcp_packet_type_counts.nack_packets;
-    info.plis_rcvd += stream_stats.rtcp_packet_type_counts.pli_packets;
-  }
-
-  if (!stats.substreams.empty()) {
-    // TODO(pbos): Report fraction lost per SSRC.
-    webrtc::VideoSendStream::StreamStats first_stream_stats =
-        stats.substreams.begin()->second;
-    info.fraction_lost =
-        static_cast<float>(first_stream_stats.rtcp_stats.fraction_lost) /
-        (1 << 8);
-  }
-
-  return info;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::FillBitrateInfo(
-    BandwidthEstimationInfo* bwe_info) {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (stream_ == NULL) {
-    return;
-  }
-  webrtc::VideoSendStream::Stats stats = stream_->GetStats();
-  for (std::map<uint32_t, webrtc::VideoSendStream::StreamStats>::iterator it =
-           stats.substreams.begin();
-       it != stats.substreams.end(); ++it) {
-    bwe_info->transmit_bitrate += it->second.total_bitrate_bps;
-    bwe_info->retransmit_bitrate += it->second.retransmit_bitrate_bps;
-  }
-  bwe_info->target_enc_bitrate += stats.target_media_bitrate_bps;
-  bwe_info->actual_enc_bitrate += stats.media_bitrate_bps;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
-  RTC_DCHECK_RUN_ON(&thread_checker_);
-  if (stream_ != NULL) {
-    call_->DestroyVideoSendStream(stream_);
-  }
-
-  RTC_CHECK(parameters_.codec_settings);
-  RTC_DCHECK_EQ((parameters_.encoder_config.content_type ==
-                 webrtc::VideoEncoderConfig::ContentType::kScreen),
-                parameters_.options.is_screencast.value_or(false))
-      << "encoder content type inconsistent with screencast option";
-  parameters_.encoder_config.encoder_specific_settings =
-      ConfigureVideoEncoderSettings(parameters_.codec_settings->codec);
-
-  webrtc::VideoSendStream::Config config = parameters_.config.Copy();
-  if (!config.rtp.rtx.ssrcs.empty() && config.rtp.rtx.payload_type == -1) {
-    LOG(LS_WARNING) << "RTX SSRCs configured but there's no configured RTX "
-                       "payload type the set codec. Ignoring RTX.";
-    config.rtp.rtx.ssrcs.clear();
-  }
-  stream_ = call_->CreateVideoSendStream(std::move(config),
-                                         parameters_.encoder_config.Copy());
-
-  parameters_.encoder_config.encoder_specific_settings = NULL;
-
-  if (source_) {
-    stream_->SetSource(this, GetDegradationPreference());
-  }
-
-  // Call stream_->Start() if necessary conditions are met.
-  UpdateSendState();
-}
-
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
-    webrtc::Call* call,
-    const StreamParams& sp,
-    webrtc::VideoReceiveStream::Config config,
-    WebRtcVideoDecoderFactory* external_decoder_factory,
-    bool default_stream,
-    const std::vector<VideoCodecSettings>& recv_codecs,
-    const webrtc::FlexfecReceiveStream::Config& flexfec_config)
-    : call_(call),
-      stream_params_(sp),
-      stream_(NULL),
-      default_stream_(default_stream),
-      config_(std::move(config)),
-      flexfec_config_(flexfec_config),
-      flexfec_stream_(nullptr),
-      external_decoder_factory_(external_decoder_factory),
-      sink_(NULL),
-      first_frame_timestamp_(-1),
-      estimated_remote_start_ntp_time_ms_(0) {
-  config_.renderer = this;
-  std::vector<AllocatedDecoder> old_decoders;
-  ConfigureCodecs(recv_codecs, &old_decoders);
-  ConfigureFlexfecCodec(flexfec_config.payload_type);
-  MaybeRecreateWebRtcFlexfecStream();
-  RecreateWebRtcVideoStream();
-  RTC_DCHECK(old_decoders.empty());
-}
-
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder::
-    AllocatedDecoder(webrtc::VideoDecoder* decoder,
-                     webrtc::VideoCodecType type,
-                     bool external)
-    : decoder(decoder),
-      external_decoder(nullptr),
-      type(type),
-      external(external) {
-  if (external) {
-    external_decoder = decoder;
-    this->decoder =
-        new webrtc::VideoDecoderSoftwareFallbackWrapper(type, external_decoder);
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::~WebRtcVideoReceiveStream() {
-  if (flexfec_stream_) {
-    call_->DestroyFlexfecReceiveStream(flexfec_stream_);
-  }
-  call_->DestroyVideoReceiveStream(stream_);
-  ClearDecoders(&allocated_decoders_);
-}
-
-const std::vector<uint32_t>&
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetSsrcs() const {
-  return stream_params_.ssrcs;
-}
-
-rtc::Optional<uint32_t>
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const {
-  std::vector<uint32_t> primary_ssrcs;
-  stream_params_.GetPrimarySsrcs(&primary_ssrcs);
-
-  if (primary_ssrcs.empty()) {
-    LOG(LS_WARNING) << "Empty primary ssrcs vector, returning empty optional";
-    return rtc::Optional<uint32_t>();
-  } else {
-    return rtc::Optional<uint32_t>(primary_ssrcs[0]);
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
-    std::vector<AllocatedDecoder>* old_decoders,
-    const VideoCodec& codec) {
-  webrtc::VideoCodecType type = webrtc::PayloadNameToCodecType(codec.name)
-                                    .value_or(webrtc::kVideoCodecUnknown);
-
-  for (size_t i = 0; i < old_decoders->size(); ++i) {
-    if ((*old_decoders)[i].type == type) {
-      AllocatedDecoder decoder = (*old_decoders)[i];
-      (*old_decoders)[i] = old_decoders->back();
-      old_decoders->pop_back();
-      return decoder;
-    }
-  }
-
-  if (external_decoder_factory_ != NULL) {
-    webrtc::VideoDecoder* decoder =
-        external_decoder_factory_->CreateVideoDecoderWithParams(
-            type, {stream_params_.id});
-    if (decoder != NULL) {
-      return AllocatedDecoder(decoder, type, true /* is_external */);
-    }
-  }
-
-  InternalDecoderFactory internal_decoder_factory;
-  return AllocatedDecoder(internal_decoder_factory.CreateVideoDecoderWithParams(
-                              type, {stream_params_.id}),
-                          type, false /* is_external */);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ConfigureCodecs(
-    const std::vector<VideoCodecSettings>& recv_codecs,
-    std::vector<AllocatedDecoder>* old_decoders) {
-  *old_decoders = allocated_decoders_;
-  allocated_decoders_.clear();
-  config_.decoders.clear();
-  for (size_t i = 0; i < recv_codecs.size(); ++i) {
-    AllocatedDecoder allocated_decoder =
-        CreateOrReuseVideoDecoder(old_decoders, recv_codecs[i].codec);
-    allocated_decoders_.push_back(allocated_decoder);
-
-    webrtc::VideoReceiveStream::Decoder decoder;
-    decoder.decoder = allocated_decoder.decoder;
-    decoder.payload_type = recv_codecs[i].codec.id;
-    decoder.payload_name = recv_codecs[i].codec.name;
-    decoder.codec_params = recv_codecs[i].codec.params;
-    config_.decoders.push_back(decoder);
-  }
-
-  config_.rtp.rtx_payload_types.clear();
-  for (const VideoCodecSettings& recv_codec : recv_codecs) {
-    config_.rtp.rtx_payload_types[recv_codec.codec.id] =
-        recv_codec.rtx_payload_type;
-  }
-
-  config_.rtp.ulpfec = recv_codecs.front().ulpfec;
-
-  config_.rtp.nack.rtp_history_ms =
-      HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ConfigureFlexfecCodec(
-    int flexfec_payload_type) {
-  flexfec_config_.payload_type = flexfec_payload_type;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetLocalSsrc(
-    uint32_t local_ssrc) {
-  // TODO(pbos): Consider turning this sanity check into a RTC_DCHECK. You
-  // should not be able to create a sender with the same SSRC as a receiver, but
-  // right now this can't be done due to unittests depending on receiving what
-  // they are sending from the same MediaChannel.
-  if (local_ssrc == config_.rtp.remote_ssrc) {
-    LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
-                    "unchanged; local_ssrc=" << local_ssrc;
-    return;
-  }
-
-  config_.rtp.local_ssrc = local_ssrc;
-  flexfec_config_.local_ssrc = local_ssrc;
-  LOG(LS_INFO)
-      << "RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc="
-      << local_ssrc;
-  MaybeRecreateWebRtcFlexfecStream();
-  RecreateWebRtcVideoStream();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFeedbackParameters(
-    bool nack_enabled,
-    bool remb_enabled,
-    bool transport_cc_enabled,
-    webrtc::RtcpMode rtcp_mode) {
-  int nack_history_ms = nack_enabled ? kNackHistoryMs : 0;
-  if (config_.rtp.nack.rtp_history_ms == nack_history_ms &&
-      config_.rtp.remb == remb_enabled &&
-      config_.rtp.transport_cc == transport_cc_enabled &&
-      config_.rtp.rtcp_mode == rtcp_mode) {
-    LOG(LS_INFO)
-        << "Ignoring call to SetFeedbackParameters because parameters are "
-           "unchanged; nack="
-        << nack_enabled << ", remb=" << remb_enabled
-        << ", transport_cc=" << transport_cc_enabled;
-    return;
-  }
-  config_.rtp.remb = remb_enabled;
-  config_.rtp.nack.rtp_history_ms = nack_history_ms;
-  config_.rtp.transport_cc = transport_cc_enabled;
-  config_.rtp.rtcp_mode = rtcp_mode;
-  // TODO(brandtr): We should be spec-compliant and set |transport_cc| here
-  // based on the rtcp-fb for the FlexFEC codec, not the media codec.
-  flexfec_config_.transport_cc = config_.rtp.transport_cc;
-  flexfec_config_.rtcp_mode = config_.rtp.rtcp_mode;
-  LOG(LS_INFO)
-      << "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack="
-      << nack_enabled << ", remb=" << remb_enabled
-      << ", transport_cc=" << transport_cc_enabled;
-  MaybeRecreateWebRtcFlexfecStream();
-  RecreateWebRtcVideoStream();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters(
-    const ChangedRecvParameters& params) {
-  bool video_needs_recreation = false;
-  bool flexfec_needs_recreation = false;
-  std::vector<AllocatedDecoder> old_decoders;
-  if (params.codec_settings) {
-    ConfigureCodecs(*params.codec_settings, &old_decoders);
-    video_needs_recreation = true;
-  }
-  if (params.rtp_header_extensions) {
-    config_.rtp.extensions = *params.rtp_header_extensions;
-    flexfec_config_.rtp_header_extensions = *params.rtp_header_extensions;
-    video_needs_recreation = true;
-    flexfec_needs_recreation = true;
-  }
-  if (params.flexfec_payload_type) {
-    ConfigureFlexfecCodec(*params.flexfec_payload_type);
-    flexfec_needs_recreation = true;
-  }
-  if (flexfec_needs_recreation) {
-    LOG(LS_INFO) << "MaybeRecreateWebRtcFlexfecStream (recv) because of "
-                    "SetRecvParameters";
-    MaybeRecreateWebRtcFlexfecStream();
-  }
-  if (video_needs_recreation) {
-    LOG(LS_INFO)
-        << "RecreateWebRtcVideoStream (recv) because of SetRecvParameters";
-    RecreateWebRtcVideoStream();
-    ClearDecoders(&old_decoders);
-  }
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::
-    RecreateWebRtcVideoStream() {
-  if (stream_) {
-    call_->DestroyVideoReceiveStream(stream_);
-    stream_ = nullptr;
-  }
-  webrtc::VideoReceiveStream::Config config = config_.Copy();
-  config.rtp.protected_by_flexfec = (flexfec_stream_ != nullptr);
-  stream_ = call_->CreateVideoReceiveStream(std::move(config));
-  stream_->Start();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::
-    MaybeRecreateWebRtcFlexfecStream() {
-  if (flexfec_stream_) {
-    call_->DestroyFlexfecReceiveStream(flexfec_stream_);
-    flexfec_stream_ = nullptr;
-  }
-  if (flexfec_config_.IsCompleteAndEnabled()) {
-    flexfec_stream_ = call_->CreateFlexfecReceiveStream(flexfec_config_);
-    flexfec_stream_->Start();
-  }
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::ClearDecoders(
-    std::vector<AllocatedDecoder>* allocated_decoders) {
-  for (size_t i = 0; i < allocated_decoders->size(); ++i) {
-    if ((*allocated_decoders)[i].external) {
-      external_decoder_factory_->DestroyVideoDecoder(
-          (*allocated_decoders)[i].external_decoder);
-    }
-    delete (*allocated_decoders)[i].decoder;
-  }
-  allocated_decoders->clear();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::OnFrame(
-    const webrtc::VideoFrame& frame) {
-  rtc::CritScope crit(&sink_lock_);
-
-  if (first_frame_timestamp_ < 0)
-    first_frame_timestamp_ = frame.timestamp();
-  int64_t rtp_time_elapsed_since_first_frame =
-      (timestamp_wraparound_handler_.Unwrap(frame.timestamp()) -
-       first_frame_timestamp_);
-  int64_t elapsed_time_ms = rtp_time_elapsed_since_first_frame /
-                            (cricket::kVideoCodecClockrate / 1000);
-  if (frame.ntp_time_ms() > 0)
-    estimated_remote_start_ntp_time_ms_ = frame.ntp_time_ms() - elapsed_time_ms;
-
-  if (sink_ == NULL) {
-    LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoSink.";
-    return;
-  }
-
-  sink_->OnFrame(frame);
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const {
-  return default_stream_;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSink(
-    rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
-  rtc::CritScope crit(&sink_lock_);
-  sink_ = sink;
-}
-
-std::string
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetCodecNameFromPayloadType(
-    int payload_type) {
-  for (const webrtc::VideoReceiveStream::Decoder& decoder : config_.decoders) {
-    if (decoder.payload_type == payload_type) {
-      return decoder.payload_name;
-    }
-  }
-  return "";
-}
-
-VideoReceiverInfo
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
-    bool log_stats) {
-  VideoReceiverInfo info;
-  info.ssrc_groups = stream_params_.ssrc_groups;
-  info.add_ssrc(config_.rtp.remote_ssrc);
-  webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
-  info.decoder_implementation_name = stats.decoder_implementation_name;
-  if (stats.current_payload_type != -1) {
-    info.codec_payload_type = rtc::Optional<int>(
-        stats.current_payload_type);
-  }
-  info.bytes_rcvd = stats.rtp_stats.transmitted.payload_bytes +
-                    stats.rtp_stats.transmitted.header_bytes +
-                    stats.rtp_stats.transmitted.padding_bytes;
-  info.packets_rcvd = stats.rtp_stats.transmitted.packets;
-  info.packets_lost = stats.rtcp_stats.cumulative_lost;
-  info.fraction_lost =
-      static_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8);
-
-  info.framerate_rcvd = stats.network_frame_rate;
-  info.framerate_decoded = stats.decode_frame_rate;
-  info.framerate_output = stats.render_frame_rate;
-  info.frame_width = stats.width;
-  info.frame_height = stats.height;
-
-  {
-    rtc::CritScope frame_cs(&sink_lock_);
-    info.capture_start_ntp_time_ms = estimated_remote_start_ntp_time_ms_;
-  }
-
-  info.decode_ms = stats.decode_ms;
-  info.max_decode_ms = stats.max_decode_ms;
-  info.current_delay_ms = stats.current_delay_ms;
-  info.target_delay_ms = stats.target_delay_ms;
-  info.jitter_buffer_ms = stats.jitter_buffer_ms;
-  info.min_playout_delay_ms = stats.min_playout_delay_ms;
-  info.render_delay_ms = stats.render_delay_ms;
-  info.frames_received = stats.frame_counts.key_frames +
-                         stats.frame_counts.delta_frames;
-  info.frames_decoded = stats.frames_decoded;
-  info.frames_rendered = stats.frames_rendered;
-  info.qp_sum = stats.qp_sum;
-
-  info.codec_name = GetCodecNameFromPayloadType(stats.current_payload_type);
-
-  info.firs_sent = stats.rtcp_packet_type_counts.fir_packets;
-  info.plis_sent = stats.rtcp_packet_type_counts.pli_packets;
-  info.nacks_sent = stats.rtcp_packet_type_counts.nack_packets;
-
-  if (log_stats)
-    LOG(LS_INFO) << stats.ToString(rtc::TimeMillis());
-
-  return info;
-}
-
-WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
-    : flexfec_payload_type(-1), rtx_payload_type(-1) {}
-
-bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
-    const WebRtcVideoChannel2::VideoCodecSettings& other) const {
-  return codec == other.codec && ulpfec == other.ulpfec &&
-         flexfec_payload_type == other.flexfec_payload_type &&
-         rtx_payload_type == other.rtx_payload_type;
-}
-
-bool WebRtcVideoChannel2::VideoCodecSettings::EqualsDisregardingFlexfec(
-    const WebRtcVideoChannel2::VideoCodecSettings& a,
-    const WebRtcVideoChannel2::VideoCodecSettings& b) {
-  return a.codec == b.codec && a.ulpfec == b.ulpfec &&
-         a.rtx_payload_type == b.rtx_payload_type;
-}
-
-bool WebRtcVideoChannel2::VideoCodecSettings::operator!=(
-    const WebRtcVideoChannel2::VideoCodecSettings& other) const {
-  return !(*this == other);
-}
-
-std::vector<WebRtcVideoChannel2::VideoCodecSettings>
-WebRtcVideoChannel2::MapCodecs(const std::vector<VideoCodec>& codecs) {
-  RTC_DCHECK(!codecs.empty());
-
-  std::vector<VideoCodecSettings> video_codecs;
-  std::map<int, bool> payload_used;
-  std::map<int, VideoCodec::CodecType> payload_codec_type;
-  // |rtx_mapping| maps video payload type to rtx payload type.
-  std::map<int, int> rtx_mapping;
-
-  webrtc::UlpfecConfig ulpfec_config;
-  int flexfec_payload_type = -1;
-
-  for (size_t i = 0; i < codecs.size(); ++i) {
-    const VideoCodec& in_codec = codecs[i];
-    int payload_type = in_codec.id;
-
-    if (payload_used[payload_type]) {
-      LOG(LS_ERROR) << "Payload type already registered: "
-                    << in_codec.ToString();
-      return std::vector<VideoCodecSettings>();
-    }
-    payload_used[payload_type] = true;
-    payload_codec_type[payload_type] = in_codec.GetCodecType();
-
-    switch (in_codec.GetCodecType()) {
-      case VideoCodec::CODEC_RED: {
-        // RED payload type, should not have duplicates.
-        RTC_DCHECK_EQ(-1, ulpfec_config.red_payload_type);
-        ulpfec_config.red_payload_type = in_codec.id;
-        continue;
-      }
-
-      case VideoCodec::CODEC_ULPFEC: {
-        // ULPFEC payload type, should not have duplicates.
-        RTC_DCHECK_EQ(-1, ulpfec_config.ulpfec_payload_type);
-        ulpfec_config.ulpfec_payload_type = in_codec.id;
-        continue;
-      }
-
-      case VideoCodec::CODEC_FLEXFEC: {
-        // FlexFEC payload type, should not have duplicates.
-        RTC_DCHECK_EQ(-1, flexfec_payload_type);
-        flexfec_payload_type = in_codec.id;
-        continue;
-      }
-
-      case VideoCodec::CODEC_RTX: {
-        int associated_payload_type;
-        if (!in_codec.GetParam(kCodecParamAssociatedPayloadType,
-                               &associated_payload_type) ||
-            !IsValidRtpPayloadType(associated_payload_type)) {
-          LOG(LS_ERROR)
-              << "RTX codec with invalid or no associated payload type: "
-              << in_codec.ToString();
-          return std::vector<VideoCodecSettings>();
-        }
-        rtx_mapping[associated_payload_type] = in_codec.id;
-        continue;
-      }
-
-      case VideoCodec::CODEC_VIDEO:
-        break;
-    }
-
-    video_codecs.push_back(VideoCodecSettings());
-    video_codecs.back().codec = in_codec;
-  }
-
-  // One of these codecs should have been a video codec. Only having FEC
-  // parameters into this code is a logic error.
-  RTC_DCHECK(!video_codecs.empty());
-
-  for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
-       it != rtx_mapping.end();
-       ++it) {
-    if (!payload_used[it->first]) {
-      LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
-      return std::vector<VideoCodecSettings>();
-    }
-    if (payload_codec_type[it->first] != VideoCodec::CODEC_VIDEO &&
-        payload_codec_type[it->first] != VideoCodec::CODEC_RED) {
-      LOG(LS_ERROR) << "RTX not mapped to regular video codec or RED codec.";
-      return std::vector<VideoCodecSettings>();
-    }
-
-    if (it->first == ulpfec_config.red_payload_type) {
-      ulpfec_config.red_rtx_payload_type = it->second;
-    }
-  }
-
-  for (size_t i = 0; i < video_codecs.size(); ++i) {
-    video_codecs[i].ulpfec = ulpfec_config;
-    video_codecs[i].flexfec_payload_type = flexfec_payload_type;
-    if (rtx_mapping[video_codecs[i].codec.id] != 0 &&
-        rtx_mapping[video_codecs[i].codec.id] !=
-            ulpfec_config.red_payload_type) {
-      video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
-    }
-  }
-
-  return video_codecs;
-}
-
-}  // namespace cricket