Index: webrtc/audio/audio_receive_stream.h |
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h |
index adac88387613a43a7f6220567a75c8e881fe313c..01bf728bf934708111049f2347c37cfd4de5d6c5 100644 |
--- a/webrtc/audio/audio_receive_stream.h |
+++ b/webrtc/audio/audio_receive_stream.h |
@@ -12,6 +12,7 @@ |
#define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_ |
#include <memory> |
+#include <vector> |
#include "webrtc/api/audio/audio_mixer.h" |
#include "webrtc/audio/audio_state.h" |
@@ -23,6 +24,7 @@ |
namespace webrtc { |
class PacketRouter; |
class RtcEventLog; |
+class RtpContributingSource; |
class RtpPacketReceived; |
namespace voe { |
@@ -49,6 +51,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream, |
int GetOutputLevel() const override; |
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override; |
void SetGain(float gain) override; |
+ const std::vector<RtpContributingSource*>& GetContributingSources() override; |
// TODO(nisse): Intended to be part of an RtpPacketReceiver interface. |
void OnRtpPacket(const RtpPacketReceived& packet); |