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Side by Side Diff: webrtc/webrtc_tests.gypi

Issue 2059703002: Move webrtc/audio_*.h to webrtc/api/call (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Changed to webrtc/api/call Created 4 years, 6 months ago
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1 # Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 1 # Copyright (c) 2013 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 'targets': [ 9 'targets': [
10 { 10 {
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374 'video/stream_synchronization_unittest.cc', 374 'video/stream_synchronization_unittest.cc',
375 'video/video_capture_input_unittest.cc', 375 'video/video_capture_input_unittest.cc',
376 'video/video_decoder_unittest.cc', 376 'video/video_decoder_unittest.cc',
377 'video/video_encoder_unittest.cc', 377 'video/video_encoder_unittest.cc',
378 'video/video_send_stream_tests.cc', 378 'video/video_send_stream_tests.cc',
379 'video/vie_remb_unittest.cc', 379 'video/vie_remb_unittest.cc',
380 ], 380 ],
381 'dependencies': [ 381 'dependencies': [
382 '<(DEPTH)/testing/gmock.gyp:gmock', 382 '<(DEPTH)/testing/gmock.gyp:gmock',
383 '<(DEPTH)/testing/gtest.gyp:gtest', 383 '<(DEPTH)/testing/gtest.gyp:gtest',
384 '<(webrtc_root)/api/api.gyp:call_api',
384 '<(webrtc_root)/common.gyp:webrtc_common', 385 '<(webrtc_root)/common.gyp:webrtc_common',
385 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', 386 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
386 '<(webrtc_root)/modules/modules.gyp:video_capture', 387 '<(webrtc_root)/modules/modules.gyp:video_capture',
387 '<(webrtc_root)/test/test.gyp:channel_transport', 388 '<(webrtc_root)/test/test.gyp:channel_transport',
388 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', 389 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
389 'test/metrics.gyp:metrics', 390 'test/metrics.gyp:metrics',
390 'test/test.gyp:test_common', 391 'test/test.gyp:test_common',
391 'test/test.gyp:test_main', 392 'test/test.gyp:test_main',
392 'webrtc', 393 'webrtc',
393 ], 394 ],
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664 'build/isolate.gypi', 665 'build/isolate.gypi',
665 ], 666 ],
666 'sources': [ 667 'sources': [
667 'webrtc_perf_tests.isolate', 668 'webrtc_perf_tests.isolate',
668 ], 669 ],
669 }, 670 },
670 ], 671 ],
671 }], 672 }],
672 ], 673 ],
673 } 674 }
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