| 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());
|
|
|