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

Issue 2838873002: Creating webrtc/modules:module_api (Closed)
Patch Set: fixing gn coding standards 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",
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
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
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
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
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 }
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