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Issue 1818023002: Delete class webrtc::VideoRenderer and its header file. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased. 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 98 matching lines...) Expand 10 before | Expand all | Expand 10 after
109 'sources': [ 109 'sources': [
110 'audio_receive_stream.h', 110 'audio_receive_stream.h',
111 'audio_send_stream.h', 111 'audio_send_stream.h',
112 'audio_state.h', 112 'audio_state.h',
113 'call.h', 113 'call.h',
114 'config.h', 114 'config.h',
115 'frame_callback.h', 115 'frame_callback.h',
116 'stream.h', 116 'stream.h',
117 'transport.h', 117 'transport.h',
118 'video_receive_stream.h', 118 'video_receive_stream.h',
119 'video_renderer.h',
120 'video_send_stream.h', 119 'video_send_stream.h',
121 120
122 '<@(webrtc_audio_sources)', 121 '<@(webrtc_audio_sources)',
123 '<@(webrtc_call_sources)', 122 '<@(webrtc_call_sources)',
124 '<@(webrtc_video_sources)', 123 '<@(webrtc_video_sources)',
125 ], 124 ],
126 'dependencies': [ 125 'dependencies': [
127 'common.gyp:*', 126 'common.gyp:*',
128 '<@(webrtc_audio_dependencies)', 127 '<@(webrtc_audio_dependencies)',
129 '<@(webrtc_call_dependencies)', 128 '<@(webrtc_call_dependencies)',
(...skipping 27 matching lines...) Expand all
157 ], 156 ],
158 'defines': [ 157 'defines': [
159 'ENABLE_RTC_EVENT_LOG', 158 'ENABLE_RTC_EVENT_LOG',
160 ], 159 ],
161 }], 160 }],
162 ], 161 ],
163 }, 162 },
164 163
165 ], 164 ],
166 } 165 }
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