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_; |