a8eb756a34
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
41 lines
1.1 KiB
C++
41 lines
1.1 KiB
C++
/*
|
|
* 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 WEBRTC_API_CALL_TRANSPORT_H_
|
|
#define WEBRTC_API_CALL_TRANSPORT_H_
|
|
|
|
#include <stddef.h>
|
|
#include <stdint.h>
|
|
|
|
namespace webrtc {
|
|
|
|
// TODO(holmer): Look into unifying this with the PacketOptions in
|
|
// asyncpacketsocket.h.
|
|
struct PacketOptions {
|
|
// A 16 bits positive id. Negative ids are invalid and should be interpreted
|
|
// as packet_id not being set.
|
|
int packet_id = -1;
|
|
};
|
|
|
|
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 // WEBRTC_API_CALL_TRANSPORT_H_
|