Reland of Add task queue to Call.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <stdio.h>
@@ skipping 1180 matching lines @@
   test::LayerFilteringTransport transport(
       params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
       params.common.selected_tl, params_.ss.selected_sl);
   // TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
   // least share as much code as possible. That way this test would also match
   // the full stack tests better.
   transport.SetReceiver(call->Receiver());
 
   SetupCommon(&transport, &transport);
 
-  video_send_config_.local_renderer = local_preview.get();
+  video_send_config_.pre_encode_callback = local_preview.get();
   video_receive_configs_[stream_id].renderer = loopback_video.get();
   if (params_.audio && params_.audio_video_sync)
     video_receive_configs_[stream_id].sync_group = kSyncGroup;
 
   video_send_config_.suspend_below_min_bitrate =
       params_.common.suspend_below_min_bitrate;
 
   if (params.common.fec) {
     video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
     video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
     video_receive_configs_[stream_id].rtp.fec.red_payload_type =
       kRedPayloadType;
     video_receive_configs_[stream_id].rtp.fec.ulpfec_payload_type =
       kUlpfecPayloadType;
   }
 
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
-  video_send_stream_ =
-      call->CreateVideoSendStream(video_send_config_, video_encoder_config_);
+  video_send_stream_ = call->CreateVideoSendStream(
+      video_send_config_.Copy(), video_encoder_config_.Copy());
   VideoReceiveStream* video_receive_stream =
       call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
   CreateCapturer(video_send_stream_->Input());
 
   AudioReceiveStream* audio_receive_stream = nullptr;
   if (params_.audio) {
     audio_send_config_ = AudioSendStream::Config(&transport);
     audio_send_config_.voe_channel_id = voe.send_channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
 
@@ skipping 68 matching lines @@
      call->DestroyAudioSendStream(audio_send_stream_);
      call->DestroyAudioReceiveStream(audio_receive_stream);
   }
 
   transport.StopSending();
   if (params_.audio)
     DestroyVoiceEngine(&voe);
 }
 
 }  // namespace webrtc