Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index 3fbbb6207285d5a1baf0be387072b107d5f494bf..dba83ca524b26d931308973aaf148f03cefc81c6 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -25,6 +25,7 @@ |
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h" |
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h" |
#include "webrtc/modules/pacing/paced_sender.h" |
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
#include "webrtc/test/gtest.h" |
#include "webrtc/test/mock_voe_channel_proxy.h" |
@@ -217,6 +218,12 @@ struct ConfigHelper { |
void SetupDefaultChannelProxy(bool audio_bwe_enabled) { |
using testing::StrEq; |
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>(); |
+ EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _)) |
+ .WillRepeatedly(Invoke( |
+ [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) { |
+ *rtp_rtcp_module = &this->rtp_rtcp_; |
+ *rtp_receiver = nullptr; // Not deemed necessary for tests yet. |
+ })); |
EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1); |
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1); |
EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1); |
@@ -321,6 +328,7 @@ struct ConfigHelper { |
AudioProcessing::AudioProcessingStatistics audio_processing_stats_; |
FakeRtpTransportController fake_transport_; |
MockRtcEventLog event_log_; |
+ MockRtpRtcp rtp_rtcp_; |
MockRtcpRttStats rtcp_rtt_stats_; |
testing::NiceMock<MockLimitObserver> limit_observer_; |
BitrateAllocator bitrate_allocator_; |