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webrtc/talk/libjingle.gyp
kjellander@webrtc.org f58fe0ab2b Rename GYP and GN targets for video capture+render. 
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.

Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)

GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture

video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render

BUG=456815
R=andresp@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35099004

Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-11 07:47:47 +00:00

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#
# libjingle
# Copyright 2012 Google Inc.
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions are met:
#
# 1. Redistributions of source code must retain the above copyright notice,
# this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright notice,
# this list of conditions and the following disclaimer in the documentation
# and/or other materials provided with the distribution.
# 3. The name of the author may not be used to endorse or promote products
# derived from this software without specific prior written permission.
#
# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
{
'includes': ['build/common.gypi'],
'conditions': [
['os_posix == 1 and OS != "mac" and OS != "ios"', {
'conditions': [
['sysroot!=""', {
'variables': {
'pkg-config': '../../../build/linux/pkg-config-wrapper "<(sysroot)" "<(target_arch)"',
},
}, {
'variables': {
'pkg-config': 'pkg-config'
},
}],
],
}],
['OS=="linux" or OS=="android"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_so',
'type': 'shared_library',
'dependencies': [
'libjingle_peerconnection',
'<(DEPTH)/third_party/icu/icu.gyp:icuuc',
],
'sources': [
'app/webrtc/java/jni/peerconnection_jni.cc'
],
'include_dirs': [
'<(DEPTH)/third_party/libyuv/include',
],
'conditions': [
['OS=="linux"', {
'defines': [
'HAVE_GTK',
],
'include_dirs': [
'<(java_home)/include',
'<(java_home)/include/linux',
],
'link_settings': {
'libraries': [
'<!@(pkg-config --libs-only-l gobject-2.0 gthread-2.0'
' gtk+-2.0)',
],
},
}],
],
},
{
'target_name': 'libjingle_peerconnection_jar',
'type': 'none',
'actions': [
{
'variables': {
'java_src_dir': 'app/webrtc/java/src',
'webrtc_modules_dir': '<(webrtc_root)/modules',
'build_jar_log': '<(INTERMEDIATE_DIR)/build_jar.log',
'peerconnection_java_files': [
'app/webrtc/java/src/org/webrtc/AudioSource.java',
'app/webrtc/java/src/org/webrtc/AudioTrack.java',
'app/webrtc/java/src/org/webrtc/DataChannel.java',
'app/webrtc/java/src/org/webrtc/IceCandidate.java',
'app/webrtc/java/src/org/webrtc/Logging.java',
'app/webrtc/java/src/org/webrtc/MediaConstraints.java',
'app/webrtc/java/src/org/webrtc/MediaSource.java',
'app/webrtc/java/src/org/webrtc/MediaStream.java',
'app/webrtc/java/src/org/webrtc/MediaStreamTrack.java',
'app/webrtc/java/src/org/webrtc/PeerConnectionFactory.java',
'app/webrtc/java/src/org/webrtc/PeerConnection.java',
'app/webrtc/java/src/org/webrtc/SdpObserver.java',
'app/webrtc/java/src/org/webrtc/StatsObserver.java',
'app/webrtc/java/src/org/webrtc/StatsReport.java',
'app/webrtc/java/src/org/webrtc/SessionDescription.java',
'app/webrtc/java/src/org/webrtc/VideoCapturer.java',
'app/webrtc/java/src/org/webrtc/VideoRenderer.java',
'app/webrtc/java/src/org/webrtc/VideoSource.java',
'app/webrtc/java/src/org/webrtc/VideoTrack.java',
],
# TODO(fischman): extract this into a webrtc gyp var that can be
# included here, or better yet, build a proper .jar in webrtc
# and include it here.
'android_java_files': [
'app/webrtc/java/android/org/webrtc/VideoRendererGui.java',
'app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.java',
'app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java',
'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/AudioManagerAndroid.java',
'<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureAndroid.java',
'<(webrtc_modules_dir)/video_capture/android/java/src/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java',
'<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViEAndroidGLES20.java',
'<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViERenderer.java',
'<(webrtc_modules_dir)/video_render/android/java/src/org/webrtc/videoengine/ViESurfaceRenderer.java',
'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java',
'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java',
],
},
'action_name': 'create_jar',
'inputs': [
'build/build_jar.sh',
'<@(java_files)',
],
'outputs': [
'<(PRODUCT_DIR)/libjingle_peerconnection.jar',
],
'conditions': [
['OS=="android"', {
'variables': {
'java_files': ['<@(peerconnection_java_files)', '<@(android_java_files)'],
'build_classpath': '<(java_src_dir):<(DEPTH)/third_party/android_tools/sdk/platforms/android-<(android_sdk_version)/android.jar',
},
}, {
'variables': {
'java_files': ['<@(peerconnection_java_files)'],
'build_classpath': '<(java_src_dir)',
},
}],
],
'action': [
'bash', '-ec',
'mkdir -p <(INTERMEDIATE_DIR) && '
'{ build/build_jar.sh <(java_home) <@(_outputs) '
' <(INTERMEDIATE_DIR)/build_jar.tmp '
' <(build_classpath) <@(java_files) '
' > <(build_jar_log) 2>&1 || '
' { cat <(build_jar_log) ; exit 1; } }'
],
},
],
'dependencies': [
'libjingle_peerconnection_so',
],
},
],
}],
['OS=="android"', {
'targets': [
{
# |libjingle_peerconnection_java| builds a jar file with name
# libjingle_peerconnection_java.jar using Chromes build system.
# It includes all Java files needed to setup a PeeerConnection call
# from Android.
# TODO(perkj): Consider replacing the use of
# libjingle_peerconnection_jar with this target everywhere.
'target_name': 'libjingle_peerconnection_java',
'type': 'none',
'dependencies': [
'libjingle_peerconnection_so',
],
'variables': {
'java_in_dir': 'app/webrtc/java',
'webrtc_modules_dir': '<(webrtc_root)/modules',
'additional_src_dirs' : [
'app/webrtc/java/android',
'<(webrtc_modules_dir)/audio_device/android/java/src',
'<(webrtc_modules_dir)/video_capture/android/java/src',
'<(webrtc_modules_dir)/video_render/android/java/src',
],
},
'includes': ['../build/java.gypi'],
}, # libjingle_peerconnection_java
]
}],
['OS=="ios" or (OS=="mac" and target_arch!="ia32" and mac_sdk>="10.7")', {
# The >= 10.7 above is required for ARC.
'targets': [
{
'target_name': 'libjingle_peerconnection_objc',
'type': 'static_library',
'dependencies': [
'libjingle_peerconnection',
],
'sources': [
'app/webrtc/objc/RTCAudioTrack+Internal.h',
'app/webrtc/objc/RTCAudioTrack.mm',
'app/webrtc/objc/RTCDataChannel+Internal.h',
'app/webrtc/objc/RTCDataChannel.mm',
'app/webrtc/objc/RTCEnumConverter.h',
'app/webrtc/objc/RTCEnumConverter.mm',
'app/webrtc/objc/RTCI420Frame+Internal.h',
'app/webrtc/objc/RTCI420Frame.mm',
'app/webrtc/objc/RTCICECandidate+Internal.h',
'app/webrtc/objc/RTCICECandidate.mm',
'app/webrtc/objc/RTCICEServer+Internal.h',
'app/webrtc/objc/RTCICEServer.mm',
'app/webrtc/objc/RTCMediaConstraints+Internal.h',
'app/webrtc/objc/RTCMediaConstraints.mm',
'app/webrtc/objc/RTCMediaConstraintsNative.cc',
'app/webrtc/objc/RTCMediaConstraintsNative.h',
'app/webrtc/objc/RTCMediaSource+Internal.h',
'app/webrtc/objc/RTCMediaSource.mm',
'app/webrtc/objc/RTCMediaStream+Internal.h',
'app/webrtc/objc/RTCMediaStream.mm',
'app/webrtc/objc/RTCMediaStreamTrack+Internal.h',
'app/webrtc/objc/RTCMediaStreamTrack.mm',
'app/webrtc/objc/RTCOpenGLVideoRenderer.mm',
'app/webrtc/objc/RTCPair.m',
'app/webrtc/objc/RTCPeerConnection+Internal.h',
'app/webrtc/objc/RTCPeerConnection.mm',
'app/webrtc/objc/RTCPeerConnectionFactory.mm',
'app/webrtc/objc/RTCPeerConnectionObserver.h',
'app/webrtc/objc/RTCPeerConnectionObserver.mm',
'app/webrtc/objc/RTCSessionDescription+Internal.h',
'app/webrtc/objc/RTCSessionDescription.mm',
'app/webrtc/objc/RTCStatsReport+Internal.h',
'app/webrtc/objc/RTCStatsReport.mm',
'app/webrtc/objc/RTCVideoCapturer+Internal.h',
'app/webrtc/objc/RTCVideoCapturer.mm',
'app/webrtc/objc/RTCVideoRendererAdapter.h',
'app/webrtc/objc/RTCVideoRendererAdapter.mm',
'app/webrtc/objc/RTCVideoSource+Internal.h',
'app/webrtc/objc/RTCVideoSource.mm',
'app/webrtc/objc/RTCVideoTrack+Internal.h',
'app/webrtc/objc/RTCVideoTrack.mm',
'app/webrtc/objc/public/RTCAudioSource.h',
'app/webrtc/objc/public/RTCAudioTrack.h',
'app/webrtc/objc/public/RTCDataChannel.h',
'app/webrtc/objc/public/RTCI420Frame.h',
'app/webrtc/objc/public/RTCICECandidate.h',
'app/webrtc/objc/public/RTCICEServer.h',
'app/webrtc/objc/public/RTCMediaConstraints.h',
'app/webrtc/objc/public/RTCMediaSource.h',
'app/webrtc/objc/public/RTCMediaStream.h',
'app/webrtc/objc/public/RTCMediaStreamTrack.h',
'app/webrtc/objc/public/RTCOpenGLVideoRenderer.h',
'app/webrtc/objc/public/RTCPair.h',
'app/webrtc/objc/public/RTCPeerConnection.h',
'app/webrtc/objc/public/RTCPeerConnectionDelegate.h',
'app/webrtc/objc/public/RTCPeerConnectionFactory.h',
'app/webrtc/objc/public/RTCSessionDescription.h',
'app/webrtc/objc/public/RTCSessionDescriptionDelegate.h',
'app/webrtc/objc/public/RTCStatsDelegate.h',
'app/webrtc/objc/public/RTCStatsReport.h',
'app/webrtc/objc/public/RTCTypes.h',
'app/webrtc/objc/public/RTCVideoCapturer.h',
'app/webrtc/objc/public/RTCVideoRenderer.h',
'app/webrtc/objc/public/RTCVideoSource.h',
'app/webrtc/objc/public/RTCVideoTrack.h',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/talk/app/webrtc/objc/public',
],
},
'include_dirs': [
'<(DEPTH)/talk/app/webrtc',
'<(DEPTH)/talk/app/webrtc/objc',
'<(DEPTH)/talk/app/webrtc/objc/public',
],
'link_settings': {
'libraries': [
'-lstdc++',
],
},
'all_dependent_settings': {
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
},
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
# common.gypi enables this for mac but we want this to be disabled
# like it is for ios.
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
},
'conditions': [
['OS=="ios"', {
'sources': [
'app/webrtc/objc/RTCEAGLVideoView.m',
'app/webrtc/objc/public/RTCEAGLVideoView.h',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreGraphics',
'-framework GLKit',
],
},
},
}],
['OS=="mac"', {
'sources': [
'app/webrtc/objc/RTCNSGLVideoView.m',
'app/webrtc/objc/public/RTCNSGLVideoView.h',
],
'xcode_settings': {
# Need to build against 10.7 framework for full ARC support
# on OSX.
'MACOSX_DEPLOYMENT_TARGET' : '10.7',
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework Cocoa',
],
},
},
}],
],
}, # target libjingle_peerconnection_objc
],
}],
],
'targets': [
{
'target_name': 'libjingle',
'type': 'none',
'dependencies': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
'<(webrtc_root)/base/base.gyp:rtc_base',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
}, # target libjingle
{
'target_name': 'libjingle_media',
'type': 'static_library',
'include_dirs': [
# TODO(jiayl): move this into the direct_dependent_settings of
# usrsctp.gyp.
'<(DEPTH)/third_party/usrsctp',
],
'dependencies': [
'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
'<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
'<(webrtc_root)/modules/modules.gyp:video_render_module',
'<(webrtc_root)/webrtc.gyp:webrtc',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(webrtc_root)/sound/sound.gyp:rtc_sound',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/libjingle/xmllite/xmllite.gyp:rtc_xmllite',
'<(webrtc_root)/libjingle/xmpp/xmpp.gyp:rtc_xmpp',
'<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',
'libjingle',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/third_party/libyuv/include',
],
},
'sources': [
'media/base/audioframe.h',
'media/base/audiorenderer.h',
'media/base/capturemanager.cc',
'media/base/capturemanager.h',
'media/base/capturerenderadapter.cc',
'media/base/capturerenderadapter.h',
'media/base/codec.cc',
'media/base/codec.h',
'media/base/constants.cc',
'media/base/constants.h',
'media/base/cpuid.cc',
'media/base/cpuid.h',
'media/base/cryptoparams.h',
'media/base/device.h',
'media/base/fakescreencapturerfactory.h',
'media/base/filemediaengine.cc',
'media/base/filemediaengine.h',
'media/base/hybriddataengine.h',
'media/base/mediachannel.h',
'media/base/mediacommon.h',
'media/base/mediaengine.cc',
'media/base/mediaengine.h',
'media/base/rtpdataengine.cc',
'media/base/rtpdataengine.h',
'media/base/rtpdump.cc',
'media/base/rtpdump.h',
'media/base/rtputils.cc',
'media/base/rtputils.h',
'media/base/screencastid.h',
'media/base/streamparams.cc',
'media/base/streamparams.h',
'media/base/videoadapter.cc',
'media/base/videoadapter.h',
'media/base/videocapturer.cc',
'media/base/videocapturer.h',
'media/base/videocapturerfactory.h',
'media/base/videocommon.cc',
'media/base/videocommon.h',
'media/base/videoframe.cc',
'media/base/videoframe.h',
'media/base/videoframefactory.cc',
'media/base/videoframefactory.h',
'media/base/videoprocessor.h',
'media/base/videorenderer.h',
'media/base/voiceprocessor.h',
'media/base/yuvframegenerator.cc',
'media/base/yuvframegenerator.h',
'media/devices/deviceinfo.h',
'media/devices/devicemanager.cc',
'media/devices/devicemanager.h',
'media/devices/dummydevicemanager.h',
'media/devices/filevideocapturer.cc',
'media/devices/filevideocapturer.h',
'media/devices/videorendererfactory.h',
'media/devices/yuvframescapturer.cc',
'media/devices/yuvframescapturer.h',
'media/other/linphonemediaengine.h',
'media/sctp/sctpdataengine.cc',
'media/sctp/sctpdataengine.h',
'media/webrtc/simulcast.cc',
'media/webrtc/simulcast.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',
'<(webrtc_root)/modules/modules.gyp:video_render',
],
}, {
'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/media/audiomonitor.cc',
'session/media/audiomonitor.h',
'session/media/bundlefilter.cc',
'session/media/bundlefilter.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/mediamonitor.cc',
'session/media/mediamonitor.h',
'session/media/mediarecorder.cc',
'session/media/mediarecorder.h',
'session/media/mediasession.cc',
'session/media/mediasession.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/peerconnectionfactoryproxy.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.cc',
'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
],
}
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