Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i…

webrtc/call/call_unittest.cc

 /*
  *  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.
  */
 
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
+#include "webrtc/audio_state.h"
 #include "webrtc/call.h"
 #include "webrtc/test/fake_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   CallHelper() : voice_engine_(new webrtc::test::FakeVoiceEngine()) {
+    webrtc::AudioState::Config audio_state_config;
+    audio_state_config.voice_engine = voice_engine_.get();
+    audio_state_.reset(webrtc::AudioState::Create(audio_state_config));
     webrtc::Call::Config config;
-    config.voice_engine = voice_engine_.get();
+    config.audio_state = audio_state_.get();
     call_.reset(webrtc::Call::Create(config));
   }
 
   webrtc::Call* operator->() { return call_.get(); }
 
  private:
   rtc::scoped_ptr<webrtc::test::FakeVoiceEngine> voice_engine_;
+  rtc::scoped_ptr<webrtc::AudioState> audio_state_;
   rtc::scoped_ptr<webrtc::Call> call_;
 };
 }  // namespace
 
 namespace webrtc {
 
 TEST(CallTest, ConstructDestruct) {
   CallHelper call;
 }
 
@@ skipping 56 matching lines @@
         streams.push_front(stream);
       }
     }
     for (auto s : streams) {
       call->DestroyAudioReceiveStream(s);
     }
     streams.clear();
   }
 }
 }  // namespace webrtc