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