Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1)

Side by Side Diff: talk/libjingle.gyp

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated location for peerconnection_unittests.isolate Created 4 years, 11 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 # 1 #
2 # libjingle 2 # libjingle
3 # Copyright 2012 Google Inc. 3 # Copyright 2012 Google Inc.
4 # 4 #
5 # Redistribution and use in source and binary forms, with or without 5 # Redistribution and use in source and binary forms, with or without
6 # modification, are permitted provided that the following conditions are met: 6 # modification, are permitted provided that the following conditions are met:
7 # 7 #
8 # 1. Redistributions of source code must retain the above copyright notice, 8 # 1. Redistributions of source code must retain the above copyright notice,
9 # this list of conditions and the following disclaimer. 9 # this list of conditions and the following disclaimer.
10 # 2. Redistributions in binary form must reproduce the above copyright notice, 10 # 2. Redistributions in binary form must reproduce the above copyright notice,
11 # this list of conditions and the following disclaimer in the documentation 11 # this list of conditions and the following disclaimer in the documentation
12 # and/or other materials provided with the distribution. 12 # and/or other materials provided with the distribution.
13 # 3. The name of the author may not be used to endorse or promote products 13 # 3. The name of the author may not be used to endorse or promote products
14 # derived from this software without specific prior written permission. 14 # derived from this software without specific prior written permission.
15 # 15 #
16 # THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED 16 # THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 # WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF 17 # WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 # MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO 18 # MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 # EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, 19 # EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 # SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, 20 # SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 # PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; 21 # PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 # OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, 22 # OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 # WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 23 # WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 # OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF 24 # OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 # ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 # ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 26
27 { 27 {
28 'includes': ['build/common.gypi'], 28 'includes': ['build/common.gypi'],
29 'conditions': [ 29 'conditions': [
30 ['os_posix == 1 and OS != "mac" and OS != "ios"', {
31 'conditions': [
32 ['sysroot!=""', {
33 'variables': {
34 'pkg-config': '../../../build/linux/pkg-config-wrapper "<(sysroot)" " <(target_arch)"',
35 },
36 }, {
37 'variables': {
38 'pkg-config': 'pkg-config'
39 },
40 }],
41 ],
42 }],
43 ['OS=="linux" or OS=="android"', {
44 'targets': [
45 {
46 'target_name': 'libjingle_peerconnection_jni',
47 'type': 'static_library',
48 'dependencies': [
49 '<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_defa ult',
50 'libjingle_peerconnection',
51 ],
52 'sources': [
53 'app/webrtc/java/jni/classreferenceholder.cc',
54 'app/webrtc/java/jni/classreferenceholder.h',
55 'app/webrtc/java/jni/jni_helpers.cc',
56 'app/webrtc/java/jni/jni_helpers.h',
57 'app/webrtc/java/jni/native_handle_impl.cc',
58 'app/webrtc/java/jni/native_handle_impl.h',
59 'app/webrtc/java/jni/peerconnection_jni.cc',
60 ],
61 'include_dirs': [
62 '<(libyuv_dir)/include',
63 ],
64 'conditions': [
65 ['OS=="linux"', {
66 'include_dirs': [
67 '<(java_home)/include',
68 '<(java_home)/include/linux',
69 ],
70 }],
71 ['build_json==1', {
72 'dependencies': [
73 '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
74 ],
75 'export_dependent_settings': [
76 '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
77 ],
78 }],
79 ['OS=="android"', {
80 'sources': [
81 'app/webrtc/androidvideocapturer.cc',
82 'app/webrtc/androidvideocapturer.h',
83 'app/webrtc/java/jni/androidmediacodeccommon.h',
84 'app/webrtc/java/jni/androidmediadecoder_jni.cc',
85 'app/webrtc/java/jni/androidmediadecoder_jni.h',
86 'app/webrtc/java/jni/androidmediaencoder_jni.cc',
87 'app/webrtc/java/jni/androidmediaencoder_jni.h',
88 'app/webrtc/java/jni/androidnetworkmonitor_jni.cc',
89 'app/webrtc/java/jni/androidnetworkmonitor_jni.h',
90 'app/webrtc/java/jni/androidvideocapturer_jni.cc',
91 'app/webrtc/java/jni/androidvideocapturer_jni.h',
92 'app/webrtc/java/jni/surfacetexturehelper_jni.cc',
93 'app/webrtc/java/jni/surfacetexturehelper_jni.h',
94 ]
95 }],
96 ],
97 },
98 {
99 'target_name': 'libjingle_peerconnection_so',
100 'type': 'shared_library',
101 'dependencies': [
102 'libjingle_peerconnection',
103 'libjingle_peerconnection_jni',
104 ],
105 'sources': [
106 'app/webrtc/java/jni/jni_onload.cc',
107 ],
108 'variables': {
109 # This library uses native JNI exports; tell GYP so that the
110 # required symbols will be kept.
111 'use_native_jni_exports': 1,
112 },
113 'conditions': [
114 ['OS=="linux"', {
115 'defines': [
116 'HAVE_GTK',
117 ],
118 'include_dirs': [
119 '<(java_home)/include',
120 '<(java_home)/include/linux',
121 ],
122 'conditions': [
123 ['use_gtk==1', {
124 'link_settings': {
125 'libraries': [
126 '<!@(pkg-config --libs-only-l gobject-2.0 gthread-2.0'
127 ' gtk+-2.0)',
128 ],
129 },
130 }],
131 ],
132 }],
133 ],
134 },
135 {
136 'target_name': 'libjingle_peerconnection_jar',
137 'type': 'none',
138 'actions': [
139 {
140 # TODO(jiayl): extract peerconnection_java_files and android_java_ files into a webrtc
141 # gyp var that can be included here, or better yet, build a proper .jar in webrtc
142 # and include it here.
143 'variables': {
144 'java_src_dir': 'app/webrtc/java/src',
145 'webrtc_base_dir': '<(webrtc_root)/base',
146 'webrtc_modules_dir': '<(webrtc_root)/modules',
147 'build_jar_log': '<(INTERMEDIATE_DIR)/build_jar.log',
148 'peerconnection_java_files': [
149 'app/webrtc/java/src/org/webrtc/AudioSource.java',
150 'app/webrtc/java/src/org/webrtc/AudioTrack.java',
151 'app/webrtc/java/src/org/webrtc/CallSessionFileRotatingLogSink .java',
152 'app/webrtc/java/src/org/webrtc/DataChannel.java',
153 'app/webrtc/java/src/org/webrtc/IceCandidate.java',
154 'app/webrtc/java/src/org/webrtc/MediaConstraints.java',
155 'app/webrtc/java/src/org/webrtc/MediaSource.java',
156 'app/webrtc/java/src/org/webrtc/MediaStream.java',
157 'app/webrtc/java/src/org/webrtc/MediaStreamTrack.java',
158 'app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java',
159 'app/webrtc/java/src/org/webrtc/PeerConnection.java',
160 'app/webrtc/java/src/org/webrtc/RtpReceiver.java',
161 'app/webrtc/java/src/org/webrtc/RtpSender.java',
162 'app/webrtc/java/src/org/webrtc/SdpObserver.java',
163 'app/webrtc/java/src/org/webrtc/StatsObserver.java',
164 'app/webrtc/java/src/org/webrtc/StatsReport.java',
165 'app/webrtc/java/src/org/webrtc/SessionDescription.java',
166 'app/webrtc/java/src/org/webrtc/VideoCapturer.java',
167 'app/webrtc/java/src/org/webrtc/VideoRenderer.java',
168 'app/webrtc/java/src/org/webrtc/VideoSource.java',
169 'app/webrtc/java/src/org/webrtc/VideoTrack.java',
170 '<(webrtc_base_dir)/java/src/org/webrtc/Logging.java',
171 ],
172 'android_java_files': [
173 'app/webrtc/java/android/org/webrtc/Camera2Enumerator.java',
174 'app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.j ava',
175 'app/webrtc/java/android/org/webrtc/CameraEnumerator.java',
176 'app/webrtc/java/android/org/webrtc/EglBase.java',
177 'app/webrtc/java/android/org/webrtc/EglBase10.java',
178 'app/webrtc/java/android/org/webrtc/EglBase14.java',
179 'app/webrtc/java/android/org/webrtc/GlRectDrawer.java',
180 'app/webrtc/java/android/org/webrtc/GlShader.java',
181 'app/webrtc/java/android/org/webrtc/GlUtil.java',
182 'app/webrtc/java/android/org/webrtc/GlTextureFrameBuffer.java' ,
183 'app/webrtc/java/android/org/webrtc/NetworkMonitor.java',
184 'app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.j ava',
185 'app/webrtc/java/android/org/webrtc/RendererCommon.java',
186 'app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java' ,
187 'app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java',
188 'app/webrtc/java/android/org/webrtc/ThreadUtils.java',
189 'app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java' ,
190 'app/webrtc/java/android/org/webrtc/VideoRendererGui.java',
191 'app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java',
192 'app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java',
193 '<(webrtc_modules_dir)/video_render/android/java/src/org/webrt c/videoengine/ViEAndroidGLES20.java',
194 '<(webrtc_modules_dir)/video_render/android/java/src/org/webrt c/videoengine/ViERenderer.java',
195 '<(webrtc_modules_dir)/video_render/android/java/src/org/webrt c/videoengine/ViESurfaceRenderer.java',
196 '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrt c/voiceengine/BuildInfo.java',
197 '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrt c/voiceengine/WebRtcAudioEffects.java',
198 '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrt c/voiceengine/WebRtcAudioManager.java',
199 '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrt c/voiceengine/WebRtcAudioUtils.java',
200 '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrt c/voiceengine/WebRtcAudioRecord.java',
201 '<(webrtc_modules_dir)/audio_device/android/java/src/org/webrt c/voiceengine/WebRtcAudioTrack.java',
202 ],
203 },
204 'action_name': 'create_jar',
205 'inputs': [
206 'build/build_jar.sh',
207 '<@(java_files)',
208 ],
209 'outputs': [
210 '<(PRODUCT_DIR)/libjingle_peerconnection.jar',
211 ],
212 'conditions': [
213 ['OS=="android"', {
214 'variables': {
215 'java_files': ['<@(peerconnection_java_files)', '<@(android_ java_files)'],
216 'build_classpath': '<(java_src_dir):<(DEPTH)/third_party/and roid_tools/sdk/platforms/android-<(android_sdk_version)/android.jar',
217 },
218 }, {
219 'variables': {
220 'java_files': ['<@(peerconnection_java_files)'],
221 'build_classpath': '<(java_src_dir)',
222 },
223 }],
224 ],
225 'action': [
226 'bash', '-ec',
227 'mkdir -p <(INTERMEDIATE_DIR) && '
228 '{ build/build_jar.sh <(java_home) <@(_outputs) '
229 ' <(INTERMEDIATE_DIR)/build_jar.tmp '
230 ' <(build_classpath) <@(java_files) '
231 ' > <(build_jar_log) 2>&1 || '
232 ' { cat <(build_jar_log) ; exit 1; } }'
233 ],
234 },
235 ],
236 'dependencies': [
237 'libjingle_peerconnection_so',
238 ],
239 },
240 ],
241 }],
242 ['OS=="android"', {
243 'targets': [
244 {
245 # |libjingle_peerconnection_java| builds a jar file with name
246 # libjingle_peerconnection_java.jar using Chromes build system.
247 # It includes all Java files needed to setup a PeeerConnection call
248 # from Android.
249 # TODO(perkj): Consider replacing the use of
250 # libjingle_peerconnection_jar with this target everywhere.
251 'target_name': 'libjingle_peerconnection_java',
252 'type': 'none',
253 'dependencies': [
254 'libjingle_peerconnection_so',
255 ],
256 'variables': {
257 # Designate as Chromium code and point to our lint settings to
258 # enable linting of the WebRTC code (this is the only way to make
259 # lint_action invoke the Android linter).
260 'android_manifest_path': '<(webrtc_root)/build/android/AndroidManife st.xml',
261 'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml' ,
262 'chromium_code': 1,
263 'java_in_dir': 'app/webrtc/java',
264 'webrtc_base_dir': '<(webrtc_root)/base',
265 'webrtc_modules_dir': '<(webrtc_root)/modules',
266 'additional_src_dirs' : [
267 'app/webrtc/java/android',
268 '<(webrtc_base_dir)/java/src',
269 '<(webrtc_modules_dir)/audio_device/android/java/src',
270 '<(webrtc_modules_dir)/video_render/android/java/src',
271 ],
272 },
273 'includes': ['../build/java.gypi'],
274 }, # libjingle_peerconnection_java
275 ]
276 }],
277 ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', { 30 ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
278 # The >= 10.7 above is required for ARC. 31 # The >= 10.7 above is required for ARC.
pthatcher1 2016/02/05 21:18:49 Shouldn't we (re)move this target also? I thought
kjellander_webrtc 2016/02/08 09:12:06 No, Zeke wanted to let the ObjC files stay, since
279 'targets': [ 32 'targets': [
280 { 33 {
281 'target_name': 'libjingle_peerconnection_objc', 34 'target_name': 'libjingle_peerconnection_objc',
282 'type': 'static_library', 35 'type': 'static_library',
283 'dependencies': [ 36 'dependencies': [
284 'libjingle_peerconnection', 37 '<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
285 ], 38 ],
286 'sources': [ 39 'sources': [
287 'app/webrtc/objc/RTCAudioTrack+Internal.h', 40 'app/webrtc/objc/RTCAudioTrack+Internal.h',
288 'app/webrtc/objc/RTCAudioTrack.mm', 41 'app/webrtc/objc/RTCAudioTrack.mm',
289 'app/webrtc/objc/RTCDataChannel+Internal.h', 42 'app/webrtc/objc/RTCDataChannel+Internal.h',
290 'app/webrtc/objc/RTCDataChannel.mm', 43 'app/webrtc/objc/RTCDataChannel.mm',
291 'app/webrtc/objc/RTCEnumConverter.h', 44 'app/webrtc/objc/RTCEnumConverter.h',
292 'app/webrtc/objc/RTCEnumConverter.mm', 45 'app/webrtc/objc/RTCEnumConverter.mm',
293 'app/webrtc/objc/RTCFileLogger.mm', 46 'app/webrtc/objc/RTCFileLogger.mm',
294 'app/webrtc/objc/RTCI420Frame+Internal.h', 47 'app/webrtc/objc/RTCI420Frame+Internal.h',
(...skipping 61 matching lines...) Expand 10 before | Expand all | Expand 10 after
356 'app/webrtc/objc/public/RTCVideoRenderer.h', 109 'app/webrtc/objc/public/RTCVideoRenderer.h',
357 'app/webrtc/objc/public/RTCVideoSource.h', 110 'app/webrtc/objc/public/RTCVideoSource.h',
358 'app/webrtc/objc/public/RTCVideoTrack.h', 111 'app/webrtc/objc/public/RTCVideoTrack.h',
359 ], 112 ],
360 'direct_dependent_settings': { 113 'direct_dependent_settings': {
361 'include_dirs': [ 114 'include_dirs': [
362 '<(DEPTH)/talk/app/webrtc/objc/public', 115 '<(DEPTH)/talk/app/webrtc/objc/public',
363 ], 116 ],
364 }, 117 },
365 'include_dirs': [ 118 'include_dirs': [
366 '<(DEPTH)/talk/app/webrtc', 119 '<(webrtc_root)/webrtc/api',
367 '<(DEPTH)/talk/app/webrtc/objc', 120 '<(DEPTH)/talk/app/webrtc/objc',
368 '<(DEPTH)/talk/app/webrtc/objc/public', 121 '<(DEPTH)/talk/app/webrtc/objc/public',
369 ], 122 ],
370 'link_settings': { 123 'link_settings': {
371 'libraries': [ 124 'libraries': [
372 '-lstdc++', 125 '-lstdc++',
373 ], 126 ],
374 }, 127 },
375 'all_dependent_settings': { 128 'all_dependent_settings': {
376 'xcode_settings': { 129 'xcode_settings': {
(...skipping 51 matching lines...) Expand 10 before | Expand all | Expand 10 after
428 '-framework Cocoa', 181 '-framework Cocoa',
429 ], 182 ],
430 }, 183 },
431 }, 184 },
432 }], 185 }],
433 ], 186 ],
434 }, # target libjingle_peerconnection_objc 187 }, # target libjingle_peerconnection_objc
435 ], 188 ],
436 }], 189 }],
437 ], 190 ],
438
439 'targets': [ 191 'targets': [
440 { 192 {
441 'target_name': 'libjingle', 193 'target_name': 'libjingle',
442 'type': 'none', 194 'type': 'none',
443 'dependencies': [ 195 'dependencies': [
444 '<(webrtc_root)/base/base.gyp:rtc_base', 196 '<(webrtc_root)/base/base.gyp:rtc_base',
445 ], 197 ],
446 'conditions': [ 198 'conditions': [
447 ['build_json==1', { 199 ['build_json==1', {
448 'dependencies': [ 200 'dependencies': [
(...skipping 292 matching lines...) Expand 10 before | Expand all | Expand 10 after
741 'session/media/mediasession.cc', 493 'session/media/mediasession.cc',
742 'session/media/mediasession.h', 494 'session/media/mediasession.h',
743 'session/media/mediasink.h', 495 'session/media/mediasink.h',
744 'session/media/rtcpmuxfilter.cc', 496 'session/media/rtcpmuxfilter.cc',
745 'session/media/rtcpmuxfilter.h', 497 'session/media/rtcpmuxfilter.h',
746 'session/media/srtpfilter.cc', 498 'session/media/srtpfilter.cc',
747 'session/media/srtpfilter.h', 499 'session/media/srtpfilter.h',
748 'session/media/voicechannel.h', 500 'session/media/voicechannel.h',
749 ], 501 ],
750 }, # target libjingle_p2p 502 }, # target libjingle_p2p
751 {
752 'target_name': 'libjingle_peerconnection',
753 'type': 'static_library',
754 'dependencies': [
755 'libjingle',
756 'libjingle_media',
757 'libjingle_p2p',
758 ],
759 'sources': [
760 'app/webrtc/audiotrack.cc',
761 'app/webrtc/audiotrack.h',
762 'app/webrtc/datachannel.cc',
763 'app/webrtc/datachannel.h',
764 'app/webrtc/datachannelinterface.h',
765 'app/webrtc/dtlsidentitystore.cc',
766 'app/webrtc/dtlsidentitystore.h',
767 'app/webrtc/dtmfsender.cc',
768 'app/webrtc/dtmfsender.h',
769 'app/webrtc/dtmfsenderinterface.h',
770 'app/webrtc/jsep.h',
771 'app/webrtc/jsepicecandidate.cc',
772 'app/webrtc/jsepicecandidate.h',
773 'app/webrtc/jsepsessiondescription.cc',
774 'app/webrtc/jsepsessiondescription.h',
775 'app/webrtc/localaudiosource.cc',
776 'app/webrtc/localaudiosource.h',
777 'app/webrtc/mediaconstraintsinterface.cc',
778 'app/webrtc/mediaconstraintsinterface.h',
779 'app/webrtc/mediacontroller.cc',
780 'app/webrtc/mediacontroller.h',
781 'app/webrtc/mediastream.cc',
782 'app/webrtc/mediastream.h',
783 'app/webrtc/mediastreaminterface.h',
784 'app/webrtc/mediastreamobserver.cc',
785 'app/webrtc/mediastreamobserver.h',
786 'app/webrtc/mediastreamprovider.h',
787 'app/webrtc/mediastreamproxy.h',
788 'app/webrtc/mediastreamtrack.h',
789 'app/webrtc/mediastreamtrackproxy.h',
790 'app/webrtc/notifier.h',
791 'app/webrtc/peerconnection.cc',
792 'app/webrtc/peerconnection.h',
793 'app/webrtc/peerconnectionfactory.cc',
794 'app/webrtc/peerconnectionfactory.h',
795 'app/webrtc/peerconnectionfactoryproxy.h',
796 'app/webrtc/peerconnectioninterface.h',
797 'app/webrtc/peerconnectionproxy.h',
798 'app/webrtc/proxy.h',
799 'app/webrtc/remoteaudiosource.cc',
800 'app/webrtc/remoteaudiosource.h',
801 'app/webrtc/remotevideocapturer.cc',
802 'app/webrtc/remotevideocapturer.h',
803 'app/webrtc/rtpreceiver.cc',
804 'app/webrtc/rtpreceiver.h',
805 'app/webrtc/rtpreceiverinterface.h',
806 'app/webrtc/rtpsender.cc',
807 'app/webrtc/rtpsender.h',
808 'app/webrtc/rtpsenderinterface.h',
809 'app/webrtc/sctputils.cc',
810 'app/webrtc/sctputils.h',
811 'app/webrtc/statscollector.cc',
812 'app/webrtc/statscollector.h',
813 'app/webrtc/statstypes.cc',
814 'app/webrtc/statstypes.h',
815 'app/webrtc/streamcollection.h',
816 'app/webrtc/videosource.cc',
817 'app/webrtc/videosource.h',
818 'app/webrtc/videosourceinterface.h',
819 'app/webrtc/videosourceproxy.h',
820 'app/webrtc/videotrack.cc',
821 'app/webrtc/videotrack.h',
822 'app/webrtc/videotrackrenderers.cc',
823 'app/webrtc/videotrackrenderers.h',
824 'app/webrtc/webrtcsdp.cc',
825 'app/webrtc/webrtcsdp.h',
826 'app/webrtc/webrtcsession.cc',
827 'app/webrtc/webrtcsession.h',
828 'app/webrtc/webrtcsessiondescriptionfactory.cc',
829 'app/webrtc/webrtcsessiondescriptionfactory.h',
830 ],
831 }, # target libjingle_peerconnection
832 ], 503 ],
833 } 504 }
OLDNEW
« no previous file with comments | « talk/build/common.gypi ('k') | talk/libjingle_tests.gyp » ('j') | webrtc/api/OWNERS » ('J')

Powered by Google App Engine
This is Rietveld 408576698