Bug 1393119 - Remove webrtc gyp files; r=jesup

This removes the gyp files to build webrtc. It looks like part of Bug 1371485 is
to vendor gyp elsewhere in tree at which time we can complete cleaning this up.

MozReview-Commit-ID: 8MqatafniN5

--HG--
extra : rebase_source : 1cf7a41f0b8a1a95dc008f4a39536ee7e76027c4
This commit is contained in:
Dan Minor 2017-12-19 09:21:03 -05:00
Родитель 22eb0295df
Коммит 2e81df0e1c
83 изменённых файлов: 0 добавлений и 15436 удалений

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@ -1,255 +0,0 @@
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'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'
},
}],
],
}],
# Excluded from the Chromium build since they cannot be built due to
# incompability with Chromium's logging implementation.
['OS=="android" and build_with_chromium==0 and build_with_mozilla==0', {
'targets': [
{
'target_name': 'libjingle_peerconnection_jni',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
'libjingle_peerconnection',
],
'sources': [
'android/jni/androidmediacodeccommon.h',
'android/jni/androidmediadecoder_jni.cc',
'android/jni/androidmediadecoder_jni.h',
'android/jni/androidmediaencoder_jni.cc',
'android/jni/androidmediaencoder_jni.h',
'android/jni/androidmetrics_jni.cc',
'android/jni/androidnetworkmonitor_jni.cc',
'android/jni/androidnetworkmonitor_jni.h',
'android/jni/androidvideotracksource_jni.cc',
'android/jni/classreferenceholder.cc',
'android/jni/classreferenceholder.h',
'android/jni/jni_helpers.cc',
'android/jni/jni_helpers.h',
'android/jni/native_handle_impl.cc',
'android/jni/native_handle_impl.h',
'android/jni/peerconnection_jni.cc',
'android/jni/surfacetexturehelper_jni.cc',
'android/jni/surfacetexturehelper_jni.h',
'androidvideotracksource.cc',
'androidvideotracksource.h',
],
'include_dirs': [
'<(libyuv_dir)/include',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
'-Wno-sign-compare',
'-Wno-unused-variable',
],
'cflags!': [
'-Wextra',
],
'msvs_disabled_warnings': [
4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch.
4267, # conversion from 'size_t' to 'int', possible loss of data.
4389, # signed/unsigned mismatch.
],
},
{
'target_name': 'libjingle_peerconnection_so',
'type': 'shared_library',
'dependencies': [
'libjingle_peerconnection',
'libjingle_peerconnection_jni',
],
'sources': [
'android/jni/jni_onload.cc',
],
'variables': {
# This library uses native JNI exports; tell GYP so that the
# required symbols will be kept.
'use_native_jni_exports': 1,
},
},
]
}],
], # conditions
'targets': [
{
'target_name': 'call_api',
'type': 'static_library',
'dependencies': [
# TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:audio_encoder_interface',
],
'sources': [
'call/audio_receive_stream.h',
'call/audio_send_stream.h',
'call/audio_sink.h',
'call/audio_state.h',
],
},
{
'target_name': 'video_frame_api',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
],
'sources': [
'call/transport.h',
'video/i420_buffer.cc',
'video/i420_buffer.h',
'video/video_frame.cc',
'video/video_frame.h',
'video/video_frame_buffer.h',
'video/video_rotation.h',
],
'include_dirs': [
'<(libyuv_dir)/include',
],
},
{
'target_name': 'libjingle_peerconnection',
'type': 'static_library',
'dependencies': [
':call_api',
':rtc_stats_api',
'<(webrtc_root)/media/media.gyp:rtc_media',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
'<(webrtc_root)/stats/stats.gyp:rtc_stats',
],
'sources': [
'audiotrack.cc',
'audiotrack.h',
'datachannel.cc',
'datachannel.h',
'datachannelinterface.h',
'dtmfsender.cc',
'dtmfsender.h',
'dtmfsenderinterface.h',
'jsep.h',
'jsepicecandidate.cc',
'jsepicecandidate.h',
'jsepsessiondescription.cc',
'jsepsessiondescription.h',
'localaudiosource.cc',
'localaudiosource.h',
'mediaconstraintsinterface.cc',
'mediaconstraintsinterface.h',
'mediacontroller.cc',
'mediacontroller.h',
'mediastream.cc',
'mediastream.h',
'mediastreaminterface.h',
'mediastreamobserver.cc',
'mediastreamobserver.h',
'mediastreamproxy.h',
'mediastreamtrack.h',
'mediastreamtrackproxy.h',
'notifier.h',
'peerconnection.cc',
'peerconnection.h',
'peerconnectionfactory.cc',
'peerconnectionfactory.h',
'peerconnectionfactoryproxy.h',
'peerconnectioninterface.h',
'peerconnectionproxy.h',
'proxy.h',
'remoteaudiosource.cc',
'remoteaudiosource.h',
'rtcstatscollector.cc',
'rtcstatscollector.h',
'rtpparameters.h',
'rtpreceiver.cc',
'rtpreceiver.h',
'rtpreceiverinterface.h',
'rtpsender.cc',
'rtpsender.h',
'rtpsenderinterface.h',
'sctputils.cc',
'sctputils.h',
'statscollector.cc',
'statscollector.h',
'statstypes.cc',
'statstypes.h',
'streamcollection.h',
'videocapturertracksource.cc',
'videocapturertracksource.h',
'videosourceproxy.h',
'videotrack.cc',
'videotrack.h',
'videotracksource.cc',
'videotracksource.h',
'webrtcsdp.cc',
'webrtcsdp.h',
'webrtcsession.cc',
'webrtcsession.h',
'webrtcsessiondescriptionfactory.cc',
'webrtcsessiondescriptionfactory.h',
],
'conditions': [
['clang==1', {
'cflags!': [
'-Wextra',
],
'xcode_settings': {
'WARNING_CFLAGS!': ['-Wextra'],
},
}, {
'cflags': [
'-Wno-maybe-uninitialized', # Only exists for GCC.
],
}],
['use_quic==1', {
'dependencies': [
'<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
],
'sources': [
'quicdatachannel.cc',
'quicdatachannel.h',
'quicdatatransport.cc',
'quicdatatransport.h',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
],
}],
],
}, # target libjingle_peerconnection
{
# GN version: webrtc/api:rtc_stats_api
'target_name': 'rtc_stats_api',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'sources': [
'stats/rtcstats.h',
'stats/rtcstats_objects.h',
'stats/rtcstatsreport.h',
],
}, # target rtc_stats_api
], # targets
}

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@ -1,50 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# This file exists only because there's no other way to avoid errors in the
# Chromium build due to the inclusion of build/java.gypi. GYP unfortunately
# processes all 'includes' for all .gyp files, ignoring conditions. This
# processing takes place early in the cycle, before it's possible to use
# variables. It will go away when WebRTC has fully migrated to GN.
{
'includes': [ '../build/common.gypi', ],
'conditions': [
['OS=="android"', {
'targets': [
{
# |libjingle_peerconnection_java| builds a jar file with name
# libjingle_peerconnection_java.jar using Chrome's build system.
# It includes all Java files needed to setup a PeeerConnection call
# from Android.
'target_name': 'libjingle_peerconnection_java',
'type': 'none',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection_so',
],
'variables': {
# Designate as Chromium code and point to our lint settings to
# enable linting of the WebRTC code (this is the only way to make
# lint_action invoke the Android linter).
'android_manifest_path': '<(webrtc_root)/build/android/AndroidManifest.xml',
'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml',
'chromium_code': 1,
'java_in_dir': 'android/java',
'webrtc_base_dir': '<(webrtc_root)/base',
'webrtc_modules_dir': '<(webrtc_root)/modules',
'additional_src_dirs' : [
'<(webrtc_base_dir)/java',
'<(webrtc_modules_dir)/audio_device/android/java/src',
],
},
'includes': ['../../build/java.gypi'],
},
], # targets
}], # OS=="android"
], # conditions
}

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# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'webrtc_audio_dependencies': [
'<(webrtc_root)/api/api.gyp:call_api',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
],
'webrtc_audio_sources': [
'audio/audio_receive_stream.cc',
'audio/audio_receive_stream.h',
'audio/audio_send_stream.cc',
'audio/audio_send_stream.h',
'audio/audio_state.cc',
'audio/audio_state.h',
'audio/audio_transport_proxy.cc',
'audio/audio_transport_proxy.h',
'audio/conversion.h',
'audio/scoped_voe_interface.h',
'audio/utility/audio_frame_operations.cc',
'audio/utility/audio_frame_operations.h',
],
},
}

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# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'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'
},
}],
],
}],
],
'targets': [
{
# The subset of rtc_base approved for use outside of libjingle.
'target_name': 'rtc_base_approved',
'type': 'static_library',
'sources': [
'arraysize.h',
'array_view.h',
'atomicops.h',
'basictypes.h',
'base64.cc',
'base64.h',
'bind.h',
'bitbuffer.cc',
'bitbuffer.h',
'buffer.h',
'bufferqueue.cc',
'bufferqueue.h',
'bytebuffer.cc',
'bytebuffer.h',
'byteorder.h',
'checks.cc',
'checks.h',
'common.cc',
'common.h',
'constructormagic.h',
'copyonwritebuffer.cc',
'copyonwritebuffer.h',
'criticalsection.cc',
'criticalsection.h',
'deprecation.h',
'event.cc',
'event.h',
'event_tracer.cc',
'event_tracer.h',
'numerics/exp_filter.cc',
'numerics/exp_filter.h',
'numerics/percentile_filter.h',
'file.cc',
'file.h',
'flags.cc',
'flags.h',
'format_macros.h',
'function_view.h',
'ignore_wundef.h',
'location.h',
'location.cc',
'md5.cc',
'md5.h',
'md5digest.cc',
'md5digest.h',
'mod_ops.h',
'onetimeevent.h',
'optional.cc',
'optional.h',
'pathutils.cc',
'pathutils.h',
'platform_file.cc',
'platform_file.h',
'platform_thread.cc',
'platform_thread.h',
'platform_thread_types.h',
'race_checker.cc',
'race_checker.h',
'random.cc',
'random.h',
'rate_statistics.cc',
'rate_statistics.h',
'rate_limiter.cc',
'rate_limiter.h',
'ratetracker.cc',
'ratetracker.h',
'refcount.h',
'refcountedobject.h',
'safe_compare.h',
'safe_conversions.h',
'safe_conversions_impl.h',
'sanitizer.h',
'scoped_ref_ptr.h',
'stringencode.cc',
'stringencode.h',
'stringutils.cc',
'stringutils.h',
'swap_queue.h',
'template_util.h',
'thread_annotations.h',
'thread_checker.h',
'thread_checker_impl.cc',
'thread_checker_impl.h',
'timestampaligner.cc',
'timestampaligner.h',
'timeutils.cc',
'timeutils.h',
'trace_event.h',
'type_traits.h',
],
'conditions': [
['os_posix==1', {
'sources': [
'file_posix.cc',
],
}],
['OS=="win"', {
'sources': [
'file_win.cc',
'win32.cc',
'win32.h',
],
}],
['OS=="mac"', {
'sources': [
'macutils.cc',
'macutils.h',
],
}],
['build_with_chromium==1', {
'dependencies': [
'<(DEPTH)/base/base.gyp:base',
],
'include_dirs': [
'../../webrtc_overrides',
],
'sources': [
'../../webrtc_overrides/webrtc/base/logging.cc',
'../../webrtc_overrides/webrtc/base/logging.h',
],
}, {
'sources': [
'logging.cc',
'logging.h',
'logging_mac.mm',
],
}],
['OS=="mac" and build_with_chromium==0', {
'all_dependent_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
# needed for logging_mac.mm
'-framework Foundation',
],
},
},
}], # OS=="mac" and build_with_chromium==0
['OS=="android"', {
'link_settings': {
'libraries': [
'-llog',
],
},
}],
],
},
{
'target_name': 'rtc_task_queue',
'type': 'static_library',
'dependencies': [
'rtc_base_approved',
],
'sources': [
'sequenced_task_checker.h',
'sequenced_task_checker_impl.cc',
'sequenced_task_checker_impl.h',
'weak_ptr.cc',
'weak_ptr.h',
],
'conditions': [
['build_with_chromium==1', {
'include_dirs': [
'../../webrtc_overrides'
],
'sources' : [
'../../webrtc_overrides/webrtc/base/task_queue.cc',
'../../webrtc_overrides/webrtc/base/task_queue.h',
]
} , {
# If not build for chromium, use our own implementation.
'sources' : [
'task_queue.h',
'task_queue_posix.h',
],
'conditions': [
#TODO: no libevent.gyp
['build_libevent==1', {
'dependencies': [
'<(DEPTH)/base/third_party/libevent/libevent.gyp:libevent',
],
} , {
'include_dirs': [
'$(MOZ_LIBEVENT_CFLAGS)',
'<(libevent_dir)/',
'<(libevent_dir)/../',
'<(libevent_dir)/include/',
],
'conditions': [
['OS=="mac" or OS=="ios"', {
'include_dirs': [
'<(libevent_dir)/mac/',
],
}],
['OS=="linux"', {
'include_dirs': [
'<(libevent_dir)/linux/',
],
}],
['os_bsd==1', {
'include_dirs': [
'<(libevent_dir)/bsd/',
],
}],
['OS=="win"', {
'include_dirs': [
'<(libevent_dir)/WIN32-Code/',
],
}],
['OS=="android"', {
'include_dirs': [
'<(libevent_dir)/android/',
],
}],
],
}],
['enable_libevent==1', {
'sources': [
'task_queue_libevent.cc',
'task_queue_posix.cc',
],
'defines': [ 'WEBRTC_BUILD_LIBEVENT', ],
'all_dependent_settings': {
'defines': [ 'WEBRTC_BUILD_LIBEVENT' ],
},
}, {
# If not libevent, fall back to the other task queues.
'conditions': [
['OS=="mac" or OS=="ios"', {
'sources': [
'task_queue_gcd.cc',
'task_queue_posix.cc',
],
}],
['OS=="win"', {
'sources': [ 'task_queue_win.cc' ],
}]
],
}],
]
}],
],
},
{
'target_name': 'rtc_base',
'type': 'static_library',
'dependencies': [
'../common.gyp:webrtc_common',
'rtc_base_approved',
],
'export_dependent_settings': [
'rtc_base_approved',
],
'defines': [
'FEATURE_ENABLE_SSL',
'SSL_USE_OPENSSL',
'HAVE_OPENSSL_SSL_H',
'LOGGING=1',
],
'sources': [
'applefilesystem.mm',
'asyncinvoker.cc',
'asyncinvoker.h',
'asyncinvoker-inl.h',
'asyncpacketsocket.cc',
'asyncpacketsocket.h',
'asyncresolverinterface.cc',
'asyncresolverinterface.h',
'asyncsocket.cc',
'asyncsocket.h',
'asynctcpsocket.cc',
'asynctcpsocket.h',
'asyncudpsocket.cc',
'asyncudpsocket.h',
'autodetectproxy.cc',
'autodetectproxy.h',
'crc32.cc',
'crc32.h',
'cryptstring.cc',
'cryptstring.h',
'diskcache.cc',
'diskcache.h',
'filerotatingstream.cc',
'filerotatingstream.h',
'fileutils.cc',
'fileutils.h',
'firewallsocketserver.cc',
'firewallsocketserver.h',
'gunit_prod.h',
'helpers.cc',
'helpers.h',
'httpbase.cc',
'httpbase.h',
'httpclient.cc',
'httpclient.h',
'httpcommon-inl.h',
'httpcommon.cc',
'httpcommon.h',
'ipaddress.cc',
'ipaddress.h',
'linked_ptr.h',
'messagedigest.cc',
'messagedigest.h',
'messagehandler.cc',
'messagehandler.h',
'messagequeue.cc',
'messagequeue.h',
'nethelpers.cc',
'nethelpers.h',
'network.cc',
'network.h',
'networkmonitor.cc',
'networkmonitor.h',
'nullsocketserver.cc',
'nullsocketserver.h',
'openssl.h',
'openssladapter.cc',
'openssladapter.h',
'openssldigest.cc',
'openssldigest.h',
'opensslidentity.cc',
'opensslidentity.h',
'opensslstreamadapter.cc',
'opensslstreamadapter.h',
'physicalsocketserver.cc',
'physicalsocketserver.h',
'proxydetect.cc',
'proxydetect.h',
'proxyinfo.cc',
'proxyinfo.h',
'ratelimiter.cc',
'ratelimiter.h',
'rtccertificate.cc',
'rtccertificate.h',
'rtccertificategenerator.cc',
'rtccertificategenerator.h',
'sha1.cc',
'sha1.h',
'sha1digest.cc',
'sha1digest.h',
'sharedexclusivelock.cc',
'sharedexclusivelock.h',
'signalthread.cc',
'signalthread.h',
'sigslot.cc',
'sigslot.h',
'sigslotrepeater.h',
'socket.h',
'socketadapters.cc',
'socketadapters.h',
'socketaddress.cc',
'socketaddress.h',
'socketaddresspair.cc',
'socketaddresspair.h',
'socketfactory.h',
'socketpool.cc',
'socketpool.h',
'socketserver.h',
'socketstream.cc',
'socketstream.h',
'ssladapter.cc',
'ssladapter.h',
'sslfingerprint.cc',
'sslfingerprint.h',
'sslidentity.cc',
'sslidentity.h',
'sslsocketfactory.cc',
'sslsocketfactory.h',
'sslstreamadapter.cc',
'sslstreamadapter.h',
'stream.cc',
'stream.h',
'task.cc',
'task.h',
'taskparent.cc',
'taskparent.h',
'taskrunner.cc',
'taskrunner.h',
'thread.cc',
'thread.h',
],
# TODO(henrike): issue 3307, make rtc_base build without disabling
# these flags.
'cflags!': [
'-Wextra',
'-Wall',
],
'direct_dependent_settings': {
'defines': [
'FEATURE_ENABLE_SSL',
'SSL_USE_OPENSSL',
'HAVE_OPENSSL_SSL_H',
],
},
'conditions': [
['build_with_chromium==1', {
'include_dirs': [
'../../webrtc_overrides',
'../../boringssl/src/include',
],
'conditions': [
['OS=="win"', {
'sources': [
'../../webrtc_overrides/webrtc/base/win32socketinit.cc',
],
}],
],
'defines': [
'NO_MAIN_THREAD_WRAPPING',
],
'direct_dependent_settings': {
'defines': [
'NO_MAIN_THREAD_WRAPPING',
],
},
}, {
'sources': [
'callback.h',
'fileutils_mock.h',
'httpserver.cc',
'httpserver.h',
'json.cc',
'json.h',
'logsinks.cc',
'logsinks.h',
'mathutils.h',
'natserver.cc',
'natserver.h',
'natsocketfactory.cc',
'natsocketfactory.h',
'nattypes.cc',
'nattypes.h',
'optionsfile.cc',
'optionsfile.h',
'proxyserver.cc',
'proxyserver.h',
'referencecountedsingletonfactory.h',
'rollingaccumulator.h',
'scopedptrcollection.h',
'sslconfig.h',
'sslroots.h',
'testbase64.h',
'testclient.cc',
'testclient.h',
'transformadapter.cc',
'transformadapter.h',
'virtualsocketserver.cc',
'virtualsocketserver.h',
'window.h',
'windowpicker.h',
'windowpickerfactory.h',
],
'conditions': [
['build_json==1', {
'dependencies': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
}, {
'include_dirs': [
'<(json_root)',
],
'defines': [
# When defined changes the include path for json.h to where it
# is expected to be when building json outside of the standalone
# build.
'WEBRTC_EXTERNAL_JSON',
],
}],
['OS=="mac"', {
'sources': [
'macasyncsocket.cc',
'macasyncsocket.h',
'maccocoasocketserver.h',
'maccocoasocketserver.mm',
'macsocketserver.cc',
'macsocketserver.h',
'macwindowpicker.cc',
'macwindowpicker.h',
],
}],
['OS=="win"', {
'sources': [
'diskcache_win32.cc',
'diskcache_win32.h',
'win32regkey.cc',
'win32regkey.h',
'win32socketinit.cc',
'win32socketinit.h',
'win32socketserver.cc',
'win32socketserver.h',
],
}],
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-sign-compare',
'-Wno-missing-braces',
],
},
},
}],
], # conditions
}], # build_with_chromium==0
['OS=="android"', {
'sources': [
'ifaddrs-android.cc',
'ifaddrs-android.h',
],
'link_settings': {
'libraries': [
'-llog',
'-lGLESv2',
],
},
}],
['(OS=="mac" or OS=="ios") and nacl_untrusted_build==0', {
'sources': [
'maccocoathreadhelper.h',
'maccocoathreadhelper.mm',
'macconversion.cc',
'macconversion.h',
'macifaddrs_converter.cc',
'scoped_autorelease_pool.h',
'scoped_autorelease_pool.mm',
],
}],
['OS=="ios"', {
'all_dependent_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CFNetwork',
'-framework Foundation',
'-framework Security',
'-framework SystemConfiguration',
'-framework UIKit',
],
},
},
}],
['use_x11==1', {
'sources': [
],
'link_settings': {
'libraries': [
'-ldl',
'-lrt',
'-lXext',
'-lX11',
'-lXcomposite',
'-lXrender',
],
},
}],
['OS=="linux"', {
'link_settings': {
'libraries': [
'-ldl',
'-lrt',
],
},
}],
['OS=="mac"', {
# moved by mozilla
# 'sources': [
# 'macutils.cc',
# 'macutils.h',
# ],
'all_dependent_settings': {
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework Cocoa',
'-framework Foundation',
'-framework IOKit',
'-framework Security',
'-framework SystemConfiguration',
],
},
},
},
}],
['OS=="win" and nacl_untrusted_build==0', {
'sources': [
# moved by mozilla
# 'win32.cc',
# 'win32.h',
'win32filesystem.cc',
'win32filesystem.h',
'win32securityerrors.cc',
'win32window.cc',
'win32window.h',
'win32windowpicker.cc',
'win32windowpicker.h',
'winfirewall.cc',
'winfirewall.h',
'winping.cc',
'winping.h',
],
'link_settings': {
'libraries': [
'-lcrypt32.lib',
'-liphlpapi.lib',
'-lsecur32.lib',
],
},
# Suppress warnings about WIN32_LEAN_AND_MEAN.
'msvs_disabled_warnings': [4005, 4703],
'defines': [
'_CRT_NONSTDC_NO_DEPRECATE',
],
}],
['os_posix==1', {
'sources': [
'ifaddrs_converter.cc',
'ifaddrs_converter.h',
'unixfilesystem.cc',
'unixfilesystem.h',
],
'configurations': {
'Debug_Base': {
'defines': [
# Chromium's build/common.gypi defines this for all posix
# _except_ for ios & mac. We want it there as well, e.g.
# because ASSERT and friends trigger off of it.
'_DEBUG',
],
},
}
}],
['build_ssl==1', {
'dependencies': [
'<(DEPTH)/third_party/boringssl/boringssl.gyp:boringssl',
],
}, {
'include_dirs': [
'<(ssl_root)',
],
}],
],
},
{
'target_name': 'gtest_prod',
'type': 'static_library',
'sources': [
'gtest_prod_util.h',
],
},
],
}

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@ -1,48 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [ '../build/common.gypi', ],
'targets': [
{
'target_name': 'rtc_base_tests_utils',
'type': 'static_library',
'sources': [
'unittest_main.cc',
# Also use this as a convenient dumping ground for misc files that are
# included by multiple targets below.
'fakeclock.cc',
'fakeclock.h',
'fakenetwork.h',
'fakesslidentity.h',
'faketaskrunner.h',
'gunit.h',
'testbase64.h',
'testechoserver.h',
'testutils.h',
'timedelta.h',
],
'defines': [
'GTEST_RELATIVE_PATH',
],
'dependencies': [
'base.gyp:rtc_base',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:field_trial',
'<(webrtc_root)/test/test.gyp:test_support',
],
'direct_dependent_settings': {
'defines': [
'GTEST_RELATIVE_PATH',
],
},
'export_dependent_settings': [
'<(DEPTH)/testing/gtest.gyp:gtest',
],
},
],
}

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@ -1,60 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# This file sets correct neon flags. Include it if you want to build
# source with neon intrinsics.
# To use this, create a gyp target with the following form:
# {
# 'target_name': 'my_lib',
# 'type': 'static_library',
# 'sources': [
# 'foo.c',
# 'bar.cc',
# ],
# 'includes': ['path/to/this/gypi/file'],
# }
{
'cflags!': [
'-mfpu=vfpv3-d16',
],
'cflags_mozilla!': [
'-mfpu=vfpv3-d16',
],
'asflags!': [
'-mfpu=vfpv3-d16',
],
'asflags_mozilla!': [
'-mfpu=vfpv3-d16',
],
'conditions': [
# "-mfpu=neon" is not required for arm64 in GCC.
['target_arch!="arm64"', {
'cflags': [
'-mfpu=neon',
],
'cflags_mozilla': [
'-mfpu=neon',
],
'asflags': [
'-mfpu=neon',
],
'asflags_mozilla': [
'-mfpu=neon',
],
}],
# Disable GCC LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
['clang==0 and use_lto==1', {
'cflags!': [
'-flto',
'-ffat-lto-objects',
],
}],
],
}

Разница между файлами не показана из-за своего большого размера Загрузить разницу

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@ -1,665 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# This file contains common settings for building WebRTC components.
{
# Nesting is required in order to use variables for setting other variables.
'variables': {
'variables': {
'variables': {
'variables': {
# This will already be set to zero by supplement.gypi
'build_with_chromium%': 1,
# Enable to use the Mozilla internal settings.
'build_with_mozilla%': 0,
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla%)',
'include_opus%': 1,
'rtc_opus_variable_complexity%': 0,
'conditions': [
# Include the iLBC audio codec?
['build_with_chromium==1 or build_with_mozilla==1', {
'include_ilbc%': 0,
}, {
'include_ilbc%': 1,
}],
['build_with_chromium==1', {
'webrtc_root%': '<(DEPTH)/third_party/webrtc',
}, {
'webrtc_root%': '<(DEPTH)/webrtc',
}],
# Controls whether we use libevent on posix platforms.
# TODO(phoglund): should arguably be controlled by platform #ifdefs
# in the code instead.
['OS=="win" or OS=="mac" or OS=="ios"', {
'build_libevent%': 0,
'enable_libevent%': 0,
}, {
'build_libevent%': 1,
'enable_libevent%': 1,
'libevent_dir%': '<(DEPTH)/third_party/libevent',
}],
],
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla)',
'build_libevent%': '<(build_libevent)',
'enable_libevent%': '<(enable_libevent)',
'webrtc_root%': '<(webrtc_root)',
'webrtc_vp8_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp8',
'webrtc_vp9_dir%': '<(webrtc_root)/modules/video_coding/codecs/vp9',
'webrtc_h264_dir%': '<(webrtc_root)/modules/video_coding/codecs/h264',
'libevent_dir%': '<(DEPTH)/third_party/libevent',
'include_g711%': 1,
'include_g722%': 1,
'include_ilbc%': '<(include_ilbc)',
'include_opus%': '<(include_opus)',
'include_isac%': 1,
'include_pcm16b%': 1,
'opus_dir%': '<(DEPTH)/third_party/opus',
},
'build_with_chromium%': '<(build_with_chromium)',
'build_with_mozilla%': '<(build_with_mozilla)',
'build_libevent%': '<(build_libevent)',
'enable_libevent%': '<(enable_libevent)',
'webrtc_root%': '<(webrtc_root)',
'test_runner_path': '<(DEPTH)/webrtc/build/android/test_runner.py',
'webrtc_vp8_dir%': '<(webrtc_vp8_dir)',
'webrtc_vp9_dir%': '<(webrtc_vp9_dir)',
'webrtc_h264_dir%': '<(webrtc_h264_dir)',
'libevent_dir%': '<(libevent_dir)',
'include_g711%': '<(include_g711)',
'include_g722%': '<(include_g722)',
'include_ilbc%': '<(include_ilbc)',
'include_opus%': '<(include_opus)',
'include_isac%': '<(include_isac)',
'include_pcm16b%': '<(include_pcm16b)',
'rtc_relative_path%': 1,
'external_libraries%': '0',
'json_root%': '<(DEPTH)/third_party/jsoncpp/source/include/',
# openssl needs to be defined or gyp will complain. Is is only used when
# when providing external libraries so just use current directory as a
# placeholder.
'ssl_root%': '.',
# The Chromium common.gypi we use treats all gyp files without
# chromium_code==1 as third party code. This disables many of the
# preferred warning settings.
#
# We can set this here to have WebRTC code treated as Chromium code. Our
# third party code will still have the reduced warning settings.
'chromium_code': 1,
# Targets are by default not NaCl untrusted code. Use this variable exclude
# code that uses libraries that aren't available in the NaCl sandbox.
'nacl_untrusted_build%': 0,
# Set to 1 to enable code coverage on Linux using the gcov library.
'coverage%': 0,
# Set to "func", "block", "edge" for coverage generation.
# At unit test runtime set UBSAN_OPTIONS="coverage=1".
# It is recommend to set include_examples=0.
# Use llvm's sancov -html-report for human readable reports.
# See http://clang.llvm.org/docs/SanitizerCoverage.html .
'webrtc_sanitize_coverage%': "",
# Remote bitrate estimator logging/plotting.
'enable_bwe_test_logging%': 0,
# Selects fixed-point code where possible.
'prefer_fixed_point%': 0,
# Enable data logging. Produces text files with data logged within engines
# which can be easily parsed for offline processing.
'enable_data_logging%': 0,
# Enables the use of protocol buffers for debug recordings.
'enable_protobuf%': 1,
# Disable the code for the intelligibility enhancer by default.
'enable_intelligibility_enhancer%': 0,
# Selects whether debug dumps for the audio processing module
# should be generated.
'apm_debug_dump%': 0,
# Disable these to not build components which can be externally provided.
'build_expat%': 1,
'build_json%': 1,
'build_libsrtp%': 1,
'build_libvpx%': 1,
'libvpx_build_vp9%': 1,
'build_libyuv%': 1,
'build_openmax_dl%': 1,
'build_opus%': 1,
'build_protobuf%': 1,
'build_ssl%': 1,
'build_usrsctp%': 1,
# Disable by default
'have_dbus_glib%': 0,
# Make it possible to provide custom locations for some libraries.
'libvpx_dir%': '<(DEPTH)/third_party/libvpx',
'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
'libevent_dir%': '<(DEPTH)/third_party/libevent',
'opus_dir%': '<(opus_dir)',
# Use Java based audio layer as default for Android.
# Change this setting to 1 to use Open SL audio instead.
# TODO(henrika): add support for Open SL ES.
'enable_android_opensl%': 0,
# Link-Time Optimizations
# Executes code generation at link-time instead of compile-time
# https://gcc.gnu.org/wiki/LinkTimeOptimization
'use_lto%': 0,
# Defer ssl perference to that specified through sslconfig.h instead of
# choosing openssl or nss directly. In practice, this can be used to
# enable schannel on windows.
'use_legacy_ssl_defaults%': 0,
# Determines whether NEON code will be built.
'build_with_neon%': 0,
# Disable this to skip building source requiring GTK.
'use_gtk%': 1,
# Enable this to prevent extern symbols from being hidden on iOS builds.
# The chromium settings we inherit hide symbols by default on Release
# builds. We want our symbols to be visible when distributing WebRTC via
# static libraries to avoid linker warnings.
'ios_override_visibility%': 0,
# Determines whether QUIC code will be built.
'use_quic%': 0,
# By default, use normal platform audio support or dummy audio, but don't
# use file-based audio playout and record.
'use_dummy_audio_file_devices%': 0,
'conditions': [
# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported
# on all platforms except Android and iOS. Because FFmpeg can be built
# with/without H.264 support, |ffmpeg_branding| has to separately be set
# to a value that includes H.264, for example "Chrome". If FFmpeg is built
# without H.264, compilation succeeds but |H264DecoderImpl| fails to
# initialize. See also: |rtc_initialize_ffmpeg|.
# CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
# http://www.openh264.org, https://www.ffmpeg.org/
# TODO: proprietary_codecs is undefined here?
#['proprietary_codecs==1 and OS!="android" and OS!="ios"', {
# 'rtc_use_h264%': 1,
#}, {
# 'rtc_use_h264%': 0,
#}],
# FFmpeg must be initialized for |H264DecoderImpl| to work. This can be
# done by WebRTC during |H264DecoderImpl::InitDecode| or externally.
# FFmpeg must only be initialized once. Projects that initialize FFmpeg
# externally, such as Chromium, must turn this flag off so that WebRTC
# does not also initialize.
['build_with_chromium==0', {
'rtc_initialize_ffmpeg%': 1,
}, {
'rtc_initialize_ffmpeg%': 0,
}],
['build_with_chromium==1', {
# Build sources requiring GTK. NOTICE: This is not present in Chrome OS
# build environments, even if available for Chromium builds.
'use_gtk%': 0,
# Exclude pulse audio on Chromium since its prerequisites don't require
# pulse audio.
'include_pulse_audio%': 0,
# Exclude internal ADM since Chromium uses its own IO handling.
'include_internal_audio_device%': 0,
'include_ndk_cpu_features%': 0,
# Remove tests for Chromium to avoid slowing down GYP generation.
'include_tests%': 0,
'restrict_webrtc_logging%': 1,
}, { # Settings for the standalone (not-in-Chromium) build.
'use_gtk%': 1,
# TODO(andrew): For now, disable the Chrome plugins, which causes a
# flood of chromium-style warnings. Investigate enabling them:
# http://code.google.com/p/webrtc/issues/detail?id=163
'clang_use_chrome_plugins%': 0,
'include_pulse_audio%': 1,
'include_internal_audio_device%': 1,
'include_ndk_cpu_features%': 0,
'conditions': [
['build_with_mozilla==1', {
'include_tests%': 0,
'conditions': [
# silly gyp won't let me do 'a': !'b'
# suppress TRACE logging in non-debug builds
['debug==1', {
'restrict_webrtc_logging%': 0,
}, {
'restrict_webrtc_logging%': 1,
}],
],
}, {
'include_tests%': 1,
'restrict_webrtc_logging%': 0,
}],
],
}],
['OS=="linux"', {
'include_alsa_audio%': 1,
}, {
'include_alsa_audio%': 0,
}],
['OS=="openbsd"', {
'include_sndio_audio%': 1,
}, {
'include_sndio_audio%': 0,
}],
['OS=="solaris" or (OS!="openbsd" and os_bsd==1)', {
'include_pulse_audio%': 1,
}, {
'include_pulse_audio%': 0,
}],
['OS=="linux" or OS=="solaris" or os_bsd==1', {
'include_v4l2_video_capture%': 1,
}, {
'include_v4l2_video_capture%': 0,
}],
['target_arch=="arm" or target_arch=="arm64" or target_arch=="mipsel"', {
'prefer_fixed_point%': 1,
}],
['(target_arch=="arm" and arm_neon==1) or target_arch=="arm64"', {
'build_with_neon%': 1,
}],
['OS!="ios" and (target_arch!="arm" or arm_version>=7) and target_arch!="mips64el" and build_with_mozilla==0', {
'rtc_use_openmax_dl%': 1,
}, {
'rtc_use_openmax_dl%': 0,
}],
], # conditions
},
'target_defaults': {
'conditions': [
['restrict_webrtc_logging==1', {
'defines': ['WEBRTC_RESTRICT_LOGGING',],
}],
['build_with_mozilla==1', {
'defines': [
# Changes settings for Mozilla build.
'WEBRTC_MOZILLA_BUILD',
'WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT',
],
}],
['have_dbus_glib==1', {
'defines': [
'HAVE_DBUS_GLIB',
],
'cflags': [
'<!@(pkg-config --cflags dbus-glib-1)',
],
}],
['rtc_relative_path==1', {
'defines': ['EXPAT_RELATIVE_PATH',],
}],
['os_posix==1', {
'configurations': {
'Debug_Base': {
'defines': [
# Chromium's build/common.gypi defines _DEBUG for all posix
# _except_ for ios & mac. The size of data types such as
# pthread_mutex_t varies between release and debug builds
# and is controlled via this flag. Since we now share code
# between base/base.gyp and build/common.gypi (this file),
# both gyp(i) files, must consistently set this flag uniformly
# or else we'll run in to hard-to-figure-out problems where
# one compilation unit uses code from another but expects
# differently laid out types.
# For WebRTC, we want it there as well, because ASSERT and
# friends trigger off of it.
'_DEBUG',
],
},
},
}],
['build_with_chromium==1', {
'defines': [
# Changes settings for Chromium build.
# TODO(kjellander): Cleanup unused ones and move defines closer to the
# source when webrtc:4256 is completed.
'ENABLE_EXTERNAL_AUTH',
'FEATURE_ENABLE_SSL',
'HAVE_OPENSSL_SSL_H',
'HAVE_SCTP',
'HAVE_SRTP',
'HAVE_WEBRTC_VIDEO',
'HAVE_WEBRTC_VOICE',
'LOGGING_INSIDE_WEBRTC',
'NO_MAIN_THREAD_WRAPPING',
'NO_SOUND_SYSTEM',
'SRTP_RELATIVE_PATH',
'SSL_USE_OPENSSL',
'USE_WEBRTC_DEV_BRANCH',
'WEBRTC_CHROMIUM_BUILD',
],
'include_dirs': [
# Include the top-level directory when building in Chrome, so we can
# use full paths (e.g. headers inside testing/ or third_party/).
'<(DEPTH)',
# The overrides must be included before the WebRTC root as that's the
# mechanism for selecting the override headers in Chromium.
'../../webrtc_overrides',
# The WebRTC root is needed to allow includes in the WebRTC code base
# to be prefixed with webrtc/.
'../..',
],
}, {
'includes': [
# Rules for excluding e.g. foo_win.cc from the build on non-Windows.
'filename_rules.gypi',
],
# Include the top-level dir so the WebRTC code can use full paths.
'include_dirs': [
'../..',
],
'conditions': [
['os_posix==1', {
'conditions': [
# -Wextra is currently disabled in Chromium's common.gypi. Enable
# for targets that can handle it. For Android/arm64 right now
# there will be an 'enumeral and non-enumeral type in conditional
# expression' warning in android_tools/ndk_experimental's version
# of stlport.
# See: https://code.google.com/p/chromium/issues/detail?id=379699
['target_arch!="arm64" or OS!="android"', {
'cflags': [
'-Wextra',
# We need to repeat some flags from Chromium's common.gypi
# here that get overridden by -Wextra.
'-Wno-unused-parameter',
'-Wno-missing-field-initializers',
'-Wno-strict-overflow',
],
}],
],
'cflags_cc': [
'-Wnon-virtual-dtor',
# This is enabled for clang; enable for gcc as well.
'-Woverloaded-virtual',
],
}],
['clang==1', {
'cflags': [
'-Wimplicit-fallthrough',
'-Wthread-safety',
'-Winconsistent-missing-override',
],
'cflags_mozilla': [
'-Wthread-safety',
],
}],
],
}],
['target_arch=="arm64"', {
'defines': [
'WEBRTC_ARCH_ARM64',
'WEBRTC_HAS_NEON',
],
}],
['target_arch=="arm"', {
'build_with_neon%': 1,
'defines': [
'WEBRTC_ARCH_ARM',
],
'conditions': [
['arm_version>=7', {
'defines': ['WEBRTC_ARCH_ARM_V7',
'WEBRTC_BUILD_NEON_LIBS',
'WEBRTC_HAS_NEON'],
'cflags_mozilla': ['-mfloat-abi=softfp',
'-mfpu=neon'],
}],
],
}],
['os_bsd==1', {
'defines': [
'WEBRTC_BSD',
],
}],
['OS=="openbsd"', {
'defines' : [
'WEBRTC_AUDIO_SNDIO',
],
}],
# Mozilla: if we support Mozilla on MIPS, we'll need to mod the cflags entries here
['target_arch=="mipsel" and mips_arch_variant!="r6"', {
'defines': [
'MIPS32_LE',
],
'conditions': [
['mips_float_abi=="hard"', {
'defines': [
'MIPS_FPU_LE',
],
}],
['mips_arch_variant=="r2"', {
'defines': [
'MIPS32_R2_LE',
],
}],
['mips_dsp_rev==1', {
'defines': [
'MIPS_DSP_R1_LE',
],
}],
['mips_dsp_rev==2', {
'defines': [
'MIPS_DSP_R1_LE',
'MIPS_DSP_R2_LE',
],
}],
],
}],
['coverage==1 and OS=="linux"', {
'cflags': [ '-ftest-coverage',
'-fprofile-arcs' ],
'ldflags': [ '--coverage' ],
'link_settings': { 'libraries': [ '-lgcov' ] },
}],
['webrtc_sanitize_coverage!=""', {
'cflags': [ '-fsanitize-coverage=<(webrtc_sanitize_coverage)' ],
'ldflags': [ '-fsanitize-coverage=<(webrtc_sanitize_coverage)' ],
}],
['webrtc_sanitize_coverage!="" and OS=="mac"', {
'xcode_settings': {
'OTHER_CFLAGS': [
'-fsanitize-coverage=func',
],
},
}],
['os_posix==1', {
# For access to standard POSIXish features, use WEBRTC_POSIX instead of
# a more specific macro.
'defines': [
'WEBRTC_POSIX',
],
}],
['OS=="ios"', {
'defines': [
'WEBRTC_MAC',
'WEBRTC_IOS',
],
}],
['OS=="ios" and ios_override_visibility==1', {
'xcode_settings': {
'GCC_INLINES_ARE_PRIVATE_EXTERN': 'NO',
'GCC_SYMBOLS_PRIVATE_EXTERN': 'NO',
}
}],
['OS=="linux"', {
'defines': [
'WEBRTC_LINUX',
],
}],
['OS=="mac"', {
'defines': [
'WEBRTC_MAC',
],
}],
['OS=="win"', {
'defines': [
'WEBRTC_WIN',
],
# TODO(andrew): enable all warnings when possible.
# TODO(phoglund): get rid of 4373 supression when
# http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
'msvs_disabled_warnings': [
4373, # legacy warning for ignoring const / volatile in signatures.
4389, # Signed/unsigned mismatch.
],
# Re-enable some warnings that Chromium disables.
'msvs_disabled_warnings!': [4189,],
}],
['enable_android_opensl==1 and OS=="android"', {
'defines': [
'WEBRTC_ANDROID_OPENSLES',
],
}],
['OS=="android"', {
'defines': [
'WEBRTC_LINUX',
'WEBRTC_ANDROID',
],
'conditions': [
['clang==0', {
# The Android NDK doesn't provide optimized versions of these
# functions. Ensure they are disabled for all compilers.
'cflags': [
'-fno-builtin-cos',
'-fno-builtin-sin',
'-fno-builtin-cosf',
'-fno-builtin-sinf',
],
}],
],
}],
['chromeos==1', {
'defines': [
'CHROMEOS',
],
}],
['os_bsd==1', {
'defines': [
'WEBRTC_BSD',
],
}],
['include_internal_audio_device==1', {
'defines': [
'WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE',
],
}],
['libvpx_build_vp9==0', {
'defines': [
'RTC_DISABLE_VP9',
],
}],
], # conditions
'direct_dependent_settings': {
'conditions': [
['build_with_mozilla==1', {
'defines': [
# Changes settings for Mozilla build.
'WEBRTC_MOZILLA_BUILD',
],
}],
['build_with_chromium==1', {
'defines': [
# Changes settings for Chromium build.
# TODO(kjellander): Cleanup unused ones and move defines closer to
# the source when webrtc:4256 is completed.
'FEATURE_ENABLE_SSL',
'FEATURE_ENABLE_VOICEMAIL',
'EXPAT_RELATIVE_PATH',
'GTEST_RELATIVE_PATH',
'NO_MAIN_THREAD_WRAPPING',
'NO_SOUND_SYSTEM',
'WEBRTC_CHROMIUM_BUILD',
],
'include_dirs': [
# The overrides must be included first as that is the mechanism for
# selecting the override headers in Chromium.
'../../webrtc_overrides',
'../..',
],
}, {
'include_dirs': [
'../..',
],
}],
['OS=="mac"', {
'defines': [
'WEBRTC_MAC',
],
}],
['OS=="ios"', {
'defines': [
'WEBRTC_MAC',
'WEBRTC_IOS',
],
}],
['OS=="win"', {
'defines': [
'WEBRTC_WIN',
'_CRT_SECURE_NO_WARNINGS', # Suppress warnings about _vsnprinf
],
}],
['OS=="linux"', {
'defines': [
'WEBRTC_LINUX',
],
}],
['OS=="android"', {
'defines': [
'WEBRTC_LINUX',
'WEBRTC_ANDROID',
],
}],
['os_posix==1', {
# For access to standard POSIXish features, use WEBRTC_POSIX instead
# of a more specific macro.
'defines': [
'WEBRTC_POSIX',
],
}],
['chromeos==1', {
'defines': [
'CHROMEOS',
],
}],
['os_bsd==1', {
'defines': [
'WEBRTC_BSD',
],
}],
],
},
}, # target_defaults
}

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@ -1,107 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# This is a copy of the deleted build/filename_includes.gypi that
# has now been dropped from Chromium.
{
'target_conditions': [
['OS!="win" or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_win(_browsertest|_unittest|_test)?\\.(h|cc)$'],
['exclude', '(^|/)win/'],
['exclude', '(^|/)win_[^/]*\\.(h|cc)$'] ],
}],
['OS!="mac" or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_(cocoa|mac|mach)(_unittest|_test)?\\.(h|cc|c|mm?)$'],
['exclude', '(^|/)(cocoa|mac|mach)/'] ],
}],
['OS!="ios" or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_ios(_unittest|_test)?\\.(h|cc|mm?)$'],
['exclude', '(^|/)ios/'] ],
}],
['(OS!="mac" and OS!="ios") or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '\\.mm?$' ] ],
}],
# Do not exclude the linux files on *BSD since most of them can be
# shared at this point.
# In case a file is not needed, it is going to be excluded later on.
# TODO(evan): the above is not correct; we shouldn't build _linux
# files on non-linux.
['OS!="linux" and OS!="solaris" and <(os_bsd)!=1 or >(nacl_untrusted_build)==1', {
'sources/': [
['exclude', '_linux(_unittest|_test)?\\.(h|cc)$'],
['exclude', '(^|/)linux/'],
],
}],
['OS!="android" or _toolset=="host" or >(nacl_untrusted_build)==1', {
'sources/': [
['exclude', '_android(_unittest|_test)?\\.(h|cc)$'],
['exclude', '(^|/)android/'],
],
}],
['OS=="win" and >(nacl_untrusted_build)==0', {
'sources/': [
['exclude', '_posix(_unittest|_test)?\\.(h|cc)$'],
['exclude', '(^|/)posix/'],
],
}],
['<(chromeos)!=1 or >(nacl_untrusted_build)==1', {
'sources/': [
['exclude', '_chromeos(_unittest|_test)?\\.(h|cc)$'],
['exclude', '(^|/)chromeos/'],
],
}],
['>(nacl_untrusted_build)==0', {
'sources/': [
['exclude', '_nacl(_unittest)?\\.(h|cc)$'],
],
}],
['OS!="linux" and OS!="solaris" and <(os_bsd)!=1 or >(nacl_untrusted_build)==1', {
'sources/': [
['exclude', '_xdg(_unittest)?\\.(h|cc)$'],
],
}],
['<(use_x11)!=1 or >(nacl_untrusted_build)==1', {
'sources/': [
['exclude', '_(x|x11)(_interactive_uitest|_unittest)?\\.(h|cc)$'],
['exclude', '(^|/)x11_[^/]*\\.(h|cc)$'],
['exclude', '(^|/)x11/'],
['exclude', '(^|/)x/'],
],
}],
['<(toolkit_views)==0 or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_views(_browsertest|_unittest)?\\.(h|cc)$'] ]
}],
['<(use_aura)==0 or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_aura(_browsertest|_unittest)?\\.(h|cc)$'],
['exclude', '(^|/)aura/'],
['exclude', '_ash(_browsertest|_unittest)?\\.(h|cc)$'],
['exclude', '(^|/)ash/'],
]
}],
['<(use_aura)==0 or <(use_x11)==0 or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_aurax11(_browsertest|_unittest)?\\.(h|cc)$'] ]
}],
['<(use_aura)==0 or OS!="win" or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_aurawin\\.(h|cc)$'],
['exclude', '_ashwin\\.(h|cc)$']
]
}],
['<(use_aura)==0 or OS!="linux" or >(nacl_untrusted_build)==1', {
'sources/': [ ['exclude', '_auralinux\\.(h|cc)$'] ]
}],
#TODO: use_ozone is undefined here
#['<(use_ozone)==0 or >(nacl_untrusted_build)==1', {
# 'sources/': [ ['exclude', '_ozone(_browsertest|_unittest)?\\.(h|cc)$'] ]
#}],
#TODO: use_pango is undefined here
#['<(use_pango)==0', {
# 'sources/': [ ['exclude', '(^|_)pango(_util|_browsertest|_unittest)?\\.(h|cc)$'], ],
#}],
]
}

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@ -1,26 +0,0 @@
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': ['../common.gypi',],
'conditions': [
['OS=="ios" or OS=="mac"', {
'targets': [
{
'target_name': 'rtc_sdk_peerconnection_objc_no_op',
'includes': [ 'objc_app.gypi' ],
'type': 'executable',
'dependencies': [
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
],
'sources': ['no_op.cc',],
},
],
}]
],
}

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@ -1,33 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# Include this .gypi in an ObjC target's definition to allow it to be
# used as an iOS or OS/X application.
{
'variables': {
'infoplist_file': './objc_app.plist',
},
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
],
'mac_bundle': 1,
'mac_bundle_resources': [
'<(infoplist_file)',
],
# The plist is listed above so that it appears in XCode's file list,
# but we don't actually want to bundle it.
'mac_bundle_resources!': [
'<(infoplist_file)',
],
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
'INFOPLIST_FILE': '<(infoplist_file)',
},
}

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@ -1,142 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# Copied from Chromium's src/build/isolate.gypi
#
# It was necessary to copy this file because the path to build/common.gypi is
# different for the standalone and Chromium builds. Gyp doesn't permit
# conditional inclusion or variable expansion in include paths.
# http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
#
# Local modifications:
# * Removed include of '../chrome/version.gypi'.
# * Removed passing of version_full variable created in version.gypi:
# '--extra-variable', 'version_full=<(version_full)',
# This file is meant to be included into a target to provide a rule
# to "build" .isolate files into a .isolated file.
#
# To use this, create a gyp target with the following form:
# 'conditions': [
# ['test_isolation_mode != "noop"', {
# 'targets': [
# {
# 'target_name': 'foo_test_run',
# 'type': 'none',
# 'dependencies': [
# 'foo_test',
# ],
# 'includes': [
# '../build/isolate.gypi',
# 'foo_test.isolate',
# ],
# 'sources': [
# 'foo_test.isolate',
# ],
# },
# ],
# }],
# ],
#
# Note: foo_test.isolate is included and a source file. It is an inherent
# property of the .isolate format. This permits to define GYP variables but is
# a stricter format than GYP so isolate.py can read it.
#
# The generated .isolated file will be:
# <(PRODUCT_DIR)/foo_test.isolated
#
# See http://dev.chromium.org/developers/testing/isolated-testing/for-swes
# for more information.
{
'rules': [
{
'rule_name': 'isolate',
'extension': 'isolate',
'inputs': [
# Files that are known to be involved in this step.
'<(DEPTH)/tools/isolate_driver.py',
'<(DEPTH)/tools/swarming_client/isolate.py',
'<(DEPTH)/tools/swarming_client/run_isolated.py',
],
'outputs': [],
'action': [
'python',
'<(DEPTH)/tools/isolate_driver.py',
'<(test_isolation_mode)',
'--isolated', '<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated',
'--isolate', '<(RULE_INPUT_PATH)',
# Variables should use the -V FOO=<(FOO) form so frequent values,
# like '0' or '1', aren't stripped out by GYP. Run 'isolate.py help' for
# more details.
# Path variables are used to replace file paths when loading a .isolate
# file
'--path-variable', 'DEPTH', '<(DEPTH)',
'--path-variable', 'PRODUCT_DIR', '<(PRODUCT_DIR) ',
# Note: This list must match DefaultConfigVariables()
# in build/android/pylib/utils/isolator.py
'--config-variable', 'CONFIGURATION_NAME=<(CONFIGURATION_NAME)',
'--config-variable', 'OS=<(OS)',
'--config-variable', 'asan=<(asan)',
'--config-variable', 'branding=<(branding)',
'--config-variable', 'chromeos=<(chromeos)',
'--config-variable', 'component=<(component)',
'--config-variable', 'disable_nacl=<(disable_nacl)',
'--config-variable', 'enable_pepper_cdms=<(enable_pepper_cdms)',
'--config-variable', 'enable_plugins=<(enable_plugins)',
'--config-variable', 'fastbuild=<(fastbuild)',
'--config-variable', 'icu_use_data_file_flag=<(icu_use_data_file_flag)',
# TODO(kbr): move this to chrome_tests.gypi:gles2_conform_tests_run
# once support for user-defined config variables is added.
'--config-variable',
'internal_gles2_conform_tests=<(internal_gles2_conform_tests)',
'--config-variable', 'kasko=<(kasko)',
'--config-variable', 'lsan=<(lsan)',
'--config-variable', 'msan=<(msan)',
'--config-variable', 'target_arch=<(target_arch)',
'--config-variable', 'tsan=<(tsan)',
'--config-variable', 'use_custom_libcxx=<(use_custom_libcxx)',
'--config-variable', 'use_instrumented_libraries=<(use_instrumented_libraries)',
'--config-variable',
'use_prebuilt_instrumented_libraries=<(use_prebuilt_instrumented_libraries)',
'--config-variable', 'use_ozone=<(use_ozone)',
'--config-variable', 'use_x11=<(use_x11)',
'--config-variable', 'v8_use_external_startup_data=<(v8_use_external_startup_data)',
],
'conditions': [
# Note: When gyp merges lists, it appends them to the old value.
['OS=="mac"', {
'action': [
'--extra-variable', 'mac_product_name=<(mac_product_name)',
],
}],
["test_isolation_mode == 'prepare'", {
'outputs': [
'<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated.gen.json',
],
}, {
'outputs': [
'<(PRODUCT_DIR)/<(RULE_INPUT_ROOT).isolated',
],
}],
['OS=="win"', {
'action': [
'--config-variable', 'msvs_version=<(MSVS_VERSION)',
],
}, {
'action': [
'--config-variable', 'msvs_version=0',
],
}],
],
},
],
}

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@ -1,53 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': ['common.gypi',],
'variables': {
'merge_libs_dependencies': [
],
},
'targets': [
{
'target_name': 'no_op',
'type': 'executable',
'dependencies': [
'<@(merge_libs_dependencies)',
'../webrtc.gyp:webrtc',
'../p2p/p2p.gyp:rtc_p2p',
],
'sources': ['no_op.cc',],
},
{
'target_name': 'merged_lib',
'type': 'none',
'dependencies': [
'no_op',
],
'actions': [
{
'variables': {
'output_lib_name': 'webrtc_merged',
'output_lib': '<(PRODUCT_DIR)/<(STATIC_LIB_PREFIX)<(output_lib_name)<(STATIC_LIB_SUFFIX)',
},
'action_name': 'merge_libs',
'inputs': ['<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)no_op<(EXECUTABLE_SUFFIX)'],
'outputs': ['<(output_lib)'],
'action': ['python',
'merge_libs.py',
'<(PRODUCT_DIR)',
'<(output_lib)',],
},
],
},
],
# }],
# ],
# }],
# ],
}

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@ -1,48 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': ['common.gypi',],
'variables': {
'merge_libs_dependencies': [
],
},
'targets': [
{
'target_name': 'no_op_voice',
'type': 'executable',
'dependencies': [
'<@(merge_libs_dependencies)',
'../voice_engine/voice_engine.gyp:voice_engine'
],
'sources': ['no_op.cc',],
},
{
'target_name': 'merged_lib_voice',
'type': 'none',
'dependencies': [
'no_op_voice',
],
'actions': [
{
'variables': {
'output_lib_name': 'rtc_voice_merged',
'output_lib': '<(PRODUCT_DIR)/<(STATIC_LIB_PREFIX)<(output_lib_name)<(STATIC_LIB_SUFFIX)',
},
'action_name': 'merge_libs_voice',
'inputs': ['<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)no_op_voice<(EXECUTABLE_SUFFIX)'],
'outputs': ['<(output_lib)'],
'action': ['python',
'merge_libs.py',
'<(PRODUCT_DIR)',
'<(output_lib)',],
},
],
},
],
}

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@ -1,43 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [ 'common.gypi', ],
'targets': [
{
'target_name': 'no_op',
'type': 'executable',
'dependencies': [
'../voice_engine/voice_engine.gyp:voice_engine',
],
'sources': [ 'no_op.cc', ],
},
{
'target_name': 'merge_voice_libs',
'type': 'none',
'dependencies': [
'no_op',
],
'actions': [
{
'variables': {
'output_lib_name': 'webrtc_voice',
'output_lib': '<(PRODUCT_DIR)/<(STATIC_LIB_PREFIX)<(output_lib_name)<(STATIC_LIB_SUFFIX)',
},
'action_name': 'merge_libs',
'inputs': ['<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)no_op<(EXECUTABLE_SUFFIX)'],
'outputs': ['<(output_lib)'],
'action': ['python',
'merge_libs.py',
'<(PRODUCT_DIR)',
'<(output_lib)',],
},
],
},
],
}

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@ -1,21 +0,0 @@
# Copyright 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# ObjC target common prefix header.
{
'variables': {
'objc_prefix_file': '../sdk/objc/WebRTC-Prefix.pch',
},
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'YES',
'GCC_PREFIX_HEADER': '<(objc_prefix_file)',
'GCC_PRECOMPILE_PREFIX_HEADER': 'YES'
},
}

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@ -1,136 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# Copied from Chromium's src/build/protoc.gypi
#
# It was necessary to copy this file to WebRTC, because the path to
# build/common.gypi is different for the standalone and Chromium builds. Gyp
# doesn't permit conditional inclusion or variable expansion in include paths.
# http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
#
# Local changes:
# * Removed <(DEPTH) from include_dir due to difficulties with generated
# downstream code.
# This file is meant to be included into a target to provide a rule
# to invoke protoc in a consistent manner. For Java-targets, see
# protoc_java.gypi.
#
# To use this, create a gyp target with the following form:
# {
# 'target_name': 'my_proto_lib',
# 'type': 'static_library',
# 'sources': [
# 'foo.proto',
# 'bar.proto',
# ],
# 'variables': {
# # Optional, see below: 'proto_in_dir': '.'
# 'proto_out_dir': 'dir/for/my_proto_lib'
# },
# 'includes': ['path/to/this/gypi/file'],
# }
# If necessary, you may add normal .cc files to the sources list or other gyp
# dependencies. The proto headers are guaranteed to be generated before any
# source files, even within this target, are compiled.
#
# The 'proto_in_dir' variable must be the relative path to the
# directory containing the .proto files. If left out, it defaults to '.'.
#
# The 'proto_out_dir' variable specifies the path suffix that output
# files are generated under. Targets that gyp-depend on my_proto_lib
# will be able to include the resulting proto headers with an include
# like:
# #include "dir/for/my_proto_lib/foo.pb.h"
#
# If you need to add an EXPORT macro to a protobuf's c++ header, set the
# 'cc_generator_options' variable with the value: 'dllexport_decl=FOO_EXPORT:'
# e.g. 'dllexport_decl=BASE_EXPORT:'
#
# It is likely you also need to #include a file for the above EXPORT macro to
# work. You can do so with the 'cc_include' variable.
# e.g. 'base/base_export.h'
#
# Implementation notes:
# A proto_out_dir of foo/bar produces
# <(SHARED_INTERMEDIATE_DIR)/protoc_out/foo/bar/{file1,file2}.pb.{cc,h}
# <(SHARED_INTERMEDIATE_DIR)/pyproto/foo/bar/{file1,file2}_pb2.py
{
'variables': {
'protoc_wrapper': '<(DEPTH)/tools/protoc_wrapper/protoc_wrapper.py',
'cc_dir': '<(SHARED_INTERMEDIATE_DIR)/protoc_out/<(proto_out_dir)',
'py_dir': '<(PRODUCT_DIR)/pyproto/<(proto_out_dir)',
'cc_generator_options%': '',
'cc_include%': '',
'proto_in_dir%': '.',
'conditions': [
['use_system_protobuf==0', {
'protoc': '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
}, { # use_system_protobuf==1
'protoc': '<!(which protoc)',
}],
],
},
'rules': [
{
'rule_name': 'genproto',
'extension': 'proto',
'inputs': [
'<(protoc_wrapper)',
'<(protoc)',
],
'outputs': [
'<(py_dir)/<(RULE_INPUT_ROOT)_pb2.py',
'<(cc_dir)/<(RULE_INPUT_ROOT).pb.cc',
'<(cc_dir)/<(RULE_INPUT_ROOT).pb.h',
],
'action': [
'python',
'<(protoc_wrapper)',
'--protoc',
'<(protoc)',
# Using the --arg val form (instead of --arg=val) allows gyp's msvs rule
# generation to correct 'val' which is a path.
'--proto-in-dir','<(proto_in_dir)',
# Naively you'd use <(RULE_INPUT_PATH) here, but protoc requires
# --proto_path is a strict prefix of the path given as an argument.
'--cc-out-dir', '<(cc_generator_options)<(cc_dir)',
'--py-out-dir', '<(py_dir)',
'<(RULE_INPUT_ROOT)<(RULE_INPUT_EXT)',
],
'message': 'Generating C++ and Python code from <(RULE_INPUT_PATH)',
'process_outputs_as_sources': 1,
},
],
'include_dirs': [
'<(SHARED_INTERMEDIATE_DIR)/protoc_out',
],
'direct_dependent_settings': {
'include_dirs': [
'<(SHARED_INTERMEDIATE_DIR)/protoc_out',
]
},
# This target exports a hard dependency because it generates header
# files.
'hard_dependency': 1,
'conditions': [
['build_protobuf==1', {
'dependencies': [
'<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
'<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
],
'export_dependent_settings': [
# The generated headers reference headers within protobuf_lite,
# so dependencies must be able to find those headers too.
'<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
],
}],
],
}

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@ -1,27 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'webrtc_call_dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:congestion_controller',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/webrtc.gyp:rtc_event_log_impl',
],
'webrtc_call_sources': [
'call/audio_receive_stream.h',
'call/audio_send_stream_call.cc',
'call/bitrate_allocator.cc',
'call/call.h',
'call/call.cc',
'call/flexfec_receive_stream.h',
'call/flexfec_receive_stream_impl.cc',
],
},
}

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@ -1,24 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': ['build/common.gypi'],
'targets': [
{
'target_name': 'webrtc_common',
'type': 'static_library',
'sources': [
'common_types.cc',
'common_types.h',
'config.h',
'config.cc',
'engine_configurations.h',
'typedefs.h',
],
},
],
}

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@ -1,235 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../build/common.gypi',
],
'targets': [
{
'target_name': 'common_audio',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
'resampler/include',
'signal_processing/include',
],
'direct_dependent_settings': {
'include_dirs': [
'resampler/include',
'signal_processing/include',
'vad/include',
],
},
'sources': [
'audio_converter.cc',
'audio_converter.h',
'audio_ring_buffer.cc',
'audio_ring_buffer.h',
'audio_util.cc',
'blocker.cc',
'blocker.h',
'channel_buffer.cc',
'channel_buffer.h',
'fft4g.c',
'fft4g.h',
'fir_filter.cc',
'fir_filter.h',
'fir_filter_neon.h',
'fir_filter_sse.h',
'include/audio_util.h',
'lapped_transform.cc',
'lapped_transform.h',
'real_fourier.cc',
'real_fourier.h',
'real_fourier_ooura.cc',
'real_fourier_ooura.h',
'resampler/include/push_resampler.h',
'resampler/include/resampler.h',
'resampler/push_resampler.cc',
'resampler/push_sinc_resampler.cc',
'resampler/push_sinc_resampler.h',
'resampler/resampler.cc',
'resampler/sinc_resampler.cc',
'resampler/sinc_resampler.h',
'ring_buffer.c',
'ring_buffer.h',
'signal_processing/include/real_fft.h',
'signal_processing/include/signal_processing_library.h',
'signal_processing/include/spl_inl.h',
'signal_processing/auto_corr_to_refl_coef.c',
'signal_processing/auto_correlation.c',
'signal_processing/complex_fft.c',
'signal_processing/complex_fft_tables.h',
'signal_processing/complex_bit_reverse.c',
'signal_processing/copy_set_operations.c',
'signal_processing/cross_correlation.c',
'signal_processing/division_operations.c',
'signal_processing/dot_product_with_scale.c',
'signal_processing/downsample_fast.c',
'signal_processing/energy.c',
'signal_processing/filter_ar.c',
'signal_processing/filter_ar_fast_q12.c',
'signal_processing/filter_ma_fast_q12.c',
'signal_processing/get_hanning_window.c',
'signal_processing/get_scaling_square.c',
'signal_processing/ilbc_specific_functions.c',
'signal_processing/levinson_durbin.c',
'signal_processing/lpc_to_refl_coef.c',
'signal_processing/min_max_operations.c',
'signal_processing/randomization_functions.c',
'signal_processing/refl_coef_to_lpc.c',
'signal_processing/real_fft.c',
'signal_processing/resample.c',
'signal_processing/resample_48khz.c',
'signal_processing/resample_by_2.c',
'signal_processing/resample_by_2_internal.c',
'signal_processing/resample_by_2_internal.h',
'signal_processing/resample_fractional.c',
'signal_processing/spl_init.c',
'signal_processing/spl_inl.c',
'signal_processing/spl_sqrt.c',
'signal_processing/spl_sqrt_floor.c',
'signal_processing/splitting_filter.c',
'signal_processing/sqrt_of_one_minus_x_squared.c',
'signal_processing/vector_scaling_operations.c',
'smoothing_filter.cc',
'smoothing_filter.h',
'sparse_fir_filter.cc',
'sparse_fir_filter.h',
'vad/include/vad.h',
'vad/vad.cc',
'vad/webrtc_vad.c',
'vad/vad_core.c',
'vad/vad_core.h',
'vad/vad_filterbank.c',
'vad/vad_filterbank.h',
'vad/vad_gmm.c',
'vad/vad_gmm.h',
'vad/vad_sp.c',
'vad/vad_sp.h',
'wav_header.cc',
'wav_header.h',
'wav_file.cc',
'wav_file.h',
'window_generator.cc',
'window_generator.h',
],
'conditions': [
#TODO: not defined
#['rtc_use_openmax_dl==1', {
# 'sources': [
# 'real_fourier_openmax.cc',
# 'real_fourier_openmax.h',
# ],
# 'defines': ['RTC_USE_OPENMAX_DL',],
# 'conditions': [
# ['build_openmax_dl==1', {
# 'dependencies': ['<(DEPTH)/third_party/openmax_dl/dl/dl.gyp:openmax_dl',],
# }],
# ],
#}],
['target_arch=="ia32" or target_arch=="x64"', {
'dependencies': ['common_audio_sse2',],
}],
['build_with_neon==1', {
'dependencies': ['common_audio_neon',],
}],
['target_arch=="arm"', {
'sources': [
'signal_processing/complex_bit_reverse_arm.S',
'signal_processing/spl_sqrt_floor_arm.S',
],
'sources!': [
'signal_processing/complex_bit_reverse.c',
'signal_processing/spl_sqrt_floor.c',
],
'conditions': [
['arm_version>=7', {
'sources': [
'signal_processing/filter_ar_fast_q12_armv7.S',
],
'sources!': [
'signal_processing/filter_ar_fast_q12.c',
],
}],
], # conditions
}],
['target_arch=="mipsel" and mips_arch_variant!="r6"', {
'sources': [
'signal_processing/include/spl_inl_mips.h',
'signal_processing/complex_bit_reverse_mips.c',
'signal_processing/complex_fft_mips.c',
'signal_processing/cross_correlation_mips.c',
'signal_processing/downsample_fast_mips.c',
'signal_processing/filter_ar_fast_q12_mips.c',
'signal_processing/min_max_operations_mips.c',
'signal_processing/resample_by_2_mips.c',
'signal_processing/spl_sqrt_floor_mips.c',
],
'sources!': [
'signal_processing/complex_bit_reverse.c',
'signal_processing/complex_fft.c',
'signal_processing/filter_ar_fast_q12.c',
'signal_processing/spl_sqrt_floor.c',
],
'conditions': [
['mips_dsp_rev>0', {
'sources': [
'signal_processing/vector_scaling_operations_mips.c',
],
}],
],
}],
], # conditions
# Ignore warning on shift operator promotion.
'msvs_disabled_warnings': [ 4334, ],
},
], # targets
'conditions': [
['target_arch=="ia32" or target_arch=="x64"', {
'targets': [
{
'target_name': 'common_audio_sse2',
'type': 'static_library',
'sources': [
'fir_filter_sse.cc',
'resampler/sinc_resampler_sse.cc',
],
'conditions': [
['os_posix==1', {
'cflags': [ '-msse2', ],
'cflags_mozilla': ['-msse2',],
'xcode_settings': {
'OTHER_CFLAGS': [ '-msse2', ],
},
}],
],
},
], # targets
}],
['build_with_neon==1', {
'targets': [
{
'target_name': 'common_audio_neon',
'type': 'static_library',
'includes': ['../build/arm_neon.gypi',],
'sources': [
'fir_filter_neon.cc',
'resampler/sinc_resampler_neon.cc',
'signal_processing/cross_correlation_neon.c',
'signal_processing/downsample_fast_neon.c',
'signal_processing/min_max_operations_neon.c',
],
},
], # targets
}],
], # conditions
}

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@ -1,83 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': ['../build/common.gypi'],
'targets': [
{
'target_name': 'common_video',
'type': 'static_library',
'include_dirs': [
'<(webrtc_root)/modules/interface/',
'include',
'libyuv/include',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_task_queue',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
'libyuv/include',
],
},
'conditions': [
['build_libyuv==1', {
'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',],
'export_dependent_settings': [
'<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',
],
}, {
# Need to add a directory normally exported by libyuv.gyp.
'include_dirs': ['<(libyuv_dir)/include',],
}],
['OS=="ios" or OS=="mac"', {
'sources': [
'corevideo_frame_buffer.cc',
'include/corevideo_frame_buffer.h',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreVideo',
],
},
},
}],
],
'sources': [
'bitrate_adjuster.cc',
'h264/sps_vui_rewriter.cc',
'h264/sps_vui_rewriter.h',
'h264/h264_common.cc',
'h264/h264_common.h',
'h264/profile_level_id.cc',
'h264/pps_parser.cc',
'h264/pps_parser.h',
'h264/sps_parser.cc',
'h264/sps_parser.h',
'i420_buffer_pool.cc',
'video_frame.cc',
'incoming_video_stream.cc',
'include/bitrate_adjuster.h',
'include/frame_callback.h',
'include/i420_buffer_pool.h',
'include/incoming_video_stream.h',
'include/video_bitrate_allocator.h',
'include/video_frame_buffer.h',
'libyuv/include/webrtc_libyuv.h',
'libyuv/webrtc_libyuv.cc',
'video_frame_buffer.cc',
'video_render_frames.cc',
'video_render_frames.h',
],
},
], # targets
}

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@ -1,219 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [ '../build/common.gypi', ],
'targets': [
{
'target_name': 'mozilla_rtc_media',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'direct_dependent_settings': {
'include_dirs': [
'<(libyuv_dir)/include',
],
},
'sources': [
'base/videoadapter.cc',
'base/videoadapter.h',
'base/videobroadcaster.cc',
'base/videobroadcaster.h',
'base/videosourcebase.cc',
'base/videosourcebase.h',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
'-Wno-deprecated-declarations',
],
'cflags!': [
'-Wextra',
],
'cflags_cc!': [
'-Woverloaded-virtual',
],
},
{
'target_name': 'rtc_media',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/webrtc.gyp:webrtc_lib',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/p2p/p2p.gyp:rtc_p2p',
],
'direct_dependent_settings': {
'include_dirs': [
'<(libyuv_dir)/include',
],
},
'sources': [
'base/adaptedvideotracksource.cc',
'base/adaptedvideotracksource.h',
'base/audiosource.h',
'base/codec.cc',
'base/codec.h',
'base/cpuid.cc',
'base/cpuid.h',
'base/cryptoparams.h',
'base/device.h',
'base/hybriddataengine.h',
'base/mediachannel.h',
'base/mediacommon.h',
'base/mediaconstants.cc',
'base/mediaconstants.h',
'base/mediaengine.cc',
'base/mediaengine.h',
'base/rtpdataengine.cc',
'base/rtpdataengine.h',
'base/rtpdump.cc',
'base/rtpdump.h',
'base/rtputils.cc',
'base/rtputils.h',
'base/screencastid.h',
'base/streamparams.cc',
'base/streamparams.h',
'base/turnutils.cc',
'base/turnutils.h',
'base/videoadapter.cc',
'base/videoadapter.h',
'base/videobroadcaster.cc',
'base/videobroadcaster.h',
'base/videocapturer.cc',
'base/videocapturer.h',
'base/videocapturerfactory.h',
'base/videocommon.cc',
'base/videocommon.h',
'base/videoframe.cc',
'base/videoframe.h',
'base/videosourcebase.cc',
'base/videosourcebase.h',
'devices/videorendererfactory.h',
'engine/internaldecoderfactory.cc',
'engine/internaldecoderfactory.h',
'engine/internalencoderfactory.cc',
'engine/internalencoderfactory.h',
'engine/nullwebrtcvideoengine.h',
'engine/payload_type_mapper.cc',
'engine/payload_type_mapper.h',
'engine/simulcast.cc',
'engine/simulcast.h',
'engine/videodecodersoftwarefallbackwrapper.cc',
'engine/videodecodersoftwarefallbackwrapper.h',
'engine/videoencodersoftwarefallbackwrapper.cc',
'engine/videoencodersoftwarefallbackwrapper.h',
'engine/webrtccommon.h',
'engine/webrtcmediaengine.cc',
'engine/webrtcmediaengine.h',
'engine/webrtcmediaengine.cc',
'engine/webrtcvideocapturer.cc',
'engine/webrtcvideocapturer.h',
'engine/webrtcvideocapturerfactory.h',
'engine/webrtcvideocapturerfactory.cc',
'engine/webrtcvideodecoderfactory.h',
'engine/webrtcvideoencoderfactory.h',
'engine/webrtcvideoengine2.cc',
'engine/webrtcvideoengine2.h',
'engine/webrtcvideoframe.cc',
'engine/webrtcvideoframe.h',
'engine/webrtcvoe.h',
'engine/webrtcvoiceengine.cc',
'engine/webrtcvoiceengine.h',
'sctp/sctptransportinternal.h',
# 'sctp/sctpdataengine.cc',
# 'sctp/sctpdataengine.h',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
'-Wno-deprecated-declarations',
],
'cflags!': [
'-Wextra',
],
'cflags_cc!': [
'-Woverloaded-virtual',
],
'msvs_disabled_warnings': [
4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch.
4267, # conversion from 'size_t' to 'int', possible loss of data.
4389, # signed/unsigned mismatch.
],
'conditions': [
['build_libyuv==1', {
'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv',],
}],
#TODO: build_usrsctp not defined
#['build_usrsctp==1', {
# 'include_dirs': [
# # TODO(jiayl): move this into the direct_dependent_settings of
# # usrsctp.gyp.
# '<(DEPTH)/third_party/usrsctp/usrsctplib',
# ],
# 'dependencies': [
# '<(DEPTH)/third_party/usrsctp/usrsctp.gyp:usrsctplib',
# ],
#}],
['enable_intelligibility_enhancer==1', {
'defines': ['WEBRTC_INTELLIGIBILITY_ENHANCER=1',],
}, {
'defines': ['WEBRTC_INTELLIGIBILITY_ENHANCER=0',],
}],
['build_with_chromium==1', {
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:video_capture',
],
}, {
'defines': [
'HAVE_WEBRTC_VIDEO',
'HAVE_WEBRTC_VOICE',
],
'direct_dependent_settings': {
'defines': [
'HAVE_WEBRTC_VIDEO',
'HAVE_WEBRTC_VOICE',
],
},
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
],
}],
['OS=="linux" and use_gtk==1', {
'sources': [
'devices/gtkvideorenderer.cc',
'devices/gtkvideorenderer.h',
],
'cflags': [
'<!@(pkg-config --cflags gobject-2.0 gthread-2.0 gtk+-2.0)',
],
}],
['OS=="win"', {
'sources': [
'devices/gdivideorenderer.cc',
'devices/gdivideorenderer.h',
],
'msvs_settings': {
'VCLibrarianTool': {
'AdditionalDependencies': [
'd3d9.lib',
'gdi32.lib',
'strmiids.lib',
],
},
},
}],
],
}, # target rtc_media
], # targets.
}

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@ -1,257 +0,0 @@
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../build/common.gypi',
'audio_network_adaptor/audio_network_adaptor.gypi',
'codecs/interfaces.gypi',
'codecs/cng/cng.gypi',
'codecs/g711/g711.gypi',
'codecs/g722/g722.gypi',
'codecs/ilbc/ilbc.gypi',
'codecs/isac/isac.gypi',
'codecs/isac/isac_common.gypi',
'codecs/isac/isacfix.gypi',
'codecs/pcm16b/pcm16b.gypi',
'codecs/red/red.gypi',
'neteq/neteq.gypi',
],
'variables': {
'variables': {
'audio_codec_dependencies': [
'cng',
'g711',
'pcm16b',
],
'audio_codec_defines': [],
'conditions': [
['include_g722==1', {
'audio_coding_dependencies': ['g722',],
'audio_coding_defines': ['WEBRTC_CODEC_G722',],
}],
['include_isac==1', {
'audio_coding_dependencies': ['isac',],
'audio_coding_defines': ['WEBRTC_CODEC_ISAC',],
}],
['include_ilbc==1', {
'audio_codec_dependencies': ['ilbc',],
'audio_codec_defines': ['WEBRTC_CODEC_ILBC',],
}],
['include_opus==1', {
'audio_codec_dependencies': ['webrtc_opus',],
'audio_codec_defines': ['WEBRTC_CODEC_OPUS',],
'conditions': [
['rtc_opus_variable_complexity==1', {
'audio_codec_defines': [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
},{
'audio_codec_defines': [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
}],
],
}],
['build_with_mozilla==0', {
'conditions': [
['target_arch=="arm"', {
'audio_codec_dependencies': ['isac_fix',],
'audio_codec_defines': ['WEBRTC_CODEC_ISACFX',],
}, {
'audio_codec_dependencies': ['isac',],
'audio_codec_defines': ['WEBRTC_CODEC_ISAC',],
}],
],
'audio_codec_dependencies': ['g722',],
'audio_codec_defines': ['WEBRTC_CODEC_G722',],
}],
['build_with_mozilla==0 and build_with_chromium==0', {
'audio_codec_dependencies': ['red',],
'audio_codec_defines': ['WEBRTC_CODEC_RED',],
}],
['build_with_mozilla==1', {
'audio_codec_dependencies': ['g722',],
'audio_codec_defines': ['WEBRTC_CODEC_G722',],
}],
],
},
'audio_codec_dependencies': '<(audio_codec_dependencies)',
'audio_codec_defines': '<(audio_codec_defines)',
'audio_coding_dependencies': [
'<@(audio_codec_dependencies)',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'audio_coding_defines': '<(audio_codec_defines)',
},
'targets': [
{
'target_name': 'audio_decoder_factory_interface',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'audio_format',
],
'include_dirs': [
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'<(webrtc_root)',
],
},
'sources': [
'codecs/audio_decoder_factory.h',
],
},
{
'target_name': 'audio_format',
'type': 'static_library',
'defines': [
'<@(audio_codec_defines)',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
],
'include_dirs': [
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'<(webrtc_root)',
],
},
'sources': [
'codecs/audio_format.cc',
'codecs/audio_format.h',
],
},
{
'target_name': 'audio_format_conversion',
'type': 'static_library',
'defines': [
'<@(audio_codec_defines)',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'audio_format',
],
'include_dirs': [
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'<(webrtc_root)',
],
},
'sources': [
'codecs/audio_format_conversion.cc',
'codecs/audio_format_conversion.h',
],
},
{
'target_name': 'builtin_audio_decoder_factory',
'type': 'static_library',
'defines': [
'<@(audio_codec_defines)',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<@(audio_codec_dependencies)',
'audio_decoder_factory_interface',
],
'include_dirs': [
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'<(webrtc_root)',
],
},
'sources': [
'codecs/builtin_audio_decoder_factory.cc',
'codecs/builtin_audio_decoder_factory.h',
],
},
{
'target_name': 'rent_a_codec',
'type': 'static_library',
'defines': [
'<@(audio_codec_defines)',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<@(audio_codec_dependencies)',
],
'include_dirs': [
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'<(webrtc_root)',
],
},
'sources': [
'acm2/acm_codec_database.cc',
'acm2/acm_codec_database.h',
'acm2/rent_a_codec.cc',
'acm2/rent_a_codec.h',
],
},
{
'target_name': 'audio_coding_module',
'type': 'static_library',
'defines': [
'<@(audio_coding_defines)',
],
'dependencies': [
'<@(audio_coding_dependencies)',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
'audio_network_adaptor',
'neteq',
'rent_a_codec',
'audio_format_conversion',
],
'include_dirs': [
'include',
'../include',
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
'../include',
'<(webrtc_root)',
],
},
'conditions': [
['include_opus==1', {
'export_dependent_settings': ['webrtc_opus'],
}],
],
'sources': [
'acm2/acm_common_defs.h',
'acm2/acm_receiver.cc',
'acm2/acm_receiver.h',
'acm2/acm_resampler.cc',
'acm2/acm_resampler.h',
'acm2/audio_coding_module.cc',
'acm2/call_statistics.cc',
'acm2/call_statistics.h',
'acm2/codec_manager.cc',
'acm2/codec_manager.h',
'include/audio_coding_module.h',
'include/audio_coding_module_typedefs.h',
],
},
],
'conditions': [
['include_opus==1', {
'includes': ['codecs/opus/opus.gypi',],
}],
],
}

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@ -1,42 +0,0 @@
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../build/common.gypi',
'codecs/isac/isac_test.gypi',
'codecs/isac/isacfix_test.gypi',
],
'targets': [
{
'target_name': 'audio_codec_speed_tests',
'type': '<(gtest_target_type)',
'dependencies': [
'audio_processing',
'isac_fix',
'webrtc_opus',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support_main',
],
'sources': [
'codecs/isac/fix/test/isac_speed_test.cc',
'codecs/opus/opus_speed_test.cc',
'codecs/tools/audio_codec_speed_test.h',
'codecs/tools/audio_codec_speed_test.cc',
],
'conditions': [
['OS=="android"', {
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
],
}],
],
},
],
}

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@ -1,77 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{ 'target_name': 'audio_network_adaptor',
'type': 'static_library',
'sources': [
'audio_network_adaptor.cc',
'audio_network_adaptor_impl.cc',
'audio_network_adaptor_impl.h',
'bitrate_controller.h',
'bitrate_controller.cc',
'channel_controller.cc',
'channel_controller.h',
'controller.h',
'controller.cc',
'controller_manager.cc',
'controller_manager.h',
'debug_dump_writer.cc',
'debug_dump_writer.h',
'dtx_controller.h',
'dtx_controller.cc',
'fec_controller.h',
'fec_controller.cc',
'frame_length_controller.cc',
'frame_length_controller.h',
'include/audio_network_adaptor.h',
], # sources
'conditions': [
['enable_protobuf==1', {
'dependencies': [
'ana_config_proto',
'ana_debug_dump_proto',
],
'defines': ['WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP'],
}],
], # conditions
},
], # targets
'conditions': [
['enable_protobuf==1', {
'targets': [
{ 'target_name': 'ana_debug_dump_proto',
'type': 'static_library',
'sources': ['debug_dump.proto',],
'variables': {
'proto_in_dir': '.',
# Workaround to protect against gyp's pathname relativization when
# this file is included by modules.gyp.
'proto_out_protected': 'webrtc/modules/audio_coding/audio_network_adaptor',
'proto_out_dir': '<(proto_out_protected)',
},
'includes': ['../../../build/protoc.gypi',],
},
{ 'target_name': 'ana_config_proto',
'type': 'static_library',
'sources': ['config.proto',],
'variables': {
'proto_in_dir': '.',
# Workaround to protect against gyp's pathname relativization when
# this file is included by modules.gyp.
'proto_out_protected': 'webrtc/modules/audio_coding/audio_network_adaptor',
'proto_out_dir': '<(proto_out_protected)',
},
'includes': ['../../../build/protoc.gypi',],
},
], # targets
}],
], # conditions
}

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@ -1,26 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'cng',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'audio_encoder_interface',
],
'sources': [
'audio_encoder_cng.cc',
'audio_encoder_cng.h',
'webrtc_cng.cc',
'webrtc_cng.h',
],
},
], # targets
}

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@ -1,30 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'g711',
'type': 'static_library',
'dependencies': [
'audio_encoder_interface',
'audio_decoder_interface',
],
'sources': [
'audio_decoder_pcm.cc',
'audio_decoder_pcm.h',
'audio_encoder_pcm.cc',
'audio_encoder_pcm.h',
'g711_interface.c',
'g711_interface.h',
'g711.c',
'g711.h',
],
},
], # targets
}

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@ -1,30 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'g722',
'type': 'static_library',
'dependencies': [
'audio_encoder_interface',
'audio_decoder_interface',
],
'sources': [
'audio_decoder_g722.cc',
'audio_decoder_g722.h',
'audio_encoder_g722.cc',
'audio_encoder_g722.h',
'g722_interface.c',
'g722_interface.h',
'g722_decode.c',
'g722_enc_dec.h',
'g722_encode.c',
],
},
], # targets
}

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@ -1,166 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'ilbc',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'audio_encoder_interface',
],
'sources': [
'abs_quant.c',
'abs_quant_loop.c',
'audio_decoder_ilbc.cc',
'audio_decoder_ilbc.h',
'audio_encoder_ilbc.cc',
'audio_encoder_ilbc.h',
'augmented_cb_corr.c',
'bw_expand.c',
'cb_construct.c',
'cb_mem_energy.c',
'cb_mem_energy_augmentation.c',
'cb_mem_energy_calc.c',
'cb_search.c',
'cb_search_core.c',
'cb_update_best_index.c',
'chebyshev.c',
'comp_corr.c',
'constants.c',
'create_augmented_vec.c',
'decode.c',
'decode_residual.c',
'decoder_interpolate_lsf.c',
'do_plc.c',
'encode.c',
'energy_inverse.c',
'enh_upsample.c',
'enhancer.c',
'enhancer_interface.c',
'filtered_cb_vecs.c',
'frame_classify.c',
'gain_dequant.c',
'gain_quant.c',
'get_cd_vec.c',
'get_lsp_poly.c',
'get_sync_seq.c',
'hp_input.c',
'hp_output.c',
'ilbc.c',
'ilbc.h',
'index_conv_dec.c',
'index_conv_enc.c',
'init_decode.c',
'init_encode.c',
'interpolate.c',
'interpolate_samples.c',
'lpc_encode.c',
'lsf_check.c',
'lsf_interpolate_to_poly_dec.c',
'lsf_interpolate_to_poly_enc.c',
'lsf_to_lsp.c',
'lsf_to_poly.c',
'lsp_to_lsf.c',
'my_corr.c',
'nearest_neighbor.c',
'pack_bits.c',
'poly_to_lsf.c',
'poly_to_lsp.c',
'refiner.c',
'simple_interpolate_lsf.c',
'simple_lpc_analysis.c',
'simple_lsf_dequant.c',
'simple_lsf_quant.c',
'smooth.c',
'smooth_out_data.c',
'sort_sq.c',
'split_vq.c',
'state_construct.c',
'state_search.c',
'swap_bytes.c',
'unpack_bits.c',
'vq3.c',
'vq4.c',
'window32_w32.c',
'xcorr_coef.c',
'abs_quant.h',
'abs_quant_loop.h',
'augmented_cb_corr.h',
'bw_expand.h',
'cb_construct.h',
'cb_mem_energy.h',
'cb_mem_energy_augmentation.h',
'cb_mem_energy_calc.h',
'cb_search.h',
'cb_search_core.h',
'cb_update_best_index.h',
'chebyshev.h',
'comp_corr.h',
'constants.h',
'create_augmented_vec.h',
'decode.h',
'decode_residual.h',
'decoder_interpolate_lsf.h',
'do_plc.h',
'encode.h',
'energy_inverse.h',
'enh_upsample.h',
'enhancer.h',
'enhancer_interface.h',
'filtered_cb_vecs.h',
'frame_classify.h',
'gain_dequant.h',
'gain_quant.h',
'get_cd_vec.h',
'get_lsp_poly.h',
'get_sync_seq.h',
'hp_input.h',
'hp_output.h',
'defines.h',
'index_conv_dec.h',
'index_conv_enc.h',
'init_decode.h',
'init_encode.h',
'interpolate.h',
'interpolate_samples.h',
'lpc_encode.h',
'lsf_check.h',
'lsf_interpolate_to_poly_dec.h',
'lsf_interpolate_to_poly_enc.h',
'lsf_to_lsp.h',
'lsf_to_poly.h',
'lsp_to_lsf.h',
'my_corr.h',
'nearest_neighbor.h',
'pack_bits.h',
'poly_to_lsf.h',
'poly_to_lsp.h',
'refiner.h',
'simple_interpolate_lsf.h',
'simple_lpc_analysis.h',
'simple_lsf_dequant.h',
'simple_lsf_quant.h',
'smooth.h',
'smooth_out_data.h',
'sort_sq.h',
'split_vq.h',
'state_construct.h',
'state_search.h',
'swap_bytes.h',
'unpack_bits.h',
'vq3.h',
'vq4.h',
'window32_w32.h',
'xcorr_coef.h',
], # sources
}, # ilbc
], # targets
}

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@ -1,39 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'audio_decoder_interface',
'type': 'static_library',
'sources': [
'audio_decoder.cc',
'audio_decoder.h',
'legacy_encoded_audio_frame.cc',
'legacy_encoded_audio_frame.h',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
],
},
{
'target_name': 'audio_encoder_interface',
'type': 'static_library',
'sources': [
'audio_encoder.cc',
'audio_encoder.h',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
],
},
],
}

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@ -1,95 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'isac',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'audio_decoder_interface',
'audio_encoder_interface',
'isac_common',
],
'include_dirs': [
'main/include',
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'main/include',
'<(webrtc_root)',
],
},
'sources': [
'main/include/audio_decoder_isac.h',
'main/include/audio_encoder_isac.h',
'main/include/isac.h',
'main/source/arith_routines.c',
'main/source/arith_routines_hist.c',
'main/source/arith_routines_logist.c',
'main/source/audio_decoder_isac.cc',
'main/source/audio_encoder_isac.cc',
'main/source/bandwidth_estimator.c',
'main/source/crc.c',
'main/source/decode.c',
'main/source/decode_bwe.c',
'main/source/encode.c',
'main/source/encode_lpc_swb.c',
'main/source/entropy_coding.c',
'main/source/fft.c',
'main/source/filter_functions.c',
'main/source/filterbank_tables.c',
'main/source/intialize.c',
'main/source/isac.c',
'main/source/isac_float_type.h',
'main/source/filterbanks.c',
'main/source/pitch_lag_tables.c',
'main/source/lattice.c',
'main/source/lpc_gain_swb_tables.c',
'main/source/lpc_analysis.c',
'main/source/lpc_shape_swb12_tables.c',
'main/source/lpc_shape_swb16_tables.c',
'main/source/lpc_tables.c',
'main/source/pitch_estimator.c',
'main/source/pitch_filter.c',
'main/source/pitch_gain_tables.c',
'main/source/spectrum_ar_model_tables.c',
'main/source/transform.c',
'main/source/arith_routines.h',
'main/source/bandwidth_estimator.h',
'main/source/codec.h',
'main/source/crc.h',
'main/source/encode_lpc_swb.h',
'main/source/entropy_coding.h',
'main/source/fft.h',
'main/source/filterbank_tables.h',
'main/source/lpc_gain_swb_tables.h',
'main/source/lpc_analysis.h',
'main/source/lpc_shape_swb12_tables.h',
'main/source/lpc_shape_swb16_tables.h',
'main/source/lpc_tables.h',
'main/source/pitch_estimator.h',
'main/source/pitch_gain_tables.h',
'main/source/pitch_lag_tables.h',
'main/source/settings.h',
'main/source/spectrum_ar_model_tables.h',
'main/source/structs.h',
'main/source/os_specific_inline.h',
],
'conditions': [
['OS=="linux"', {
'link_settings': {
'libraries': ['-lm',],
},
}],
],
},
],
}

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@ -1,22 +0,0 @@
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'isac_common',
'type': 'static_library',
'sources': [
'audio_encoder_isac_t.h',
'audio_encoder_isac_t_impl.h',
'locked_bandwidth_info.cc',
'locked_bandwidth_info.h',
],
},
],
}

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@ -1,83 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
# simple kenny
{
'target_name': 'isac_test',
'type': 'executable',
'dependencies': [
'isac',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'include_dirs': [
'./main/include',
'./main/test',
'./main/util',
'<(webrtc_root)',
],
'sources': [
'empty.cc', # force build system to use C++ linker
'./main/test/simpleKenny.c',
'./main/util/utility.c',
],
'conditions': [
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-format',
],
},
},
}],
], # conditions.
},
# ReleaseTest-API
{
'target_name': 'isac_api_test',
'type': 'executable',
'dependencies': [
'isac',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'include_dirs': [
'./main/test',
'./main/include',
'./main/util',
'<(webrtc_root)',
],
'sources': [
'./main/test/ReleaseTest-API/ReleaseTest-API.cc',
'./main/util/utility.c',
],
},
# SwitchingSampRate
{
'target_name': 'isac_switch_samprate_test',
'type': 'executable',
'dependencies': [
'isac',
],
'include_dirs': [
'./main/test',
'./main/include',
'../../../../common_audio/signal_processing/include',
'./main/util',
'<(webrtc_root)',
],
'sources': [
'./main/test/SwitchingSampRate/SwitchingSampRate.cc',
'./main/util/utility.c',
],
},
],
}

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@ -1,149 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../../build/common.gypi',
],
'targets': [
{
'target_name': 'isac_fix',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'isac_common',
],
'include_dirs': [
'fix/include',
'<(webrtc_root)'
],
'direct_dependent_settings': {
'include_dirs': [
'fix/include',
'<(webrtc_root)',
],
},
'sources': [
'fix/include/audio_decoder_isacfix.h',
'fix/include/audio_encoder_isacfix.h',
'fix/include/isacfix.h',
'fix/source/arith_routines.c',
'fix/source/arith_routines_hist.c',
'fix/source/arith_routines_logist.c',
'fix/source/audio_decoder_isacfix.cc',
'fix/source/audio_encoder_isacfix.cc',
'fix/source/bandwidth_estimator.c',
'fix/source/decode.c',
'fix/source/decode_bwe.c',
'fix/source/decode_plc.c',
'fix/source/encode.c',
'fix/source/entropy_coding.c',
'fix/source/fft.c',
'fix/source/filterbank_tables.c',
'fix/source/filterbanks.c',
'fix/source/filters.c',
'fix/source/initialize.c',
'fix/source/isac_fix_type.h',
'fix/source/isacfix.c',
'fix/source/lattice.c',
'fix/source/lattice_c.c',
'fix/source/lpc_masking_model.c',
'fix/source/lpc_tables.c',
'fix/source/pitch_estimator.c',
'fix/source/pitch_estimator_c.c',
'fix/source/pitch_filter.c',
'fix/source/pitch_filter_c.c',
'fix/source/pitch_gain_tables.c',
'fix/source/pitch_lag_tables.c',
'fix/source/spectrum_ar_model_tables.c',
'fix/source/transform.c',
'fix/source/transform_tables.c',
'fix/source/arith_routins.h',
'fix/source/bandwidth_estimator.h',
'fix/source/codec.h',
'fix/source/entropy_coding.h',
'fix/source/fft.h',
'fix/source/filterbank_tables.h',
'fix/source/lpc_masking_model.h',
'fix/source/lpc_tables.h',
'fix/source/pitch_estimator.h',
'fix/source/pitch_gain_tables.h',
'fix/source/pitch_lag_tables.h',
'fix/source/settings.h',
'fix/source/spectrum_ar_model_tables.h',
'fix/source/structs.h',
],
'conditions': [
['target_arch=="arm" and arm_version>=7', {
'sources': [
'fix/source/lattice_armv7.S',
'fix/source/pitch_filter_armv6.S',
],
'sources!': [
'fix/source/lattice_c.c',
'fix/source/pitch_filter_c.c',
],
}],
['build_with_neon==1', {
'dependencies': ['isac_neon', ],
}],
['target_arch=="mipsel" and mips_arch_variant!="r6"', {
'sources': [
'fix/source/entropy_coding_mips.c',
'fix/source/filters_mips.c',
'fix/source/lattice_mips.c',
'fix/source/pitch_estimator_mips.c',
'fix/source/transform_mips.c',
],
'sources!': [
'fix/source/lattice_c.c',
'fix/source/pitch_estimator_c.c',
],
'conditions': [
['mips_dsp_rev>0', {
'sources': [
'fix/source/filterbanks_mips.c',
],
}],
['mips_dsp_rev>1', {
'sources': [
'fix/source/lpc_masking_model_mips.c',
'fix/source/pitch_filter_mips.c',
],
'sources!': [
'fix/source/pitch_filter_c.c',
],
}],
],
}],
],
},
],
'conditions': [
['build_with_neon==1', {
'targets': [
{
'target_name': 'isac_neon',
'type': 'static_library',
'includes': ['../../../../build/arm_neon.gypi',],
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'fix/source/entropy_coding_neon.c',
'fix/source/filterbanks_neon.c',
'fix/source/filters_neon.c',
'fix/source/lattice_neon.c',
'fix/source/transform_neon.c',
],
},
],
}],
],
}

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@ -1,35 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
# kenny
{
'target_name': 'isac_fix_test',
'type': 'executable',
'dependencies': [
'isac_fix',
'<(webrtc_root)/test/test.gyp:test_support',
],
'include_dirs': [
'./fix/test',
'./fix/include',
'<(webrtc_root)',
],
'sources': [
'./fix/test/kenny.cc',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
],
}
# TODO(kma): Add bit-exact test for iSAC-fix.

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@ -1,68 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'opus_complexity%': 0,
},
'targets': [
{
'target_name': 'webrtc_opus',
'type': 'static_library',
'conditions': [
['build_opus==1', {
'dependencies': [
'<(opus_dir)/opus.gyp:opus'
],
'export_dependent_settings': [
'<(opus_dir)/opus.gyp:opus',
],
'direct_dependent_settings': {
'include_dirs': [ # need by Neteq audio classifier.
'<(opus_dir)/src/src',
'<(opus_dir)/src/celt',
],
},
}, {
'conditions': [
['build_with_mozilla==1', {
# Mozilla provides its own build of the opus library.
'include_dirs': [
'/media/libopus/include',
'/media/libopus/src',
'/media/libopus/celt',
],
'direct_dependent_settings': {
'include_dirs': [
'/media/libopus/include',
'/media/libopus/src',
'/media/libopus/celt',
],
},
}],
],
}],
],
'dependencies': [
'audio_encoder_interface',
],
'defines': [
'OPUS_COMPLEXITY=<(opus_complexity)'
],
'sources': [
'audio_decoder_opus.cc',
'audio_decoder_opus.h',
'audio_encoder_opus.cc',
'audio_encoder_opus.h',
'opus_inst.h',
'opus_interface.c',
'opus_interface.h',
],
},
],
}

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@ -1,29 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'pcm16b',
'type': 'static_library',
'dependencies': [
'audio_encoder_interface',
'audio_decoder_interface',
'g711',
],
'sources': [
'audio_decoder_pcm16b.cc',
'audio_decoder_pcm16b.h',
'audio_encoder_pcm16b.cc',
'audio_encoder_pcm16b.h',
'pcm16b.c',
'pcm16b.h',
],
},
], # targets
}

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@ -1,33 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'red',
'type': 'static_library',
'dependencies': [
'audio_encoder_interface',
],
'include_dirs': [
'include',
'<(webrtc_root)',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
'<(webrtc_root)',
],
},
'sources': [
'audio_encoder_copy_red.h',
'audio_encoder_copy_red.cc',
],
},
], # targets
}

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@ -1,130 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'codecs': [
'cng',
'g711',
'pcm16b',
],
'neteq_defines': [],
'conditions': [
['include_ilbc==1', {
'codecs': ['ilbc',],
'neteq_defines': ['WEBRTC_CODEC_ILBC',],
}],
['include_opus==1', {
'codecs': ['webrtc_opus',],
'neteq_defines': ['WEBRTC_CODEC_OPUS',],
}],
['include_g722==1', {
'codecs': ['g722',],
'neteq_defines': ['WEBRTC_CODEC_G722',],
}],
['include_isac==1', {
'codecs': ['isac', 'isac_fix',],
'neteq_defines': ['WEBRTC_CODEC_ISAC', 'WEBRTC_CODEC_ISACFIX',],
}],
],
'neteq_dependencies': [
'<@(codecs)',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'audio_decoder_interface',
],
},
'targets': [
{
'target_name': 'neteq',
'type': 'static_library',
'dependencies': [
'<@(neteq_dependencies)',
'<(webrtc_root)/common.gyp:webrtc_common',
'builtin_audio_decoder_factory',
'rent_a_codec',
],
'defines': [
'<@(neteq_defines)',
],
'sources': [
'include/neteq.h',
'accelerate.cc',
'accelerate.h',
'audio_decoder_impl.cc',
'audio_decoder_impl.h',
'audio_multi_vector.cc',
'audio_multi_vector.h',
'audio_vector.cc',
'audio_vector.h',
'background_noise.cc',
'background_noise.h',
'buffer_level_filter.cc',
'buffer_level_filter.h',
'comfort_noise.cc',
'comfort_noise.h',
'cross_correlation.cc',
'cross_correlation.h',
'decision_logic.cc',
'decision_logic.h',
'decision_logic_fax.cc',
'decision_logic_fax.h',
'decision_logic_normal.cc',
'decision_logic_normal.h',
'decoder_database.cc',
'decoder_database.h',
'defines.h',
'delay_manager.cc',
'delay_manager.h',
'delay_peak_detector.cc',
'delay_peak_detector.h',
'dsp_helper.cc',
'dsp_helper.h',
'dtmf_buffer.cc',
'dtmf_buffer.h',
'dtmf_tone_generator.cc',
'dtmf_tone_generator.h',
'expand.cc',
'expand.h',
'merge.cc',
'merge.h',
'nack_tracker.h',
'nack_tracker.cc',
'neteq_impl.cc',
'neteq_impl.h',
'neteq.cc',
'statistics_calculator.cc',
'statistics_calculator.h',
'normal.cc',
'normal.h',
'packet.cc',
'packet.h',
'packet_buffer.cc',
'packet_buffer.h',
'red_payload_splitter.cc',
'red_payload_splitter.h',
'post_decode_vad.cc',
'post_decode_vad.h',
'preemptive_expand.cc',
'preemptive_expand.h',
'random_vector.cc',
'random_vector.h',
'rtcp.cc',
'rtcp.h',
'sync_buffer.cc',
'sync_buffer.h',
'tick_timer.cc',
'tick_timer.h',
'timestamp_scaler.cc',
'timestamp_scaler.h',
'time_stretch.cc',
'time_stretch.h',
],
},
], # targets
}

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@ -1,312 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['enable_protobuf==1', {
'targets': [
{
'target_name': 'neteq_rtpplay',
'type': 'executable',
'dependencies': [
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
'neteq',
'neteq_unittest_tools',
],
'sources': [
'tools/neteq_rtpplay.cc',
],
'defines': [
],
}, # neteq_rtpplay
],
}],
],
'targets': [
{
'target_name': 'RTPencode',
'type': 'executable',
'dependencies': [
# TODO(hlundin): Make RTPencode use ACM to encode files.
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'cng',
'g711',
'g722',
'ilbc',
'isac',
'neteq_test_tools', # Test helpers
'pcm16b',
'webrtc_opus',
],
'defines': [
'CODEC_ILBC',
'CODEC_PCM16B',
'CODEC_G711',
'CODEC_G722',
'CODEC_ISAC',
'CODEC_PCM16B_WB',
'CODEC_ISAC_SWB',
'CODEC_PCM16B_32KHZ',
'CODEC_PCM16B_48KHZ',
'CODEC_CNGCODEC8',
'CODEC_CNGCODEC16',
'CODEC_CNGCODEC32',
'CODEC_ATEVENT_DECODE',
'CODEC_RED',
'CODEC_OPUS',
],
'include_dirs': [
'include',
'test',
'<(webrtc_root)',
],
'sources': [
'test/RTPencode.cc',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
{
'target_name': 'RTPjitter',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [
'test/RTPjitter.cc',
],
},
{
'target_name': 'rtp_analyze',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'neteq_unittest_tools',
],
'sources': [
'tools/rtp_analyze.cc',
],
},
{
'target_name': 'RTPchange',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'neteq_test_tools',
],
'sources': [
'test/RTPchange.cc',
],
},
{
'target_name': 'RTPtimeshift',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'neteq_test_tools',
],
'sources': [
'test/RTPtimeshift.cc',
],
},
{
'target_name': 'rtpcat',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/test/test.gyp:rtp_test_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
],
'sources': [
'tools/rtpcat.cc',
],
},
{
'target_name': 'audio_classifier_test',
'type': 'executable',
'dependencies': [
'neteq',
'webrtc_opus',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
],
'sources': [
'test/audio_classifier_test.cc',
],
},
{
'target_name': 'neteq_test_support',
'type': 'static_library',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'neteq',
'neteq_unittest_tools',
'pcm16b',
],
'sources': [
'tools/neteq_external_decoder_test.cc',
'tools/neteq_external_decoder_test.h',
'tools/neteq_performance_test.cc',
'tools/neteq_performance_test.h',
],
}, # neteq_test_support
{
'target_name': 'neteq_quality_test_support',
'type': 'static_library',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'neteq',
'neteq_unittest_tools',
],
'sources': [
'tools/neteq_quality_test.cc',
'tools/neteq_quality_test.h',
],
}, # neteq_test_support
{
'target_name': 'neteq_speed_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:test_support',
'neteq',
'neteq_test_support',
],
'sources': [
'test/neteq_speed_test.cc',
],
},
{
'target_name': 'neteq_opus_quality_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/test/test.gyp:test_support_main',
'neteq',
'neteq_quality_test_support',
'webrtc_opus',
],
'sources': [
'test/neteq_opus_quality_test.cc',
],
},
{
'target_name': 'neteq_isac_quality_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/test/test.gyp:test_support_main',
'isac_fix',
'neteq',
'neteq_quality_test_support',
],
'sources': [
'test/neteq_isac_quality_test.cc',
],
},
{
'target_name': 'neteq_pcmu_quality_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/test/test.gyp:test_support_main',
'g711',
'neteq',
'neteq_quality_test_support',
],
'sources': [
'test/neteq_pcmu_quality_test.cc',
],
},
{
'target_name': 'neteq_ilbc_quality_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:test_support_main',
'neteq',
'neteq_quality_test_support',
'ilbc',
],
'sources': [
'test/neteq_ilbc_quality_test.cc',
],
},
{
'target_name': 'neteq_test_tools',
# Collection of useful functions used in other tests.
'type': 'static_library',
'variables': {
# Expects RTP packets without payloads when enabled.
'neteq_dummy_rtp%': 0,
},
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common.gyp:webrtc_common',
'cng',
'g711',
'g722',
'ilbc',
'isac',
'pcm16b',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
'test',
'<(webrtc_root)',
],
},
'defines': [
],
'include_dirs': [
'include',
'test',
'<(webrtc_root)',
],
'sources': [
'test/NETEQTEST_DummyRTPpacket.cc',
'test/NETEQTEST_DummyRTPpacket.h',
'test/NETEQTEST_RTPpacket.cc',
'test/NETEQTEST_RTPpacket.h',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
], # targets
}

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@ -1,34 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'audio_conference_mixer',
'type': 'static_library',
'dependencies': [
'audio_processing',
'webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/audio_conference_mixer.h',
'include/audio_conference_mixer_defines.h',
'source/audio_frame_manipulator.cc',
'source/audio_frame_manipulator.h',
'source/memory_pool.h',
'source/memory_pool_posix.h',
'source/memory_pool_win.h',
'source/audio_conference_mixer_impl.cc',
'source/audio_conference_mixer_impl.h',
'source/time_scheduler.cc',
'source/time_scheduler.h',
],
},
], # targets
}

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@ -1,316 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'audio_device',
'type': 'static_library',
'dependencies': [
'webrtc_utility',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
'.',
'../include',
'include',
'dummy', # Contains dummy audio device implementations.
],
'direct_dependent_settings': {
'include_dirs': [
'../include',
'include',
],
},
# TODO(xians): Rename files to e.g. *_linux.{ext}, remove sources in conditions section
'sources': [
'include/audio_device.h',
'include/audio_device_defines.h',
'audio_device_buffer.cc',
'audio_device_buffer.h',
'audio_device_generic.cc',
'audio_device_generic.h',
'audio_device_config.h',
'dummy/audio_device_dummy.cc',
'dummy/audio_device_dummy.h',
'dummy/file_audio_device.cc',
'dummy/file_audio_device.h',
'fine_audio_buffer.cc',
'fine_audio_buffer.h',
],
'conditions': [
['build_with_mozilla==1', {
'cflags_mozilla': [
'$(NSPR_CFLAGS)',
],
}],
['hardware_aec_ns==1', {
'defines': [
'WEBRTC_HARDWARE_AEC_NS',
],
}],
['include_sndio_audio==1', {
'include_dirs': [
'sndio',
],
}], # include_sndio_audio==1
['OS=="linux" or include_alsa_audio==1 or include_pulse_audio==1', {
'include_dirs': [
'linux',
],
}], # OS=="linux" or include_alsa_audio==1 or include_pulse_audio==1
['OS=="ios"', {
'include_dirs': [
'ios',
],
}], # OS==ios
['OS=="mac"', {
'include_dirs': [
'mac',
],
}], # OS==mac
['OS=="win"', {
'include_dirs': [
'win',
],
}],
['OS=="android"', {
'include_dirs': [
'/widget/android',
'android',
],
}], # OS==android
['enable_android_opensl==1', {
'include_dirs': [
'dom/media/systemservices',
'opensl',
],
}], # enable_android_opensl
['include_internal_audio_device==0', {
'defines': [
'WEBRTC_DUMMY_AUDIO_BUILD',
],
}],
['build_with_chromium==0', {
'sources': [
# Don't link these into Chrome since they contain static data.
'dummy/file_audio_device_factory.cc',
'dummy/file_audio_device_factory.h',
],
}],
['include_internal_audio_device==1', {
'sources': [
'audio_device_impl.cc',
'audio_device_impl.h',
# used externally for getUserMedia
'opensl/single_rw_fifo.cc',
'opensl/single_rw_fifo.h',
],
'conditions': [
['use_dummy_audio_file_devices==1', {
'defines': [
'WEBRTC_DUMMY_FILE_DEVICES',
],
}, { # use_dummy_audio_file_devices==0, so use a platform device.
'conditions': [
['OS=="android"', {
'sources': [
'android/audio_device_template.h',
'android/audio_manager.cc',
'android/audio_manager.h',
'android/audio_record_jni.cc',
'android/audio_record_jni.h',
'android/audio_track_jni.cc',
'android/audio_track_jni.h',
'android/build_info.cc',
'android/build_info.h',
'android/opensles_common.cc',
'android/opensles_common.h',
'android/opensles_player.cc',
'android/opensles_player.h',
'android/opensles_recorder.cc',
'android/opensles_recorder.h',
],
'link_settings': {
'libraries': [
'-llog',
'-lOpenSLES',
],
},
}],
['OS=="linux"', {
'link_settings': {
'libraries': [
'-ldl',
],
},
}],
['include_sndio_audio==1', {
'link_settings': {
'libraries': [
'-lsndio',
],
},
'sources': [
'sndio/audio_device_sndio.cc',
'sndio/audio_device_sndio.h',
],
}],
['include_alsa_audio==1', {
'cflags_mozilla': [
'$(MOZ_ALSA_CFLAGS)',
],
'defines': [
'LINUX_ALSA',
],
'link_settings': {
'libraries': [
'-lX11',
],
},
'sources': [
'linux/alsasymboltable_linux.cc',
'linux/alsasymboltable_linux.h',
'linux/audio_device_alsa_linux.cc',
'linux/audio_device_alsa_linux.h',
'linux/audio_mixer_manager_alsa_linux.cc',
'linux/audio_mixer_manager_alsa_linux.h',
'linux/latebindingsymboltable_linux.cc',
'linux/latebindingsymboltable_linux.h',
],
}],
['include_pulse_audio==1', {
'cflags_mozilla': [
'$(MOZ_PULSEAUDIO_CFLAGS)',
],
'defines': [
'LINUX_PULSE',
],
'link_settings': {
'libraries': [
'-lX11',
],
},
'sources': [
'linux/audio_device_pulse_linux.cc',
'linux/audio_device_pulse_linux.h',
'linux/audio_mixer_manager_pulse_linux.cc',
'linux/audio_mixer_manager_pulse_linux.h',
'linux/latebindingsymboltable_linux.cc',
'linux/latebindingsymboltable_linux.h',
'linux/pulseaudiosymboltable_linux.cc',
'linux/pulseaudiosymboltable_linux.h',
],
}],
['OS=="mac"', {
'sources': [
'mac/audio_device_mac.cc',
'mac/audio_device_mac.h',
'mac/audio_mixer_manager_mac.cc',
'mac/audio_mixer_manager_mac.h',
'mac/portaudio/pa_memorybarrier.h',
'mac/portaudio/pa_ringbuffer.c',
'mac/portaudio/pa_ringbuffer.h',
],
'link_settings': {
'libraries': [
'$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework',
'$(SDKROOT)/System/Library/Frameworks/CoreAudio.framework',
],
},
}],
['OS=="ios"', {
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
],
'export_dependent_settings': [
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
],
'sources': [
'ios/audio_device_ios.h',
'ios/audio_device_ios.mm',
'ios/audio_device_not_implemented_ios.mm',
'ios/audio_session_observer.h',
'ios/objc/RTCAudioSession+Configuration.mm',
'ios/objc/RTCAudioSession+Private.h',
'ios/objc/RTCAudioSession.h',
'ios/objc/RTCAudioSession.mm',
'ios/objc/RTCAudioSessionConfiguration.h',
'ios/objc/RTCAudioSessionConfiguration.m',
'ios/objc/RTCAudioSessionDelegateAdapter.h',
'ios/objc/RTCAudioSessionDelegateAdapter.mm',
'ios/voice_processing_audio_unit.h',
'ios/voice_processing_audio_unit.mm',
],
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework AudioToolbox',
'-framework AVFoundation',
'-framework Foundation',
'-framework UIKit',
],
},
},
}],
['OS=="win"', {
'sources': [
'win/audio_device_core_win.cc',
'win/audio_device_core_win.h',
'win/audio_device_wave_win.cc',
'win/audio_device_wave_win.h',
'win/audio_mixer_manager_win.cc',
'win/audio_mixer_manager_win.h',
],
'link_settings': {
'libraries': [
# Required for the built-in WASAPI AEC.
'-ldmoguids.lib',
'-lwmcodecdspuuid.lib',
'-lamstrmid.lib',
'-lmsdmo.lib',
],
},
}],
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-bool-conversion',
'-Wno-delete-non-virtual-dtor',
'-Wno-logical-op-parentheses',
'-Wno-microsoft-extra-qualification',
'-Wno-microsoft-goto',
'-Wno-missing-braces',
'-Wno-parentheses-equality',
'-Wno-reorder',
'-Wno-shift-overflow',
'-Wno-tautological-compare',
'-Wno-unused-private-field',
],
},
},
}],
], # conditions (for non-dummy devices)
}], # use_dummy_audio_file_devices check
], # conditions
}], # include_internal_audio_device==1
], # conditions
},
],
}

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@ -1,33 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'audio_mixer',
'type': 'static_library',
'dependencies': [
'audio_processing',
'webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/voice_engine/voice_engine.gyp:level_indicator',
],
'sources': [
'audio_frame_manipulator.cc',
'audio_frame_manipulator.h',
'audio_mixer.h',
'audio_mixer_impl.cc',
'audio_mixer_impl.h',
'default_output_rate_calculator.cc',
'default_output_rate_calculator.h',
'output_rate_calculator.h',
],
},
], # targets
}

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@ -1,351 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../build/common.gypi',
],
'variables': {
'shared_generated_dir': '<(SHARED_INTERMEDIATE_DIR)/audio_processing/asm_offsets',
'apm_debug_dump%': 1,
},
'targets': [
{
'target_name': 'audio_processing',
'type': 'static_library',
'variables': {
# Outputs some low-level debug files.
'agc_debug_dump%': 0,
# Disables the usual mode where we trust the reported system delay
# values the AEC receives. The corresponding define is set appropriately
# in the code, but it can be force-enabled here for testing.
'aec_untrusted_delay_for_testing%': 0,
},
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
# '<(webrtc_root)/modules/modules.gyp:isac',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'aec/aec_core.cc',
'aec/aec_core.h',
'aec/aec_core_optimized_methods.h',
'aec/aec_resampler.cc',
'aec/aec_resampler.h',
'aec/echo_cancellation.cc',
'aec/echo_cancellation.h',
'aec3/aec3_constants.h',
'aec3/block_framer.cc',
'aec3/block_framer.h',
'aec3/block_processor.cc',
'aec3/block_processor.h',
'aec3/cascaded_biquad_filter.cc',
'aec3/cascaded_biquad_filter.h',
'aec3/echo_canceller3.cc',
'aec3/echo_canceller3.h',
'aec3/frame_blocker.cc',
'aec3/frame_blocker.h',
'aecm/aecm_core.cc',
'aecm/aecm_core.h',
'aecm/echo_control_mobile.cc',
'aecm/echo_control_mobile.h',
'agc/agc.cc',
'agc/agc.h',
'agc/agc_manager_direct.cc',
'agc/agc_manager_direct.h',
'agc/gain_map_internal.h',
'agc/legacy/analog_agc.c',
'agc/legacy/analog_agc.h',
'agc/legacy/digital_agc.c',
'agc/legacy/digital_agc.h',
'agc/legacy/gain_control.h',
'agc/loudness_histogram.cc',
'agc/loudness_histogram.h',
'agc/legacy/gain_control.h',
'agc/utility.cc',
'agc/utility.h',
'audio_buffer.cc',
'audio_buffer.h',
'audio_processing_impl.cc',
'audio_processing_impl.h',
'beamformer/array_util.cc',
'beamformer/array_util.h',
'beamformer/complex_matrix.h',
'beamformer/covariance_matrix_generator.cc',
'beamformer/covariance_matrix_generator.h',
'beamformer/matrix.h',
'beamformer/nonlinear_beamformer.cc',
'beamformer/nonlinear_beamformer.h',
'common.h',
'echo_cancellation_impl.cc',
'echo_cancellation_impl.h',
'echo_control_mobile_impl.cc',
'echo_control_mobile_impl.h',
'echo_detector/circular_buffer.cc',
'echo_detector/circular_buffer.h',
'echo_detector/mean_variance_estimator.cc',
'echo_detector/mean_variance_estimator.h',
'echo_detector/moving_max.cc',
'echo_detector/moving_max.h',
'echo_detector/normalized_covariance_estimator.cc',
'echo_detector/normalized_covariance_estimator.h',
'gain_control_for_experimental_agc.cc',
'gain_control_for_experimental_agc.h',
'gain_control_impl.cc',
'gain_control_impl.h',
'include/audio_processing.cc',
'include/audio_processing.h',
'include/config.cc',
'include/config.h',
'level_controller/biquad_filter.cc',
'level_controller/biquad_filter.h',
'level_controller/down_sampler.cc',
'level_controller/down_sampler.h',
'level_controller/gain_applier.cc',
'level_controller/gain_applier.h',
'level_controller/gain_selector.cc',
'level_controller/gain_selector.h',
'level_controller/lc_constants.h',
'level_controller/level_controller.cc',
'level_controller/level_controller.h',
'level_controller/noise_spectrum_estimator.cc',
'level_controller/noise_spectrum_estimator.h',
'level_controller/noise_level_estimator.cc',
'level_controller/noise_level_estimator.h',
'level_controller/peak_level_estimator.cc',
'level_controller/peak_level_estimator.h',
'level_controller/saturating_gain_estimator.cc',
'level_controller/saturating_gain_estimator.h',
'level_controller/signal_classifier.cc',
'level_controller/signal_classifier.h',
'level_estimator_impl.cc',
'level_estimator_impl.h',
'logging/apm_data_dumper.cc',
'logging/apm_data_dumper.h',
'low_cut_filter.cc',
'low_cut_filter.h',
'noise_suppression_impl.cc',
'noise_suppression_impl.h',
'render_queue_item_verifier.h',
'residual_echo_detector.cc',
'residual_echo_detector.h',
'rms_level.cc',
'rms_level.h',
'splitting_filter.cc',
'splitting_filter.h',
'three_band_filter_bank.cc',
'three_band_filter_bank.h',
'transient/common.h',
'transient/daubechies_8_wavelet_coeffs.h',
'transient/dyadic_decimator.h',
'transient/moving_moments.cc',
'transient/moving_moments.h',
'transient/transient_detector.cc',
'transient/transient_detector.h',
'transient/transient_suppressor.cc',
'transient/transient_suppressor.h',
'transient/wpd_node.cc',
'transient/wpd_node.h',
'transient/wpd_tree.cc',
'transient/wpd_tree.h',
'typing_detection.cc',
'typing_detection.h',
'utility/block_mean_calculator.cc',
'utility/block_mean_calculator.h',
'utility/delay_estimator.cc',
'utility/delay_estimator.h',
'utility/delay_estimator_internal.h',
'utility/delay_estimator_wrapper.cc',
'utility/delay_estimator_wrapper.h',
'utility/ooura_fft.cc',
'utility/ooura_fft.h',
'utility/ooura_fft_tables_common.h',
'vad/common.h',
'vad/gmm.cc',
'vad/gmm.h',
'vad/noise_gmm_tables.h',
'vad/pitch_based_vad.cc',
'vad/pitch_based_vad.h',
'vad/pitch_internal.cc',
'vad/pitch_internal.h',
'vad/pole_zero_filter.cc',
'vad/pole_zero_filter.h',
'vad/standalone_vad.cc',
'vad/standalone_vad.h',
'vad/vad_audio_proc.cc',
'vad/vad_audio_proc.h',
'vad/vad_audio_proc_internal.h',
'vad/vad_circular_buffer.cc',
'vad/vad_circular_buffer.h',
'vad/voice_activity_detector.cc',
'vad/voice_activity_detector.h',
'vad/voice_gmm_tables.h',
'voice_detection_impl.cc',
'voice_detection_impl.h',
],
'conditions': [
['apm_debug_dump==1', {
'defines': ['WEBRTC_APM_DEBUG_DUMP=1',],
}, {
'defines': ['WEBRTC_APM_DEBUG_DUMP=0',],
}],
['aec_untrusted_delay_for_testing==1', {
'defines': ['WEBRTC_UNTRUSTED_DELAY',],
}],
['agc_debug_dump==1', {
'defines': ['WEBRTC_AGC_DEBUG_DUMP',],
}],
['enable_protobuf==1', {
'dependencies': ['audioproc_debug_proto'],
'defines': ['WEBRTC_AUDIOPROC_DEBUG_DUMP'],
}],
['enable_intelligibility_enhancer==1', {
'defines': ['WEBRTC_INTELLIGIBILITY_ENHANCER=1',],
'sources': [
'intelligibility/intelligibility_enhancer.cc',
'intelligibility/intelligibility_enhancer.h',
'intelligibility/intelligibility_utils.cc',
'intelligibility/intelligibility_utils.h',
],
}, {
'defines': ['WEBRTC_INTELLIGIBILITY_ENHANCER=0',],
}],
['prefer_fixed_point==1', {
'defines': ['WEBRTC_NS_FIXED'],
'sources': [
'ns/noise_suppression_x.h',
'ns/noise_suppression_x.c',
'ns/nsx_core.c',
'ns/nsx_core.h',
'ns/nsx_defines.h',
],
'conditions': [
['target_arch=="mipsel" and mips_arch_variant!="r6"', {
'sources': [
'ns/nsx_core_mips.c',
],
}, {
'sources': [
'ns/nsx_core_c.c',
],
}],
],
}, {
'defines': ['WEBRTC_NS_FLOAT'],
'sources': [
'ns/defines.h',
'ns/noise_suppression.h',
'ns/noise_suppression.c',
'ns/ns_core.c',
'ns/ns_core.h',
'ns/windows_private.h',
],
}],
['target_arch=="ia32" or target_arch=="x64"', {
'dependencies': ['audio_processing_sse2',],
}],
['build_with_neon==1', {
'dependencies': ['audio_processing_neon',],
}],
['target_arch=="mipsel" and mips_arch_variant!="r6"', {
'sources': [
'aecm/aecm_core_mips.cc',
],
'conditions': [
['mips_float_abi=="hard"', {
'sources': [
'aec/aec_core_mips.cc',
'aec/aec_rdft_mips.cc',
'utility/ooura_fft_mips.cc',
],
}],
],
}, {
'sources': [
'aecm/aecm_core_c.cc',
],
}],
],
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
'conditions': [
['enable_protobuf==1', {
'targets': [
{
'target_name': 'audioproc_debug_proto',
'type': 'static_library',
'sources': ['debug.proto',],
'variables': {
'proto_in_dir': '.',
# Workaround to protect against gyp's pathname relativization when
# this file is included by modules.gyp.
'proto_out_protected': 'webrtc/modules/audio_processing',
'proto_out_dir': '<(proto_out_protected)',
},
'includes': ['../../build/protoc.gypi',],
},
],
}],
['target_arch=="ia32" or target_arch=="x64"', {
'targets': [
{
'target_name': 'audio_processing_sse2',
'type': 'static_library',
'sources': [
'aec/aec_core_sse2.cc',
'utility/ooura_fft_sse2.cc',
'utility/ooura_fft_tables_neon_sse2.h',
],
'conditions': [
['apm_debug_dump==1', {
'defines': ['WEBRTC_APM_DEBUG_DUMP=1',],
}, {
'defines': ['WEBRTC_APM_DEBUG_DUMP=0',],
}],
['os_posix==1', {
'cflags': [ '-msse2', ],
'cflags_mozilla': [ '-msse2', ],
'xcode_settings': {
'OTHER_CFLAGS': [ '-msse2', ],
},
}],
],
},
],
}],
['build_with_neon==1', {
'targets': [{
'target_name': 'audio_processing_neon',
'type': 'static_library',
'includes': ['../../build/arm_neon.gypi',],
'dependencies': [
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'aec/aec_core_neon.cc',
'aecm/aecm_core_neon.cc',
'ns/nsx_core_neon.c',
'utility/ooura_fft_neon.cc',
],
'conditions': [
['apm_debug_dump==1', {
'defines': ['WEBRTC_APM_DEBUG_DUMP=1',],
}],
['apm_debug_dump==0', {
'defines': ['WEBRTC_APM_DEBUG_DUMP=0',],
}],
],
}],
}],
],
}

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@ -1,166 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'audioproc_test_utils',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'test/audio_buffer_tools.cc',
'test/audio_buffer_tools.h',
'test/test_utils.cc',
'test/test_utils.h',
],
},
{
'target_name': 'transient_suppression_test',
'type': 'executable',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
],
'sources': [
'transient/transient_suppression_test.cc',
'transient/file_utils.cc',
'transient/file_utils.h',
],
}, # transient_suppression_test
{
'target_name': 'click_annotate',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:audio_processing',
],
'sources': [
'transient/click_annotate.cc',
'transient/file_utils.cc',
'transient/file_utils.h',
],
}, # click_annotate
{
'target_name': 'nonlinear_beamformer_test',
'type': 'executable',
'dependencies': [
'audioproc_test_utils',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
],
'sources': [
'beamformer/nonlinear_beamformer_test.cc',
],
}, # nonlinear_beamformer_test
],
'conditions': [
['enable_intelligibility_enhancer==1', {
'defines': ['WEBRTC_INTELLIGIBILITY_ENHANCER=1',],
'targets': [
{
'target_name': 'intelligibility_proc',
'type': 'executable',
'dependencies': [
'audioproc_test_utils',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/test/test.gyp:test_support',
],
'sources': [
'intelligibility/test/intelligibility_proc.cc',
],
},
],
}, {
'defines': ['WEBRTC_INTELLIGIBILITY_ENHANCER=0',],
}],
['enable_protobuf==1', {
'targets': [
{
'target_name': 'audioproc_unittest_proto',
'type': 'static_library',
'sources': [ 'test/unittest.proto', ],
'variables': {
'proto_in_dir': 'test',
# Workaround to protect against gyp's pathname relativization when
# this file is included by modules.gyp.
'proto_out_protected': 'webrtc/modules/audio_processing',
'proto_out_dir': '<(proto_out_protected)',
},
'includes': [ '../../build/protoc.gypi', ],
},
{
'target_name': 'audioproc_protobuf_utils',
'type': 'static_library',
'dependencies': [
'audioproc_debug_proto',
],
'sources': [
'test/protobuf_utils.cc',
'test/protobuf_utils.h',
],
},
{
'target_name': 'audioproc',
'type': 'executable',
'dependencies': [
'audio_processing',
'audioproc_debug_proto',
'audioproc_test_utils',
'audioproc_protobuf_utils',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:test_support',
],
'sources': [ 'test/process_test.cc', ],
},
{
'target_name': 'audioproc_f',
'type': 'executable',
'dependencies': [
'audio_processing',
'audioproc_debug_proto',
'audioproc_test_utils',
'audioproc_protobuf_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:test_support',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [
'test/audio_processing_simulator.cc',
'test/audio_processing_simulator.h',
'test/aec_dump_based_simulator.cc',
'test/aec_dump_based_simulator.h',
'test/wav_based_simulator.cc',
'test/wav_based_simulator.h',
'test/audioproc_float.cc',
],
},
{
'target_name': 'unpack_aecdump',
'type': 'executable',
'dependencies': [
'audioproc_debug_proto',
'audioproc_test_utils',
'audioproc_protobuf_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
],
'sources': [ 'test/unpack.cc', ],
},
],
}],
],
}

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@ -1,35 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'bitrate_controller',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'bitrate_controller_impl.cc',
'bitrate_controller_impl.h',
'include/bitrate_controller.h',
'send_side_bandwidth_estimation.cc',
'send_side_bandwidth_estimation.h',
],
'conditions': [
['enable_bwe_test_logging==1', {
'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
}, {
'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
}],
],
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
], # targets
}

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@ -1,40 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'congestion_controller',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:bitrate_controller',
'<(webrtc_root)/modules/modules.gyp:paced_sender',
],
'sources': [
'congestion_controller.cc',
'delay_based_bwe.cc',
'delay_based_bwe.h',
'include/congestion_controller.h',
'median_slope_estimator.cc',
'median_slope_estimator.h',
'probe_bitrate_estimator.cc',
'probe_bitrate_estimator.h',
'probe_controller.cc',
'probe_controller.h',
'probing_interval_estimator.cc',
'probing_interval_estimator.h',
'transport_feedback_adapter.cc',
'transport_feedback_adapter.h',
'trendline_estimator.cc',
'trendline_estimator.h',
],
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
], # targets
}

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@ -1,300 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'multi_monitor_screenshare%' : 1,
},
'targets': [
{
'target_name': 'primitives',
'type': 'static_library',
'sources': [
'desktop_capture_types.h',
'desktop_frame.cc',
'desktop_frame.h',
'desktop_geometry.cc',
'desktop_geometry.h',
'desktop_region.cc',
'desktop_region.h',
],
},
{
'target_name': 'desktop_capture',
'type': 'static_library',
'dependencies': [
':primitives',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'include_dirs': [
'../../../../../libyuv/libyuv/include/',
],
'sources': [
'cropped_desktop_frame.cc',
'cropped_desktop_frame.h',
'cropping_window_capturer.cc',
'cropping_window_capturer.h',
'cropping_window_capturer_win.cc',
'desktop_and_cursor_composer.cc',
'desktop_and_cursor_composer.h',
'desktop_capture_options.h',
'desktop_capture_options.cc',
'desktop_capturer.h',
'desktop_capturer_differ_wrapper.cc',
'desktop_capturer_differ_wrapper.h',
'desktop_frame_rotation.cc',
'desktop_frame_rotation.h',
# 'desktop_frame_win.cc',
# 'desktop_frame_win.h',
'differ_block.cc',
'differ_block.h',
# 'mac/desktop_configuration.h',
# 'mac/desktop_configuration.mm',
# 'mac/desktop_configuration_monitor.h',
# 'mac/desktop_configuration_monitor.cc',
# 'mac/full_screen_chrome_window_detector.cc',
# 'mac/full_screen_chrome_window_detector.h',
# 'mac/scoped_pixel_buffer_object.cc',
# 'mac/scoped_pixel_buffer_object.h',
# 'mac/window_list_utils.cc',
# 'mac/window_list_utils.h',
'mouse_cursor.cc',
'mouse_cursor.h',
'mouse_cursor_monitor.h',
# 'mouse_cursor_monitor_mac.mm',
# 'mouse_cursor_monitor_win.cc',
'screen_capture_frame_queue.h',
'screen_capturer.h',
'screen_capturer_helper.cc',
'screen_capturer_helper.h',
# 'screen_capturer_mac.mm',
# 'screen_capturer_win.cc',
'shared_desktop_frame.cc',
'shared_desktop_frame.h',
'shared_memory.cc',
'shared_memory.h',
# 'win/cursor.cc',
# 'win/cursor.h',
# 'win/d3d_device.cc',
# 'win/d3d_device.h',
# 'win/desktop.cc',
# 'win/desktop.h',
# 'win/dxgi_adapter_duplicator.cc',
# 'win/dxgi_adapter_duplicator.h',
# 'win/dxgi_duplicator_controller.cc',
# 'win/dxgi_duplicator_controller.h',
# 'win/dxgi_output_duplicator.cc',
# 'win/dxgi_output_duplicator.h',
# 'win/dxgi_texture.cc',
# 'win/dxgi_texture.h',
# 'win/dxgi_texture_mapping.cc',
# 'win/dxgi_texture_mapping.h',
# 'win/dxgi_texture_staging.cc',
# 'win/dxgi_texture_staging.h',
# 'win/scoped_gdi_object.h',
# 'win/scoped_thread_desktop.cc',
# 'win/scoped_thread_desktop.h',
# 'win/screen_capture_utils.cc',
# 'win/screen_capture_utils.h',
# 'win/screen_capturer_win_directx.cc',
# 'win/screen_capturer_win_directx.h',
# 'win/screen_capturer_win_gdi.cc',
# 'win/screen_capturer_win_gdi.h',
# 'win/screen_capturer_win_magnifier.cc',
# 'win/screen_capturer_win_magnifier.h',
# 'win/window_capture_utils.cc',
# 'win/window_capture_utils.h',
'window_capturer.h',
# 'window_capturer_mac.mm',
# 'window_capturer_win.cc',
"desktop_capturer.h",
"desktop_capturer.cc",
"desktop_device_info.h",
"desktop_device_info.cc",
# "app_capturer.h",
# "app_capturer.cc",
],
'conditions': [
['OS!="android"', {
'sources': [
'../../video_engine/desktop_capture_impl.cc',
'../../video_engine/desktop_capture_impl.h',
],
}],
['multi_monitor_screenshare != 0', {
'defines': [
'MULTI_MONITOR_SCREENSHARE'
],
}],
['OS!="ios" and (target_arch=="ia32" or target_arch=="x64")', {
'dependencies': [
'desktop_capture_differ_sse2',
],
}],
['use_x11==1', {
'defines':[
'USE_X11',
],
'sources': [
'mouse_cursor_monitor_x11.cc',
'screen_capturer_x11.cc',
'window_capturer_x11.cc',
"x11/shared_x_util.h",
"x11/shared_x_util.cc",
'x11/shared_x_display.h',
'x11/shared_x_display.cc',
'x11/x_error_trap.cc',
'x11/x_error_trap.h',
'x11/x_server_pixel_buffer.cc',
'x11/x_server_pixel_buffer.h',
'x11/desktop_device_info_x11.h',
'x11/desktop_device_info_x11.cc',
'app_capturer_x11.cc',
],
'link_settings': {
'libraries': [
'-lX11',
'-lXcomposite',
'-lXdamage',
'-lXext',
'-lXfixes',
'-lXrender',
],
},
}],
['OS!="win" and OS!="mac" and use_x11==0', {
'sources': [
"app_capturer_null.cc",
"desktop_device_info_null.cc",
'mouse_cursor_monitor_null.cc',
'screen_capturer_null.cc',
'window_capturer_null.cc',
],
}],
['OS!="ios" ', {
'sources': [
'differ_block.cc',
'differ_block.h',
],
}],
['OS=="mac"', {
'sources': [
"mac/desktop_configuration.h",
"mac/desktop_configuration.mm",
"mac/desktop_configuration_monitor.h",
"mac/desktop_configuration_monitor.cc",
"mac/full_screen_chrome_window_detector.cc",
"mac/full_screen_chrome_window_detector.h",
"mac/window_list_utils.cc",
"mac/window_list_utils.h",
"mac/scoped_pixel_buffer_object.cc",
"mac/scoped_pixel_buffer_object.h",
"mac/desktop_device_info_mac.h",
"mac/desktop_device_info_mac.mm",
"mouse_cursor_monitor_mac.mm",
"screen_capturer_mac.mm",
"window_capturer_mac.mm",
"app_capturer_mac.mm",
],
'link_settings': {
'libraries': [
'$(SDKROOT)/System/Library/Frameworks/AppKit.framework',
'$(SDKROOT)/System/Library/Frameworks/IOKit.framework',
'$(SDKROOT)/System/Library/Frameworks/OpenGL.framework',
],
},
}],
['OS=="win"', {
'sources': [
"desktop_frame_win.cc",
"desktop_frame_win.h",
"mouse_cursor_monitor_win.cc",
"screen_capturer_win.cc",
"win/cursor.cc",
"win/cursor.h",
# 'win/d3d_device.cc',
# 'win/d3d_device.h',
"win/desktop.cc",
"win/desktop.h",
# 'win/dxgi_adapter_duplicator.cc',
# 'win/dxgi_adapter_duplicator.h',
# 'win/dxgi_duplicator_controller.cc',
# 'win/dxgi_duplicator_controller.h',
# 'win/dxgi_output_duplicator.cc',
# 'win/dxgi_output_duplicator.h',
# 'win/dxgi_texture.cc',
# 'win/dxgi_texture.h',
# 'win/dxgi_texture_mapping.cc',
# 'win/dxgi_texture_mapping.h',
# 'win/dxgi_texture_staging.cc',
# 'win/dxgi_texture_staging.h',
"win/scoped_gdi_object.h",
"win/scoped_thread_desktop.cc",
"win/scoped_thread_desktop.h",
"win/win_shared.h",
"win/win_shared.cc",
"win/desktop_device_info_win.h",
"win/desktop_device_info_win.cc",
# "win/screen_capturer_win_directx.cc",
# "win/screen_capturer_win_directx.h",
"win/screen_capturer_win_gdi.cc",
"win/screen_capturer_win_gdi.h",
"win/screen_capturer_win_magnifier.cc",
"win/screen_capturer_win_magnifier.h",
"win/screen_capture_utils.cc",
"win/screen_capture_utils.h",
"win/window_capture_utils.cc",
"win/window_capture_utils.h",
"window_capturer_win.cc",
"app_capturer_win.cc",
],
}],
],
'all_dependent_settings': {
'conditions': [
['OS=="win"', {
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
'd3d11.lib',
'dxgi.lib',
],
},
},
}],
],
},
},
], # targets
'conditions': [
['OS!="ios" and (target_arch=="ia32" or target_arch=="x64")', {
'targets': [
{
# Have to be compiled as a separate target because it needs to be
# compiled with SSE2 enabled.
'target_name': 'desktop_capture_differ_sse2',
'type': 'static_library',
'sources': [
'differ_vector_sse2.cc',
'differ_vector_sse2.h',
],
'conditions': [
['os_posix==1', {
'cflags': [ '-msse2', ],
'cflags_mozilla': [ '-msse2', ],
'xcode_settings': {
'OTHER_CFLAGS': [ '-msse2', ],
},
}],
],
},
], # targets
}],
],
}

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@ -1,31 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'media_file',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'media_file.h',
'media_file_defines.h',
'media_file_impl.cc',
'media_file_impl.h',
'media_file_utility.cc',
'media_file_utility.h',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
], # targets
}

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@ -1,31 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../build/common.gypi',
'audio_coding/audio_coding.gypi',
'audio_conference_mixer/audio_conference_mixer.gypi',
'audio_device/audio_device.gypi',
'audio_mixer/audio_mixer.gypi',
'audio_processing/audio_processing.gypi',
'bitrate_controller/bitrate_controller.gypi',
'congestion_controller/congestion_controller.gypi',
'desktop_capture/desktop_capture.gypi',
'media_file/media_file.gypi',
'pacing/pacing.gypi',
'remote_bitrate_estimator/remote_bitrate_estimator.gypi',
'rtp_rtcp/rtp_rtcp.gypi',
'utility/utility.gypi',
'video_coding/codecs/h264/h264.gypi',
'video_coding/codecs/i420/i420.gypi',
'video_coding/video_coding.gypi',
'video_capture/video_capture.gypi',
'video_processing/video_processing.gypi',
],
}

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@ -1,30 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'paced_sender',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/modules/modules.gyp:bitrate_controller',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
],
'sources': [
'alr_detector.cc',
'bitrate_prober.cc',
'bitrate_prober.h',
'paced_sender.cc',
'paced_sender.h',
'packet_router.cc',
'packet_router.h',
],
},
], # targets
}

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@ -1,118 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../build/common.gypi',
],
'targets': [
{
'target_name': 'remote_bitrate_estimator',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/bwe_defines.h',
'include/remote_bitrate_estimator.h',
'include/send_time_history.h',
'aimd_rate_control.cc',
'aimd_rate_control.h',
'bwe_defines.cc',
'inter_arrival.cc',
'inter_arrival.h',
'overuse_detector.cc',
'overuse_detector.h',
'overuse_estimator.cc',
'overuse_estimator.h',
'remote_bitrate_estimator_abs_send_time.cc',
'remote_bitrate_estimator_abs_send_time.h',
'remote_bitrate_estimator_single_stream.cc',
'remote_bitrate_estimator_single_stream.h',
'remote_estimator_proxy.cc',
'remote_estimator_proxy.h',
'send_time_history.cc',
'test/bwe_test_logging.h',
], # source
'conditions': [
['enable_bwe_test_logging==1', {
'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
'sources': [
'test/bwe_test_logging.cc'
],
}, {
'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
}],
],
},
], # targets
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'bwe_tools_util',
'type': 'static_library',
'dependencies': [
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'rtp_rtcp',
],
'sources': [
'tools/bwe_rtp.cc',
'tools/bwe_rtp.h',
],
},
{
'target_name': 'bwe_rtp_to_text',
'type': 'executable',
'includes': [
'../rtp_rtcp/rtp_rtcp.gypi',
],
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:rtp_test_utils',
'bwe_tools_util',
'rtp_rtcp',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
],
},
'sources': [
'tools/rtp_to_text.cc',
], # source
},
{
'target_name': 'bwe_rtp_play',
'type': 'executable',
'includes': [
'../rtp_rtcp/rtp_rtcp.gypi',
],
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:rtp_test_utils',
'bwe_tools_util',
'rtp_rtcp',
],
'direct_dependent_settings': {
'include_dirs': [
'include',
],
},
'sources': [
'tools/bwe_rtp_play.cc',
], # source
},
],
}], # include_tests==1
],
}

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@ -1,192 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'rtp_rtcp',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/modules.gyp:remote_bitrate_estimator',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
# Common
'include/fec_receiver.h',
'include/flexfec_receiver.h',
'include/flexfec_sender.h',
'include/receive_statistics.h',
'include/remote_ntp_time_estimator.h',
'include/rtp_audio_level_observer.h',
'include/rtp_header_parser.h',
'include/rtp_payload_registry.h',
'include/rtp_receiver.h',
'include/rtp_rtcp.h',
'include/rtp_rtcp_defines.h',
'source/byte_io.h',
'source/fec_private_tables_bursty.h',
'source/fec_private_tables_random.h',
'source/flexfec_header_reader_writer.cc',
'source/flexfec_header_reader_writer.h',
'source/flexfec_receiver.cc',
'source/flexfec_sender.cc',
'source/packet_loss_stats.cc',
'source/packet_loss_stats.h',
'source/playout_delay_oracle.cc',
'source/playout_delay_oracle.h',
'source/receive_statistics_impl.cc',
'source/receive_statistics_impl.h',
'source/remote_ntp_time_estimator.cc',
'source/rtcp_packet.cc',
'source/rtcp_packet.h',
'source/rtcp_packet/app.cc',
'source/rtcp_packet/app.h',
'source/rtcp_packet/bye.cc',
'source/rtcp_packet/bye.h',
'source/rtcp_packet/common_header.cc',
'source/rtcp_packet/common_header.h',
'source/rtcp_packet/compound_packet.cc',
'source/rtcp_packet/compound_packet.h',
'source/rtcp_packet/dlrr.cc',
'source/rtcp_packet/dlrr.h',
'source/rtcp_packet/extended_jitter_report.cc',
'source/rtcp_packet/extended_jitter_report.h',
'source/rtcp_packet/extended_reports.cc',
'source/rtcp_packet/extended_reports.h',
'source/rtcp_packet/fir.cc',
'source/rtcp_packet/fir.h',
'source/rtcp_packet/nack.cc',
'source/rtcp_packet/nack.h',
'source/rtcp_packet/pli.cc',
'source/rtcp_packet/pli.h',
'source/rtcp_packet/psfb.cc',
'source/rtcp_packet/psfb.h',
'source/rtcp_packet/rapid_resync_request.cc',
'source/rtcp_packet/rapid_resync_request.h',
'source/rtcp_packet/receiver_report.cc',
'source/rtcp_packet/receiver_report.h',
'source/rtcp_packet/remb.cc',
'source/rtcp_packet/remb.h',
'source/rtcp_packet/report_block.cc',
'source/rtcp_packet/report_block.h',
'source/rtcp_packet/rpsi.cc',
'source/rtcp_packet/rpsi.h',
'source/rtcp_packet/rrtr.cc',
'source/rtcp_packet/rrtr.h',
'source/rtcp_packet/rtpfb.cc',
'source/rtcp_packet/rtpfb.h',
'source/rtcp_packet/sdes.cc',
'source/rtcp_packet/sdes.h',
'source/rtcp_packet/sender_report.cc',
'source/rtcp_packet/sender_report.h',
'source/rtcp_packet/sli.cc',
'source/rtcp_packet/sli.h',
'source/rtcp_packet/target_bitrate.cc',
'source/rtcp_packet/target_bitrate.h',
'source/rtcp_packet/tmmb_item.cc',
'source/rtcp_packet/tmmb_item.h',
'source/rtcp_packet/tmmbn.cc',
'source/rtcp_packet/tmmbn.h',
'source/rtcp_packet/tmmbr.cc',
'source/rtcp_packet/tmmbr.h',
'source/rtcp_packet/transport_feedback.cc',
'source/rtcp_packet/transport_feedback.h',
'source/rtcp_packet/voip_metric.cc',
'source/rtcp_packet/voip_metric.h',
'source/rtcp_receiver.cc',
'source/rtcp_receiver.h',
'source/rtcp_sender.cc',
'source/rtcp_sender.h',
'source/rtcp_utility.cc',
'source/rtcp_utility.h',
'source/rtp_header_extension.cc',
'source/rtp_header_extension.h',
'source/rtp_header_extensions.cc',
'source/rtp_header_extensions.h',
'source/rtp_header_parser.cc',
'source/rtp_packet.cc',
'source/rtp_packet.h',
'source/rtp_packet_received.h',
'source/rtp_packet_to_send.h',
'source/rtp_receiver_impl.cc',
'source/rtp_receiver_impl.h',
'source/rtp_rtcp_config.h',
'source/rtp_rtcp_impl.cc',
'source/rtp_rtcp_impl.h',
'source/rtp_sender.cc',
'source/rtp_sender.h',
'source/rtp_utility.cc',
'source/rtp_utility.h',
'source/ssrc_database.cc',
'source/ssrc_database.h',
'source/time_util.cc',
'source/time_util.h',
'source/tmmbr_help.cc',
'source/tmmbr_help.h',
# Audio Files
'source/dtmf_queue.cc',
'source/dtmf_queue.h',
'source/rtp_receiver_audio.cc',
'source/rtp_receiver_audio.h',
'source/rtp_sender_audio.cc',
'source/rtp_sender_audio.h',
# Video Files
'source/fec_private_tables_random.h',
'source/fec_private_tables_bursty.h',
'source/flexfec_header_reader_writer.cc',
'source/flexfec_header_reader_writer.h',
'source/forward_error_correction.cc',
'source/forward_error_correction.h',
'source/forward_error_correction_internal.cc',
'source/forward_error_correction_internal.h',
'source/rtp_packet_history.cc',
'source/rtp_packet_history.h',
'source/rtp_payload_registry.cc',
'source/rtp_receiver_strategy.cc',
'source/rtp_receiver_strategy.h',
'source/rtp_receiver_video.cc',
'source/rtp_receiver_video.h',
'source/rtp_sender_video.cc',
'source/rtp_sender_video.h',
'source/video_codec_information.h',
'source/rtp_format.cc',
'source/rtp_format.h',
'source/rtp_format_h264.cc',
'source/rtp_format_h264.h',
'source/rtp_format_vp8.cc',
'source/rtp_format_vp8.h',
'source/rtp_format_vp9.cc',
'source/rtp_format_vp9.h',
'source/rtp_format_video_generic.cc',
'source/rtp_format_video_generic.h',
'source/ulpfec_generator.cc',
'source/ulpfec_generator.h',
'source/ulpfec_header_reader_writer.cc',
'source/ulpfec_header_reader_writer.h',
'source/ulpfec_receiver_impl.cc',
'source/ulpfec_receiver_impl.h',
'source/video_codec_information.h',
'source/vp8_partition_aggregator.cc',
'source/vp8_partition_aggregator.h',
# Mocks
'mocks/mock_rtp_rtcp.h',
'source/mock/mock_rtp_payload_strategy.h',
], # source
'conditions': [
['enable_bwe_test_logging==1', {
'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1' ],
}, {
'defines': [ 'BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0' ],
}],
],
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
}

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@ -1,26 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
# The test below takes long to run, no need to add it to any bot.
'target_name': 'test_packet_masks_metrics',
'type': 'executable',
'dependencies': [
'rtp_rtcp',
'<(webrtc_root)/test/test.gyp:test_support_main',
'<(DEPTH)/testing/gtest.gyp:gtest',
],
'sources': [
'test_packet_masks_metrics.cc',
'average_residual_loss_xor_codes.h',
],
},
],
}

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@ -1,37 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'webrtc_utility',
'type': 'static_library',
'dependencies': [
'audio_coding_module',
'media_file',
'<(webrtc_root)/base/base.gyp:rtc_task_queue',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/audio_frame_operations.h',
'include/file_player.h',
'include/file_recorder.h',
'include/helpers_android.h',
'include/helpers_ios.h',
'include/jvm_android.h',
'include/process_thread.h',
'source/helpers_android.cc',
'source/helpers_ios.mm',
'source/jvm_android.cc',
'source/process_thread_impl.cc',
'source/process_thread_impl.h',
],
},
], # targets
}

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@ -1,200 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
# Note this library is missing an implementation for the video capture.
# Targets must link with either 'video_capture' or
# 'video_capture_module_internal_impl' depending on whether they want to
# use the internal capturer.
'target_name': 'video_capture_module',
'type': 'static_library',
'dependencies': [
'webrtc_utility',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'cflags_mozilla': [
'$(NSPR_CFLAGS)',
],
'sources': [
'device_info_impl.cc',
'device_info_impl.h',
'video_capture.h',
'video_capture_config.h',
'video_capture_defines.h',
'video_capture_delay.h',
'video_capture_factory.h',
'video_capture_factory.cc',
'video_capture_impl.cc',
'video_capture_impl.h',
],
},
{
# Default video capture module implementation that only supports external
# capture.
'target_name': 'video_capture',
'type': 'static_library',
'dependencies': [
'video_capture_module',
],
'cflags_mozilla': [
'$(NSPR_CFLAGS)',
],
'sources': [
'external/device_info_external.cc',
'external/video_capture_external.cc',
],
},
], # targets
'conditions': [
['build_with_chromium==0', {
'targets': [
{
'target_name': 'video_capture_module_internal_impl',
'type': 'static_library',
'dependencies': [
'video_capture_module',
'<(webrtc_root)/common.gyp:webrtc_common',
],
'cflags_mozilla': [
'$(NSPR_CFLAGS)',
],
'conditions': [
['include_v4l2_video_capture==1', {
'sources': [
'linux/device_info_linux.cc',
'linux/device_info_linux.h',
'linux/video_capture_linux.cc',
'linux/video_capture_linux.h',
],
}],
['OS=="mac"', {
'sources': [
'mac/avfoundation/video_capture_avfoundation.h',
'mac/avfoundation/video_capture_avfoundation.mm',
'mac/avfoundation/video_capture_avfoundation_info.h',
'mac/avfoundation/video_capture_avfoundation_info.mm',
'mac/avfoundation/video_capture_avfoundation_info_objc.h',
'mac/avfoundation/video_capture_avfoundation_info_objc.mm',
'mac/avfoundation/video_capture_avfoundation_objc.h',
'mac/avfoundation/video_capture_avfoundation_objc.mm',
'mac/avfoundation/video_capture_avfoundation_utility.h',
'mac/video_capture_mac.mm',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework Cocoa',
'-framework CoreVideo',
],
},
},
}], # mac
['OS=="win"', {
'conditions': [
['build_with_mozilla==0', {
'dependencies': [
'<(DEPTH)/third_party/winsdk_samples/winsdk_samples.gyp:directshow_baseclasses',
],
}],
],
'sources': [
'windows/device_info_ds.cc',
'windows/device_info_ds.h',
'windows/device_info_mf.cc',
'windows/device_info_mf.h',
'windows/help_functions_ds.cc',
'windows/help_functions_ds.h',
'windows/sink_filter_ds.cc',
'windows/sink_filter_ds.h',
'windows/video_capture_ds.cc',
'windows/video_capture_ds.h',
'windows/video_capture_factory_windows.cc',
'windows/video_capture_mf.cc',
'windows/video_capture_mf.h',
'windows/BasePin.cpp',
'windows/BaseFilter.cpp',
'windows/BaseInputPin.cpp',
'windows/MediaType.cpp',
],
'link_settings': {
'libraries': [
'-lStrmiids.lib',
],
},
}], # win
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-comment',
'-Wno-ignored-attributes',
'-Wno-microsoft-extra-qualification',
'-Wno-missing-braces',
'-Wno-overloaded-virtual',
'-Wno-reorder',
'-Wno-writable-strings',
],
},
},
}],
['OS=="android"', {
'sources': [
'android/device_info_android.cc',
'android/device_info_android.h',
'android/video_capture_android.cc',
'android/video_capture_android.h',
],
}], # android
['OS=="ios"', {
'sources': [
'objc/device_info.h',
'objc/device_info.mm',
'objc/device_info_objc.h',
'objc/device_info_objc.mm',
'objc/rtc_video_capture_objc.h',
'objc/rtc_video_capture_objc.mm',
'objc/video_capture.h',
'objc/video_capture.mm',
],
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
'cflags_mozilla': [
'-fobjc-arc',
],
'all_dependent_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework AVFoundation',
'-framework CoreMedia',
'-framework CoreVideo',
],
},
},
}], # ios
['OS=="ios"', {
'all_dependent_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework UIKit',
],
},
},
}], # ios
], # conditions
},
],
}], # build_with_chromium==0
],
}

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@ -1,97 +0,0 @@
# Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../../build/common.gypi',
],
'targets': [
{
'target_name': 'webrtc_h264',
'type': 'static_library',
'conditions': [
['OS=="ios"', {
'dependencies': [
'webrtc_h264_video_toolbox',
],
'sources': [
'h264_objc.mm',
],
}],
#TODO: rtc_use_h264 not defined here
# TODO(hbos): Consider renaming this flag and the below macro to
# something which helps distinguish OpenH264/FFmpeg from other H264
# implementations.
#['rtc_use_h264==1', {
# 'defines': [
# 'WEBRTC_USE_H264',
# ],
# 'conditions': [
# ['rtc_initialize_ffmpeg==1', {
# 'defines': [
# 'WEBRTC_INITIALIZE_FFMPEG',
# ],
# }],
# ],
# 'dependencies': [
# '<(DEPTH)/third_party/ffmpeg/ffmpeg.gyp:ffmpeg',
# '<(DEPTH)/third_party/openh264/openh264.gyp:openh264_encoder',
# '<(webrtc_root)/common_video/common_video.gyp:common_video',
# ],
# 'sources': [
# 'h264_decoder_impl.cc',
# 'h264_decoder_impl.h',
# 'h264_encoder_impl.cc',
# 'h264_encoder_impl.h',
# ],
#}],
],
'sources': [
'h264.cc',
'include/h264.h',
],
}, # webrtc_h264
],
'conditions': [
['OS=="ios"', {
'targets': [
{
'target_name': 'webrtc_h264_video_toolbox',
'type': 'static_library',
'includes': [ '../../../../build/objc_common.gypi' ],
'dependencies': [
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreFoundation',
'-framework CoreMedia',
'-framework CoreVideo',
'-framework VideoToolbox',
],
},
},
'sources': [
'h264_video_toolbox_decoder.cc',
'h264_video_toolbox_decoder.h',
'h264_video_toolbox_encoder.h',
'h264_video_toolbox_encoder.mm',
'h264_video_toolbox_nalu.cc',
'h264_video_toolbox_nalu.h',
],
'conditions': [
['build_libyuv==1', {
'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv'],
}],
],
}, # webrtc_h264_video_toolbox
], # targets
}], # OS=="ios"
], # conditions
}

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@ -1,23 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'webrtc_i420',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/i420.h',
'i420.cc',
],
},
],
}

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@ -1,34 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'video_codecs_test_framework',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/test/test.gyp:test_support',
],
'sources': [
'mock/mock_packet_manipulator.h',
'packet_manipulator.h',
'packet_manipulator.cc',
'predictive_packet_manipulator.h',
'predictive_packet_manipulator.cc',
'stats.h',
'stats.cc',
'videoprocessor.h',
'videoprocessor.cc',
],
},
], # targets
}], # include_tests
], # conditions
}

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@ -1,36 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'video_quality_measurement',
'type': 'executable',
'dependencies': [
'video_codecs_test_framework',
'webrtc_video_coding',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:test_support',
'<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8',
],
'sources': [
'video_quality_measurement.cc',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
], # targets
}], # include_tests
], # conditions
}

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@ -1,82 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../../build/common.gypi',
],
'targets': [
{
'target_name': 'webrtc_vp8',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'include_dirs': [
'../../../../../../../libyuv/libyuv/include',
],
'conditions': [
['build_libvpx==1', {
'dependencies': [
'<(libvpx_dir)/libvpx.gyp:libvpx',
],
},{
'link_settings': {
'libraries': [
'$(LIBVPX_OBJ)/libvpx.a',
],
},
}],
],
'sources': [
'default_temporal_layers.cc',
'default_temporal_layers.h',
'include/vp8.h',
'include/vp8_common_types.h',
'realtime_temporal_layers.cc',
'reference_picture_selection.cc',
'reference_picture_selection.h',
'screenshare_layers.cc',
'screenshare_layers.h',
'simulcast_encoder_adapter.cc',
'simulcast_encoder_adapter.h',
'temporal_layers.h',
'vp8_impl.cc',
'vp8_impl.h',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
},
], # targets
'conditions': [
['include_tests==1', {
'targets': [
{
'target_name': 'vp8_coder',
'type': 'executable',
'dependencies': [
'webrtc_vp8',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/test/test.gyp:test_support_main',
'<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
],
'sources': [
'vp8_sequence_coder.cc',
],
},
], # targets
}], # include_tests
],
}

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@ -1,54 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../../build/common.gypi',
],
'targets': [
{
'target_name': 'webrtc_vp9',
'type': 'static_library',
'conditions': [
['build_libvpx==1', {
'dependencies': [
'<(libvpx_dir)/libvpx.gyp:libvpx',
],
}, {
'include_dirs': [
'$(MOZ_LIBVPX_CFLAGS)',
'<(libvpx_dir)',
],
}],
['libvpx_build_vp9==1', {
'sources': [
'screenshare_layers.cc',
'screenshare_layers.h',
'vp9_frame_buffer_pool.cc',
'vp9_frame_buffer_pool.h',
'vp9_impl.cc',
'vp9_impl.h',
],
}, {
'sources': [
'vp9_noop.cc',
],
}
],
],
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/vp9.h',
],
},
],
}

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@ -1,40 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../../../build/common.gypi',
],
'targets': [
{
'target_name': 'video_coding_utility',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'default_video_bitrate_allocator.cc',
'frame_dropper.cc',
'frame_dropper.h',
'ivf_file_writer.cc',
'ivf_file_writer.h',
'moving_average.cc',
'moving_average.h',
'qp_parser.cc',
'qp_parser.h',
'quality_scaler.cc',
'quality_scaler.h',
'simulcast_rate_allocator.cc',
'simulcast_rate_allocator.h',
'vp8_header_parser.cc',
'vp8_header_parser.h',
],
},
], # targets
}

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@ -1,98 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'webrtc_video_coding',
'type': 'static_library',
'dependencies': [
'webrtc_h264',
'webrtc_i420',
'../base/base.gyp:rtc_task_queue',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/video_coding/utility/video_coding_utility.gyp:video_coding_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_vp8_dir)/vp8.gyp:webrtc_vp8',
'<(webrtc_vp9_dir)/vp9.gyp:webrtc_vp9',
],
'sources': [
# interfaces
'include/video_coding.h',
'include/video_coding_defines.h',
# headers
'codec_database.h',
'codec_timer.h',
'decoding_state.h',
'encoded_frame.h',
'fec_rate_table.h',
'frame_buffer.h',
'frame_buffer2.h',
'frame_object.h',
'rtp_frame_reference_finder.h',
'generic_decoder.h',
'generic_encoder.h',
'histogram.h',
'inter_frame_delay.h',
'internal_defines.h',
'jitter_buffer.h',
'jitter_buffer_common.h',
'jitter_estimator.h',
'media_opt_util.h',
'media_optimization.h',
'nack_fec_tables.h',
'nack_module.h',
'packet.h',
'packet_buffer.h',
'protection_bitrate_calculator.h',
'receiver.h',
'rtt_filter.h',
'session_info.h',
'timestamp_map.h',
'timing.h',
'video_coding_impl.h',
# sources
'codec_database.cc',
'codec_timer.cc',
'decoding_state.cc',
'encoded_frame.cc',
'frame_buffer.cc',
'frame_buffer2.cc',
'frame_object.cc',
'rtp_frame_reference_finder.cc',
'generic_decoder.cc',
'generic_encoder.cc',
'h264_sprop_parameter_sets.cc',
'h264_sps_pps_tracker.cc',
'inter_frame_delay.cc',
'histogram.cc',
'jitter_buffer.cc',
'jitter_estimator.cc',
'media_opt_util.cc',
'media_optimization.cc',
'protection_bitrate_calculator.cc',
'nack_module.cc',
'packet.cc',
'packet_buffer.cc',
'receiver.cc',
'rtt_filter.cc',
'session_info.cc',
'timestamp_map.cc',
'timing.cc',
'video_codec_initializer.cc',
'video_coding_impl.cc',
'video_sender.cc',
'video_receiver.cc',
], # source
# TODO(jschuh): Bug 1348: fix size_t to int truncations.
'msvs_disabled_warnings': [ 4267, ],
},
],
}

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@ -1,35 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'rtp_player',
'type': 'executable',
'dependencies': [
'rtp_rtcp',
'webrtc_video_coding',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'<(webrtc_root)/test/test.gyp:test_common',
],
'sources': [
# headers
'test/receiver_tests.h',
'test/rtp_player.h',
'test/vcm_payload_sink_factory.h',
# sources
'test/rtp_player.cc',
'test/test_util.cc',
'test/tester_main.cc',
'test/vcm_payload_sink_factory.cc',
'test/video_rtp_play.cc',
], # sources
},
],
}

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@ -1,86 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'video_processing',
'type': 'static_library',
'include_dirs': [
'<(libyuv_dir)/include',
],
'dependencies': [
'webrtc_utility',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
'sources': [
'include/video_processing.h',
'include/video_processing_defines.h',
'video_denoiser.cc',
'video_denoiser.h',
'util/denoiser_filter.cc',
'util/denoiser_filter.h',
'util/denoiser_filter_c.cc',
'util/denoiser_filter_c.h',
'util/noise_estimation.cc',
'util/noise_estimation.h',
'util/skin_detection.cc',
'util/skin_detection.h',
],
'conditions': [
['target_arch=="ia32" or target_arch=="x64"', {
'dependencies': [ 'video_processing_sse2', ],
}],
['target_arch=="arm" or target_arch == "arm64"', {
'dependencies': [ 'video_processing_neon', ],
}],
],
},
],
'conditions': [
['target_arch=="ia32" or target_arch=="x64"', {
'targets': [
{
'target_name': 'video_processing_sse2',
'type': 'static_library',
'sources': [
'util/denoiser_filter_sse2.cc',
'util/denoiser_filter_sse2.h',
],
'conditions': [
['os_posix==1 and OS!="mac"', {
'cflags': [ '-msse2', ],
'cflags_mozilla': [ '-msse2', ],
}],
['OS=="mac"', {
'xcode_settings': {
'OTHER_CFLAGS': [ '-msse2', ],
},
}],
],
},
],
}],
['target_arch=="arm" or target_arch == "arm64"', {
'targets': [
{
'target_name': 'video_processing_neon',
'type': 'static_library',
'includes': [ '../../build/arm_neon.gypi', ],
'sources': [
'util/denoiser_filter_neon.cc',
'util/denoiser_filter_neon.h',
],
},
],
}],
],
}

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@ -1,145 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [ '../build/common.gypi', ],
'targets': [
{
'target_name': 'rtc_p2p',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/common.gyp:webrtc_common',
],
'sources': [
'base/asyncstuntcpsocket.cc',
'base/asyncstuntcpsocket.h',
'base/basicpacketsocketfactory.cc',
'base/basicpacketsocketfactory.h',
'base/candidate.h',
'base/common.h',
'base/dtlstransport.h',
'base/dtlstransportchannel.cc',
'base/dtlstransportchannel.h',
'base/p2pconstants.cc',
'base/p2pconstants.h',
'base/p2ptransport.cc',
'base/p2ptransport.h',
'base/p2ptransportchannel.cc',
'base/p2ptransportchannel.h',
'base/packetsocketfactory.h',
'base/port.cc',
'base/port.h',
'base/portallocator.cc',
'base/portallocator.h',
'base/portinterface.h',
'base/pseudotcp.cc',
'base/pseudotcp.h',
'base/relayport.cc',
'base/relayport.h',
'base/session.cc',
'base/session.h',
'base/sessiondescription.cc',
'base/sessiondescription.h',
'base/stun.cc',
'base/stun.h',
'base/stunport.cc',
'base/stunport.h',
'base/stunrequest.cc',
'base/stunrequest.h',
'base/tcpport.cc',
'base/tcpport.h',
'base/transport.cc',
'base/transport.h',
'base/transportchannel.cc',
'base/transportchannel.h',
'base/transportchannelimpl.h',
'base/transportcontroller.cc',
'base/transportcontroller.h',
'base/transportdescription.cc',
'base/transportdescription.h',
'base/transportdescriptionfactory.cc',
'base/transportdescriptionfactory.h',
'base/transportinfo.h',
'base/turnport.cc',
'base/turnport.h',
'base/udpport.h',
'client/basicportallocator.cc',
'client/basicportallocator.h',
'client/httpportallocator.cc',
'client/httpportallocator.h',
'client/socketmonitor.cc',
'client/socketmonitor.h',
],
'direct_dependent_settings': {
'defines': [
'FEATURE_ENABLE_VOICEMAIL',
],
},
'conditions': [
['build_with_chromium==0', {
'sources': [
'base/relayserver.cc',
'base/relayserver.h',
'base/stunserver.cc',
'base/stunserver.h',
'base/turnserver.cc',
'base/turnserver.h',
],
'defines': [
'FEATURE_ENABLE_VOICEMAIL',
'FEATURE_ENABLE_PSTN',
],
}],
['use_quic==1', {
'dependencies': [
'<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
],
'sources': [
'quic/quicconnectionhelper.cc',
'quic/quicconnectionhelper.h',
'quic/quicsession.cc',
'quic/quicsession.h',
'quic/quictransport.cc',
'quic/quictransport.h',
'quic/quictransportchannel.cc',
'quic/quictransportchannel.h',
'quic/reliablequicstream.cc',
'quic/reliablequicstream.h',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/libquic/libquic.gyp:libquic',
],
}],
],
},
{
'target_name': 'libstunprober',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/common.gyp:webrtc_common',
],
'sources': [
'stunprober/stunprober.cc',
],
},
{
'target_name': 'stun_prober',
'type': 'executable',
'dependencies': [
'libstunprober',
'rtc_p2p'
],
'sources': [
'stunprober/main.cc',
],
},
], # targets
}

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@ -1,77 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': ['../build/common.gypi'],
'variables': {
'rtc_pc_defines': [
'SRTP_RELATIVE_PATH',
'HAVE_SCTP',
'HAVE_SRTP',
],
},
'targets': [
{
'target_name': 'rtc_pc',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/media/media.gyp:rtc_media',
],
'conditions': [
['build_with_chromium==1', {
'sources': [
'externalhmac.h',
'externalhmac.cc',
],
}],
# TODO: lisrtp.gyp not found
#['build_libsrtp==1', {
# 'dependencies': [
# '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
# ],
#}],
],
'defines': [
'<@(rtc_pc_defines)',
],
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
'direct_dependent_settings': {
'defines': [
'<@(rtc_pc_defines)'
],
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
},
'sources': [
'audiomonitor.cc',
'audiomonitor.h',
'bundlefilter.cc',
'bundlefilter.h',
'channel.cc',
'channel.h',
'channelmanager.cc',
'channelmanager.h',
'currentspeakermonitor.cc',
'currentspeakermonitor.h',
'mediamonitor.cc',
'mediamonitor.h',
'mediasession.cc',
'mediasession.h',
'rtcpmuxfilter.cc',
'rtcpmuxfilter.h',
'srtpfilter.cc',
'srtpfilter.h',
'voicechannel.h',
],
}, # target rtc_pc
], # targets
}

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@ -1,342 +0,0 @@
# Copyright 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../build/common.gypi',
'sdk.gypi',
],
'conditions': [
['OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")', {
'targets': [
{
'target_name': 'rtc_sdk_common_objc',
'type': 'static_library',
'includes': [ '../build/objc_common.gypi' ],
'dependencies': [
'../base/base.gyp:rtc_base',
],
'include_dirs': [
'objc/Framework/Classes',
'objc/Framework/Headers',
],
'direct_dependent_settings': {
'include_dirs': [
'objc/Framework/Classes',
'objc/Framework/Headers',
],
},
'sources': [
'objc/Framework/Classes/NSString+StdString.h',
'objc/Framework/Classes/NSString+StdString.mm',
'objc/Framework/Classes/RTCDispatcher.m',
'objc/Framework/Classes/RTCFieldTrials.mm',
'objc/Framework/Classes/RTCLogging.mm',
'objc/Framework/Classes/RTCMetrics.mm',
'objc/Framework/Classes/RTCMetricsSampleInfo+Private.h',
'objc/Framework/Classes/RTCMetricsSampleInfo.mm',
'objc/Framework/Classes/RTCSSLAdapter.mm',
'objc/Framework/Classes/RTCTracing.mm',
'objc/Framework/Headers/WebRTC/RTCDispatcher.h',
'objc/Framework/Headers/WebRTC/RTCFieldTrials.h',
'objc/Framework/Headers/WebRTC/RTCLogging.h',
'objc/Framework/Headers/WebRTC/RTCMacros.h',
'objc/Framework/Headers/WebRTC/RTCMetrics.h',
'objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h',
'objc/Framework/Headers/WebRTC/RTCSSLAdapter.h',
'objc/Framework/Headers/WebRTC/RTCTracing.h',
],
'conditions': [
['OS=="ios"', {
'sources': [
'objc/Framework/Classes/RTCCameraPreviewView.m',
'objc/Framework/Classes/RTCUIApplication.h',
'objc/Framework/Classes/RTCUIApplication.mm',
'objc/Framework/Classes/UIDevice+RTCDevice.mm',
'objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h',
'objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework AVFoundation',
],
},
},
}], # OS=="ios"
['build_with_chromium==0', {
'sources': [
'objc/Framework/Classes/RTCFileLogger.mm',
'objc/Framework/Headers/WebRTC/RTCFileLogger.h',
],
}],
],
},
{
'target_name': 'rtc_sdk_peerconnection_objc',
'type': 'static_library',
'includes': [ '../build/objc_common.gypi' ],
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
'rtc_sdk_common_objc',
],
'include_dirs': [
'objc/Framework/Classes',
'objc/Framework/Headers',
],
'direct_dependent_settings': {
'include_dirs': [
'objc/Framework/Classes',
'objc/Framework/Headers',
],
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework AVFoundation',
],
},
'libraries': [
'-lstdc++',
],
}, # link_settings
'sources': [
'objc/Framework/Classes/RTCAVFoundationVideoSource+Private.h',
'objc/Framework/Classes/RTCAVFoundationVideoSource.mm',
'objc/Framework/Classes/RTCAudioSource+Private.h',
'objc/Framework/Classes/RTCAudioSource.mm',
'objc/Framework/Classes/RTCAudioTrack+Private.h',
'objc/Framework/Classes/RTCAudioTrack.mm',
'objc/Framework/Classes/RTCConfiguration+Private.h',
'objc/Framework/Classes/RTCConfiguration.mm',
'objc/Framework/Classes/RTCDataChannel+Private.h',
'objc/Framework/Classes/RTCDataChannel.mm',
'objc/Framework/Classes/RTCDataChannelConfiguration+Private.h',
'objc/Framework/Classes/RTCDataChannelConfiguration.mm',
'objc/Framework/Classes/RTCI420Shader.mm',
'objc/Framework/Classes/RTCIceCandidate+Private.h',
'objc/Framework/Classes/RTCIceCandidate.mm',
'objc/Framework/Classes/RTCIceServer+Private.h',
'objc/Framework/Classes/RTCIceServer.mm',
'objc/Framework/Classes/RTCLegacyStatsReport+Private.h',
'objc/Framework/Classes/RTCLegacyStatsReport.mm',
'objc/Framework/Classes/RTCMediaConstraints+Private.h',
'objc/Framework/Classes/RTCMediaConstraints.mm',
'objc/Framework/Classes/RTCMediaSource+Private.h',
'objc/Framework/Classes/RTCMediaSource.mm',
'objc/Framework/Classes/RTCMediaStream+Private.h',
'objc/Framework/Classes/RTCMediaStream.mm',
'objc/Framework/Classes/RTCMediaStreamTrack+Private.h',
'objc/Framework/Classes/RTCMediaStreamTrack.mm',
'objc/Framework/Classes/RTCOpenGLDefines.h',
'objc/Framework/Classes/RTCOpenGLVideoRenderer.h',
'objc/Framework/Classes/RTCOpenGLVideoRenderer.mm',
'objc/Framework/Classes/RTCPeerConnection+DataChannel.mm',
'objc/Framework/Classes/RTCPeerConnection+Private.h',
'objc/Framework/Classes/RTCPeerConnection+Stats.mm',
'objc/Framework/Classes/RTCPeerConnection.mm',
'objc/Framework/Classes/RTCPeerConnectionFactory+Private.h',
'objc/Framework/Classes/RTCPeerConnectionFactory.mm',
'objc/Framework/Classes/RTCRtpCodecParameters+Private.h',
'objc/Framework/Classes/RTCRtpCodecParameters.mm',
'objc/Framework/Classes/RTCRtpEncodingParameters+Private.h',
'objc/Framework/Classes/RTCRtpEncodingParameters.mm',
'objc/Framework/Classes/RTCRtpParameters+Private.h',
'objc/Framework/Classes/RTCRtpParameters.mm',
'objc/Framework/Classes/RTCRtpReceiver+Private.h',
'objc/Framework/Classes/RTCRtpReceiver.mm',
'objc/Framework/Classes/RTCRtpSender+Private.h',
'objc/Framework/Classes/RTCRtpSender.mm',
'objc/Framework/Classes/RTCSessionDescription+Private.h',
'objc/Framework/Classes/RTCSessionDescription.mm',
'objc/Framework/Classes/RTCShader+Private.h',
'objc/Framework/Classes/RTCShader.h',
'objc/Framework/Classes/RTCShader.mm',
'objc/Framework/Classes/RTCVideoFrame+Private.h',
'objc/Framework/Classes/RTCVideoFrame.mm',
'objc/Framework/Classes/RTCVideoRendererAdapter+Private.h',
'objc/Framework/Classes/RTCVideoRendererAdapter.h',
'objc/Framework/Classes/RTCVideoRendererAdapter.mm',
'objc/Framework/Classes/RTCVideoSource+Private.h',
'objc/Framework/Classes/RTCVideoSource.mm',
'objc/Framework/Classes/RTCVideoTrack+Private.h',
'objc/Framework/Classes/RTCVideoTrack.mm',
'objc/Framework/Classes/avfoundationvideocapturer.h',
'objc/Framework/Classes/avfoundationvideocapturer.mm',
'objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h',
'objc/Framework/Headers/WebRTC/RTCAudioSource.h',
'objc/Framework/Headers/WebRTC/RTCAudioTrack.h',
'objc/Framework/Headers/WebRTC/RTCConfiguration.h',
'objc/Framework/Headers/WebRTC/RTCDataChannel.h',
'objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h',
'objc/Framework/Headers/WebRTC/RTCIceCandidate.h',
'objc/Framework/Headers/WebRTC/RTCIceServer.h',
'objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h',
'objc/Framework/Headers/WebRTC/RTCMediaConstraints.h',
'objc/Framework/Headers/WebRTC/RTCMediaSource.h',
'objc/Framework/Headers/WebRTC/RTCMediaStream.h',
'objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h',
'objc/Framework/Headers/WebRTC/RTCPeerConnection.h',
'objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h',
'objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h',
'objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h',
'objc/Framework/Headers/WebRTC/RTCRtpParameters.h',
'objc/Framework/Headers/WebRTC/RTCRtpReceiver.h',
'objc/Framework/Headers/WebRTC/RTCRtpSender.h',
'objc/Framework/Headers/WebRTC/RTCSessionDescription.h',
'objc/Framework/Headers/WebRTC/RTCVideoFrame.h',
'objc/Framework/Headers/WebRTC/RTCVideoRenderer.h',
'objc/Framework/Headers/WebRTC/RTCVideoSource.h',
'objc/Framework/Headers/WebRTC/RTCVideoTrack.h',
], # sources
'conditions': [
['build_libyuv==1', {
'dependencies': ['<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv'],
}],
['OS=="ios"', {
'sources': [
'objc/Framework/Classes/RTCEAGLVideoView.m',
'objc/Framework/Classes/RTCNativeNV12Shader.mm',
'objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreGraphics',
'-framework GLKit',
'-framework OpenGLES',
'-framework QuartzCore',
],
},
}, # link_settings
}], # OS=="ios"
['OS=="mac"', {
'sources': [
'objc/Framework/Classes/RTCNSGLVideoView.m',
'objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreMedia',
'-framework OpenGL',
],
},
},
}],
], # conditions
}, # rtc_sdk_peerconnection_objc
{
'target_name': 'rtc_sdk_framework_objc',
'type': 'shared_library',
'product_name': 'WebRTC',
'mac_bundle': 1,
'includes': [ '../build/objc_common.gypi' ],
# Slightly hacky, but we need to re-declare files here that are C
# interfaces because otherwise they will be dead-stripped during
# linking (ObjC classes cannot be dead-stripped). We might consider
# just only using ObjC interfaces.
'sources': [
'objc/Framework/Classes/RTCFieldTrials.mm',
'objc/Framework/Classes/RTCLogging.mm',
'objc/Framework/Classes/RTCMetrics.mm',
'objc/Framework/Classes/RTCSSLAdapter.mm',
'objc/Framework/Classes/RTCTracing.mm',
'objc/Framework/Headers/WebRTC/RTCFieldTrials.h',
'objc/Framework/Headers/WebRTC/RTCLogging.h',
'objc/Framework/Headers/WebRTC/RTCSSLAdapter.h',
'objc/Framework/Headers/WebRTC/RTCTracing.h',
'objc/Framework/Headers/WebRTC/WebRTC.h',
'objc/Framework/Modules/module.modulemap',
],
'mac_framework_headers': [
'objc/Framework/Headers/WebRTC/RTCAudioSource.h',
'objc/Framework/Headers/WebRTC/RTCAudioTrack.h',
'objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h',
'objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h',
'objc/Framework/Headers/WebRTC/RTCConfiguration.h',
'objc/Framework/Headers/WebRTC/RTCDataChannel.h',
'objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h',
'objc/Framework/Headers/WebRTC/RTCDispatcher.h',
'objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h',
'objc/Framework/Headers/WebRTC/RTCFieldTrials.h',
'objc/Framework/Headers/WebRTC/RTCFileLogger.h',
'objc/Framework/Headers/WebRTC/RTCIceCandidate.h',
'objc/Framework/Headers/WebRTC/RTCIceServer.h',
'objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h',
'objc/Framework/Headers/WebRTC/RTCLogging.h',
'objc/Framework/Headers/WebRTC/RTCMacros.h',
'objc/Framework/Headers/WebRTC/RTCMediaConstraints.h',
'objc/Framework/Headers/WebRTC/RTCMediaSource.h',
'objc/Framework/Headers/WebRTC/RTCMediaStream.h',
'objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h',
'objc/Framework/Headers/WebRTC/RTCMetrics.h',
'objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h',
'objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h',
'objc/Framework/Headers/WebRTC/RTCPeerConnection.h',
'objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h',
'objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h',
'objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h',
'objc/Framework/Headers/WebRTC/RTCRtpParameters.h',
'objc/Framework/Headers/WebRTC/RTCRtpReceiver.h',
'objc/Framework/Headers/WebRTC/RTCRtpSender.h',
'objc/Framework/Headers/WebRTC/RTCSessionDescription.h',
'objc/Framework/Headers/WebRTC/RTCSSLAdapter.h',
'objc/Framework/Headers/WebRTC/RTCTracing.h',
'objc/Framework/Headers/WebRTC/RTCVideoFrame.h',
'objc/Framework/Headers/WebRTC/RTCVideoRenderer.h',
'objc/Framework/Headers/WebRTC/RTCVideoSource.h',
'objc/Framework/Headers/WebRTC/RTCVideoTrack.h',
'objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h',
'objc/Framework/Headers/WebRTC/WebRTC.h',
],
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
'rtc_sdk_peerconnection_objc',
],
'xcode_settings': {
'CODE_SIGNING_REQUIRED': 'NO',
'CODE_SIGN_IDENTITY': '',
'DEFINES_MODULE': 'YES',
'INFOPLIST_FILE': 'objc/Framework/Info.plist',
'LD_DYLIB_INSTALL_NAME': '@rpath/WebRTC.framework/WebRTC',
'MODULEMAP_FILE': '<(webrtc_root)/sdk/Framework/Modules/module.modulemap',
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework AVFoundation',
'-framework AudioToolbox',
'-framework CoreGraphics',
'-framework CoreMedia',
'-framework GLKit',
'-framework VideoToolbox',
],
},
}, # link_settings
'conditions': [
# TODO(tkchin): Generate WebRTC.h based off of
# mac_framework_headers instead of hard-coding. Ok for now since we
# only really care about dynamic lib on iOS outside of chromium.
['OS!="mac"', {
'mac_framework_headers!': [
'objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h',
],
}],
['build_with_chromium==1', {
'mac_framework_headers!': [
'objc/Framework/Headers/WebRTC/RTCFileLogger.h',
],
}],
], # conditions
}, # rtc_sdk_framework_objc
], # targets
}], # OS=="ios" or (OS=="mac" and mac_deployment_target=="10.7")
],
}

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@ -1,26 +0,0 @@
# Copyright 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'target_defaults': {
'configurations': {
'Profile': {
'xcode_settings': {
'DEBUG_INFORMARTION_FORMAT': 'dwarf-with-dsym',
# We manually strip using strip -S and strip -x. We need to run
# dsymutil ourselves so we need symbols around before that.
'DEPLOYMENT_POSTPROCESSING': 'NO',
'GCC_OPTIMIZATION_LEVEL': 's',
'GCC_SYMBOLS_PRIVATE_EXTERN': 'YES',
'STRIP_INSTALLED_PRODUCT': 'NO',
'USE_HEADERMAP': 'YES',
},
},
},
},
}

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@ -1,27 +0,0 @@
# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [ '../build/common.gypi', ],
'targets': [
{
# GN version: webrtc/stats:rtc_stats
'target_name': 'rtc_stats',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/api/api.gyp:rtc_stats_api',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
],
'sources': [
'rtcstats.cc',
'rtcstats_objects.cc',
'rtcstatsreport.cc',
],
},
],
}

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@ -1,64 +0,0 @@
{
'variables': {
'variables': {
'webrtc_root%': '<(DEPTH)', # '<(DEPTH)/webrtc',
# Override the defaults in Chromium's build/common.gypi.
# Needed for ARC and libc++.
'mac_sdk_min%': '10.11',
'mac_deployment_target%': '10.7',
# Disable use of sysroot for Linux. It's enabled by default in Chromium,
# but it currently lacks the libudev-dev package.
# TODO(kjellander): Remove when crbug.com/561584 is fixed.
'use_sysroot': 0,
},
'webrtc_root%': '<(webrtc_root)',
'mac_deployment_target%': '<(mac_deployment_target)',
'use_sysroot%': '<(use_sysroot)',
'build_with_chromium': 0,
'conditions': [
['OS=="ios"', {
# Set target_subarch for if not already set. This is needed because the
# Chromium iOS toolchain relies on target_subarch being set.
'conditions': [
['target_arch=="arm" or target_arch=="ia32"', {
'target_subarch%': 'arm32',
}],
['target_arch=="arm64" or target_arch=="x64"', {
'target_subarch%': 'arm64',
}],
],
}],
['OS=="android"', {
# MJPEG capture is not used on Android. Disable to reduce
# libjingle_peerconnection_so file size.
'libyuv_disable_jpeg%': 1,
}],
],
},
'target_defaults': {
'target_conditions': [
['_target_name=="sanitizer_options"', {
'conditions': [
['lsan==1', {
# Replace Chromium's LSan suppressions with our own for WebRTC.
'sources/': [
['exclude', 'lsan_suppressions.cc'],
],
'sources': [
'<(webrtc_root)/build/sanitizers/lsan_suppressions_webrtc.cc',
],
}],
['tsan==1', {
# Replace Chromium's TSan v2 suppressions with our own for WebRTC.
'sources/': [
['exclude', 'tsan_suppressions.cc'],
],
'sources': [
'<(webrtc_root)/build/sanitizers/tsan_suppressions_webrtc.cc',
],
}],
],
}],
],
},
}

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@ -1,26 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'cpu_features_android',
'type': 'static_library',
'sources': [
'source/cpu_features_android.c',
],
'dependencies': [
'../../../build/android/ndk.gyp:cpu_features',
],
},
],
}],
], # conditions
}

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@ -1,56 +0,0 @@
# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'include_ndk_cpu_features%': 0,
},
'includes': [ '../build/common.gypi', ],
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'cpu_features_android',
'type': 'static_library',
'sources': [
'source/cpu_features_android.c',
],
'conditions': [
['include_ndk_cpu_features==1', {
'includes': [
'../../build/android/cpufeatures.gypi',
],
}, {
'sources': [
'source/droid-cpu-features.c',
'source/droid-cpu-features.h',
],
}],
'dependencies': [
# Not supported, please refer to the GN build.
#'../../build/android/ndk.gyp:cpu_features',
],
},
],
}],
['OS=="linux"', {
'targets': [
{
'target_name': 'cpu_features_linux',
'type': 'static_library',
'sources': [
'source/cpu_features_linux.c',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
],
},
],
}],
], # conditions
}

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@ -1,194 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [ '../build/common.gypi', ],
'targets': [
{
'target_name': 'system_wrappers',
'type': 'static_library',
'dependencies': [
# '<(webrtc_root)/common.gyp:webrtc_common',
# '../base/base.gyp:rtc_base_approved',
'field_trial_default',
],
'sources': [
'include/aligned_array.h',
'include/aligned_malloc.h',
'include/atomic32.h',
'include/clock.h',
'include/cpu_features_wrapper.h',
'include/cpu_info.h',
'include/critical_section_wrapper.h',
'include/data_log.h',
'include/data_log_c.h',
'include/data_log_impl.h',
'include/event_wrapper.h',
'include/field_trial.h',
'include/file_wrapper.h',
'include/fix_interlocked_exchange_pointer_win.h',
'include/logging.h',
'include/metrics.h',
'include/ntp_time.h',
'include/rtp_to_ntp.h',
'include/rw_lock_wrapper.h',
'include/sleep.h',
'include/sort.h',
'include/static_instance.h',
'include/stl_util.h',
'include/stringize_macros.h',
'include/timestamp_extrapolator.h',
'include/trace.h',
'include/utf_util_win.h',
'source/aligned_malloc.cc',
'source/atomic32_win.cc',
'source/clock.cc',
'source/condition_variable_event_win.cc',
'source/condition_variable_event_win.h',
'source/cpu_features.cc',
'source/cpu_info.cc',
# TODO: removed
# 'source/data_log_c.cc',
'source/event.cc',
'source/event_timer_posix.cc',
'source/event_timer_posix.h',
'source/event_timer_win.cc',
'source/event_timer_win.h',
'source/file_impl.cc',
'source/logging.cc',
'source/rtp_to_ntp_estimator.cc',
'source/rw_lock.cc',
'source/rw_lock_posix.cc',
'source/rw_lock_posix.h',
'source/rw_lock_win.cc',
'source/rw_lock_win.h',
'source/rw_lock_winxp_win.cc',
'source/rw_lock_winxp_win.h',
'source/sleep.cc',
# TODO: removed
# 'source/sort.cc',
'source/timestamp_extrapolator.cc',
'source/trace_impl.cc',
'source/trace_impl.h',
'source/trace_posix.cc',
'source/trace_posix.h',
'source/trace_win.cc',
'source/trace_win.h',
],
'conditions': [
#TODO: missing
#['enable_data_logging==1', {
# 'sources': [ 'source/data_log.cc', ],
#}, {
# 'sources': [ 'source/data_log_no_op.cc', ],
#},],
['build_with_mozilla', {
'sources': [
'source/metrics_default.cc',
],
}],
['OS=="android"', {
'defines': [
'WEBRTC_THREAD_RR',
],
'conditions': [
['build_with_chromium==1', {
'dependencies': [
# 'cpu_features_chromium.gyp:cpu_features_android',
],
}, {
'dependencies': [
# 'cpu_features_webrtc.gyp:cpu_features_android',
],
}],
],
'link_settings': {
'libraries': [
'-llog',
],
},
'sources': [
'include/logcat_trace_context.h',
'source/logcat_trace_context.cc',
],
}],
['OS=="linux"', {
'defines': [
'WEBRTC_THREAD_RR',
],
'conditions': [
['build_with_chromium==0', {
'dependencies': [
# 'cpu_features_webrtc.gyp:cpu_features_linux',
],
}],
],
'link_settings': {
'libraries': [ '-lrt', ],
},
}],
['OS=="mac"', {
'link_settings': {
'libraries': [ '$(SDKROOT)/System/Library/Frameworks/ApplicationServices.framework', ],
},
}],
['os_bsd==1', {
'defines': [
'WEBRTC_THREAD_RR',
],
}],
['OS=="linux" or OS=="android" or os_bsd==1', {
'sources': [
'source/atomic32_non_darwin_unix.cc',
],
}],
['OS=="ios" or OS=="mac"', {
'defines': [
'WEBRTC_THREAD_RR',
],
'sources': [
'source/atomic32_darwin.cc',
],
}],
['OS=="win"', {
'link_settings': {
'libraries': [ '-lwinmm.lib', ],
},
}],
], # conditions
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
4334, # Ignore warning on shift operator promotion.
],
}, {
'target_name': 'field_trial_default',
'type': 'static_library',
'sources': [
'include/field_trial_default.h',
'source/field_trial_default.cc',
]
}, {
'target_name': 'metrics_default',
'type': 'static_library',
'sources': [
'include/metrics_default.h',
'source/metrics_default.cc',
],
}, {
'target_name': 'system_wrappers_default',
'type': 'static_library',
'dependencies': [
# 'system_wrappers',
# 'field_trial_default',
# 'metrics_default',
]
},
], # targets
}

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@ -1,296 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# TODO(andrew): consider moving test_support to src/base/test.
{
'includes': [
'../build/common.gypi',
],
'targets': [
{
'target_name': 'video_test_common',
'type': 'static_library',
'sources': [
'fake_texture_frame.cc',
'fake_texture_frame.h',
'frame_generator.cc',
'frame_generator.h',
'frame_utils.cc',
'frame_utils.h',
],
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
],
},
{
'target_name': 'rtp_test_utils',
'type': 'static_library',
'sources': [
'rtcp_packet_parser.cc',
'rtcp_packet_parser.h',
'rtp_file_reader.cc',
'rtp_file_reader.h',
'rtp_file_writer.cc',
'rtp_file_writer.h',
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
],
},
{
'target_name': 'field_trial',
'type': 'static_library',
'sources': [
'field_trial.cc',
'field_trial.h',
],
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
],
},
{
'target_name': 'test_main',
'type': 'static_library',
'sources': [
'test_main.cc',
],
'dependencies': [
'field_trial',
'test_support',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
],
},
{
'target_name': 'test_support',
'type': 'static_library',
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/base/base.gyp:gtest_prod',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'video_test_common',
],
'sources': [
'gmock.h',
'gtest.h',
'testsupport/fileutils.cc',
'testsupport/fileutils.h',
'testsupport/frame_reader.cc',
'testsupport/frame_reader.h',
'testsupport/frame_writer.cc',
'testsupport/frame_writer.h',
'testsupport/iosfileutils.mm',
'testsupport/metrics/video_metrics.h',
'testsupport/metrics/video_metrics.cc',
'testsupport/mock/mock_frame_reader.h',
'testsupport/mock/mock_frame_writer.h',
'testsupport/packet_reader.cc',
'testsupport/packet_reader.h',
'testsupport/perf_test.cc',
'testsupport/perf_test.h',
'testsupport/trace_to_stderr.cc',
'testsupport/trace_to_stderr.h',
],
'conditions': [
['OS=="ios"', {
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
}],
['use_x11==1', {
'dependencies': [
'<(DEPTH)/tools/xdisplaycheck/xdisplaycheck.gyp:xdisplaycheck',
],
}],
],
},
{
# Depend on this target when you want to have test_support but also the
# main method needed for gtest to execute!
'target_name': 'test_support_main',
'type': 'static_library',
'dependencies': [
'field_trial',
'test_support',
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
],
'sources': [
'run_all_unittests.cc',
'test_suite.cc',
'test_suite.h',
],
},
{
# Depend on this target when you want to have test_support and a special
# main for mac which will run your test on a worker thread and consume
# events on the main thread. Useful if you want to access a webcam.
# This main will provide all the scaffolding and objective-c black magic
# for you. All you need to do is to implement a function in the
# run_threaded_main_mac.h file (ImplementThisToRunYourTest).
'target_name': 'test_support_main_threaded_mac',
'type': 'static_library',
'dependencies': [
'test_support',
],
'sources': [
'testsupport/mac/run_threaded_main_mac.h',
'testsupport/mac/run_threaded_main_mac.mm',
],
},
{
'target_name': 'test_common',
'type': 'static_library',
'sources': [
'call_test.cc',
'call_test.h',
'configurable_frame_size_encoder.cc',
'configurable_frame_size_encoder.h',
'constants.cc',
'constants.h',
'direct_transport.cc',
'direct_transport.h',
'drifting_clock.cc',
'drifting_clock.h',
'encoder_settings.cc',
'encoder_settings.h',
'fake_audio_device.cc',
'fake_audio_device.h',
'fake_decoder.cc',
'fake_decoder.h',
'fake_encoder.cc',
'fake_encoder.h',
'fake_network_pipe.cc',
'fake_network_pipe.h',
'fake_videorenderer.h',
'frame_generator_capturer.cc',
'frame_generator_capturer.h',
'layer_filtering_transport.cc',
'layer_filtering_transport.h',
'mock_transport.h',
'mock_voe_channel_proxy.h',
'mock_voice_engine.h',
'null_transport.cc',
'null_transport.h',
'rtp_rtcp_observer.h',
'statistics.cc',
'statistics.h',
'vcm_capturer.cc',
'vcm_capturer.h',
'video_capturer.h',
'win/run_loop_win.cc',
],
'conditions': [
['OS!="win"', {
'sources': [
'run_loop.h',
'run_loop.cc',
],
}],
],
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/webrtc.gyp:webrtc',
'rtp_test_utils',
'test_support',
],
},
{
'target_name': 'test_renderer',
'type': 'static_library',
'sources': [
'linux/glx_renderer.cc',
'linux/glx_renderer.h',
'linux/video_renderer_linux.cc',
'mac/video_renderer_mac.h',
'mac/video_renderer_mac.mm',
'video_renderer.cc',
'video_renderer.h',
'win/d3d_renderer.cc',
'win/d3d_renderer.h',
],
'conditions': [
['OS!="linux" and OS!="mac" and OS!="win"', {
'sources': [
'null_platform_renderer.cc',
],
}],
['OS=="linux" or OS=="mac"', {
'sources' : [
'gl/gl_renderer.cc',
'gl/gl_renderer.h',
],
}],
['OS=="win"', {
'include_dirs': [
'<(directx_sdk_path)/Include',
],
}],
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-bool-conversion',
'-Wno-comment',
'-Wno-delete-non-virtual-dtor',
],
},
},
}],
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:media_file',
'test_support',
'video_test_common',
],
'direct_dependent_settings': {
'conditions': [
['OS=="linux"', {
'libraries': [
'-lXext',
'-lX11',
'-lGL',
],
}],
['OS=="android"', {
'libraries' : [
'-lGLESv2', '-llog',
],
}],
['OS=="mac"', {
'xcode_settings' : {
'OTHER_LDFLAGS' : [
'-framework Cocoa',
'-framework OpenGL',
'-framework CoreVideo',
],
},
}],
],
},
},
],
}

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@ -1,28 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
# This file is used for internal tools used by the WebRTC code only.
{
'includes': [
'../build/common.gypi',
],
'targets': [
{
'target_name': 'command_line_parser',
'type': 'static_library',
'sources': [
'simple_command_line_parser.h',
'simple_command_line_parser.cc',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:gtest_prod',
],
}, # command_line_parser
],
}

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@ -1,146 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../build/common.gypi',
],
'targets': [
{
'target_name': 'video_quality_analysis',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
],
'export_dependent_settings': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
],
'sources': [
'frame_analyzer/video_quality_analysis.h',
'frame_analyzer/video_quality_analysis.cc',
],
}, # video_quality_analysis
{
'target_name': 'frame_analyzer',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
'video_quality_analysis',
],
'sources': [
'frame_analyzer/frame_analyzer.cc',
],
}, # frame_analyzer
{
'target_name': 'psnr_ssim_analyzer',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
'video_quality_analysis',
],
'sources': [
'psnr_ssim_analyzer/psnr_ssim_analyzer.cc',
],
}, # psnr_ssim_analyzer
{
'target_name': 'rgba_to_i420_converter',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
],
'sources': [
'converter/converter.h',
'converter/converter.cc',
'converter/rgba_to_i420_converter.cc',
],
}, # rgba_to_i420_converter
{
'target_name': 'frame_editing_lib',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common_video/common_video.gyp:common_video',
],
'sources': [
'frame_editing/frame_editing_lib.cc',
'frame_editing/frame_editing_lib.h',
],
# Disable warnings to enable Win64 build, issue 1323.
'msvs_disabled_warnings': [
4267, # size_t to int truncation.
],
}, # frame_editing_lib
{
'target_name': 'frame_editor',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
'frame_editing_lib',
],
'sources': [
'frame_editing/frame_editing.cc',
],
}, # frame_editing
{
'target_name': 'force_mic_volume_max',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:audio_device',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
],
'sources': [
'force_mic_volume_max/force_mic_volume_max.cc',
],
}, # force_mic_volume_max
],
'conditions': [
['enable_protobuf==1', {
'targets': [
{
'target_name': 'chart_proto',
'type': 'static_library',
'sources': [
'event_log_visualizer/chart.proto',
],
'variables': {
'proto_in_dir': 'event_log_visualizer',
'proto_out_dir': 'webrtc/tools/event_log_visualizer',
},
'includes': ['../build/protoc.gypi'],
},
{
# RTC event log visualization library
'target_name': 'event_log_visualizer_utils',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/webrtc.gyp:rtc_event_log_impl',
'<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
'<(webrtc_root)/modules/modules.gyp:congestion_controller',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
':chart_proto',
],
'sources': [
'event_log_visualizer/analyzer.cc',
'event_log_visualizer/analyzer.h',
'event_log_visualizer/plot_base.cc',
'event_log_visualizer/plot_base.h',
'event_log_visualizer/plot_protobuf.cc',
'event_log_visualizer/plot_protobuf.h',
'event_log_visualizer/plot_python.cc',
'event_log_visualizer/plot_python.h',
],
'export_dependent_settings': [
'<(webrtc_root)/webrtc.gyp:rtc_event_log_parser',
':chart_proto',
],
},
],
}],
], # conditions
}

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@ -1,69 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'variables': {
'webrtc_video_dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_video/common_video.gyp:common_video',
'<(webrtc_root)/modules/modules.gyp:bitrate_controller',
'<(webrtc_root)/modules/modules.gyp:congestion_controller',
'<(webrtc_root)/modules/modules.gyp:paced_sender',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:video_capture_module',
'<(webrtc_root)/modules/modules.gyp:video_processing',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
'<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
'<(webrtc_root)/api/api.gyp:video_frame_api',
'<(webrtc_root)/media/media.gyp:mozilla_rtc_media'
],
'webrtc_video_sources': [
'video/call_stats.cc',
'video/call_stats.h',
'video/encoder_rtcp_feedback.cc',
'video/encoder_rtcp_feedback.h',
'video/overuse_frame_detector.cc',
'video/overuse_frame_detector.h',
'video/payload_router.cc',
'video/payload_router.h',
'video/quality_threshold.cc',
'video/quality_threshold.h',
'video/receive_statistics_proxy.cc',
'video/receive_statistics_proxy.h',
'video/report_block_stats.cc',
'video/report_block_stats.h',
'video/rtp_stream_receiver.cc',
'video/rtp_stream_receiver.h',
'video/rtp_streams_synchronizer.cc',
'video/rtp_streams_synchronizer.h',
'video/send_delay_stats.cc',
'video/send_delay_stats.h',
'video/send_statistics_proxy.cc',
'video/send_statistics_proxy.h',
'video/stats_counter.cc',
'video/stats_counter.h',
'video/stream_synchronization.cc',
'video/stream_synchronization.h',
'video/transport_adapter.cc',
'video/transport_adapter.h',
'video/video_receive_stream.cc',
'video/video_receive_stream.h',
'video/video_send_stream.cc',
'video/video_send_stream.h',
'video/video_stream_decoder.cc',
'video/video_stream_decoder.h',
'video/vie_encoder.cc',
'video/vie_encoder.h',
'video/vie_remb.cc',
'video/vie_remb.h',
],
},
}

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@ -1,146 +0,0 @@
# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'../build/common.gypi',
],
'targets': [
{
'target_name': 'audio_coder',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
],
'sources': [
'coder.cc',
'coder.h',
]
},
{
'target_name': 'file_player',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
],
'sources': [
'file_player.cc',
'file_player.h',
]
},
{
'target_name': 'file_recorder',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/common.gyp:webrtc_common',
],
'sources': [
'file_recorder.cc',
'file_recorder.h',
]
},
{
'target_name': 'voice_engine',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/api/api.gyp:call_api',
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
'<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
'<(webrtc_root)/modules/modules.gyp:audio_device',
'<(webrtc_root)/modules/modules.gyp:audio_processing',
'<(webrtc_root)/modules/modules.gyp:bitrate_controller',
'<(webrtc_root)/modules/modules.gyp:media_file',
'<(webrtc_root)/modules/modules.gyp:paced_sender',
'<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
'<(webrtc_root)/modules/modules.gyp:webrtc_utility',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'<(webrtc_root)/webrtc.gyp:rtc_event_log_api',
'audio_coder',
'file_player',
'file_recorder',
'level_indicator',
],
'export_dependent_settings': [
'<(webrtc_root)/modules/modules.gyp:audio_coding_module',
],
'sources': [
'include/voe_audio_processing.h',
'include/voe_base.h',
'include/voe_codec.h',
'include/voe_errors.h',
'include/voe_external_media.h',
'include/voe_file.h',
'include/voe_hardware.h',
'include/voe_neteq_stats.h',
'include/voe_network.h',
'include/voe_rtp_rtcp.h',
'include/voe_video_sync.h',
'include/voe_volume_control.h',
'channel.cc',
'channel.h',
'channel_manager.cc',
'channel_manager.h',
'channel_proxy.cc',
'channel_proxy.h',
'monitor_module.cc',
'monitor_module.h',
'output_mixer.cc',
'output_mixer.h',
'shared_data.cc',
'shared_data.h',
'statistics.cc',
'statistics.h',
'transmit_mixer.cc',
'transmit_mixer.h',
'utility.cc',
'utility.h',
'voe_audio_processing_impl.cc',
'voe_audio_processing_impl.h',
'voe_base_impl.cc',
'voe_base_impl.h',
'voe_codec_impl.cc',
'voe_codec_impl.h',
'voe_external_media_impl.cc',
'voe_external_media_impl.h',
'voe_file_impl.cc',
'voe_file_impl.h',
'voe_hardware_impl.cc',
'voe_hardware_impl.h',
'voe_neteq_stats_impl.cc',
'voe_neteq_stats_impl.h',
'voe_network_impl.cc',
'voe_network_impl.h',
'voe_rtp_rtcp_impl.cc',
'voe_rtp_rtcp_impl.h',
'voe_video_sync_impl.cc',
'voe_video_sync_impl.h',
'voe_volume_control_impl.cc',
'voe_volume_control_impl.h',
'voice_engine_defines.h',
'voice_engine_impl.cc',
'voice_engine_impl.h',
],
},
{
'target_name': 'level_indicator',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/common_audio/common_audio.gyp:common_audio',
],
'sources': [
'level_indicator.cc',
'level_indicator.h',
]
},
],
}

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@ -1,129 +0,0 @@
# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'includes': [
'build/common.gypi',
'audio/webrtc_audio.gypi',
'call/webrtc_call.gypi',
'video/webrtc_video.gypi',
],
'targets': [
{
'target_name': 'webrtc_lib',
'type': 'static_library',
'sources': [
'call.h',
'config.h',
'video_receive_stream.h',
'video_send_stream.h',
'<@(webrtc_audio_sources)',
'<@(webrtc_call_sources)',
'<@(webrtc_video_sources)',
],
'dependencies': [
'common.gyp:*',
'<@(webrtc_audio_dependencies)',
'<@(webrtc_call_dependencies)',
'<@(webrtc_video_dependencies)',
'rtc_event_log_impl',
'<(webrtc_root)/modules/modules.gyp:audio_mixer',
],
'conditions': [
# TODO(andresp): Chromium should link directly with this and no if
# conditions should be needed on webrtc build files.
['build_with_chromium==1', {
'dependencies': [
'<(webrtc_root)/modules/modules.gyp:video_capture',
],
}],
],
},
{
'target_name': 'rtc_event_log_api',
'type': 'static_library',
'sources': [
'logging/rtc_event_log/rtc_event_log.h',
],
},
{
'target_name': 'rtc_event_log_impl',
'type': 'static_library',
'sources': [
'logging/rtc_event_log/ringbuffer.h',
'logging/rtc_event_log/rtc_event_log.cc',
'logging/rtc_event_log/rtc_event_log_helper_thread.cc',
'logging/rtc_event_log/rtc_event_log_helper_thread.h',
],
'conditions': [
# If enable_protobuf is defined, we want to compile the protobuf
# and add rtc_event_log.pb.h and rtc_event_log.pb.cc to the sources.
['enable_protobuf==1', {
'dependencies': [
'rtc_event_log_api',
'rtc_event_log_proto',
],
'defines': [
'ENABLE_RTC_EVENT_LOG',
],
}],
],
},
], # targets
'conditions': [
['include_tests==1', {
'includes': [
'webrtc_tests.gypi',
],
}],
['enable_protobuf==1', {
'targets': [
{
# This target should only be built if enable_protobuf is defined
'target_name': 'rtc_event_log_proto',
'type': 'static_library',
'sources': ['logging/rtc_event_log/rtc_event_log.proto',],
'variables': {
'proto_in_dir': 'logging/rtc_event_log',
'proto_out_dir': 'webrtc/logging/rtc_event_log',
},
'includes': ['build/protoc.gypi'],
},
{
'target_name': 'rtc_event_log_parser',
'type': 'static_library',
'sources': [
'logging/rtc_event_log/rtc_event_log_parser.cc',
'logging/rtc_event_log/rtc_event_log_parser.h',
],
'dependencies': [
'rtc_event_log_proto',
],
'export_dependent_settings': [
'rtc_event_log_proto',
],
},
],
}],
['include_tests==1 and enable_protobuf==1', {
'targets': [
{
'target_name': 'rtc_event_log2rtp_dump',
'type': 'executable',
'sources': ['logging/rtc_event_log2rtp_dump.cc',],
'dependencies': [
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'rtc_event_log_parser',
'rtc_event_log_proto',
'test/test.gyp:rtp_test_utils'
],
},
],
}],
], # conditions
}

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@ -1,471 +0,0 @@
# Copyright (c) 2012 The WebRTC Project Authors. All rights reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'conditions': [
['OS=="linux" or OS=="win"', {
'targets': [
{
'target_name': 'relayserver',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'examples/relayserver/relayserver_main.cc',
],
}, # target relayserver
{
'target_name': 'stunserver',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'examples/stunserver/stunserver_main.cc',
],
}, # target stunserver
{
'target_name': 'turnserver',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'examples/turnserver/turnserver_main.cc',
],
}, # target turnserver
{
'target_name': 'peerconnection_server',
'type': 'executable',
'sources': [
'examples/peerconnection/server/data_socket.cc',
'examples/peerconnection/server/data_socket.h',
'examples/peerconnection/server/main.cc',
'examples/peerconnection/server/peer_channel.cc',
'examples/peerconnection/server/peer_channel.h',
'examples/peerconnection/server/utils.cc',
'examples/peerconnection/server/utils.h',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_approved',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/tools/internal_tools.gyp:command_line_parser',
],
# TODO(ronghuawu): crbug.com/167187 fix size_t to int truncations.
'msvs_disabled_warnings': [ 4309, ],
}, # target peerconnection_server
{
'target_name': 'peerconnection_client',
'type': 'executable',
'sources': [
'examples/peerconnection/client/conductor.cc',
'examples/peerconnection/client/conductor.h',
'examples/peerconnection/client/defaults.cc',
'examples/peerconnection/client/defaults.h',
'examples/peerconnection/client/peer_connection_client.cc',
'examples/peerconnection/client/peer_connection_client.h',
],
'dependencies': [
'api/api.gyp:libjingle_peerconnection',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
],
'conditions': [
['build_json==1', {
'dependencies': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
}],
# TODO(ronghuawu): Move these files to a win/ directory then they
# can be excluded automatically.
['OS=="win"', {
'sources': [
'examples/peerconnection/client/flagdefs.h',
'examples/peerconnection/client/main.cc',
'examples/peerconnection/client/main_wnd.cc',
'examples/peerconnection/client/main_wnd.h',
],
'msvs_settings': {
'VCLinkerTool': {
'SubSystem': '2', # Windows
},
},
}], # OS=="win"
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-reorder',
'-Wno-unused-function',
],
},
},
}], # OS=="win" and clang==1
['OS=="linux"', {
'sources': [
'examples/peerconnection/client/linux/main.cc',
'examples/peerconnection/client/linux/main_wnd.cc',
'examples/peerconnection/client/linux/main_wnd.h',
],
'cflags': [
'<!@(pkg-config --cflags glib-2.0 gobject-2.0 gtk+-2.0)',
],
'link_settings': {
'ldflags': [
'<!@(pkg-config --libs-only-L --libs-only-other glib-2.0'
' gobject-2.0 gthread-2.0 gtk+-2.0)',
],
'libraries': [
'<!@(pkg-config --libs-only-l glib-2.0 gobject-2.0'
' gthread-2.0 gtk+-2.0)',
'-lX11',
'-lXcomposite',
'-lXext',
'-lXrender',
],
},
}], # OS=="linux"
['OS=="linux" and target_arch=="ia32"', {
'cflags': [
'-Wno-sentinel',
],
}], # OS=="linux" and target_arch=="ia32"
], # conditions
}, # target peerconnection_client
], # targets
}], # OS=="linux" or OS=="win"
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
'targets': [
{
'target_name': 'apprtc_common',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_common_objc',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:metrics_default',
],
'sources': [
'examples/objc/AppRTCMobile/common/ARDUtilities.h',
'examples/objc/AppRTCMobile/common/ARDUtilities.m',
],
'include_dirs': [
'examples/objc/AppRTCMobile/common',
],
'direct_dependent_settings': {
'include_dirs': [
'examples/objc/AppRTCMobile/common',
],
},
'conditions': [
['OS=="ios"', {
'xcode_settings': {
'WARNING_CFLAGS': [
# Suppress compiler warnings about deprecated that triggered
# when moving from ios_deployment_target 7.0 to 9.0.
# See webrtc:5549 for more details.
'-Wno-deprecated-declarations',
],
},
}],
['OS=="mac"', {
'xcode_settings': {
'MACOSX_DEPLOYMENT_TARGET' : '10.8',
},
}],
],
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework QuartzCore',
],
},
},
},
{
'target_name': 'apprtc_signaling',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
'apprtc_common',
'socketrocket',
],
'sources': [
'examples/objc/AppRTCMobile/ARDAppClient.h',
'examples/objc/AppRTCMobile/ARDAppClient.m',
'examples/objc/AppRTCMobile/ARDAppClient+Internal.h',
'examples/objc/AppRTCMobile/ARDAppEngineClient.h',
'examples/objc/AppRTCMobile/ARDAppEngineClient.m',
'examples/objc/AppRTCMobile/ARDBitrateTracker.h',
'examples/objc/AppRTCMobile/ARDBitrateTracker.m',
'examples/objc/AppRTCMobile/ARDCEODTURNClient.h',
'examples/objc/AppRTCMobile/ARDCEODTURNClient.m',
'examples/objc/AppRTCMobile/ARDJoinResponse.h',
'examples/objc/AppRTCMobile/ARDJoinResponse.m',
'examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h',
'examples/objc/AppRTCMobile/ARDMessageResponse.h',
'examples/objc/AppRTCMobile/ARDMessageResponse.m',
'examples/objc/AppRTCMobile/ARDMessageResponse+Internal.h',
'examples/objc/AppRTCMobile/ARDRoomServerClient.h',
'examples/objc/AppRTCMobile/ARDSDPUtils.h',
'examples/objc/AppRTCMobile/ARDSDPUtils.m',
'examples/objc/AppRTCMobile/ARDSignalingChannel.h',
'examples/objc/AppRTCMobile/ARDSignalingMessage.h',
'examples/objc/AppRTCMobile/ARDSignalingMessage.m',
'examples/objc/AppRTCMobile/ARDStatsBuilder.h',
'examples/objc/AppRTCMobile/ARDStatsBuilder.m',
'examples/objc/AppRTCMobile/ARDTURNClient.h',
'examples/objc/AppRTCMobile/ARDWebSocketChannel.h',
'examples/objc/AppRTCMobile/ARDWebSocketChannel.m',
'examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h',
'examples/objc/AppRTCMobile/RTCIceCandidate+JSON.m',
'examples/objc/AppRTCMobile/RTCIceServer+JSON.h',
'examples/objc/AppRTCMobile/RTCIceServer+JSON.m',
'examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.h',
'examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.m',
'examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h',
'examples/objc/AppRTCMobile/RTCSessionDescription+JSON.m',
],
'include_dirs': [
'examples/objc/AppRTCMobile',
],
'direct_dependent_settings': {
'include_dirs': [
'examples/objc/AppRTCMobile',
],
},
'export_dependent_settings': [
'<(webrtc_root)/sdk/sdk.gyp:rtc_sdk_peerconnection_objc',
],
'conditions': [
['OS=="ios"', {
'xcode_settings': {
'WARNING_CFLAGS': [
# Suppress compiler warnings about deprecated that triggered
# when moving from ios_deployment_target 7.0 to 9.0.
# See webrtc:5549 for more details.
'-Wno-deprecated-declarations',
],
},
}],
['OS=="mac"', {
'xcode_settings': {
'MACOSX_DEPLOYMENT_TARGET' : '10.8',
},
}],
],
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
},
{
'target_name': 'AppRTCMobile',
'type': 'executable',
'product_name': 'AppRTCMobile',
'mac_bundle': 1,
'dependencies': [
'apprtc_common',
'apprtc_signaling',
],
'conditions': [
['OS=="ios"', {
'mac_bundle_resources': [
'examples/objc/AppRTCMobile/ios/resources/Roboto-Regular.ttf',
'examples/objc/AppRTCMobile/ios/resources/iPhone5@2x.png',
'examples/objc/AppRTCMobile/ios/resources/iPhone6@2x.png',
'examples/objc/AppRTCMobile/ios/resources/iPhone6p@3x.png',
'examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp.png',
'examples/objc/AppRTCMobile/ios/resources/ic_call_end_black_24dp@2x.png',
'examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp.png',
'examples/objc/AppRTCMobile/ios/resources/ic_clear_black_24dp@2x.png',
'examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp.png',
'examples/objc/AppRTCMobile/ios/resources/ic_surround_sound_black_24dp@2x.png',
'examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp.png',
'examples/objc/AppRTCMobile/ios/resources/ic_switch_video_black_24dp@2x.png',
'examples/objc/AppRTCMobile/ios/resources/mozart.mp3',
'examples/objc/Icon.png',
],
'sources': [
'examples/objc/AppRTCMobile/ios/ARDAppDelegate.h',
'examples/objc/AppRTCMobile/ios/ARDAppDelegate.m',
'examples/objc/AppRTCMobile/ios/ARDMainView.h',
'examples/objc/AppRTCMobile/ios/ARDMainView.m',
'examples/objc/AppRTCMobile/ios/ARDMainViewController.h',
'examples/objc/AppRTCMobile/ios/ARDMainViewController.m',
'examples/objc/AppRTCMobile/ios/ARDStatsView.h',
'examples/objc/AppRTCMobile/ios/ARDStatsView.m',
'examples/objc/AppRTCMobile/ios/ARDVideoCallView.h',
'examples/objc/AppRTCMobile/ios/ARDVideoCallView.m',
'examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.h',
'examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m',
'examples/objc/AppRTCMobile/ios/AppRTCMobile-Prefix.pch',
'examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.h',
'examples/objc/AppRTCMobile/ios/UIImage+ARDUtilities.m',
'examples/objc/AppRTCMobile/ios/main.m',
],
'xcode_settings': {
'INFOPLIST_FILE': 'examples/objc/AppRTCMobile/ios/Info.plist',
'WARNING_CFLAGS': [
# Suppress compiler warnings about deprecated that triggered
# when moving from ios_deployment_target 7.0 to 9.0.
# See webrtc:5549 for more details.
'-Wno-deprecated-declarations',
],
},
}],
['OS=="mac"', {
'sources': [
'examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.h',
'examples/objc/AppRTCMobile/mac/APPRTCAppDelegate.m',
'examples/objc/AppRTCMobile/mac/APPRTCViewController.h',
'examples/objc/AppRTCMobile/mac/APPRTCViewController.m',
'examples/objc/AppRTCMobile/mac/main.m',
],
'xcode_settings': {
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
'INFOPLIST_FILE': 'examples/objc/AppRTCMobile/mac/Info.plist',
'MACOSX_DEPLOYMENT_TARGET' : '10.8',
'OTHER_LDFLAGS': [
'-framework AVFoundation',
],
},
}],
['target_arch=="ia32"', {
'dependencies' : [
'<(DEPTH)/testing/iossim/iossim.gyp:iossim#host',
],
}],
],
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
}, # target AppRTCMobile
{
# TODO(tkchin): move this into the real third party location and
# have it mirrored on chrome infra.
'target_name': 'socketrocket',
'type': 'static_library',
'sources': [
'examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.h',
'examples/objc/AppRTCMobile/third_party/SocketRocket/SRWebSocket.m',
],
'conditions': [
['OS=="mac"', {
'xcode_settings': {
# SocketRocket autosynthesizes some properties. Disable the
# warning so we can compile successfully.
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
'MACOSX_DEPLOYMENT_TARGET' : '10.8',
# SRWebSocket.m uses code with partial availability.
# https://code.google.com/p/webrtc/issues/detail?id=4695
'WARNING_CFLAGS!': [
'-Wpartial-availability',
],
},
}],
],
'direct_dependent_settings': {
'include_dirs': [
'examples/objc/AppRTCMobile/third_party/SocketRocket',
],
},
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
'WARNING_CFLAGS': [
'-Wno-deprecated-declarations',
'-Wno-nonnull',
# Hide the warning for SecRandomCopyBytes(), till we update
# to upstream.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=6396
'-Wno-unused-result',
],
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CFNetwork',
'-licucore',
],
},
}
}, # target socketrocket
], # targets
}], # OS=="ios" or (OS=="mac" and target_arch!="ia32")
['OS=="android"', {
'targets': [
{
'target_name': 'AppRTCMobile',
'type': 'none',
'dependencies': [
'api/api_java.gyp:libjingle_peerconnection_java',
],
'variables': {
'apk_name': 'AppRTCMobile',
'java_in_dir': 'examples/androidapp',
'has_java_resources': 1,
'resource_dir': 'examples/androidapp/res',
'R_package': 'org.appspot.apprtc',
'R_package_relpath': 'org/appspot/apprtc',
'input_jars_paths': [
'examples/androidapp/third_party/autobanh/lib/autobanh.jar',
],
'library_dexed_jars_paths': [
'examples/androidapp/third_party/autobanh/lib/autobanh.jar',
],
'native_lib_target': 'libjingle_peerconnection_so',
'add_to_dependents_classpaths':1,
},
'includes': [ '../build/java_apk.gypi' ],
}, # target AppRTCMobile
{
# AppRTCMobile creates a .jar as a side effect. Any java targets
# that need that .jar in their classpath should depend on this target,
# AppRTCMobile_apk. Dependents of AppRTCMobile_apk receive its
# jar path in the variable 'apk_output_jar_path'.
# This target should only be used by targets which instrument
# AppRTCMobile_apk.
'target_name': 'AppRTCMobile_apk',
'type': 'none',
'dependencies': [
'AppRTCMobile',
],
'includes': [ '../build/apk_fake_jar.gypi' ],
}, # target AppRTCMobile_apk
{
'target_name': 'AppRTCMobileTest',
'type': 'none',
'dependencies': [
'AppRTCMobile_apk',
],
'variables': {
'apk_name': 'AppRTCMobileTest',
'java_in_dir': 'examples/androidtests',
'is_test_apk': 1,
'test_type': 'instrumentation',
'test_runner_path': '<(DEPTH)/webrtc/build/android/test_runner.py',
},
'includes': [
'../build/java_apk.gypi',
'../build/android/test_runner.gypi',
],
},
], # targets
}], # OS=="android"
],
}

Просмотреть файл

@ -1,84 +0,0 @@
# Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
{
'targets': [
{
'target_name': 'video_quality_test',
'type': 'static_library',
'sources': [
'video/video_quality_test.cc',
'video/video_quality_test.h',
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers',
'webrtc',
],
'conditions': [
['OS=="android"', {
'dependencies!': [
'<(webrtc_root)/modules/modules.gyp:video_capture_module_internal_impl',
],
}],
],
},
{
'target_name': 'screenshare_loopback',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/screenshare_loopback.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
'video_quality_test',
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/test.gyp:test_common',
'test/test.gyp:test_main',
'test/test.gyp:test_renderer',
'webrtc',
],
},
{
'target_name': 'video_replay',
'type': 'executable',
'sources': [
'test/mac/run_test.mm',
'test/run_test.cc',
'test/run_test.h',
'video/replay.cc',
],
'conditions': [
['OS=="mac"', {
'sources!': [
'test/run_test.cc',
],
}],
],
'dependencies': [
'<(DEPTH)/testing/gtest.gyp:gtest',
'<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
'test/test.gyp:test_common',
'test/test.gyp:test_renderer',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:system_wrappers_default',
'webrtc',
],
},
],
}