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Issue 1436553004: Created a test that reports the statistics for the duration of APM stream processing API calls. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updates to handle changes in external code used in this test Created 5 years 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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213 ], 213 ],
214 }], 214 }],
215 ], 215 ],
216 }, 216 },
217 { 217 {
218 'target_name': 'webrtc_perf_tests', 218 'target_name': 'webrtc_perf_tests',
219 'type': '<(gtest_target_type)', 219 'type': '<(gtest_target_type)',
220 'sources': [ 220 'sources': [
221 'call/call_perf_tests.cc', 221 'call/call_perf_tests.cc',
222 'modules/audio_coding/neteq/test/neteq_performance_unittest.cc', 222 'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
223 'modules/audio_processing/audio_processing_performance_unittest.cc',
223 'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc', 224 'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
224 'video/full_stack.cc', 225 'video/full_stack.cc',
225 'video/rampup_tests.cc', 226 'video/rampup_tests.cc',
226 'video/rampup_tests.h', 227 'video/rampup_tests.h',
227 ], 228 ],
228 'dependencies': [ 229 'dependencies': [
229 '<(DEPTH)/testing/gmock.gyp:gmock', 230 '<(DEPTH)/testing/gmock.gyp:gmock',
230 '<(DEPTH)/testing/gtest.gyp:gtest', 231 '<(DEPTH)/testing/gtest.gyp:gtest',
232 '<(webrtc_root)/modules/modules.gyp:audio_processing',
233 '<(webrtc_root)/modules/modules.gyp:audioproc_test_utils',
231 '<(webrtc_root)/modules/modules.gyp:video_capture', 234 '<(webrtc_root)/modules/modules.gyp:video_capture',
232 '<(webrtc_root)/test/test.gyp:channel_transport', 235 '<(webrtc_root)/test/test.gyp:channel_transport',
233 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine', 236 '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
234 'video_quality_test', 237 'video_quality_test',
235 'modules/modules.gyp:neteq_test_support', 238 'modules/modules.gyp:neteq_test_support',
236 'modules/modules.gyp:bwe_simulator', 239 'modules/modules.gyp:bwe_simulator',
237 'modules/modules.gyp:rtp_rtcp', 240 'modules/modules.gyp:rtp_rtcp',
238 'test/test.gyp:test_main', 241 'test/test.gyp:test_main',
239 'test/webrtc_test_common.gyp:webrtc_test_common', 242 'test/webrtc_test_common.gyp:webrtc_test_common',
240 'test/webrtc_test_common.gyp:webrtc_test_renderer', 243 'test/webrtc_test_common.gyp:webrtc_test_renderer',
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384 'build/isolate.gypi', 387 'build/isolate.gypi',
385 ], 388 ],
386 'sources': [ 389 'sources': [
387 'webrtc_perf_tests.isolate', 390 'webrtc_perf_tests.isolate',
388 ], 391 ],
389 }, 392 },
390 ], 393 ],
391 }], 394 }],
392 ], 395 ],
393 } 396 }
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