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| 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("../webrtc.gni") | 9 import("../webrtc.gni") |
| 10 | 10 |
| 11 rtc_static_library("audio_coder") { | 11 rtc_static_library("audio_coder") { |
| 12 sources = [ | 12 sources = [ |
| 13 "coder.cc", | 13 "coder.cc", |
| 14 "coder.h", | 14 "coder.h", |
| 15 ] | 15 ] |
| 16 deps = [ | 16 deps = [ |
| 17 "..:webrtc_common", | 17 "..:webrtc_common", |
| 18 "../api/audio_codecs:builtin_audio_decoder_factory", | 18 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 19 "../modules:module_api", | |
| 20 "../modules/audio_coding", | 19 "../modules/audio_coding", |
| 21 "../modules/audio_coding:audio_encoder_factory_interface", | 20 "../modules/audio_coding:audio_encoder_factory_interface", |
| 22 "../modules/audio_coding:audio_format_conversion", | 21 "../modules/audio_coding:audio_format_conversion", |
| 23 "../modules/audio_coding:builtin_audio_encoder_factory", | 22 "../modules/audio_coding:builtin_audio_encoder_factory", |
| 24 "../modules/audio_coding:rent_a_codec", | 23 "../modules/audio_coding:rent_a_codec", |
| 25 ] | 24 ] |
| 26 | 25 |
| 27 if (!build_with_chromium && is_clang) { | 26 if (!build_with_chromium && is_clang) { |
| 28 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 27 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 29 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 28 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 30 } | 29 } |
| 31 } | 30 } |
| 32 | 31 |
| 33 rtc_static_library("file_player") { | 32 rtc_static_library("file_player") { |
| 34 sources = [ | 33 sources = [ |
| 35 "file_player.cc", | 34 "file_player.cc", |
| 36 "file_player.h", | 35 "file_player.h", |
| 37 ] | 36 ] |
| 38 deps = [ | 37 deps = [ |
| 39 ":audio_coder", | 38 ":audio_coder", |
| 40 "..:webrtc_common", | 39 "..:webrtc_common", |
| 41 "../base:rtc_base_approved", | 40 "../base:rtc_base_approved", |
| 42 "../common_audio", | 41 "../common_audio", |
| 43 "../modules:module_api", | |
| 44 "../modules/media_file", | 42 "../modules/media_file", |
| 45 ] | 43 ] |
| 46 | 44 |
| 47 if (!build_with_chromium && is_clang) { | 45 if (!build_with_chromium && is_clang) { |
| 48 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 46 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 49 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 47 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 50 } | 48 } |
| 51 } | 49 } |
| 52 | 50 |
| 53 rtc_static_library("file_recorder") { | 51 rtc_static_library("file_recorder") { |
| 54 sources = [ | 52 sources = [ |
| 55 "file_recorder.cc", | 53 "file_recorder.cc", |
| 56 "file_recorder.h", | 54 "file_recorder.h", |
| 57 ] | 55 ] |
| 58 deps = [ | 56 deps = [ |
| 59 ":audio_coder", | 57 ":audio_coder", |
| 60 "..:webrtc_common", | 58 "..:webrtc_common", |
| 61 "../base:rtc_base_approved", | 59 "../base:rtc_base_approved", |
| 62 "../common_audio", | 60 "../common_audio", |
| 63 "../modules:module_api", | |
| 64 "../modules/media_file:media_file", | 61 "../modules/media_file:media_file", |
| 65 "../system_wrappers", | 62 "../system_wrappers", |
| 66 ] | 63 ] |
| 67 | 64 |
| 68 if (!build_with_chromium && is_clang) { | 65 if (!build_with_chromium && is_clang) { |
| 69 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 66 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
| 70 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 67 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 71 } | 68 } |
| 72 } | 69 } |
| 73 | 70 |
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| 137 "..:webrtc_common", | 134 "..:webrtc_common", |
| 138 "../api:audio_mixer_api", | 135 "../api:audio_mixer_api", |
| 139 "../api:call_api", | 136 "../api:call_api", |
| 140 "../api:libjingle_peerconnection_api", | 137 "../api:libjingle_peerconnection_api", |
| 141 "../api:transport_api", | 138 "../api:transport_api", |
| 142 "../api/audio_codecs:audio_codecs_api", | 139 "../api/audio_codecs:audio_codecs_api", |
| 143 "../api/audio_codecs:builtin_audio_decoder_factory", | 140 "../api/audio_codecs:builtin_audio_decoder_factory", |
| 144 "../audio/utility:audio_frame_operations", | 141 "../audio/utility:audio_frame_operations", |
| 145 "../base:rtc_base_approved", | 142 "../base:rtc_base_approved", |
| 146 "../base:rtc_task_queue", | 143 "../base:rtc_task_queue", |
| 147 "../modules:module_api", | |
| 148 | 144 |
| 149 # TODO(nisse): Delete when declaration of RtpTransportController | 145 # TODO(nisse): Delete when declaration of RtpTransportController |
| 150 # and related interfaces move to api/. | 146 # and related interfaces move to api/. |
| 151 "../call:call_interfaces", | 147 "../call:call_interfaces", |
| 152 "../common_audio", | 148 "../common_audio", |
| 153 "../logging:rtc_event_log_api", | 149 "../logging:rtc_event_log_api", |
| 154 "../modules/audio_coding:audio_encoder_interface", | 150 "../modules/audio_coding:audio_encoder_interface", |
| 155 "../modules/audio_coding:audio_format_conversion", | 151 "../modules/audio_coding:audio_format_conversion", |
| 156 "../modules/audio_coding:rent_a_codec", | 152 "../modules/audio_coding:rent_a_codec", |
| 157 "../modules/audio_conference_mixer", | 153 "../modules/audio_conference_mixer", |
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| 169 rtc_static_library("audio_level") { | 165 rtc_static_library("audio_level") { |
| 170 sources = [ | 166 sources = [ |
| 171 "audio_level.cc", | 167 "audio_level.cc", |
| 172 "audio_level.h", | 168 "audio_level.h", |
| 173 ] | 169 ] |
| 174 | 170 |
| 175 deps = [ | 171 deps = [ |
| 176 "..:webrtc_common", | 172 "..:webrtc_common", |
| 177 "../base:rtc_base_approved", | 173 "../base:rtc_base_approved", |
| 178 "../common_audio", | 174 "../common_audio", |
| 179 "../modules:module_api", | |
| 180 ] | 175 ] |
| 181 } | 176 } |
| 182 | 177 |
| 183 if (rtc_include_tests) { | 178 if (rtc_include_tests) { |
| 184 rtc_test("voice_engine_unittests") { | 179 rtc_test("voice_engine_unittests") { |
| 185 deps = [ | 180 deps = [ |
| 186 ":file_player", | 181 ":file_player", |
| 187 ":voice_engine", | 182 ":voice_engine", |
| 188 "../base:rtc_base_approved", | 183 "../base:rtc_base_approved", |
| 189 "../modules:module_api", | |
| 190 "../test:test_common", | 184 "../test:test_common", |
| 191 "//testing/gmock", | 185 "//testing/gmock", |
| 192 "//testing/gtest", | 186 "//testing/gtest", |
| 193 "//third_party/gflags", | 187 "//third_party/gflags", |
| 194 "//webrtc/common_audio", | 188 "//webrtc/common_audio", |
| 195 "//webrtc/modules/audio_coding", | 189 "//webrtc/modules/audio_coding", |
| 196 "//webrtc/modules/audio_conference_mixer", | 190 "//webrtc/modules/audio_conference_mixer", |
| 197 "//webrtc/modules/audio_device", | 191 "//webrtc/modules/audio_device", |
| 198 "//webrtc/modules/audio_processing", | 192 "//webrtc/modules/audio_processing", |
| 199 "//webrtc/modules/media_file", | 193 "//webrtc/modules/media_file", |
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| 243 } | 237 } |
| 244 | 238 |
| 245 if (!is_ios) { | 239 if (!is_ios) { |
| 246 rtc_executable("voe_auto_test") { | 240 rtc_executable("voe_auto_test") { |
| 247 testonly = true | 241 testonly = true |
| 248 | 242 |
| 249 deps = [ | 243 deps = [ |
| 250 ":voice_engine", | 244 ":voice_engine", |
| 251 "..:webrtc_common", | 245 "..:webrtc_common", |
| 252 "../base:rtc_base_approved", | 246 "../base:rtc_base_approved", |
| 253 "../modules:module_api", | |
| 254 "../modules/audio_device:audio_device", | 247 "../modules/audio_device:audio_device", |
| 255 "../modules/audio_processing:audio_processing", | 248 "../modules/audio_processing:audio_processing", |
| 256 "../modules/rtp_rtcp:rtp_rtcp", | 249 "../modules/rtp_rtcp:rtp_rtcp", |
| 257 "//testing/gmock", | 250 "//testing/gmock", |
| 258 "//testing/gtest", | 251 "//testing/gtest", |
| 259 "//third_party/gflags", | 252 "//third_party/gflags", |
| 260 "//webrtc/logging:rtc_event_log_api", | 253 "//webrtc/logging:rtc_event_log_api", |
| 261 "//webrtc/modules/video_capture", | 254 "//webrtc/modules/video_capture", |
| 262 "//webrtc/system_wrappers", | 255 "//webrtc/system_wrappers", |
| 263 "//webrtc/system_wrappers/:system_wrappers_default", | 256 "//webrtc/system_wrappers/:system_wrappers_default", |
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| 308 ] | 301 ] |
| 309 } | 302 } |
| 310 | 303 |
| 311 if (!build_with_chromium && is_clang) { | 304 if (!build_with_chromium && is_clang) { |
| 312 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. | 305 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
. |
| 313 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 306 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
| 314 } | 307 } |
| 315 } | 308 } |
| 316 } | 309 } |
| 317 } | 310 } |
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