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platform-external-webrtc/talk/libjingle.gyp
Fredrik Solenberg 709ed67c38 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels. 
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1269863005 .

Cr-Commit-Position: refs/heads/master@{#9939}
2015-09-15 10:26:45 +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',
],
'sources': [
'app/webrtc/java/jni/classreferenceholder.cc',
'app/webrtc/java/jni/classreferenceholder.h',
'app/webrtc/java/jni/jni_helpers.cc',
'app/webrtc/java/jni/jni_helpers.h',
'app/webrtc/java/jni/native_handle_impl.h',
'app/webrtc/java/jni/peerconnection_jni.cc',
],
'include_dirs': [
'<(libyuv_dir)/include',
],
'conditions': [
['build_icu==1', {
'dependencies': [
'<(DEPTH)/third_party/icu/icu.gyp:icuuc',
],
}],
['OS=="linux"', {
'defines': [
'HAVE_GTK',
],
'include_dirs': [
'<(java_home)/include',
'<(java_home)/include/linux',
],
'conditions': [
['use_gtk==1', {
'link_settings': {
'libraries': [
'<!@(pkg-config --libs-only-l gobject-2.0 gthread-2.0'
' gtk+-2.0)',
],
},
}],
],
}],
['OS=="android"', {
'sources': [
'app/webrtc/java/jni/androidvideocapturer_jni.cc',
'app/webrtc/java/jni/androidvideocapturer_jni.h',
],
'variables': {
# This library uses native JNI exports; tell GYP so that the
# required symbols will be kept.
'use_native_jni_exports': 1,
},
}],
['OS=="android" and build_with_chromium==0', {
'sources': [
'app/webrtc/java/jni/androidmediacodeccommon.h',
'app/webrtc/java/jni/androidmediadecoder_jni.cc',
'app/webrtc/java/jni/androidmediadecoder_jni.h',
'app/webrtc/java/jni/androidmediaencoder_jni.cc',
'app/webrtc/java/jni/androidmediaencoder_jni.h',
]
}],
],
},
{
'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/CallSessionFileRotatingLogSink.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/Camera2Enumerator.java',
'app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java',
'app/webrtc/java/android/org/webrtc/CameraEnumerator.java',
'app/webrtc/java/android/org/webrtc/EglBase.java',
'app/webrtc/java/android/org/webrtc/GlRectDrawer.java',
'app/webrtc/java/android/org/webrtc/GlShader.java',
'app/webrtc/java/android/org/webrtc/GlUtil.java',
'app/webrtc/java/android/org/webrtc/RendererCommon.java',
'app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java',
'app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java',
'app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java',
'app/webrtc/java/android/org/webrtc/VideoRendererGui.java',
'app/webrtc/java/src/org/webrtc/MediaCodecVideoDecoder.java',
'app/webrtc/java/src/org/webrtc/MediaCodecVideoEncoder.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/BuildInfo.java',
'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java',
'<(webrtc_modules_dir)/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.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/RTCFileLogger.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/RTCLogging.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/RTCPeerConnectionInterface+Internal.h',
'app/webrtc/objc/RTCPeerConnectionInterface.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/RTCFileLogger.h',
'app/webrtc/objc/public/RTCI420Frame.h',
'app/webrtc/objc/public/RTCICECandidate.h',
'app/webrtc/objc/public/RTCICEServer.h',
'app/webrtc/objc/public/RTCLogging.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/RTCPeerConnectionInterface.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/avfoundationvideocapturer.h',
'app/webrtc/objc/avfoundationvideocapturer.mm',
'app/webrtc/objc/RTCAVFoundationVideoSource+Internal.h',
'app/webrtc/objc/RTCAVFoundationVideoSource.mm',
'app/webrtc/objc/RTCEAGLVideoView.m',
'app/webrtc/objc/public/RTCEAGLVideoView.h',
'app/webrtc/objc/public/RTCAVFoundationVideoSource.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',
# RTCVideoTrack.mm uses code with partial availability.
# https://code.google.com/p/webrtc/issues/detail?id=4695
'WARNING_CFLAGS!': ['-Wpartial-availability'],
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework Cocoa',
],
},
},
}],
],
}, # target libjingle_peerconnection_objc
],
}],
],
'targets': [
{
'target_name': 'libjingle',
'type': 'none',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
],
'conditions': [
['build_json==1', {
'dependencies': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
}],
['build_expat==1', {
'dependencies': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/expat/expat.gyp:expat',
],
}],
],
}, # target libjingle
{
'target_name': 'libjingle_media',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(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': [
'<(libyuv_dir)/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/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/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/sctp/sctpdataengine.cc',
'media/sctp/sctpdataengine.h',
'media/webrtc/simulcast.cc',
'media/webrtc/simulcast.h',
'media/webrtc/webrtccommon.h',
'media/webrtc/webrtcmediaengine.cc',
'media/webrtc/webrtcmediaengine.h',
'media/webrtc/webrtcmediaengine.cc',
'media/webrtc/webrtcpassthroughrender.cc',
'media/webrtc/webrtcpassthroughrender.h',
'media/webrtc/webrtcvideocapturer.cc',
'media/webrtc/webrtcvideocapturer.h',
'media/webrtc/webrtcvideocapturerfactory.h',
'media/webrtc/webrtcvideocapturerfactory.cc',
'media/webrtc/webrtcvideodecoderfactory.h',
'media/webrtc/webrtcvideoencoderfactory.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/webrtcvoe.h',
'media/webrtc/webrtcvoiceengine.cc',
'media/webrtc/webrtcvoiceengine.h',
],
'conditions': [
['build_libyuv==1', {
'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',],
}],
['build_usrsctp==1', {
'include_dirs': [
# TODO(jiayl): move this into the direct_dependent_settings of
# usrsctp.gyp.
'<(DEPTH)/third_party/usrsctp',
],
'dependencies': [
'<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
],
}],
['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/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',
],
'conditions': [
['use_gtk==1', {
'sources': [
'media/devices/gtkvideorenderer.cc',
'media/devices/gtkvideorenderer.h',
],
'cflags': [
'<!@(pkg-config --cflags gobject-2.0 gthread-2.0 gtk+-2.0)',
],
}],
],
'include_dirs': [
'third_party/libudev'
],
'libraries': [
'-lrt',
],
}],
['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',
],
# Disable partial availability warning to prevent errors
# in macdevicemanagermm.mm using AVFoundation.
# https://code.google.com/p/webrtc/issues/detail?id=4695
'WARNING_CFLAGS!': ['-Wpartial-availability'],
},
'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': [
'libjingle',
'libjingle_media',
],
'conditions': [
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
],
'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/mediasession.cc',
'session/media/mediasession.h',
'session/media/mediasink.h',
'session/media/rtcpmuxfilter.cc',
'session/media/rtcpmuxfilter.h',
'session/media/srtpfilter.cc',
'session/media/srtpfilter.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/dtlsidentitystore.cc',
'app/webrtc/dtlsidentitystore.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/mediacontroller.cc',
'app/webrtc/mediacontroller.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',
],
'conditions': [
['OS=="android" and build_with_chromium==0', {
'sources': [
'app/webrtc/androidvideocapturer.h',
'app/webrtc/androidvideocapturer.cc',
],
}],
],
}, # target libjingle_peerconnection
],
}
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