Implement AudioReceiveStream::GetStats().

webrtc/audio/audio_receive_stream.h

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 
 #include "webrtc/audio_receive_stream.h"
+#include "webrtc/audio/scoped_voe_interface.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/voice_engine/include/voe_base.h"
 
 namespace webrtc {
 
 class RemoteBitrateEstimator;
+class VoiceEngine;
 
 namespace internal {
 
 class AudioReceiveStream : public webrtc::AudioReceiveStream {
  public:
   AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
-                     const webrtc::AudioReceiveStream::Config& config);
+                     const webrtc::AudioReceiveStream::Config& config,
+                     VoiceEngine* voice_engine);
   ~AudioReceiveStream() override;
 
   // webrtc::ReceiveStream implementation.
   void Start() override;
   void Stop() override;
   void SignalNetworkState(NetworkState state) override;
   bool DeliverRtcp(const uint8_t* packet, size_t length) override;
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const PacketTime& packet_time) override;
 
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
 
   const webrtc::AudioReceiveStream::Config& config() const;
 
  private:
+  rtc::ThreadChecker thread_checker_;
   RemoteBitrateEstimator* const remote_bitrate_estimator_;
   const webrtc::AudioReceiveStream::Config config_;
+  VoiceEngine* voice_engine_;
+  // We hold one interface pointer to the VoE to make sure it is kept alive.
+  ScopedVoEInterface<VoEBase> voe_base_;
   rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_