blob: 292aa11da37026a54ea117f0cffa1c6f8ff456c0 [file] [log] [blame]
* 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.
#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(const 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 {
virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
const PacketOptions& options) = 0;
virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) = 0;
virtual ~Transport() {}
} // namespace webrtc