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/*
* Copyright (c) 2012 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_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
#include <map>
#include "webrtc/base/deprecation.h"
#include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/typedefs.h"
namespace webrtc {
const uint8_t kRtpMarkerBitMask = 0x80;
RtpData* NullObjectRtpData();
RtpFeedback* NullObjectRtpFeedback();
RtpAudioFeedback* NullObjectRtpAudioFeedback();
ReceiveStatistics* NullObjectReceiveStatistics();
namespace RtpUtility {
struct Payload {
char name[RTP_PAYLOAD_NAME_SIZE];
bool audio;
PayloadUnion typeSpecific;
};
typedef std::map<int8_t, Payload*> PayloadTypeMap;
bool StringCompare(const char* str1, const char* str2, const uint32_t length);
// Round up to the nearest size that is a multiple of 4.
size_t Word32Align(size_t size);
class RtpHeaderParser {
public:
RtpHeaderParser(const uint8_t* rtpData, size_t rtpDataLength);
~RtpHeaderParser();
bool RTCP() const;
bool ParseRtcp(RTPHeader* header) const;
bool Parse(RTPHeader* parsedPacket,
RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
RTC_DEPRECATED bool Parse(
RTPHeader& parsedPacket, // NOLINT(runtime/references)
RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const {
return Parse(&parsedPacket, ptrExtensionMap);
}
private:
void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
const RtpHeaderExtensionMap* ptrExtensionMap,
const uint8_t* ptrRTPDataExtensionEnd,
const uint8_t* ptr) const;
uint8_t ParsePaddingBytes(const uint8_t* ptrRTPDataExtensionEnd,
const uint8_t* ptr) const;
const uint8_t* const _ptrRTPDataBegin;
const uint8_t* const _ptrRTPDataEnd;
};
} // namespace RtpUtility
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
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