Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(229)

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

Issue 2245413002: Revert of Move FilePlayer and FileRecorder to Voice Engine (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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
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
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',
(...skipping 151 matching lines...) Expand 10 before | Expand all | Expand 10 after
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 }
OLDNEW
« no previous file with comments | « webrtc/voice_engine/transmit_mixer.h ('k') | webrtc/voice_engine/voice_engine_unittests.isolate » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698