Split out audio debug recording from RenderProcessHostImpl.

content/content_tests.gypi

Index: content/content_tests.gypi
diff --git a/content/content_tests.gypi b/content/content_tests.gypi
index 23b6e370cc4285e61dc0857f72fe4300b02976a5..d8a60c30a024cea351c94ce6066c1a3993936783 100644
--- a/content/content_tests.gypi
+++ b/content/content_tests.gypi
@@ -124,6 +124,8 @@
       'test/fake_plugin_service.h',
       'test/fake_renderer_scheduler.cc',
       'test/fake_renderer_scheduler.h',
+      'test/fake_service_registry.cc',
+      'test/fake_service_registry.h',
       'test/gpu_memory_buffer_factory_test_template.h',
       'test/mock_google_streaming_server.cc',
       'test/mock_google_streaming_server.h',
@@ -711,6 +713,7 @@
       'renderer/input/input_event_filter_unittest.cc',
       'renderer/manifest/manifest_parser_unittest.cc',
       'renderer/media/android/media_info_loader_unittest.cc',
+      'renderer/media/audio_debug_recorder_unittest.cc',
       'renderer/media/audio_message_filter_unittest.cc',
       'renderer/media/audio_renderer_mixer_manager_unittest.cc',
       'renderer/media/midi_message_filter_unittest.cc',
@@ -760,6 +763,7 @@
     ],
     # WebRTC-specific sources. Put WebRTC plugin-related stuff further below.
     'content_unittests_webrtc_sources': [
+      'browser/media/audio_debug_recording_impl_unittest.cc',
       'browser/media/webrtc_internals_unittest.cc',
       'browser/renderer_host/media/webrtc_identity_service_host_unittest.cc',
       'browser/renderer_host/p2p/socket_host_tcp_server_unittest.cc',