Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/api/ortc/ortcrtpsenderinterface.h

+/*
+ *  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.
+ */
+
+// This file contains interfaces for RtpSenders:
+// http://publications.ortc.org/2016/20161202/#rtcrtpsender*
+//
+// However, underneath the RtpSender is an RtpTransport, rather than a
+// DtlsTransport. This is to allow different types of RTP transports (besides
+// DTLS-SRTP) to be used.
+
+#ifndef WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
+#define WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_
+
+#include "webrtc/api/mediatypes.h"
+#include "webrtc/api/mediastreaminterface.h"
+#include "webrtc/api/ortc/rtptransportinterface.h"
+#include "webrtc/api/rtcerror.h"
+#include "webrtc/api/rtpparameters.h"
+
+namespace webrtc {
+
+// Note: Since sender capabilities may depend on how the OrtcFactory was
+// created, instead of a static "GetCapabilities" method on this interface,
+// there is a "GetRtpSenderCapabilities" method on the OrtcFactory.
+class OrtcRtpSenderInterface {
+ public:
+  virtual ~OrtcRtpSenderInterface() {}
+
+  // Sets the source of media that will be sent by this sender.
+  //
+  // If Send has already been called, will immediately switch to sending this
+  // track. If |track| is null, will stop sending media.
+  //
+  // Returns INVALID_PARAMETER error if an audio track is set on a video
+  // RtpSender, or vice-versa.
+  virtual RTCError SetTrack(MediaStreamTrackInterface* track) = 0;
+  // Returns previously set (or constructed-with) track.
+  virtual rtc::scoped_refptr<MediaStreamTrackInterface> GetTrack() const = 0;
+
+  // Once supported, will switch to sending media on a new transport. However,
+  // this is not currently supported and will always return an error.
+  virtual RTCError SetTransport(RtpTransportInterface* transport) = 0;
+  // Returns previously set (or constructed-with) transport.
+  virtual RtpTransportInterface* GetTransport() const = 0;
+
+  // Start sending media with |parameters| (if |parameters| contains an active
+  // encoding).
+  //
+  // There are no limitations to how the parameters can be changed after the
+  // initial call to Send, as long as they're valid (for example, they can't
+  // use the same payload type for two codecs).
+  virtual RTCError Send(const RtpParameters& parameters) = 0;
+  // Returns parameters that were last successfully passed into Send, or empty
+  // parameters if that hasn't yet occurred.
+  //
+  // Note that for parameters that are described as having an "implementation
+  // default" value chosen, GetParameters() will return those chosen defaults,
+  // with the exception of SSRCs which have special behavior. See
+  // rtpparameters.h for more details.
+  virtual RtpParameters GetParameters() const = 0;
+
+  // Audio or video sender?
+  virtual cricket::MediaType GetKind() const = 0;
+
+  // TODO(deadbeef): SSRC conflict signal.
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_API_ORTC_ORTCRTPSENDERINTERFACE_H_