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1 # Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2011 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 { | 9 { |
10 'includes': [ | 10 'includes': [ |
11 '../build/common.gypi', | 11 '../build/common.gypi', |
12 ], | 12 ], |
13 'targets': [ | 13 'targets': [ |
14 { | 14 { |
15 'target_name': 'voice_engine', | 15 'target_name': 'voice_engine', |
16 'type': 'static_library', | 16 'type': 'static_library', |
17 'dependencies': [ | 17 'dependencies': [ |
18 '<(webrtc_root)/common.gyp:webrtc_common', | 18 '<(webrtc_root)/common.gyp:webrtc_common', |
19 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | 19 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
20 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 20 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
21 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 21 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
22 '<(webrtc_root)/modules/modules.gyp:audio_device', | 22 '<(webrtc_root)/modules/modules.gyp:audio_device', |
23 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 23 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
24 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', | 24 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', |
25 '<(webrtc_root)/modules/modules.gyp:media_file', | 25 '<(webrtc_root)/modules/modules.gyp:media_file', |
26 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 26 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
27 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 27 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
28 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | 28 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', |
| 29 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
29 ], | 30 ], |
30 'sources': [ | 31 'sources': [ |
31 'include/voe_audio_processing.h', | 32 'include/voe_audio_processing.h', |
32 'include/voe_base.h', | 33 'include/voe_base.h', |
33 'include/voe_codec.h', | 34 'include/voe_codec.h', |
34 'include/voe_dtmf.h', | 35 'include/voe_dtmf.h', |
35 'include/voe_errors.h', | 36 'include/voe_errors.h', |
36 'include/voe_external_media.h', | 37 'include/voe_external_media.h', |
37 'include/voe_file.h', | 38 'include/voe_file.h', |
38 'include/voe_hardware.h', | 39 'include/voe_hardware.h', |
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146 'type': 'executable', | 147 'type': 'executable', |
147 'dependencies': [ | 148 'dependencies': [ |
148 'voice_engine', | 149 'voice_engine', |
149 '<(DEPTH)/testing/gmock.gyp:gmock', | 150 '<(DEPTH)/testing/gmock.gyp:gmock', |
150 '<(DEPTH)/testing/gtest.gyp:gtest', | 151 '<(DEPTH)/testing/gtest.gyp:gtest', |
151 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', | 152 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', |
152 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, | 153 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, |
153 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_
default', | 154 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_
default', |
154 '<(webrtc_root)/test/test.gyp:channel_transport', | 155 '<(webrtc_root)/test/test.gyp:channel_transport', |
155 '<(webrtc_root)/test/test.gyp:test_support', | 156 '<(webrtc_root)/test/test.gyp:test_support', |
| 157 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
156 ], | 158 ], |
157 'sources': [ | 159 'sources': [ |
158 'test/auto_test/automated_mode.cc', | 160 'test/auto_test/automated_mode.cc', |
159 'test/auto_test/extended/agc_config_test.cc', | 161 'test/auto_test/extended/agc_config_test.cc', |
160 'test/auto_test/extended/ec_metrics_test.cc', | 162 'test/auto_test/extended/ec_metrics_test.cc', |
161 'test/auto_test/fakes/conference_transport.cc', | 163 'test/auto_test/fakes/conference_transport.cc', |
162 'test/auto_test/fakes/conference_transport.h', | 164 'test/auto_test/fakes/conference_transport.h', |
163 'test/auto_test/fakes/fake_external_transport.cc', | 165 'test/auto_test/fakes/fake_external_transport.cc', |
164 'test/auto_test/fakes/fake_external_transport.h', | 166 'test/auto_test/fakes/fake_external_transport.h', |
165 'test/auto_test/fixtures/after_initialization_fixture.cc', | 167 'test/auto_test/fixtures/after_initialization_fixture.cc', |
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199 'test/auto_test/voe_test_defines.h', | 201 'test/auto_test/voe_test_defines.h', |
200 'test/auto_test/voe_test_interface.h', | 202 'test/auto_test/voe_test_interface.h', |
201 ], | 203 ], |
202 'conditions': [ | 204 'conditions': [ |
203 ['OS=="android"', { | 205 ['OS=="android"', { |
204 # some tests are not supported on android yet, exclude these tests
. | 206 # some tests are not supported on android yet, exclude these tests
. |
205 'sources!': [ | 207 'sources!': [ |
206 'test/auto_test/standard/hardware_before_streaming_test.cc', | 208 'test/auto_test/standard/hardware_before_streaming_test.cc', |
207 ], | 209 ], |
208 }], | 210 }], |
| 211 ['enable_protobuf==1', { |
| 212 'defines': [ |
| 213 'ENABLE_RTC_EVENT_LOG', |
| 214 ], |
| 215 }], |
209 ], | 216 ], |
210 # Disable warnings to enable Win64 build, issue 1323. | 217 # Disable warnings to enable Win64 build, issue 1323. |
211 'msvs_disabled_warnings': [ | 218 'msvs_disabled_warnings': [ |
212 4267, # size_t to int truncation. | 219 4267, # size_t to int truncation. |
213 ], | 220 ], |
214 }, | 221 }, |
215 { | 222 { |
216 # command line test that should work on linux/mac/win | 223 # command line test that should work on linux/mac/win |
217 'target_name': 'voe_cmd_test', | 224 'target_name': 'voe_cmd_test', |
218 'type': 'executable', | 225 'type': 'executable', |
219 'dependencies': [ | 226 'dependencies': [ |
220 'voice_engine', | 227 'voice_engine', |
221 '<(DEPTH)/testing/gtest.gyp:gtest', | 228 '<(DEPTH)/testing/gtest.gyp:gtest', |
222 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', | 229 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', |
223 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, | 230 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, |
224 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_
default', | 231 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_
default', |
225 '<(webrtc_root)/test/test.gyp:channel_transport', | 232 '<(webrtc_root)/test/test.gyp:channel_transport', |
226 '<(webrtc_root)/test/test.gyp:test_support', | 233 '<(webrtc_root)/test/test.gyp:test_support', |
| 234 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
227 ], | 235 ], |
228 'sources': [ | 236 'sources': [ |
229 'test/cmd_test/voe_cmd_test.cc', | 237 'test/cmd_test/voe_cmd_test.cc', |
230 ], | 238 ], |
231 }, | 239 }, |
232 ], # targets | 240 ], # targets |
233 'conditions': [ | 241 'conditions': [ |
234 ['OS=="android"', { | 242 ['OS=="android"', { |
235 'targets': [ | 243 'targets': [ |
236 { | 244 { |
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269 'sources': [ | 277 'sources': [ |
270 'voe_auto_test.isolate', | 278 'voe_auto_test.isolate', |
271 ], | 279 ], |
272 }, | 280 }, |
273 ], | 281 ], |
274 }], | 282 }], |
275 ], # conditions | 283 ], # conditions |
276 }], # include_tests | 284 }], # include_tests |
277 ], # conditions | 285 ], # conditions |
278 } | 286 } |
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