| Index: webrtc/audio/audio_send_stream.cc
|
| diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
|
| index 438d1cc78a5aca5d7657b6368bfbac03fa5aed8e..f2cfcc29998408c323bf301cc9645b560e81f0b5 100644
|
| --- a/webrtc/audio/audio_send_stream.cc
|
| +++ b/webrtc/audio/audio_send_stream.cc
|
| @@ -20,7 +20,7 @@
|
| #include "webrtc/base/logging.h"
|
| #include "webrtc/base/task_queue.h"
|
| #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
|
| -#include "webrtc/modules/congestion_controller/include/congestion_controller.h"
|
| +#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
|
| #include "webrtc/modules/pacing/paced_sender.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
| #include "webrtc/voice_engine/channel_proxy.h"
|
| @@ -45,7 +45,7 @@ AudioSendStream::AudioSendStream(
|
| const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
|
| rtc::TaskQueue* worker_queue,
|
| PacketRouter* packet_router,
|
| - CongestionController* congestion_controller,
|
| + SendSideCongestionController* congestion_controller,
|
| BitrateAllocator* bitrate_allocator,
|
| RtcEventLog* event_log,
|
| RtcpRttStats* rtcp_rtt_stats)
|
|
|