| Index: webrtc/api/ortcfactoryinterface.h
|
| diff --git a/webrtc/api/ortcfactoryinterface.h b/webrtc/api/ortcfactoryinterface.h
|
| index 8d46d6865eb1e69b0422a751a74bd2818fc050f1..caaf8abfaf0e21f60a1c4a297ede01af1764d90d 100644
|
| --- a/webrtc/api/ortcfactoryinterface.h
|
| +++ b/webrtc/api/ortcfactoryinterface.h
|
| @@ -13,13 +13,28 @@
|
|
|
| #include <memory>
|
|
|
| +#include "webrtc/api/mediaconstraintsinterface.h"
|
| +#include "webrtc/api/mediastreaminterface.h"
|
| +#include "webrtc/api/mediatypes.h"
|
| +#include "webrtc/api/ortcrtpreceiverinterface.h"
|
| +#include "webrtc/api/ortcrtpsenderinterface.h"
|
| +#include "webrtc/api/packettransportinterface.h"
|
| +#include "webrtc/api/rtcerror.h"
|
| +#include "webrtc/api/rtpparameters.h"
|
| +#include "webrtc/api/rtptransportcontrollerinterface.h"
|
| +#include "webrtc/api/rtptransportinterface.h"
|
| #include "webrtc/api/udptransportinterface.h"
|
| #include "webrtc/base/network.h"
|
| +#include "webrtc/base/scoped_ref_ptr.h"
|
| #include "webrtc/base/thread.h"
|
| #include "webrtc/p2p/base/packetsocketfactory.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.
|
| //
|
| @@ -29,6 +44,9 @@ namespace webrtc {
|
| // 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: The factory must not be destroyed before destroying the
|
| // objects it created, and the objects passed into the factory must not be
|
| // destroyed before destroying the factory.
|
| @@ -50,28 +68,148 @@ class OrtcFactoryInterface {
|
| // 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.
|
| + //
|
| // Note that the OrtcFactoryInterface does not take ownership of any of the
|
| - // objects
|
| - // passed in, and as previously stated, these objects can't be destroyed
|
| - // before the factory is.
|
| - static std::unique_ptr<OrtcFactoryInterface> Create(
|
| + // objects passed in, 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);
|
| + rtc::PacketSocketFactory* socket_factory,
|
| + AudioDeviceModule* adm);
|
| +
|
| // Constructor for convenience which uses default implementations of
|
| // everything (though does still require that the current thread runs a
|
| // message loop; see above).
|
| - static std::unique_ptr<OrtcFactoryInterface> Create() {
|
| - return Create(nullptr, nullptr, nullptr, nullptr);
|
| + static RTCErrorOr<std::unique_ptr<OrtcFactoryInterface>> Create() {
|
| + return Create(nullptr, nullptr, nullptr, nullptr, nullptr);
|
| }
|
|
|
| virtual ~OrtcFactoryInterface() {}
|
|
|
| - virtual std::unique_ptr<UdpTransportInterface>
|
| + // Creates an RTP transport controller, which is required for calls to
|
| + // CreateRtpTransport methods. If your application has some notion of a
|
| + // "call", you should create one transport controller per call.
|
| + // 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| can if RTCP muxing is being used immediately,
|
| + // meaning |rtcp_parameters.mux| is true.
|
| + //
|
| + // |transport_controller| must not be null.
|
| + virtual RTCErrorOr<std::unique_ptr<RtpTransportInterface>> CreateRtpTransport(
|
| + const RtcpParameters& rtcp_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 and starts sending the provided |track| (assuming an
|
| + // active encoding exists in |rtp_parameters|).
|
| + //
|
| + // |track| and |transport| must not be null.
|
| + virtual RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>> CreateRtpSender(
|
| + rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
| + const RtpParameters& rtp_parameters,
|
| + RtpTransportInterface* transport) = 0;
|
| +
|
| + // Same as above, but 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,
|
| + const RtpParameters& rtp_parameters,
|
| + 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 and prepares to receive a MediaStreamTrack
|
| + // described by |rtp_parameters|.
|
| + //
|
| + // |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,
|
| + const RtpParameters& rtp_parameters,
|
| + 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;
|
| +
|
| + // 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;
|
| +
|
| // Method for convenience that has no port range restrictions.
|
| - std::unique_ptr<UdpTransportInterface> CreateUdpTransport(int family) {
|
| + RTCErrorOr<std::unique_ptr<UdpTransportInterface>> CreateUdpTransport(
|
| + int family) {
|
| return CreateUdpTransport(family, 0, 0);
|
| }
|
| };
|
|
|