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Side by Side Diff: webrtc/voice_engine/BUILD.gn

Issue 2839963005: Revert of Creating webrtc/modules:module_api (Closed)
Patch Set: Created 3 years, 7 months ago
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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
(...skipping 63 matching lines...) Expand 10 before | Expand all | Expand 10 after
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",
(...skipping 11 matching lines...) Expand all
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",
(...skipping 43 matching lines...) Expand 10 before | Expand all | Expand 10 after
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",
(...skipping 44 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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