Enable send-side BWE by default for video in call tests.

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 182 matching lines @@
   RTC_DCHECK_LE(num_audio_streams, 1);
   RTC_DCHECK_LE(num_flexfec_streams, 1);
   RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
   if (num_video_streams > 0) {
     video_send_config_ = VideoSendStream::Config(send_transport);
     video_send_config_.encoder_settings.encoder = &fake_encoder_;
     video_send_config_.encoder_settings.payload_name = "FAKE";
     video_send_config_.encoder_settings.payload_type =
         kFakeVideoSendPayloadType;
     video_send_config_.rtp.extensions.push_back(
-        RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
+        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
+                     kTransportSequenceNumberExtensionId));
     FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
 
     for (size_t i = 0; i < num_video_streams; ++i)
       video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
   }
 
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
@@ skipping 10 matching lines @@
     video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
   }
 }
 
 void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
   RTC_DCHECK(video_receive_configs_.empty());
   RTC_DCHECK(allocated_decoders_.empty());
   if (num_video_streams_ > 0) {
     RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
     VideoReceiveStream::Config video_config(rtcp_send_transport);
-    video_config.rtp.remb = true;
+    video_config.rtp.remb = false;
+    video_config.rtp.transport_cc = true;
     video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       video_config.rtp.extensions.push_back(extension);
     video_config.renderer = &fake_renderer_;
     for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
       VideoReceiveStream::Decoder decoder =
           test::CreateMatchingDecoder(video_send_config_.encoder_settings);
       allocated_decoders_.push_back(
           std::unique_ptr<VideoDecoder>(decoder.decoder));
       video_config.decoders.clear();
@@ skipping 248 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc