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1 # Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. | 1 # Copyright (c) 2016 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("//build/config/linux/pkg_config.gni") | 9 import("//build/config/linux/pkg_config.gni") |
10 import("../build/webrtc.gni") | 10 import("../build/webrtc.gni") |
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47 libs = [] | 47 libs = [] |
48 deps = [] | 48 deps = [] |
49 sources = [ | 49 sources = [ |
50 "base/adaptedvideotracksource.cc", | 50 "base/adaptedvideotracksource.cc", |
51 "base/adaptedvideotracksource.h", | 51 "base/adaptedvideotracksource.h", |
52 "base/audiosource.h", | 52 "base/audiosource.h", |
53 "base/codec.cc", | 53 "base/codec.cc", |
54 "base/codec.h", | 54 "base/codec.h", |
55 "base/cryptoparams.h", | 55 "base/cryptoparams.h", |
56 "base/device.h", | 56 "base/device.h", |
57 "base/hybriddataengine.h", | |
58 "base/mediachannel.h", | 57 "base/mediachannel.h", |
59 "base/mediaconstants.cc", | 58 "base/mediaconstants.cc", |
60 "base/mediaconstants.h", | 59 "base/mediaconstants.h", |
61 "base/mediaengine.cc", | 60 "base/mediaengine.cc", |
62 "base/mediaengine.h", | 61 "base/mediaengine.h", |
63 "base/rtpdataengine.cc", | 62 "base/rtpdataengine.cc", |
64 "base/rtpdataengine.h", | 63 "base/rtpdataengine.h", |
65 "base/rtpdump.cc", | 64 "base/rtpdump.cc", |
66 "base/rtpdump.h", | 65 "base/rtpdump.h", |
67 "base/rtputils.cc", | 66 "base/rtputils.cc", |
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136 "engine/webrtcvideocapturerfactory.h", | 135 "engine/webrtcvideocapturerfactory.h", |
137 "engine/webrtcvideodecoderfactory.h", | 136 "engine/webrtcvideodecoderfactory.h", |
138 "engine/webrtcvideoencoderfactory.cc", | 137 "engine/webrtcvideoencoderfactory.cc", |
139 "engine/webrtcvideoencoderfactory.h", | 138 "engine/webrtcvideoencoderfactory.h", |
140 "engine/webrtcvideoengine2.cc", | 139 "engine/webrtcvideoengine2.cc", |
141 "engine/webrtcvideoengine2.h", | 140 "engine/webrtcvideoengine2.h", |
142 "engine/webrtcvideoframe.h", | 141 "engine/webrtcvideoframe.h", |
143 "engine/webrtcvoe.h", | 142 "engine/webrtcvoe.h", |
144 "engine/webrtcvoiceengine.cc", | 143 "engine/webrtcvoiceengine.cc", |
145 "engine/webrtcvoiceengine.h", | 144 "engine/webrtcvoiceengine.h", |
| 145 "sctp/sctptransportinternal.h", |
146 ] | 146 ] |
147 | 147 |
148 if (rtc_enable_sctp) { | 148 if (rtc_enable_sctp) { |
149 sources += [ | 149 sources += [ |
150 "sctp/sctpdataengine.cc", | 150 "sctp/sctptransport.cc", |
151 "sctp/sctpdataengine.h", | 151 "sctp/sctptransport.h", |
152 ] | 152 ] |
153 } | 153 } |
154 | 154 |
155 configs += [ ":rtc_media_warnings_config" ] | 155 configs += [ ":rtc_media_warnings_config" ] |
156 | 156 |
157 if (!build_with_chromium && is_clang) { | 157 if (!build_with_chromium && is_clang) { |
158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). | 158 # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163). |
159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] | 159 suppressed_configs += [ "//build/config/clang:find_bad_constructs" ] |
160 } | 160 } |
161 | 161 |
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339 "engine/simulcast_unittest.cc", | 339 "engine/simulcast_unittest.cc", |
340 "engine/videodecodersoftwarefallbackwrapper_unittest.cc", | 340 "engine/videodecodersoftwarefallbackwrapper_unittest.cc", |
341 "engine/videoencodersoftwarefallbackwrapper_unittest.cc", | 341 "engine/videoencodersoftwarefallbackwrapper_unittest.cc", |
342 "engine/webrtcmediaengine_unittest.cc", | 342 "engine/webrtcmediaengine_unittest.cc", |
343 "engine/webrtcvideocapturer_unittest.cc", | 343 "engine/webrtcvideocapturer_unittest.cc", |
344 "engine/webrtcvideoengine2_unittest.cc", | 344 "engine/webrtcvideoengine2_unittest.cc", |
345 "engine/webrtcvoiceengine_unittest.cc", | 345 "engine/webrtcvoiceengine_unittest.cc", |
346 ] | 346 ] |
347 | 347 |
348 if (rtc_enable_sctp) { | 348 if (rtc_enable_sctp) { |
349 sources += [ "sctp/sctpdataengine_unittest.cc" ] | 349 sources += [ "sctp/sctptransport_unittest.cc" ] |
350 } | 350 } |
351 | 351 |
352 configs += [ ":rtc_media_unittests_config" ] | 352 configs += [ ":rtc_media_unittests_config" ] |
353 | 353 |
354 if (rtc_use_h264) { | 354 if (rtc_use_h264) { |
355 defines += [ "WEBRTC_USE_H264" ] | 355 defines += [ "WEBRTC_USE_H264" ] |
356 } | 356 } |
357 if (is_win) { | 357 if (is_win) { |
358 cflags = [ | 358 cflags = [ |
359 "/wd4245", # conversion from int to size_t, signed/unsigned mismatch. | 359 "/wd4245", # conversion from int to size_t, signed/unsigned mismatch. |
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386 # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243. | 386 # TODO(kjellander): Move as part of work in bugs.webrtc.org/4243. |
387 ":rtc_media", | 387 ":rtc_media", |
388 ":rtc_unittest_main", | 388 ":rtc_unittest_main", |
389 "../audio", | 389 "../audio", |
390 "../base:rtc_base_tests_utils", | 390 "../base:rtc_base_tests_utils", |
391 "../modules/audio_device:mock_audio_device", | 391 "../modules/audio_device:mock_audio_device", |
392 "../system_wrappers:metrics_default", | 392 "../system_wrappers:metrics_default", |
393 ] | 393 ] |
394 } | 394 } |
395 } | 395 } |
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