Implement AudioReceiveStream::GetStats().

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
@@ skipping 303 matching lines @@
   voe_base->DeleteChannel(channel);
   voe_base->Release();
   voe_codec->Release();
   voe_network->Release();
   voe_sync->Release();
 
   DestroyStreams();
 
   receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
 
+  DestroyCalls();
+
   VoiceEngine::Delete(voice_engine);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
   TestAudioVideoSync(false, true);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
   TestAudioVideoSync(false, false);
 }
@@ skipping 369 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc