| Index: webrtc/media/engine/webrtcvideoengine2_unittest.cc
|
| diff --git a/webrtc/media/engine/webrtcvideoengine2_unittest.cc b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
|
| index 43dbbc6d1880b425c0fa2944cdae2f3ac904194a..b8fbe39c1c73b00dab9752d2cabfa405aeb6eb87 100644
|
| --- a/webrtc/media/engine/webrtcvideoengine2_unittest.cc
|
| +++ b/webrtc/media/engine/webrtcvideoengine2_unittest.cc
|
| @@ -16,8 +16,10 @@
|
| #include "webrtc/api/video_codecs/video_encoder.h"
|
| #include "webrtc/base/arraysize.h"
|
| #include "webrtc/base/gunit.h"
|
| +#include "webrtc/base/ptr_util.h"
|
| #include "webrtc/base/stringutils.h"
|
| #include "webrtc/call/flexfec_receive_stream.h"
|
| +#include "webrtc/call/rtp_transport_controller_send.h"
|
| #include "webrtc/common_video/h264/profile_level_id.h"
|
| #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
|
| #include "webrtc/media/base/mediaconstants.h"
|
| @@ -119,8 +121,16 @@ class WebRtcVideoEngine2Test : public ::testing::Test {
|
| WebRtcVideoEngine2Test() : WebRtcVideoEngine2Test("") {}
|
| explicit WebRtcVideoEngine2Test(const char* field_trials)
|
| : override_field_trials_(field_trials),
|
| - call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))),
|
| + rtp_transport_controller_send_(
|
| + rtc::MakeUnique<webrtc::RtpTransportControllerSend>(
|
| + webrtc::Clock::GetRealTimeClock(), &event_log_)),
|
| engine_() {
|
| + 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));
|
| +
|
| std::vector<VideoCodec> engine_codecs = engine_.codecs();
|
| RTC_DCHECK(!engine_codecs.empty());
|
| bool codec_set = false;
|
| @@ -156,6 +166,8 @@ class WebRtcVideoEngine2Test : public ::testing::Test {
|
|
|
| webrtc::test::ScopedFieldTrials override_field_trials_;
|
| webrtc::RtcEventLogNullImpl event_log_;
|
| + const std::unique_ptr<webrtc::RtpTransportControllerSendInterface>
|
| + rtp_transport_controller_send_;
|
| // Used in WebRtcVideoEngine2VoiceTest, but defined here so it's properly
|
| // initialized when the constructor is called.
|
| std::unique_ptr<webrtc::Call> call_;
|
|
|