webrtc/media/base/rtputils.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

90 lines
3.2 KiB
C++

/*
* Copyright (c) 2011 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 MEDIA_BASE_RTPUTILS_H_
#define MEDIA_BASE_RTPUTILS_H_
#include "rtc_base/byteorder.h"
namespace rtc {
struct PacketTimeUpdateParams;
} // namespace rtc
namespace cricket {
const size_t kMinRtpPacketLen = 12;
const size_t kMaxRtpPacketLen = 2048;
const size_t kMinRtcpPacketLen = 4;
struct RtpHeader {
int payload_type;
int seq_num;
uint32_t timestamp;
uint32_t ssrc;
};
enum RtcpTypes {
kRtcpTypeSR = 200, // Sender report payload type.
kRtcpTypeRR = 201, // Receiver report payload type.
kRtcpTypeSDES = 202, // SDES payload type.
kRtcpTypeBye = 203, // BYE payload type.
kRtcpTypeApp = 204, // APP payload type.
kRtcpTypeRTPFB = 205, // Transport layer Feedback message payload type.
kRtcpTypePSFB = 206, // Payload-specific Feedback message payload type.
};
bool GetRtpPayloadType(const void* data, size_t len, int* value);
bool GetRtpSeqNum(const void* data, size_t len, int* value);
bool GetRtpTimestamp(const void* data, size_t len, uint32_t* value);
bool GetRtpSsrc(const void* data, size_t len, uint32_t* value);
bool GetRtpHeaderLen(const void* data, size_t len, size_t* value);
bool GetRtcpType(const void* data, size_t len, int* value);
bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value);
bool GetRtpHeader(const void* data, size_t len, RtpHeader* header);
bool SetRtpSsrc(void* data, size_t len, uint32_t value);
// Assumes version 2, no padding, no extensions, no csrcs.
bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
bool IsRtpPacket(const void* data, size_t len);
// True if |payload type| is 0-127.
bool IsValidRtpPayloadType(int payload_type);
// True if |size| is appropriate for the indicated packet type.
bool IsValidRtpRtcpPacketSize(bool rtcp, size_t size);
// TODO(zstein): Consider using an enum instead of a bool to differentiate
// between RTP and RTCP.
// Returns "RTCP" or "RTP" according to |rtcp|.
const char* RtpRtcpStringLiteral(bool rtcp);
// Verifies that a packet has a valid RTP header.
bool ValidateRtpHeader(const uint8_t* rtp,
size_t length,
size_t* header_length);
// Helper method which updates the absolute send time extension if present.
bool UpdateRtpAbsSendTimeExtension(uint8_t* rtp,
size_t length,
int extension_id,
uint64_t time_us);
// Applies specified |options| to the packet. It updates the absolute send time
// extension header if it is present present then updates HMAC.
bool ApplyPacketOptions(uint8_t* data,
size_t length,
const rtc::PacketTimeUpdateParams& packet_time_params,
uint64_t time_us);
} // namespace cricket
#endif // MEDIA_BASE_RTPUTILS_H_