diff --git a/PRESUBMIT.py b/PRESUBMIT.py index 24a50348ec..8ea0f3a86a 100755 --- a/PRESUBMIT.py +++ b/PRESUBMIT.py @@ -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', diff --git a/all.gyp b/all.gyp index 0b11c8f7d6..a688afba9e 100644 --- a/all.gyp +++ b/all.gyp @@ -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:*', ], }], ], diff --git a/talk/app/webrtc/legacy_objc_api.gyp b/talk/app/webrtc/legacy_objc_api.gyp new file mode 100755 index 0000000000..8aaa0f14e9 --- /dev/null +++ b/talk/app/webrtc/legacy_objc_api.gyp @@ -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 + ], + }], + ], +} diff --git a/talk/libjingle_tests.gyp b/talk/app/webrtc/legacy_objc_api_tests.gyp similarity index 60% rename from talk/libjingle_tests.gyp rename to talk/app/webrtc/legacy_objc_api_tests.gyp index f45a99c025..1d8b682c39 100755 --- a/talk/libjingle_tests.gyp +++ b/talk/app/webrtc/legacy_objc_api_tests.gyp @@ -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', - ], - }, - ], - }], ], } diff --git a/talk/build/common.gypi b/talk/build/common.gypi index 6da1449da2..4ee78f1c47 100644 --- a/talk/build/common.gypi +++ b/talk/build/common.gypi @@ -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': [ diff --git a/talk/build/isolate.gypi b/talk/build/isolate.gypi deleted file mode 100644 index 65cf34e2b6..0000000000 --- a/talk/build/isolate.gypi +++ /dev/null @@ -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', - ], - }], - ], - }, - ], -} diff --git a/talk/build/merge_ios_libs.gyp b/talk/build/merge_ios_libs.gyp index 0ec6e85304..76313fb61a 100644 --- a/talk/build/merge_ios_libs.gyp +++ b/talk/build/merge_ios_libs.gyp @@ -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',], }, diff --git a/talk/libjingle.gyp b/talk/libjingle.gyp deleted file mode 100755 index 09daed7f17..0000000000 --- a/talk/libjingle.gyp +++ /dev/null @@ -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 - ], -} diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp index 1efd531456..34a75f60f5 100644 --- a/webrtc/api/api.gyp +++ b/webrtc/api/api.gyp @@ -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', diff --git a/webrtc/api/api_tests.gyp b/webrtc/api/api_tests.gyp index 31bc6999ff..5af1cabfeb 100644 --- a/webrtc/api/api_tests.gyp +++ b/webrtc/api/api_tests.gyp @@ -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': [ diff --git a/webrtc/api/datachannel.h b/webrtc/api/datachannel.h index eeac1d0bbb..62e3eaf221 100644 --- a/webrtc/api/datachannel.h +++ b/webrtc/api/datachannel.h @@ -15,13 +15,13 @@ #include #include -#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 { diff --git a/webrtc/api/jsepsessiondescription.cc b/webrtc/api/jsepsessiondescription.cc index 6ea62499a6..b76b167193 100644 --- a/webrtc/api/jsepsessiondescription.cc +++ b/webrtc/api/jsepsessiondescription.cc @@ -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; diff --git a/webrtc/api/jsepsessiondescription_unittest.cc b/webrtc/api/jsepsessiondescription_unittest.cc index c49f5f57b4..7e505a122c 100644 --- a/webrtc/api/jsepsessiondescription_unittest.cc +++ b/webrtc/api/jsepsessiondescription_unittest.cc @@ -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; diff --git a/webrtc/api/mediacontroller.cc b/webrtc/api/mediacontroller.cc index 28438fb2f6..0c7d73389e 100644 --- a/webrtc/api/mediacontroller.cc +++ b/webrtc/api/mediacontroller.cc @@ -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 { diff --git a/webrtc/api/peerconnection.cc b/webrtc/api/peerconnection.cc index a076f3ca3a..70800880fb 100644 --- a/webrtc/api/peerconnection.cc +++ b/webrtc/api/peerconnection.cc @@ -15,7 +15,6 @@ #include #include -#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 { diff --git a/webrtc/api/peerconnection_unittest.cc b/webrtc/api/peerconnection_unittest.cc index 9cb0faf5a3..2b4e3e3e3b 100644 --- a/webrtc/api/peerconnection_unittest.cc +++ b/webrtc/api/peerconnection_unittest.cc @@ -16,7 +16,6 @@ #include #include -#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())) { \ diff --git a/webrtc/api/peerconnectionfactory.h b/webrtc/api/peerconnectionfactory.h index b33aa37c10..548986cb10 100644 --- a/webrtc/api/peerconnectionfactory.h +++ b/webrtc/api/peerconnectionfactory.h @@ -13,7 +13,6 @@ #include -#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; diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc index 7178ac3b5c..b9629f5043 100644 --- a/webrtc/api/peerconnectioninterface_unittest.cc +++ b/webrtc/api/peerconnectioninterface_unittest.cc @@ -11,7 +11,6 @@ #include #include -#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"; diff --git a/webrtc/api/rtpsenderinterface.h b/webrtc/api/rtpsenderinterface.h index 762eb4be58..c3dd52fe16 100644 --- a/webrtc/api/rtpsenderinterface.h +++ b/webrtc/api/rtpsenderinterface.h @@ -16,11 +16,11 @@ #include -#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 { diff --git a/webrtc/api/statscollector.cc b/webrtc/api/statscollector.cc index a47e8d7af9..ca1548fa0d 100644 --- a/webrtc/api/statscollector.cc +++ b/webrtc/api/statscollector.cc @@ -14,11 +14,11 @@ #include #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; diff --git a/webrtc/api/statscollector_unittest.cc b/webrtc/api/statscollector_unittest.cc index 0514c67114..cdb4bd0d75 100644 --- a/webrtc/api/statscollector_unittest.cc +++ b/webrtc/api/statscollector_unittest.cc @@ -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::_; diff --git a/webrtc/api/videosource.cc b/webrtc/api/videosource.cc index b37f998a6b..3a55cd5018 100644 --- a/webrtc/api/videosource.cc +++ b/webrtc/api/videosource.cc @@ -14,8 +14,8 @@ #include #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; diff --git a/webrtc/api/videosource_unittest.cc b/webrtc/api/videosource_unittest.cc index d8ca1a5787..2d163ef9a0 100644 --- a/webrtc/api/videosource_unittest.cc +++ b/webrtc/api/videosource_unittest.cc @@ -11,7 +11,6 @@ #include #include -#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; diff --git a/webrtc/api/videotrack_unittest.cc b/webrtc/api/videotrack_unittest.cc index face9f90b4..0f82e1830b 100644 --- a/webrtc/api/videotrack_unittest.cc +++ b/webrtc/api/videotrack_unittest.cc @@ -10,7 +10,6 @@ #include -#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; diff --git a/webrtc/api/webrtcsdp.cc b/webrtc/api/webrtcsdp.cc index 19e571c35b..5fd200a576 100644 --- a/webrtc/api/webrtcsdp.cc +++ b/webrtc/api/webrtcsdp.cc @@ -17,7 +17,6 @@ #include #include -#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; diff --git a/webrtc/api/webrtcsdp_unittest.cc b/webrtc/api/webrtcsdp_unittest.cc index 30efa1eee9..7f54a9dab3 100644 --- a/webrtc/api/webrtcsdp_unittest.cc +++ b/webrtc/api/webrtcsdp_unittest.cc @@ -12,7 +12,6 @@ #include #include -#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; diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc index 92a5f0aec4..56bea6403d 100644 --- a/webrtc/api/webrtcsession.cc +++ b/webrtc/api/webrtcsession.cc @@ -17,9 +17,6 @@ #include #include -#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; diff --git a/webrtc/api/webrtcsession.h b/webrtc/api/webrtcsession.h index ae68cfca30..e8b2a05af6 100644 --- a/webrtc/api/webrtcsession.h +++ b/webrtc/api/webrtcsession.h @@ -14,7 +14,6 @@ #include #include -#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 { diff --git a/webrtc/api/webrtcsession_unittest.cc b/webrtc/api/webrtcsession_unittest.cc index 56699baafa..1fe3a81592 100644 --- a/webrtc/api/webrtcsession_unittest.cc +++ b/webrtc/api/webrtcsession_unittest.cc @@ -11,8 +11,6 @@ #include #include -#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())) { \ diff --git a/webrtc/api/webrtcsessiondescriptionfactory.h b/webrtc/api/webrtcsessiondescriptionfactory.h index a7ff949c80..77771687a3 100644 --- a/webrtc/api/webrtcsessiondescriptionfactory.h +++ b/webrtc/api/webrtcsessiondescriptionfactory.h @@ -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; diff --git a/webrtc/build/common.gypi b/webrtc/build/common.gypi index 429dd6baba..3459a55d09 100644 --- a/webrtc/build/common.gypi +++ b/webrtc/build/common.gypi @@ -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, diff --git a/webrtc/libjingle/libjingle.gyp b/webrtc/libjingle/libjingle.gyp index 3c3996905d..b2322c68ec 100644 --- a/webrtc/libjingle/libjingle.gyp +++ b/webrtc/libjingle/libjingle.gyp @@ -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', { diff --git a/talk/libjingle_p2p_unittest.isolate b/webrtc/libjingle_p2p_unittest.isolate similarity index 100% rename from talk/libjingle_p2p_unittest.isolate rename to webrtc/libjingle_p2p_unittest.isolate diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h index 2c7696ceb9..e32eb6078e 100644 --- a/webrtc/media/base/mediachannel.h +++ b/webrtc/media/base/mediachannel.h @@ -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; diff --git a/webrtc/media/engine/webrtcvoiceengine.h b/webrtc/media/engine/webrtcvoiceengine.h index 75d71f67c3..fac7fa6f30 100644 --- a/webrtc/media/engine/webrtcvoiceengine.h +++ b/webrtc/media/engine/webrtcvoiceengine.h @@ -15,7 +15,6 @@ #include #include -#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 { diff --git a/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/webrtc/media/engine/webrtcvoiceengine_unittest.cc index 15ac43d2e5..2eafb47961 100644 --- a/webrtc/media/engine/webrtcvoiceengine_unittest.cc +++ b/webrtc/media/engine/webrtcvoiceengine_unittest.cc @@ -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" diff --git a/talk/session/media/audiomonitor.cc b/webrtc/pc/audiomonitor.cc similarity index 97% rename from talk/session/media/audiomonitor.cc rename to webrtc/pc/audiomonitor.cc index bdb71cb2f6..7eb6c36394 100644 --- a/talk/session/media/audiomonitor.cc +++ b/webrtc/pc/audiomonitor.cc @@ -26,8 +26,8 @@ */ #include -#include "talk/session/media/audiomonitor.h" -#include "talk/session/media/voicechannel.h" +#include "webrtc/pc/audiomonitor.h" +#include "webrtc/pc/voicechannel.h" namespace cricket { diff --git a/talk/session/media/audiomonitor.h b/webrtc/pc/audiomonitor.h similarity index 100% rename from talk/session/media/audiomonitor.h rename to webrtc/pc/audiomonitor.h diff --git a/talk/session/media/bundlefilter.cc b/webrtc/pc/bundlefilter.cc similarity index 98% rename from talk/session/media/bundlefilter.cc rename to webrtc/pc/bundlefilter.cc index 2cb4701bf0..9221a668a1 100644 --- a/talk/session/media/bundlefilter.cc +++ b/webrtc/pc/bundlefilter.cc @@ -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" diff --git a/talk/session/media/bundlefilter.h b/webrtc/pc/bundlefilter.h similarity index 100% rename from talk/session/media/bundlefilter.h rename to webrtc/pc/bundlefilter.h diff --git a/talk/session/media/bundlefilter_unittest.cc b/webrtc/pc/bundlefilter_unittest.cc similarity index 98% rename from talk/session/media/bundlefilter_unittest.cc rename to webrtc/pc/bundlefilter_unittest.cc index f2c35fc1d8..07377677ad 100755 --- a/talk/session/media/bundlefilter_unittest.cc +++ b/webrtc/pc/bundlefilter_unittest.cc @@ -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; diff --git a/talk/session/media/channel.cc b/webrtc/pc/channel.cc similarity index 99% rename from talk/session/media/channel.cc rename to webrtc/pc/channel.cc index bee37718ad..892f2b270f 100644 --- a/talk/session/media/channel.cc +++ b/webrtc/pc/channel.cc @@ -27,9 +27,8 @@ #include -#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; diff --git a/talk/session/media/channel.h b/webrtc/pc/channel.h similarity index 98% rename from talk/session/media/channel.h rename to webrtc/pc/channel.h index dfc0ce517d..5d42076903 100644 --- a/talk/session/media/channel.h +++ b/webrtc/pc/channel.h @@ -34,12 +34,6 @@ #include #include -#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; diff --git a/talk/session/media/channel_unittest.cc b/webrtc/pc/channel_unittest.cc similarity index 99% rename from talk/session/media/channel_unittest.cc rename to webrtc/pc/channel_unittest.cc index 5bbe1e7d8c..655925b5e9 100644 --- a/talk/session/media/channel_unittest.cc +++ b/webrtc/pc/channel_unittest.cc @@ -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())) { \ diff --git a/talk/session/media/channelmanager.cc b/webrtc/pc/channelmanager.cc similarity index 99% rename from talk/session/media/channelmanager.cc rename to webrtc/pc/channelmanager.cc index 8124f28364..ac2df8b0b8 100644 --- a/talk/session/media/channelmanager.cc +++ b/webrtc/pc/channelmanager.cc @@ -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 @@ -33,7 +33,6 @@ #include -#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 { diff --git a/talk/session/media/channelmanager.h b/webrtc/pc/channelmanager.h similarity index 99% rename from talk/session/media/channelmanager.h rename to webrtc/pc/channelmanager.h index de8555208f..2b1e9ea543 100644 --- a/talk/session/media/channelmanager.h +++ b/webrtc/pc/channelmanager.h @@ -31,13 +31,13 @@ #include #include -#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; diff --git a/talk/session/media/channelmanager_unittest.cc b/webrtc/pc/channelmanager_unittest.cc similarity index 99% rename from talk/session/media/channelmanager_unittest.cc rename to webrtc/pc/channelmanager_unittest.cc index 8137436ea9..70d9c6f23e 100644 --- a/talk/session/media/channelmanager_unittest.cc +++ b/webrtc/pc/channelmanager_unittest.cc @@ -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 { diff --git a/talk/session/media/currentspeakermonitor.cc b/webrtc/pc/currentspeakermonitor.cc similarity index 98% rename from talk/session/media/currentspeakermonitor.cc rename to webrtc/pc/currentspeakermonitor.cc index 4008f0e901..cc21023415 100644 --- a/talk/session/media/currentspeakermonitor.cc +++ b/webrtc/pc/currentspeakermonitor.cc @@ -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 { diff --git a/talk/session/media/currentspeakermonitor.h b/webrtc/pc/currentspeakermonitor.h similarity index 100% rename from talk/session/media/currentspeakermonitor.h rename to webrtc/pc/currentspeakermonitor.h diff --git a/talk/session/media/currentspeakermonitor_unittest.cc b/webrtc/pc/currentspeakermonitor_unittest.cc similarity index 98% rename from talk/session/media/currentspeakermonitor_unittest.cc rename to webrtc/pc/currentspeakermonitor_unittest.cc index fa2a3ce893..41e2f8d633 100644 --- a/talk/session/media/currentspeakermonitor_unittest.cc +++ b/webrtc/pc/currentspeakermonitor_unittest.cc @@ -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 { diff --git a/talk/session/media/externalhmac.cc b/webrtc/pc/externalhmac.cc similarity index 99% rename from talk/session/media/externalhmac.cc rename to webrtc/pc/externalhmac.cc index dd991d4848..28dbed736a 100644 --- a/talk/session/media/externalhmac.cc +++ b/webrtc/pc/externalhmac.cc @@ -27,7 +27,7 @@ #if defined(HAVE_SRTP) && defined(ENABLE_EXTERNAL_AUTH) -#include "talk/session/media/externalhmac.h" +#include "webrtc/pc/externalhmac.h" #include // For malloc/free. diff --git a/talk/session/media/externalhmac.h b/webrtc/pc/externalhmac.h similarity index 100% rename from talk/session/media/externalhmac.h rename to webrtc/pc/externalhmac.h diff --git a/talk/session/media/mediamonitor.cc b/webrtc/pc/mediamonitor.cc similarity index 97% rename from talk/session/media/mediamonitor.cc rename to webrtc/pc/mediamonitor.cc index f717062201..d6cb89dbf0 100644 --- a/talk/session/media/mediamonitor.cc +++ b/webrtc/pc/mediamonitor.cc @@ -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 { diff --git a/talk/session/media/mediamonitor.h b/webrtc/pc/mediamonitor.h similarity index 100% rename from talk/session/media/mediamonitor.h rename to webrtc/pc/mediamonitor.h diff --git a/talk/session/media/mediasession.cc b/webrtc/pc/mediasession.cc similarity index 99% rename from talk/session/media/mediasession.cc rename to webrtc/pc/mediasession.cc index b6f52c2b62..8512eb2527 100644 --- a/talk/session/media/mediasession.cc +++ b/webrtc/pc/mediasession.cc @@ -25,15 +25,13 @@ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ -#include "talk/session/media/mediasession.h" +#include "webrtc/pc/mediasession.h" #include #include #include #include -#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" diff --git a/talk/session/media/mediasession.h b/webrtc/pc/mediasession.h similarity index 100% rename from talk/session/media/mediasession.h rename to webrtc/pc/mediasession.h diff --git a/talk/session/media/mediasession_unittest.cc b/webrtc/pc/mediasession_unittest.cc similarity index 99% rename from talk/session/media/mediasession_unittest.cc rename to webrtc/pc/mediasession_unittest.cc index 7c61d64d3b..b9fb66492a 100644 --- a/talk/session/media/mediasession_unittest.cc +++ b/webrtc/pc/mediasession_unittest.cc @@ -28,8 +28,6 @@ #include #include -#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) \ diff --git a/talk/session/media/mediasink.h b/webrtc/pc/mediasink.h similarity index 100% rename from talk/session/media/mediasink.h rename to webrtc/pc/mediasink.h diff --git a/webrtc/pc/pc.gyp b/webrtc/pc/pc.gyp new file mode 100755 index 0000000000..40874b1a45 --- /dev/null +++ b/webrtc/pc/pc.gyp @@ -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 + ], +} diff --git a/webrtc/pc/pc_tests.gypi b/webrtc/pc/pc_tests.gypi new file mode 100755 index 0000000000..0245dde055 --- /dev/null +++ b/webrtc/pc/pc_tests.gypi @@ -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 + ], +} diff --git a/talk/session/media/planarfunctions_unittest.cc b/webrtc/pc/planarfunctions_unittest.cc similarity index 100% rename from talk/session/media/planarfunctions_unittest.cc rename to webrtc/pc/planarfunctions_unittest.cc diff --git a/talk/session/media/rtcpmuxfilter.cc b/webrtc/pc/rtcpmuxfilter.cc similarity index 99% rename from talk/session/media/rtcpmuxfilter.cc rename to webrtc/pc/rtcpmuxfilter.cc index f21e0eeb5b..eef5f0c657 100644 --- a/talk/session/media/rtcpmuxfilter.cc +++ b/webrtc/pc/rtcpmuxfilter.cc @@ -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" diff --git a/talk/session/media/rtcpmuxfilter.h b/webrtc/pc/rtcpmuxfilter.h similarity index 100% rename from talk/session/media/rtcpmuxfilter.h rename to webrtc/pc/rtcpmuxfilter.h diff --git a/talk/session/media/rtcpmuxfilter_unittest.cc b/webrtc/pc/rtcpmuxfilter_unittest.cc similarity index 99% rename from talk/session/media/rtcpmuxfilter_unittest.cc rename to webrtc/pc/rtcpmuxfilter_unittest.cc index edc4b39918..e16c9567fb 100644 --- a/talk/session/media/rtcpmuxfilter_unittest.cc +++ b/webrtc/pc/rtcpmuxfilter_unittest.cc @@ -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; diff --git a/talk/session/media/srtpfilter.cc b/webrtc/pc/srtpfilter.cc similarity index 99% rename from talk/session/media/srtpfilter.cc rename to webrtc/pc/srtpfilter.cc index 4e6cbe846e..e81d95bdf6 100644 --- a/talk/session/media/srtpfilter.cc +++ b/webrtc/pc/srtpfilter.cc @@ -27,7 +27,7 @@ #undef HAVE_CONFIG_H -#include "talk/session/media/srtpfilter.h" +#include "webrtc/pc/srtpfilter.h" #include @@ -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; diff --git a/talk/session/media/srtpfilter.h b/webrtc/pc/srtpfilter.h similarity index 100% rename from talk/session/media/srtpfilter.h rename to webrtc/pc/srtpfilter.h diff --git a/talk/session/media/srtpfilter_unittest.cc b/webrtc/pc/srtpfilter_unittest.cc similarity index 99% rename from talk/session/media/srtpfilter_unittest.cc rename to webrtc/pc/srtpfilter_unittest.cc index af1ea33f27..869673c681 100644 --- a/talk/session/media/srtpfilter_unittest.cc +++ b/webrtc/pc/srtpfilter_unittest.cc @@ -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" diff --git a/talk/session/media/typewrapping.h.pump b/webrtc/pc/typewrapping.h.pump similarity index 100% rename from talk/session/media/typewrapping.h.pump rename to webrtc/pc/typewrapping.h.pump diff --git a/talk/session/media/voicechannel.h b/webrtc/pc/voicechannel.h similarity index 97% rename from talk/session/media/voicechannel.h rename to webrtc/pc/voicechannel.h index 6c1b6afdd5..5608767f87 100644 --- a/talk/session/media/voicechannel.h +++ b/webrtc/pc/voicechannel.h @@ -28,6 +28,6 @@ #ifndef _VOICECHANNEL_H_ #define _VOICECHANNEL_H_ -#include "talk/session/media/channel.h" +#include "webrtc/pc/channel.h" #endif // _VOICECHANNEL_H_ diff --git a/talk/session/media/yuvscaler_unittest.cc b/webrtc/pc/yuvscaler_unittest.cc similarity index 100% rename from talk/session/media/yuvscaler_unittest.cc rename to webrtc/pc/yuvscaler_unittest.cc diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp index be911088d4..1adae73f9e 100644 --- a/webrtc/webrtc.gyp +++ b/webrtc/webrtc.gyp @@ -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', ], diff --git a/webrtc/webrtc_examples.gyp b/webrtc/webrtc_examples.gyp index 51cd7929d5..1ce450b0f1 100755 --- a/webrtc/webrtc_examples.gyp +++ b/webrtc/webrtc_examples.gyp @@ -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"', { diff --git a/webrtc/webrtc_tests.gypi b/webrtc/webrtc_tests.gypi index add7f6e13c..59bfe40b9d 100644 --- a/webrtc/webrtc_tests.gypi +++ b/webrtc/webrtc_tests.gypi @@ -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',