| 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 318ea955cf79dd0906c0d17164235d9330ff6d16..b605a4fb0789036030276c71e1eb42ec9c624bf4 100644
|
| --- a/webrtc/audio/audio_send_stream_unittest.cc
|
| +++ b/webrtc/audio/audio_send_stream_unittest.cc
|
| @@ -74,7 +74,6 @@ struct ConfigHelper {
|
| stream_config_(nullptr),
|
| congestion_controller_(&simulated_clock_,
|
| &bitrate_observer_,
|
| - nullptr,
|
| &event_log_,
|
| &packet_router_),
|
| bitrate_allocator_(&limit_observer_),
|
| @@ -126,7 +125,7 @@ struct ConfigHelper {
|
| rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
|
| MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
|
| PacketRouter* packet_router() { return &packet_router_; }
|
| - CongestionController* congestion_controller() {
|
| + SendSideCongestionController* congestion_controller() {
|
| return &congestion_controller_;
|
| }
|
| BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
|
| @@ -249,7 +248,7 @@ struct ConfigHelper {
|
| MockTransmitMixer transmit_mixer_;
|
| AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
|
| PacketRouter packet_router_;
|
| - CongestionController congestion_controller_;
|
| + SendSideCongestionController congestion_controller_;
|
| MockRtcEventLog event_log_;
|
| MockRtcpRttStats rtcp_rtt_stats_;
|
| testing::NiceMock<MockLimitObserver> limit_observer_;
|
|
|