Repo created

TMessagesProj/jni/voip/webrtc/api/call/transport.h

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+/*
+ *  Copyright (c) 2013 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 API_CALL_TRANSPORT_H_
+#define API_CALL_TRANSPORT_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include "api/array_view.h"
+#include "api/ref_counted_base.h"
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+// TODO(holmer): Look into unifying this with the PacketOptions in
+// asyncpacketsocket.h.
+struct PacketOptions {
+  PacketOptions();
+  PacketOptions(const PacketOptions&);
+  ~PacketOptions();
+
+  // A 16 bits positive id. Negative ids are invalid and should be interpreted
+  // as packet_id not being set.
+  int packet_id = -1;
+  // Additional data bound to the RTP packet for use in application code,
+  // outside of WebRTC.
+  rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
+  // Whether this is a retransmission of an earlier packet.
+  bool is_retransmit = false;
+  bool included_in_feedback = false;
+  bool included_in_allocation = false;
+  // Whether this packet can be part of a packet batch at lower levels.
+  bool batchable = false;
+  // Whether this packet is the last of a batch.
+  bool last_packet_in_batch = false;
+};
+
+class Transport {
+ public:
+  virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
+                       const PacketOptions& options) = 0;
+  virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) = 0;
+
+ protected:
+  virtual ~Transport() {}
+};
+
+}  // namespace webrtc
+
+#endif  // API_CALL_TRANSPORT_H_