Index: webrtc/audio/audio_send_stream.h |
diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h |
index 5ee49da91a7a1469285d1fcf675c39c53a43ac5c..436c49824cc2325921c560e59f2f56bc3d439f99 100644 |
--- a/webrtc/audio/audio_send_stream.h |
+++ b/webrtc/audio/audio_send_stream.h |
@@ -20,7 +20,7 @@ |
#include "webrtc/call/bitrate_allocator.h" |
namespace webrtc { |
-class CongestionController; |
+class SendSideCongestionController; |
class VoiceEngine; |
class RtcEventLog; |
class RtcpBandwidthObserver; |
@@ -39,7 +39,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
const rtc::scoped_refptr<webrtc::AudioState>& audio_state, |
rtc::TaskQueue* worker_queue, |
PacketRouter* packet_router, |
- CongestionController* congestion_controller, |
+ SendSideCongestionController* send_side_cc, |
BitrateAllocator* bitrate_allocator, |
RtcEventLog* event_log, |
RtcpRttStats* rtcp_rtt_stats); |
@@ -77,7 +77,7 @@ class AudioSendStream final : public webrtc::AudioSendStream, |
std::unique_ptr<voe::ChannelProxy> channel_proxy_; |
BitrateAllocator* const bitrate_allocator_; |
- CongestionController* const congestion_controller_; |
+ SendSideCongestionController* const send_side_cc_; |
std::unique_ptr<RtcpBandwidthObserver> bandwidth_observer_; |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream); |