269fb4bc90
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder. BUG=3379 Review URL: https://webrtc-codereview.appspot.com/26999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
676 lines
27 KiB
Python
Executable File
676 lines
27 KiB
Python
Executable File
#
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# libjingle
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# Copyright 2012, Google Inc.
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#
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# Redistribution and use in source and binary forms, with or without
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# modification, are permitted provided that the following conditions are met:
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#
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# 1. Redistributions of source code must retain the above copyright notice,
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# this list of conditions and the following disclaimer.
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# 2. Redistributions in binary form must reproduce the above copyright notice,
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# this list of conditions and the following disclaimer in the documentation
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# and/or other materials provided with the distribution.
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# 3. The name of the author may not be used to endorse or promote products
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# derived from this software without specific prior written permission.
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#
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# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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#
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{
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'includes': ['build/common.gypi'],
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'conditions': [
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['os_posix == 1 and OS != "mac" and OS != "ios"', {
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'conditions': [
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['sysroot!=""', {
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'variables': {
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'pkg-config': '../../../build/linux/pkg-config-wrapper "<(sysroot)" "<(target_arch)"',
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},
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}, {
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'variables': {
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'pkg-config': 'pkg-config'
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},
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}],
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],
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}],
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['OS=="linux" or OS=="android"', {
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'targets': [
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{
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'target_name': 'libjingle_peerconnection_so',
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'type': 'loadable_module',
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'dependencies': [
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'libjingle_peerconnection',
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'<(DEPTH)/third_party/icu/icu.gyp:icuuc',
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],
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'sources': [
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'app/webrtc/java/jni/peerconnection_jni.cc'
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],
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'include_dirs': [
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'<(DEPTH)/third_party/libyuv/include',
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],
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'conditions': [
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['OS=="linux"', {
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'defines': [
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'HAVE_GTK',
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],
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'include_dirs': [
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'<(java_home)/include',
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'<(java_home)/include/linux',
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],
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'link_settings': {
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'libraries': [
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'<!@(pkg-config --libs-only-l gobject-2.0 gthread-2.0'
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' gtk+-2.0)',
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],
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},
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}],
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],
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},
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{
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'target_name': 'libjingle_peerconnection_jar',
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'type': 'none',
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'actions': [
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{
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'variables': {
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'java_src_dir': 'app/webrtc/java/src',
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'webrtc_modules_dir': '<(webrtc_root)/modules',
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'build_jar_log': '<(INTERMEDIATE_DIR)/build_jar.log',
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'peerconnection_java_files': [
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'app/webrtc/java/src/org/webrtc/AudioSource.java',
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'app/webrtc/java/src/org/webrtc/AudioTrack.java',
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'app/webrtc/java/src/org/webrtc/DataChannel.java',
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'app/webrtc/java/src/org/webrtc/IceCandidate.java',
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'app/webrtc/java/src/org/webrtc/Logging.java',
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'app/webrtc/java/src/org/webrtc/MediaConstraints.java',
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'app/webrtc/java/src/org/webrtc/MediaSource.java',
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'app/webrtc/java/src/org/webrtc/MediaStream.java',
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'app/webrtc/java/src/org/webrtc/MediaStreamTrack.java',
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'app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java',
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'app/webrtc/java/src/org/webrtc/PeerConnection.java',
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'app/webrtc/java/src/org/webrtc/SdpObserver.java',
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'app/webrtc/java/src/org/webrtc/StatsObserver.java',
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'app/webrtc/java/src/org/webrtc/StatsReport.java',
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'app/webrtc/java/src/org/webrtc/SessionDescription.java',
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'app/webrtc/java/src/org/webrtc/VideoCapturer.java',
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'app/webrtc/java/src/org/webrtc/VideoRenderer.java',
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'app/webrtc/java/src/org/webrtc/VideoSource.java',
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'app/webrtc/java/src/org/webrtc/VideoTrack.java',
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],
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# TODO(fischman): extract this into a webrtc gyp var that can be
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# included here, or better yet, build a proper .jar in webrtc
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# and include it here.
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'android_java_files': [
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'app/webrtc/java/android/org/webrtc/VideoRendererGui.java',
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'app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java',
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'app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java',
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'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java',
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'<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java',
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'<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java',
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'<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java',
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'<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java',
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'<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java',
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'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java',
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'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java',
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],
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},
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'action_name': 'create_jar',
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'inputs': [
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'build/build_jar.sh',
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'<@(java_files)',
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],
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'outputs': [
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'<(PRODUCT_DIR)/libjingle_peerconnection.jar',
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],
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'conditions': [
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['OS=="android"', {
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'variables': {
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'java_files': ['<@(peerconnection_java_files)', '<@(android_java_files)'],
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'build_classpath': '<(java_src_dir):<(DEPTH)/third_party/android_tools/sdk/platforms/android-<(android_sdk_version)/android.jar',
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},
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}, {
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'variables': {
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'java_files': ['<@(peerconnection_java_files)'],
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'build_classpath': '<(java_src_dir)',
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},
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}],
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],
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'action': [
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'bash', '-ec',
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'mkdir -p <(INTERMEDIATE_DIR) && '
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'{ build/build_jar.sh <(java_home) <@(_outputs) '
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' <(INTERMEDIATE_DIR)/build_jar.tmp '
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' <(build_classpath) <@(java_files) '
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' > <(build_jar_log) 2>&1 || '
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' { cat <(build_jar_log) ; exit 1; } }'
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],
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},
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],
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'dependencies': [
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'libjingle_peerconnection_so',
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],
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},
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],
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}],
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['OS=="ios" or (OS=="mac" and target_arch!="ia32" and mac_sdk>="10.7")', {
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# The >= 10.7 above is required for ARC.
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'targets': [
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{
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'target_name': 'libjingle_peerconnection_objc',
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'type': 'static_library',
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'dependencies': [
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'libjingle_peerconnection',
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],
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'sources': [
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'app/webrtc/objc/RTCAudioTrack+Internal.h',
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'app/webrtc/objc/RTCAudioTrack.mm',
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'app/webrtc/objc/RTCDataChannel+Internal.h',
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'app/webrtc/objc/RTCDataChannel.mm',
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'app/webrtc/objc/RTCEnumConverter.h',
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'app/webrtc/objc/RTCEnumConverter.mm',
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'app/webrtc/objc/RTCI420Frame+Internal.h',
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'app/webrtc/objc/RTCI420Frame.mm',
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'app/webrtc/objc/RTCICECandidate+Internal.h',
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'app/webrtc/objc/RTCICECandidate.mm',
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'app/webrtc/objc/RTCICEServer+Internal.h',
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'app/webrtc/objc/RTCICEServer.mm',
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'app/webrtc/objc/RTCMediaConstraints+Internal.h',
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'app/webrtc/objc/RTCMediaConstraints.mm',
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'app/webrtc/objc/RTCMediaConstraintsNative.cc',
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'app/webrtc/objc/RTCMediaConstraintsNative.h',
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'app/webrtc/objc/RTCMediaSource+Internal.h',
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'app/webrtc/objc/RTCMediaSource.mm',
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'app/webrtc/objc/RTCMediaStream+Internal.h',
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'app/webrtc/objc/RTCMediaStream.mm',
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'app/webrtc/objc/RTCMediaStreamTrack+Internal.h',
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'app/webrtc/objc/RTCMediaStreamTrack.mm',
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'app/webrtc/objc/RTCOpenGLVideoRenderer.mm',
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'app/webrtc/objc/RTCPair.m',
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'app/webrtc/objc/RTCPeerConnection+Internal.h',
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'app/webrtc/objc/RTCPeerConnection.mm',
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'app/webrtc/objc/RTCPeerConnectionFactory.mm',
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'app/webrtc/objc/RTCPeerConnectionObserver.h',
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'app/webrtc/objc/RTCPeerConnectionObserver.mm',
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'app/webrtc/objc/RTCSessionDescription+Internal.h',
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'app/webrtc/objc/RTCSessionDescription.mm',
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'app/webrtc/objc/RTCStatsReport+Internal.h',
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'app/webrtc/objc/RTCStatsReport.mm',
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'app/webrtc/objc/RTCVideoCapturer+Internal.h',
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'app/webrtc/objc/RTCVideoCapturer.mm',
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'app/webrtc/objc/RTCVideoRenderer+Internal.h',
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'app/webrtc/objc/RTCVideoRenderer.mm',
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'app/webrtc/objc/RTCVideoSource+Internal.h',
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'app/webrtc/objc/RTCVideoSource.mm',
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'app/webrtc/objc/RTCVideoTrack+Internal.h',
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'app/webrtc/objc/RTCVideoTrack.mm',
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'app/webrtc/objc/public/RTCAudioSource.h',
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'app/webrtc/objc/public/RTCAudioTrack.h',
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'app/webrtc/objc/public/RTCDataChannel.h',
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'app/webrtc/objc/public/RTCI420Frame.h',
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'app/webrtc/objc/public/RTCICECandidate.h',
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'app/webrtc/objc/public/RTCICEServer.h',
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'app/webrtc/objc/public/RTCMediaConstraints.h',
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'app/webrtc/objc/public/RTCMediaSource.h',
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'app/webrtc/objc/public/RTCMediaStream.h',
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'app/webrtc/objc/public/RTCMediaStreamTrack.h',
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'app/webrtc/objc/public/RTCOpenGLVideoRenderer.h',
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'app/webrtc/objc/public/RTCPair.h',
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'app/webrtc/objc/public/RTCPeerConnection.h',
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'app/webrtc/objc/public/RTCPeerConnectionDelegate.h',
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'app/webrtc/objc/public/RTCPeerConnectionFactory.h',
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'app/webrtc/objc/public/RTCSessionDescription.h',
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'app/webrtc/objc/public/RTCSessionDescriptionDelegate.h',
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'app/webrtc/objc/public/RTCStatsDelegate.h',
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'app/webrtc/objc/public/RTCStatsReport.h',
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'app/webrtc/objc/public/RTCTypes.h',
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'app/webrtc/objc/public/RTCVideoCapturer.h',
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'app/webrtc/objc/public/RTCVideoRenderer.h',
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'app/webrtc/objc/public/RTCVideoSource.h',
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'app/webrtc/objc/public/RTCVideoTrack.h',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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'<(DEPTH)/talk/app/webrtc/objc/public',
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],
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},
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'include_dirs': [
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'<(DEPTH)/talk/app/webrtc',
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'<(DEPTH)/talk/app/webrtc/objc',
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'<(DEPTH)/talk/app/webrtc/objc/public',
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],
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'link_settings': {
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'libraries': [
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'-lstdc++',
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],
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},
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'xcode_settings': {
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'CLANG_ENABLE_OBJC_ARC': 'YES',
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# common.gypi enables this for mac but we want this to be disabled
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# like it is for ios.
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'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
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},
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'conditions': [
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['OS=="ios"', {
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'sources': [
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'app/webrtc/objc/RTCEAGLVideoView+Internal.h',
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'app/webrtc/objc/RTCEAGLVideoView.m',
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'app/webrtc/objc/public/RTCEAGLVideoView.h',
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],
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'link_settings': {
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'xcode_settings': {
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'OTHER_LDFLAGS': [
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'-framework CoreGraphics',
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'-framework GLKit',
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],
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},
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},
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}],
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['OS=="mac"', {
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'sources': [
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'app/webrtc/objc/RTCNSGLVideoView.m',
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'app/webrtc/objc/public/RTCNSGLVideoView.h',
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],
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'xcode_settings': {
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# Need to build against 10.7 framework for full ARC support
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# on OSX.
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'MACOSX_DEPLOYMENT_TARGET' : '10.7',
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},
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'link_settings': {
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'xcode_settings': {
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'OTHER_LDFLAGS': [
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'-framework Cocoa',
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],
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},
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},
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}],
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],
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}, # target libjingle_peerconnection_objc
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],
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}],
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],
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'targets': [
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{
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'target_name': 'libjingle',
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'type': 'none',
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'dependencies': [
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'<(DEPTH)/third_party/expat/expat.gyp:expat',
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'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
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'<(webrtc_root)/base/base.gyp:rtc_base',
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],
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'export_dependent_settings': [
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'<(DEPTH)/third_party/expat/expat.gyp:expat',
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'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
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],
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}, # target libjingle
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{
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'target_name': 'libjingle_media',
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'type': 'static_library',
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'include_dirs': [
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# TODO(jiayl): move this into the direct_dependent_settings of
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# usrsctp.gyp.
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'<(DEPTH)/third_party/usrsctp',
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],
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'dependencies': [
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'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
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'<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
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'<(webrtc_root)/modules/modules.gyp:video_render_module',
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'<(webrtc_root)/webrtc.gyp:webrtc',
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'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
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'<(webrtc_root)/sound/sound.gyp:rtc_sound',
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
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'<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers_default',
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'<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
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'<(webrtc_root)/libjingle/xmpp/xmpp.gyp:rtc_xmpp',
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'<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',
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'libjingle',
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],
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'direct_dependent_settings': {
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'include_dirs': [
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'<(DEPTH)/third_party/libyuv/include',
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],
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},
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'sources': [
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'media/base/audioframe.h',
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'media/base/audiorenderer.h',
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'media/base/capturemanager.cc',
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'media/base/capturemanager.h',
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'media/base/capturerenderadapter.cc',
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'media/base/capturerenderadapter.h',
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'media/base/codec.cc',
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'media/base/codec.h',
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'media/base/constants.cc',
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'media/base/constants.h',
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'media/base/cpuid.cc',
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'media/base/cpuid.h',
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'media/base/cryptoparams.h',
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'media/base/device.h',
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'media/base/fakescreencapturerfactory.h',
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'media/base/filemediaengine.cc',
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'media/base/filemediaengine.h',
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'media/base/hybriddataengine.h',
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'media/base/mediachannel.h',
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'media/base/mediacommon.h',
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'media/base/mediaengine.cc',
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'media/base/mediaengine.h',
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'media/base/rtpdataengine.cc',
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'media/base/rtpdataengine.h',
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'media/base/rtpdump.cc',
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'media/base/rtpdump.h',
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'media/base/rtputils.cc',
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'media/base/rtputils.h',
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'media/base/screencastid.h',
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'media/base/streamparams.cc',
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'media/base/streamparams.h',
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'media/base/videoadapter.cc',
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'media/base/videoadapter.h',
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'media/base/videocapturer.cc',
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'media/base/videocapturer.h',
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'media/base/videocapturerfactory.h',
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'media/base/videocommon.cc',
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'media/base/videocommon.h',
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'media/base/videoframe.cc',
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'media/base/videoframe.h',
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'media/base/videoframefactory.h',
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'media/base/videoprocessor.h',
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'media/base/videorenderer.h',
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'media/base/voiceprocessor.h',
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'media/base/yuvframegenerator.cc',
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'media/base/yuvframegenerator.h',
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'media/devices/deviceinfo.h',
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'media/devices/devicemanager.cc',
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'media/devices/devicemanager.h',
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'media/devices/dummydevicemanager.h',
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'media/devices/filevideocapturer.cc',
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'media/devices/filevideocapturer.h',
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'media/devices/videorendererfactory.h',
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'media/devices/yuvframescapturer.cc',
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'media/devices/yuvframescapturer.h',
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'media/other/linphonemediaengine.h',
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'media/sctp/sctpdataengine.cc',
|
|
'media/sctp/sctpdataengine.h',
|
|
'media/webrtc/webrtccommon.h',
|
|
'media/webrtc/webrtcexport.h',
|
|
'media/webrtc/webrtcmediaengine.cc',
|
|
'media/webrtc/webrtcmediaengine.h',
|
|
'media/webrtc/webrtcmediaengine.cc',
|
|
'media/webrtc/webrtcpassthroughrender.cc',
|
|
'media/webrtc/webrtcpassthroughrender.h',
|
|
'media/webrtc/webrtctexturevideoframe.cc',
|
|
'media/webrtc/webrtctexturevideoframe.h',
|
|
'media/webrtc/webrtcvideocapturer.cc',
|
|
'media/webrtc/webrtcvideocapturerfactory.h',
|
|
'media/webrtc/webrtcvideocapturerfactory.cc',
|
|
'media/webrtc/webrtcvideocapturer.h',
|
|
'media/webrtc/webrtcvideodecoderfactory.h',
|
|
'media/webrtc/webrtcvideoencoderfactory.h',
|
|
'media/webrtc/webrtcvideoengine.cc',
|
|
'media/webrtc/webrtcvideoengine.h',
|
|
'media/webrtc/webrtcvideoengine2.cc',
|
|
'media/webrtc/webrtcvideoengine2.h',
|
|
'media/webrtc/webrtcvideoframe.cc',
|
|
'media/webrtc/webrtcvideoframe.h',
|
|
'media/webrtc/webrtcvideoframefactory.cc',
|
|
'media/webrtc/webrtcvideoframefactory.h',
|
|
'media/webrtc/webrtcvie.h',
|
|
'media/webrtc/webrtcvoe.h',
|
|
'media/webrtc/webrtcvoiceengine.cc',
|
|
'media/webrtc/webrtcvoiceengine.h',
|
|
],
|
|
'conditions': [
|
|
['build_with_chromium==1', {
|
|
'dependencies': [
|
|
'<(webrtc_root)/modules/modules.gyp:video_capture_module_impl',
|
|
'<(webrtc_root)/modules/modules.gyp:video_render_module_impl',
|
|
],
|
|
}, {
|
|
'dependencies': [
|
|
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
|
|
'<(webrtc_root)/modules/modules.gyp:video_render_module_internal_impl',
|
|
],
|
|
}],
|
|
['OS=="linux"', {
|
|
'sources': [
|
|
'media/devices/gtkvideorenderer.cc',
|
|
'media/devices/gtkvideorenderer.h',
|
|
'media/devices/libudevsymboltable.cc',
|
|
'media/devices/libudevsymboltable.h',
|
|
'media/devices/linuxdeviceinfo.cc',
|
|
'media/devices/linuxdevicemanager.cc',
|
|
'media/devices/linuxdevicemanager.h',
|
|
'media/devices/v4llookup.cc',
|
|
'media/devices/v4llookup.h',
|
|
],
|
|
'include_dirs': [
|
|
'third_party/libudev'
|
|
],
|
|
'cflags': [
|
|
'<!@(pkg-config --cflags gobject-2.0 gthread-2.0 gtk+-2.0)',
|
|
],
|
|
'libraries': [
|
|
'-lrt',
|
|
'-lXext',
|
|
'-lX11',
|
|
],
|
|
}],
|
|
['OS=="win"', {
|
|
'sources': [
|
|
'media/devices/gdivideorenderer.cc',
|
|
'media/devices/gdivideorenderer.h',
|
|
'media/devices/win32deviceinfo.cc',
|
|
'media/devices/win32devicemanager.cc',
|
|
'media/devices/win32devicemanager.h',
|
|
],
|
|
'msvs_settings': {
|
|
'VCLibrarianTool': {
|
|
'AdditionalDependencies': [
|
|
'd3d9.lib',
|
|
'gdi32.lib',
|
|
'strmiids.lib',
|
|
'winmm.lib',
|
|
],
|
|
},
|
|
},
|
|
}],
|
|
['OS=="mac"', {
|
|
'sources': [
|
|
'media/devices/macdeviceinfo.cc',
|
|
'media/devices/macdevicemanager.cc',
|
|
'media/devices/macdevicemanager.h',
|
|
'media/devices/macdevicemanagermm.mm',
|
|
],
|
|
'conditions': [
|
|
['target_arch=="ia32"', {
|
|
'sources': [
|
|
'media/devices/carbonvideorenderer.cc',
|
|
'media/devices/carbonvideorenderer.h',
|
|
],
|
|
'link_settings': {
|
|
'xcode_settings': {
|
|
'OTHER_LDFLAGS': [
|
|
'-framework Carbon',
|
|
],
|
|
},
|
|
},
|
|
}],
|
|
],
|
|
'xcode_settings': {
|
|
'WARNING_CFLAGS': [
|
|
# TODO(ronghuawu): Update macdevicemanager.cc to stop using
|
|
# deprecated functions and remove this flag.
|
|
'-Wno-deprecated-declarations',
|
|
],
|
|
},
|
|
'link_settings': {
|
|
'xcode_settings': {
|
|
'OTHER_LDFLAGS': [
|
|
'-weak_framework AVFoundation',
|
|
'-framework Cocoa',
|
|
'-framework CoreAudio',
|
|
'-framework CoreVideo',
|
|
'-framework OpenGL',
|
|
'-framework QTKit',
|
|
],
|
|
},
|
|
},
|
|
}],
|
|
['OS=="ios"', {
|
|
'sources': [
|
|
'media/devices/mobiledevicemanager.cc',
|
|
],
|
|
'include_dirs': [
|
|
# TODO(sjlee) Remove when vp8 is building for iOS. vp8 pulls in
|
|
# libjpeg which pulls in libyuv which currently disabled.
|
|
'../third_party/libyuv/include',
|
|
],
|
|
}],
|
|
['OS=="android"', {
|
|
'sources': [
|
|
'media/devices/mobiledevicemanager.cc',
|
|
],
|
|
}],
|
|
],
|
|
}, # target libjingle_media
|
|
{
|
|
'target_name': 'libjingle_p2p',
|
|
'type': 'static_library',
|
|
'dependencies': [
|
|
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
|
|
'libjingle',
|
|
'libjingle_media',
|
|
],
|
|
'include_dirs': [
|
|
'<(DEPTH)/testing/gtest/include',
|
|
],
|
|
'direct_dependent_settings': {
|
|
'include_dirs': [
|
|
'<(DEPTH)/testing/gtest/include',
|
|
],
|
|
},
|
|
'sources': [
|
|
'session/tunnel/pseudotcpchannel.cc',
|
|
'session/tunnel/pseudotcpchannel.h',
|
|
'session/tunnel/tunnelsessionclient.cc',
|
|
'session/tunnel/tunnelsessionclient.h',
|
|
'session/tunnel/securetunnelsessionclient.cc',
|
|
'session/tunnel/securetunnelsessionclient.h',
|
|
'session/media/audiomonitor.cc',
|
|
'session/media/audiomonitor.h',
|
|
'session/media/bundlefilter.cc',
|
|
'session/media/bundlefilter.h',
|
|
'session/media/call.cc',
|
|
'session/media/call.h',
|
|
'session/media/channel.cc',
|
|
'session/media/channel.h',
|
|
'session/media/channelmanager.cc',
|
|
'session/media/channelmanager.h',
|
|
'session/media/currentspeakermonitor.cc',
|
|
'session/media/currentspeakermonitor.h',
|
|
'session/media/mediamessages.cc',
|
|
'session/media/mediamessages.h',
|
|
'session/media/mediamonitor.cc',
|
|
'session/media/mediamonitor.h',
|
|
'session/media/mediarecorder.cc',
|
|
'session/media/mediarecorder.h',
|
|
'session/media/mediasession.cc',
|
|
'session/media/mediasession.h',
|
|
'session/media/mediasessionclient.cc',
|
|
'session/media/mediasessionclient.h',
|
|
'session/media/mediasink.h',
|
|
'session/media/rtcpmuxfilter.cc',
|
|
'session/media/rtcpmuxfilter.h',
|
|
'session/media/soundclip.cc',
|
|
'session/media/soundclip.h',
|
|
'session/media/srtpfilter.cc',
|
|
'session/media/srtpfilter.h',
|
|
'session/media/typingmonitor.cc',
|
|
'session/media/typingmonitor.h',
|
|
'session/media/voicechannel.h',
|
|
],
|
|
}, # target libjingle_p2p
|
|
{
|
|
'target_name': 'libjingle_peerconnection',
|
|
'type': 'static_library',
|
|
'dependencies': [
|
|
'libjingle',
|
|
'libjingle_media',
|
|
'libjingle_p2p',
|
|
],
|
|
'sources': [
|
|
'app/webrtc/audiotrack.cc',
|
|
'app/webrtc/audiotrack.h',
|
|
'app/webrtc/audiotrackrenderer.cc',
|
|
'app/webrtc/audiotrackrenderer.h',
|
|
'app/webrtc/datachannel.cc',
|
|
'app/webrtc/datachannel.h',
|
|
'app/webrtc/datachannelinterface.h',
|
|
'app/webrtc/dtmfsender.cc',
|
|
'app/webrtc/dtmfsender.h',
|
|
'app/webrtc/dtmfsenderinterface.h',
|
|
'app/webrtc/fakeportallocatorfactory.h',
|
|
'app/webrtc/jsep.h',
|
|
'app/webrtc/jsepicecandidate.cc',
|
|
'app/webrtc/jsepicecandidate.h',
|
|
'app/webrtc/jsepsessiondescription.cc',
|
|
'app/webrtc/jsepsessiondescription.h',
|
|
'app/webrtc/localaudiosource.cc',
|
|
'app/webrtc/localaudiosource.h',
|
|
'app/webrtc/mediaconstraintsinterface.cc',
|
|
'app/webrtc/mediaconstraintsinterface.h',
|
|
'app/webrtc/mediastream.cc',
|
|
'app/webrtc/mediastream.h',
|
|
'app/webrtc/mediastreamhandler.cc',
|
|
'app/webrtc/mediastreamhandler.h',
|
|
'app/webrtc/mediastreaminterface.h',
|
|
'app/webrtc/mediastreamprovider.h',
|
|
'app/webrtc/mediastreamproxy.h',
|
|
'app/webrtc/mediastreamsignaling.cc',
|
|
'app/webrtc/mediastreamsignaling.h',
|
|
'app/webrtc/mediastreamtrack.h',
|
|
'app/webrtc/mediastreamtrackproxy.h',
|
|
'app/webrtc/notifier.h',
|
|
'app/webrtc/peerconnection.cc',
|
|
'app/webrtc/peerconnection.h',
|
|
'app/webrtc/peerconnectionfactory.cc',
|
|
'app/webrtc/peerconnectionfactory.h',
|
|
'app/webrtc/peerconnectioninterface.h',
|
|
'app/webrtc/peerconnectionproxy.h',
|
|
'app/webrtc/portallocatorfactory.cc',
|
|
'app/webrtc/portallocatorfactory.h',
|
|
'app/webrtc/proxy.h',
|
|
'app/webrtc/remoteaudiosource.cc',
|
|
'app/webrtc/remoteaudiosource.h',
|
|
'app/webrtc/remotevideocapturer.cc',
|
|
'app/webrtc/remotevideocapturer.h',
|
|
'app/webrtc/sctputils.cc',
|
|
'app/webrtc/sctputils.h',
|
|
'app/webrtc/statscollector.cc',
|
|
'app/webrtc/statscollector.h',
|
|
'app/webrtc/statstypes.h',
|
|
'app/webrtc/streamcollection.h',
|
|
'app/webrtc/videosource.cc',
|
|
'app/webrtc/videosource.h',
|
|
'app/webrtc/videosourceinterface.h',
|
|
'app/webrtc/videosourceproxy.h',
|
|
'app/webrtc/videotrack.cc',
|
|
'app/webrtc/videotrack.h',
|
|
'app/webrtc/videotrackrenderers.cc',
|
|
'app/webrtc/videotrackrenderers.h',
|
|
'app/webrtc/webrtcsdp.cc',
|
|
'app/webrtc/webrtcsdp.h',
|
|
'app/webrtc/webrtcsession.cc',
|
|
'app/webrtc/webrtcsession.h',
|
|
'app/webrtc/webrtcsessiondescriptionfactory.cc',
|
|
'app/webrtc/webrtcsessiondescriptionfactory.h',
|
|
],
|
|
}, # target libjingle_peerconnection
|
|
],
|
|
}
|