Change kEnableAudioTrackProcessing to a disable flag and remove finch code

content/renderer/media/webrtc_audio_capturer_unittest.cc

 // Copyright 2013 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 "base/logging.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"
@@ skipping 66 matching lines @@
 #if defined(OS_ANDROID)
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) {
     // Android works with a buffer size bigger than 20ms.
 #else
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) {
 #endif
   }
 
-  void EnableAudioTrackProcessing() {
+  void DisableAudioTrackProcessing() {
     CommandLine::ForCurrentProcess()->AppendSwitch(
-        switches::kEnableAudioTrackProcessing);
+        switches::kDisableAudioTrackProcessing);
   }
 
   void VerifyAudioParams(const blink::WebMediaConstraints& constraints,
                          bool need_audio_processing) {
     capturer_ = WebRtcAudioCapturer::CreateCapturer(
         -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
                              "", "", params_.sample_rate(),
                              params_.channel_layout(),
                              params_.frames_per_buffer()),
         constraints, NULL, NULL);
@@ skipping 49 matching lines @@
   }
 
   media::AudioParameters params_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
   scoped_ptr<WebRtcLocalAudioTrack> track_;
 };
 
 // Pass the delay value, volume and key_pressed info via capture callback, and
 // those values should be correctly stored and passed to the track.
-TEST_F(WebRtcAudioCapturerTest, VerifyAudioParams) {
+TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithoutAudioProcessing) {
+  DisableAudioTrackProcessing();
   // Use constraints with default settings.
   MockMediaConstraintFactory constraint_factory;
   VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), true);
 }
 
 TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) {
-  EnableAudioTrackProcessing();
   // Turn off the default constraints to verify that the sink will get packets
   // with a buffer size smaller than 10ms.
   MockMediaConstraintFactory constraint_factory;
   constraint_factory.DisableDefaultAudioConstraints();
   VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false);
 }
 
 TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) {
-  EnableAudioTrackProcessing();
   MockMediaConstraintFactory constraint_factory;
   const std::string dummy_constraint = "dummy";
   constraint_factory.AddMandatory(dummy_constraint, true);
 
   scoped_refptr<WebRtcAudioCapturer> capturer(
       WebRtcAudioCapturer::CreateCapturer(
           0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE,
                                "", "", params_.sample_rate(),
                                params_.channel_layout(),
                                params_.frames_per_buffer()),
           constraint_factory.CreateWebMediaConstraints(), NULL, NULL)
   );
   EXPECT_TRUE(capturer == NULL);
 }
 
 
 }  // namespace content