Swiftgram/submodules/TgVoipWebrtcCustom/Sources/tg_peer_connection_factory.h
2020-06-02 19:49:54 +04:00

262 lines
10 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 TG_PC_PEER_CONNECTION_FACTORY_H_
#define TG_PC_PEER_CONNECTION_FACTORY_H_
#include <memory>
#include <string>
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
#include "api/scoped_refptr.h"
#include "api/transport/media/media_transport_interface.h"
#include "media/sctp/sctp_transport_internal.h"
#include "pc/channel_manager.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/thread.h"
#include "pc/peer_connection_factory.h"
namespace rtc {
class BasicNetworkManager;
class BasicPacketSocketFactory;
} // namespace rtc
namespace webrtc {
class RtcEventLog;
class TgPeerConnection;
class TgPeerConnectionInterface;
class RTC_EXPORT TgPeerConnectionFactoryInterface
: public rtc::RefCountInterface {
public:
// Set the options to be used for subsequently created PeerConnections.
virtual void SetOptions(const PeerConnectionFactoryInterface::Options& options) = 0;
// The preferred way to create a new peer connection. Simply provide the
// configuration and a PeerConnectionDependencies structure.
// TODO(benwright): Make pure virtual once downstream mock PC factory classes
// are updated.
virtual rtc::scoped_refptr<TgPeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
// Deprecated; |allocator| and |cert_generator| may be null, in which case
// default implementations will be used.
//
// |observer| must not be null.
//
// Note that this method does not take ownership of |observer|; it's the
// responsibility of the caller to delete it. It can be safely deleted after
// Close has been called on the returned PeerConnection, which ensures no
// more observer callbacks will be invoked.
virtual rtc::scoped_refptr<TgPeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer);
// Returns the capabilities of an RTP sender of type |kind|.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpSenderCapabilities(
cricket::MediaType kind) const;
// Returns the capabilities of an RTP receiver of type |kind|.
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
// TODO(orphis): Make pure virtual when all subclasses implement it.
virtual RtpCapabilities GetRtpReceiverCapabilities(
cricket::MediaType kind) const;
virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id) = 0;
// Creates an AudioSourceInterface.
// |options| decides audio processing settings.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) = 0;
// Creates a new local VideoTrack. The same |source| can be used in several
// tracks.
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label,
VideoTrackSourceInterface* source) = 0;
// Creates an new AudioTrack. At the moment |source| can be null.
virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& label,
AudioSourceInterface* source) = 0;
// Starts AEC dump using existing file. Takes ownership of |file| and passes
// it on to VoiceEngine (via other objects) immediately, which will take
// the ownerhip. If the operation fails, the file will be closed.
// A maximum file size in bytes can be specified. When the file size limit is
// reached, logging is stopped automatically. If max_size_bytes is set to a
// value <= 0, no limit will be used, and logging will continue until the
// StopAecDump function is called.
// TODO(webrtc:6463): Delete default implementation when downstream mocks
// classes are updated.
virtual bool StartAecDump(FILE* file, int64_t max_size_bytes) {
return false;
}
// Stops logging the AEC dump.
virtual void StopAecDump() = 0;
protected:
// Dtor and ctor protected as objects shouldn't be created or deleted via
// this interface.
TgPeerConnectionFactoryInterface() {}
~TgPeerConnectionFactoryInterface() override = default;
};
class TgPeerConnectionFactory: public TgPeerConnectionFactoryInterface {
public:
void SetOptions(const PeerConnectionFactoryInterface::Options& options);
rtc::scoped_refptr<TgPeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
std::unique_ptr<cricket::PortAllocator> allocator,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
PeerConnectionObserver* observer);
rtc::scoped_refptr<TgPeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& configuration,
PeerConnectionDependencies dependencies);
bool Initialize();
RtpCapabilities GetRtpSenderCapabilities(
cricket::MediaType kind) const;
RtpCapabilities GetRtpReceiverCapabilities(
cricket::MediaType kind) const;
rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
const std::string& stream_id);
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options);
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& id,
VideoTrackSourceInterface* video_source);
rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& id,
AudioSourceInterface* audio_source);
bool StartAecDump(FILE* file, int64_t max_size_bytes);
void StopAecDump();
virtual std::unique_ptr<cricket::SctpTransportInternalFactory>
CreateSctpTransportInternalFactory();
virtual cricket::ChannelManager* channel_manager();
rtc::Thread* signaling_thread() {
// This method can be called on a different thread when the factory is
// created in CreatePeerConnectionFactory().
return signaling_thread_;
}
rtc::Thread* worker_thread() { return worker_thread_; }
rtc::Thread* network_thread() { return network_thread_; }
const PeerConnectionFactoryInterface::Options& options() const { return options_; }
MediaTransportFactory* media_transport_factory() {
return media_transport_factory_.get();
}
protected:
// This structure allows simple management of all new dependencies being added
// to the PeerConnectionFactory.
explicit TgPeerConnectionFactory(
PeerConnectionFactoryDependencies dependencies);
// Hook to let testing framework insert actions between
// "new RTCPeerConnection" and "pc.Initialize"
virtual void ActionsBeforeInitializeForTesting(PeerConnectionInterface*) {}
virtual ~TgPeerConnectionFactory();
private:
bool IsTrialEnabled(absl::string_view key) const;
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_;
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
PeerConnectionFactoryInterface::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_;
std::unique_ptr<NetworkStatePredictorFactoryInterface>
network_state_predictor_factory_;
std::unique_ptr<NetworkControllerFactoryInterface>
injected_network_controller_factory_;
std::unique_ptr<MediaTransportFactory> media_transport_factory_;
std::unique_ptr<NetEqFactory> neteq_factory_;
const std::unique_ptr<WebRtcKeyValueConfig> trials_;
};
BEGIN_SIGNALING_PROXY_MAP(TgPeerConnectionFactory)
PROXY_SIGNALING_THREAD_DESTRUCTOR()
PROXY_METHOD1(void, SetOptions, const PeerConnectionFactory::Options&)
PROXY_METHOD4(rtc::scoped_refptr<TgPeerConnectionInterface>,
CreatePeerConnection,
const PeerConnectionInterface::RTCConfiguration&,
std::unique_ptr<cricket::PortAllocator>,
std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
PeerConnectionObserver*)
PROXY_METHOD2(rtc::scoped_refptr<TgPeerConnectionInterface>,
CreatePeerConnection,
const PeerConnectionInterface::RTCConfiguration&,
PeerConnectionDependencies)
PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
GetRtpSenderCapabilities,
cricket::MediaType)
PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
GetRtpReceiverCapabilities,
cricket::MediaType)
PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
CreateLocalMediaStream,
const std::string&)
PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
CreateAudioSource,
const cricket::AudioOptions&)
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, FILE*, int64_t)
PROXY_METHOD0(void, StopAecDump)
END_PROXY_MAP()
} // namespace webrtc
#endif // PC_PEER_CONNECTION_FACTORY_H_