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Side by Side Diff: webrtc/voice_engine/voice_engine.gyp

Issue 1267683002: Hooked up RtcEventLog. It lives in Voice Engine and pointers are propagated to ACM and Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Another rebase Created 5 years, 3 months ago
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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',
(...skipping 107 matching lines...) Expand 10 before | Expand all | Expand 10 after
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/fakes/loudest_filter.cc', 167 'test/auto_test/fakes/loudest_filter.cc',
(...skipping 35 matching lines...) Expand 10 before | Expand all | Expand 10 after
201 'test/auto_test/voe_test_defines.h', 203 'test/auto_test/voe_test_defines.h',
202 'test/auto_test/voe_test_interface.h', 204 'test/auto_test/voe_test_interface.h',
203 ], 205 ],
204 'conditions': [ 206 'conditions': [
205 ['OS=="android"', { 207 ['OS=="android"', {
206 # some tests are not supported on android yet, exclude these tests . 208 # some tests are not supported on android yet, exclude these tests .
207 'sources!': [ 209 'sources!': [
208 'test/auto_test/standard/hardware_before_streaming_test.cc', 210 'test/auto_test/standard/hardware_before_streaming_test.cc',
209 ], 211 ],
210 }], 212 }],
213 ['enable_protobuf==1', {
214 'defines': [
215 'ENABLE_RTC_EVENT_LOG',
216 ],
217 }],
211 ], 218 ],
212 # Disable warnings to enable Win64 build, issue 1323. 219 # Disable warnings to enable Win64 build, issue 1323.
213 'msvs_disabled_warnings': [ 220 'msvs_disabled_warnings': [
214 4267, # size_t to int truncation. 221 4267, # size_t to int truncation.
215 ], 222 ],
216 }, 223 },
217 { 224 {
218 # command line test that should work on linux/mac/win 225 # command line test that should work on linux/mac/win
219 'target_name': 'voe_cmd_test', 226 'target_name': 'voe_cmd_test',
220 'type': 'executable', 227 'type': 'executable',
221 'dependencies': [ 228 'dependencies': [
222 'voice_engine', 229 'voice_engine',
223 '<(DEPTH)/testing/gtest.gyp:gtest', 230 '<(DEPTH)/testing/gtest.gyp:gtest',
224 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', 231 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
225 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers' , 232 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers' ,
226 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_ default', 233 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_ default',
227 '<(webrtc_root)/test/test.gyp:channel_transport', 234 '<(webrtc_root)/test/test.gyp:channel_transport',
228 '<(webrtc_root)/test/test.gyp:test_support', 235 '<(webrtc_root)/test/test.gyp:test_support',
236 '<(webrtc_root)/webrtc.gyp:rtc_event_log',
229 ], 237 ],
230 'sources': [ 238 'sources': [
231 'test/cmd_test/voe_cmd_test.cc', 239 'test/cmd_test/voe_cmd_test.cc',
232 ], 240 ],
233 }, 241 },
234 ], # targets 242 ], # targets
235 'conditions': [ 243 'conditions': [
236 ['OS=="android"', { 244 ['OS=="android"', {
237 'targets': [ 245 'targets': [
238 { 246 {
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after
271 'sources': [ 279 'sources': [
272 'voe_auto_test.isolate', 280 'voe_auto_test.isolate',
273 ], 281 ],
274 }, 282 },
275 ], 283 ],
276 }], 284 }],
277 ], # conditions 285 ], # conditions
278 }], # include_tests 286 }], # include_tests
279 ], # conditions 287 ], # conditions
280 } 288 }
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