Chromium Code Reviews| 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> |
|
the sun
2016/01/21 12:24:52
lint warning?
stefan-webrtc
2016/01/21 13:17:43
Yes
|
| #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()); |