Index: webrtc/audio/audio_receive_stream_unittest.cc |
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc |
index eb008b3045408c32cedd5a1d9b9d2cec5208a32f..b241bed6b51e259c8e095e449bb5d150485a0433 100644 |
--- a/webrtc/audio/audio_receive_stream_unittest.cc |
+++ b/webrtc/audio/audio_receive_stream_unittest.cc |
@@ -9,6 +9,7 @@ |
*/ |
#include <string> |
+#include <vector> |
#include "testing/gtest/include/gtest/gtest.h" |
@@ -90,6 +91,9 @@ struct ConfigHelper { |
EXPECT_CALL(*channel_proxy_, |
SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId)) |
.Times(1); |
+ EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber( |
+ kTransportSequenceNumberId)) |
+ .Times(1); |
EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects( |
nullptr, nullptr, &packet_router_)) |
.Times(1); |
@@ -107,6 +111,8 @@ struct ConfigHelper { |
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); |
stream_config_.rtp.extensions.push_back( |
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId)); |
+ stream_config_.rtp.extensions.push_back(RtpExtension( |
+ RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); |
} |
MockCongestionController* congestion_controller() { |
@@ -261,8 +267,6 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) { |
ConfigHelper helper; |
helper.config().combined_audio_video_bwe = true; |
helper.config().rtp.transport_cc = true; |
- helper.config().rtp.extensions.push_back(RtpExtension( |
- RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId)); |
helper.SetupMockForBweFeedback(true); |
internal::AudioReceiveStream recv_stream( |
helper.congestion_controller(), helper.config(), helper.audio_state()); |