/* * 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.
*/
// TODO(holmer): Look into unifying this with the PacketOptions in // asyncpacketsocket.h. struct PacketOptions {
PacketOptions();
PacketOptions(const PacketOptions&);
~PacketOptions();
// Negative ids are invalid and should be interpreted // as packet_id not being set.
int64_t packet_id = -1; // Whether this is an audio or video packet, excluding retransmissions. bool is_media = true; 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: virtualbool SendRtp(rtc::ArrayView<const uint8_t> packet, const PacketOptions& options) = 0; virtualbool SendRtcp(rtc::ArrayView<const uint8_t> packet) = 0;
protected: virtual ~Transport() {}
};
} // namespace webrtc
#endif// API_CALL_TRANSPORT_H_
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