Clean up the media stream audio track code

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

 // Copyright 2014 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#include "base/command_line.h"
-#include "content/public/common/content_switches.h"
 #include "content/renderer/media/mock_media_constraint_factory.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_local_audio_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 
 namespace content {
 
@@ skipping 48 matching lines @@
   // Send a packet via |track_| and it data should reach the sink of the
   // |adapter_|.
   const int length = params_.frames_per_buffer() * params_.channels();
   scoped_ptr<int16[]> data(new int16[length]);
   // Initialize the data to 0 to avoid Memcheck:Uninitialized warning.
   memset(data.get(), 0, length * sizeof(data[0]));
 
   EXPECT_CALL(*sink,
               OnData(_, 16, params_.sample_rate(), params_.channels(),
                      params_.frames_per_buffer()));
-  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
+  track_->Capture(data.get(), false);
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
   sink.reset();
 
   // Verify that no more callback gets into the sink.
-  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
+  track_->Capture(data.get(), false);
 }
 
 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
   webrtc::AudioTrackInterface* webrtc_track =
       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
   int signal_level = 0;
   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
-
-  // Disable the audio processing in the audio track.
-  CommandLine::ForCurrentProcess()->AppendSwitch(
-      switches::kDisableAudioTrackProcessing);
-  EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
 }
 
 }  // namespace content