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); |
} |
}; |