Index: webrtc/voice_engine/channel.h |
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h |
index fb9d61ddc84b744c7c1cd3d787e20b30519c86d8..13d8e49877a3d74f924dcded2105013babfb10ae 100644 |
--- a/webrtc/voice_engine/channel.h |
+++ b/webrtc/voice_engine/channel.h |
@@ -28,7 +28,6 @@ |
#include "webrtc/modules/audio_processing/rms_level.h" |
#include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
#include "webrtc/voice_engine/audio_level.h" |
#include "webrtc/voice_engine/file_player.h" |
@@ -53,6 +52,7 @@ |
class RemoteNtpTimeEstimator; |
class RtcEventLog; |
class RTPPayloadRegistry; |
+class RtpReceiver; |
class RTPReceiverAudio; |
class RtpPacketReceived; |
class RtpRtcp; |
@@ -399,10 +399,6 @@ |
void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate); |
- std::vector<RtpSource> GetSources() const { |
- return rtp_receiver_->GetSources(); |
- } |
- |
private: |
class ProcessAndEncodeAudioTask; |