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Move talk/session/media -> webrtc/pc

Browse Source 
The libjingle_p2p target is renamed to rtc_pc.
The libjingle_p2p_unittest test will be renamed in a
separate follow-up CL, to make it possible to run all
trybots successfully for this CL.

BUG=webrtc:5419
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1691463002 .

Cr-Commit-Position: refs/heads/master@{#11592}
This commit is contained in:
kjellander@webrtc.org 2016-02-12 06:47:59 +01:00
parent 5ad129741c
commit 9b8df25c73
73 changed files with 427 additions and 568 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
PRESUBMIT.py
View File

@ -208,8 +208,6 @@ def _CheckNoRtcBaseDeps(input_api, gyp_files, output_api):
gyp_exceptions = (
'base_tests.gyp',
'desktop_capture.gypi',
'libjingle.gyp',
'libjingle_tests.gyp',
'p2p.gyp',
'sound.gyp',
'webrtc_test_common.gyp',

5
all.gyp
View File

@ -18,7 +18,6 @@
'type': 'none',
'dependencies': [
'webrtc/webrtc.gyp:*',
'talk/libjingle.gyp:*',
'<@(webrtc_root_additional_dependencies)',
],
'conditions': [
@ -27,9 +26,9 @@
'webrtc/webrtc_examples.gyp:*',
],
}],
['include_tests==1', {
['OS=="ios" or (OS=="mac" and target_arch!="ia32") and include_tests==1', {
'dependencies': [
'talk/libjingle_tests.gyp:*',
'talk/app/webrtc/legacy_objc_api_tests.gyp:*',
],
}],
],

191
talk/app/webrtc/legacy_objc_api.gyp Executable file
View File

@ -0,0 +1,191 @@
#
# libjingle
# Copyright 2012 Google Inc.
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions are met:
#
# 1. Redistributions of source code must retain the above copyright notice,
# this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright notice,
# this list of conditions and the following disclaimer in the documentation
# and/or other materials provided with the distribution.
# 3. The name of the author may not be used to endorse or promote products
# derived from this software without specific prior written permission.
#
# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
{
'includes': ['../../build/common.gypi'],
'conditions': [
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >= 10.7 above is required for ARC.
'targets': [
{
'target_name': 'libjingle_peerconnection_objc',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
],
'sources': [
'objc/RTCAudioTrack+Internal.h',
'objc/RTCAudioTrack.mm',
'objc/RTCDataChannel+Internal.h',
'objc/RTCDataChannel.mm',
'objc/RTCEnumConverter.h',
'objc/RTCEnumConverter.mm',
'objc/RTCFileLogger.mm',
'objc/RTCI420Frame+Internal.h',
'objc/RTCI420Frame.mm',
'objc/RTCICECandidate+Internal.h',
'objc/RTCICECandidate.mm',
'objc/RTCICEServer+Internal.h',
'objc/RTCICEServer.mm',
'objc/RTCLogging.mm',
'objc/RTCMediaConstraints+Internal.h',
'objc/RTCMediaConstraints.mm',
'objc/RTCMediaConstraintsNative.cc',
'objc/RTCMediaConstraintsNative.h',
'objc/RTCMediaSource+Internal.h',
'objc/RTCMediaSource.mm',
'objc/RTCMediaStream+Internal.h',
'objc/RTCMediaStream.mm',
'objc/RTCMediaStreamTrack+Internal.h',
'objc/RTCMediaStreamTrack.mm',
'objc/RTCOpenGLVideoRenderer.mm',
'objc/RTCPair.m',
'objc/RTCPeerConnection+Internal.h',
'objc/RTCPeerConnection.mm',
'objc/RTCPeerConnectionFactory.mm',
'objc/RTCPeerConnectionInterface+Internal.h',
'objc/RTCPeerConnectionInterface.mm',
'objc/RTCPeerConnectionObserver.h',
'objc/RTCPeerConnectionObserver.mm',
'objc/RTCSessionDescription+Internal.h',
'objc/RTCSessionDescription.mm',
'objc/RTCStatsReport+Internal.h',
'objc/RTCStatsReport.mm',
'objc/RTCVideoCapturer+Internal.h',
'objc/RTCVideoCapturer.mm',
'objc/RTCVideoRendererAdapter.h',
'objc/RTCVideoRendererAdapter.mm',
'objc/RTCVideoSource+Internal.h',
'objc/RTCVideoSource.mm',
'objc/RTCVideoTrack+Internal.h',
'objc/RTCVideoTrack.mm',
'objc/public/RTCAudioSource.h',
'objc/public/RTCAudioTrack.h',
'objc/public/RTCDataChannel.h',
'objc/public/RTCFileLogger.h',
'objc/public/RTCI420Frame.h',
'objc/public/RTCICECandidate.h',
'objc/public/RTCICEServer.h',
'objc/public/RTCLogging.h',
'objc/public/RTCMediaConstraints.h',
'objc/public/RTCMediaSource.h',
'objc/public/RTCMediaStream.h',
'objc/public/RTCMediaStreamTrack.h',
'objc/public/RTCOpenGLVideoRenderer.h',
'objc/public/RTCPair.h',
'objc/public/RTCPeerConnection.h',
'objc/public/RTCPeerConnectionDelegate.h',
'objc/public/RTCPeerConnectionFactory.h',
'objc/public/RTCPeerConnectionInterface.h',
'objc/public/RTCSessionDescription.h',
'objc/public/RTCSessionDescriptionDelegate.h',
'objc/public/RTCStatsDelegate.h',
'objc/public/RTCStatsReport.h',
'objc/public/RTCTypes.h',
'objc/public/RTCVideoCapturer.h',
'objc/public/RTCVideoRenderer.h',
'objc/public/RTCVideoSource.h',
'objc/public/RTCVideoTrack.h',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/talk/app/webrtc/objc/public',
],
},
'include_dirs': [
'<(webrtc_root)/webrtc/api',
'<(DEPTH)/talk/app/webrtc/objc',
'<(DEPTH)/talk/app/webrtc/objc/public',
],
'link_settings': {
'libraries': [
'-lstdc++',
],
},
'all_dependent_settings': {
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
},
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
# common.gypi enables this for mac but we want this to be disabled
# like it is for ios.
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
# Disabled due to failing when compiled with -Wall, see
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5397
'WARNING_CFLAGS': ['-Wno-unused-property-ivar'],
},
'conditions': [
['OS=="ios"', {
'sources': [
'objc/avfoundationvideocapturer.h',
'objc/avfoundationvideocapturer.mm',
'objc/RTCAVFoundationVideoSource+Internal.h',
'objc/RTCAVFoundationVideoSource.mm',
'objc/RTCEAGLVideoView.m',
'objc/public/RTCEAGLVideoView.h',
'objc/public/RTCAVFoundationVideoSource.h',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_objc',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreGraphics',
'-framework GLKit',
],
},
},
}],
['OS=="mac"', {
'sources': [
'objc/RTCNSGLVideoView.m',
'objc/public/RTCNSGLVideoView.h',
],
'xcode_settings': {
# Need to build against 10.7 framework for full ARC support
# on OSX.
'MACOSX_DEPLOYMENT_TARGET' : '10.7',
# RTCVideoTrack.mm uses code with partial availability.
# https://code.google.com/p/webrtc/issues/detail?id=4695
'WARNING_CFLAGS!': ['-Wpartial-availability'],
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework Cocoa',
],
},
},
}],
],
}, # target libjingle_peerconnection_objc
],
}],
],
}

79
talk/libjingle_tests.gyp → talk/app/webrtc/legacy_objc_api_tests.gyp
View File

@ -25,47 +25,7 @@
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
{
'includes': ['build/common.gypi'],
'targets': [
{
'target_name': 'libjingle_p2p_unittest',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/webrtc.gyp:rtc_unittest_main',
'libjingle.gyp:libjingle_p2p',
],
'include_dirs': [
'<(DEPTH)/third_party/libsrtp/srtp',
],
'sources': [
'session/media/bundlefilter_unittest.cc',
'session/media/channel_unittest.cc',
'session/media/channelmanager_unittest.cc',
'session/media/currentspeakermonitor_unittest.cc',
'session/media/mediasession_unittest.cc',
'session/media/rtcpmuxfilter_unittest.cc',
'session/media/srtpfilter_unittest.cc',
],
'conditions': [
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
['OS=="win"', {
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
'strmiids.lib',
],
},
},
}],
],
}, # target libjingle_p2p_unittest
],
'includes': ['../../build/common.gypi'],
'conditions': [
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >=10.7 above is required to make ARC link cleanly (e.g. as
@ -75,21 +35,21 @@
{
'target_name': 'libjingle_peerconnection_objc_test',
'type': 'executable',
'includes': [ 'build/objc_app.gypi' ],
'includes': [ '../../build/objc_app.gypi' ],
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'libjingle.gyp:libjingle_peerconnection_objc',
'legacy_objc_api.gyp:libjingle_peerconnection_objc',
],
'sources': [
'app/webrtc/objctests/RTCPeerConnectionSyncObserver.h',
'app/webrtc/objctests/RTCPeerConnectionSyncObserver.m',
'app/webrtc/objctests/RTCPeerConnectionTest.mm',
'app/webrtc/objctests/RTCSessionDescriptionSyncObserver.h',
'app/webrtc/objctests/RTCSessionDescriptionSyncObserver.m',
'objctests/RTCPeerConnectionSyncObserver.h',
'objctests/RTCPeerConnectionSyncObserver.m',
'objctests/RTCPeerConnectionTest.mm',
'objctests/RTCSessionDescriptionSyncObserver.h',
'objctests/RTCSessionDescriptionSyncObserver.m',
# TODO(fischman): figure out if this works for ios or if it
# needs a GUI driver.
'app/webrtc/objctests/mac/main.mm',
'objctests/mac/main.mm',
],
'conditions': [
['OS=="mac"', {
@ -107,7 +67,7 @@
{
'target_name': 'apprtc_signaling_gunit_test',
'type': 'executable',
'includes': [ 'build/objc_app.gypi' ],
'includes': [ '../../build/objc_app.gypi' ],
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
@ -115,7 +75,7 @@
'<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock',
],
'sources': [
'app/webrtc/objctests/mac/main.mm',
'objctests/mac/main.mm',
'<(webrtc_root)/examples/objc/AppRTCDemo/tests/ARDAppClientTest.mm',
],
'conditions': [
@ -128,22 +88,5 @@
}, # target apprtc_signaling_gunit_test
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'libjingle_p2p_unittest_run',
'type': 'none',
'dependencies': [
'libjingle_p2p_unittest',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'libjingle_p2p_unittest.isolate',
],
},
],
}],
],
}

14
talk/build/common.gypi
View File

@ -34,14 +34,7 @@
# flood of chromium-style warnings.
'clang_use_chrome_plugins%': 0,
# Disable these to not build components which can be externally provided.
'build_expat%': 1,
'build_json%': 1,
'build_libsrtp%': 1,
# Make it possible to provide custom locations for some libraries.
'libyuv_dir%': '<(DEPTH)/third_party/libyuv',
# Disable this to skip building source requiring GTK.
'use_gtk%': 1,
},
'target_defaults': {
'include_dirs': [
@ -51,13 +44,6 @@
'../../third_party/webrtc',
'../../webrtc',
],
'defines': [
'SRTP_RELATIVE_PATH',
# Feature selection
'HAVE_SCTP',
'HAVE_SRTP',
],
'conditions': [
['OS=="linux"', {
'defines': [

161
talk/build/isolate.gypi
View File

@ -1,161 +0,0 @@
#
# libjingle
# Copyright 2013 Google Inc.
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions are met:
#
# 1. Redistributions of source code must retain the above copyright notice,
# this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright notice,
# this list of conditions and the following disclaimer in the documentation
# and/or other materials provided with the distribution.
# 3. The name of the author may not be used to endorse or promote products
# derived from this software without specific prior written permission.
#
# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
# 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_openssl=<(use_openssl)',
'--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',
],
}],
],
},
],
}

2
talk/build/merge_ios_libs.gyp
View File

@ -35,7 +35,7 @@
'type': 'executable',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'../libjingle.gyp:libjingle_peerconnection_objc',
'../app/webrtc/legacy_objc_api.gyp:libjingle_peerconnection_objc',
],
'sources': ['<(webrtc_root)/build/no_op.cc',],
},

244
talk/libjingle.gyp
View File

@ -1,244 +0,0 @@
#
# libjingle
# Copyright 2012 Google Inc.
#
# Redistribution and use in source and binary forms, with or without
# modification, are permitted provided that the following conditions are met:
#
# 1. Redistributions of source code must retain the above copyright notice,
# this list of conditions and the following disclaimer.
# 2. Redistributions in binary form must reproduce the above copyright notice,
# this list of conditions and the following disclaimer in the documentation
# and/or other materials provided with the distribution.
# 3. The name of the author may not be used to endorse or promote products
# derived from this software without specific prior written permission.
#
# THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
# WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
# MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
# EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
# SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
# PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
# OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
# WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
# OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
# ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
{
'includes': ['build/common.gypi'],
'conditions': [
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >= 10.7 above is required for ARC.
'targets': [
{
'target_name': 'libjingle_peerconnection_objc',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
],
'sources': [
'app/webrtc/objc/RTCAudioTrack+Internal.h',
'app/webrtc/objc/RTCAudioTrack.mm',
'app/webrtc/objc/RTCDataChannel+Internal.h',
'app/webrtc/objc/RTCDataChannel.mm',
'app/webrtc/objc/RTCEnumConverter.h',
'app/webrtc/objc/RTCEnumConverter.mm',
'app/webrtc/objc/RTCFileLogger.mm',
'app/webrtc/objc/RTCI420Frame+Internal.h',
'app/webrtc/objc/RTCI420Frame.mm',
'app/webrtc/objc/RTCICECandidate+Internal.h',
'app/webrtc/objc/RTCICECandidate.mm',
'app/webrtc/objc/RTCICEServer+Internal.h',
'app/webrtc/objc/RTCICEServer.mm',
'app/webrtc/objc/RTCLogging.mm',
'app/webrtc/objc/RTCMediaConstraints+Internal.h',
'app/webrtc/objc/RTCMediaConstraints.mm',
'app/webrtc/objc/RTCMediaConstraintsNative.cc',
'app/webrtc/objc/RTCMediaConstraintsNative.h',
'app/webrtc/objc/RTCMediaSource+Internal.h',
'app/webrtc/objc/RTCMediaSource.mm',
'app/webrtc/objc/RTCMediaStream+Internal.h',
'app/webrtc/objc/RTCMediaStream.mm',
'app/webrtc/objc/RTCMediaStreamTrack+Internal.h',
'app/webrtc/objc/RTCMediaStreamTrack.mm',
'app/webrtc/objc/RTCOpenGLVideoRenderer.mm',
'app/webrtc/objc/RTCPair.m',
'app/webrtc/objc/RTCPeerConnection+Internal.h',
'app/webrtc/objc/RTCPeerConnection.mm',
'app/webrtc/objc/RTCPeerConnectionFactory.mm',
'app/webrtc/objc/RTCPeerConnectionInterface+Internal.h',
'app/webrtc/objc/RTCPeerConnectionInterface.mm',
'app/webrtc/objc/RTCPeerConnectionObserver.h',
'app/webrtc/objc/RTCPeerConnectionObserver.mm',
'app/webrtc/objc/RTCSessionDescription+Internal.h',
'app/webrtc/objc/RTCSessionDescription.mm',
'app/webrtc/objc/RTCStatsReport+Internal.h',
'app/webrtc/objc/RTCStatsReport.mm',
'app/webrtc/objc/RTCVideoCapturer+Internal.h',
'app/webrtc/objc/RTCVideoCapturer.mm',
'app/webrtc/objc/RTCVideoRendererAdapter.h',
'app/webrtc/objc/RTCVideoRendererAdapter.mm',
'app/webrtc/objc/RTCVideoSource+Internal.h',
'app/webrtc/objc/RTCVideoSource.mm',
'app/webrtc/objc/RTCVideoTrack+Internal.h',
'app/webrtc/objc/RTCVideoTrack.mm',
'app/webrtc/objc/public/RTCAudioSource.h',
'app/webrtc/objc/public/RTCAudioTrack.h',
'app/webrtc/objc/public/RTCDataChannel.h',
'app/webrtc/objc/public/RTCFileLogger.h',
'app/webrtc/objc/public/RTCI420Frame.h',
'app/webrtc/objc/public/RTCICECandidate.h',
'app/webrtc/objc/public/RTCICEServer.h',
'app/webrtc/objc/public/RTCLogging.h',
'app/webrtc/objc/public/RTCMediaConstraints.h',
'app/webrtc/objc/public/RTCMediaSource.h',
'app/webrtc/objc/public/RTCMediaStream.h',
'app/webrtc/objc/public/RTCMediaStreamTrack.h',
'app/webrtc/objc/public/RTCOpenGLVideoRenderer.h',
'app/webrtc/objc/public/RTCPair.h',
'app/webrtc/objc/public/RTCPeerConnection.h',
'app/webrtc/objc/public/RTCPeerConnectionDelegate.h',
'app/webrtc/objc/public/RTCPeerConnectionFactory.h',
'app/webrtc/objc/public/RTCPeerConnectionInterface.h',
'app/webrtc/objc/public/RTCSessionDescription.h',
'app/webrtc/objc/public/RTCSessionDescriptionDelegate.h',
'app/webrtc/objc/public/RTCStatsDelegate.h',
'app/webrtc/objc/public/RTCStatsReport.h',
'app/webrtc/objc/public/RTCTypes.h',
'app/webrtc/objc/public/RTCVideoCapturer.h',
'app/webrtc/objc/public/RTCVideoRenderer.h',
'app/webrtc/objc/public/RTCVideoSource.h',
'app/webrtc/objc/public/RTCVideoTrack.h',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/talk/app/webrtc/objc/public',
],
},
'include_dirs': [
'<(webrtc_root)/webrtc/api',
'<(DEPTH)/talk/app/webrtc/objc',
'<(DEPTH)/talk/app/webrtc/objc/public',
],
'link_settings': {
'libraries': [
'-lstdc++',
],
},
'all_dependent_settings': {
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
},
},
'xcode_settings': {
'CLANG_ENABLE_OBJC_ARC': 'YES',
# common.gypi enables this for mac but we want this to be disabled
# like it is for ios.
'CLANG_WARN_OBJC_MISSING_PROPERTY_SYNTHESIS': 'NO',
# Disabled due to failing when compiled with -Wall, see
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5397
'WARNING_CFLAGS': ['-Wno-unused-property-ivar'],
},
'conditions': [
['OS=="ios"', {
'sources': [
'app/webrtc/objc/avfoundationvideocapturer.h',
'app/webrtc/objc/avfoundationvideocapturer.mm',
'app/webrtc/objc/RTCAVFoundationVideoSource+Internal.h',
'app/webrtc/objc/RTCAVFoundationVideoSource.mm',
'app/webrtc/objc/RTCEAGLVideoView.m',
'app/webrtc/objc/public/RTCEAGLVideoView.h',
'app/webrtc/objc/public/RTCAVFoundationVideoSource.h',
],
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_objc',
],
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework CoreGraphics',
'-framework GLKit',
],
},
},
}],
['OS=="mac"', {
'sources': [
'app/webrtc/objc/RTCNSGLVideoView.m',
'app/webrtc/objc/public/RTCNSGLVideoView.h',
],
'xcode_settings': {
# Need to build against 10.7 framework for full ARC support
# on OSX.
'MACOSX_DEPLOYMENT_TARGET' : '10.7',
# RTCVideoTrack.mm uses code with partial availability.
# https://code.google.com/p/webrtc/issues/detail?id=4695
'WARNING_CFLAGS!': ['-Wpartial-availability'],
},
'link_settings': {
'xcode_settings': {
'OTHER_LDFLAGS': [
'-framework Cocoa',
],
},
},
}],
],
}, # target libjingle_peerconnection_objc
],
}],
],
'targets': [
{
'target_name': 'libjingle_p2p',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/media/media.gyp:rtc_media',
],
'conditions': [
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
],
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
'include_dirs!': [
'<(DEPTH)/webrtc',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/testing/gtest/include',
],
'include_dirs!': [
'<(DEPTH)/webrtc',
],
},
'sources': [
'session/media/audiomonitor.cc',
'session/media/audiomonitor.h',
'session/media/bundlefilter.cc',
'session/media/bundlefilter.h',
'session/media/channel.cc',
'session/media/channel.h',
'session/media/channelmanager.cc',
'session/media/channelmanager.h',
'session/media/currentspeakermonitor.cc',
'session/media/currentspeakermonitor.h',
'session/media/mediamonitor.cc',
'session/media/mediamonitor.h',
'session/media/mediasession.cc',
'session/media/mediasession.h',
'session/media/mediasink.h',
'session/media/rtcpmuxfilter.cc',
'session/media/rtcpmuxfilter.h',
'session/media/srtpfilter.cc',
'session/media/srtpfilter.h',
'session/media/voicechannel.h',
],
}, # target libjingle_p2p
],
}

2
webrtc/api/api.gyp
View File

@ -255,7 +255,7 @@
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/media/media.gyp:rtc_media',
'../../talk/libjingle.gyp:libjingle_p2p',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'audiotrack.cc',

2
webrtc/api/api_tests.gyp
View File

@ -18,7 +18,7 @@
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/webrtc.gyp:rtc_unittest_main',
'../../talk/libjingle.gyp:libjingle_p2p',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'direct_dependent_settings': {
'include_dirs': [

2
webrtc/api/datachannel.h
View File

@ -15,13 +15,13 @@
#include <set>
#include <string>
#include "talk/session/media/channel.h"
#include "webrtc/api/datachannelinterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/sigslot.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/pc/channel.h"
namespace webrtc {

2
webrtc/api/jsepsessiondescription.cc
View File

@ -11,9 +11,9 @@
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/webrtcsdp.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/pc/mediasession.h"
using rtc::scoped_ptr;
using cricket::SessionDescription;

2
webrtc/api/jsepsessiondescription_unittest.cc
View File

@ -12,7 +12,6 @@
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/scoped_ptr.h"
@ -21,6 +20,7 @@
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/pc/mediasession.h"
using webrtc::IceCandidateCollection;
using webrtc::IceCandidateInterface;

2
webrtc/api/mediacontroller.cc
View File

@ -10,10 +10,10 @@
#include "webrtc/api/mediacontroller.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/call.h"
#include "webrtc/pc/channelmanager.h"
namespace {

2
webrtc/api/peerconnection.cc
View File

@ -15,7 +15,6 @@
#include <utility>
#include <vector>
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/dtmfsender.h"
#include "webrtc/api/jsepicecandidate.h"
@ -39,6 +38,7 @@
#include "webrtc/base/trace_event.h"
#include "webrtc/media/sctp/sctpdataengine.h"
#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/system_wrappers/include/field_trial.h"
namespace {

2
webrtc/api/peerconnection_unittest.cc
View File

@ -16,7 +16,6 @@
#include <utility>
#include <vector>
#include "talk/session/media/mediasession.h"
#include "webrtc/api/dtmfsender.h"
#include "webrtc/api/fakemetricsobserver.h"
#include "webrtc/api/localaudiosource.h"
@ -42,6 +41,7 @@
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/p2p/client/fakeportallocator.h"
#include "webrtc/pc/mediasession.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \

2
webrtc/api/peerconnectionfactory.h
View File

@ -13,7 +13,6 @@
#include <string>
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/dtlsidentitystore.h"
#include "webrtc/api/mediacontroller.h"
#include "webrtc/api/mediastreaminterface.h"
@ -21,6 +20,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"
#include "webrtc/pc/channelmanager.h"
namespace rtc {
class BasicNetworkManager;

2
webrtc/api/peerconnectioninterface_unittest.cc
View File

@ -11,7 +11,6 @@
#include <string>
#include <utility>
#include "talk/session/media/mediasession.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediastream.h"
@ -39,6 +38,7 @@
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/sctp/sctpdataengine.h"
#include "webrtc/p2p/client/fakeportallocator.h"
#include "webrtc/pc/mediasession.h"
static const char kStreamLabel1[] = "local_stream_1";
static const char kStreamLabel2[] = "local_stream_2";

2
webrtc/api/rtpsenderinterface.h
View File

@ -16,11 +16,11 @@
#include <string>
#include "talk/session/media/mediasession.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/base/refcount.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/pc/mediasession.h"
namespace webrtc {

2
webrtc/api/statscollector.cc
View File

@ -14,11 +14,11 @@
#include <vector>
#include "webrtc/api/peerconnection.h"
#include "talk/session/media/channel.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/timing.h"
#include "webrtc/pc/channel.h"
using rtc::scoped_ptr;

2
webrtc/api/statscollector_unittest.cc
View File

@ -14,7 +14,6 @@
#include "webrtc/api/statscollector.h"
#include "talk/session/media/channelmanager.h"
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/api/mediastream.h"
@ -30,6 +29,7 @@
#include "webrtc/base/network.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/channelmanager.h"
using rtc::scoped_ptr;
using testing::_;

2
webrtc/api/videosource.cc
View File

@ -14,8 +14,8 @@
#include <vector>
#include "webrtc/api/mediaconstraintsinterface.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/pc/channelmanager.h"
using cricket::CaptureState;
using webrtc::MediaConstraintsInterface;

2
webrtc/api/videosource_unittest.cc
View File

@ -11,7 +11,6 @@
#include <string>
#include <vector>
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/remotevideocapturer.h"
#include "webrtc/api/test/fakeconstraints.h"
#include "webrtc/api/videosource.h"
@ -20,6 +19,7 @@
#include "webrtc/media/base/fakevideocapturer.h"
#include "webrtc/media/base/fakevideorenderer.h"
#include "webrtc/media/engine/webrtcvideoframe.h"
#include "webrtc/pc/channelmanager.h"
using webrtc::FakeConstraints;
using webrtc::VideoSource;

2
webrtc/api/videotrack_unittest.cc
View File

@ -10,7 +10,6 @@
#include <string>
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/remotevideocapturer.h"
#include "webrtc/api/test/fakevideotrackrenderer.h"
#include "webrtc/api/videosource.h"
@ -19,6 +18,7 @@
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/media/base/fakemediaengine.h"
#include "webrtc/media/engine/webrtcvideoframe.h"
#include "webrtc/pc/channelmanager.h"
using webrtc::FakeVideoTrackRenderer;
using webrtc::VideoSource;

2
webrtc/api/webrtcsdp.cc
View File

@ -17,7 +17,6 @@
#include <string>
#include <vector>
#include "talk/session/media/mediasession.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/base/arraysize.h"
@ -33,6 +32,7 @@
#include "webrtc/p2p/base/candidate.h"
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/port.h"
#include "webrtc/pc/mediasession.h"
using cricket::AudioContentDescription;
using cricket::Candidate;

2
webrtc/api/webrtcsdp_unittest.cc
View File

@ -12,7 +12,6 @@
#include <string>
#include <vector>
#include "talk/session/media/mediasession.h"
#include "webrtc/api/jsepsessiondescription.h"
#ifdef WEBRTC_ANDROID
#include "webrtc/api/test/androidtestinitializer.h"
@ -27,6 +26,7 @@
#include "webrtc/base/stringutils.h"
#include "webrtc/media/base/constants.h"
#include "webrtc/p2p/base/constants.h"
#include "webrtc/pc/mediasession.h"
using cricket::AudioCodec;
using cricket::AudioContentDescription;

6
webrtc/api/webrtcsession.cc
View File

@ -17,9 +17,6 @@
#include <utility>
#include <vector>
#include "talk/session/media/channel.h"
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/api/jsepicecandidate.h"
#include "webrtc/api/jsepsessiondescription.h"
#include "webrtc/api/mediaconstraintsinterface.h"
@ -38,6 +35,9 @@
#include "webrtc/media/base/videocapturer.h"
#include "webrtc/p2p/base/portallocator.h"
#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/pc/channel.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediasession.h"
using cricket::ContentInfo;
using cricket::ContentInfos;

2
webrtc/api/webrtcsession.h
View File

@ -14,7 +14,6 @@
#include <string>
#include <vector>
#include "talk/session/media/mediasession.h"
#include "webrtc/api/datachannel.h"
#include "webrtc/api/dtmfsender.h"
#include "webrtc/api/mediacontroller.h"
@ -26,6 +25,7 @@
#include "webrtc/base/thread.h"
#include "webrtc/media/base/mediachannel.h"
#include "webrtc/p2p/base/transportcontroller.h"
#include "webrtc/pc/mediasession.h"
namespace cricket {

4
webrtc/api/webrtcsession_unittest.cc
View File

@ -11,8 +11,6 @@
#include <utility>
#include <vector>
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediasession.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/fakemediacontroller.h"
#include "webrtc/api/fakemetricsobserver.h"
@ -48,6 +46,8 @@
#include "webrtc/p2p/base/testturnserver.h"
#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/p2p/client/basicportallocator.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediasession.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(feature())) { \

2
webrtc/api/webrtcsessiondescriptionfactory.h
View File

@ -11,12 +11,12 @@
#ifndef WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
#define WEBRTC_API_WEBRTCSESSIONDESCRIPTIONFACTORY_H_
#include "talk/session/media/mediasession.h"
#include "webrtc/api/dtlsidentitystore.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/rtccertificate.h"
#include "webrtc/p2p/base/transportdescriptionfactory.h"
#include "webrtc/pc/mediasession.h"
namespace cricket {
class ChannelManager;

1
webrtc/build/common.gypi
View File

@ -99,6 +99,7 @@
'build_expat%': 1,
'build_json%': 1,
'build_libjpeg%': 1,
'build_libsrtp%': 1,
'build_libvpx%': 1,
'build_libyuv%': 1,
'build_openmax_dl%': 1,

6
webrtc/libjingle/libjingle.gyp
View File

@ -8,16 +8,12 @@
{
'includes': ['../build/common.gypi'],
'variables': {
'talk_root%': '<(webrtc_root)/../talk',
},
'targets': [
{
'target_name': 'libjingle_xmpphelp',
'type': 'static_library',
'dependencies': [
'<(talk_root)/libjingle.gyp:libjingle',
'<(talk_root)/libjingle.gyp:libjingle_p2p',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'conditions': [
['build_expat==1', {

0
talk/libjingle_p2p_unittest.isolate → webrtc/libjingle_p2p_unittest.isolate
View File

2
webrtc/media/base/mediachannel.h
View File

@ -27,7 +27,7 @@
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/base/videosinkinterface.h"
// TODO(juberti): re-evaluate this include
#include "talk/session/media/audiomonitor.h"
#include "webrtc/pc/audiomonitor.h"
namespace rtc {
class Buffer;

2
webrtc/media/engine/webrtcvoiceengine.h
View File

@ -15,7 +15,6 @@
#include <string>
#include <vector>
#include "talk/session/media/channel.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/scoped_ptr.h"
@ -27,6 +26,7 @@
#include "webrtc/media/base/rtputils.h"
#include "webrtc/media/engine/webrtccommon.h"
#include "webrtc/media/engine/webrtcvoe.h"
#include "webrtc/pc/channel.h"
namespace cricket {

2
webrtc/media/engine/webrtcvoiceengine_unittest.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "talk/session/media/channel.h"
#include "webrtc/pc/channel.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/gunit.h"

4
talk/session/media/audiomonitor.cc → webrtc/pc/audiomonitor.cc
View File

@ -26,8 +26,8 @@
*/
#include <assert.h>
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/voicechannel.h"
#include "webrtc/pc/audiomonitor.h"
#include "webrtc/pc/voicechannel.h"
namespace cricket {

0
talk/session/media/audiomonitor.h → webrtc/pc/audiomonitor.h
View File

2
talk/session/media/bundlefilter.cc → webrtc/pc/bundlefilter.cc
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/bundlefilter.h"
#include "webrtc/pc/bundlefilter.h"
#include "webrtc/base/logging.h"
#include "webrtc/media/base/rtputils.h"

0
talk/session/media/bundlefilter.h → webrtc/pc/bundlefilter.h
View File

2
talk/session/media/bundlefilter_unittest.cc → webrtc/pc/bundlefilter_unittest.cc
View File

@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/bundlefilter.h"
#include "webrtc/base/gunit.h"
#include "webrtc/pc/bundlefilter.h"
using cricket::StreamParams;

4
talk/session/media/channel.cc → webrtc/pc/channel.cc
View File

@ -27,9 +27,8 @@
#include <utility>
#include "talk/session/media/channel.h"
#include "webrtc/pc/channel.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/buffer.h"
@ -41,6 +40,7 @@
#include "webrtc/media/base/constants.h"
#include "webrtc/media/base/rtputils.h"
#include "webrtc/p2p/base/transportchannel.h"
#include "webrtc/pc/channelmanager.h"
namespace cricket {
using rtc::Bind;

12
talk/session/media/channel.h → webrtc/pc/channel.h
View File

@ -34,12 +34,6 @@
#include <utility>
#include <vector>
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/bundlefilter.h"
#include "talk/session/media/mediamonitor.h"
#include "talk/session/media/mediasession.h"
#include "talk/session/media/rtcpmuxfilter.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/asyncudpsocket.h"
#include "webrtc/base/criticalsection.h"
@ -53,6 +47,12 @@
#include "webrtc/media/base/videosinkinterface.h"
#include "webrtc/p2p/base/transportcontroller.h"
#include "webrtc/p2p/client/socketmonitor.h"
#include "webrtc/pc/audiomonitor.h"
#include "webrtc/pc/bundlefilter.h"
#include "webrtc/pc/mediamonitor.h"
#include "webrtc/pc/mediasession.h"
#include "webrtc/pc/rtcpmuxfilter.h"
#include "webrtc/pc/srtpfilter.h"
namespace webrtc {
class AudioSinkInterface;

2
talk/session/media/channel_unittest.cc → webrtc/pc/channel_unittest.cc
View File

@ -25,7 +25,6 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/channel.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/gunit.h"
@ -45,6 +44,7 @@
#include "webrtc/media/base/screencastid.h"
#include "webrtc/media/base/testutils.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/channel.h"
#define MAYBE_SKIP_TEST(feature) \
if (!(rtc::SSLStreamAdapter::feature())) { \

4
talk/session/media/channelmanager.cc → webrtc/pc/channelmanager.cc
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/channelmanager.h"
#include "webrtc/pc/channelmanager.h"
#ifdef HAVE_CONFIG_H
#include <config.h>
@ -33,7 +33,6 @@
#include <algorithm>
#include "talk/session/media/srtpfilter.h"
#include "webrtc/api/mediacontroller.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
@ -50,6 +49,7 @@
#ifdef HAVE_SCTP
#include "webrtc/media/sctp/sctpdataengine.h"
#endif
#include "webrtc/pc/srtpfilter.h"
namespace cricket {

2
talk/session/media/channelmanager.h → webrtc/pc/channelmanager.h
View File

@ -31,13 +31,13 @@
#include <string>
#include <vector>
#include "talk/session/media/voicechannel.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/fileutils.h"
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/base/thread.h"
#include "webrtc/media/base/capturemanager.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/pc/voicechannel.h"
namespace webrtc {
class MediaControllerInterface;

2
talk/session/media/channelmanager_unittest.cc → webrtc/pc/channelmanager_unittest.cc
View File

@ -25,7 +25,6 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/fakemediacontroller.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
@ -36,6 +35,7 @@
#include "webrtc/media/base/testutils.h"
#include "webrtc/media/engine/fakewebrtccall.h"
#include "webrtc/p2p/base/faketransportcontroller.h"
#include "webrtc/pc/channelmanager.h"
namespace cricket {

4
talk/session/media/currentspeakermonitor.cc → webrtc/pc/currentspeakermonitor.cc
View File

@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/currentspeakermonitor.h"
#include "webrtc/pc/currentspeakermonitor.h"
#include "talk/session/media/audiomonitor.h"
#include "webrtc/base/logging.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/pc/audiomonitor.h"
namespace cricket {

0
talk/session/media/currentspeakermonitor.h → webrtc/pc/currentspeakermonitor.h
View File

4
talk/session/media/currentspeakermonitor_unittest.cc → webrtc/pc/currentspeakermonitor_unittest.cc
View File

@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/audiomonitor.h"
#include "talk/session/media/currentspeakermonitor.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
#include "webrtc/pc/audiomonitor.h"
#include "webrtc/pc/currentspeakermonitor.h"
namespace cricket {

2
talk/session/media/externalhmac.cc → webrtc/pc/externalhmac.cc
View File

@ -27,7 +27,7 @@
#if defined(HAVE_SRTP) && defined(ENABLE_EXTERNAL_AUTH)
#include "talk/session/media/externalhmac.h"
#include "webrtc/pc/externalhmac.h"
#include <stdlib.h> // For malloc/free.

0
talk/session/media/externalhmac.h → webrtc/pc/externalhmac.h
View File

4
talk/session/media/mediamonitor.cc → webrtc/pc/mediamonitor.cc
View File

@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/mediamonitor.h"
#include "webrtc/base/common.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/mediamonitor.h"
namespace cricket {

0
talk/session/media/mediamonitor.h → webrtc/pc/mediamonitor.h
View File

6
talk/session/media/mediasession.cc → webrtc/pc/mediasession.cc
View File

@ -25,15 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/mediasession.h"
#include "webrtc/pc/mediasession.h"
#include <functional>
#include <map>
#include <set>
#include <utility>
#include "talk/session/media/channelmanager.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/scoped_ptr.h"
@ -41,6 +39,8 @@
#include "webrtc/media/base/constants.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/p2p/base/constants.h"
#include "webrtc/pc/channelmanager.h"
#include "webrtc/pc/srtpfilter.h"
#ifdef HAVE_SCTP
#include "webrtc/media/sctp/sctpdataengine.h"

0
talk/session/media/mediasession.h → webrtc/pc/mediasession.h
View File

4
talk/session/media/mediasession_unittest.cc → webrtc/pc/mediasession_unittest.cc
View File

@ -28,8 +28,6 @@
#include <string>
#include <vector>
#include "talk/session/media/mediasession.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/messagedigest.h"
@ -39,6 +37,8 @@
#include "webrtc/p2p/base/constants.h"
#include "webrtc/p2p/base/transportdescription.h"
#include "webrtc/p2p/base/transportinfo.h"
#include "webrtc/pc/mediasession.h"
#include "webrtc/pc/srtpfilter.h"
#ifdef HAVE_SRTP
#define ASSERT_CRYPTO(cd, s, cs) \

0
talk/session/media/mediasink.h → webrtc/pc/mediasink.h
View File

76
webrtc/pc/pc.gyp Executable file
View File

@ -0,0 +1,76 @@
# 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',
'<(webrtc_root)/media/media.gyp:rtc_media',
],
'conditions': [
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
],
'defines': [
'<@(rtc_pc_defines)',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags_cc!': [
'-Wnon-virtual-dtor',
],
'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',
'mediasink.h',
'rtcpmuxfilter.cc',
'rtcpmuxfilter.h',
'srtpfilter.cc',
'srtpfilter.h',
'voicechannel.h',
],
}, # target rtc_pc
],
}

61
webrtc/pc/pc_tests.gypi Executable file
View File

@ -0,0 +1,61 @@
# 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': 'libjingle_p2p_unittest',
'type': 'executable',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/webrtc.gyp:rtc_unittest_main',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'include_dirs': [
'<(DEPTH)/third_party/libsrtp/srtp',
],
'sources': [
'bundlefilter_unittest.cc',
'channel_unittest.cc',
'channelmanager_unittest.cc',
'currentspeakermonitor_unittest.cc',
'mediasession_unittest.cc',
'rtcpmuxfilter_unittest.cc',
'srtpfilter_unittest.cc',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags_cc!': [
'-Wnon-virtual-dtor',
],
'conditions': [
['clang==0', {
'cflags': [
'-Wno-maybe-uninitialized', # Only exists for GCC.
],
}],
['build_libsrtp==1', {
'dependencies': [
'<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
],
}],
['OS=="win"', {
'msvs_settings': {
'VCLinkerTool': {
'AdditionalDependencies': [
'strmiids.lib',
],
},
},
}],
],
}, # target libjingle_p2p_unittest
],
}

0
talk/session/media/planarfunctions_unittest.cc → webrtc/pc/planarfunctions_unittest.cc
View File

2
talk/session/media/rtcpmuxfilter.cc → webrtc/pc/rtcpmuxfilter.cc
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/rtcpmuxfilter.h"
#include "webrtc/pc/rtcpmuxfilter.h"
#include "webrtc/base/logging.h"

0
talk/session/media/rtcpmuxfilter.h → webrtc/pc/rtcpmuxfilter.h
View File

3
talk/session/media/rtcpmuxfilter_unittest.cc → webrtc/pc/rtcpmuxfilter_unittest.cc
View File

@ -25,10 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/rtcpmuxfilter.h"
#include "webrtc/base/gunit.h"
#include "webrtc/media/base/testutils.h"
#include "webrtc/pc/rtcpmuxfilter.h"
TEST(RtcpMuxFilterTest, DemuxRtcpSender) {
cricket::RtcpMuxFilter filter;

4
talk/session/media/srtpfilter.cc → webrtc/pc/srtpfilter.cc
View File

@ -27,7 +27,7 @@
#undef HAVE_CONFIG_H
#include "talk/session/media/srtpfilter.h"
#include "webrtc/pc/srtpfilter.h"
#include <string.h>
@ -55,7 +55,7 @@ extern "C" {
#endif // SRTP_RELATIVE_PATH
}
#ifdef ENABLE_EXTERNAL_AUTH
#include "talk/session/media/externalhmac.h"
#include "webrtc/pc/externalhmac.h"
#endif // ENABLE_EXTERNAL_AUTH
#if !defined(NDEBUG)
extern "C" debug_module_t mod_srtp;

0
talk/session/media/srtpfilter.h → webrtc/pc/srtpfilter.h
View File

2
talk/session/media/srtpfilter_unittest.cc → webrtc/pc/srtpfilter_unittest.cc
View File

@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/thread.h"
#include "webrtc/media/base/cryptoparams.h"
#include "webrtc/media/base/fakertp.h"
#include "webrtc/p2p/base/sessiondescription.h"
#include "webrtc/pc/srtpfilter.h"
extern "C" {
#ifdef SRTP_RELATIVE_PATH
#include "crypto/include/err.h"

0
talk/session/media/typewrapping.h.pump → webrtc/pc/typewrapping.h.pump
View File

2
talk/session/media/voicechannel.h → webrtc/pc/voicechannel.h
View File

@ -28,6 +28,6 @@
#ifndef _VOICECHANNEL_H_
#define _VOICECHANNEL_H_
#include "talk/session/media/channel.h"
#include "webrtc/pc/channel.h"
#endif // _VOICECHANNEL_H_

0
talk/session/media/yuvscaler_unittest.cc → webrtc/pc/yuvscaler_unittest.cc
View File

1
webrtc/webrtc.gyp
View File

@ -41,6 +41,7 @@
'libjingle/xmpp/xmpp_tests.gypi',
'media/media_tests.gypi',
'p2p/p2p_tests.gypi',
'pc/pc_tests.gypi',
'sound/sound_tests.gypi',
'webrtc_tests.gypi',
],

12
webrtc/webrtc_examples.gyp
View File

@ -15,7 +15,7 @@
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'../talk/libjingle.gyp:libjingle_p2p',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'examples/relayserver/relayserver_main.cc',
@ -26,7 +26,7 @@
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'../talk/libjingle.gyp:libjingle_p2p',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'examples/stunserver/stunserver_main.cc',
@ -37,7 +37,7 @@
'type': 'executable',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'../talk/libjingle.gyp:libjingle_p2p',
'<(webrtc_root)/pc/pc.gyp:rtc_pc',
],
'sources': [
'examples/turnserver/turnserver_main.cc',
@ -150,7 +150,7 @@
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'../talk/libjingle.gyp:libjingle_peerconnection_objc',
'../talk/app/webrtc/legacy_objc_api.gyp:libjingle_peerconnection_objc',
],
'sources': [
'examples/objc/AppRTCDemo/common/ARDUtilities.h',
@ -184,7 +184,7 @@
'type': 'static_library',
'dependencies': [
'apprtc_common',
'../talk/libjingle.gyp:libjingle_peerconnection_objc',
'../talk/app/webrtc/legacy_objc_api.gyp:libjingle_peerconnection_objc',
'socketrocket',
],
'sources': [
@ -232,7 +232,7 @@
],
},
'export_dependent_settings': [
'../talk/libjingle.gyp:libjingle_peerconnection_objc',
'../talk/app/webrtc/legacy_objc_api.gyp:libjingle_peerconnection_objc',
],
'conditions': [
['OS=="mac"', {

13
webrtc/webrtc_tests.gypi
View File

@ -368,6 +368,19 @@
'libjingle_media_unittest.isolate',
],
},
{
'target_name': 'libjingle_p2p_unittest_run',
'type': 'none',
'dependencies': [
'libjingle_p2p_unittest',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'libjingle_p2p_unittest.isolate',
],
},
{
'target_name': 'video_engine_tests_run',
'type': 'none',