webrtc/call/rtp_config.h
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

48 lines
1.4 KiB
C++

/*
* Copyright (c) 2017 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 CALL_RTP_CONFIG_H_
#define CALL_RTP_CONFIG_H_
#include <string>
namespace webrtc {
// Settings for NACK, see RFC 4585 for details.
struct NackConfig {
NackConfig() : rtp_history_ms(0) {}
std::string ToString() const;
// Send side: the time RTP packets are stored for retransmissions.
// Receive side: the time the receiver is prepared to wait for
// retransmissions.
// Set to '0' to disable.
int rtp_history_ms;
};
// Settings for ULPFEC forward error correction.
// Set the payload types to '-1' to disable.
struct UlpfecConfig {
UlpfecConfig()
: ulpfec_payload_type(-1),
red_payload_type(-1),
red_rtx_payload_type(-1) {}
std::string ToString() const;
bool operator==(const UlpfecConfig& other) const;
// Payload type used for ULPFEC packets.
int ulpfec_payload_type;
// Payload type used for RED packets.
int red_payload_type;
// RTX payload type for RED payload.
int red_rtx_payload_type;
};
} // namespace webrtc
#endif // CALL_RTP_CONFIG_H_