Test upload to see if the ADM is passed correctly to audio_state.

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.
  */
 
@@ skipping 148 matching lines @@
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   int send_channel_id = voe_base->CreateChannel(config);
   int recv_channel_id = voe_base->CreateChannel();
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
-  send_audio_state_config.audio_device_module = &fake_audio_device;
+  // send_audio_state_config.audio_device_module = &fake_audio_device;
   Call::Config sender_config;
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
   Call::Config receiver_config;
   receiver_config.audio_state = sender_config.audio_state;
   CreateCalls(sender_config, receiver_config);
 
 
   VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock());
 
   // Helper class to ensure we deliver correct media_type to the receiving call.
@@ skipping 518 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc