Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver
They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:
* You can only have one of each type of sender and receiver (audio/video) on top
of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.
Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:
ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine
And later we hope to have simply:
PeerConnection -> "Real" ORTC objects -> Media engine
See the linked bug for more context.
BUG=webrtc:7013
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
diff --git a/webrtc/api/ortc/rtptransportcontrollerinterface.h b/webrtc/api/ortc/rtptransportcontrollerinterface.h
new file mode 100644
index 0000000..d1d0e44
--- /dev/null
+++ b/webrtc/api/ortc/rtptransportcontrollerinterface.h
@@ -0,0 +1,57 @@
+/*
+ * 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 WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
+#define WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_
+
+#include <vector>
+
+#include "webrtc/api/ortc/rtptransportinterface.h"
+
+namespace webrtc {
+
+class RtpTransportControllerAdapter;
+
+// Used to group RTP transports between a local endpoint and the same remote
+// endpoint, for the purpose of sharing bandwidth estimation and other things.
+//
+// Comparing this to the PeerConnection model, non-budled audio/video would use
+// two RtpTransports with a single RtpTransportController, whereas bundled
+// media would use a single RtpTransport, and two PeerConnections would use
+// independent RtpTransportControllers.
+//
+// RtpTransports are associated with this controller when they're created, by
+// passing the controller into OrtcFactory's relevant "CreateRtpTransport"
+// method. When a transport is destroyed, it's automatically disassociated.
+// GetTransports returns all currently associated transports.
+//
+// This is the RTP equivalent of "IceTransportController" in ORTC; RtpTransport
+// is to RtpTransportController as IceTransport is to IceTransportController.
+class RtpTransportControllerInterface {
+ public:
+ virtual ~RtpTransportControllerInterface() {}
+
+ // Returns all transports associated with this controller (see explanation
+ // above). No ordering is guaranteed.
+ virtual std::vector<RtpTransportInterface*> GetTransports() const = 0;
+
+ protected:
+ // Only for internal use. Returns a pointer to an internal interface, for use
+ // by the implementation.
+ virtual RtpTransportControllerAdapter* GetInternal() = 0;
+
+ // Classes that can use this internal interface.
+ friend class OrtcFactory;
+ friend class RtpTransportAdapter;
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_API_ORTC_RTPTRANSPORTCONTROLLERINTERFACE_H_