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webrtc_m130/webrtc/api/peerconnectionfactoryproxy.h

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Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
/*
* libjingle
Update libjingle license statements at top of talk files for consistency BUG=2133 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
* Copyright 2014 Google Inc.
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
*
* 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.
*/
Update libjingle license statements at top of talk files for consistency BUG=2133 R=juberti@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39559004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-20 21:36:13 +00:00
Move talk/app/webrtc to webrtc/api 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}
2016-02-10 10:53:12 +01:00
#ifndef WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
#define WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
#include <string>
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector We can now use std::move instead! This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them. Review URL: https://codereview.webrtc.org/1460043002 Cr-Commit-Position: refs/heads/master@{#11064}
2015-12-17 03:04:15 -08:00
#include <utility>
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
Move talk/app/webrtc to webrtc/api 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}
2016-02-10 10:53:12 +01:00
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/proxy.h"
DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface). DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands: https://codereview.webrtc.org/1189583002 The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore. Where a service was previously passed around, a store is now passed around. Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur. For more information about the steps being taken to land this without breaking Chromium, see referenced bug. BUG=webrtc:4899 R=magjed@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1176383004 . Cr-Commit-Position: refs/heads/master@{#9696}
2015-08-11 10:33:13 +02:00
#include "webrtc/base/bind.h"
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
namespace webrtc {
BEGIN_PROXY_MAP(PeerConnectionFactory)
PROXY_METHOD1(void, SetOptions, const Options&)
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector We can now use std::move instead! This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them. Review URL: https://codereview.webrtc.org/1460043002 Cr-Commit-Position: refs/heads/master@{#11064}
2015-12-17 03:04:15 -08:00
// Can't use PROXY_METHOD5 because scoped_ptr must be moved.
DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface). DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands: https://codereview.webrtc.org/1189583002 The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore. Where a service was previously passed around, a store is now passed around. Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur. For more information about the steps being taken to land this without breaking Chromium, see referenced bug. BUG=webrtc:4899 R=magjed@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1176383004 . Cr-Commit-Position: refs/heads/master@{#9696}
2015-08-11 10:33:13 +02:00
// TODO(tommi,hbos): Use of templates to support scoped_ptr?
Adding CreatePeerConnection method that uses new PC Initialize method. This will let us transition to the new Initialize method in Chromium, and then get rid of the old one. Review URL: https://codereview.webrtc.org/1462253002 Cr-Commit-Position: refs/heads/master@{#10860}
2015-12-01 15:01:24 -08:00
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& a1,
const MediaConstraintsInterface* a2,
rtc::scoped_ptr<cricket::PortAllocator> a3,
rtc::scoped_ptr<DtlsIdentityStoreInterface> a4,
PeerConnectionObserver* a5) override {
return owner_thread_->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
Removing webrtc::PortAllocatorFactoryInterface. ICE servers are now passed directly into PortAllocator, making PortAllocatorFactoryInterface redundant. This CL also moves SetNetworkIgnoreMask to PortAllocator. R=phoglund@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1520963002 . Cr-Commit-Position: refs/heads/master@{#11139}
2015-12-29 14:14:52 -08:00
rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot, this,
Adding CreatePeerConnection method that uses new PC Initialize method. This will let us transition to the new Initialize method in Chromium, and then get rid of the old one. Review URL: https://codereview.webrtc.org/1462253002 Cr-Commit-Position: refs/heads/master@{#10860}
2015-12-01 15:01:24 -08:00
a1, a2, a3.release(), a4.release(), a5));
}
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream, const std::string&)
PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
CreateAudioSource, const MediaConstraintsInterface*)
PROXY_METHOD2(rtc::scoped_refptr<VideoSourceInterface>,
CreateVideoSource, cricket::VideoCapturer*,
const MediaConstraintsInterface*)
PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
CreateVideoTrack, const std::string&, VideoSourceInterface*)
PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
CreateAudioTrack, const std::string&, AudioSourceInterface*)
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. The revert of the original CL was commit 36d4c545007129446e551c45c17b25377dce89a4. Original review: https://codereview.webrtc.org/1413483003/ The original CL changes a function on audio_processing.h that is used by Chrome, this CL adds back the old function. TBR=glaznev@webrtc.org, henrik.lundin@webrtc.org, solenberg@google.com, henrikg@webrtc.org, perkj@webrtc.org BUG=webrtc:4741 Committed: https://crrev.com/f4f5cb09277d5ef6aeac8341e5f54a055867803a Cr-Commit-Position: refs/heads/master@{#11093} Review URL: https://codereview.webrtc.org/1540103002 Cr-Commit-Position: refs/heads/master@{#11267}
2016-01-15 03:06:36 -08:00
PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
Added StopAecDump function to PeerConnectionFactory. The function to stop recording an AEC dump was missing from the PeerConnectionFactory interface (only a start function was provided). This CL adds the missing stop function. BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1415733005 Cr-Commit-Position: refs/heads/master@{#10372}
2015-10-22 03:25:41 -07:00
PROXY_METHOD0(void, StopAecDump)
Added functions on libjingle API to start and stop the recording of an RtcEventLog. BUG=webrtc:4741 Review URL: https://codereview.webrtc.org/1374253002 Cr-Commit-Position: refs/heads/master@{#10297}
2015-10-16 02:22:18 -07:00
PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
PROXY_METHOD0(void, StopRtcEventLog)
DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface). DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands: https://codereview.webrtc.org/1189583002 The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore. Where a service was previously passed around, a store is now passed around. Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur. For more information about the steps being taken to land this without breaking Chromium, see referenced bug. BUG=webrtc:4899 R=magjed@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1176383004 . Cr-Commit-Position: refs/heads/master@{#9696}
2015-08-11 10:33:13 +02:00
private:
Removing webrtc::PortAllocatorFactoryInterface. ICE servers are now passed directly into PortAllocator, making PortAllocatorFactoryInterface redundant. This CL also moves SetNetworkIgnoreMask to PortAllocator. R=phoglund@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org Review URL: https://codereview.webrtc.org/1520963002 . Cr-Commit-Position: refs/heads/master@{#11139}
2015-12-29 14:14:52 -08:00
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_ot(
Adding CreatePeerConnection method that uses new PC Initialize method. This will let us transition to the new Initialize method in Chromium, and then get rid of the old one. Review URL: https://codereview.webrtc.org/1462253002 Cr-Commit-Position: refs/heads/master@{#10860}
2015-12-01 15:01:24 -08:00
const PeerConnectionInterface::RTCConfiguration& a1,
const MediaConstraintsInterface* a2,
cricket::PortAllocator* a3,
DtlsIdentityStoreInterface* a4,
PeerConnectionObserver* a5) {
rtc::scoped_ptr<cricket::PortAllocator> ptr_a3(a3);
rtc::scoped_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector We can now use std::move instead! This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them. Review URL: https://codereview.webrtc.org/1460043002 Cr-Commit-Position: refs/heads/master@{#11064}
2015-12-17 03:04:15 -08:00
return c_->CreatePeerConnection(a1, a2, std::move(ptr_a3),
std::move(ptr_a4), a5);
Adding CreatePeerConnection method that uses new PC Initialize method. This will let us transition to the new Initialize method in Chromium, and then get rid of the old one. Review URL: https://codereview.webrtc.org/1462253002 Cr-Commit-Position: refs/heads/master@{#10860}
2015-12-01 15:01:24 -08:00
}
Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. In order to do that, the signaling thread is also changed to wrap the current thread unless an external signaling thread thread is specified in the call to CreatePeerConnectionFactory. This cleans up the PeerConnectionFactory and makes sure a user of the API will always access the factory on the signaling thread. Note that both Chrome and the Android implementation use an external signaling thread. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35429004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8039 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-12 08:30:16 +00:00
END_PROXY()
} // namespace webrtc
Move talk/app/webrtc to webrtc/api 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}
2016-02-10 10:53:12 +01:00
#endif // WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
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