Replacing the legacy tool RTPjitter with a new rtp_jitter
This new tool provides the similar functionality as the legacy tool, but
is implemented using less legacy helpers. It also replaces RTPtimeshift
and RTPchange. The most significant change versus the old RTPjitter tool
is that the new tool takes the timing data in the form of integers in a
text file (instead of the binary data file used by the old tool). This
should make it easier to create custom timing files when needed.
Bug: webrtc:2692
Change-Id: I5e46fe7abdd9ca8c04a04de87555204fca36e287
Reviewed-on: https://webrtc-review.googlesource.com/25700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20868}diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index a0b1525..04683d1 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -1189,9 +1189,6 @@
group("audio_coding_tests") {
testonly = true
public_deps = [
- ":RTPchange",
- ":RTPjitter",
- ":RTPtimeshift",
":acm_receive_test",
":acm_send_test",
":audio_codec_speed_tests",
@@ -1213,6 +1210,7 @@
":neteq_speed_test",
":rtp_analyze",
":rtp_encode",
+ ":rtp_jitter",
":rtpcat",
":webrtc_opus_fec_test",
]
@@ -1611,40 +1609,6 @@
]
}
- rtc_source_set("neteq_test_tools_deprecated") {
- testonly = true
- sources = [
- "neteq/test/NETEQTEST_DummyRTPpacket.cc",
- "neteq/test/NETEQTEST_DummyRTPpacket.h",
- "neteq/test/NETEQTEST_RTPpacket.cc",
- "neteq/test/NETEQTEST_RTPpacket.h",
- ]
-
- deps = [
- ":cng",
- ":g711",
- ":g722",
- ":ilbc",
- ":isac",
- ":pcm16b",
- "..:module_api",
- "../..:webrtc_common",
- "//testing/gtest",
- ]
-
- include_dirs = [
- "neteq/include",
- "neteq/test",
- ]
-
- if (is_win) {
- cflags = [
- # Disable warnings to enable Win64 build, issue 1323.
- "/wd4267", # size_t to int truncation
- ]
- }
- }
-
rtc_executable("rtp_encode") {
testonly = true
@@ -1668,16 +1632,21 @@
defines = audio_coding_defines
}
- rtc_executable("RTPchange") {
+ rtc_executable("rtp_jitter") {
testonly = true
+ deps = audio_coding_deps + [
+ "../rtp_rtcp:rtp_rtcp_format",
+ "../../api:array_view",
+ "../../rtc_base:rtc_base_approved",
+ "../../system_wrappers:system_wrappers_default",
+ ]
+
sources = [
- "neteq/test/RTPchange.cc",
+ "neteq/tools/rtp_jitter.cc",
]
- deps = [
- ":neteq_test_tools_deprecated",
- ]
+ defines = audio_coding_defines
}
rtc_executable("rtpcat") {
@@ -1695,33 +1664,6 @@
]
}
- rtc_executable("RTPtimeshift") {
- testonly = true
-
- sources = [
- "neteq/test/RTPtimeshift.cc",
- ]
-
- deps = [
- ":neteq_test_tools_deprecated",
- "../../test:test_support",
- "//testing/gtest",
- ]
- }
-
- rtc_executable("RTPjitter") {
- testonly = true
- deps = [
- "../..:webrtc_common",
- "../../rtc_base:rtc_base_approved",
- "../../test:test_support",
- "//testing/gtest",
- ]
- sources = [
- "neteq/test/RTPjitter.cc",
- ]
- }
-
rtc_executable("rtp_analyze") {
testonly = true
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc b/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc
deleted file mode 100644
index 37e215b..0000000
--- a/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.cc
+++ /dev/null
@@ -1,204 +0,0 @@
-/*
- * 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.
- */
-
-#include "NETEQTEST_DummyRTPpacket.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <string.h>
-
-#ifdef WIN32
-#include <winsock2.h>
-#else
-#include <netinet/in.h> // for htons, htonl, etc
-#endif
-
-int NETEQTEST_DummyRTPpacket::readFromFile(FILE *fp)
-{
- if (!fp)
- {
- return -1;
- }
-
- uint16_t length, plen;
- uint32_t offset;
- int packetLen = 0;
-
- bool readNextPacket = true;
- while (readNextPacket) {
- readNextPacket = false;
- if (fread(&length, 2, 1, fp) == 0)
- {
- reset();
- return -2;
- }
- length = ntohs(length);
-
- if (fread(&plen, 2, 1, fp) == 0)
- {
- reset();
- return -1;
- }
- packetLen = ntohs(plen);
-
- if (fread(&offset, 4, 1, fp) == 0)
- {
- reset();
- return -1;
- }
- // Store in local variable until we have passed the reset below.
- uint32_t receiveTime = ntohl(offset);
-
- // Use length here because a plen of 0 specifies rtcp.
- length = (uint16_t) (length - _kRDHeaderLen);
-
- // check buffer size
- if (_datagram && _memSize < length + 1)
- {
- reset();
- }
-
- if (!_datagram)
- {
- // Add one extra byte, to be able to fake a dummy payload of 1 byte.
- _datagram = new uint8_t[length + 1];
- _memSize = length + 1;
- }
- memset(_datagram, 0, length + 1);
-
- if (length == 0)
- {
- _datagramLen = 0;
- _rtpParsed = false;
- return packetLen;
- }
-
- // Read basic header
- if (fread((unsigned short *) _datagram, 1, _kBasicHeaderLen, fp)
- != (size_t)_kBasicHeaderLen)
- {
- reset();
- return -1;
- }
- _receiveTime = receiveTime;
- _datagramLen = _kBasicHeaderLen;
-
- // Parse the basic header
- webrtc::WebRtcRTPHeader tempRTPinfo;
- int P, X, CC;
- parseBasicHeader(&tempRTPinfo, &P, &X, &CC);
-
- // Check if we have to extend the header
- if (X != 0 || CC != 0)
- {
- int newLen = _kBasicHeaderLen + CC * 4 + X * 4;
- assert(_memSize >= newLen);
-
- // Read extension from file
- size_t readLen = newLen - _kBasicHeaderLen;
- if (fread(&_datagram[_kBasicHeaderLen], 1, readLen, fp) != readLen)
- {
- reset();
- return -1;
- }
- _datagramLen = newLen;
-
- if (X != 0)
- {
- int totHdrLen = calcHeaderLength(X, CC);
- assert(_memSize >= totHdrLen);
-
- // Read extension from file
- size_t readLen = totHdrLen - newLen;
- if (fread(&_datagram[newLen], 1, readLen, fp) != readLen)
- {
- reset();
- return -1;
- }
- _datagramLen = totHdrLen;
- }
- }
- _datagramLen = length;
-
- if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
- {
- readNextPacket = true;
- }
- }
-
- _rtpParsed = false;
- assert(_memSize > _datagramLen);
- _payloadLen = 1; // Set the length to 1 byte.
- return packetLen;
-
-}
-
-int NETEQTEST_DummyRTPpacket::writeToFile(FILE *fp)
-{
- if (!fp)
- {
- return -1;
- }
-
- uint16_t length, plen;
- uint32_t offset;
-
- // length including RTPplay header
- length = htons(_datagramLen + _kRDHeaderLen);
- if (fwrite(&length, 2, 1, fp) != 1)
- {
- return -1;
- }
-
- // payload length
- plen = htons(_datagramLen);
- if (fwrite(&plen, 2, 1, fp) != 1)
- {
- return -1;
- }
-
- // offset (=receive time)
- offset = htonl(_receiveTime);
- if (fwrite(&offset, 4, 1, fp) != 1)
- {
- return -1;
- }
-
- // Figure out the length of the RTP header.
- int headerLen;
- if (_datagramLen == 0)
- {
- // No payload at all; we are done writing to file.
- headerLen = 0;
- }
- else
- {
- parseHeader();
- headerLen = _payloadPtr - _datagram;
- assert(headerLen >= 0);
- }
-
- // write RTP header
- if (fwrite((unsigned short *) _datagram, 1, headerLen, fp) !=
- static_cast<size_t>(headerLen))
- {
- return -1;
- }
-
- return (headerLen + _kRDHeaderLen); // total number of bytes written
-
-}
-
-void NETEQTEST_DummyRTPpacket::parseHeader() {
- NETEQTEST_RTPpacket::parseHeader();
- // Change _payloadLen to 1 byte. The memory should always be big enough.
- assert(_memSize > _datagramLen);
- _payloadLen = 1;
-}
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h b/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h
deleted file mode 100644
index 1ac6c9a..0000000
--- a/modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * 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_DUMMYRTPPACKET_H
-#define NETEQTEST_DUMMYRTPPACKET_H
-
-#include "NETEQTEST_RTPpacket.h"
-
-class NETEQTEST_DummyRTPpacket : public NETEQTEST_RTPpacket {
- public:
- int readFromFile(FILE* fp) override;
- int writeToFile(FILE* fp) override;
- void parseHeader() override;
-};
-
-#endif // NETEQTEST_DUMMYRTPPACKET_H
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
deleted file mode 100644
index 3d44fbc..0000000
--- a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
+++ /dev/null
@@ -1,851 +0,0 @@
-/*
- * 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.
- */
-
-#include "NETEQTEST_RTPpacket.h"
-
-#include <assert.h>
-#include <stdlib.h> // rand
-#include <string.h>
-
-#ifdef WIN32
-#include <winsock2.h>
-#else
-#include <netinet/in.h> // for htons, htonl, etc
-#endif
-
-const int NETEQTEST_RTPpacket::_kRDHeaderLen = 8;
-const int NETEQTEST_RTPpacket::_kBasicHeaderLen = 12;
-
-NETEQTEST_RTPpacket::NETEQTEST_RTPpacket()
-:
-_datagram(NULL),
-_payloadPtr(NULL),
-_memSize(0),
-_datagramLen(-1),
-_payloadLen(0),
-_rtpParsed(false),
-_receiveTime(0),
-_lost(false)
-{
- memset(&_rtpInfo, 0, sizeof(_rtpInfo));
- _blockList.clear();
-}
-
-NETEQTEST_RTPpacket::~NETEQTEST_RTPpacket()
-{
- if(_datagram)
- {
- delete [] _datagram;
- }
-}
-
-void NETEQTEST_RTPpacket::reset()
-{
- if(_datagram) {
- delete [] _datagram;
- }
- _datagram = NULL;
- _memSize = 0;
- _datagramLen = -1;
- _payloadLen = 0;
- _payloadPtr = NULL;
- _receiveTime = 0;
- memset(&_rtpInfo, 0, sizeof(_rtpInfo));
- _rtpParsed = false;
-
-}
-
-int NETEQTEST_RTPpacket::skipFileHeader(FILE *fp)
-{
- if (!fp) {
- return -1;
- }
-
- const int kFirstLineLength = 40;
- char firstline[kFirstLineLength];
- if (fgets(firstline, kFirstLineLength, fp) == NULL) {
- return -1;
- }
- if (strncmp(firstline, "#!rtpplay", 9) == 0) {
- if (strncmp(firstline, "#!rtpplay1.0", 12) != 0) {
- return -1;
- }
- }
- else if (strncmp(firstline, "#!RTPencode", 11) == 0) {
- if (strncmp(firstline, "#!RTPencode1.0", 14) != 0) {
- return -1;
- }
- }
- else
- {
- return -1;
- }
-
- const int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- if (fseek(fp, kRtpDumpHeaderSize, SEEK_CUR) != 0)
- {
- return -1;
- }
- return 0;
-}
-
-int NETEQTEST_RTPpacket::readFromFile(FILE *fp)
-{
- if(!fp)
- {
- return(-1);
- }
-
- uint16_t length, plen;
- uint32_t offset;
- int packetLen = 0;
-
- bool readNextPacket = true;
- while (readNextPacket) {
- readNextPacket = false;
- if (fread(&length,2,1,fp)==0)
- {
- reset();
- return(-2);
- }
- length = ntohs(length);
-
- if (fread(&plen,2,1,fp)==0)
- {
- reset();
- return(-1);
- }
- packetLen = ntohs(plen);
-
- if (fread(&offset,4,1,fp)==0)
- {
- reset();
- return(-1);
- }
- // store in local variable until we have passed the reset below
- uint32_t receiveTime = ntohl(offset);
-
- // Use length here because a plen of 0 specifies rtcp
- length = (uint16_t) (length - _kRDHeaderLen);
-
- // check buffer size
- if (_datagram && _memSize < length)
- {
- reset();
- }
-
- if (!_datagram)
- {
- _datagram = new uint8_t[length];
- _memSize = length;
- }
-
- if (fread((unsigned short *) _datagram,1,length,fp) != length)
- {
- reset();
- return(-1);
- }
-
- _datagramLen = length;
- _receiveTime = receiveTime;
-
- if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
- {
- readNextPacket = true;
- }
- }
-
- _rtpParsed = false;
- return(packetLen);
-
-}
-
-
-int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, size_t length)
-{
- if (!fp)
- {
- return -1;
- }
-
- // check buffer size
- if (_datagram && _memSize < static_cast<int>(length))
- {
- reset();
- }
-
- if (!_datagram)
- {
- _datagram = new uint8_t[length];
- _memSize = length;
- }
-
- if (fread(_datagram, 1, length, fp) != length)
- {
- reset();
- return -1;
- }
-
- _datagramLen = length;
- _receiveTime = 0;
-
- if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
- {
- // discard this payload
- return readFromFile(fp);
- }
-
- _rtpParsed = false;
- return length;
-
-}
-
-
-int NETEQTEST_RTPpacket::writeToFile(FILE *fp)
-{
- if (!fp)
- {
- return -1;
- }
-
- uint16_t length, plen;
- uint32_t offset;
-
- // length including RTPplay header
- length = htons(_datagramLen + _kRDHeaderLen);
- if (fwrite(&length, 2, 1, fp) != 1)
- {
- return -1;
- }
-
- // payload length
- plen = htons(_datagramLen);
- if (fwrite(&plen, 2, 1, fp) != 1)
- {
- return -1;
- }
-
- // offset (=receive time)
- offset = htonl(_receiveTime);
- if (fwrite(&offset, 4, 1, fp) != 1)
- {
- return -1;
- }
-
-
- // write packet data
- if (fwrite(_datagram, 1, _datagramLen, fp) !=
- static_cast<size_t>(_datagramLen))
- {
- return -1;
- }
-
- return _datagramLen + _kRDHeaderLen; // total number of bytes written
-
-}
-
-
-void NETEQTEST_RTPpacket::blockPT(uint8_t pt)
-{
- _blockList[pt] = true;
-}
-
-
-void NETEQTEST_RTPpacket::parseHeader()
-{
- if (_rtpParsed)
- {
- // nothing to do
- return;
- }
-
- if (_datagramLen < _kBasicHeaderLen)
- {
- // corrupt packet?
- return;
- }
-
- _payloadLen = parseRTPheader(&_payloadPtr);
-
- _rtpParsed = true;
-
- return;
-
-}
-
-void NETEQTEST_RTPpacket::parseHeader(webrtc::WebRtcRTPHeader* rtp_header) {
- if (!_rtpParsed) {
- parseHeader();
- }
- if (rtp_header) {
- rtp_header->header.markerBit = _rtpInfo.header.markerBit;
- rtp_header->header.payloadType = _rtpInfo.header.payloadType;
- rtp_header->header.sequenceNumber = _rtpInfo.header.sequenceNumber;
- rtp_header->header.timestamp = _rtpInfo.header.timestamp;
- rtp_header->header.ssrc = _rtpInfo.header.ssrc;
- }
-}
-
-const webrtc::WebRtcRTPHeader* NETEQTEST_RTPpacket::RTPinfo() const
-{
- if (_rtpParsed)
- {
- return &_rtpInfo;
- }
- else
- {
- return NULL;
- }
-}
-
-uint8_t * NETEQTEST_RTPpacket::datagram() const
-{
- if (_datagramLen > 0)
- {
- return _datagram;
- }
- else
- {
- return NULL;
- }
-}
-
-uint8_t * NETEQTEST_RTPpacket::payload() const
-{
- if (_payloadLen > 0)
- {
- return _payloadPtr;
- }
- else
- {
- return NULL;
- }
-}
-
-size_t NETEQTEST_RTPpacket::payloadLen()
-{
- parseHeader();
- return _payloadLen;
-}
-
-int16_t NETEQTEST_RTPpacket::dataLen() const
-{
- return _datagramLen;
-}
-
-bool NETEQTEST_RTPpacket::isParsed() const
-{
- return _rtpParsed;
-}
-
-bool NETEQTEST_RTPpacket::isLost() const
-{
- return _lost;
-}
-
-uint8_t NETEQTEST_RTPpacket::payloadType() const
-{
- if(_datagram && _datagramLen >= _kBasicHeaderLen)
- {
- webrtc::WebRtcRTPHeader tempRTPinfo;
- parseRTPheader(&tempRTPinfo);
- return tempRTPinfo.header.payloadType;
- }
- else
- {
- return 0;
- }
-}
-
-uint16_t NETEQTEST_RTPpacket::sequenceNumber() const
-{
- if(_datagram && _datagramLen >= _kBasicHeaderLen)
- {
- webrtc::WebRtcRTPHeader tempRTPinfo;
- parseRTPheader(&tempRTPinfo);
- return tempRTPinfo.header.sequenceNumber;
- }
- else
- {
- return 0;
- }
-}
-
-uint32_t NETEQTEST_RTPpacket::timeStamp() const
-{
- if(_datagram && _datagramLen >= _kBasicHeaderLen)
- {
- webrtc::WebRtcRTPHeader tempRTPinfo;
- parseRTPheader(&tempRTPinfo);
- return tempRTPinfo.header.timestamp;
- }
- else
- {
- return 0;
- }
-}
-
-uint32_t NETEQTEST_RTPpacket::SSRC() const
-{
- if(_datagram && _datagramLen >= _kBasicHeaderLen)
- {
- webrtc::WebRtcRTPHeader tempRTPinfo;
- parseRTPheader(&tempRTPinfo);
- return tempRTPinfo.header.ssrc;
- }
- else
- {
- return 0;
- }
-}
-
-uint8_t NETEQTEST_RTPpacket::markerBit() const
-{
- if(_datagram && _datagramLen >= _kBasicHeaderLen)
- {
- webrtc::WebRtcRTPHeader tempRTPinfo;
- parseRTPheader(&tempRTPinfo);
- return tempRTPinfo.header.markerBit;
- }
- else
- {
- return 0;
- }
-}
-
-
-
-int NETEQTEST_RTPpacket::setPayloadType(uint8_t pt)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.header.payloadType = pt;
- }
-
- _datagram[1] = pt;
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setSequenceNumber(uint16_t sn)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.header.sequenceNumber = sn;
- }
-
- _datagram[2]=(unsigned char)((sn>>8)&0xFF);
- _datagram[3]=(unsigned char)((sn)&0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setTimeStamp(uint32_t ts)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.header.timestamp = ts;
- }
-
- _datagram[4]=(unsigned char)((ts>>24)&0xFF);
- _datagram[5]=(unsigned char)((ts>>16)&0xFF);
- _datagram[6]=(unsigned char)((ts>>8)&0xFF);
- _datagram[7]=(unsigned char)(ts & 0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setSSRC(uint32_t ssrc)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.header.ssrc = ssrc;
- }
-
- _datagram[8]=(unsigned char)((ssrc>>24)&0xFF);
- _datagram[9]=(unsigned char)((ssrc>>16)&0xFF);
- _datagram[10]=(unsigned char)((ssrc>>8)&0xFF);
- _datagram[11]=(unsigned char)(ssrc & 0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setMarkerBit(uint8_t mb)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (_rtpParsed)
- {
- _rtpInfo.header.markerBit = mb;
- }
-
- if (mb)
- {
- _datagram[0] |= 0x01;
- }
- else
- {
- _datagram[0] &= 0xFE;
- }
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setRTPheader(const webrtc::WebRtcRTPHeader* RTPinfo)
-{
- if (_datagramLen < 12)
- {
- // this packet is not ok
- return -1;
- }
-
- makeRTPheader(_datagram,
- RTPinfo->header.payloadType,
- RTPinfo->header.sequenceNumber,
- RTPinfo->header.timestamp,
- RTPinfo->header.ssrc,
- RTPinfo->header.markerBit);
-
- return 0;
-}
-
-
-int NETEQTEST_RTPpacket::splitStereo(NETEQTEST_RTPpacket* slaveRtp,
- enum stereoModes mode)
-{
- // if mono, do nothing
- if (mode == stereoModeMono)
- {
- return 0;
- }
-
- // check that the RTP header info is parsed
- parseHeader();
-
- // start by copying the main rtp packet
- *slaveRtp = *this;
-
- if(_payloadLen == 0)
- {
- // do no more
- return 0;
- }
-
- if(_payloadLen%2 != 0)
- {
- // length must be a factor of 2
- return -1;
- }
-
- switch(mode)
- {
- case stereoModeSample1:
- {
- // sample based codec with 1-byte samples
- splitStereoSample(slaveRtp, 1 /* 1 byte/sample */);
- break;
- }
- case stereoModeSample2:
- {
- // sample based codec with 2-byte samples
- splitStereoSample(slaveRtp, 2 /* 2 bytes/sample */);
- break;
- }
- case stereoModeFrame:
- {
- // frame based codec
- splitStereoFrame(slaveRtp);
- break;
- }
- case stereoModeDuplicate:
- {
- // frame based codec, send the whole packet to both master and slave
- splitStereoDouble(slaveRtp);
- break;
- }
- case stereoModeMono:
- {
- assert(false);
- return -1;
- }
- }
-
- return 0;
-}
-
-
-void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data,
- uint8_t payloadType,
- uint16_t seqNo,
- uint32_t timestamp,
- uint32_t ssrc,
- uint8_t markerBit) const
-{
- rtp_data[0] = markerBit ? 0x81 : 0x80;
- rtp_data[1] = payloadType;
- rtp_data[2] = seqNo >> 8;
- rtp_data[3] = seqNo & 0xFF;
- rtp_data[4] = timestamp >> 24;
- rtp_data[5] = (timestamp >> 16) & 0xFF;
- rtp_data[6] = (timestamp >> 8) & 0xFF;
- rtp_data[7] = timestamp & 0xFF;
- rtp_data[8] = ssrc >> 24;
- rtp_data[9] = (ssrc >> 16) & 0xFF;
- rtp_data[10] = (ssrc >> 8) & 0xFF;
- rtp_data[11] = ssrc & 0xFF;
-}
-
-uint16_t NETEQTEST_RTPpacket::parseRTPheader(webrtc::WebRtcRTPHeader* RTPinfo,
- uint8_t **payloadPtr) const
-{
- uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram);
- int i_P, i_X, i_CC;
-
- assert(_datagramLen >= 12);
- parseBasicHeader(RTPinfo, &i_P, &i_X, &i_CC);
-
- int i_startPosition = calcHeaderLength(i_X, i_CC);
-
- int i_padlength = calcPadLength(i_P);
-
- if (payloadPtr)
- {
- *payloadPtr =
- reinterpret_cast<uint8_t*>(&rtp_data[i_startPosition >> 1]);
- }
-
- return static_cast<uint16_t>(_datagramLen - i_startPosition - i_padlength);
-}
-
-
-void NETEQTEST_RTPpacket::parseBasicHeader(webrtc::WebRtcRTPHeader* RTPinfo,
- int *i_P, int *i_X, int *i_CC) const
-{
- uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram);
- if (_datagramLen < 12)
- {
- assert(false);
- return;
- }
-
- *i_P = (rtp_data[0] >> 5) & 0x01;
- *i_X = (rtp_data[0] >> 4) & 0x01;
- *i_CC = rtp_data[0] & 0xF;
- RTPinfo->header.markerBit = (rtp_data[0] >> 15) & 0x01;
- RTPinfo->header.payloadType = (rtp_data[0] >> 8) & 0x7F;
- RTPinfo->header.sequenceNumber =
- (rtp_data[1] >> 8) | ((rtp_data[1] & 0xFF) << 8);
- RTPinfo->header.timestamp =
- ((rtp_data[2] & 0xFF) << 24) | ((rtp_data[2] & 0xFF00) << 8) |
- (rtp_data[3] >> 8) | ((rtp_data[3] & 0xFF) << 8);
- RTPinfo->header.ssrc =
- ((rtp_data[4] & 0xFF) << 24) | ((rtp_data[4] & 0xFF00) << 8) |
- (rtp_data[5] >> 8) | ((rtp_data[5] & 0xFF) << 8);
-}
-
-int NETEQTEST_RTPpacket::calcHeaderLength(int i_X, int i_CC) const
-{
- int i_extlength = 0;
- uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram);
-
- if (i_X == 1)
- {
- // Extension header exists.
- // Find out how many int32_t it consists of.
- int offset = 7 + 2 * i_CC;
- assert(_datagramLen > 2 * offset);
- if (_datagramLen > 2 * offset)
- {
- i_extlength = 1 +
- (((rtp_data[offset]) >> 8) | ((rtp_data[offset] & 0xFF) << 8));
- }
- }
-
- return 12 + 4 * i_extlength + 4 * i_CC;
-}
-
-int NETEQTEST_RTPpacket::calcPadLength(int i_P) const
-{
- uint16_t* rtp_data = reinterpret_cast<uint16_t*>(_datagram);
- if (i_P == 1)
- {
- /* Padding exists. Find out how many bytes the padding consists of. */
- if (_datagramLen & 0x1)
- {
- /* odd number of bytes => last byte in higher byte */
- return rtp_data[_datagramLen >> 1] & 0xFF;
- }
- else
- {
- /* even number of bytes => last byte in lower byte */
- return rtp_data[(_datagramLen >> 1) - 1] >> 8;
- }
- }
- return 0;
-}
-
-void NETEQTEST_RTPpacket::splitStereoSample(NETEQTEST_RTPpacket* slaveRtp,
- int stride)
-{
- if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
- || _payloadLen == 0 || slaveRtp->_memSize < _memSize)
- {
- return;
- }
-
- uint8_t *readDataPtr = _payloadPtr;
- uint8_t *writeDataPtr = _payloadPtr;
- uint8_t *slaveData = slaveRtp->_payloadPtr;
-
- while (readDataPtr - _payloadPtr < static_cast<ptrdiff_t>(_payloadLen))
- {
- // master data
- for (int ix = 0; ix < stride; ix++) {
- *writeDataPtr = *readDataPtr;
- writeDataPtr++;
- readDataPtr++;
- }
-
- // slave data
- for (int ix = 0; ix < stride; ix++) {
- *slaveData = *readDataPtr;
- slaveData++;
- readDataPtr++;
- }
- }
-
- _payloadLen /= 2;
- slaveRtp->_payloadLen = _payloadLen;
-}
-
-
-void NETEQTEST_RTPpacket::splitStereoFrame(NETEQTEST_RTPpacket* slaveRtp)
-{
- if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
- || _payloadLen == 0 || slaveRtp->_memSize < _memSize)
- {
- return;
- }
-
- memmove(slaveRtp->_payloadPtr, _payloadPtr + _payloadLen/2, _payloadLen/2);
-
- _payloadLen /= 2;
- slaveRtp->_payloadLen = _payloadLen;
-}
-void NETEQTEST_RTPpacket::splitStereoDouble(NETEQTEST_RTPpacket* slaveRtp)
-{
- if(!_payloadPtr || !slaveRtp || !slaveRtp->_payloadPtr
- || _payloadLen == 0 || slaveRtp->_memSize < _memSize)
- {
- return;
- }
-
- memcpy(slaveRtp->_payloadPtr, _payloadPtr, _payloadLen);
- slaveRtp->_payloadLen = _payloadLen;
-}
-
-// Get the RTP header for the RED payload indicated by argument index.
-// The first RED payload is index = 0.
-int NETEQTEST_RTPpacket::extractRED(int index, webrtc::WebRtcRTPHeader& red)
-{
-//
-// 0 1 2 3
-// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |1| block PT | timestamp offset | block length |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |1| ... |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |0| block PT |
-// +-+-+-+-+-+-+-+-+
-//
-
- parseHeader();
-
- uint8_t* ptr = payload();
- uint8_t* payloadEndPtr = ptr + payloadLen();
- int num_encodings = 0;
- int total_len = 0;
-
- while ((ptr < payloadEndPtr) && (*ptr & 0x80))
- {
- int len = ((ptr[2] & 0x03) << 8) + ptr[3];
- if (num_encodings == index)
- {
- // Header found.
- red.header.payloadType = ptr[0] & 0x7F;
- uint32_t offset = (ptr[1] << 6) + (ptr[2] >> 2);
- red.header.sequenceNumber = sequenceNumber();
- red.header.timestamp = timeStamp() - offset;
- red.header.markerBit = markerBit();
- red.header.ssrc = SSRC();
- return len;
- }
- ++num_encodings;
- total_len += len;
- ptr += 4;
- }
- if ((ptr < payloadEndPtr) && (num_encodings == index))
- {
- // Last header.
- red.header.payloadType = ptr[0] & 0x7F;
- red.header.sequenceNumber = sequenceNumber();
- red.header.timestamp = timeStamp();
- red.header.markerBit = markerBit();
- red.header.ssrc = SSRC();
- ++ptr;
- return payloadLen() - (ptr - payload()) - total_len;
- }
- return -1;
-}
-
-// Randomize the payload, not the RTP header.
-void NETEQTEST_RTPpacket::scramblePayload(void)
-{
- parseHeader();
-
- for (size_t i = 0; i < _payloadLen; ++i)
- {
- _payloadPtr[i] = static_cast<uint8_t>(rand());
- }
-}
diff --git a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
deleted file mode 100644
index e4fe9c4..0000000
--- a/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * 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 "typedefs.h" // NOLINT(build/include)
-#include "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
diff --git a/modules/audio_coding/neteq/test/RTPchange.cc b/modules/audio_coding/neteq/test/RTPchange.cc
deleted file mode 100644
index 3e90002..0000000
--- a/modules/audio_coding/neteq/test/RTPchange.cc
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-
-#include <algorithm>
-#include <vector>
-
-#include "modules/audio_coding/neteq/test/NETEQTEST_DummyRTPpacket.h"
-#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-
-#define FIRSTLINELEN 40
-//#define WEBRTC_DUMMY_RTP
-
-static bool pktCmp(NETEQTEST_RTPpacket *a, NETEQTEST_RTPpacket *b) {
- return (a->time() < b->time());
-}
-
-int main(int argc, char* argv[]) {
- FILE* in_file = fopen(argv[1], "rb");
- if (!in_file) {
- printf("Cannot open input file %s\n", argv[1]);
- return -1;
- }
- printf("Input RTP file: %s\n", argv[1]);
-
- FILE* stat_file = fopen(argv[2], "rt");
- if (!stat_file) {
- printf("Cannot open timing file %s\n", argv[2]);
- return -1;
- }
- printf("Timing file: %s\n", argv[2]);
-
- FILE* out_file = fopen(argv[3], "wb");
- if (!out_file) {
- printf("Cannot open output file %s\n", argv[3]);
- return -1;
- }
- printf("Output RTP file: %s\n\n", argv[3]);
-
- // Read all statistics and insert into map.
- // Read first line.
- char temp_str[100];
- if (fgets(temp_str, 100, stat_file) == NULL) {
- printf("Failed to read timing file %s\n", argv[2]);
- return -1;
- }
- // Define map.
- std::map<std::pair<uint16_t, uint32_t>, uint32_t> packet_stats;
- uint16_t seq_no;
- uint32_t ts;
- uint32_t send_time;
-
- while (fscanf(stat_file,
- "%hu %u %u %*i %*i\n", &seq_no, &ts, &send_time) == 3) {
- std::pair<uint16_t, uint32_t>
- temp_pair = std::pair<uint16_t, uint32_t>(seq_no, ts);
-
- packet_stats[temp_pair] = send_time;
- }
-
- fclose(stat_file);
-
- // Read file header and write directly to output file.
- char first_line[FIRSTLINELEN];
- if (fgets(first_line, FIRSTLINELEN, in_file) == NULL) {
- printf("Failed to read first line of input file %s\n", argv[1]);
- return -1;
- }
- fputs(first_line, out_file);
- // start_sec + start_usec + source + port + padding
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- if (fread(first_line, 1, kRtpDumpHeaderSize, in_file)
- != kRtpDumpHeaderSize) {
- printf("Failed to read RTP dump header from input file %s\n", argv[1]);
- return -1;
- }
- if (fwrite(first_line, 1, kRtpDumpHeaderSize, out_file)
- != kRtpDumpHeaderSize) {
- printf("Failed to write RTP dump header to output file %s\n", argv[3]);
- return -1;
- }
-
- std::vector<NETEQTEST_RTPpacket *> packet_vec;
-
- while (1) {
- // Insert in vector.
-#ifdef WEBRTC_DUMMY_RTP
- NETEQTEST_RTPpacket *new_packet = new NETEQTEST_DummyRTPpacket();
-#else
- NETEQTEST_RTPpacket *new_packet = new NETEQTEST_RTPpacket();
-#endif
- if (new_packet->readFromFile(in_file) < 0) {
- // End of file.
- break;
- }
-
- // Look for new send time in statistics vector.
- std::pair<uint16_t, uint32_t> temp_pair =
- std::pair<uint16_t, uint32_t>(new_packet->sequenceNumber(),
- new_packet->timeStamp());
-
- uint32_t new_send_time = packet_stats[temp_pair];
- new_packet->setTime(new_send_time); // Set new send time.
- packet_vec.push_back(new_packet); // Insert in vector.
- }
-
- // Sort the vector according to send times.
- std::sort(packet_vec.begin(), packet_vec.end(), pktCmp);
-
- std::vector<NETEQTEST_RTPpacket *>::iterator it;
- for (it = packet_vec.begin(); it != packet_vec.end(); it++) {
- // Write to out file.
- if ((*it)->writeToFile(out_file) < 0) {
- printf("Error writing to file\n");
- return -1;
- }
- // Delete packet.
- delete *it;
- }
-
- fclose(in_file);
- fclose(out_file);
-
- return 0;
-}
diff --git a/modules/audio_coding/neteq/test/RTPjitter.cc b/modules/audio_coding/neteq/test/RTPjitter.cc
deleted file mode 100644
index ca73292..0000000
--- a/modules/audio_coding/neteq/test/RTPjitter.cc
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- * 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.
- */
-
-//TODO(hlundin): Reformat file to meet style guide.
-
-/* header includes */
-#include <float.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#ifdef WIN32
-#include <winsock2.h>
-#include <io.h>
-#endif
-#ifdef WEBRTC_LINUX
-#include <netinet/in.h>
-#endif
-
-#include <assert.h>
-
-#include "test/gtest.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-/*********************/
-/* Misc. definitions */
-/*********************/
-
-#define FIRSTLINELEN 40
-#define CHECK_NOT_NULL(a) if((a)==NULL){ \
- fprintf(stderr,"\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a ); \
- return(-1);}
-
-struct arr_time {
- float time;
- uint32_t ix;
-};
-
-int filelen(FILE *fid)
-{
- fpos_t cur_pos;
- int len;
-
- if (!fid || fgetpos(fid, &cur_pos)) {
- return(-1);
- }
-
- fseek(fid, 0, SEEK_END);
- len = ftell(fid);
-
- fsetpos(fid, &cur_pos);
-
- return (len);
-}
-
-int compare_arr_time(const void *x, const void *y);
-
-int main(int argc, char* argv[])
-{
- unsigned int dat_len, rtp_len, Npack, k;
- arr_time *time_vec;
- char firstline[FIRSTLINELEN];
- unsigned char* rtp_vec = NULL;
- unsigned char** packet_ptr = NULL;
- unsigned char* temp_packet = NULL;
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- uint16_t len;
- uint32_t *offset;
-
-/* check number of parameters */
- if (argc != 4) {
- /* print help text and exit */
- printf("Apply jitter on RTP stream.\n");
- printf("Reads an RTP stream and packet timing from two files.\n");
- printf("The RTP stream is modified to have the same jitter as described in "
- "the timing files.\n");
- printf("The format of the RTP stream file should be the same as for \n");
- printf("rtpplay, and can be obtained e.g., from Ethereal by using\n");
- printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> "
- "Save As\n\n");
- printf("Usage:\n\n");
- printf("%s RTP_infile dat_file RTP_outfile\n", argv[0]);
- printf("where:\n");
-
- printf("RTP_infile : RTP stream input file\n\n");
-
- printf("dat_file : file with packet arrival times in ms\n\n");
-
- printf("RTP_outfile : RTP stream output file\n\n");
-
- return(0);
- }
-
- FILE* in_file=fopen(argv[1],"rb");
- CHECK_NOT_NULL(in_file);
- printf("Input file: %s\n",argv[1]);
- FILE* dat_file=fopen(argv[2],"rb");
- CHECK_NOT_NULL(dat_file);
- printf("Dat-file: %s\n",argv[2]);
- FILE* out_file=fopen(argv[3],"wb");
- CHECK_NOT_NULL(out_file);
- printf("Output file: %s\n\n",argv[3]);
-
- // add 1000 bytes to avoid (rare) strange error.
- time_vec = (arr_time *) malloc(sizeof(arr_time)
- *(filelen(dat_file)/sizeof(float)) + 1000);
- if (time_vec==NULL) {
- fprintf(stderr, "Error: could not allocate memory for reading dat file\n");
- goto closing;
- }
-
- dat_len=0;
- while(fread(&(time_vec[dat_len].time),sizeof(float),1,dat_file)>0) {
- time_vec[dat_len].ix=dat_len;
- dat_len++;
- }
-
- if (dat_len == 0) {
- fprintf(stderr, "Error: dat_file is empty, no arrival time is given.\n");
- goto closing;
- }
-
- qsort(time_vec,dat_len,sizeof(arr_time),compare_arr_time);
-
-
- rtp_vec = (unsigned char *) malloc(sizeof(unsigned char)*filelen(in_file));
- if (rtp_vec==NULL) {
- fprintf(stderr,"Error: could not allocate memory for reading rtp file\n");
- goto closing;
- }
-
- // read file header and write directly to output file
- EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
- EXPECT_GT(fputs(firstline, out_file), 0);
- EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
- in_file));
- EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
- out_file));
-
- // read all RTP packets into vector
- rtp_len=0;
- Npack=0;
-
- // read length of first packet.
- len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file);
- while(len==2) {
- len = ntohs(*((uint16_t *)(rtp_vec + rtp_len)));
- rtp_len += 2;
- if(fread(&rtp_vec[rtp_len], sizeof(unsigned char),
- len-2, in_file)!=(unsigned) (len-2)) {
- fprintf(stderr,"Error: currupt packet length\n");
- goto closing;
- }
- rtp_len += len-2;
- Npack++;
-
- // read length of next packet.
- len=(uint16_t) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file);
- }
-
- if (Npack == 0) {
- fprintf(stderr, "Error: No RTP packet found.\n");
- goto closing;
- }
-
- packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
-
- packet_ptr[0]=rtp_vec;
- k=1;
- while(k<Npack) {
- len = ntohs(*((uint16_t *) packet_ptr[k-1]));
- packet_ptr[k]=packet_ptr[k-1]+len;
- k++;
- }
-
- for(k=0; k<dat_len && k<Npack; k++) {
- if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){
- temp_packet = packet_ptr[time_vec[k].ix];
- offset = (uint32_t *) (temp_packet+4);
- if ( time_vec[k].time >= 0 ) {
- *offset = htonl((uint32_t) time_vec[k].time);
- }
- else {
- *offset = htonl((uint32_t) 0);
- fprintf(stderr, "Warning: negative receive time in dat file transformed"
- " to 0.\n");
- }
-
- // write packet to file
- if (fwrite(temp_packet, sizeof(unsigned char),
- ntohs(*((uint16_t*) temp_packet)),
- out_file) !=
- ntohs(*((uint16_t*) temp_packet))) {
- return -1;
- }
- }
- }
-
-
-closing:
- free(time_vec);
- free(rtp_vec);
- if (packet_ptr != NULL) {
- free(packet_ptr);
- }
- fclose(in_file);
- fclose(dat_file);
- fclose(out_file);
-
- return(0);
-}
-
-
-
-int compare_arr_time(const void *xp, const void *yp) {
-
- if(((arr_time *)xp)->time == ((arr_time *)yp)->time)
- return(0);
- else if(((arr_time *)xp)->time > ((arr_time *)yp)->time)
- return(1);
-
- return(-1);
-}
diff --git a/modules/audio_coding/neteq/test/RTPtimeshift.cc b/modules/audio_coding/neteq/test/RTPtimeshift.cc
deleted file mode 100644
index a2429f1..0000000
--- a/modules/audio_coding/neteq/test/RTPtimeshift.cc
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * 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.
- */
-
-#include <stdio.h>
-#include <algorithm>
-#include <vector>
-
-#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-#include "test/gtest.h"
-
-#define FIRSTLINELEN 40
-
-int main(int argc, char* argv[]) {
- if (argc < 4 || argc > 6) {
- printf(
- "Usage: RTPtimeshift in.rtp out.rtp newStartTS [newStartSN "
- "[newStartArrTime]]\n");
- exit(1);
- }
-
- FILE* inFile = fopen(argv[1], "rb");
- if (!inFile) {
- printf("Cannot open input file %s\n", argv[1]);
- return (-1);
- }
- printf("Input RTP file: %s\n", argv[1]);
-
- FILE* outFile = fopen(argv[2], "wb");
- if (!outFile) {
- printf("Cannot open output file %s\n", argv[2]);
- return (-1);
- }
- printf("Output RTP file: %s\n\n", argv[2]);
-
- // Read file header and write directly to output file.
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- char firstline[FIRSTLINELEN];
- EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
- EXPECT_GT(fputs(firstline, outFile), 0);
- EXPECT_EQ(kRtpDumpHeaderSize,
- fread(firstline, 1, kRtpDumpHeaderSize, inFile));
- EXPECT_EQ(kRtpDumpHeaderSize,
- fwrite(firstline, 1, kRtpDumpHeaderSize, outFile));
- NETEQTEST_RTPpacket packet;
- int packLen = packet.readFromFile(inFile);
- if (packLen < 0) {
- exit(1);
- }
-
- // Get new start TS and start SeqNo from arguments.
- uint32_t TSdiff = atoi(argv[3]) - packet.timeStamp();
- uint16_t SNdiff = 0;
- uint32_t ATdiff = 0;
- if (argc > 4) {
- int startSN = atoi(argv[4]);
- if (startSN >= 0)
- SNdiff = startSN - packet.sequenceNumber();
- if (argc > 5) {
- int startTS = atoi(argv[5]);
- if (startTS >= 0)
- ATdiff = startTS - packet.time();
- }
- }
-
- while (packLen >= 0) {
- packet.setTimeStamp(packet.timeStamp() + TSdiff);
- packet.setSequenceNumber(packet.sequenceNumber() + SNdiff);
- packet.setTime(packet.time() + ATdiff);
-
- packet.writeToFile(outFile);
-
- packLen = packet.readFromFile(inFile);
- }
-
- fclose(inFile);
- fclose(outFile);
-
- return 0;
-}
diff --git a/modules/audio_coding/neteq/tools/rtp_jitter.cc b/modules/audio_coding/neteq/tools/rtp_jitter.cc
new file mode 100644
index 0000000..d92fed0
--- /dev/null
+++ b/modules/audio_coding/neteq/tools/rtp_jitter.cc
@@ -0,0 +1,152 @@
+/*
+ * 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.
+ */
+
+#include <stdio.h>
+#include <algorithm>
+#include <fstream>
+#include <iostream>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/rtp_rtcp/source/byte_io.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/flags.h"
+#include "typedefs.h" // NOLINT(build/include)
+
+namespace webrtc {
+namespace test {
+namespace {
+
+DEFINE_bool(help, false, "Print help message");
+
+constexpr size_t kRtpDumpHeaderLength = 8;
+
+// Returns the next packet or an empty buffer if end of file was encountered.
+rtc::Buffer ReadNextPacket(FILE* file) {
+ // Read the rtpdump header for the next packet.
+ rtc::Buffer buffer;
+ buffer.SetData(kRtpDumpHeaderLength, [&](rtc::ArrayView<uint8_t> x) {
+ return fread(x.data(), 1, x.size(), file);
+ });
+ if (buffer.size() != kRtpDumpHeaderLength) {
+ return rtc::Buffer();
+ }
+
+ // Get length field. This is the total length for this packet written to file,
+ // including the kRtpDumpHeaderLength bytes already read.
+ const uint16_t len = ByteReader<uint16_t>::ReadBigEndian(buffer.data());
+ RTC_CHECK_GE(len, kRtpDumpHeaderLength);
+
+ // Read remaining data from file directly into buffer.
+ buffer.AppendData(len - kRtpDumpHeaderLength, [&](rtc::ArrayView<uint8_t> x) {
+ return fread(x.data(), 1, x.size(), file);
+ });
+ if (buffer.size() != len) {
+ buffer.Clear();
+ }
+ return buffer;
+}
+
+struct PacketAndTime {
+ rtc::Buffer packet;
+ int time;
+};
+
+void WritePacket(const PacketAndTime& packet, FILE* file) {
+ // Write the first 4 bytes from the original packet.
+ const auto* payload_ptr = packet.packet.data();
+ RTC_CHECK_EQ(fwrite(payload_ptr, 4, 1, file), 1);
+ payload_ptr += 4;
+
+ // Convert the new time offset to network endian, and write to file.
+ uint8_t time[sizeof(uint32_t)];
+ ByteWriter<uint32_t, sizeof(uint32_t)>::WriteBigEndian(time, packet.time);
+ RTC_CHECK_EQ(fwrite(time, sizeof(uint32_t), 1, file), 1);
+ payload_ptr += 4; // Skip the old time in the original payload.
+
+ // Write the remaining part of the payload.
+ RTC_DCHECK_EQ(payload_ptr - packet.packet.data(), kRtpDumpHeaderLength);
+ RTC_CHECK_EQ(
+ fwrite(payload_ptr, packet.packet.size() - kRtpDumpHeaderLength, 1, file),
+ 1);
+}
+
+int RunRtpJitter(int argc, char* argv[]) {
+ const std::string program_name = argv[0];
+ const std::string usage =
+ "Tool for alternating the arrival times in an RTP dump file.\n"
+ "Example usage:\n" +
+ program_name + " input.rtp arrival_times_ms.txt output.rtp\n\n";
+ if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
+ argc != 4) {
+ printf("%s", usage.c_str());
+ return FLAG_help ? 0 : 1;
+ }
+
+ printf("Input RTP file: %s\n", argv[1]);
+ FILE* in_file = fopen(argv[1], "rb");
+ RTC_CHECK(in_file) << "Could not open file " << argv[1] << " for reading";
+ printf("Timing file: %s\n", argv[2]);
+ std::ifstream timing_file(argv[2]);
+ printf("Output file: %s\n", argv[3]);
+ FILE* out_file = fopen(argv[3], "wb");
+ RTC_CHECK(out_file) << "Could not open file " << argv[2] << " for writing";
+
+ // Copy the RTP file header to the output file.
+ char header_string[30];
+ RTC_CHECK(fgets(header_string, 30, in_file));
+ fprintf(out_file, "%s", header_string);
+ uint8_t file_header[16];
+ RTC_CHECK_EQ(fread(file_header, sizeof(file_header), 1, in_file), 1);
+ RTC_CHECK_EQ(fwrite(file_header, sizeof(file_header), 1, out_file), 1);
+
+ // Read all time values from the timing file. Store in a vector.
+ std::vector<int> new_arrival_times;
+ int new_time;
+ while (timing_file >> new_time) {
+ new_arrival_times.push_back(new_time);
+ }
+
+ // Read all packets from the input RTP file, but no more than the number of
+ // new time values. Store RTP packets together with new time values.
+ auto time_it = new_arrival_times.begin();
+ std::vector<PacketAndTime> packets;
+ while (1) {
+ auto packet = ReadNextPacket(in_file);
+ if (packet.empty() || time_it == new_arrival_times.end()) {
+ break;
+ }
+ packets.push_back({std::move(packet), *time_it});
+ ++time_it;
+ }
+
+ // Sort on new time values.
+ std::sort(packets.begin(), packets.end(),
+ [](const PacketAndTime& a, const PacketAndTime& b) {
+ return a.time < b.time;
+ });
+
+ // Write packets to output file.
+ for (const auto& p : packets) {
+ WritePacket(p, out_file);
+ }
+
+ fclose(in_file);
+ fclose(out_file);
+ return 0;
+}
+
+} // namespace
+} // namespace test
+} // namespace webrtc
+
+int main(int argc, char* argv[]) {
+ return webrtc::test::RunRtpJitter(argc, argv);
+}
