webrtc_m130/pc/peerconnectionfactory.h
Zhi Huang e830e683c4 Use new TransportController implementation in PeerConnection.
The TransportController will be replaced by the JsepTransportController
and JsepTransport will be replace be JsepTransport2.

The JsepTransportController will take the entire SessionDescription
and handle the RtcpMux, Sdes and BUNDLE internally.

The ownership model is also changed. The P2P layer transports are not
ref-counted and will be owned by the JsepTransport2.

In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport
or SrtpTransport and it implements the public and internal interface
by calling the transport underneath.

Bug: webrtc:8587
Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df
Reviewed-on: https://webrtc-review.googlesource.com/59586
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22693}
2018-03-30 18:41:19 +00:00

141 lines
5.4 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 PC_PEERCONNECTIONFACTORY_H_
#define PC_PEERCONNECTIONFACTORY_H_
#include <memory>
#include <string>
#include "api/mediastreaminterface.h"
#include "api/peerconnectioninterface.h"
#include "media/sctp/sctptransportinternal.h"
#include "pc/channelmanager.h"
#include "rtc_base/rtccertificategenerator.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
namespace rtc {
class BasicNetworkManager;
class BasicPacketSocketFactory;
}
namespace webrtc {
class RtcEventLog;
class PeerConnectionFactory : public PeerConnectionFactoryInterface {
public:
// Use the overloads of CreateVideoSource that take raw VideoCapturer
// pointers from PeerConnectionFactoryInterface.
// TODO(deadbeef): Remove this using statement once those overloads are
// removed.
using PeerConnectionFactoryInterface::CreateVideoSource;
void SetOptions(const Options& options) override;
// 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<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) override;
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer) override;
bool Initialize();
rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) override;
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) override;
// Deprecated, use version without constraints.
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const MediaConstraintsInterface* constraints) override;
rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<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(
std::unique_ptr<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;
virtual std::unique_ptr<cricket::SctpTransportInternalFactory>
CreateSctpTransportInternalFactory();
virtual cricket::ChannelManager* channel_manager();
virtual rtc::Thread* signaling_thread();
virtual rtc::Thread* worker_thread();
virtual rtc::Thread* network_thread();
const Options& options() const { return options_; }
protected:
PeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
PeerConnectionFactory(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory,
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory);
virtual ~PeerConnectionFactory();
private:
std::unique_ptr<RtcEventLog> CreateRtcEventLog_w();
std::unique_ptr<Call> CreateCall_w(RtcEventLog* event_log);
bool wraps_current_thread_;
rtc::Thread* network_thread_;
rtc::Thread* worker_thread_;
rtc::Thread* signaling_thread_;
std::unique_ptr<rtc::Thread> owned_network_thread_;
std::unique_ptr<rtc::Thread> owned_worker_thread_;
Options options_;
std::unique_ptr<cricket::ChannelManager> channel_manager_;
std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_;
std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
std::unique_ptr<cricket::MediaEngineInterface> media_engine_;
std::unique_ptr<webrtc::CallFactoryInterface> call_factory_;
std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory_;
std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
};
} // namespace webrtc
#endif // PC_PEERCONNECTIONFACTORY_H_