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Issue 1739783002: Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
2 # 2 #
3 # Use of this source code is governed by a BSD-style license 3 # Use of this source code is governed by a BSD-style license
4 # that can be found in the LICENSE file in the root of the source 4 # that can be found in the LICENSE file in the root of the source
5 # tree. An additional intellectual property rights grant can be found 5 # tree. An additional intellectual property rights grant can be found
6 # in the file PATENTS. All contributing project authors may 6 # in the file PATENTS. All contributing project authors may
7 # be found in the AUTHORS file in the root of the source tree. 7 # be found in the AUTHORS file in the root of the source tree.
8 { 8 {
9 'variables': { 9 'variables': {
10 'webrtc_all_dependencies': [ 10 'webrtc_all_dependencies': [
(...skipping 92 matching lines...) Expand 10 before | Expand all | Expand 10 after
103 'test/webrtc_test_common.gyp:*', 103 'test/webrtc_test_common.gyp:*',
104 'webrtc_tests', 104 'webrtc_tests',
105 ], 105 ],
106 }], 106 }],
107 ], 107 ],
108 }, 108 },
109 { 109 {
110 'target_name': 'webrtc', 110 'target_name': 'webrtc',
111 'type': 'static_library', 111 'type': 'static_library',
112 'sources': [ 112 'sources': [
113 'audio_receive_stream.h',
113 'audio_send_stream.h', 114 'audio_send_stream.h',
114 'audio_state.h', 115 'audio_state.h',
115 'call.h', 116 'call.h',
116 'video_frame.h', 117 'config.h',
117 'video_decoder.h', 118 'frame_callback.h',
118 'video_encoder.h', 119 'stream.h',
120 'transport.h',
121 'video_receive_stream.h',
122 'video_renderer.h',
123 'video_send_stream.h',
124
119 '<@(webrtc_audio_sources)', 125 '<@(webrtc_audio_sources)',
120 '<@(webrtc_call_sources)', 126 '<@(webrtc_call_sources)',
121 '<@(webrtc_video_sources)', 127 '<@(webrtc_video_sources)',
122 ], 128 ],
123 'dependencies': [ 129 'dependencies': [
124 'common.gyp:*', 130 'common.gyp:*',
125 '<@(webrtc_audio_dependencies)', 131 '<@(webrtc_audio_dependencies)',
126 '<@(webrtc_call_dependencies)', 132 '<@(webrtc_call_dependencies)',
127 '<@(webrtc_video_dependencies)', 133 '<@(webrtc_video_dependencies)',
128 'rtc_event_log', 134 'rtc_event_log',
(...skipping 25 matching lines...) Expand all
154 ], 160 ],
155 'defines': [ 161 'defines': [
156 'ENABLE_RTC_EVENT_LOG', 162 'ENABLE_RTC_EVENT_LOG',
157 ], 163 ],
158 }], 164 }],
159 ], 165 ],
160 }, 166 },
161 167
162 ], 168 ],
163 } 169 }
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