Fix/Add content_unittests to work for the code behind --enable-audio-track-processing

content/renderer/media/media_stream_audio_processor_unittest.cc

Index: content/renderer/media/media_stream_audio_processor_unittest.cc
diff --git a/content/renderer/media/media_stream_audio_processor_unittest.cc b/content/renderer/media/media_stream_audio_processor_unittest.cc
index 3f46406c65366c42788536a8d6f2e943c425c7f8..72e4010157a9f0363f3c64a351e97ce879c5cd07 100644
--- a/content/renderer/media/media_stream_audio_processor_unittest.cc
+++ b/content/renderer/media/media_stream_audio_processor_unittest.cc
@@ -11,6 +11,7 @@
 #include "content/public/common/content_switches.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
+#include "content/renderer/media/mock_media_constraint_factory.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gmock/include/gmock/gmock.h"
@@ -227,4 +228,30 @@ TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) {
   audio_processor = NULL;
 }
 
+TEST_F(MediaStreamAudioProcessorTest, TurnOffDefaultConstraints) {
+  // Setup the audio processor with enabling the flag.

[phoglund_chromium, 2014/03/11 14:43:21]

Nit: Set up the audio processor with the flag enabled.

[no longer working on chromium, 2014/03/11 15:22:30]

Done.

+  CommandLine::ForCurrentProcess()->AppendSwitch(
+      switches::kEnableAudioTrackProcessing);
+
+  // Turn off the default constraints and pass it to MediaStreamAudioProcessor.
+  MockMediaConstraintFactory constraint_factory;
+  constraint_factory.DisableDefaultAudioConstraints();
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl());
+  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
+      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+          constraint_factory.CreateWebMediaConstraints(), 0,
+          MEDIA_DEVICE_AUDIO_CAPTURE, webrtc_audio_device.get()));
+  EXPECT_FALSE(audio_processor->has_audio_processing());
+  audio_processor->OnCaptureFormatChanged(params_);
+
+  ProcessDataAndVerifyFormat(audio_processor,
+                             params_.sample_rate(),
+                             params_.channels(),
+                             params_.sample_rate() / 100);
+  // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
+  // |audio_processor|.
+  audio_processor = NULL;
+}
+
 }  // namespace content