Index: webrtc/api/ortc/ortcrtpsenderinterface.h |
diff --git a/webrtc/api/ortc/ortcrtpsenderinterface.h b/webrtc/api/ortc/ortcrtpsenderinterface.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..e369b539e41febc7e193f32b807b2634ad8cfacb |
--- /dev/null |
+++ b/webrtc/api/ortc/ortcrtpsenderinterface.h |
@@ -0,0 +1,77 @@ |
+/* |
+ * 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_ |