Split out audio debug recording from RenderProcessHostImpl.

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 <vector>
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/memory/aligned_memory.h"
 #include "base/path_service.h"
 #include "base/time/time.h"
+#include "content/common/mojo/service_registry_impl.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.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"
 #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"
@@ skipping 448 matching lines @@
                                kAudioProcessingSampleRate,
                                kAudioProcessingNumberOfChannel,
                                kAudioProcessingSampleRate / 100);
   }
 
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device|
   // outlives |audio_processor|.
   audio_processor = NULL;
 }
 
-// Test that if we have an AEC dump message filter created, we are getting it
+// Test that if we have an audio debug recorder created, we are getting it
 // correctly in MSAP. Any IPC messages will be deleted since no sender in the
 // filter will be created.
-TEST_F(MediaStreamAudioProcessorTest, GetAecDumpMessageFilter) {
+TEST_F(MediaStreamAudioProcessorTest, GetAudioDebugRecorder) {
   base::MessageLoopForUI message_loop;
-  scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_(
-      new AecDumpMessageFilter(message_loop.task_runner(),
-                               message_loop.task_runner()));
+  ServiceRegistryImpl service_registry;
+  scoped_refptr<AudioDebugRecorder> audio_debug_recorder(
+      new AudioDebugRecorder(&service_registry));
 
   MockMediaConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
 
-  EXPECT_TRUE(audio_processor->aec_dump_message_filter_.get());
+  EXPECT_TRUE(audio_processor->audio_debug_recorder_.get());
+  EXPECT_EQ(audio_debug_recorder, audio_processor->audio_debug_recorder_);
 
   audio_processor = NULL;
 }
 
 TEST_F(MediaStreamAudioProcessorTest, TestStereoAudio) {
   // Set up the correct constraints to turn off the audio processing and turn
   // on the stereo channels mirroring.
   MockMediaConstraintFactory constraint_factory;
   constraint_factory.AddMandatory(MediaAudioConstraints::kEchoCancellation,
                                   false);
@@ skipping 79 matching lines @@
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
   // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
   // |audio_processor|.
   audio_processor = NULL;
 }
 
 }  // namespace content