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Move talk/app/webrtc to webrtc/api

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The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.

License headers will be updated in a follow-up CL.

Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
  except for these files:
  talk/app/webrtc/peerconnectionendtoend_unittest.cc
  talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
  talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
  webrtc/media/devices/win32devicemanager.cc

The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.

I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.

For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002

BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1610243002 .

Cr-Commit-Position: refs/heads/master@{#11545}
This commit is contained in:
Henrik Kjellander
2016-02-10 10:53:12 +01:00
parent f5368ab81d
commit 15583c19d7
263 changed files with 1171 additions and 996 deletions
Download Patch File Download Diff File Expand all files Collapse all files

5
talk/app/webrtc/OWNERS
View File

@ -1,5 +0,0 @@
glaznev@webrtc.org
juberti@webrtc.org
perkj@webrtc.org
tkchin@webrtc.org
tommi@webrtc.org

2
talk/app/webrtc/objc/RTCAudioTrack+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCAudioTrack.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
@interface RTCAudioTrack (Internal)

2
talk/app/webrtc/objc/RTCDataChannel+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCDataChannel.h"
#include "talk/app/webrtc/datachannelinterface.h"
#include "webrtc/api/datachannelinterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
@interface RTCDataBuffer (Internal)

2
talk/app/webrtc/objc/RTCDataChannel.mm
View File

@ -31,7 +31,7 @@
#import "RTCDataChannel+Internal.h"
#include "talk/app/webrtc/datachannelinterface.h"
#include "webrtc/api/datachannelinterface.h"
namespace webrtc {

2
talk/app/webrtc/objc/RTCEnumConverter.mm
View File

@ -27,7 +27,7 @@
#import "RTCEnumConverter.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
@implementation RTCEnumConverter

2
talk/app/webrtc/objc/RTCICECandidate+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCICECandidate.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
@interface RTCICECandidate (Internal)

2
talk/app/webrtc/objc/RTCICEServer+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCICEServer.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
@interface RTCICEServer (Internal)

2
talk/app/webrtc/objc/RTCMediaConstraints+Internal.h
View File

@ -29,7 +29,7 @@
#import "RTCMediaConstraintsNative.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
@interface RTCMediaConstraints (Internal)

2
talk/app/webrtc/objc/RTCMediaConstraintsNative.h
View File

@ -28,7 +28,7 @@
#ifndef TALK_APP_WEBRTC_OBJC_RTCMEDIACONSTRAINTSNATIVE_H_
#define TALK_APP_WEBRTC_OBJC_RTCMEDIACONSTRAINTSNATIVE_H_
#include "talk/app/webrtc/mediaconstraintsinterface.h"
#include "webrtc/api/mediaconstraintsinterface.h"
namespace webrtc {
class RTCMediaConstraintsNative : public MediaConstraintsInterface {

2
talk/app/webrtc/objc/RTCMediaSource+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCMediaSource.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
@interface RTCMediaSource (Internal)

2
talk/app/webrtc/objc/RTCMediaStream+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCMediaStream.h"
#include "talk/app/webrtc/mediastreamtrack.h"
#include "webrtc/api/mediastreamtrack.h"
@interface RTCMediaStream (Internal)

2
talk/app/webrtc/objc/RTCMediaStream.mm
View File

@ -35,7 +35,7 @@
#import "RTCMediaStreamTrack+Internal.h"
#import "RTCVideoTrack+Internal.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
@implementation RTCMediaStream {
NSMutableArray* _audioTracks;

2
talk/app/webrtc/objc/RTCMediaStreamTrack+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCMediaStreamTrack.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
@interface RTCMediaStreamTrack (Internal)

2
talk/app/webrtc/objc/RTCPeerConnection+Internal.h
View File

@ -29,7 +29,7 @@
#import "RTCPeerConnectionDelegate.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
@interface RTCPeerConnection (Internal)

2
talk/app/webrtc/objc/RTCPeerConnection.mm
View File

@ -45,7 +45,7 @@
#import "RTCStatsDelegate.h"
#import "RTCStatsReport+Internal.h"
#include "talk/app/webrtc/jsep.h"
#include "webrtc/api/jsep.h"
NSString* const kRTCSessionDescriptionDelegateErrorDomain = @"RTCSDPError";
int const kRTCSessionDescriptionDelegateErrorCode = -1;

2
talk/app/webrtc/objc/RTCPeerConnectionFactory+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCPeerConnectionFactory.h"
#include "talk/app/webrtc/peerconnectionfactory.h"
#include "webrtc/api/peerconnectionfactory.h"
#include "webrtc/base/scoped_ptr.h"
@interface RTCPeerConnectionFactory ()

10
talk/app/webrtc/objc/RTCPeerConnectionFactory.mm
View File

@ -46,11 +46,11 @@
#import "RTCVideoSource+Internal.h"
#import "RTCVideoTrack+Internal.h"
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "talk/app/webrtc/videotrack.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/videosourceinterface.h"
#include "webrtc/api/videotrack.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"

2
talk/app/webrtc/objc/RTCPeerConnectionInterface+Internal.h
View File

@ -27,7 +27,7 @@
#import "talk/app/webrtc/objc/public/RTCPeerConnectionInterface.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
@interface RTCConfiguration ()

2
talk/app/webrtc/objc/RTCPeerConnectionObserver.h
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#import "RTCPeerConnection.h"
#import "RTCPeerConnectionDelegate.h"

4
talk/app/webrtc/objc/RTCSessionDescription+Internal.h
View File

@ -27,8 +27,8 @@
#import "RTCSessionDescription.h"
#include "talk/app/webrtc/jsep.h"
#include "talk/app/webrtc/webrtcsession.h"
#include "webrtc/api/jsep.h"
#include "webrtc/api/webrtcsession.h"
@interface RTCSessionDescription (Internal)

2
talk/app/webrtc/objc/RTCStatsReport+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCStatsReport.h"
#include "talk/app/webrtc/statstypes.h"
#include "webrtc/api/statstypes.h"
@interface RTCStatsReport (Internal)

2
talk/app/webrtc/objc/RTCVideoCapturer+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCVideoCapturer.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "webrtc/api/videosourceinterface.h"
@interface RTCVideoCapturer (Internal)

2
talk/app/webrtc/objc/RTCVideoRendererAdapter.h
View File

@ -27,7 +27,7 @@
#import "RTCVideoRenderer.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
@interface RTCVideoRendererAdapter : NSObject

2
talk/app/webrtc/objc/RTCVideoRendererAdapter.mm
View File

@ -29,8 +29,8 @@
#error "This file requires ARC support."
#endif
#import "RTCVideoRendererAdapter.h"
#import "RTCI420Frame+Internal.h"
#import "RTCVideoRendererAdapter.h"
namespace webrtc {

2
talk/app/webrtc/objc/RTCVideoSource+Internal.h
View File

@ -27,7 +27,7 @@
#import "RTCVideoSource.h"
#include "talk/app/webrtc/videosourceinterface.h"
#include "webrtc/api/videosourceinterface.h"
@interface RTCVideoSource (Internal)

4
talk/app/webrtc/objc/RTCVideoTrack+Internal.h
View File

@ -27,8 +27,8 @@
#import "RTCVideoTrack.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
@class RTCVideoRenderer;

2
talk/app/webrtc/objc/RTCVideoTrack.mm
View File

@ -31,10 +31,10 @@
#import "RTCVideoTrack+Internal.h"
#import "RTCMediaSource+Internal.h"
#import "RTCMediaStreamTrack+Internal.h"
#import "RTCPeerConnectionFactory+Internal.h"
#import "RTCVideoRendererAdapter.h"
#import "RTCMediaSource+Internal.h"
#import "RTCVideoSource+Internal.h"
@implementation RTCVideoTrack {

5
talk/build/common.gypi
View File

@ -33,11 +33,6 @@
# TODO(ronghuawu): For now, disable the Chrome plugins, which causes a
# flood of chromium-style warnings.
'clang_use_chrome_plugins%': 0,
'conditions': [
['OS=="android" or OS=="linux"', {
'java_home%': '<!(python -c "import os; dir=os.getenv(\'JAVA_HOME\', \'/usr/lib/jvm/java-7-openjdk-amd64\'); assert os.path.exists(os.path.join(dir, \'include/jni.h\')), \'Point \\$JAVA_HOME or the java_home gyp variable to a directory containing include/jni.h!\'; print dir")',
}],
],
# Disable these to not build components which can be externally provided.
'build_expat%': 1,
'build_json%': 1,

193
talk/libjingle.gyp
View File

@ -27,113 +27,6 @@
{
'includes': ['build/common.gypi'],
'conditions': [
['os_posix == 1 and OS != "mac" and OS != "ios"', {
'conditions': [
['sysroot!=""', {
'variables': {
'pkg-config': '../../../build/linux/pkg-config-wrapper "<(sysroot)" "<(target_arch)"',
},
}, {
'variables': {
'pkg-config': 'pkg-config'
},
}],
],
}],
['OS=="android"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_jni',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'libjingle_peerconnection',
],
'sources': [
'app/webrtc/androidvideocapturer.cc',
'app/webrtc/androidvideocapturer.h',
'app/webrtc/java/jni/androidmediacodeccommon.h',
'app/webrtc/java/jni/androidmediadecoder_jni.cc',
'app/webrtc/java/jni/androidmediadecoder_jni.h',
'app/webrtc/java/jni/androidmediaencoder_jni.cc',
'app/webrtc/java/jni/androidmediaencoder_jni.h',
'app/webrtc/java/jni/androidnetworkmonitor_jni.cc',
'app/webrtc/java/jni/androidnetworkmonitor_jni.h',
'app/webrtc/java/jni/androidvideocapturer_jni.cc',
'app/webrtc/java/jni/androidvideocapturer_jni.h',
'app/webrtc/java/jni/classreferenceholder.cc',
'app/webrtc/java/jni/classreferenceholder.h',
'app/webrtc/java/jni/eglbase_jni.cc',
'app/webrtc/java/jni/eglbase_jni.h',
'app/webrtc/java/jni/jni_helpers.cc',
'app/webrtc/java/jni/jni_helpers.h',
'app/webrtc/java/jni/native_handle_impl.cc',
'app/webrtc/java/jni/native_handle_impl.h',
'app/webrtc/java/jni/peerconnection_jni.cc',
'app/webrtc/java/jni/surfacetexturehelper_jni.cc',
'app/webrtc/java/jni/surfacetexturehelper_jni.h',
],
'include_dirs': [
'<(libyuv_dir)/include',
],
'conditions': [
['build_json==1', {
'dependencies': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
}],
],
},
{
'target_name': 'libjingle_peerconnection_so',
'type': 'shared_library',
'dependencies': [
'libjingle_peerconnection',
'libjingle_peerconnection_jni',
],
'sources': [
'app/webrtc/java/jni/jni_onload.cc',
],
'variables': {
# This library uses native JNI exports; tell GYP so that the
# required symbols will be kept.
'use_native_jni_exports': 1,
},
},
{
# |libjingle_peerconnection_java| builds a jar file with name
# libjingle_peerconnection_java.jar using Chromes build system.
# It includes all Java files needed to setup a PeeerConnection call
# from Android.
'target_name': 'libjingle_peerconnection_java',
'type': 'none',
'dependencies': [
'libjingle_peerconnection_so',
],
'variables': {
# Designate as Chromium code and point to our lint settings to
# enable linting of the WebRTC code (this is the only way to make
# lint_action invoke the Android linter).
'android_manifest_path': '<(webrtc_root)/build/android/AndroidManifest.xml',
'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml',
'chromium_code': 1,
'java_in_dir': 'app/webrtc/java',
'webrtc_base_dir': '<(webrtc_root)/base',
'webrtc_modules_dir': '<(webrtc_root)/modules',
'additional_src_dirs' : [
'app/webrtc/java/android',
'<(webrtc_base_dir)/java/src',
'<(webrtc_modules_dir)/audio_device/android/java/src',
'<(webrtc_modules_dir)/video_render/android/java/src',
],
},
'includes': ['../build/java.gypi'],
}, # libjingle_peerconnection_java
]
}],
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >= 10.7 above is required for ARC.
'targets': [
@ -141,7 +34,7 @@
'target_name': 'libjingle_peerconnection_objc',
'type': 'static_library',
'dependencies': [
'libjingle_peerconnection',
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
],
'sources': [
'app/webrtc/objc/RTCAudioTrack+Internal.h',
@ -223,7 +116,7 @@
],
},
'include_dirs': [
'<(DEPTH)/talk/app/webrtc',
'<(webrtc_root)/webrtc/api',
'<(DEPTH)/talk/app/webrtc/objc',
'<(DEPTH)/talk/app/webrtc/objc/public',
],
@ -295,7 +188,6 @@
],
}],
],
'targets': [
{
'target_name': 'libjingle_p2p',
@ -348,86 +240,5 @@
'session/media/voicechannel.h',
],
}, # target libjingle_p2p
{
'target_name': 'libjingle_peerconnection',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base',
'<(webrtc_root)/media/media.gyp:rtc_media',
'libjingle_p2p',
],
'sources': [
'app/webrtc/audiotrack.cc',
'app/webrtc/audiotrack.h',
'app/webrtc/datachannel.cc',
'app/webrtc/datachannel.h',
'app/webrtc/datachannelinterface.h',
'app/webrtc/dtlsidentitystore.cc',
'app/webrtc/dtlsidentitystore.h',
'app/webrtc/dtmfsender.cc',
'app/webrtc/dtmfsender.h',
'app/webrtc/dtmfsenderinterface.h',
'app/webrtc/jsep.h',
'app/webrtc/jsepicecandidate.cc',
'app/webrtc/jsepicecandidate.h',
'app/webrtc/jsepsessiondescription.cc',
'app/webrtc/jsepsessiondescription.h',
'app/webrtc/localaudiosource.cc',
'app/webrtc/localaudiosource.h',
'app/webrtc/mediaconstraintsinterface.cc',
'app/webrtc/mediaconstraintsinterface.h',
'app/webrtc/mediacontroller.cc',
'app/webrtc/mediacontroller.h',
'app/webrtc/mediastream.cc',
'app/webrtc/mediastream.h',
'app/webrtc/mediastreaminterface.h',
'app/webrtc/mediastreamobserver.cc',
'app/webrtc/mediastreamobserver.h',
'app/webrtc/mediastreamprovider.h',
'app/webrtc/mediastreamproxy.h',
'app/webrtc/mediastreamtrack.h',
'app/webrtc/mediastreamtrackproxy.h',
'app/webrtc/notifier.h',
'app/webrtc/peerconnection.cc',
'app/webrtc/peerconnection.h',
'app/webrtc/peerconnectionfactory.cc',
'app/webrtc/peerconnectionfactory.h',
'app/webrtc/peerconnectionfactoryproxy.h',
'app/webrtc/peerconnectioninterface.h',
'app/webrtc/peerconnectionproxy.h',
'app/webrtc/proxy.h',
'app/webrtc/remoteaudiosource.cc',
'app/webrtc/remoteaudiosource.h',
'app/webrtc/remotevideocapturer.cc',
'app/webrtc/remotevideocapturer.h',
'app/webrtc/rtpreceiver.cc',
'app/webrtc/rtpreceiver.h',
'app/webrtc/rtpreceiverinterface.h',
'app/webrtc/rtpsender.cc',
'app/webrtc/rtpsender.h',
'app/webrtc/rtpsenderinterface.h',
'app/webrtc/sctputils.cc',
'app/webrtc/sctputils.h',
'app/webrtc/statscollector.cc',
'app/webrtc/statscollector.h',
'app/webrtc/statstypes.cc',
'app/webrtc/statstypes.h',
'app/webrtc/streamcollection.h',
'app/webrtc/videosource.cc',
'app/webrtc/videosource.h',
'app/webrtc/videosourceinterface.h',
'app/webrtc/videosourceproxy.h',
'app/webrtc/videotrack.cc',
'app/webrtc/videotrack.h',
'app/webrtc/videotrackrenderers.cc',
'app/webrtc/videotrackrenderers.h',
'app/webrtc/webrtcsdp.cc',
'app/webrtc/webrtcsdp.h',
'app/webrtc/webrtcsession.cc',
'app/webrtc/webrtcsession.h',
'app/webrtc/webrtcsessiondescriptionfactory.cc',
'app/webrtc/webrtcsessiondescriptionfactory.h',
],
}, # target libjingle_peerconnection
],
}

121
talk/libjingle_tests.gyp
View File

@ -31,9 +31,9 @@
'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_peerconnection',
'libjingle.gyp:libjingle_p2p',
],
'include_dirs': [
@ -65,101 +65,8 @@
}],
],
}, # target libjingle_p2p_unittest
{
'target_name': 'peerconnection_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/common.gyp:webrtc_common',
'<(webrtc_root)/webrtc.gyp:rtc_unittest_main',
'libjingle.gyp:libjingle_p2p',
'libjingle.gyp:libjingle_peerconnection',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/testing/gmock/include',
],
},
'sources': [
'app/webrtc/datachannel_unittest.cc',
'app/webrtc/dtlsidentitystore_unittest.cc',
'app/webrtc/dtmfsender_unittest.cc',
'app/webrtc/fakemetricsobserver.cc',
'app/webrtc/fakemetricsobserver.h',
'app/webrtc/jsepsessiondescription_unittest.cc',
'app/webrtc/localaudiosource_unittest.cc',
'app/webrtc/mediastream_unittest.cc',
'app/webrtc/peerconnection_unittest.cc',
'app/webrtc/peerconnectionendtoend_unittest.cc',
'app/webrtc/peerconnectionfactory_unittest.cc',
'app/webrtc/peerconnectioninterface_unittest.cc',
# 'app/webrtc/peerconnectionproxy_unittest.cc',
'app/webrtc/remotevideocapturer_unittest.cc',
'app/webrtc/rtpsenderreceiver_unittest.cc',
'app/webrtc/statscollector_unittest.cc',
'app/webrtc/test/fakeaudiocapturemodule.cc',
'app/webrtc/test/fakeaudiocapturemodule.h',
'app/webrtc/test/fakeaudiocapturemodule_unittest.cc',
'app/webrtc/test/fakeconstraints.h',
'app/webrtc/test/fakedatachannelprovider.h',
'app/webrtc/test/fakedtlsidentitystore.h',
'app/webrtc/test/fakeperiodicvideocapturer.h',
'app/webrtc/test/fakevideotrackrenderer.h',
'app/webrtc/test/mockpeerconnectionobservers.h',
'app/webrtc/test/peerconnectiontestwrapper.h',
'app/webrtc/test/peerconnectiontestwrapper.cc',
'app/webrtc/test/testsdpstrings.h',
'app/webrtc/videosource_unittest.cc',
'app/webrtc/videotrack_unittest.cc',
'app/webrtc/webrtcsdp_unittest.cc',
'app/webrtc/webrtcsession_unittest.cc',
],
'conditions': [
['OS=="android"', {
'sources': [
'app/webrtc/test/androidtestinitializer.cc',
'app/webrtc/test/androidtestinitializer.h',
],
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
'libjingle.gyp:libjingle_peerconnection_jni',
],
}],
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-unused-function',
],
},
},
}],
],
}, # target peerconnection_unittests
],
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_android_unittest',
'type': 'none',
'dependencies': [
'libjingle.gyp:libjingle_peerconnection_java',
],
'variables': {
'apk_name': 'libjingle_peerconnection_android_unittest',
'java_in_dir': 'app/webrtc/androidtests',
'resource_dir': 'app/webrtc/androidtests/res',
'native_lib_target': 'libjingle_peerconnection_so',
'is_test_apk': 1,
},
'includes': [ '../build/java_apk.gypi' ],
},
], # targets
}], # OS=="android"
['OS=="ios" or (OS=="mac" and target_arch!="ia32")', {
# The >=10.7 above is required to make ARC link cleanly (e.g. as
# opposed to _compile_ cleanly, which the library under test
@ -204,8 +111,8 @@
'dependencies': [
'<(webrtc_root)/base/base_tests.gyp:rtc_base_tests_utils',
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock',
'<(webrtc_root)/webrtc_examples.gyp:apprtc_signaling',
'<(DEPTH)/third_party/ocmock/ocmock.gyp:ocmock',
],
'sources': [
'app/webrtc/objctests/mac/main.mm',
@ -221,17 +128,6 @@
}, # target apprtc_signaling_gunit_test
],
}],
['OS=="android"', {
'targets': [
{
'target_name': 'peerconnection_unittests_apk_target',
'type': 'none',
'dependencies': [
'<(DEPTH)/webrtc/build/apk_tests.gyp:peerconnection_unittests_apk',
],
},
],
}],
['test_isolation_mode != "noop"', {
'targets': [
{
@ -247,19 +143,6 @@
'libjingle_p2p_unittest.isolate',
],
},
{
'target_name': 'peerconnection_unittests_run',
'type': 'none',
'dependencies': [
'peerconnection_unittests',
],
'includes': [
'build/isolate.gypi',
],
'sources': [
'peerconnection_unittests.isolate',
],
},
],
}],
],

18
talk/session/media/channelmanager.cc
View File

@ -33,7 +33,15 @@
#include <algorithm>
#include "talk/app/webrtc/mediacontroller.h"
#include "talk/session/media/srtpfilter.h"
#include "webrtc/api/mediacontroller.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/media/base/capturemanager.h"
#include "webrtc/media/base/device.h"
#include "webrtc/media/base/hybriddataengine.h"
@ -42,14 +50,6 @@
#ifdef HAVE_SCTP
#include "webrtc/media/sctp/sctpdataengine.h"
#endif
#include "talk/session/media/srtpfilter.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/sigslotrepeater.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
namespace cricket {

2
talk/session/media/channelmanager_unittest.cc
View File

@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/fakemediacontroller.h"
#include "talk/session/media/channelmanager.h"
#include "webrtc/api/fakemediacontroller.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"

5
webrtc/api/OWNERS
View File

@ -1 +1,6 @@
pthatcher@webrtc.org
glaznev@webrtc.org
juberti@webrtc.org
perkj@webrtc.org
tkchin@webrtc.org
tommi@webrtc.org

0
talk/app/webrtc/androidtests/AndroidManifest.xml → webrtc/api/androidtests/AndroidManifest.xml
View File

0
talk/app/webrtc/androidtests/OWNERS → webrtc/api/androidtests/OWNERS
View File

0
talk/app/webrtc/androidtests/ant.properties → webrtc/api/androidtests/ant.properties
View File

0
talk/app/webrtc/androidtests/build.xml → webrtc/api/androidtests/build.xml
View File

0
talk/app/webrtc/androidtests/project.properties → webrtc/api/androidtests/project.properties
View File

0
talk/app/webrtc/androidtests/res/drawable-hdpi/ic_launcher.png → webrtc/api/androidtests/res/drawable-hdpi/ic_launcher.png
View File

Side by Side Swipe Overlay

Before

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After

Width:  |  Height:  |  Size: 9.2 KiB

0
talk/app/webrtc/androidtests/res/drawable-ldpi/ic_launcher.png → webrtc/api/androidtests/res/drawable-ldpi/ic_launcher.png
View File

Side by Side Swipe Overlay

Before

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After

Width:  |  Height:  |  Size: 2.7 KiB

0
talk/app/webrtc/androidtests/res/drawable-mdpi/ic_launcher.png → webrtc/api/androidtests/res/drawable-mdpi/ic_launcher.png
View File

Side by Side Swipe Overlay

Before

Width:  |  Height:  |  Size: 5.1 KiB

After

Width:  |  Height:  |  Size: 5.1 KiB

0
talk/app/webrtc/androidtests/res/drawable-xhdpi/ic_launcher.png → webrtc/api/androidtests/res/drawable-xhdpi/ic_launcher.png
View File

Side by Side Swipe Overlay

Before

Width:  |  Height:  |  Size: 14 KiB

After

Width:  |  Height:  |  Size: 14 KiB

0
talk/app/webrtc/androidtests/res/values/strings.xml → webrtc/api/androidtests/res/values/strings.xml
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/GlRectDrawerTest.java → webrtc/api/androidtests/src/org/webrtc/GlRectDrawerTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java → webrtc/api/androidtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/NetworkMonitorTest.java → webrtc/api/androidtests/src/org/webrtc/NetworkMonitorTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/PeerConnectionTest.java → webrtc/api/androidtests/src/org/webrtc/PeerConnectionTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/RendererCommonTest.java → webrtc/api/androidtests/src/org/webrtc/RendererCommonTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java → webrtc/api/androidtests/src/org/webrtc/SurfaceTextureHelperTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java → webrtc/api/androidtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java → webrtc/api/androidtests/src/org/webrtc/VideoCapturerAndroidTest.java
View File

0
talk/app/webrtc/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java → webrtc/api/androidtests/src/org/webrtc/VideoCapturerAndroidTestFixtures.java
View File

5
talk/app/webrtc/androidvideocapturer.cc → webrtc/api/androidvideocapturer.cc
View File

@ -24,9 +24,10 @@
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/androidvideocapturer.h"
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
#include "webrtc/api/androidvideocapturer.h"
#include "webrtc/api/java/jni/native_handle_impl.h"
#include "webrtc/base/common.h"
#include "webrtc/base/json.h"
#include "webrtc/base/timeutils.h"

7
talk/app/webrtc/androidvideocapturer.h → webrtc/api/androidvideocapturer.h
View File

@ -24,8 +24,9 @@
* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_ANDROIDVIDEOCAPTURER_H_
#define TALK_APP_WEBRTC_ANDROIDVIDEOCAPTURER_H_
#ifndef WEBRTC_API_ANDROIDVIDEOCAPTURER_H_
#define WEBRTC_API_ANDROIDVIDEOCAPTURER_H_
#include <string>
#include <vector>
@ -105,4 +106,4 @@ class AndroidVideoCapturer : public cricket::VideoCapturer {
} // namespace webrtc
#endif // TALK_APP_WEBRTC_ANDROIDVIDEOCAPTURER_H_
#endif // WEBRTC_API_ANDROIDVIDEOCAPTURER_H_

249
webrtc/api/api.gyp
View File

@ -9,6 +9,131 @@
{
'includes': [ '../build/common.gypi', ],
'conditions': [
['os_posix == 1 and OS != "mac" and OS != "ios"', {
'conditions': [
['sysroot!=""', {
'variables': {
'pkg-config': '../../../build/linux/pkg-config-wrapper "<(sysroot)" "<(target_arch)"',
},
}, {
'variables': {
'pkg-config': 'pkg-config'
},
}],
],
}],
['OS=="android"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_jni',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/system_wrappers/system_wrappers.gyp:field_trial_default',
'libjingle_peerconnection',
],
'sources': [
'androidvideocapturer.cc',
'androidvideocapturer.h',
'java/jni/androidmediacodeccommon.h',
'java/jni/androidmediadecoder_jni.cc',
'java/jni/androidmediadecoder_jni.h',
'java/jni/androidmediaencoder_jni.cc',
'java/jni/androidmediaencoder_jni.h',
'java/jni/androidnetworkmonitor_jni.cc',
'java/jni/androidnetworkmonitor_jni.h',
'java/jni/androidvideocapturer_jni.cc',
'java/jni/androidvideocapturer_jni.h',
'java/jni/eglbase_jni.cc',
'java/jni/eglbase_jni.h',
'java/jni/surfacetexturehelper_jni.cc',
'java/jni/surfacetexturehelper_jni.h',
'java/jni/classreferenceholder.cc',
'java/jni/classreferenceholder.h',
'java/jni/jni_helpers.cc',
'java/jni/jni_helpers.h',
'java/jni/native_handle_impl.cc',
'java/jni/native_handle_impl.h',
'java/jni/peerconnection_jni.cc',
],
'include_dirs': [
'<(libyuv_dir)/include',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
'-Wno-sign-compare',
'-Wno-unused-variable',
],
'cflags!': [
'-Wextra',
],
'cflags_cc!': [
'-Wnon-virtual-dtor',
'-Woverloaded-virtual',
],
'msvs_disabled_warnings': [
4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch.
4267, # conversion from 'size_t' to 'int', possible loss of data.
4389, # signed/unsigned mismatch.
],
'conditions': [
['build_json==1', {
'dependencies': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
'export_dependent_settings': [
'<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
],
}],
],
},
{
'target_name': 'libjingle_peerconnection_so',
'type': 'shared_library',
'dependencies': [
'libjingle_peerconnection',
'libjingle_peerconnection_jni',
],
'sources': [
'java/jni/jni_onload.cc',
],
'variables': {
# This library uses native JNI exports; tell GYP so that the
# required symbols will be kept.
'use_native_jni_exports': 1,
},
},
{
# |libjingle_peerconnection_java| builds a jar file with name
# libjingle_peerconnection_java.jar using Chrome's build system.
# It includes all Java files needed to setup a PeeerConnection call
# from Android.
'target_name': 'libjingle_peerconnection_java',
'type': 'none',
'dependencies': [
'libjingle_peerconnection_so',
],
'variables': {
# Designate as Chromium code and point to our lint settings to
# enable linting of the WebRTC code (this is the only way to make
# lint_action invoke the Android linter).
'android_manifest_path': '<(webrtc_root)/build/android/AndroidManifest.xml',
'suppressions_file': '<(webrtc_root)/build/android/suppressions.xml',
'chromium_code': 1,
'java_in_dir': 'java',
'webrtc_base_dir': '<(webrtc_root)/base',
'webrtc_modules_dir': '<(webrtc_root)/modules',
'additional_src_dirs' : [
'java/android',
'<(webrtc_base_dir)/java/src',
'<(webrtc_modules_dir)/audio_device/android/java/src',
'<(webrtc_modules_dir)/video_render/android/java/src',
],
},
'includes': ['../../build/java.gypi'],
}, # libjingle_peerconnection_java
]
}],
['OS=="ios"', {
'targets': [
{
@ -16,7 +141,7 @@
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/base/base.gyp:rtc_base_objc',
'../../talk/libjingle.gyp:libjingle_peerconnection',
'libjingle_peerconnection',
],
'sources': [
'objc/RTCAVFoundationVideoSource+Private.h',
@ -118,5 +243,127 @@
}
],
}], # OS=="ios"
], # conditions
'targets': [
{
'target_name': 'libjingle_peerconnection',
'type': 'static_library',
'dependencies': [
'<(webrtc_root)/media/media.gyp:rtc_media',
'../../talk/libjingle.gyp:libjingle_p2p',
],
'sources': [
'audiotrack.cc',
'audiotrack.h',
'datachannel.cc',
'datachannel.h',
'datachannelinterface.h',
'dtlsidentitystore.cc',
'dtlsidentitystore.h',
'dtmfsender.cc',
'dtmfsender.h',
'dtmfsenderinterface.h',
'jsep.h',
'jsepicecandidate.cc',
'jsepicecandidate.h',
'jsepsessiondescription.cc',
'jsepsessiondescription.h',
'localaudiosource.cc',
'localaudiosource.h',
'mediaconstraintsinterface.cc',
'mediaconstraintsinterface.h',
'mediacontroller.cc',
'mediacontroller.h',
'mediastream.cc',
'mediastream.h',
'mediastreaminterface.h',
'mediastreamobserver.cc',
'mediastreamobserver.h',
'mediastreamprovider.h',
'mediastreamproxy.h',
'mediastreamtrack.h',
'mediastreamtrackproxy.h',
'notifier.h',
'peerconnection.cc',
'peerconnection.h',
'peerconnectionfactory.cc',
'peerconnectionfactory.h',
'peerconnectionfactoryproxy.h',
'peerconnectioninterface.h',
'peerconnectionproxy.h',
'proxy.h',
'remoteaudiosource.cc',
'remoteaudiosource.h',
'remotevideocapturer.cc',
'remotevideocapturer.h',
'rtpreceiver.cc',
'rtpreceiver.h',
'rtpreceiverinterface.h',
'rtpsender.cc',
'rtpsender.h',
'rtpsenderinterface.h',
'sctputils.cc',
'sctputils.h',
'statscollector.cc',
'statscollector.h',
'statstypes.cc',
'statstypes.h',
'streamcollection.h',
'videosource.cc',
'videosource.h',
'videosourceinterface.h',
'videosourceproxy.h',
'videotrack.cc',
'videotrack.h',
'videotrackrenderers.cc',
'videotrackrenderers.h',
'webrtcsdp.cc',
'webrtcsdp.h',
'webrtcsession.cc',
'webrtcsession.h',
'webrtcsessiondescriptionfactory.cc',
'webrtcsessiondescriptionfactory.h',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
'-Wno-sign-compare',
],
'cflags_cc!': [
'-Wnon-virtual-dtor',
'-Woverloaded-virtual',
],
'conditions': [
['clang==1', {
'cflags!': [
'-Wextra',
],
'xcode_settings': {
'WARNING_CFLAGS!': ['-Wextra'],
},
}, {
'cflags': [
'-Wno-maybe-uninitialized', # Only exists for GCC.
],
}],
['OS=="win"', {
# Disable warning for signed/unsigned mismatch.
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions!': ['/we4389'],
},
},
}],
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
'-Wno-sign-compare',
],
},
},
}],
],
}, # target libjingle_peerconnection
], # targets
}

158
webrtc/api/api_tests.gyp
View File

@ -8,7 +8,135 @@
{
'includes': [ '../build/common.gypi', ],
'targets': [
{
'target_name': 'peerconnection_unittests',
'type': '<(gtest_target_type)',
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection',
'<(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',
],
'direct_dependent_settings': {
'include_dirs': [
'<(DEPTH)/testing/gmock/include',
],
},
'defines': [
# Feature selection.
'HAVE_SCTP',
],
'sources': [
'datachannel_unittest.cc',
'dtlsidentitystore_unittest.cc',
'dtmfsender_unittest.cc',
'fakemetricsobserver.cc',
'fakemetricsobserver.h',
'jsepsessiondescription_unittest.cc',
'localaudiosource_unittest.cc',
'mediastream_unittest.cc',
'peerconnection_unittest.cc',
'peerconnectionendtoend_unittest.cc',
'peerconnectionfactory_unittest.cc',
'peerconnectioninterface_unittest.cc',
# 'peerconnectionproxy_unittest.cc',
'remotevideocapturer_unittest.cc',
'rtpsenderreceiver_unittest.cc',
'statscollector_unittest.cc',
'test/fakeaudiocapturemodule.cc',
'test/fakeaudiocapturemodule.h',
'test/fakeaudiocapturemodule_unittest.cc',
'test/fakeconstraints.h',
'test/fakedatachannelprovider.h',
'test/fakedtlsidentitystore.h',
'test/fakeperiodicvideocapturer.h',
'test/fakevideotrackrenderer.h',
'test/mockpeerconnectionobservers.h',
'test/peerconnectiontestwrapper.h',
'test/peerconnectiontestwrapper.cc',
'test/testsdpstrings.h',
'videosource_unittest.cc',
'videotrack_unittest.cc',
'webrtcsdp_unittest.cc',
'webrtcsession_unittest.cc',
],
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags': [
'-Wno-sign-compare',
],
'cflags!': [
'-Wextra',
],
'cflags_cc!': [
'-Wnon-virtual-dtor',
'-Woverloaded-virtual',
],
'msvs_disabled_warnings': [
4245, # conversion from 'int' to 'size_t', signed/unsigned mismatch.
4267, # conversion from 'size_t' to 'int', possible loss of data.
4389, # signed/unsigned mismatch.
],
'conditions': [
['clang==1', {
# TODO(kjellander): Make the code compile without disabling these flags.
# See https://bugs.chromium.org/p/webrtc/issues/detail?id=3307
'cflags!': [
'-Wextra',
],
'xcode_settings': {
'WARNING_CFLAGS!': ['-Wextra'],
},
}],
['OS=="android"', {
'sources': [
'test/androidtestinitializer.cc',
'test/androidtestinitializer.h',
],
'dependencies': [
'<(DEPTH)/testing/android/native_test.gyp:native_test_native_code',
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection_jni',
],
}],
['OS=="win" and clang==1', {
'msvs_settings': {
'VCCLCompilerTool': {
'AdditionalOptions': [
# Disable warnings failing when compiling with Clang on Windows.
# https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
'-Wno-sign-compare',
'-Wno-unused-function',
],
},
},
}],
], # conditions
}, # target peerconnection_unittests
], # targets
'conditions': [
['OS=="android"', {
'targets': [
{
'target_name': 'libjingle_peerconnection_android_unittest',
'type': 'none',
'dependencies': [
'<(webrtc_root)/api/api.gyp:libjingle_peerconnection_java',
],
'variables': {
'apk_name': 'libjingle_peerconnection_android_unittest',
'java_in_dir': 'androidtests',
'resource_dir': 'androidtests/res',
'native_lib_target': 'libjingle_peerconnection_so',
'is_test_apk': 1,
'never_lint': 1,
},
'includes': [ '../../build/java_apk.gypi' ],
},
], # targets
}], # OS=="android"
['OS=="ios"', {
'targets': [
{
@ -35,8 +163,36 @@
# https://developer.apple.com/library/mac/qa/qa1490/_index.html
'OTHER_LDFLAGS': ['-ObjC'],
},
}
},
],
}], # OS=="ios"
['OS=="android"', {
'targets': [
{
'target_name': 'peerconnection_unittests_apk_target',
'type': 'none',
'dependencies': [
'<(apk_tests_path):peerconnection_unittests_apk',
],
},
],
}], # OS=="android"
['test_isolation_mode != "noop"', {
'targets': [
{
'target_name': 'peerconnection_unittests_run',
'type': 'none',
'dependencies': [
'peerconnection_unittests',
],
'includes': [
'../build/isolate.gypi',
],
'sources': [
'peerconnection_unittests.isolate',
],
},
], # targets
}], # test_isolation_mode != "noop"
], # conditions
}

4
talk/app/webrtc/audiotrack.cc → webrtc/api/audiotrack.cc
View File

@ -1,6 +1,6 @@
/*
* libjingle
* Copyright 2004--2011 Google Inc.
* Copyright 2011 Google Inc.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions are met:
@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/audiotrack.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/base/checks.h"

12
talk/app/webrtc/audiotrack.h → webrtc/api/audiotrack.h
View File

@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_AUDIOTRACK_H_
#define TALK_APP_WEBRTC_AUDIOTRACK_H_
#ifndef WEBRTC_API_AUDIOTRACK_H_
#define WEBRTC_API_AUDIOTRACK_H_
#include <string>
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/mediastreamtrack.h"
#include "talk/app/webrtc/notifier.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/mediastreamtrack.h"
#include "webrtc/api/notifier.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread_checker.h"
@ -73,4 +73,4 @@ class AudioTrack : public MediaStreamTrack<AudioTrackInterface>,
} // namespace webrtc
#endif // TALK_APP_WEBRTC_AUDIOTRACK_H_
#endif // WEBRTC_API_AUDIOTRACK_H_

6
talk/app/webrtc/datachannel.cc → webrtc/api/datachannel.cc
View File

@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/datachannel.h"
#include "webrtc/api/datachannel.h"
#include <string>
#include "talk/app/webrtc/mediastreamprovider.h"
#include "talk/app/webrtc/sctputils.h"
#include "webrtc/api/mediastreamprovider.h"
#include "webrtc/api/sctputils.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/refcount.h"
#include "webrtc/media/sctp/sctpdataengine.h"

10
talk/app/webrtc/datachannel.h → webrtc/api/datachannel.h
View File

@ -25,16 +25,16 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_DATACHANNEL_H_
#define TALK_APP_WEBRTC_DATACHANNEL_H_
#ifndef WEBRTC_API_DATACHANNEL_H_
#define WEBRTC_API_DATACHANNEL_H_
#include <deque>
#include <set>
#include <string>
#include "talk/app/webrtc/datachannelinterface.h"
#include "talk/app/webrtc/proxy.h"
#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"
@ -296,4 +296,4 @@ END_PROXY()
} // namespace webrtc
#endif // TALK_APP_WEBRTC_DATACHANNEL_H_
#endif // WEBRTC_API_DATACHANNEL_H_

6
talk/app/webrtc/datachannel_unittest.cc → webrtc/api/datachannel_unittest.cc
View File

@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/datachannel.h"
#include "talk/app/webrtc/sctputils.h"
#include "talk/app/webrtc/test/fakedatachannelprovider.h"
#include "webrtc/api/datachannel.h"
#include "webrtc/api/sctputils.h"
#include "webrtc/api/test/fakedatachannelprovider.h"
#include "webrtc/base/gunit.h"
using webrtc::DataChannel;

6
talk/app/webrtc/datachannelinterface.h → webrtc/api/datachannelinterface.h
View File

@ -28,8 +28,8 @@
// This file contains interfaces for DataChannels
// http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcdatachannel
#ifndef TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_
#define TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_
#ifndef WEBRTC_API_DATACHANNELINTERFACE_H_
#define WEBRTC_API_DATACHANNELINTERFACE_H_
#include <string>
@ -156,4 +156,4 @@ class DataChannelInterface : public rtc::RefCountInterface {
} // namespace webrtc
#endif // TALK_APP_WEBRTC_DATACHANNELINTERFACE_H_
#endif // WEBRTC_API_DATACHANNELINTERFACE_H_

4
talk/app/webrtc/dtlsidentitystore.cc → webrtc/api/dtlsidentitystore.cc
View File

@ -25,11 +25,11 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "webrtc/api/dtlsidentitystore.h"
#include <utility>
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
#include "webrtc/api/webrtcsessiondescriptionfactory.h"
#include "webrtc/base/logging.h"
using webrtc::DtlsIdentityRequestObserver;

6
talk/app/webrtc/dtlsidentitystore.h → webrtc/api/dtlsidentitystore.h
View File

@ -25,8 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_DTLSIDENTITYSTORE_H_
#define TALK_APP_WEBRTC_DTLSIDENTITYSTORE_H_
#ifndef WEBRTC_API_DTLSIDENTITYSTORE_H_
#define WEBRTC_API_DTLSIDENTITYSTORE_H_
#include <queue>
#include <string>
@ -162,4 +162,4 @@ class DtlsIdentityStoreImpl : public DtlsIdentityStoreInterface,
} // namespace webrtc
#endif // TALK_APP_WEBRTC_DTLSIDENTITYSTORE_H_
#endif // WEBRTC_API_DTLSIDENTITYSTORE_H_

4
talk/app/webrtc/dtlsidentitystore_unittest.cc → webrtc/api/dtlsidentitystore_unittest.cc
View File

@ -25,9 +25,9 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "webrtc/api/dtlsidentitystore.h"
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
#include "webrtc/api/webrtcsessiondescriptionfactory.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/ssladapter.h"

2
talk/app/webrtc/dtmfsender.cc → webrtc/api/dtmfsender.cc
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/dtmfsender.h"
#include "webrtc/api/dtmfsender.h"
#include <ctype.h>

12
talk/app/webrtc/dtmfsender.h → webrtc/api/dtmfsender.h
View File

@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_DTMFSENDER_H_
#define TALK_APP_WEBRTC_DTMFSENDER_H_
#ifndef WEBRTC_API_DTMFSENDER_H_
#define WEBRTC_API_DTMFSENDER_H_
#include <string>
#include "talk/app/webrtc/dtmfsenderinterface.h"
#include "talk/app/webrtc/mediastreaminterface.h"
#include "talk/app/webrtc/proxy.h"
#include "webrtc/api/dtmfsenderinterface.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/base/common.h"
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/refcount.h"
@ -136,4 +136,4 @@ bool GetDtmfCode(char tone, int* code);
} // namespace webrtc
#endif // TALK_APP_WEBRTC_DTMFSENDER_H_
#endif // WEBRTC_API_DTMFSENDER_H_

4
talk/app/webrtc/dtmfsender_unittest.cc → webrtc/api/dtmfsender_unittest.cc
View File

@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/dtmfsender.h"
#include "webrtc/api/dtmfsender.h"
#include <set>
#include <string>
#include <vector>
#include "talk/app/webrtc/audiotrack.h"
#include "webrtc/api/audiotrack.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/timeutils.h"

8
talk/app/webrtc/dtmfsenderinterface.h → webrtc/api/dtmfsenderinterface.h
View File

@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
#define TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
#ifndef WEBRTC_API_DTMFSENDERINTERFACE_H_
#define WEBRTC_API_DTMFSENDERINTERFACE_H_
#include <string>
#include "talk/app/webrtc/mediastreaminterface.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/base/common.h"
#include "webrtc/base/refcount.h"
@ -102,4 +102,4 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
} // namespace webrtc
#endif // TALK_APP_WEBRTC_DTMFSENDERINTERFACE_H_
#endif // WEBRTC_API_DTMFSENDERINTERFACE_H_

8
talk/app/webrtc/fakemediacontroller.h → webrtc/api/fakemediacontroller.h
View File

@ -25,10 +25,10 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_
#define TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_
#ifndef WEBRTC_API_FAKEMEDIACONTROLLER_H_
#define WEBRTC_API_FAKEMEDIACONTROLLER_H_
#include "talk/app/webrtc/mediacontroller.h"
#include "webrtc/api/mediacontroller.h"
#include "webrtc/base/checks.h"
namespace cricket {
@ -52,4 +52,4 @@ class FakeMediaController : public webrtc::MediaControllerInterface {
webrtc::Call* call_;
};
} // namespace cricket
#endif // TALK_APP_WEBRTC_FAKEMEDIACONTROLLER_H_
#endif // WEBRTC_API_FAKEMEDIACONTROLLER_H_

2
talk/app/webrtc/fakemetricsobserver.cc → webrtc/api/fakemetricsobserver.cc
View File

@ -25,7 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/fakemetricsobserver.h"
#include "webrtc/api/fakemetricsobserver.h"
#include "webrtc/base/checks.h"
namespace webrtc {

8
talk/app/webrtc/fakemetricsobserver.h → webrtc/api/fakemetricsobserver.h
View File

@ -25,13 +25,13 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
#define TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
#ifndef WEBRTC_API_FAKEMETRICSOBSERVER_H_
#define WEBRTC_API_FAKEMETRICSOBSERVER_H_
#include <map>
#include <string>
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/thread_checker.h"
namespace webrtc {
@ -65,4 +65,4 @@ class FakeMetricsObserver : public MetricsObserverInterface {
} // namespace webrtc
#endif // TALK_APP_WEBRTC_FAKEMETRICSOBSERVER_H_
#endif // WEBRTC_API_FAKEMETRICSOBSERVER_H_

0
talk/app/webrtc/java/README → webrtc/api/java/README
View File

0
talk/app/webrtc/java/android/org/webrtc/Camera2Enumerator.java → webrtc/api/java/android/org/webrtc/Camera2Enumerator.java
View File

0
talk/app/webrtc/java/android/org/webrtc/CameraEnumerationAndroid.java → webrtc/api/java/android/org/webrtc/CameraEnumerationAndroid.java
View File

0
talk/app/webrtc/java/android/org/webrtc/CameraEnumerator.java → webrtc/api/java/android/org/webrtc/CameraEnumerator.java
View File

0
talk/app/webrtc/java/android/org/webrtc/EglBase.java → webrtc/api/java/android/org/webrtc/EglBase.java
View File

0
talk/app/webrtc/java/android/org/webrtc/EglBase10.java → webrtc/api/java/android/org/webrtc/EglBase10.java
View File

0
talk/app/webrtc/java/android/org/webrtc/EglBase14.java → webrtc/api/java/android/org/webrtc/EglBase14.java
View File

0
talk/app/webrtc/java/android/org/webrtc/GlRectDrawer.java → webrtc/api/java/android/org/webrtc/GlRectDrawer.java
View File

0
talk/app/webrtc/java/android/org/webrtc/GlShader.java → webrtc/api/java/android/org/webrtc/GlShader.java
View File

0
talk/app/webrtc/java/android/org/webrtc/GlTextureFrameBuffer.java → webrtc/api/java/android/org/webrtc/GlTextureFrameBuffer.java
View File

0
talk/app/webrtc/java/android/org/webrtc/GlUtil.java → webrtc/api/java/android/org/webrtc/GlUtil.java
View File

0
talk/app/webrtc/java/android/org/webrtc/NetworkMonitor.java → webrtc/api/java/android/org/webrtc/NetworkMonitor.java
View File

0
talk/app/webrtc/java/android/org/webrtc/NetworkMonitorAutoDetect.java → webrtc/api/java/android/org/webrtc/NetworkMonitorAutoDetect.java
View File

0
talk/app/webrtc/java/android/org/webrtc/OWNERS → webrtc/api/java/android/org/webrtc/OWNERS
View File

0
talk/app/webrtc/java/android/org/webrtc/RendererCommon.java → webrtc/api/java/android/org/webrtc/RendererCommon.java
View File

0
talk/app/webrtc/java/android/org/webrtc/SurfaceTextureHelper.java → webrtc/api/java/android/org/webrtc/SurfaceTextureHelper.java
View File

0
talk/app/webrtc/java/android/org/webrtc/SurfaceViewRenderer.java → webrtc/api/java/android/org/webrtc/SurfaceViewRenderer.java
View File

0
talk/app/webrtc/java/android/org/webrtc/ThreadUtils.java → webrtc/api/java/android/org/webrtc/ThreadUtils.java
View File

0
talk/app/webrtc/java/android/org/webrtc/VideoCapturerAndroid.java → webrtc/api/java/android/org/webrtc/VideoCapturerAndroid.java
View File

0
talk/app/webrtc/java/android/org/webrtc/VideoRendererGui.java → webrtc/api/java/android/org/webrtc/VideoRendererGui.java
View File

0
talk/app/webrtc/java/jni/OWNERS → webrtc/api/java/jni/OWNERS
View File

10
talk/app/webrtc/java/jni/androidmediacodeccommon.h → webrtc/api/java/jni/androidmediacodeccommon.h
View File

@ -26,15 +26,15 @@
*
*/
#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#ifndef WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#include <android/log.h>
#include <string>
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/jni_helpers.h"
#include "webrtc/base/thread.h"
#include "webrtc/api/java/jni/classreferenceholder.h"
#include "webrtc/api/java/jni/jni_helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/thread.h"
#include "webrtc/system_wrappers/include/tick_util.h"
@ -110,4 +110,4 @@ static inline bool CheckException(JNIEnv* jni) {
} // namespace webrtc_jni
#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_
#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIACODECCOMMON_H_

11
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc → webrtc/api/java/jni/androidmediadecoder_jni.cc
View File

@ -29,16 +29,17 @@
#include <algorithm>
#include <vector>
#include "talk/app/webrtc/java/jni/androidmediadecoder_jni.h"
// NOTICE: androidmediadecoder_jni.h must be included before
// androidmediacodeccommon.h to avoid build errors.
#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h"
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
#include "talk/app/webrtc/java/jni/surfacetexturehelper_jni.h"
#include "webrtc/api/java/jni/androidmediadecoder_jni.h"
#include "third_party/libyuv/include/libyuv/convert.h"
#include "third_party/libyuv/include/libyuv/convert_from.h"
#include "third_party/libyuv/include/libyuv/video_common.h"
#include "webrtc/api/java/jni/androidmediacodeccommon.h"
#include "webrtc/api/java/jni/classreferenceholder.h"
#include "webrtc/api/java/jni/native_handle_impl.h"
#include "webrtc/api/java/jni/surfacetexturehelper_jni.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"

8
talk/app/webrtc/java/jni/androidmediadecoder_jni.h → webrtc/api/java/jni/androidmediadecoder_jni.h
View File

@ -26,10 +26,10 @@
*
*/
#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_
#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_
#ifndef WEBRTC_API_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_
#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_
#include "talk/app/webrtc/java/jni/eglbase_jni.h"
#include "webrtc/api/java/jni/eglbase_jni.h"
#include "webrtc/media/webrtc/webrtcvideodecoderfactory.h"
namespace webrtc_jni {
@ -56,4 +56,4 @@ class MediaCodecVideoDecoderFactory
} // namespace webrtc_jni
#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_
#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIADECODER_JNI_H_

9
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc → webrtc/api/java/jni/androidmediaencoder_jni.cc
View File

@ -26,15 +26,16 @@
*
*/
#include "talk/app/webrtc/java/jni/androidmediaencoder_jni.h"
// NOTICE: androidmediaencoder_jni.h must be included before
// androidmediacodeccommon.h to avoid build errors.
#include "talk/app/webrtc/java/jni/androidmediacodeccommon.h"
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/native_handle_impl.h"
#include "webrtc/api/java/jni/androidmediaencoder_jni.h"
#include "third_party/libyuv/include/libyuv/convert.h"
#include "third_party/libyuv/include/libyuv/convert_from.h"
#include "third_party/libyuv/include/libyuv/video_common.h"
#include "webrtc/api/java/jni/androidmediacodeccommon.h"
#include "webrtc/api/java/jni/classreferenceholder.h"
#include "webrtc/api/java/jni/native_handle_impl.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"

8
talk/app/webrtc/java/jni/androidmediaencoder_jni.h → webrtc/api/java/jni/androidmediaencoder_jni.h
View File

@ -26,12 +26,12 @@
*
*/
#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_
#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_
#ifndef WEBRTC_API_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_
#define WEBRTC_API_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_
#include <vector>
#include "talk/app/webrtc/java/jni/eglbase_jni.h"
#include "webrtc/api/java/jni/eglbase_jni.h"
#include "webrtc/media/webrtc/webrtcvideoencoderfactory.h"
namespace webrtc_jni {
@ -60,4 +60,4 @@ class MediaCodecVideoEncoderFactory
} // namespace webrtc_jni
#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_
#endif // WEBRTC_API_JAVA_JNI_ANDROIDMEDIAENCODER_JNI_H_

6
talk/app/webrtc/java/jni/androidnetworkmonitor_jni.cc → webrtc/api/java/jni/androidnetworkmonitor_jni.cc
View File

@ -25,12 +25,12 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#include "talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h"
#include "webrtc/api/java/jni/androidnetworkmonitor_jni.h"
#include <dlfcn.h>
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/jni_helpers.h"
#include "webrtc/api/java/jni/classreferenceholder.h"
#include "webrtc/api/java/jni/jni_helpers.h"
#include "webrtc/base/bind.h"
#include "webrtc/base/common.h"
#include "webrtc/base/ipaddress.h"

8
talk/app/webrtc/java/jni/androidnetworkmonitor_jni.h → webrtc/api/java/jni/androidnetworkmonitor_jni.h
View File

@ -25,14 +25,14 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef TALK_APP_WEBRTC_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_
#define TALK_APP_WEBRTC_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_
#ifndef WEBRTC_API_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_
#define WEBRTC_API_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_
#include "webrtc/base/networkmonitor.h"
#include <map>
#include "talk/app/webrtc/java/jni/jni_helpers.h"
#include "webrtc/api/java/jni/jni_helpers.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/thread_checker.h"
@ -105,4 +105,4 @@ class AndroidNetworkMonitorFactory : public rtc::NetworkMonitorFactory {
} // namespace webrtc_jni
#endif // TALK_APP_WEBRTC_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_
#endif // WEBRTC_API_JAVA_JNI_ANDROIDNETWORKMONITOR_JNI_H_

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