| Index: webrtc/audio/audio_receive_stream.h
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| diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
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| index adac88387613a43a7f6220567a75c8e881fe313c..20ed4613a7ede16d1ad3aa8897cb957c8c282c34 100644
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| --- a/webrtc/audio/audio_receive_stream.h
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| +++ b/webrtc/audio/audio_receive_stream.h
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| @@ -12,6 +12,7 @@
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|  #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
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|  
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|  #include <memory>
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| +#include <vector>
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|  
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|  #include "webrtc/api/audio/audio_mixer.h"
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|  #include "webrtc/audio/audio_state.h"
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| @@ -49,6 +50,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
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|    int GetOutputLevel() const override;
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|    void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
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|    void SetGain(float gain) override;
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| +  std::vector<webrtc::RtpSource> GetSources() const override;
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|  
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|    // TODO(nisse): Intended to be part of an RtpPacketReceiver interface.
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|    void OnRtpPacket(const RtpPacketReceived& packet);
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| 
 |