5312a8f532
As an alternative to attaching custom array of bytes. Bug: b/178094662 Change-Id: I92dcbf04998d8206091125febc520ebfcc4bcebf Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203264 Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33069}
59 lines
1.7 KiB
C++
59 lines
1.7 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_CALL_TRANSPORT_H_
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#define API_CALL_TRANSPORT_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <vector>
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#include "api/ref_counted_base.h"
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#include "api/scoped_refptr.h"
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namespace webrtc {
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// TODO(holmer): Look into unifying this with the PacketOptions in
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// asyncpacketsocket.h.
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struct PacketOptions {
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PacketOptions();
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PacketOptions(const PacketOptions&);
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~PacketOptions();
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// A 16 bits positive id. Negative ids are invalid and should be interpreted
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// as packet_id not being set.
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int packet_id = -1;
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// Deprecated: use `additional_data` instead of `application_data`.
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std::vector<uint8_t> application_data;
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// Additional data bound to the RTP packet for use in application code,
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// outside of WebRTC.
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rtc::scoped_refptr<rtc::RefCountedBase> additional_data;
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// Whether this is a retransmission of an earlier packet.
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bool is_retransmit = false;
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bool included_in_feedback = false;
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bool included_in_allocation = false;
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};
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class Transport {
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public:
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virtual bool SendRtp(const uint8_t* packet,
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size_t length,
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const PacketOptions& options) = 0;
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virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
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protected:
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virtual ~Transport() {}
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};
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} // namespace webrtc
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#endif // API_CALL_TRANSPORT_H_
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