webrtc_m130/webrtc/api/peerconnectionfactoryproxy.h
ivoc 9e03c3b372 Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
Reason for revert:
Reverting all CLs related to moving the eventlog, as they break Chromium tests.

Original issue's description:
> Move RtcEventLog object from inside VoiceEngine to Call.
>
> In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced.
> The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface.
>
> BUG=webrtc:4741,webrtc:5603,chromium:609749
> R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org
>
> Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016
> Cr-Commit-Position: refs/heads/master@{#13321}

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4741,webrtc:5603,chromium:609749

Review-Url: https://codereview.webrtc.org/2111813002
Cr-Commit-Position: refs/heads/master@{#13340}
2016-06-30 07:59:49 +00:00

105 lines
4.4 KiB
C++

/*
* Copyright 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
#define WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_
#include <memory>
#include <string>
#include <utility>
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/api/proxy.h"
#include "webrtc/base/bind.h"
namespace webrtc {
BEGIN_SIGNALING_PROXY_MAP(PeerConnectionFactory)
PROXY_METHOD1(void, SetOptions, const Options&)
// Can't use PROXY_METHOD5 because unique_ptr must be moved.
// TODO(tommi,hbos): Use of templates to support unique_ptr?
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& a1,
const MediaConstraintsInterface* a2,
std::unique_ptr<cricket::PortAllocator> a3,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> a4,
PeerConnectionObserver* a5) override {
return signaling_thread_
->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot,
this, a1, a2, a3.release(), a4.release(), a5));
}
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& a1,
std::unique_ptr<cricket::PortAllocator> a3,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> a4,
PeerConnectionObserver* a5) override {
return signaling_thread_
->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot,
this, a1, a3.release(), a4.release(), a5));
}
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream, const std::string&)
PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
CreateAudioSource, const MediaConstraintsInterface*)
PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
CreateAudioSource,
const cricket::AudioOptions&)
PROXY_METHOD2(rtc::scoped_refptr<VideoTrackSourceInterface>,
CreateVideoSource,
cricket::VideoCapturer*,
const MediaConstraintsInterface*)
PROXY_METHOD1(rtc::scoped_refptr<VideoTrackSourceInterface>,
CreateVideoSource,
cricket::VideoCapturer*)
PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
CreateVideoTrack,
const std::string&,
VideoTrackSourceInterface*)
PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
CreateAudioTrack, const std::string&, AudioSourceInterface*)
PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
PROXY_METHOD0(void, StopAecDump)
PROXY_METHOD1(bool, StartRtcEventLog, rtc::PlatformFile)
PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
PROXY_METHOD0(void, StopRtcEventLog)
private:
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_ot(
const PeerConnectionInterface::RTCConfiguration& a1,
const MediaConstraintsInterface* a2,
cricket::PortAllocator* a3,
rtc::RTCCertificateGeneratorInterface* a4,
PeerConnectionObserver* a5) {
std::unique_ptr<cricket::PortAllocator> ptr_a3(a3);
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> ptr_a4(a4);
return c_->CreatePeerConnection(a1, a2, std::move(ptr_a3),
std::move(ptr_a4), a5);
}
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_ot(
const PeerConnectionInterface::RTCConfiguration& a1,
cricket::PortAllocator* a3,
rtc::RTCCertificateGeneratorInterface* a4,
PeerConnectionObserver* a5) {
std::unique_ptr<cricket::PortAllocator> ptr_a3(a3);
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> ptr_a4(a4);
return c_->CreatePeerConnection(a1, std::move(ptr_a3), std::move(ptr_a4),
a5);
}
END_SIGNALING_PROXY()
} // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONFACTORYPROXY_H_