Movable support for VideoReceiveStream::Config and avoid copies

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.
  */
 #include "webrtc/base/checks.h"
@@ skipping 199 matching lines @@
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       video_config.rtp.extensions.push_back(extension);
     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();
       video_config.decoders.push_back(decoder);
       video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
-      video_receive_configs_.push_back(video_config);
+      video_receive_configs_.push_back(video_config.Copy());
     }
   }
 
   RTC_DCHECK(num_audio_streams_ <= 1);
   if (num_audio_streams_ == 1) {
     RTC_DCHECK(voe_send_.channel_id >= 0);
     AudioReceiveStream::Config audio_config;
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = rtcp_send_transport;
     audio_config.voe_channel_id = voe_recv_.channel_id;
@@ skipping 28 matching lines @@
 
 void CallTest::CreateVideoStreams() {
   RTC_DCHECK(video_send_stream_ == nullptr);
   RTC_DCHECK(video_receive_streams_.empty());
   RTC_DCHECK(audio_send_stream_ == nullptr);
   RTC_DCHECK(audio_receive_streams_.empty());
 
   video_send_stream_ = sender_call_->CreateVideoSendStream(
       video_send_config_, video_encoder_config_);
   for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
-    video_receive_streams_.push_back(
-        receiver_call_->CreateVideoReceiveStream(video_receive_configs_[i]));
+    video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
+        video_receive_configs_[i].Copy()));
   }
 }
 
 void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
   frame_generator_capturer_->SetFakeRotation(rotation);
 }
 
 void CallTest::CreateAudioStreams() {
   audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
   for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
@@ skipping 144 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc