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

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 223 matching lines @@
 
   FakeNetworkPipe::Config net_config;
   net_config.queue_delay_ms = 500;
   net_config.loss_percent = 5;
   SyncRtcpObserver audio_observer(net_config);
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(),
                                     channel,
                                     voe_sync,
                                     &audio_observer);
 
+  AudioState::Config audio_state_config;
+  audio_state_config.voice_engine = voice_engine;
+  rtc::scoped_ptr<AudioState> audio_state(
+      AudioState::Create(audio_state_config));
   Call::Config receiver_config;
-  receiver_config.voice_engine = voice_engine;
+  receiver_config.audio_state = audio_state.get();
   CreateCalls(Call::Config(), receiver_config);
 
   CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
   EXPECT_EQ(0, voe_codec->SetSendCodec(channel, isac));
 
   AudioPacketReceiver voe_packet_receiver(channel, voe_network);
   audio_observer.SetReceivers(&voe_packet_receiver, &voe_packet_receiver);
 
   internal::TransportAdapter transport_adapter(audio_observer.SendTransport());
   transport_adapter.Enable();
@@ skipping 449 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc