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Issue 1929933002: Drop build dependency on webrtc's legacy video_render module. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 months ago
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1 # Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 # Copyright (c) 2012 The Chromium Authors. All rights reserved.
2 # Use of this source code is governed by a BSD-style license that can be 2 # Use of this source code is governed by a BSD-style license that can be
3 # found in the LICENSE file. 3 # found in the LICENSE file.
4 4
5 { 5 {
6 'includes': [ 6 'includes': [
7 '../../build/win_precompile.gypi', 7 '../../build/win_precompile.gypi',
8 ], 8 ],
9 # Most of these settings have been split according to their scope into 9 # Most of these settings have been split according to their scope into
10 # :jingle_unexported_configs and :jingle_public_config in the GN build. 10 # :jingle_unexported_configs and :jingle_public_config in the GN build.
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358 '<(DEPTH)/third_party/webrtc/pc/rtcpmuxfilter.h', 358 '<(DEPTH)/third_party/webrtc/pc/rtcpmuxfilter.h',
359 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.cc', 359 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.cc',
360 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.h', 360 '<(DEPTH)/third_party/webrtc/pc/srtpfilter.h',
361 '<(DEPTH)/third_party/webrtc/pc/voicechannel.h', 361 '<(DEPTH)/third_party/webrtc/pc/voicechannel.h',
362 ], 362 ],
363 'dependencies': [ 363 'dependencies': [
364 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp', 364 '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
365 '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib', 365 '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
366 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file', 366 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:media_file',
367 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture', 367 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_capture',
368 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:video_render',
369 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng ine', 368 '<(DEPTH)/third_party/webrtc/voice_engine/voice_engine.gyp:voice_eng ine',
370 '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc', 369 '<(DEPTH)/third_party/webrtc/webrtc.gyp:webrtc',
371 'libjingle', 370 'libjingle',
372 ], 371 ],
373 }, # target libjingle_webrtc_common 372 }, # target libjingle_webrtc_common
374 { 373 {
375 # GN version: //third_party/libjingle:libjingle_webrtc 374 # GN version: //third_party/libjingle:libjingle_webrtc
376 'target_name': 'libjingle_webrtc', 375 'target_name': 'libjingle_webrtc',
377 'type': 'static_library', 376 'type': 'static_library',
378 'sources': [ 377 'sources': [
379 'overrides/init_webrtc.cc', 378 'overrides/init_webrtc.cc',
380 'overrides/init_webrtc.h', 379 'overrides/init_webrtc.h',
381 ], 380 ],
382 'dependencies': [ 381 'dependencies': [
383 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing', 382 '<(DEPTH)/third_party/webrtc/modules/modules.gyp:audio_processing',
384 'libjingle_webrtc_common', 383 'libjingle_webrtc_common',
385 ], 384 ],
386 }, 385 },
387 ], 386 ],
388 }], 387 }],
389 ], 388 ],
390 } 389 }
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