Files
platform-external-webrtc/webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
Henrik Kjellander ff761fba82 modules: more interface -> include renames
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface

To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
                "use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)

BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc

R=stefan@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1417683006 .

Cr-Commit-Position: refs/heads/master@{#10500}
2015-11-04 07:32:04 +00:00

105 lines
3.3 KiB
C++

/*
* 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 NETEQTEST_RTPPACKET_H
#define NETEQTEST_RTPPACKET_H
#include <map>
#include <stdio.h>
#include "webrtc/typedefs.h"
#include "webrtc/modules/include/module_common_types.h"
enum stereoModes {
stereoModeMono,
stereoModeSample1,
stereoModeSample2,
stereoModeFrame,
stereoModeDuplicate
};
class NETEQTEST_RTPpacket
{
public:
NETEQTEST_RTPpacket();
bool operator !() const { return (dataLen() < 0); };
virtual ~NETEQTEST_RTPpacket();
void reset();
static int skipFileHeader(FILE *fp);
virtual int readFromFile(FILE *fp);
int readFixedFromFile(FILE *fp, size_t len);
virtual int writeToFile(FILE *fp);
void blockPT(uint8_t pt);
virtual void parseHeader();
void parseHeader(webrtc::WebRtcRTPHeader* rtp_header);
const webrtc::WebRtcRTPHeader* RTPinfo() const;
uint8_t * datagram() const;
uint8_t * payload() const;
size_t payloadLen();
int16_t dataLen() const;
bool isParsed() const;
bool isLost() const;
uint32_t time() const { return _receiveTime; };
uint8_t payloadType() const;
uint16_t sequenceNumber() const;
uint32_t timeStamp() const;
uint32_t SSRC() const;
uint8_t markerBit() const;
int setPayloadType(uint8_t pt);
int setSequenceNumber(uint16_t sn);
int setTimeStamp(uint32_t ts);
int setSSRC(uint32_t ssrc);
int setMarkerBit(uint8_t mb);
void setTime(uint32_t receiveTime) { _receiveTime = receiveTime; };
int setRTPheader(const webrtc::WebRtcRTPHeader* RTPinfo);
int splitStereo(NETEQTEST_RTPpacket* slaveRtp, enum stereoModes mode);
int extractRED(int index, webrtc::WebRtcRTPHeader& red);
void scramblePayload(void);
uint8_t * _datagram;
uint8_t * _payloadPtr;
int _memSize;
int16_t _datagramLen;
size_t _payloadLen;
webrtc::WebRtcRTPHeader _rtpInfo;
bool _rtpParsed;
uint32_t _receiveTime;
bool _lost;
std::map<uint8_t, bool> _blockList;
protected:
static const int _kRDHeaderLen;
static const int _kBasicHeaderLen;
void parseBasicHeader(webrtc::WebRtcRTPHeader* RTPinfo, int *i_P, int *i_X,
int *i_CC) const;
int calcHeaderLength(int i_X, int i_CC) const;
private:
void makeRTPheader(unsigned char* rtp_data, uint8_t payloadType,
uint16_t seqNo, uint32_t timestamp,
uint32_t ssrc, uint8_t markerBit) const;
uint16_t parseRTPheader(webrtc::WebRtcRTPHeader* RTPinfo,
uint8_t **payloadPtr = NULL) const;
uint16_t parseRTPheader(uint8_t **payloadPtr = NULL)
{ return parseRTPheader(&_rtpInfo, payloadPtr);};
int calcPadLength(int i_P) const;
void splitStereoSample(NETEQTEST_RTPpacket* slaveRtp, int stride);
void splitStereoFrame(NETEQTEST_RTPpacket* slaveRtp);
void splitStereoDouble(NETEQTEST_RTPpacket* slaveRtp);
};
#endif //NETEQTEST_RTPPACKET_H