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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/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;
};
class Transport {
public:
virtual bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) = 0;
virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
protected:
virtual ~Transport() {}
};
} // namespace webrtc
#endif // API_CALL_TRANSPORT_H_
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