Only turn on the audio processing by default for MEDIA_DEVICE_AUDIO_CAPTURE

content/renderer/media/media_stream_audio_processor_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/file_util.h"
 #include "base/files/file_path.h"
 #include "base/logging.h"
 #include "base/path_service.h"
 #include "base/time/time.h"
 #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 "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
@@ skipping 127 matching lines @@
   media::AudioParameters params_;
 };
 
 TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
   // Setup the audio processor without enabling the flag.
   blink::WebMediaConstraints constraints;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-          constraints, 0, webrtc_audio_device.get()));
+          constraints, 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;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
   // Setup the audio processor with enabling the flag.
   CommandLine::ForCurrentProcess()->AppendSwitch(
       switches::kEnableAudioTrackProcessing);
   blink::WebMediaConstraints constraints;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
-          constraints, 0, webrtc_audio_device.get()));
+          constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
+          webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
   audio_processor->OnCaptureFormatChanged(params_);
   VerifyDefaultComponents(audio_processor);
 
   ProcessDataAndVerifyFormat(audio_processor,
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
+TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) {
+  // Setup the audio processor with enabling the flag.
+  CommandLine::ForCurrentProcess()->AppendSwitch(
+      switches::kEnableAudioTrackProcessing);
+  blink::WebMediaConstraints constraints;
+  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
+      new WebRtcAudioDeviceImpl());
+  // Create MediaStreamAudioProcessor instance for MEDIA_TAB_AUDIO_CAPTURE type.
+  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
+      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+          constraints, 0, MEDIA_TAB_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);
+
+  // Create MediaStreamAudioProcessor instance for MEDIA_LOOPBACK_AUDIO_CAPTURE.
+  audio_processor =
+      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
+          constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE,
+          webrtc_audio_device.get());
+  EXPECT_FALSE(audio_processor->has_audio_processing());
+
+  // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
+  // |audio_processor|.
+  audio_processor = NULL;
+}
+
 }  // namespace content