blob: d228341ea485652bbb4dede1691e6c47d13e537b [file] [log] [blame]
xians@google.comd3185fe2011-09-12 12:24:39 +00001# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
2#
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
5# tree. An additional intellectual property rights grant can be found
6# in the file PATENTS. All contributing project authors may
7# be found in the AUTHORS file in the root of the source tree.
8
9{
10 'includes': [
andrew@webrtc.org3ce62fc2011-10-06 01:03:18 +000011 '../build/common.gypi',
xians@google.comd3185fe2011-09-12 12:24:39 +000012 ],
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +000013 'targets': [
14 {
15 'target_name': 'voice_engine',
16 'type': 'static_library',
17 'dependencies': [
18 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
19 '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
20 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
21 '<(webrtc_root)/modules/modules.gyp:audio_device',
22 '<(webrtc_root)/modules/modules.gyp:audio_processing',
23 '<(webrtc_root)/modules/modules.gyp:media_file',
24 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
25 '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
26 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
andrew@webrtc.org3ce62fc2011-10-06 01:03:18 +000027 ],
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +000028 'include_dirs': [
29 'include',
30 '<(webrtc_root)/modules/audio_device',
31 ],
32 'direct_dependent_settings': {
33 'include_dirs': [
34 'include',
35 ],
36 },
37 'sources': [
38 '../common_types.h',
39 '../engine_configurations.h',
40 '../typedefs.h',
41 'include/voe_audio_processing.h',
42 'include/voe_base.h',
43 'include/voe_call_report.h',
44 'include/voe_codec.h',
45 'include/voe_dtmf.h',
46 'include/voe_encryption.h',
47 'include/voe_errors.h',
48 'include/voe_external_media.h',
49 'include/voe_file.h',
50 'include/voe_hardware.h',
51 'include/voe_neteq_stats.h',
52 'include/voe_network.h',
53 'include/voe_rtp_rtcp.h',
54 'include/voe_video_sync.h',
55 'include/voe_volume_control.h',
56 'channel.cc',
57 'channel.h',
58 'channel_manager.cc',
59 'channel_manager.h',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +000060 'dtmf_inband.cc',
61 'dtmf_inband.h',
62 'dtmf_inband_queue.cc',
63 'dtmf_inband_queue.h',
64 'level_indicator.cc',
65 'level_indicator.h',
66 'monitor_module.cc',
67 'monitor_module.h',
68 'output_mixer.cc',
69 'output_mixer.h',
70 'output_mixer_internal.cc',
71 'output_mixer_internal.h',
72 'shared_data.cc',
73 'shared_data.h',
74 'statistics.cc',
75 'statistics.h',
76 'transmit_mixer.cc',
77 'transmit_mixer.h',
78 'utility.cc',
79 'utility.h',
80 'voe_audio_processing_impl.cc',
81 'voe_audio_processing_impl.h',
82 'voe_base_impl.cc',
83 'voe_base_impl.h',
84 'voe_call_report_impl.cc',
85 'voe_call_report_impl.h',
86 'voe_codec_impl.cc',
87 'voe_codec_impl.h',
88 'voe_dtmf_impl.cc',
89 'voe_dtmf_impl.h',
90 'voe_encryption_impl.cc',
91 'voe_encryption_impl.h',
92 'voe_external_media_impl.cc',
93 'voe_external_media_impl.h',
94 'voe_file_impl.cc',
95 'voe_file_impl.h',
96 'voe_hardware_impl.cc',
97 'voe_hardware_impl.h',
98 'voe_neteq_stats_impl.cc',
99 'voe_neteq_stats_impl.h',
100 'voe_network_impl.cc',
101 'voe_network_impl.h',
102 'voe_rtp_rtcp_impl.cc',
103 'voe_rtp_rtcp_impl.h',
104 'voe_video_sync_impl.cc',
105 'voe_video_sync_impl.h',
106 'voe_volume_control_impl.cc',
107 'voe_volume_control_impl.h',
108 'voice_engine_defines.h',
109 'voice_engine_impl.cc',
110 'voice_engine_impl.h',
111 ],
112 },
andrew@webrtc.org3ce62fc2011-10-06 01:03:18 +0000113 ],
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000114 'conditions': [
115 ['OS=="win"', {
116 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',],
117 }],
118 ['include_tests==1', {
119 'targets': [
120 {
121 'target_name': 'voice_engine_unittests',
henrike@webrtc.org89c67402013-08-02 16:53:47 +0000122 'type': '<(gtest_target_type)',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000123 'dependencies': [
124 'voice_engine',
125 '<(DEPTH)/testing/gtest.gyp:gtest',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000126 # The rest are to satisfy the unittests' include chain.
127 # This would be unnecessary if we used qualified includes.
128 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
129 '<(webrtc_root)/modules/modules.gyp:audio_device',
130 '<(webrtc_root)/modules/modules.gyp:audio_processing',
131 '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
132 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
133 '<(webrtc_root)/modules/modules.gyp:media_file',
134 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
135 '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
136 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
kjellander@webrtc.org6c35e0b2013-06-11 08:29:17 +0000137 '<(webrtc_root)/test/test.gyp:test_support_main',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000138 ],
139 'include_dirs': [
140 'include',
141 ],
142 'sources': [
143 'channel_unittest.cc',
144 'output_mixer_unittest.cc',
145 'transmit_mixer_unittest.cc',
146 'voe_audio_processing_unittest.cc',
147 'voe_base_unittest.cc',
148 'voe_codec_unittest.cc',
149 ],
henrike@webrtc.org89c67402013-08-02 16:53:47 +0000150 'conditions': [
151 # TODO(henrike): remove build_with_chromium==1 when the bots are
152 # using Chromium's buildbots.
153 ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', {
154 'dependencies': [
155 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
156 ],
157 }],
158 ],
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000159 },
160 {
161 'target_name': 'voe_auto_test',
162 'type': 'executable',
163 'dependencies': [
164 'voice_engine',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000165 '<(DEPTH)/testing/gmock.gyp:gmock',
kjellander@webrtc.org6c35e0b2013-06-11 08:29:17 +0000166 '<(DEPTH)/testing/gtest.gyp:gtest',
fischman@webrtc.orgdde7d4c2013-08-15 23:31:30 +0000167 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
kjellander@webrtc.org6c35e0b2013-06-11 08:29:17 +0000168 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000169 '<(webrtc_root)/test/libtest/libtest.gyp:libtest',
kjellander@webrtc.org6c35e0b2013-06-11 08:29:17 +0000170 '<(webrtc_root)/test/test.gyp:channel_transport',
171 '<(webrtc_root)/test/test.gyp:test_support',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000172 ],
173 'include_dirs': [
174 'auto_test',
175 'auto_test/fixtures',
176 '<(webrtc_root)/modules/interface',
177 # TODO(phoglund): We only depend on voice_engine_defines.h here -
178 # move that file to interface and then remove this dependency.
179 '<(webrtc_root)/voice_engine',
180 '<(webrtc_root)/modules/audio_device/main/interface',
181 ],
182 'sources': [
183 'test/auto_test/automated_mode.cc',
184 'test/auto_test/extended/agc_config_test.cc',
185 'test/auto_test/extended/ec_metrics_test.cc',
186 'test/auto_test/fakes/fake_external_transport.cc',
187 'test/auto_test/fakes/fake_external_transport.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/fuzz/rtp_fuzz_test.cc',
195 'test/auto_test/standard/audio_processing_test.cc',
196 'test/auto_test/standard/call_report_test.cc',
197 'test/auto_test/standard/codec_before_streaming_test.cc',
198 'test/auto_test/standard/codec_test.cc',
199 'test/auto_test/standard/dtmf_test.cc',
200 'test/auto_test/standard/encryption_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_before_streaming_test.cc',
206 'test/auto_test/standard/hardware_test.cc',
207 'test/auto_test/standard/manual_hold_test.cc',
208 'test/auto_test/standard/mixing_test.cc',
209 'test/auto_test/standard/neteq_stats_test.cc',
210 'test/auto_test/standard/neteq_test.cc',
211 'test/auto_test/standard/network_test.cc',
212 'test/auto_test/standard/rtp_rtcp_before_streaming_test.cc',
213 'test/auto_test/standard/rtp_rtcp_test.cc',
214 'test/auto_test/standard/voe_base_misc_test.cc',
215 'test/auto_test/standard/video_sync_test.cc',
216 'test/auto_test/standard/volume_test.cc',
217 'test/auto_test/resource_manager.cc',
218 'test/auto_test/voe_cpu_test.cc',
219 'test/auto_test/voe_cpu_test.h',
220 'test/auto_test/voe_extended_test.cc',
221 'test/auto_test/voe_extended_test.h',
222 'test/auto_test/voe_standard_test.cc',
223 'test/auto_test/voe_standard_test.h',
224 'test/auto_test/voe_stress_test.cc',
225 'test/auto_test/voe_stress_test.h',
226 'test/auto_test/voe_test_defines.h',
227 'test/auto_test/voe_test_interface.h',
228 'test/auto_test/voe_unit_test.cc',
229 'test/auto_test/voe_unit_test.h',
230 ],
231 'conditions': [
232 ['OS=="android"', {
233 # some tests are not supported on android yet, exclude these tests.
234 'sources!': [
235 'test/auto_test/standard/hardware_before_streaming_test.cc',
236 ],
237 }],
238 ],
239 # Disable warnings to enable Win64 build, issue 1323.
240 'msvs_disabled_warnings': [
241 4267, # size_t to int truncation.
242 ],
243 },
244 {
245 # command line test that should work on linux/mac/win
246 'target_name': 'voe_cmd_test',
247 'type': 'executable',
248 'dependencies': [
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000249 'voice_engine',
kjellander@webrtc.org6c35e0b2013-06-11 08:29:17 +0000250 '<(DEPTH)/testing/gtest.gyp:gtest',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000251 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
kjellander@webrtc.org6c35e0b2013-06-11 08:29:17 +0000252 '<(webrtc_root)/test/test.gyp:channel_transport',
253 '<(webrtc_root)/test/test.gyp:test_support',
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000254 ],
255 'sources': [
256 'test/cmd_test/voe_cmd_test.cc',
257 ],
258 },
259 ], # targets
260 'conditions': [
261 # TODO(kjellander): Support UseoFMFC on VS2010.
262 # http://code.google.com/p/webrtc/issues/detail?id=709
263 ['OS=="win" and MSVS_VERSION < "2010"', {
264 'targets': [
265 # WinTest - GUI test for Windows
266 {
267 'target_name': 'voe_ui_win_test',
268 'type': 'executable',
269 'dependencies': [
270 'voice_engine',
271 '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
272 '<(webrtc_root)/test/test.gyp:test_support',
273 ],
274 'include_dirs': [
275 'win_test',
276 ],
277 'sources': [
278 'test/win_test/Resource.h',
279 'test/win_test/WinTest.cc',
280 'test/win_test/WinTest.h',
281 'test/win_test/WinTest.rc',
282 'test/win_test/WinTestDlg.cc',
283 'test/win_test/WinTestDlg.h',
284 'test/win_test/res/WinTest.ico',
285 'test/win_test/res/WinTest.rc2',
286 'test/win_test/stdafx.cc',
287 'test/win_test/stdafx.h',
288 ],
289 'configurations': {
290 'Common_Base': {
291 'msvs_configuration_attributes': {
292 'conditions': [
293 ['component=="shared_library"', {
294 'UseOfMFC': '2', # Shared DLL
295 },{
296 'UseOfMFC': '1', # Static
297 }],
298 ],
299 },
300 },
301 },
302 'msvs_settings': {
303 'VCLinkerTool': {
304 'SubSystem': '2', # Windows
305 },
306 },
307 },
308 ], # targets
309 }],
henrike@webrtc.org89c67402013-08-02 16:53:47 +0000310 # TODO(henrike): remove build_with_chromium==1 when the bots are using
311 # Chromium's buildbots.
312 ['build_with_chromium==1 and OS=="android" and gtest_target_type=="shared_library"', {
313 'targets': [
314 {
315 'target_name': 'voice_engine_unittests_apk_target',
316 'type': 'none',
317 'dependencies': [
318 '<(apk_tests_path):voice_engine_unittests_apk',
319 ],
320 },
321 ],
322 }],
andrew@webrtc.orgf791b1c2013-05-29 00:38:02 +0000323 ], # conditions
324 }], # include_tests
325 ], # conditions
xians@google.comd3185fe2011-09-12 12:24:39 +0000326}