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) |