OLD | NEW |
---|---|
1 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2014 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 import("../build/webrtc.gni") | 9 import("../build/webrtc.gni") |
10 import("//testing/test.gni") | |
10 | 11 |
11 source_set("voice_engine") { | 12 source_set("voice_engine") { |
12 sources = [ | 13 sources = [ |
13 "channel.cc", | 14 "channel.cc", |
14 "channel.h", | 15 "channel.h", |
15 "channel_manager.cc", | 16 "channel_manager.cc", |
16 "channel_manager.h", | 17 "channel_manager.h", |
17 "channel_proxy.cc", | 18 "channel_proxy.cc", |
18 "channel_proxy.h", | 19 "channel_proxy.h", |
19 "include/voe_audio_processing.h", | 20 "include/voe_audio_processing.h", |
(...skipping 79 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... | |
99 "../modules/audio_device", | 100 "../modules/audio_device", |
100 "../modules/audio_processing", | 101 "../modules/audio_processing", |
101 "../modules/bitrate_controller", | 102 "../modules/bitrate_controller", |
102 "../modules/media_file", | 103 "../modules/media_file", |
103 "../modules/pacing", | 104 "../modules/pacing", |
104 "../modules/rtp_rtcp", | 105 "../modules/rtp_rtcp", |
105 "../modules/utility", | 106 "../modules/utility", |
106 "../system_wrappers", | 107 "../system_wrappers", |
107 ] | 108 ] |
108 } | 109 } |
110 | |
111 if (rtc_include_tests) { | |
112 test("voice_engine_unittests") { | |
113 deps = [ | |
114 ":voice_engine", | |
115 "//testing/gmock", | |
116 "//testing/gtest", | |
117 "//webrtc/common_audio", | |
118 "//webrtc/modules/audio_coding", | |
119 "//webrtc/modules/audio_conference_mixer", | |
120 "//webrtc/modules/audio_device", | |
121 "//webrtc/modules/audio_processing", | |
122 "//webrtc/modules/media_file", | |
123 "//webrtc/modules/rtp_rtcp", | |
124 "//webrtc/modules/utility", | |
125 "//webrtc/system_wrappers", | |
126 "//webrtc/test:test_support_main", | |
127 ] | |
128 | |
129 if (is_android) { | |
130 deps += [ "//testing/android/native_test:native_test_native_code" ] | |
131 } | |
132 | |
133 sources = [ | |
134 "channel_unittest.cc", | |
135 "network_predictor_unittest.cc", | |
136 "transmit_mixer_unittest.cc", | |
137 "utility_unittest.cc", | |
138 "voe_audio_processing_unittest.cc", | |
139 "voe_base_unittest.cc", | |
140 "voe_codec_unittest.cc", | |
141 "voe_network_unittest.cc", | |
142 "voice_engine_fixture.cc", | |
143 "voice_engine_fixture.h", | |
144 ] | |
145 | |
146 if (is_win) { | |
147 defines = [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ] | |
148 | |
149 cflags = [ | |
150 # TODO(kjellander): Bug 261: fix this warning. | |
151 "/wd4373", # virtual function override. | |
152 ] | |
153 } | |
154 | |
155 if (is_clang) { | |
phoglund
2016/07/07 12:26:42
Where is this from? Can't find anything correspond
ossu
2016/07/07 12:42:42
I took them from the voice_engine target above.. I
| |
156 # Suppress warnings from Chrome's Clang plugins. | |
157 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. | |
158 configs -= [ "//build/config/clang:find_bad_constructs" ] | |
159 } | |
160 } | |
161 | |
162 if (!is_ios) { | |
163 executable("voe_auto_test") { | |
164 testonly = true | |
165 | |
166 deps = [ | |
167 ":voice_engine", | |
168 "//testing/gmock", | |
169 "//testing/gtest", | |
170 "//third_party/gflags", | |
171 "//webrtc/:rtc_event_log", | |
172 "//webrtc/modules/video_capture", | |
173 "//webrtc/system_wrappers", | |
174 "//webrtc/system_wrappers/:system_wrappers_default", | |
175 "//webrtc/test/:channel_transport", | |
176 "//webrtc/test/:test_common", | |
177 "//webrtc/test/:test_support", | |
178 ] | |
179 | |
180 sources = [ | |
181 "test/auto_test/automated_mode.cc", | |
182 "test/auto_test/extended/agc_config_test.cc", | |
183 "test/auto_test/extended/ec_metrics_test.cc", | |
184 "test/auto_test/fakes/conference_transport.cc", | |
185 "test/auto_test/fakes/conference_transport.h", | |
186 "test/auto_test/fakes/loudest_filter.cc", | |
187 "test/auto_test/fakes/loudest_filter.h", | |
188 "test/auto_test/fixtures/after_initialization_fixture.cc", | |
189 "test/auto_test/fixtures/after_initialization_fixture.h", | |
190 "test/auto_test/fixtures/after_streaming_fixture.cc", | |
191 "test/auto_test/fixtures/after_streaming_fixture.h", | |
192 "test/auto_test/fixtures/before_initialization_fixture.cc", | |
193 "test/auto_test/fixtures/before_initialization_fixture.h", | |
194 "test/auto_test/fixtures/before_streaming_fixture.cc", | |
195 "test/auto_test/fixtures/before_streaming_fixture.h", | |
196 "test/auto_test/resource_manager.cc", | |
197 "test/auto_test/standard/audio_processing_test.cc", | |
198 "test/auto_test/standard/codec_before_streaming_test.cc", | |
199 "test/auto_test/standard/codec_test.cc", | |
200 "test/auto_test/standard/dtmf_test.cc", | |
201 "test/auto_test/standard/external_media_test.cc", | |
202 "test/auto_test/standard/file_before_streaming_test.cc", | |
203 "test/auto_test/standard/file_test.cc", | |
204 "test/auto_test/standard/hardware_before_initializing_test.cc", | |
205 "test/auto_test/standard/hardware_test.cc", | |
206 "test/auto_test/standard/mixing_test.cc", | |
207 "test/auto_test/standard/neteq_stats_test.cc", | |
208 "test/auto_test/standard/rtp_rtcp_before_streaming_test.cc", | |
209 "test/auto_test/standard/rtp_rtcp_extensions.cc", | |
210 "test/auto_test/standard/rtp_rtcp_test.cc", | |
211 "test/auto_test/standard/video_sync_test.cc", | |
212 "test/auto_test/standard/voe_base_misc_test.cc", | |
213 "test/auto_test/standard/volume_test.cc", | |
214 "test/auto_test/voe_conference_test.cc", | |
215 "test/auto_test/voe_cpu_test.cc", | |
216 "test/auto_test/voe_cpu_test.h", | |
217 "test/auto_test/voe_output_test.cc", | |
218 "test/auto_test/voe_standard_test.cc", | |
219 "test/auto_test/voe_standard_test.h", | |
220 "test/auto_test/voe_stress_test.cc", | |
221 "test/auto_test/voe_stress_test.h", | |
222 "test/auto_test/voe_test_defines.h", | |
223 "test/auto_test/voe_test_interface.h", | |
224 ] | |
225 | |
226 if (!is_android) { | |
227 # some tests are not supported on android yet, exclude these tests. | |
phoglund
2016/07/07 12:26:42
Nit: uppercase S
ossu
2016/07/07 12:42:42
I stole these comments from gyp, but I'll fix 'em
| |
228 sources += | |
229 [ "test/auto_test/standard/hardware_before_streaming_test.cc" ] | |
230 } | |
231 | |
232 defines = [] | |
233 | |
234 if (rtc_enable_protobuf) { | |
235 defines = [ "ENABLE_RTC_EVENT_LOG" ] | |
236 } | |
237 | |
238 if (is_win) { | |
239 defines += [ "WEBRTC_DRIFT_COMPENSATION_SUPPORTED" ] | |
240 | |
241 cflags = [ | |
242 "/wd4267", # size_t to int truncation. | |
243 "/wd4373", # virtual function override. | |
phoglund
2016/07/07 12:26:42
Nit: uppercase v
ossu
2016/07/07 12:42:42
Acknowledged.
| |
244 # TODO(kjellander): Bug 261: fix this warning. | |
245 ] | |
246 } | |
247 | |
248 if (is_clang) { | |
249 # Suppress warnings from Chrome's Clang plugins. | |
250 # See http://code.google.com/p/webrtc/issues/detail?id=163 for details. | |
251 configs -= [ "//build/config/clang:find_bad_constructs" ] | |
252 } | |
253 } | |
254 } | |
255 } | |
OLD | NEW |