MediaStream audio rendering: Bypass audio processing for non-WebRTC cases.

content/content_renderer.gypi

 # Copyright (c) 2012 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.
 
 {
   'dependencies': [
     '../base/base.gyp:base',
     '../base/third_party/dynamic_annotations/dynamic_annotations.gyp:dynamic_annotations',
     '../cc/cc.gyp:cc',
     '../cc/cc.gyp:cc_proto',
@@ skipping 49 matching lines @@
       'public/renderer/browser_plugin_delegate.cc',
       'public/renderer/browser_plugin_delegate.h',
       'public/renderer/chrome_object_extensions_utils.cc',
       'public/renderer/chrome_object_extensions_utils.h',
       'public/renderer/content_renderer_client.cc',
       'public/renderer/content_renderer_client.h',
       'public/renderer/context_menu_client.h',
       'public/renderer/document_state.cc',
       'public/renderer/document_state.h',
       'public/renderer/media_stream_audio_renderer.h',
+      'public/renderer/media_stream_audio_sink.cc',

[miu, 2016/01/27 05:14:14]

Note: These no longer have dependencies on WebRTC, so they should not be
conditionally compiled.  I moved them to the group of files that is always
built.

[tommi (sloooow) - chröme, 2016/02/01 20:32:23]

Nice!

+      'public/renderer/media_stream_audio_sink.h',
+      'public/renderer/media_stream_sink.h',
       'public/renderer/navigation_state.cc',
       'public/renderer/navigation_state.h',
       'public/renderer/pepper_plugin_instance.h',
       'public/renderer/platform_event_observer.h',
       'public/renderer/render_frame.h',
       'public/renderer/render_frame_observer.cc',
       'public/renderer/render_frame_observer.h',
       'public/renderer/render_frame_observer_tracker.h',
       'public/renderer/render_process_observer.cc',
       'public/renderer/render_process_observer.h',
@@ skipping 244 matching lines @@
       'renderer/media/render_media_log.cc',
       'renderer/media/render_media_log.h',
       'renderer/media/renderer_gpu_video_accelerator_factories.cc',
       'renderer/media/renderer_gpu_video_accelerator_factories.h',
       'renderer/media/renderer_webaudiodevice_impl.cc',
       'renderer/media/renderer_webaudiodevice_impl.h',
       'renderer/media/renderer_webmediaplayer_delegate.cc',
       'renderer/media/renderer_webmediaplayer_delegate.h',
       'renderer/media/renderer_webmidiaccessor_impl.cc',
       'renderer/media/renderer_webmidiaccessor_impl.h',
+      'renderer/media/track_audio_renderer.cc',
+      'renderer/media/track_audio_renderer.h',
       'renderer/media/video_capture_impl.cc',
       'renderer/media/video_capture_impl.h',
       'renderer/media/video_capture_impl_manager.cc',
       'renderer/media/video_capture_impl_manager.h',
       'renderer/media/video_capture_message_filter.cc',
       'renderer/media/video_capture_message_filter.h',
       'renderer/media/webmediaplayer_ms.cc',
       'renderer/media/webmediaplayer_ms.h',
       'renderer/media/webmediaplayer_ms_compositor.cc',
       'renderer/media/webmediaplayer_ms_compositor.h',
@@ skipping 266 matching lines @@
       'renderer/pepper/video_decoder_shim.cc',
       'renderer/pepper/video_decoder_shim.h',
       'renderer/pepper/video_encoder_shim.cc',
       'renderer/pepper/video_encoder_shim.h',
       'renderer/render_widget_fullscreen_pepper.cc',
       'renderer/render_widget_fullscreen_pepper.h',
     ],
     'public_renderer_webrtc_sources': [
       'public/renderer/media_stream_api.cc',
       'public/renderer/media_stream_api.h',
-      'public/renderer/media_stream_audio_sink.cc',
-      'public/renderer/media_stream_audio_sink.h',
       'public/renderer/media_stream_renderer_factory.h',
-      'public/renderer/media_stream_sink.h',
       'public/renderer/media_stream_video_sink.cc',
       'public/renderer/media_stream_video_sink.h',
       'public/renderer/webrtc_log_message_delegate.h',
     ],
     # WebRTC-specific sources. Put WebRTC plugin-related stuff in the
     # Plugin+WebRTC section below.
     'private_renderer_webrtc_sources': [
       'renderer/media/audio_repetition_detector.cc',
       'renderer/media/audio_repetition_detector.h',
       'renderer/media/audio_track_recorder.cc',
@@ skipping 92 matching lines @@
       'renderer/media/webrtc_audio_capturer.cc',
       'renderer/media/webrtc_audio_capturer.h',
       'renderer/media/webrtc_audio_device_impl.cc',
       'renderer/media/webrtc_audio_device_impl.h',
       'renderer/media/webrtc_audio_device_not_impl.cc',
       'renderer/media/webrtc_audio_device_not_impl.h',
       'renderer/media/webrtc_audio_renderer.cc',
       'renderer/media/webrtc_audio_renderer.h',
       'renderer/media/webrtc_identity_service.cc',
       'renderer/media/webrtc_identity_service.h',
-      'renderer/media/webrtc_local_audio_renderer.cc',
-      'renderer/media/webrtc_local_audio_renderer.h',
       'renderer/media/webrtc_local_audio_source_provider.cc',
       'renderer/media/webrtc_local_audio_source_provider.h',
       'renderer/media/webrtc_local_audio_track.cc',
       'renderer/media/webrtc_local_audio_track.h',
       'renderer/media/webrtc_logging.cc',
       'renderer/media/webrtc_logging.h',
       'renderer/media/webrtc_uma_histograms.cc',
       'renderer/media/webrtc_uma_histograms.h',
       'renderer/p2p/empty_network_manager.cc',
       'renderer/p2p/empty_network_manager.h',
@@ skipping 172 matching lines @@
     }],
   ],
   'target_conditions': [
     ['OS=="android"', {
       'sources/': [
         ['include', '^renderer/render_view_linux\\.cc$'],
       ],
     }],
   ],
 }