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Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago
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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=="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/androidvideocapturer.cc',
54 'app/webrtc/androidvideocapturer.h',
55 'app/webrtc/java/jni/androidmediacodeccommon.h',
56 'app/webrtc/java/jni/androidmediadecoder_jni.cc',
57 'app/webrtc/java/jni/androidmediadecoder_jni.h',
58 'app/webrtc/java/jni/androidmediaencoder_jni.cc',
59 'app/webrtc/java/jni/androidmediaencoder_jni.h',
60 'app/webrtc/java/jni/androidnetworkmonitor_jni.cc',
61 'app/webrtc/java/jni/androidnetworkmonitor_jni.h',
62 'app/webrtc/java/jni/androidvideocapturer_jni.cc',
63 'app/webrtc/java/jni/androidvideocapturer_jni.h',
64 'app/webrtc/java/jni/classreferenceholder.cc',
65 'app/webrtc/java/jni/classreferenceholder.h',
66 'app/webrtc/java/jni/eglbase_jni.cc',
67 'app/webrtc/java/jni/eglbase_jni.h',
68 'app/webrtc/java/jni/jni_helpers.cc',
69 'app/webrtc/java/jni/jni_helpers.h',
70 'app/webrtc/java/jni/native_handle_impl.cc',
71 'app/webrtc/java/jni/native_handle_impl.h',
72 'app/webrtc/java/jni/peerconnection_jni.cc',
73 'app/webrtc/java/jni/surfacetexturehelper_jni.cc',
74 'app/webrtc/java/jni/surfacetexturehelper_jni.h',
75 ],
76 'include_dirs': [
77 '<(libyuv_dir)/include',
78 ],
79 'conditions': [
80 ['build_json==1', {
81 'dependencies': [
82 '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
83 ],
84 'export_dependent_settings': [
85 '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
86 ],
87 }],
88 ],
89 },
90 {
91 'target_name': 'libjingle_peerconnection_so',
92 'type': 'shared_library',
93 'dependencies': [
94 'libjingle_peerconnection',
95 'libjingle_peerconnection_jni',
96 ],
97 'sources': [
98 'app/webrtc/java/jni/jni_onload.cc',
99 ],
100 'variables': {
101 # This library uses native JNI exports; tell GYP so that the
102 # required symbols will be kept.
103 'use_native_jni_exports': 1,
104 },
105 },
106 {
107 # |libjingle_peerconnection_java| builds a jar file with name
108 # libjingle_peerconnection_java.jar using Chromes build system.
109 # It includes all Java files needed to setup a PeeerConnection call
110 # from Android.
111 'target_name': 'libjingle_peerconnection_java',
112 'type': 'none',
113 'dependencies': [
114 'libjingle_peerconnection_so',
115 ],
116 'variables': {
117 # Designate as Chromium code and point to our lint settings to
118 # enable linting of the WebRTC code (this is the only way to make
119 # lint_action invoke the Android linter).
120 'android_manifest_path': '<(webrtc_root)/build/android/AndroidManife st.xml',
121 'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml' ,
122 'chromium_code': 1,
123 'java_in_dir': 'app/webrtc/java',
124 'webrtc_base_dir': '<(webrtc_root)/base',
125 'webrtc_modules_dir': '<(webrtc_root)/modules',
126 'additional_src_dirs' : [
127 'app/webrtc/java/android',
128 '<(webrtc_base_dir)/java/src',
129 '<(webrtc_modules_dir)/audio_device/android/java/src',
130 '<(webrtc_modules_dir)/video_render/android/java/src',
131 ],
132 },
133 'includes': ['../build/java.gypi'],
134 }, # libjingle_peerconnection_java
135 ]
136 }],
137 ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', { 30 ['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
138 # The >= 10.7 above is required for ARC. 31 # The >= 10.7 above is required for ARC.
139 'targets': [ 32 'targets': [
140 { 33 {
141 'target_name': 'libjingle_peerconnection_objc', 34 'target_name': 'libjingle_peerconnection_objc',
142 'type': 'static_library', 35 'type': 'static_library',
143 'dependencies': [ 36 'dependencies': [
144 'libjingle_peerconnection', 37 '<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
145 ], 38 ],
146 'sources': [ 39 'sources': [
147 'app/webrtc/objc/RTCAudioTrack+Internal.h', 40 'app/webrtc/objc/RTCAudioTrack+Internal.h',
148 'app/webrtc/objc/RTCAudioTrack.mm', 41 'app/webrtc/objc/RTCAudioTrack.mm',
149 'app/webrtc/objc/RTCDataChannel+Internal.h', 42 'app/webrtc/objc/RTCDataChannel+Internal.h',
150 'app/webrtc/objc/RTCDataChannel.mm', 43 'app/webrtc/objc/RTCDataChannel.mm',
151 'app/webrtc/objc/RTCEnumConverter.h', 44 'app/webrtc/objc/RTCEnumConverter.h',
152 'app/webrtc/objc/RTCEnumConverter.mm', 45 'app/webrtc/objc/RTCEnumConverter.mm',
153 'app/webrtc/objc/RTCFileLogger.mm', 46 'app/webrtc/objc/RTCFileLogger.mm',
154 '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
216 'app/webrtc/objc/public/RTCVideoRenderer.h', 109 'app/webrtc/objc/public/RTCVideoRenderer.h',
217 'app/webrtc/objc/public/RTCVideoSource.h', 110 'app/webrtc/objc/public/RTCVideoSource.h',
218 'app/webrtc/objc/public/RTCVideoTrack.h', 111 'app/webrtc/objc/public/RTCVideoTrack.h',
219 ], 112 ],
220 'direct_dependent_settings': { 113 'direct_dependent_settings': {
221 'include_dirs': [ 114 'include_dirs': [
222 '<(DEPTH)/talk/app/webrtc/objc/public', 115 '<(DEPTH)/talk/app/webrtc/objc/public',
223 ], 116 ],
224 }, 117 },
225 'include_dirs': [ 118 'include_dirs': [
226 '<(DEPTH)/talk/app/webrtc', 119 '<(webrtc_root)/webrtc/api',
227 '<(DEPTH)/talk/app/webrtc/objc', 120 '<(DEPTH)/talk/app/webrtc/objc',
228 '<(DEPTH)/talk/app/webrtc/objc/public', 121 '<(DEPTH)/talk/app/webrtc/objc/public',
229 ], 122 ],
230 'link_settings': { 123 'link_settings': {
231 'libraries': [ 124 'libraries': [
232 '-lstdc++', 125 '-lstdc++',
233 ], 126 ],
234 }, 127 },
235 'all_dependent_settings': { 128 'all_dependent_settings': {
236 'xcode_settings': { 129 'xcode_settings': {
(...skipping 51 matching lines...) Expand 10 before | Expand all | Expand 10 after
288 '-framework Cocoa', 181 '-framework Cocoa',
289 ], 182 ],
290 }, 183 },
291 }, 184 },
292 }], 185 }],
293 ], 186 ],
294 }, # target libjingle_peerconnection_objc 187 }, # target libjingle_peerconnection_objc
295 ], 188 ],
296 }], 189 }],
297 ], 190 ],
298
299 'targets': [ 191 'targets': [
300 { 192 {
301 'target_name': 'libjingle_p2p', 193 'target_name': 'libjingle_p2p',
302 'type': 'static_library', 194 'type': 'static_library',
303 'dependencies': [ 195 'dependencies': [
304 '<(webrtc_root)/base/base.gyp:rtc_base', 196 '<(webrtc_root)/base/base.gyp:rtc_base',
305 '<(webrtc_root)/media/media.gyp:rtc_media', 197 '<(webrtc_root)/media/media.gyp:rtc_media',
306 ], 198 ],
307 'conditions': [ 199 'conditions': [
308 ['build_libsrtp==1', { 200 ['build_libsrtp==1', {
(...skipping 32 matching lines...) Expand 10 before | Expand all | Expand 10 after
341 'session/media/mediasession.cc', 233 'session/media/mediasession.cc',
342 'session/media/mediasession.h', 234 'session/media/mediasession.h',
343 'session/media/mediasink.h', 235 'session/media/mediasink.h',
344 'session/media/rtcpmuxfilter.cc', 236 'session/media/rtcpmuxfilter.cc',
345 'session/media/rtcpmuxfilter.h', 237 'session/media/rtcpmuxfilter.h',
346 'session/media/srtpfilter.cc', 238 'session/media/srtpfilter.cc',
347 'session/media/srtpfilter.h', 239 'session/media/srtpfilter.h',
348 'session/media/voicechannel.h', 240 'session/media/voicechannel.h',
349 ], 241 ],
350 }, # target libjingle_p2p 242 }, # target libjingle_p2p
351 {
352 'target_name': 'libjingle_peerconnection',
353 'type': 'static_library',
354 'dependencies': [
355 '<(webrtc_root)/base/base.gyp:rtc_base',
356 '<(webrtc_root)/media/media.gyp:rtc_media',
357 'libjingle_p2p',
358 ],
359 'sources': [
360 'app/webrtc/audiotrack.cc',
361 'app/webrtc/audiotrack.h',
362 'app/webrtc/datachannel.cc',
363 'app/webrtc/datachannel.h',
364 'app/webrtc/datachannelinterface.h',
365 'app/webrtc/dtlsidentitystore.cc',
366 'app/webrtc/dtlsidentitystore.h',
367 'app/webrtc/dtmfsender.cc',
368 'app/webrtc/dtmfsender.h',
369 'app/webrtc/dtmfsenderinterface.h',
370 'app/webrtc/jsep.h',
371 'app/webrtc/jsepicecandidate.cc',
372 'app/webrtc/jsepicecandidate.h',
373 'app/webrtc/jsepsessiondescription.cc',
374 'app/webrtc/jsepsessiondescription.h',
375 'app/webrtc/localaudiosource.cc',
376 'app/webrtc/localaudiosource.h',
377 'app/webrtc/mediaconstraintsinterface.cc',
378 'app/webrtc/mediaconstraintsinterface.h',
379 'app/webrtc/mediacontroller.cc',
380 'app/webrtc/mediacontroller.h',
381 'app/webrtc/mediastream.cc',
382 'app/webrtc/mediastream.h',
383 'app/webrtc/mediastreaminterface.h',
384 'app/webrtc/mediastreamobserver.cc',
385 'app/webrtc/mediastreamobserver.h',
386 'app/webrtc/mediastreamprovider.h',
387 'app/webrtc/mediastreamproxy.h',
388 'app/webrtc/mediastreamtrack.h',
389 'app/webrtc/mediastreamtrackproxy.h',
390 'app/webrtc/notifier.h',
391 'app/webrtc/peerconnection.cc',
392 'app/webrtc/peerconnection.h',
393 'app/webrtc/peerconnectionfactory.cc',
394 'app/webrtc/peerconnectionfactory.h',
395 'app/webrtc/peerconnectionfactoryproxy.h',
396 'app/webrtc/peerconnectioninterface.h',
397 'app/webrtc/peerconnectionproxy.h',
398 'app/webrtc/proxy.h',
399 'app/webrtc/remoteaudiosource.cc',
400 'app/webrtc/remoteaudiosource.h',
401 'app/webrtc/remotevideocapturer.cc',
402 'app/webrtc/remotevideocapturer.h',
403 'app/webrtc/rtpreceiver.cc',
404 'app/webrtc/rtpreceiver.h',
405 'app/webrtc/rtpreceiverinterface.h',
406 'app/webrtc/rtpsender.cc',
407 'app/webrtc/rtpsender.h',
408 'app/webrtc/rtpsenderinterface.h',
409 'app/webrtc/sctputils.cc',
410 'app/webrtc/sctputils.h',
411 'app/webrtc/statscollector.cc',
412 'app/webrtc/statscollector.h',
413 'app/webrtc/statstypes.cc',
414 'app/webrtc/statstypes.h',
415 'app/webrtc/streamcollection.h',
416 'app/webrtc/videosource.cc',
417 'app/webrtc/videosource.h',
418 'app/webrtc/videosourceinterface.h',
419 'app/webrtc/videosourceproxy.h',
420 'app/webrtc/videotrack.cc',
421 'app/webrtc/videotrack.h',
422 'app/webrtc/videotrackrenderers.cc',
423 'app/webrtc/videotrackrenderers.h',
424 'app/webrtc/webrtcsdp.cc',
425 'app/webrtc/webrtcsdp.h',
426 'app/webrtc/webrtcsession.cc',
427 'app/webrtc/webrtcsession.h',
428 'app/webrtc/webrtcsessiondescriptionfactory.cc',
429 'app/webrtc/webrtcsessiondescriptionfactory.h',
430 ],
431 }, # target libjingle_peerconnection
432 ], 243 ],
433 } 244 }
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