Chromium Code Reviews| Index: webrtc/webrtc.gyp |
| diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp |
| index b1574d22a6178e94a3b723ee5976084d808257f7..1f0412e845489ed118c0ed23c701b039a08752b3 100644 |
| --- a/webrtc/webrtc.gyp |
| +++ b/webrtc/webrtc.gyp |
| @@ -61,15 +61,16 @@ |
| ], |
| }, |
| { |
| - # TODO(pbos): This is intended to contain audio parts as well as soon as |
| - # VoiceEngine moves to the same new API format. |
| 'target_name': 'webrtc', |
| 'type': 'static_library', |
| 'sources': [ |
| + 'audio_receive_stream.h', |
| + 'audio_send_stream.h', |
| 'call.h', |
| 'config.h', |
| 'experiments.h', |
| 'frame_callback.h', |
| + 'stream.h', |
| 'transport.h', |
| 'video_receive_stream.h', |
| 'video_renderer.h', |