Move talk/media to webrtc/media

talk/media/webrtc/webrtcvideoengine2.cc

-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifdef HAVE_WEBRTC_VIDEO
-#include "talk/media/webrtc/webrtcvideoengine2.h"
-
-#include <algorithm>
-#include <set>
-#include <string>
-
-#include "talk/media/base/videocapturer.h"
-#include "talk/media/base/videorenderer.h"
-#include "talk/media/webrtc/constants.h"
-#include "talk/media/webrtc/simulcast.h"
-#include "talk/media/webrtc/webrtcmediaengine.h"
-#include "talk/media/webrtc/webrtcvideoencoderfactory.h"
-#include "talk/media/webrtc/webrtcvideoframe.h"
-#include "talk/media/webrtc/webrtcvoiceengine.h"
-#include "webrtc/base/buffer.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.h"
-#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
-#include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
-#include "webrtc/system_wrappers/include/field_trial.h"
-#include "webrtc/video_decoder.h"
-#include "webrtc/video_encoder.h"
-
-namespace cricket {
-namespace {
-
-// 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(webrtc::kVideoCodecVP8);
-  }
-
-  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<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 (codec.type == webrtc::kVideoCodecVP8) {
-        return true;
-      }
-    }
-    return false;
-  }
-
-  webrtc::VideoEncoder* CreateVideoEncoder(
-      webrtc::VideoCodecType type) override {
-    RTC_DCHECK(factory_ != NULL);
-    // If it's a codec type we can simulcast, create a wrapped encoder.
-    if (type == webrtc::kVideoCodecVP8) {
-      return new webrtc::SimulcastEncoderAdapter(
-          new EncoderFactoryAdapter(factory_));
-    }
-    webrtc::VideoEncoder* encoder = factory_->CreateVideoEncoder(type);
-    if (encoder) {
-      non_simulcast_encoders_.push_back(encoder);
-    }
-    return encoder;
-  }
-
-  const std::vector<VideoCodec>& codecs() const override {
-    return factory_->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_;
-};
-
-bool CodecIsInternallySupported(const std::string& codec_name) {
-  if (CodecNamesEq(codec_name, kVp8CodecName)) {
-    return true;
-  }
-  if (CodecNamesEq(codec_name, kVp9CodecName)) {
-    return true;
-  }
-  if (CodecNamesEq(codec_name, kH264CodecName)) {
-    return webrtc::H264Encoder::IsSupported() &&
-        webrtc::H264Decoder::IsSupported();
-  }
-  return false;
-}
-
-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 VideoCodec MakeVideoCodecWithDefaultFeedbackParams(int payload_type,
-                                                          const char* name) {
-  VideoCodec codec(payload_type, name, kDefaultVideoMaxWidth,
-                   kDefaultVideoMaxHeight, kDefaultVideoMaxFramerate, 0);
-  AddDefaultFeedbackParams(&codec);
-  return codec;
-}
-
-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;
-}
-
-inline const webrtc::RtpExtension* FindHeaderExtension(
-    const std::vector<webrtc::RtpExtension>& extensions,
-    const std::string& name) {
-  for (const auto& kv : extensions) {
-    if (kv.name == name) {
-      return &kv;
-    }
-  }
-  return NULL;
-}
-
-// Merges two fec configs and logs an error if a conflict arises
-// such that merging in different order would trigger a different output.
-static void MergeFecConfig(const webrtc::FecConfig& other,
-                           webrtc::FecConfig* output) {
-  if (other.ulpfec_payload_type != -1) {
-    if (output->ulpfec_payload_type != -1 &&
-        output->ulpfec_payload_type != other.ulpfec_payload_type) {
-      LOG(LS_WARNING) << "Conflict merging ulpfec_payload_type configs: "
-                      << output->ulpfec_payload_type << " and "
-                      << other.ulpfec_payload_type;
-    }
-    output->ulpfec_payload_type = other.ulpfec_payload_type;
-  }
-  if (other.red_payload_type != -1) {
-    if (output->red_payload_type != -1 &&
-        output->red_payload_type != other.red_payload_type) {
-      LOG(LS_WARNING) << "Conflict merging red_payload_type configs: "
-                      << output->red_payload_type << " and "
-                      << other.red_payload_type;
-    }
-    output->red_payload_type = other.red_payload_type;
-  }
-  if (other.red_rtx_payload_type != -1) {
-    if (output->red_rtx_payload_type != -1 &&
-        output->red_rtx_payload_type != other.red_rtx_payload_type) {
-      LOG(LS_WARNING) << "Conflict merging red_rtx_payload_type configs: "
-                      << output->red_rtx_payload_type << " and "
-                      << other.red_rtx_payload_type;
-    }
-    output->red_rtx_payload_type = other.red_rtx_payload_type;
-  }
-}
-
-// 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;
-  }
-}
-}  // namespace
-
-// Constants defined in talk/media/webrtc/constants.h
-// TODO(pbos): Move these to a separate constants.cc file.
-const int kMinVideoBitrate = 30;
-const int kStartVideoBitrate = 300;
-
-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;
-
-std::vector<VideoCodec> DefaultVideoCodecList() {
-  std::vector<VideoCodec> codecs;
-  codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp8PlType,
-                                                           kVp8CodecName));
-  if (CodecIsInternallySupported(kVp9CodecName)) {
-    codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultVp9PlType,
-                                                             kVp9CodecName));
-    // TODO(andresp): Add rtx codec for vp9 and verify it works.
-  }
-  if (CodecIsInternallySupported(kH264CodecName)) {
-    codecs.push_back(MakeVideoCodecWithDefaultFeedbackParams(kDefaultH264PlType,
-                                                             kH264CodecName));
-  }
-  codecs.push_back(
-      VideoCodec::CreateRtxCodec(kDefaultRtxVp8PlType, kDefaultVp8PlType));
-  codecs.push_back(VideoCodec(kDefaultRedPlType, kRedCodecName));
-  codecs.push_back(VideoCodec(kDefaultUlpfecType, kUlpfecCodecName));
-  return codecs;
-}
-
-std::vector<webrtc::VideoStream>
-WebRtcVideoChannel2::WebRtcVideoSendStream::CreateSimulcastVideoStreams(
-    const VideoCodec& codec,
-    const VideoOptions& options,
-    int max_bitrate_bps,
-    size_t num_streams) {
-  int max_qp = kDefaultQpMax;
-  codec.GetParam(kCodecParamMaxQuantization, &max_qp);
-
-  return GetSimulcastConfig(
-      num_streams, codec.width, codec.height, max_bitrate_bps, max_qp,
-      codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate);
-}
-
-std::vector<webrtc::VideoStream>
-WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoStreams(
-    const VideoCodec& codec,
-    const VideoOptions& options,
-    int max_bitrate_bps,
-    size_t num_streams) {
-  int codec_max_bitrate_kbps;
-  if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) {
-    max_bitrate_bps = codec_max_bitrate_kbps * 1000;
-  }
-  if (num_streams != 1) {
-    return CreateSimulcastVideoStreams(codec, options, max_bitrate_bps,
-                                       num_streams);
-  }
-
-  // For unset max bitrates set default bitrate for non-simulcast.
-  if (max_bitrate_bps <= 0) {
-    max_bitrate_bps =
-        GetMaxDefaultVideoBitrateKbps(codec.width, codec.height) * 1000;
-  }
-
-  webrtc::VideoStream stream;
-  stream.width = codec.width;
-  stream.height = codec.height;
-  stream.max_framerate =
-      codec.framerate != 0 ? codec.framerate : kDefaultVideoMaxFramerate;
-
-  stream.min_bitrate_bps = kMinVideoBitrate * 1000;
-  stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps;
-
-  int max_qp = kDefaultQpMax;
-  codec.GetParam(kCodecParamMaxQuantization, &max_qp);
-  stream.max_qp = max_qp;
-  std::vector<webrtc::VideoStream> streams;
-  streams.push_back(stream);
-  return streams;
-}
-
-void* WebRtcVideoChannel2::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
-    const VideoCodec& codec,
-    const VideoOptions& options,
-    bool is_screencast) {
-  // 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 = !options.video_noise_reduction;
-    denoising = options.video_noise_reduction.value_or(false);
-  }
-
-  if (CodecNamesEq(codec.name, kVp8CodecName)) {
-    encoder_settings_.vp8 = webrtc::VideoEncoder::GetDefaultVp8Settings();
-    encoder_settings_.vp8.automaticResizeOn = automatic_resize;
-    // VP8 denoising is enabled by default.
-    encoder_settings_.vp8.denoisingOn =
-        codec_default_denoising ? true : denoising;
-    encoder_settings_.vp8.frameDroppingOn = frame_dropping;
-    return &encoder_settings_.vp8;
-  }
-  if (CodecNamesEq(codec.name, kVp9CodecName)) {
-    encoder_settings_.vp9 = webrtc::VideoEncoder::GetDefaultVp9Settings();
-    // VP9 denoising is disabled by default.
-    encoder_settings_.vp9.denoisingOn =
-        codec_default_denoising ? false : denoising;
-    encoder_settings_.vp9.frameDroppingOn = frame_dropping;
-    return &encoder_settings_.vp9;
-  }
-  return NULL;
-}
-
-DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
-    : default_recv_ssrc_(0), default_renderer_(NULL) {}
-
-UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
-    WebRtcVideoChannel2* channel,
-    uint32_t ssrc) {
-  if (default_recv_ssrc_ != 0) {  // Already one default stream.
-    LOG(LS_WARNING) << "Unknown SSRC, but default receive stream already set.";
-    return kDropPacket;
-  }
-
-  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->SetRenderer(ssrc, default_renderer_);
-  default_recv_ssrc_ = ssrc;
-  return kDeliverPacket;
-}
-
-VideoRenderer* DefaultUnsignalledSsrcHandler::GetDefaultRenderer() const {
-  return default_renderer_;
-}
-
-void DefaultUnsignalledSsrcHandler::SetDefaultRenderer(
-    VideoMediaChannel* channel,
-    VideoRenderer* renderer) {
-  default_renderer_ = renderer;
-  if (default_recv_ssrc_ != 0) {
-    channel->SetRenderer(default_recv_ssrc_, default_renderer_);
-  }
-}
-
-WebRtcVideoEngine2::WebRtcVideoEngine2()
-    : initialized_(false),
-      external_decoder_factory_(NULL),
-      external_encoder_factory_(NULL) {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::WebRtcVideoEngine2()";
-  video_codecs_ = GetSupportedCodecs();
-}
-
-WebRtcVideoEngine2::~WebRtcVideoEngine2() {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::~WebRtcVideoEngine2";
-}
-
-void WebRtcVideoEngine2::Init() {
-  LOG(LS_INFO) << "WebRtcVideoEngine2::Init";
-  initialized_ = true;
-}
-
-WebRtcVideoChannel2* WebRtcVideoEngine2::CreateChannel(
-    webrtc::Call* call,
-    const VideoOptions& options) {
-  RTC_DCHECK(initialized_);
-  LOG(LS_INFO) << "CreateChannel. Options: " << options.ToString();
-  return new WebRtcVideoChannel2(call, options, video_codecs_,
-      external_encoder_factory_, external_decoder_factory_);
-}
-
-const std::vector<VideoCodec>& WebRtcVideoEngine2::codecs() const {
-  return video_codecs_;
-}
-
-RtpCapabilities WebRtcVideoEngine2::GetCapabilities() const {
-  RtpCapabilities capabilities;
-  capabilities.header_extensions.push_back(
-      RtpHeaderExtension(kRtpTimestampOffsetHeaderExtension,
-                         kRtpTimestampOffsetHeaderExtensionDefaultId));
-  capabilities.header_extensions.push_back(
-      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
-                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
-  capabilities.header_extensions.push_back(
-      RtpHeaderExtension(kRtpVideoRotationHeaderExtension,
-                         kRtpVideoRotationHeaderExtensionDefaultId));
-  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
-    capabilities.header_extensions.push_back(RtpHeaderExtension(
-        kRtpTransportSequenceNumberHeaderExtension,
-        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
-  }
-  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->codecs())) {
-    simulcast_encoder_factory_.reset(
-        new WebRtcSimulcastEncoderFactory(encoder_factory));
-    encoder_factory = simulcast_encoder_factory_.get();
-  }
-  external_encoder_factory_ = encoder_factory;
-
-  video_codecs_ = GetSupportedCodecs();
-}
-
-bool WebRtcVideoEngine2::EnableTimedRender() {
-  // TODO(pbos): Figure out whether this can be removed.
-  return true;
-}
-
-// Checks to see whether we comprehend and could receive a particular codec
-bool WebRtcVideoEngine2::FindCodec(const VideoCodec& in) {
-  // TODO(pbos): Probe encoder factory to figure out that the codec is supported
-  // if supported by the encoder factory. Add a corresponding test that fails
-  // with this code (that doesn't ask the factory).
-  for (size_t j = 0; j < video_codecs_.size(); ++j) {
-    VideoCodec codec(video_codecs_[j].id, video_codecs_[j].name, 0, 0, 0, 0);
-    if (codec.Matches(in)) {
-      return true;
-    }
-  }
-  return false;
-}
-
-// Ignore spammy trace messages, mostly from the stats API when we haven't
-// gotten RTCP info yet from the remote side.
-bool WebRtcVideoEngine2::ShouldIgnoreTrace(const std::string& trace) {
-  static const char* const kTracesToIgnore[] = {NULL};
-  for (const char* const* p = kTracesToIgnore; *p; ++p) {
-    if (trace.find(*p) == 0) {
-      return true;
-    }
-  }
-  return false;
-}
-
-std::vector<VideoCodec> WebRtcVideoEngine2::GetSupportedCodecs() const {
-  std::vector<VideoCodec> supported_codecs = DefaultVideoCodecList();
-
-  if (external_encoder_factory_ == NULL) {
-    return supported_codecs;
-  }
-
-  const std::vector<WebRtcVideoEncoderFactory::VideoCodec>& codecs =
-      external_encoder_factory_->codecs();
-  for (size_t i = 0; i < codecs.size(); ++i) {
-    // Don't add internally-supported codecs twice.
-    if (CodecIsInternallySupported(codecs[i].name)) {
-      continue;
-    }
-
-    // External video encoders are given payloads 120-127. This also means that
-    // we only support up to 8 external payload types.
-    const int kExternalVideoPayloadTypeBase = 120;
-    size_t payload_type = kExternalVideoPayloadTypeBase + i;
-    RTC_DCHECK(payload_type < 128);
-    VideoCodec codec(static_cast<int>(payload_type),
-                     codecs[i].name,
-                     codecs[i].max_width,
-                     codecs[i].max_height,
-                     codecs[i].max_fps,
-                     0);
-
-    AddDefaultFeedbackParams(&codec);
-    supported_codecs.push_back(codec);
-  }
-  return supported_codecs;
-}
-
-WebRtcVideoChannel2::WebRtcVideoChannel2(
-    webrtc::Call* call,
-    const VideoOptions& options,
-    const std::vector<VideoCodec>& recv_codecs,
-    WebRtcVideoEncoderFactory* external_encoder_factory,
-    WebRtcVideoDecoderFactory* external_decoder_factory)
-    : call_(call),
-      unsignalled_ssrc_handler_(&default_unsignalled_ssrc_handler_),
-      external_encoder_factory_(external_encoder_factory),
-      external_decoder_factory_(external_decoder_factory) {
-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  SetDefaultOptions();
-  options_.SetAll(options);
-  if (options_.cpu_overuse_detection)
-    signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
-  rtcp_receiver_report_ssrc_ = kDefaultRtcpReceiverReportSsrc;
-  sending_ = false;
-  default_send_ssrc_ = 0;
-  SetRecvCodecs(recv_codecs);
-}
-
-void WebRtcVideoChannel2::SetDefaultOptions() {
-  options_.cpu_overuse_detection = rtc::Optional<bool>(true);
-  options_.dscp = rtc::Optional<bool>(false);
-  options_.suspend_below_min_bitrate = rtc::Optional<bool>(false);
-  options_.screencast_min_bitrate = rtc::Optional<int>(0);
-}
-
-WebRtcVideoChannel2::~WebRtcVideoChannel2() {
-  for (auto& kv : send_streams_)
-    delete kv.second;
-  for (auto& kv : receive_streams_)
-    delete kv.second;
-}
-
-bool WebRtcVideoChannel2::CodecIsExternallySupported(
-    const std::string& name) const {
-  if (external_encoder_factory_ == NULL) {
-    return false;
-  }
-
-  const std::vector<WebRtcVideoEncoderFactory::VideoCodec> external_codecs =
-      external_encoder_factory_->codecs();
-  for (size_t c = 0; c < external_codecs.size(); ++c) {
-    if (CodecNamesEq(name, external_codecs[c].name)) {
-      return true;
-    }
-  }
-  return false;
-}
-
-std::vector<WebRtcVideoChannel2::VideoCodecSettings>
-WebRtcVideoChannel2::FilterSupportedCodecs(
-    const std::vector<WebRtcVideoChannel2::VideoCodecSettings>& mapped_codecs)
-    const {
-  std::vector<VideoCodecSettings> supported_codecs;
-  for (size_t i = 0; i < mapped_codecs.size(); ++i) {
-    const VideoCodecSettings& codec = mapped_codecs[i];
-    if (CodecIsInternallySupported(codec.codec.name) ||
-        CodecIsExternallySupported(codec.codec.name)) {
-      supported_codecs.push_back(codec);
-    }
-  }
-  return supported_codecs;
-}
-
-bool WebRtcVideoChannel2::ReceiveCodecsHaveChanged(
-    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);
-  for (size_t i = 0; i < before.size(); ++i) {
-    // For the same reason that we sort the codecs, we also ignore the
-    // preference.  We don't want a preference change on the receive
-    // side to cause recreation of the stream.
-    before[i].codec.preference = 0;
-    after[i].codec.preference = 0;
-    if (before[i] != after[i]) {
-      return true;
-    }
-  }
-  return false;
-}
-
-bool WebRtcVideoChannel2::SetSendParameters(const VideoSendParameters& params) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendParameters");
-  LOG(LS_INFO) << "SetSendParameters: " << params.ToString();
-  // TODO(pbos): Refactor this to only recreate the send streams once
-  // instead of 4 times.
-  if (!SetSendCodecs(params.codecs) ||
-      !SetSendRtpHeaderExtensions(params.extensions) ||
-      !SetMaxSendBandwidth(params.max_bandwidth_bps) ||
-      !SetOptions(params.options)) {
-    return false;
-  }
-  if (send_params_.rtcp.reduced_size != params.rtcp.reduced_size) {
-    rtc::CritScope stream_lock(&stream_crit_);
-    for (auto& kv : send_streams_) {
-      kv.second->SetSendParameters(params);
-    }
-  }
-  send_params_ = params;
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetRecvParameters(const VideoRecvParameters& params) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvParameters");
-  LOG(LS_INFO) << "SetRecvParameters: " << params.ToString();
-  // TODO(pbos): Refactor this to only recreate the recv streams once
-  // instead of twice.
-  if (!SetRecvCodecs(params.codecs) ||
-      !SetRecvRtpHeaderExtensions(params.extensions)) {
-    return false;
-  }
-  if (recv_params_.rtcp.reduced_size != params.rtcp.reduced_size) {
-    rtc::CritScope stream_lock(&stream_crit_);
-    for (auto& kv : receive_streams_) {
-      kv.second->SetRecvParameters(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::SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvCodecs");
-  LOG(LS_INFO) << "SetRecvCodecs: " << CodecVectorToString(codecs);
-  if (!ValidateCodecFormats(codecs)) {
-    return false;
-  }
-
-  const std::vector<VideoCodecSettings> mapped_codecs = MapCodecs(codecs);
-  if (mapped_codecs.empty()) {
-    LOG(LS_ERROR) << "SetRecvCodecs called without any video codecs.";
-    return false;
-  }
-
-  std::vector<VideoCodecSettings> supported_codecs =
-      FilterSupportedCodecs(mapped_codecs);
-
-  if (mapped_codecs.size() != supported_codecs.size()) {
-    LOG(LS_ERROR) << "SetRecvCodecs called with unsupported video codecs.";
-    return false;
-  }
-
-  // Prevent reconfiguration when setting identical receive codecs.
-  if (!ReceiveCodecsHaveChanged(recv_codecs_, supported_codecs)) {
-    LOG(LS_INFO)
-        << "Ignoring call to SetRecvCodecs because codecs haven't changed.";
-    return true;
-  }
-
-  LOG(LS_INFO) << "Changing recv codecs from "
-               << CodecSettingsVectorToString(recv_codecs_) << " to "
-               << CodecSettingsVectorToString(supported_codecs);
-  recv_codecs_ = supported_codecs;
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
-           receive_streams_.begin();
-       it != receive_streams_.end(); ++it) {
-    it->second->SetRecvCodecs(recv_codecs_);
-  }
-
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendCodecs");
-  LOG(LS_INFO) << "SetSendCodecs: " << CodecVectorToString(codecs);
-  if (!ValidateCodecFormats(codecs)) {
-    return false;
-  }
-
-  const std::vector<VideoCodecSettings> supported_codecs =
-      FilterSupportedCodecs(MapCodecs(codecs));
-
-  if (supported_codecs.empty()) {
-    LOG(LS_ERROR) << "No video codecs supported.";
-    return false;
-  }
-
-  LOG(LS_INFO) << "Using codec: " << supported_codecs.front().codec.ToString();
-
-  if (send_codec_ && supported_codecs.front() == *send_codec_) {
-    LOG(LS_INFO) << "Ignore call to SetSendCodecs because first supported "
-                    "codec hasn't changed.";
-    // Using same codec, avoid reconfiguring.
-    return true;
-  }
-
-  send_codec_ = rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(
-      supported_codecs.front());
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  LOG(LS_INFO) << "Change the send codec because SetSendCodecs has a different "
-                  "first supported codec.";
-  for (auto& kv : send_streams_) {
-    RTC_DCHECK(kv.second != nullptr);
-    kv.second->SetCodec(supported_codecs.front());
-  }
-  LOG(LS_INFO)
-      << "SetFeedbackOptions on all the receive streams because the send "
-         "codec has changed.";
-  for (auto& kv : receive_streams_) {
-    RTC_DCHECK(kv.second != nullptr);
-    kv.second->SetFeedbackParameters(
-        HasNack(supported_codecs.front().codec),
-        HasRemb(supported_codecs.front().codec),
-        HasTransportCc(supported_codecs.front().codec));
-  }
-
-  // TODO(holmer): Changing the codec parameters shouldn't necessarily mean that
-  // we change the min/max of bandwidth estimation. Reevaluate this.
-  VideoCodec codec = supported_codecs.front().codec;
-  int bitrate_kbps;
-  if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
-      bitrate_kbps > 0) {
-    bitrate_config_.min_bitrate_bps = bitrate_kbps * 1000;
-  } else {
-    bitrate_config_.min_bitrate_bps = 0;
-  }
-  if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
-      bitrate_kbps > 0) {
-    bitrate_config_.start_bitrate_bps = bitrate_kbps * 1000;
-  } else {
-    // Do not reconfigure start bitrate unless it's specified and positive.
-    bitrate_config_.start_bitrate_bps = -1;
-  }
-  if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
-      bitrate_kbps > 0) {
-    bitrate_config_.max_bitrate_bps = bitrate_kbps * 1000;
-  } else {
-    bitrate_config_.max_bitrate_bps = -1;
-  }
-  call_->SetBitrateConfig(bitrate_config_);
-
-  return true;
-}
-
-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::SetSendStreamFormat(uint32_t ssrc,
-                                              const VideoFormat& format) {
-  LOG(LS_VERBOSE) << "SetSendStreamFormat:" << ssrc << " -> "
-                  << format.ToString();
-  rtc::CritScope stream_lock(&stream_crit_);
-  if (send_streams_.find(ssrc) == send_streams_.end()) {
-    return false;
-  }
-  return send_streams_[ssrc]->SetVideoFormat(format);
-}
-
-bool WebRtcVideoChannel2::SetSend(bool send) {
-  LOG(LS_VERBOSE) << "SetSend: " << (send ? "true" : "false");
-  if (send && !send_codec_) {
-    LOG(LS_ERROR) << "SetSend(true) called before setting codec.";
-    return false;
-  }
-  if (send) {
-    StartAllSendStreams();
-  } else {
-    StopAllSendStreams();
-  }
-  sending_ = send;
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable,
-                                       const VideoOptions* options) {
-  // TODO(solenberg): The state change should be fully rolled back if any one of
-  //                  these calls fail.
-  if (!MuteStream(ssrc, !enable)) {
-    return false;
-  }
-  if (enable && options) {
-    return SetOptions(*options);
-  } else {
-    return true;
-  }
-}
-
-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.overuse_callback = this;
-
-  WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
-      call_, sp, config, external_encoder_factory_, options_,
-      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 (default_send_ssrc_ == 0) {
-    default_send_ssrc_ = ssrc;
-  }
-  if (sending_) {
-    stream->Start();
-  }
-
-  return true;
-}
-
-bool WebRtcVideoChannel2::RemoveSendStream(uint32_t ssrc) {
-  LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
-
-  if (ssrc == 0) {
-    if (default_send_ssrc_ == 0) {
-      LOG(LS_ERROR) << "No default send stream active.";
-      return false;
-    }
-
-    LOG(LS_VERBOSE) << "Removing default stream: " << default_send_ssrc_;
-    ssrc = default_send_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;
-
-  if (ssrc == default_send_ssrc_) {
-    default_send_ssrc_ = 0;
-  }
-
-  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.
-  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);
-  ConfigureReceiverRtp(&config, sp);
-
-  // Set up A/V sync group based on sync label.
-  config.sync_group = sp.sync_label;
-
-  config.rtp.remb = send_codec_ ? HasRemb(send_codec_->codec) : false;
-  config.rtp.transport_cc =
-      send_codec_ ? HasTransportCc(send_codec_->codec) : false;
-
-  receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
-      call_, sp, config, external_decoder_factory_, default_stream,
-      recv_codecs_, options_.disable_prerenderer_smoothing.value_or(false));
-
-  return true;
-}
-
-void WebRtcVideoChannel2::ConfigureReceiverRtp(
-    webrtc::VideoReceiveStream::Config* config,
-    const StreamParams& sp) const {
-  uint32_t ssrc = sp.first_ssrc();
-
-  config->rtp.remote_ssrc = ssrc;
-  config->rtp.local_ssrc = rtcp_receiver_report_ssrc_;
-
-  config->rtp.extensions = recv_rtp_extensions_;
-  config->rtp.rtcp_mode = recv_params_.rtcp.reduced_size
-                              ? webrtc::RtcpMode::kReducedSize
-                              : webrtc::RtcpMode::kCompound;
-
-  // 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;
-    }
-  }
-
-  for (size_t i = 0; i < recv_codecs_.size(); ++i) {
-    MergeFecConfig(recv_codecs_[i].fec, &config->rtp.fec);
-  }
-
-  for (size_t i = 0; i < recv_codecs_.size(); ++i) {
-    uint32_t rtx_ssrc;
-    if (recv_codecs_[i].rtx_payload_type != -1 &&
-        sp.GetFidSsrc(ssrc, &rtx_ssrc)) {
-      webrtc::VideoReceiveStream::Config::Rtp::Rtx& rtx =
-          config->rtp.rtx[recv_codecs_[i].codec.id];
-      rtx.ssrc = rtx_ssrc;
-      rtx.payload_type = recv_codecs_[i].rtx_payload_type;
-    }
-  }
-}
-
-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::SetRenderer(uint32_t ssrc, VideoRenderer* renderer) {
-  LOG(LS_INFO) << "SetRenderer: ssrc:" << ssrc << " "
-               << (renderer ? "(ptr)" : "NULL");
-  if (ssrc == 0) {
-    default_unsignalled_ssrc_handler_.SetDefaultRenderer(this, renderer);
-    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->SetRenderer(renderer);
-  return true;
-}
-
-bool WebRtcVideoChannel2::GetRenderer(uint32_t ssrc, VideoRenderer** renderer) {
-  if (ssrc == 0) {
-    *renderer = default_unsignalled_ssrc_handler_.GetDefaultRenderer();
-    return *renderer != NULL;
-  }
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
-      receive_streams_.find(ssrc);
-  if (it == receive_streams_.end()) {
-    return false;
-  }
-  *renderer = it->second->GetRenderer();
-  return true;
-}
-
-bool WebRtcVideoChannel2::GetStats(VideoMediaInfo* info) {
-  info->Clear();
-  FillSenderStats(info);
-  FillReceiverStats(info);
-  webrtc::Call::Stats stats = call_->GetStats();
-  FillBandwidthEstimationStats(stats, info);
-  if (stats.rtt_ms != -1) {
-    for (size_t i = 0; i < info->senders.size(); ++i) {
-      info->senders[i].rtt_ms = stats.rtt_ms;
-    }
-  }
-  return true;
-}
-
-void WebRtcVideoChannel2::FillSenderStats(VideoMediaInfo* video_media_info) {
-  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());
-  }
-}
-
-void WebRtcVideoChannel2::FillReceiverStats(VideoMediaInfo* video_media_info) {
-  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());
-  }
-}
-
-void WebRtcVideoChannel2::FillBandwidthEstimationStats(
-    const webrtc::Call::Stats& stats,
-    VideoMediaInfo* video_media_info) {
-  BandwidthEstimationInfo bwe_info;
-  bwe_info.available_send_bandwidth = stats.send_bandwidth_bps;
-  bwe_info.available_recv_bandwidth = stats.recv_bandwidth_bps;
-  bwe_info.bucket_delay = stats.pacer_delay_ms;
-
-  // Get send stream bitrate stats.
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator stream =
-           send_streams_.begin();
-       stream != send_streams_.end(); ++stream) {
-    stream->second->FillBandwidthEstimationInfo(&bwe_info);
-  }
-  video_media_info->bw_estimations.push_back(bwe_info);
-}
-
-bool WebRtcVideoChannel2::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
-  LOG(LS_INFO) << "SetCapturer: " << ssrc << " -> "
-               << (capturer != NULL ? "(capturer)" : "NULL");
-  RTC_DCHECK(ssrc != 0);
-  {
-    rtc::CritScope stream_lock(&stream_crit_);
-    if (send_streams_.find(ssrc) == send_streams_.end()) {
-      LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-      return false;
-    }
-    if (!send_streams_[ssrc]->SetCapturer(capturer)) {
-      return false;
-    }
-  }
-
-  if (capturer) {
-    capturer->SetApplyRotation(
-        !FindHeaderExtension(send_rtp_extensions_,
-                             kRtpVideoRotationHeaderExtension));
-  }
-  {
-    rtc::CritScope lock(&capturer_crit_);
-    capturers_[ssrc] = capturer;
-  }
-  return true;
-}
-
-bool WebRtcVideoChannel2::SendIntraFrame() {
-  // TODO(pbos): Implement.
-  LOG(LS_VERBOSE) << "SendIntraFrame().";
-  return true;
-}
-
-bool WebRtcVideoChannel2::RequestIntraFrame() {
-  // TODO(pbos): Implement.
-  LOG(LS_VERBOSE) << "SendIntraFrame().";
-  return true;
-}
-
-void WebRtcVideoChannel2::OnPacketReceived(
-    rtc::Buffer* 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,
-          reinterpret_cast<const uint8_t*>(packet->data()), 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->data(), packet->size(), &ssrc)) {
-    return;
-  }
-
-  int payload_type = 0;
-  if (!GetRtpPayloadType(packet->data(), 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 (ULPFEC). 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.fec.red_rtx_payload_type ||
-        payload_type == codec.fec.ulpfec_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,
-          reinterpret_cast<const uint8_t*>(packet->data()), 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::Buffer* 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,
-      reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
-      webrtc_packet_time);
-}
-
-void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
-  LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
-  call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
-}
-
-bool WebRtcVideoChannel2::MuteStream(uint32_t ssrc, bool mute) {
-  LOG(LS_VERBOSE) << "MuteStream: " << ssrc << " -> "
-                  << (mute ? "mute" : "unmute");
-  RTC_DCHECK(ssrc != 0);
-  rtc::CritScope stream_lock(&stream_crit_);
-  if (send_streams_.find(ssrc) == send_streams_.end()) {
-    LOG(LS_ERROR) << "No sending stream on ssrc " << ssrc;
-    return false;
-  }
-
-  send_streams_[ssrc]->MuteStream(mute);
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetRecvRtpHeaderExtensions(
-    const std::vector<RtpHeaderExtension>& extensions) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetRecvRtpHeaderExtensions");
-  if (!ValidateRtpExtensions(extensions)) {
-    return false;
-  }
-  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
-      extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
-  if (recv_rtp_extensions_ == filtered_extensions) {
-    LOG(LS_INFO) << "Ignoring call to SetRecvRtpHeaderExtensions because "
-                    "header extensions haven't changed.";
-    return true;
-  }
-  recv_rtp_extensions_.swap(filtered_extensions);
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
-           receive_streams_.begin();
-       it != receive_streams_.end(); ++it) {
-    it->second->SetRtpExtensions(recv_rtp_extensions_);
-  }
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetSendRtpHeaderExtensions(
-    const std::vector<RtpHeaderExtension>& extensions) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetSendRtpHeaderExtensions");
-  if (!ValidateRtpExtensions(extensions)) {
-    return false;
-  }
-  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
-      extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
-  if (send_rtp_extensions_ == filtered_extensions) {
-    LOG(LS_INFO) << "Ignoring call to SetRecvRtpHeaderExtensions because "
-                    "header extensions haven't changed.";
-    return true;
-  }
-  send_rtp_extensions_.swap(filtered_extensions);
-
-  const webrtc::RtpExtension* cvo_extension = FindHeaderExtension(
-      send_rtp_extensions_, kRtpVideoRotationHeaderExtension);
-
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-           send_streams_.begin();
-       it != send_streams_.end(); ++it) {
-    it->second->SetRtpExtensions(send_rtp_extensions_);
-    it->second->SetApplyRotation(!cvo_extension);
-  }
-  return true;
-}
-
-// Counter-intuitively this method doesn't only set global bitrate caps but also
-// per-stream codec max bitrates. This is to permit SetMaxSendBitrate (b=AS) to
-// raise bitrates above the 2000k default bitrate cap.
-bool WebRtcVideoChannel2::SetMaxSendBandwidth(int max_bitrate_bps) {
-  // 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.
-  LOG(LS_INFO) << "SetMaxSendBandwidth: " << max_bitrate_bps << "bps.";
-  if (max_bitrate_bps == bitrate_config_.max_bitrate_bps)
-    return true;
-
-  if (max_bitrate_bps < 0) {
-    // Option not set.
-    return true;
-  }
-  if (max_bitrate_bps == 0) {
-    // Unsetting max bitrate.
-    max_bitrate_bps = -1;
-  }
-  bitrate_config_.start_bitrate_bps = -1;
-  bitrate_config_.max_bitrate_bps = max_bitrate_bps;
-  if (max_bitrate_bps > 0 &&
-      bitrate_config_.min_bitrate_bps > max_bitrate_bps) {
-    bitrate_config_.min_bitrate_bps = max_bitrate_bps;
-  }
-  call_->SetBitrateConfig(bitrate_config_);
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (auto& kv : send_streams_)
-    kv.second->SetMaxBitrateBps(max_bitrate_bps);
-  return true;
-}
-
-bool WebRtcVideoChannel2::SetOptions(const VideoOptions& options) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoChannel2::SetOptions");
-  LOG(LS_INFO) << "SetOptions: " << options.ToString();
-  VideoOptions old_options = options_;
-  options_.SetAll(options);
-  if (options_ == old_options) {
-    // No new options to set.
-    return true;
-  }
-  {
-    rtc::CritScope lock(&capturer_crit_);
-    if (options_.cpu_overuse_detection)
-      signal_cpu_adaptation_ = *options_.cpu_overuse_detection;
-  }
-  rtc::DiffServCodePoint dscp =
-      options_.dscp.value_or(false) ? rtc::DSCP_AF41 : rtc::DSCP_DEFAULT;
-  MediaChannel::SetDscp(dscp);
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-           send_streams_.begin();
-       it != send_streams_.end(); ++it) {
-    it->second->SetOptions(options_);
-  }
-  return true;
-}
-
-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);
-}
-
-void WebRtcVideoChannel2::UpdateAspectRatio(int ratio_w, int ratio_h) {
-  // TODO(pbos): Implement.
-}
-
-void WebRtcVideoChannel2::OnMessage(rtc::Message* msg) {
-  // Ignored.
-}
-
-void WebRtcVideoChannel2::OnLoadUpdate(Load load) {
-  // OnLoadUpdate can not take any locks that are held while creating streams
-  // etc. Doing so establishes lock-order inversions between the webrtc process
-  // thread on stream creation and locks such as stream_crit_ while calling out.
-  rtc::CritScope stream_lock(&capturer_crit_);
-  if (!signal_cpu_adaptation_)
-    return;
-  // Do not adapt resolution for screen content as this will likely result in
-  // blurry and unreadable text.
-  for (auto& kv : capturers_) {
-    if (kv.second != nullptr
-        && !kv.second->IsScreencast()
-        && kv.second->video_adapter() != nullptr) {
-      kv.second->video_adapter()->OnCpuResolutionRequest(
-          load == kOveruse ? CoordinatedVideoAdapter::DOWNGRADE
-                           : CoordinatedVideoAdapter::UPGRADE);
-    }
-  }
-}
-
-bool WebRtcVideoChannel2::SendRtp(const uint8_t* data,
-                                  size_t len,
-                                  const webrtc::PacketOptions& options) {
-  rtc::Buffer 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::Buffer packet(data, len, kMaxRtpPacketLen);
-  return MediaChannel::SendRtcp(&packet, rtc::PacketOptions());
-}
-
-void WebRtcVideoChannel2::StartAllSendStreams() {
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-           send_streams_.begin();
-       it != send_streams_.end(); ++it) {
-    it->second->Start();
-  }
-}
-
-void WebRtcVideoChannel2::StopAllSendStreams() {
-  rtc::CritScope stream_lock(&stream_crit_);
-  for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
-           send_streams_.begin();
-       it != send_streams_.end(); ++it) {
-    it->second->Stop();
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::VideoSendStreamParameters::
-    VideoSendStreamParameters(
-        const webrtc::VideoSendStream::Config& config,
-        const VideoOptions& options,
-        int max_bitrate_bps,
-        const rtc::Optional<VideoCodecSettings>& codec_settings)
-    : config(config),
-      options(options),
-      max_bitrate_bps(max_bitrate_bps),
-      codec_settings(codec_settings) {}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder::AllocatedEncoder(
-    webrtc::VideoEncoder* encoder,
-    webrtc::VideoCodecType type,
-    bool external)
-    : encoder(encoder),
-      external_encoder(nullptr),
-      type(type),
-      external(external) {
-  if (external) {
-    external_encoder = encoder;
-    this->encoder =
-        new webrtc::VideoEncoderSoftwareFallbackWrapper(type, encoder);
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
-    webrtc::Call* call,
-    const StreamParams& sp,
-    const webrtc::VideoSendStream::Config& config,
-    WebRtcVideoEncoderFactory* external_encoder_factory,
-    const VideoOptions& options,
-    int max_bitrate_bps,
-    const rtc::Optional<VideoCodecSettings>& codec_settings,
-    const std::vector<webrtc::RtpExtension>& rtp_extensions,
-    // TODO(deadbeef): Don't duplicate information between send_params,
-    // rtp_extensions, options, etc.
-    const VideoSendParameters& send_params)
-    : ssrcs_(sp.ssrcs),
-      ssrc_groups_(sp.ssrc_groups),
-      call_(call),
-      external_encoder_factory_(external_encoder_factory),
-      stream_(NULL),
-      parameters_(config, options, max_bitrate_bps, codec_settings),
-      allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
-      capturer_(NULL),
-      sending_(false),
-      muted_(false),
-      old_adapt_changes_(0),
-      first_frame_timestamp_ms_(0),
-      last_frame_timestamp_ms_(0) {
-  parameters_.config.rtp.max_packet_size = kVideoMtu;
-
-  sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
-  sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
-                 &parameters_.config.rtp.rtx.ssrcs);
-  parameters_.config.rtp.c_name = sp.cname;
-  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) {
-    SetCodec(*codec_settings);
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
-  DisconnectCapturer();
-  if (stream_ != NULL) {
-    call_->DestroyVideoSendStream(stream_);
-  }
-  DestroyVideoEncoder(&allocated_encoder_);
-}
-
-static void CreateBlackFrame(webrtc::VideoFrame* video_frame,
-                             int width,
-                             int height) {
-  video_frame->CreateEmptyFrame(width, height, width, (width + 1) / 2,
-                                (width + 1) / 2);
-  memset(video_frame->buffer(webrtc::kYPlane), 16,
-         video_frame->allocated_size(webrtc::kYPlane));
-  memset(video_frame->buffer(webrtc::kUPlane), 128,
-         video_frame->allocated_size(webrtc::kUPlane));
-  memset(video_frame->buffer(webrtc::kVPlane), 128,
-         video_frame->allocated_size(webrtc::kVPlane));
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
-    VideoCapturer* capturer,
-    const VideoFrame* frame) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame");
-  webrtc::VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0,
-                                 frame->GetVideoRotation());
-  rtc::CritScope cs(&lock_);
-  if (stream_ == NULL) {
-    // Frame input before send codecs are configured, dropping frame.
-    return;
-  }
-
-  // Not sending, abort early to prevent expensive reconfigurations while
-  // setting up codecs etc.
-  if (!sending_)
-    return;
-
-  if (format_.width == 0) {  // Dropping frames.
-    RTC_DCHECK(format_.height == 0);
-    LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
-    return;
-  }
-  if (muted_) {
-    // Create a black frame to transmit instead.
-    CreateBlackFrame(&video_frame,
-                     static_cast<int>(frame->GetWidth()),
-                     static_cast<int>(frame->GetHeight()));
-  }
-
-  int64_t frame_delta_ms = frame->GetTimeStamp() / rtc::kNumNanosecsPerMillisec;
-  // frame->GetTimeStamp() is essentially a delta, align to webrtc time
-  if (first_frame_timestamp_ms_ == 0) {
-    first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms;
-  }
-
-  last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms;
-  video_frame.set_render_time_ms(last_frame_timestamp_ms_);
-  // Reconfigure codec if necessary.
-  SetDimensions(
-      video_frame.width(), video_frame.height(), capturer->IsScreencast());
-
-  stream_->Input()->IncomingCapturedFrame(video_frame);
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
-    VideoCapturer* capturer) {
-  TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
-  if (!DisconnectCapturer() && capturer == NULL) {
-    return false;
-  }
-
-  {
-    rtc::CritScope cs(&lock_);
-
-    // Reset timestamps to realign new incoming frames to a webrtc timestamp. A
-    // new capturer may have a different timestamp delta than the previous one.
-    first_frame_timestamp_ms_ = 0;
-
-    if (capturer == NULL) {
-      if (stream_ != NULL) {
-        LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
-        webrtc::VideoFrame black_frame;
-
-        CreateBlackFrame(&black_frame, last_dimensions_.width,
-                         last_dimensions_.height);
-
-        // Force this black frame not to be dropped due to timestamp order
-        // check. As IncomingCapturedFrame will drop the frame if this frame's
-        // timestamp is less than or equal to last frame's timestamp, it is
-        // necessary to give this black frame a larger timestamp than the
-        // previous one.
-        last_frame_timestamp_ms_ +=
-            format_.interval / rtc::kNumNanosecsPerMillisec;
-        black_frame.set_render_time_ms(last_frame_timestamp_ms_);
-        stream_->Input()->IncomingCapturedFrame(black_frame);
-      }
-
-      capturer_ = NULL;
-      return true;
-    }
-
-    capturer_ = capturer;
-  }
-  // Lock cannot be held while connecting the capturer to prevent lock-order
-  // violations.
-  capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame);
-  return true;
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetVideoFormat(
-    const VideoFormat& format) {
-  if ((format.width == 0 || format.height == 0) &&
-      format.width != format.height) {
-    LOG(LS_ERROR) << "Can't set VideoFormat, width or height is zero (but not "
-                     "both, 0x0 drops frames).";
-    return false;
-  }
-
-  rtc::CritScope cs(&lock_);
-  if (format.width == 0 && format.height == 0) {
-    LOG(LS_INFO)
-        << "0x0 resolution selected. Captured frames will be dropped for ssrc: "
-        << parameters_.config.rtp.ssrcs[0] << ".";
-  } else {
-    // TODO(pbos): Fix me, this only affects the last stream!
-    parameters_.encoder_config.streams.back().max_framerate =
-        VideoFormat::IntervalToFps(format.interval);
-    SetDimensions(format.width, format.height, false);
-  }
-
-  format_ = format;
-  return true;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::MuteStream(bool mute) {
-  rtc::CritScope cs(&lock_);
-  muted_ = mute;
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() {
-  cricket::VideoCapturer* capturer;
-  {
-    rtc::CritScope cs(&lock_);
-    if (capturer_ == NULL)
-      return false;
-
-    if (capturer_->video_adapter() != nullptr)
-      old_adapt_changes_ += capturer_->video_adapter()->adaptation_changes();
-
-    capturer = capturer_;
-    capturer_ = NULL;
-  }
-  capturer->SignalVideoFrame.disconnect(this);
-  return true;
-}
-
-const std::vector<uint32_t>&
-WebRtcVideoChannel2::WebRtcVideoSendStream::GetSsrcs() const {
-  return ssrcs_;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetApplyRotation(
-    bool apply_rotation) {
-  rtc::CritScope cs(&lock_);
-  if (capturer_ == NULL)
-    return;
-
-  capturer_->SetApplyRotation(apply_rotation);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetOptions(
-    const VideoOptions& options) {
-  rtc::CritScope cs(&lock_);
-  if (parameters_.codec_settings) {
-    LOG(LS_INFO) << "SetCodecAndOptions because of SetOptions; options="
-                 << options.ToString();
-    SetCodecAndOptions(*parameters_.codec_settings, options);
-  } else {
-    parameters_.options = options;
-  }
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodec(
-    const VideoCodecSettings& codec_settings) {
-  rtc::CritScope cs(&lock_);
-  LOG(LS_INFO) << "SetCodecAndOptions because of SetCodec.";
-  SetCodecAndOptions(codec_settings, parameters_.options);
-}
-
-webrtc::VideoCodecType CodecTypeFromName(const std::string& name) {
-  if (CodecNamesEq(name, kVp8CodecName)) {
-    return webrtc::kVideoCodecVP8;
-  } else if (CodecNamesEq(name, kVp9CodecName)) {
-    return webrtc::kVideoCodecVP9;
-  } else if (CodecNamesEq(name, kH264CodecName)) {
-    return webrtc::kVideoCodecH264;
-  }
-  return webrtc::kVideoCodecUnknown;
-}
-
-WebRtcVideoChannel2::WebRtcVideoSendStream::AllocatedEncoder
-WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoder(
-    const VideoCodec& codec) {
-  webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
-
-  // Do not re-create encoders of the same type.
-  if (type == allocated_encoder_.type && allocated_encoder_.encoder != NULL) {
-    return allocated_encoder_;
-  }
-
-  if (external_encoder_factory_ != NULL) {
-    webrtc::VideoEncoder* encoder =
-        external_encoder_factory_->CreateVideoEncoder(type);
-    if (encoder != NULL) {
-      return AllocatedEncoder(encoder, type, true);
-    }
-  }
-
-  if (type == webrtc::kVideoCodecVP8) {
-    return AllocatedEncoder(
-        webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp8), type, false);
-  } else if (type == webrtc::kVideoCodecVP9) {
-    return AllocatedEncoder(
-        webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kVp9), type, false);
-  } else if (type == webrtc::kVideoCodecH264) {
-    return AllocatedEncoder(
-        webrtc::VideoEncoder::Create(webrtc::VideoEncoder::kH264), type, false);
-  }
-
-  // This shouldn't happen, we should not be trying to create something we don't
-  // support.
-  RTC_DCHECK(false);
-  return AllocatedEncoder(NULL, webrtc::kVideoCodecUnknown, false);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::DestroyVideoEncoder(
-    AllocatedEncoder* encoder) {
-  if (encoder->external) {
-    external_encoder_factory_->DestroyVideoEncoder(encoder->external_encoder);
-  }
-  delete encoder->encoder;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetCodecAndOptions(
-    const VideoCodecSettings& codec_settings,
-    const VideoOptions& options) {
-  parameters_.encoder_config =
-      CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
-  if (parameters_.encoder_config.streams.empty())
-    return;
-
-  format_ = VideoFormat(codec_settings.codec.width,
-                        codec_settings.codec.height,
-                        VideoFormat::FpsToInterval(30),
-                        FOURCC_I420);
-
-  AllocatedEncoder new_encoder = CreateVideoEncoder(codec_settings.codec);
-  parameters_.config.encoder_settings.encoder = new_encoder.encoder;
-  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 = CodecTypeFromName(codec_settings.codec.name);
-    parameters_.config.encoder_settings.internal_source =
-        external_encoder_factory_->EncoderTypeHasInternalSource(type);
-  }
-  parameters_.config.rtp.fec = codec_settings.fec;
-
-  // 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;
-
-  RTC_CHECK(options.suspend_below_min_bitrate);
-  parameters_.config.suspend_below_min_bitrate =
-      *options.suspend_below_min_bitrate;
-
-  parameters_.codec_settings =
-      rtc::Optional<WebRtcVideoChannel2::VideoCodecSettings>(codec_settings);
-  parameters_.options = options;
-
-  LOG(LS_INFO)
-      << "RecreateWebRtcStream (send) because of SetCodecAndOptions; options="
-      << options.ToString();
-  RecreateWebRtcStream();
-  if (allocated_encoder_.encoder != new_encoder.encoder) {
-    DestroyVideoEncoder(&allocated_encoder_);
-    allocated_encoder_ = new_encoder;
-  }
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetRtpExtensions(
-    const std::vector<webrtc::RtpExtension>& rtp_extensions) {
-  rtc::CritScope cs(&lock_);
-  parameters_.config.rtp.extensions = rtp_extensions;
-  if (stream_ != nullptr) {
-    LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetRtpExtensions";
-    RecreateWebRtcStream();
-  }
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
-    const VideoSendParameters& send_params) {
-  rtc::CritScope cs(&lock_);
-  parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
-                                         ? webrtc::RtcpMode::kReducedSize
-                                         : webrtc::RtcpMode::kCompound;
-  if (stream_ != nullptr) {
-    LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
-    RecreateWebRtcStream();
-  }
-}
-
-webrtc::VideoEncoderConfig
-WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
-    const Dimensions& dimensions,
-    const VideoCodec& codec) const {
-  webrtc::VideoEncoderConfig encoder_config;
-  if (dimensions.is_screencast) {
-    RTC_CHECK(parameters_.options.screencast_min_bitrate);
-    encoder_config.min_transmit_bitrate_bps =
-        *parameters_.options.screencast_min_bitrate * 1000;
-    encoder_config.content_type =
-        webrtc::VideoEncoderConfig::ContentType::kScreen;
-  } else {
-    encoder_config.min_transmit_bitrate_bps = 0;
-    encoder_config.content_type =
-        webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo;
-  }
-
-  // Restrict dimensions according to codec max.
-  int width = dimensions.width;
-  int height = dimensions.height;
-  if (!dimensions.is_screencast) {
-    if (codec.width < width)
-      width = codec.width;
-    if (codec.height < height)
-      height = codec.height;
-  }
-
-  VideoCodec clamped_codec = codec;
-  clamped_codec.width = width;
-  clamped_codec.height = height;
-
-  // 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, only configure a single stream.
-  size_t stream_count = parameters_.config.rtp.ssrcs.size();
-  if (IsCodecBlacklistedForSimulcast(codec.name) || dimensions.is_screencast) {
-    stream_count = 1;
-  }
-
-  encoder_config.streams =
-      CreateVideoStreams(clamped_codec, parameters_.options,
-                         parameters_.max_bitrate_bps, stream_count);
-
-  // Conference mode screencast uses 2 temporal layers split at 100kbit.
-  if (parameters_.options.conference_mode.value_or(false) &&
-      dimensions.is_screencast && encoder_config.streams.size() == 1) {
-    ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
-
-    // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
-    // on the VideoCodec struct as target and max bitrates, respectively.
-    // See eg. webrtc::VP8EncoderImpl::SetRates().
-    encoder_config.streams[0].target_bitrate_bps =
-        config.tl0_bitrate_kbps * 1000;
-    encoder_config.streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
-    encoder_config.streams[0].temporal_layer_thresholds_bps.clear();
-    encoder_config.streams[0].temporal_layer_thresholds_bps.push_back(
-        config.tl0_bitrate_kbps * 1000);
-  }
-  return encoder_config;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::SetDimensions(
-    int width,
-    int height,
-    bool is_screencast) {
-  if (last_dimensions_.width == width && last_dimensions_.height == height &&
-      last_dimensions_.is_screencast == is_screencast) {
-    // Configured using the same parameters, do not reconfigure.
-    return;
-  }
-  LOG(LS_INFO) << "SetDimensions: " << width << "x" << height
-               << (is_screencast ? " (screencast)" : " (not screencast)");
-
-  last_dimensions_.width = width;
-  last_dimensions_.height = height;
-  last_dimensions_.is_screencast = is_screencast;
-
-  RTC_DCHECK(!parameters_.encoder_config.streams.empty());
-
-  RTC_CHECK(parameters_.codec_settings);
-  VideoCodecSettings codec_settings = *parameters_.codec_settings;
-
-  webrtc::VideoEncoderConfig encoder_config =
-      CreateVideoEncoderConfig(last_dimensions_, codec_settings.codec);
-
-  encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
-      codec_settings.codec, parameters_.options, is_screencast);
-
-  bool stream_reconfigured = stream_->ReconfigureVideoEncoder(encoder_config);
-
-  encoder_config.encoder_specific_settings = NULL;
-
-  if (!stream_reconfigured) {
-    LOG(LS_WARNING) << "Failed to reconfigure video encoder for dimensions: "
-                    << width << "x" << height;
-    return;
-  }
-
-  parameters_.encoder_config = encoder_config;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::Start() {
-  rtc::CritScope cs(&lock_);
-  RTC_DCHECK(stream_ != NULL);
-  stream_->Start();
-  sending_ = true;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::Stop() {
-  rtc::CritScope cs(&lock_);
-  if (stream_ != NULL) {
-    stream_->Stop();
-  }
-  sending_ = false;
-}
-
-VideoSenderInfo
-WebRtcVideoChannel2::WebRtcVideoSendStream::GetVideoSenderInfo() {
-  VideoSenderInfo info;
-  webrtc::VideoSendStream::Stats stats;
-  {
-    rtc::CritScope cs(&lock_);
-    for (uint32_t ssrc : parameters_.config.rtp.ssrcs)
-      info.add_ssrc(ssrc);
-
-    if (parameters_.codec_settings)
-      info.codec_name = parameters_.codec_settings->codec.name;
-    for (size_t i = 0; i < parameters_.encoder_config.streams.size(); ++i) {
-      if (i == parameters_.encoder_config.streams.size() - 1) {
-        info.preferred_bitrate +=
-            parameters_.encoder_config.streams[i].max_bitrate_bps;
-      } else {
-        info.preferred_bitrate +=
-            parameters_.encoder_config.streams[i].target_bitrate_bps;
-      }
-    }
-
-    if (stream_ == NULL)
-      return info;
-
-    stats = stream_->GetStats();
-
-    info.adapt_changes = old_adapt_changes_;
-    info.adapt_reason = CoordinatedVideoAdapter::ADAPTREASON_NONE;
-
-    if (capturer_ != NULL) {
-      if (!capturer_->IsMuted()) {
-        VideoFormat last_captured_frame_format;
-        capturer_->GetStats(&info.adapt_frame_drops, &info.effects_frame_drops,
-                            &info.capturer_frame_time,
-                            &last_captured_frame_format);
-        info.input_frame_width = last_captured_frame_format.width;
-        info.input_frame_height = last_captured_frame_format.height;
-      }
-      if (capturer_->video_adapter() != nullptr) {
-        info.adapt_changes += capturer_->video_adapter()->adaptation_changes();
-        info.adapt_reason = capturer_->video_adapter()->adapt_reason();
-      }
-    }
-  }
-
-  // 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 |= CoordinatedVideoAdapter::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.nominal_bitrate = stats.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::FillBandwidthEstimationInfo(
-    BandwidthEstimationInfo* bwe_info) {
-  rtc::CritScope cs(&lock_);
-  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::SetMaxBitrateBps(
-    int max_bitrate_bps) {
-  rtc::CritScope cs(&lock_);
-  parameters_.max_bitrate_bps = max_bitrate_bps;
-
-  // No need to reconfigure if the stream hasn't been configured yet.
-  if (parameters_.encoder_config.streams.empty())
-    return;
-
-  // Force a stream reconfigure to set the new max bitrate.
-  int width = last_dimensions_.width;
-  last_dimensions_.width = 0;
-  SetDimensions(width, last_dimensions_.height, last_dimensions_.is_screencast);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoSendStream::RecreateWebRtcStream() {
-  if (stream_ != NULL) {
-    call_->DestroyVideoSendStream(stream_);
-  }
-
-  RTC_CHECK(parameters_.codec_settings);
-  parameters_.encoder_config.encoder_specific_settings =
-      ConfigureVideoEncoderSettings(
-          parameters_.codec_settings->codec, parameters_.options,
-          parameters_.encoder_config.content_type ==
-              webrtc::VideoEncoderConfig::ContentType::kScreen);
-
-  webrtc::VideoSendStream::Config config = parameters_.config;
-  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(config, parameters_.encoder_config);
-
-  parameters_.encoder_config.encoder_specific_settings = NULL;
-
-  if (sending_) {
-    stream_->Start();
-  }
-}
-
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::WebRtcVideoReceiveStream(
-    webrtc::Call* call,
-    const StreamParams& sp,
-    const webrtc::VideoReceiveStream::Config& config,
-    WebRtcVideoDecoderFactory* external_decoder_factory,
-    bool default_stream,
-    const std::vector<VideoCodecSettings>& recv_codecs,
-    bool disable_prerenderer_smoothing)
-    : call_(call),
-      ssrcs_(sp.ssrcs),
-      ssrc_groups_(sp.ssrc_groups),
-      stream_(NULL),
-      default_stream_(default_stream),
-      config_(config),
-      external_decoder_factory_(external_decoder_factory),
-      disable_prerenderer_smoothing_(disable_prerenderer_smoothing),
-      renderer_(NULL),
-      last_width_(-1),
-      last_height_(-1),
-      first_frame_timestamp_(-1),
-      estimated_remote_start_ntp_time_ms_(0) {
-  config_.renderer = this;
-  // SetRecvCodecs will also reset (start) the VideoReceiveStream.
-  LOG(LS_INFO) << "SetRecvCodecs (recv) because we are creating the receive "
-                  "stream for the first time: "
-               << CodecSettingsVectorToString(recv_codecs);
-  SetRecvCodecs(recv_codecs);
-}
-
-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() {
-  call_->DestroyVideoReceiveStream(stream_);
-  ClearDecoders(&allocated_decoders_);
-}
-
-const std::vector<uint32_t>&
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetSsrcs() const {
-  return ssrcs_;
-}
-
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::AllocatedDecoder
-WebRtcVideoChannel2::WebRtcVideoReceiveStream::CreateOrReuseVideoDecoder(
-    std::vector<AllocatedDecoder>* old_decoders,
-    const VideoCodec& codec) {
-  webrtc::VideoCodecType type = CodecTypeFromName(codec.name);
-
-  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_->CreateVideoDecoder(type);
-    if (decoder != NULL) {
-      return AllocatedDecoder(decoder, type, true);
-    }
-  }
-
-  if (type == webrtc::kVideoCodecVP8) {
-    return AllocatedDecoder(
-        webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp8), type, false);
-  }
-
-  if (type == webrtc::kVideoCodecVP9) {
-    return AllocatedDecoder(
-        webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kVp9), type, false);
-  }
-
-  if (type == webrtc::kVideoCodecH264) {
-    return AllocatedDecoder(
-        webrtc::VideoDecoder::Create(webrtc::VideoDecoder::kH264), type, false);
-  }
-
-  // This shouldn't happen, we should not be trying to create something we don't
-  // support.
-  RTC_DCHECK(false);
-  return AllocatedDecoder(NULL, webrtc::kVideoCodecUnknown, false);
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvCodecs(
-    const std::vector<VideoCodecSettings>& recv_codecs) {
-  std::vector<AllocatedDecoder> 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;
-    config_.decoders.push_back(decoder);
-  }
-
-  // TODO(pbos): Reconfigure RTX based on incoming recv_codecs.
-  config_.rtp.fec = recv_codecs.front().fec;
-  config_.rtp.nack.rtp_history_ms =
-      HasNack(recv_codecs.begin()->codec) ? kNackHistoryMs : 0;
-
-  LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvCodecs: "
-               << CodecSettingsVectorToString(recv_codecs);
-  RecreateWebRtcStream();
-  ClearDecoders(&old_decoders);
-}
-
-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;
-  LOG(LS_INFO)
-      << "RecreateWebRtcStream (recv) because of SetLocalSsrc; local_ssrc="
-      << local_ssrc;
-  RecreateWebRtcStream();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetFeedbackParameters(
-    bool nack_enabled,
-    bool remb_enabled,
-    bool transport_cc_enabled) {
-  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) {
-    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;
-  LOG(LS_INFO)
-      << "RecreateWebRtcStream (recv) because of SetFeedbackParameters; nack="
-      << nack_enabled << ", remb=" << remb_enabled
-      << ", transport_cc=" << transport_cc_enabled;
-  RecreateWebRtcStream();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRtpExtensions(
-    const std::vector<webrtc::RtpExtension>& extensions) {
-  config_.rtp.extensions = extensions;
-  LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRtpExtensions";
-  RecreateWebRtcStream();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRecvParameters(
-    const VideoRecvParameters& recv_params) {
-  config_.rtp.rtcp_mode = recv_params.rtcp.reduced_size
-                              ? webrtc::RtcpMode::kReducedSize
-                              : webrtc::RtcpMode::kCompound;
-  LOG(LS_INFO) << "RecreateWebRtcStream (recv) because of SetRecvParameters";
-  RecreateWebRtcStream();
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::RecreateWebRtcStream() {
-  if (stream_ != NULL) {
-    call_->DestroyVideoReceiveStream(stream_);
-  }
-  stream_ = call_->CreateVideoReceiveStream(config_);
-  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::RenderFrame(
-    const webrtc::VideoFrame& frame,
-    int time_to_render_ms) {
-  rtc::CritScope crit(&renderer_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 (renderer_ == NULL) {
-    LOG(LS_WARNING) << "VideoReceiveStream not connected to a VideoRenderer.";
-    return;
-  }
-
-  if (frame.width() != last_width_ || frame.height() != last_height_) {
-    SetSize(frame.width(), frame.height());
-  }
-
-  const WebRtcVideoFrame render_frame(
-      frame.video_frame_buffer(),
-      frame.render_time_ms() * rtc::kNumNanosecsPerMillisec, frame.rotation());
-  renderer_->RenderFrame(&render_frame);
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsTextureSupported() const {
-  return true;
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::SmoothsRenderedFrames()
-    const {
-  return disable_prerenderer_smoothing_;
-}
-
-bool WebRtcVideoChannel2::WebRtcVideoReceiveStream::IsDefaultStream() const {
-  return default_stream_;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetRenderer(
-    cricket::VideoRenderer* renderer) {
-  rtc::CritScope crit(&renderer_lock_);
-  renderer_ = renderer;
-  if (renderer_ != NULL && last_width_ != -1) {
-    SetSize(last_width_, last_height_);
-  }
-}
-
-VideoRenderer* WebRtcVideoChannel2::WebRtcVideoReceiveStream::GetRenderer() {
-  // TODO(pbos): Remove GetRenderer and all uses of it, it's thread-unsafe by
-  // design.
-  rtc::CritScope crit(&renderer_lock_);
-  return renderer_;
-}
-
-void WebRtcVideoChannel2::WebRtcVideoReceiveStream::SetSize(int width,
-                                                            int height) {
-  rtc::CritScope crit(&renderer_lock_);
-  if (!renderer_->SetSize(width, height, 0)) {
-    LOG(LS_ERROR) << "Could not set renderer size.";
-  }
-  last_width_ = width;
-  last_height_ = height;
-}
-
-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() {
-  VideoReceiverInfo info;
-  info.ssrc_groups = ssrc_groups_;
-  info.add_ssrc(config_.rtp.remote_ssrc);
-  webrtc::VideoReceiveStream::Stats stats = stream_->GetStats();
-  info.decoder_implementation_name = stats.decoder_implementation_name;
-  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;
-
-  {
-    rtc::CritScope frame_cs(&renderer_lock_);
-    info.frame_width = last_width_;
-    info.frame_height = last_height_;
-    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.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;
-
-  return info;
-}
-
-WebRtcVideoChannel2::VideoCodecSettings::VideoCodecSettings()
-    : rtx_payload_type(-1) {}
-
-bool WebRtcVideoChannel2::VideoCodecSettings::operator==(
-    const WebRtcVideoChannel2::VideoCodecSettings& other) const {
-  return codec == other.codec &&
-         fec.ulpfec_payload_type == other.fec.ulpfec_payload_type &&
-         fec.red_payload_type == other.fec.red_payload_type &&
-         fec.red_rtx_payload_type == other.fec.red_rtx_payload_type &&
-         rtx_payload_type == other.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::FecConfig fec_settings;
-
-  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(fec_settings.red_payload_type == -1);
-        fec_settings.red_payload_type = in_codec.id;
-        continue;
-      }
-
-      case VideoCodec::CODEC_ULPFEC: {
-        // ULPFEC payload type, should not have duplicates.
-        RTC_DCHECK(fec_settings.ulpfec_payload_type == -1);
-        fec_settings.ulpfec_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 == fec_settings.red_payload_type) {
-      fec_settings.red_rtx_payload_type = it->second;
-    }
-  }
-
-  for (size_t i = 0; i < video_codecs.size(); ++i) {
-    video_codecs[i].fec = fec_settings;
-    if (rtx_mapping[video_codecs[i].codec.id] != 0 &&
-        rtx_mapping[video_codecs[i].codec.id] !=
-            fec_settings.red_payload_type) {
-      video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
-    }
-  }
-
-  return video_codecs;
-}
-
-}  // namespace cricket
-
-#endif  // HAVE_WEBRTC_VIDEO