mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
53 lines
1.5 KiB
C++
53 lines
1.5 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 API_CALL_TRANSPORT_H_
|
|
#define API_CALL_TRANSPORT_H_
|
|
|
|
#include <stddef.h>
|
|
#include <stdint.h>
|
|
|
|
#include <vector>
|
|
|
|
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.
|
|
std::vector<uint8_t> application_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_
|