| Index: webrtc/media/base/videoengine_unittest.h
|
| diff --git a/webrtc/media/base/videoengine_unittest.h b/webrtc/media/base/videoengine_unittest.h
|
| index ac430b2ede422c3dc81f72b6b227e591d636931a..f79b14cc09b00e80b18b43b182863409c375345a 100644
|
| --- a/webrtc/media/base/videoengine_unittest.h
|
| +++ b/webrtc/media/base/videoengine_unittest.h
|
| @@ -64,7 +64,15 @@ class VideoMediaChannelTest : public testing::Test,
|
| public sigslot::has_slots<> {
|
| protected:
|
| VideoMediaChannelTest<E, C>()
|
| - : call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))) {}
|
| + : rtp_transport_controller_send_(
|
| + rtc::MakeUnique<webrtc::RtpTransportControllerSend>(
|
| + webrtc::Clock::GetRealTimeClock(), &event_log_)) {
|
| + webrtc::Call::Config call_config(&event_log_);
|
| + call_config.audio_rtp_transport_send = rtp_transport_controller_send_.get();
|
| + call_config.video_rtp_transport_send = rtp_transport_controller_send_.get();
|
| + call_config.send_side_cc = rtp_transport_controller_send_->send_side_cc();
|
| + call_ = rtc::WrapUnique(webrtc::Call::Create(call_config));
|
| + }
|
|
|
| virtual cricket::VideoCodec DefaultCodec() = 0;
|
|
|
| @@ -929,7 +937,9 @@ class VideoMediaChannelTest : public testing::Test,
|
| }
|
|
|
| webrtc::RtcEventLogNullImpl event_log_;
|
| - const std::unique_ptr<webrtc::Call> call_;
|
| + const std::unique_ptr<webrtc::RtpTransportControllerSendInterface>
|
| + rtp_transport_controller_send_;
|
| + std::unique_ptr<webrtc::Call> call_;
|
| E engine_;
|
| std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_;
|
| std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_2_;
|
|
|