Index: webrtc/call/call_unittest.cc |
diff --git a/webrtc/call/call_unittest.cc b/webrtc/call/call_unittest.cc |
index 06ee2005b1953dac90b724fe5f1f2e47f64610ca..08bf860c2c6d3597f5f2dcc37b2c0ef4a0c767fb 100644 |
--- a/webrtc/call/call_unittest.cc |
+++ b/webrtc/call/call_unittest.cc |
@@ -18,6 +18,7 @@ |
#include "webrtc/call/audio_state.h" |
#include "webrtc/call/call.h" |
#include "webrtc/call/fake_rtp_transport_controller_send.h" |
+#include "webrtc/call/rtp_transport_controller_send.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
#include "webrtc/modules/audio_device/include/mock_audio_device.h" |
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
@@ -40,9 +41,16 @@ struct CallHelper { |
EXPECT_CALL(voice_engine_, audio_device_module()); |
EXPECT_CALL(voice_engine_, audio_processing()); |
EXPECT_CALL(voice_engine_, audio_transport()); |
+ |
+ rtp_transport_send_ = rtc::MakeUnique<webrtc::RtpTransportControllerSend>( |
+ webrtc::Clock::GetRealTimeClock(), &event_log_); |
+ |
webrtc::Call::Config config(&event_log_); |
config.audio_state = webrtc::AudioState::Create(audio_state_config); |
- call_.reset(webrtc::Call::Create(config)); |
+ config.audio_rtp_transport_send = rtp_transport_send_.get(); |
+ config.video_rtp_transport_send = rtp_transport_send_.get(); |
+ config.send_side_cc = rtp_transport_send_->send_side_cc(); |
+ call_ = rtc::WrapUnique(webrtc::Call::Create(config)); |
} |
webrtc::Call* operator->() { return call_.get(); } |
@@ -51,6 +59,7 @@ struct CallHelper { |
private: |
testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_; |
webrtc::RtcEventLogNullImpl event_log_; |
+ std::unique_ptr<webrtc::RtpTransportControllerSend> rtp_transport_send_; |
std::unique_ptr<webrtc::Call> call_; |
}; |
} // namespace |
@@ -320,15 +329,15 @@ TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) { |
namespace { |
struct CallBitrateHelper { |
- CallBitrateHelper() : CallBitrateHelper(Call::Config(&event_log_)) {} |
- |
- explicit CallBitrateHelper(const Call::Config& config) |
+ CallBitrateHelper() |
: mock_cc_(Clock::GetRealTimeClock(), &event_log_, &packet_router_), |
- call_(Call::Create( |
- config, |
- rtc::MakeUnique<FakeRtpTransportControllerSend>(&packet_router_, |
- &mock_cc_))) {} |
+ transport_send(&packet_router_, &mock_cc_) { |
+ Call::Config config(&event_log_); |
+ config.audio_rtp_transport_send = &transport_send; |
+ config.video_rtp_transport_send = &transport_send; |
+ call_ = rtc::WrapUnique(Call::Create(config)); |
+ } |
webrtc::Call* operator->() { return call_.get(); } |
testing::NiceMock<test::MockSendSideCongestionController>& mock_cc() { |
return mock_cc_; |
@@ -338,6 +347,7 @@ struct CallBitrateHelper { |
webrtc::RtcEventLogNullImpl event_log_; |
PacketRouter packet_router_; |
testing::NiceMock<test::MockSendSideCongestionController> mock_cc_; |
+ FakeRtpTransportControllerSend transport_send; |
std::unique_ptr<Call> call_; |
}; |
} // namespace |
@@ -458,8 +468,13 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) { |
audio_state_config.audio_mixer = mock_mixer; |
auto audio_state = AudioState::Create(audio_state_config); |
RtcEventLogNullImpl event_log; |
+ RtpTransportControllerSend rtp_transport_send( |
+ webrtc::Clock::GetRealTimeClock(), &event_log); |
Call::Config call_config(&event_log); |
call_config.audio_state = audio_state; |
+ call_config.audio_rtp_transport_send = &rtp_transport_send; |
+ call_config.video_rtp_transport_send = &rtp_transport_send; |
+ call_config.send_side_cc = rtp_transport_send.send_side_cc(); |
std::unique_ptr<Call> call(Call::Create(call_config)); |
auto create_stream_and_get_rtp_state = [&](uint32_t ssrc) { |