Define Stream base classes

webrtc/webrtc.gyp

 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 {
   'conditions': [
     ['include_tests==1', {
@@ skipping 43 matching lines @@
             'test/metrics.gyp:*',
             'test/test.gyp:*',
             'test/webrtc_test_common.gyp:webrtc_test_common_unittests',
             'video_engine/video_engine_core_unittests.gyp:video_engine_core_unittests',
             'webrtc_tests',
           ],
         }],
       ],
     },
     {
-      # 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',
         'video_send_stream.h',
 
         '<@(webrtc_video_sources)',
       ],
       'dependencies': [
         'common.gyp:*',
         '<@(webrtc_video_dependencies)',
       ],
       'conditions': [
         # TODO(andresp): Chromium libpeerconnection should link directly with
         # this and no if conditions should be needed on webrtc build files.
         ['build_with_chromium==1', {
           'dependencies': [
             '<(webrtc_root)/modules/modules.gyp:video_capture',
             '<(webrtc_root)/modules/modules.gyp:video_render',
           ],
         }],
       ],
     },
   ],
 }