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