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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': [ |
(...skipping 11 matching lines...) Expand all Loading... |
22 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 22 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
23 '<(webrtc_root)/modules/modules.gyp:audio_device', | 23 '<(webrtc_root)/modules/modules.gyp:audio_device', |
24 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 24 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
25 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', | 25 '<(webrtc_root)/modules/modules.gyp:bitrate_controller', |
26 '<(webrtc_root)/modules/modules.gyp:media_file', | 26 '<(webrtc_root)/modules/modules.gyp:media_file', |
27 '<(webrtc_root)/modules/modules.gyp:paced_sender', | 27 '<(webrtc_root)/modules/modules.gyp:paced_sender', |
28 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 28 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
29 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 29 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
30 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | 30 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', |
31 '<(webrtc_root)/webrtc.gyp:rtc_event_log', | 31 '<(webrtc_root)/webrtc.gyp:rtc_event_log', |
32 'file_player', | |
33 'file_recorder', | |
34 'level_indicator', | 32 'level_indicator', |
35 ], | 33 ], |
36 'export_dependent_settings': [ | 34 'export_dependent_settings': [ |
37 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 35 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
38 ], | 36 ], |
39 'sources': [ | 37 'sources': [ |
40 'include/voe_audio_processing.h', | 38 'include/voe_audio_processing.h', |
41 'include/voe_base.h', | 39 'include/voe_base.h', |
42 'include/voe_codec.h', | 40 'include/voe_codec.h', |
43 'include/voe_errors.h', | 41 'include/voe_errors.h', |
(...skipping 46 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
90 'voe_video_sync_impl.cc', | 88 'voe_video_sync_impl.cc', |
91 'voe_video_sync_impl.h', | 89 'voe_video_sync_impl.h', |
92 'voe_volume_control_impl.cc', | 90 'voe_volume_control_impl.cc', |
93 'voe_volume_control_impl.h', | 91 'voe_volume_control_impl.h', |
94 'voice_engine_defines.h', | 92 'voice_engine_defines.h', |
95 'voice_engine_impl.cc', | 93 'voice_engine_impl.cc', |
96 'voice_engine_impl.h', | 94 'voice_engine_impl.h', |
97 ], | 95 ], |
98 }, | 96 }, |
99 { | 97 { |
100 'target_name': 'audio_coder', | |
101 'type': 'static_library', | |
102 'sources': [ | |
103 'coder.cc', | |
104 'coder.h', | |
105 ], | |
106 'dependencies': [ | |
107 '<(webrtc_root)/common.gyp:webrtc_common', | |
108 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | |
109 '<(webrtc_root)/modules/modules.gyp:builtin_audio_decoder_factory', | |
110 '<(webrtc_root)/modules/modules.gyp:rent_a_codec', | |
111 ], | |
112 }, | |
113 { | |
114 'target_name': 'file_player', | |
115 'type': 'static_library', | |
116 'sources': [ | |
117 'file_player.h', | |
118 'file_player_impl.cc', | |
119 'file_player_impl.h', | |
120 ], | |
121 'dependencies': [ | |
122 '<(webrtc_root)/common.gyp:webrtc_common', | |
123 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | |
124 '<(webrtc_root)/modules/modules.gyp:media_file', | |
125 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | |
126 'audio_coder', | |
127 ], | |
128 }, | |
129 { | |
130 'target_name': 'file_recorder', | |
131 'type': 'static_library', | |
132 'sources': [ | |
133 'file_recorder.h', | |
134 'file_recorder_impl.cc', | |
135 'file_recorder_impl.h', | |
136 ], | |
137 'dependencies': [ | |
138 '<(webrtc_root)/base/base.gyp:rtc_base_approved', | |
139 '<(webrtc_root)/common.gyp:webrtc_common', | |
140 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | |
141 '<(webrtc_root)/modules/modules.gyp:media_file', | |
142 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers', | |
143 'audio_coder', | |
144 ], | |
145 }, | |
146 { | |
147 'target_name': 'level_indicator', | 98 'target_name': 'level_indicator', |
148 'type': 'static_library', | 99 'type': 'static_library', |
149 'dependencies': [ | 100 'dependencies': [ |
150 '<(webrtc_root)/base/base.gyp:rtc_base_approved', | 101 '<(webrtc_root)/base/base.gyp:rtc_base_approved', |
151 '<(webrtc_root)/common.gyp:webrtc_common', | 102 '<(webrtc_root)/common.gyp:webrtc_common', |
152 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | 103 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
153 ], | 104 ], |
154 'sources': [ | 105 'sources': [ |
155 'level_indicator.cc', | 106 'level_indicator.cc', |
156 'level_indicator.h', | 107 'level_indicator.h', |
157 ] | 108 ] |
158 } | 109 } |
159 ], | 110 ], |
160 'conditions': [ | 111 'conditions': [ |
161 ['OS=="win"', { | 112 ['OS=="win"', { |
162 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',], | 113 'defines': ['WEBRTC_DRIFT_COMPENSATION_SUPPORTED',], |
163 }], | 114 }], |
164 ['include_tests==1', { | 115 ['include_tests==1', { |
165 'targets': [ | 116 'targets': [ |
166 { | 117 { |
167 'target_name': 'voice_engine_unittests', | 118 'target_name': 'voice_engine_unittests', |
168 'type': '<(gtest_target_type)', | 119 'type': '<(gtest_target_type)', |
169 'dependencies': [ | 120 'dependencies': [ |
170 'voice_engine', | 121 'voice_engine', |
171 '<(DEPTH)/testing/gmock.gyp:gmock', | 122 '<(DEPTH)/testing/gmock.gyp:gmock', |
172 '<(DEPTH)/testing/gtest.gyp:gtest', | 123 '<(DEPTH)/testing/gtest.gyp:gtest', |
173 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', | |
174 # The rest are to satisfy the unittests' include chain. | 124 # The rest are to satisfy the unittests' include chain. |
175 # This would be unnecessary if we used qualified includes. | 125 # This would be unnecessary if we used qualified includes. |
176 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', | 126 '<(webrtc_root)/common_audio/common_audio.gyp:common_audio', |
177 '<(webrtc_root)/modules/modules.gyp:audio_device', | 127 '<(webrtc_root)/modules/modules.gyp:audio_device', |
178 '<(webrtc_root)/modules/modules.gyp:audio_processing', | 128 '<(webrtc_root)/modules/modules.gyp:audio_processing', |
179 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', | 129 '<(webrtc_root)/modules/modules.gyp:audio_coding_module', |
180 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', | 130 '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer', |
181 '<(webrtc_root)/modules/modules.gyp:media_file', | 131 '<(webrtc_root)/modules/modules.gyp:media_file', |
182 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', | 132 '<(webrtc_root)/modules/modules.gyp:rtp_rtcp', |
183 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', | 133 '<(webrtc_root)/modules/modules.gyp:webrtc_utility', |
184 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, | 134 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers'
, |
185 '<(webrtc_root)/test/test.gyp:test_support_main', | 135 '<(webrtc_root)/test/test.gyp:test_support_main', |
186 ], | 136 ], |
187 'sources': [ | 137 'sources': [ |
188 'channel_unittest.cc', | 138 'channel_unittest.cc', |
189 'file_player_unittests.cc', | |
190 'network_predictor_unittest.cc', | 139 'network_predictor_unittest.cc', |
191 'transmit_mixer_unittest.cc', | 140 'transmit_mixer_unittest.cc', |
192 'utility_unittest.cc', | 141 'utility_unittest.cc', |
193 'voe_audio_processing_unittest.cc', | 142 'voe_audio_processing_unittest.cc', |
194 'voe_base_unittest.cc', | 143 'voe_base_unittest.cc', |
195 'voe_codec_unittest.cc', | 144 'voe_codec_unittest.cc', |
196 'voe_network_unittest.cc', | 145 'voe_network_unittest.cc', |
197 'voice_engine_fixture.cc', | 146 'voice_engine_fixture.cc', |
198 'voice_engine_fixture.h', | 147 'voice_engine_fixture.h', |
199 ], | 148 ], |
200 'conditions': [ | 149 'conditions': [ |
201 ['OS=="android"', { | 150 ['OS=="android"', { |
202 'dependencies': [ | 151 'dependencies': [ |
203 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_cod
e', | 152 '<(DEPTH)/testing/android/native_test.gyp:native_test_native_cod
e', |
204 ], | 153 ], |
205 }], | 154 }], |
206 ['OS=="ios"', { | |
207 'mac_bundle_resources': [ | |
208 '<(DEPTH)/resources/utility/encapsulated_pcm16b_8khz.wav', | |
209 '<(DEPTH)/resources/utility/encapsulated_pcmu_8khz.wav', | |
210 ], | |
211 }], | |
212 ], | 155 ], |
213 }, | 156 }, |
214 { | 157 { |
215 # command line test that should work on linux/mac/win | 158 # command line test that should work on linux/mac/win |
216 'target_name': 'voe_cmd_test', | 159 'target_name': 'voe_cmd_test', |
217 'type': 'executable', | 160 'type': 'executable', |
218 'dependencies': [ | 161 'dependencies': [ |
219 'voice_engine', | 162 'voice_engine', |
220 '<(DEPTH)/testing/gtest.gyp:gtest', | 163 '<(DEPTH)/testing/gtest.gyp:gtest', |
221 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', | 164 '<(DEPTH)/third_party/gflags/gflags.gyp:gflags', |
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373 'sources': [ | 316 'sources': [ |
374 'voe_auto_test.isolate', | 317 'voe_auto_test.isolate', |
375 ], | 318 ], |
376 }, | 319 }, |
377 ], | 320 ], |
378 }], | 321 }], |
379 ], # conditions | 322 ], # conditions |
380 }], # include_tests==1 | 323 }], # include_tests==1 |
381 ], # conditions | 324 ], # conditions |
382 } | 325 } |
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