Index: webrtc/call/rtp_transport_controller_receive_interface.h |
diff --git a/webrtc/call/rtp_transport_controller_receive_interface.h b/webrtc/call/rtp_transport_controller_receive_interface.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..420b0e4f5b0f76ba2cb4b9e2ed021a96e304af3f |
--- /dev/null |
+++ b/webrtc/call/rtp_transport_controller_receive_interface.h |
@@ -0,0 +1,46 @@ |
+/* |
+ * Copyright (c) 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_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_ |
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_ |
+ |
+#include <memory> |
+ |
+#include "webrtc/call/rtp_packet_sink_interface.h" |
+ |
+namespace webrtc { |
+ |
+// An RtpTransportReceiver is responsible for the rtp-specific but |
+// media-independent state needed for receiving a stream. |
pthatcher
2017/06/02 22:44:17
"stream" is ambiguous in the RTP world. Needs to
|
+// TODO(nisse): Currently, only owns the association between ssrc and |
+// the stream's RtpPacketSinkInterface. Ownership of corresponding |
+// objects from modules/rtp_rtcp/ should move to this class (or |
+// rather, the corresponding implmentation class), and we should add |
+// have methods for getting rtp receive stats. |
+class RtpTransportReceiver { |
+ public: |
+ virtual ~RtpTransportReceiver() {} |
+}; |
+ |
+// This class acts as a factory for RtpTransportReceive objects. |
+class RtpTransportControllerReceiveInterface { |
+ public: |
+ virtual ~RtpTransportControllerReceiveInterface() {} |
+ |
+ virtual std::unique_ptr<RtpTransportReceiver> CreateReceiver( |
+ uint32_t ssrc, |
+ RtpPacketSinkInterface* sink) = 0; |
+ // For registering additional sinks, needed for FlexFEC. |
+ virtual void AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) = 0; |
+ virtual size_t RemoveSink(const RtpPacketSinkInterface* sink) = 0; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_RECEIVE_INTERFACE_H_ |