webrtc_m130/webrtc/api/peerconnectionfactory.h
hbos d7973ccdb5 Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ )
Reason for revert:
There are more CreatePeerConnection calls than I anticipated/had found in Chromium, like remoting/protocol/webrtc_transport.cc. Reverting due to broken Chromium FYI bots.

Original issue's description:
> Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface.
>
> The store was used in WebRtcSessionDescriptionFactory to generate certificates,
> now a generator is used instead (new API). PeerConnection[Factory][Interface],
> and WebRtcSession are updated to pass generators all the way down to the
> WebRtcSessionDescriptionFactory instead of stores.
>
> The webrtc implementation of a generator, RTCCertificateGenerator, is used as
> the default generator (peerconnectionfactory.cc:189) instead of the webrtc
> implementation of a store, DtlsIdentityStoreImpl.
>   The generator is fully parameterized and does not generate RSA-1024 unless you
> ask for it (which makes sense not to do beforehand since ECDSA is now default).
> The store was not fully parameterized (known filed bug).
>
> The "top" layer, PeerConnectionFactoryInterface::CreatePeerConnection, is
> updated to take a generator instead of a store. But as to not break Chromium,
> the old function signature taking a store is kept. It is implemented to invoke
> the generator version by wrapping the store in an
> RTCCertificateGeneratorStoreWrapper. As soon as Chromium is updated to use the
> new function signature we can remove the old CreatePeerConnection.
>   Due to having multiple CreatePeerConnection signatures, some calling places
> are updated to resolve the ambiguity introduced.
>
> BUG=webrtc:5707, webrtc:5708
> R=phoglund@webrtc.org, tommi@webrtc.org
> TBR=tkchin@webrc.org
>
> Committed: 400781a209

TBR=tkchin@webrtc.org,tommi@webrtc.org,phoglund@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5707, webrtc:5708

Review-Url: https://codereview.webrtc.org/2020633002
Cr-Commit-Position: refs/heads/master@{#12948}
2016-05-27 13:08:58 +00:00

138 lines
5.3 KiB
C++

/*
* Copyright 2011 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_PEERCONNECTIONFACTORY_H_
#define WEBRTC_API_PEERCONNECTIONFACTORY_H_
#include <memory>
#include <string>
#include "webrtc/api/dtlsidentitystore.h"
#include "webrtc/api/mediacontroller.h"
#include "webrtc/api/mediastreaminterface.h"
#include "webrtc/api/peerconnectioninterface.h"
#include "webrtc/base/scoped_ref_ptr.h"
#include "webrtc/base/thread.h"
#include "webrtc/pc/channelmanager.h"
namespace rtc {
class BasicNetworkManager;
class BasicPacketSocketFactory;
}
namespace webrtc {
typedef rtc::RefCountedObject<DtlsIdentityStoreImpl>
RefCountedDtlsIdentityStore;
class PeerConnectionFactory : public PeerConnectionFactoryInterface {
public:
void SetOptions(const Options& options) override {
options_ = options;
}
// Deprecated, use version without constraints.
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
const MediaConstraintsInterface* constraints,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) override;
virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
PeerConnectionObserver* observer) override;
bool Initialize();
rtc::scoped_refptr<MediaStreamInterface>
CreateLocalMediaStream(const std::string& label) override;
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) override;
// Deprecated, use version without constraints.
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) override;
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer) override;
// This version supports filtering on width, height and frame rate.
// For the "constraints=null" case, use the version without constraints.
// TODO(hta): Design a version without MediaConstraintsInterface.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
cricket::VideoCapturer* capturer,
const MediaConstraintsInterface* constraints) override;
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& id,
VideoTrackSourceInterface* video_source) override;
rtc::scoped_refptr<AudioTrackInterface>
CreateAudioTrack(const std::string& id,
AudioSourceInterface* audio_source) override;
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
void StopAecDump() override;
bool StartRtcEventLog(rtc::PlatformFile file) override {
return StartRtcEventLog(file, -1);
}
bool StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) override;
void StopRtcEventLog() override;
virtual webrtc::MediaControllerInterface* CreateMediaController(
const cricket::MediaConfig& config) const;
virtual rtc::Thread* signaling_thread();
virtual rtc::Thread* worker_thread();
virtual rtc::Thread* network_thread();
const Options& options() const { return options_; }
protected:
PeerConnectionFactory();
PeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
AudioDeviceModule* default_adm,
cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
virtual ~PeerConnectionFactory();
private:
cricket::MediaEngineInterface* CreateMediaEngine_w();
bool owns_ptrs_;
bool wraps_current_thread_;
rtc::Thread* network_thread_;
rtc::Thread* worker_thread_;
rtc::Thread* signaling_thread_;
Options options_;
// External Audio device used for audio playback.
rtc::scoped_refptr<AudioDeviceModule> default_adm_;
std::unique_ptr<cricket::ChannelManager> channel_manager_;
// External Video encoder factory. This can be NULL if the client has not
// injected any. In that case, video engine will use the internal SW encoder.
std::unique_ptr<cricket::WebRtcVideoEncoderFactory> video_encoder_factory_;
// External Video decoder factory. This can be NULL if the client has not
// injected any. In that case, video engine will use the internal SW decoder.
std::unique_ptr<cricket::WebRtcVideoDecoderFactory> video_decoder_factory_;
std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_;
std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
rtc::scoped_refptr<RefCountedDtlsIdentityStore> dtls_identity_store_;
};
} // namespace webrtc
#endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_