Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(25)

Side by Side Diff: webrtc/webrtc_tests.gypi

Issue 1510183002: Reland of Merge webrtc/video_engine/ into webrtc/video/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/video_renderer.h ('k') | no next file » | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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 {
(...skipping 142 matching lines...) Expand 10 before | Expand all | Expand 10 after
153 'sources': [ 153 'sources': [
154 'audio/audio_receive_stream_unittest.cc', 154 'audio/audio_receive_stream_unittest.cc',
155 'audio/audio_send_stream_unittest.cc', 155 'audio/audio_send_stream_unittest.cc',
156 'audio/audio_state_unittest.cc', 156 'audio/audio_state_unittest.cc',
157 'call/bitrate_allocator_unittest.cc', 157 'call/bitrate_allocator_unittest.cc',
158 'call/bitrate_estimator_tests.cc', 158 'call/bitrate_estimator_tests.cc',
159 'call/call_unittest.cc', 159 'call/call_unittest.cc',
160 'call/packet_injection_tests.cc', 160 'call/packet_injection_tests.cc',
161 'test/common_unittest.cc', 161 'test/common_unittest.cc',
162 'test/testsupport/metrics/video_metrics_unittest.cc', 162 'test/testsupport/metrics/video_metrics_unittest.cc',
163 'video/call_stats_unittest.cc',
164 'video/encoder_state_feedback_unittest.cc',
163 'video/end_to_end_tests.cc', 165 'video/end_to_end_tests.cc',
166 'video/overuse_frame_detector_unittest.cc',
167 'video/payload_router_unittest.cc',
168 'video/report_block_stats_unittest.cc',
164 'video/send_statistics_proxy_unittest.cc', 169 'video/send_statistics_proxy_unittest.cc',
170 'video/stream_synchronization_unittest.cc',
165 'video/video_capture_input_unittest.cc', 171 'video/video_capture_input_unittest.cc',
166 'video/video_decoder_unittest.cc', 172 'video/video_decoder_unittest.cc',
167 'video/video_encoder_unittest.cc', 173 'video/video_encoder_unittest.cc',
168 'video/video_send_stream_tests.cc', 174 'video/video_send_stream_tests.cc',
169 'video_engine/call_stats_unittest.cc', 175 'video/vie_codec_unittest.cc',
170 'video_engine/encoder_state_feedback_unittest.cc', 176 'video/vie_remb_unittest.cc',
171 'video_engine/overuse_frame_detector_unittest.cc',
172 'video_engine/payload_router_unittest.cc',
173 'video_engine/report_block_stats_unittest.cc',
174 'video_engine/stream_synchronization_unittest.cc',
175 'video_engine/vie_codec_unittest.cc',
176 'video_engine/vie_remb_unittest.cc',
177 ], 177 ],
178 'dependencies': [ 178 'dependencies': [
179 '<(DEPTH)/testing/gmock.gyp:gmock', 179 '<(DEPTH)/testing/gmock.gyp:gmock',
180 '<(DEPTH)/testing/gtest.gyp:gtest', 180 '<(DEPTH)/testing/gtest.gyp:gtest',
181 '<(webrtc_root)/common.gyp:webrtc_common', 181 '<(webrtc_root)/common.gyp:webrtc_common',
182 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', 182 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
183 '<(webrtc_root)/modules/modules.gyp:video_capture', 183 '<(webrtc_root)/modules/modules.gyp:video_capture',
184 '<(webrtc_root)/modules/modules.gyp:video_render', 184 '<(webrtc_root)/modules/modules.gyp:video_render',
185 '<(webrtc_root)/test/test.gyp:channel_transport', 185 '<(webrtc_root)/test/test.gyp:channel_transport',
186 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', 186 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
(...skipping 192 matching lines...) Expand 10 before | Expand all | Expand 10 after
379 'build/isolate.gypi', 379 'build/isolate.gypi',
380 ], 380 ],
381 'sources': [ 381 'sources': [
382 'webrtc_perf_tests.isolate', 382 'webrtc_perf_tests.isolate',
383 ], 383 ],
384 }, 384 },
385 ], 385 ],
386 }], 386 }],
387 ], 387 ],
388 } 388 }
OLDNEW
« no previous file with comments | « webrtc/video_renderer.h ('k') | no next file » | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698