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/*
* Copyright 2017 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 API_ORTC_ORTCFACTORYINTERFACE_H_
#define API_ORTC_ORTCFACTORYINTERFACE_H_
#include <memory>
#include <string>
#include <utility> // For std::move.
#include "api/mediaconstraintsinterface.h"
#include "api/mediastreaminterface.h"
#include "api/mediatypes.h"
#include "api/ortc/ortcrtpreceiverinterface.h"
#include "api/ortc/ortcrtpsenderinterface.h"
#include "api/ortc/packettransportinterface.h"
#include "api/ortc/rtptransportcontrollerinterface.h"
#include "api/ortc/rtptransportinterface.h"
#include "api/ortc/srtptransportinterface.h"
#include "api/ortc/udptransportinterface.h"
#include "api/rtcerror.h"
#include "api/rtpparameters.h"
#include "rtc_base/network.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
namespace webrtc {
// TODO(deadbeef): This should be part of /api/, but currently it's not and
// including its header violates checkdeps rules.
class AudioDeviceModule;
// WARNING: This is experimental/under development, so use at your own risk; no
// guarantee about API stability is guaranteed here yet.
//
// This class is the ORTC analog of PeerConnectionFactory. It acts as a factory
// for ORTC objects that can be connected to each other.
//
// Some of these objects may not be represented by the ORTC specification, but
// follow the same general principles.
//
// If one of the factory methods takes another object as an argument, it MUST
// have been created by the same OrtcFactory.
//
// On object lifetimes: objects should be destroyed in this order:
// 1. Objects created by the factory.
// 2. The factory itself.
// 3. Objects passed into OrtcFactoryInterface::Create.
class OrtcFactoryInterface {
public:
// |network_thread| is the thread on which packets are sent and received.
// If null, a new rtc::Thread with a default socket server is created.
//
// |signaling_thread| is used for callbacks to the consumer of the API. If
// null, the current thread will be used, which assumes that the API consumer
// is running a message loop on this thread (either using an existing
// rtc::Thread, or by calling rtc::Thread::Current()->ProcessMessages).
//
// |network_manager| is used to determine which network interfaces are
// available. This is used for ICE, for example. If null, a default
// implementation will be used. Only accessed on |network_thread|.
//
// |socket_factory| is used (on the network thread) for creating sockets. If
// it's null, a default implementation will be used, which assumes
// |network_thread| is a normal rtc::Thread.
//
// |adm| is optional, and allows a different audio device implementation to
// be injected; otherwise a platform-specific module will be used that will
// use the default audio input.
//
// |audio_encoder_factory| and |audio_decoder_factory| are used to
// instantiate audio codecs; they determine what codecs are supported.
//
// Note that the OrtcFactoryInterface does not take ownership of any of the
// objects passed in by raw pointer, and as previously stated, these objects
// can't be destroyed before the factory is.
static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
rtc::Thread* network_thread,
rtc::Thread* signaling_thread,
rtc::NetworkManager* network_manager,
rtc::PacketSocketFactory* socket_factory,
AudioDeviceModule* adm,
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory);
// Constructor for convenience which uses default implementations where
// possible (though does still require that the current thread runs a message
// loop; see above).
static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create(
rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
return Create(nullptr, nullptr, nullptr, nullptr, nullptr,
audio_encoder_factory, audio_decoder_factory);
}
virtual ~OrtcFactoryInterface() {}
// Creates an RTP transport controller, which is used in calls to
// CreateRtpTransport methods. If your application has some notion of a
// "call", you should create one transport controller per call.
//
// However, if you only are using one RtpTransport object, this doesn't need
// to be called explicitly; CreateRtpTransport will create one automatically
// if |rtp_transport_controller| is null. See below.
//
// TODO(deadbeef): Add MediaConfig and RtcEventLog arguments?
virtual RTCErrorOr<std::unique_ptr<RtpTransportControllerInterface>>
CreateRtpTransportController() = 0;
// Creates an RTP transport using the provided packet transports and
// transport controller.
//
// |rtp| will be used for sending RTP packets, and |rtcp| for RTCP packets.
//
// |rtp| can't be null. |rtcp| must be non-null if and only if
// |rtp_parameters.rtcp.mux| is false, indicating that RTCP muxing isn't used.
// Note that if RTCP muxing isn't enabled initially, it can still enabled
// later through SetParameters.
//
// If |transport_controller| is null, one will automatically be created, and
// its lifetime managed by the returned RtpTransport. This should only be
// done if a single RtpTransport is being used to communicate with the remote
// endpoint.
virtual RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
const RtpTransportParameters& rtp_parameters,
PacketTransportInterface* rtp,
PacketTransportInterface* rtcp,
RtpTransportControllerInterface* transport_controller) = 0;
// Creates an SrtpTransport which is an RTP transport that uses SRTP.
virtual RTCErrorOr<std::unique_ptr<SrtpTransportInterface>>
CreateSrtpTransport(
const RtpTransportParameters& rtp_parameters,
PacketTransportInterface* rtp,
PacketTransportInterface* rtcp,
RtpTransportControllerInterface* transport_controller) = 0;
// Returns the capabilities of an RTP sender of type |kind|. These
// capabilities can be used to determine what RtpParameters to use to create
// an RtpSender.
//
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
virtual RtpCapabilities GetRtpSenderCapabilities(
cricket::MediaType kind) const = 0;
// Creates an RTP sender with |track|. Will not start sending until Send is
// called. This is provided as a convenience; it's equivalent to calling
// CreateRtpSender with a kind (see below), followed by SetTrack.
//
// |track| and |transport| must not be null.
virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
rtc::scoped_refptr<MediaStreamTrackInterface> track,
RtpTransportInterface* transport) = 0;
// Overload of CreateRtpSender allows creating the sender without a track.
//
// |kind| must be MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
cricket::MediaType kind,
RtpTransportInterface* transport) = 0;
// Returns the capabilities of an RTP receiver of type |kind|. These
// capabilities can be used to determine what RtpParameters to use to create
// an RtpReceiver.
//
// If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
virtual RtpCapabilities GetRtpReceiverCapabilities(
cricket::MediaType kind) const = 0;
// Creates an RTP receiver of type |kind|. Will not start receiving media
// until Receive is called.
//
// |kind| must be MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
//
// |transport| must not be null.
virtual RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
CreateRtpReceiver(cricket::MediaType kind,
RtpTransportInterface* transport) = 0;
// Create a UDP transport with IP address family |family|, using a port
// within the specified range.
//
// |family| must be AF_INET or AF_INET6.
//
// |min_port|/|max_port| values of 0 indicate no range restriction.
//
// Returns an error if the transport wasn't successfully created.
virtual RTCErrorOr<std::unique_ptr<UdpTransportInterface>>
CreateUdpTransport(int family, uint16_t min_port, uint16_t max_port) = 0;
// Method for convenience that has no port range restrictions.
RTCErrorOr<std::unique_ptr<UdpTransportInterface>> CreateUdpTransport(
int family) {
return CreateUdpTransport(family, 0, 0);
}
// NOTE: The methods below to create tracks/sources return scoped_refptrs
// rather than unique_ptrs, because these interfaces are also used with
// PeerConnection, where everything is ref-counted.
// Creates a audio source representing the default microphone input.
// |options| decides audio processing settings.
virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
const cricket::AudioOptions& options) = 0;
// Version of the above method that uses default options.
rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource() {
return CreateAudioSource(cricket::AudioOptions());
}
// Creates a video source object wrapping and taking ownership of |capturer|.
//
// |constraints| can be used for selection of resolution and frame rate, and
// may be null if no constraints are desired.
virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer,
const MediaConstraintsInterface* constraints) = 0;
// Version of the above method that omits |constraints|.
rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer> capturer) {
return CreateVideoSource(std::move(capturer), nullptr);
}
// Creates a new local video track wrapping |source|. The same |source| can
// be used in several tracks.
virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& id,
VideoTrackSourceInterface* source) = 0;
// Creates an new local audio track wrapping |source|.
virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& id,
AudioSourceInterface* source) = 0;
};
} // namespace webrtc
#endif // API_ORTC_ORTCFACTORYINTERFACE_H_