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Adding regression test to NetEQ

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The test inputs RTP packets from an RTPdump file into NetEQ
and compares the output to the corresponding reference file.
Test files are included.

The change also includes a new method in NETEQTEST_RTPpacket
class, which reads past the initial file header in an RTPdump
file.

Finally, a few warnings are removed.
Review URL: http://webrtc-codereview.appspot.com/138012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@568 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org 2011-09-09 08:01:16 +00:00
parent c273019768
commit 35dcc23110
8 changed files with 258 additions and 64 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

152
src/modules/audio_coding/NetEQ/main/source/neteq_api_unittest.cc Normal file
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@ -0,0 +1,152 @@
/*
* 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.
*/
/*
* This file includes unit tests for NetEQ.
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h> // memset
#include <vector>
#include "gtest/gtest.h"
#include "modules/audio_coding/NetEQ/main/test/NETEQTEST_CodecClass.h"
#include "modules/audio_coding/NetEQ/main/test/NETEQTEST_NetEQClass.h"
#include "modules/audio_coding/NetEQ/main/test/NETEQTEST_RTPpacket.h"
#include "typedefs.h" // NOLINT(build/include)
#include "modules/audio_coding/NetEQ/main/interface/webrtc_neteq.h"
#include "modules/audio_coding/NetEQ/main/interface/webrtc_neteq_help_macros.h"
namespace {
class NetEqDecodingTest : public ::testing::Test {
protected:
NetEqDecodingTest();
virtual void SetUp();
virtual void TearDown();
void SelectDecoders(WebRtcNetEQDecoder* used_codec);
void LoadDecoders();
void DecodeAndCompare(const char* rtp_file, const char* ref_file);
NETEQTEST_NetEQClass* neteq_inst_;
std::vector<NETEQTEST_Decoder*> dec_;
};
NetEqDecodingTest::NetEqDecodingTest() : neteq_inst_(NULL) {}
void NetEqDecodingTest::SetUp() {
WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1];
SelectDecoders(usedCodec);
neteq_inst_ = new NETEQTEST_NetEQClass(usedCodec, dec_.size(), 8000,
kTCPLargeJitter);
ASSERT_TRUE(neteq_inst_);
LoadDecoders();
}
void NetEqDecodingTest::TearDown() {
if (neteq_inst_)
delete neteq_inst_;
for (size_t i = 0; i < dec_.size(); ++i) {
if (dec_[i])
delete dec_[i];
}
}
void NetEqDecodingTest::SelectDecoders(WebRtcNetEQDecoder* used_codec) {
*used_codec++ = kDecoderPCMu;
dec_.push_back(new decoder_PCMU(0));
*used_codec++ = kDecoderPCMa;
dec_.push_back(new decoder_PCMA(8));
*used_codec++ = kDecoderILBC;
dec_.push_back(new decoder_ILBC(102));
*used_codec++ = kDecoderISAC;
dec_.push_back(new decoder_iSAC(103));
*used_codec++ = kDecoderISACswb;
dec_.push_back(new decoder_iSACSWB(104));
*used_codec++ = kDecoderPCM16B;
dec_.push_back(new decoder_PCM16B_NB(93));
*used_codec++ = kDecoderPCM16Bwb;
dec_.push_back(new decoder_PCM16B_WB(94));
*used_codec++ = kDecoderPCM16Bswb32kHz;
dec_.push_back(new decoder_PCM16B_SWB32(95));
*used_codec++ = kDecoderCNG;
dec_.push_back(new decoder_CNG(13));
}
void NetEqDecodingTest::LoadDecoders() {
for (size_t i = 0; i < dec_.size(); ++i) {
ASSERT_EQ(0, dec_[i]->loadToNetEQ(*neteq_inst_));
}
}
void NetEqDecodingTest::DecodeAndCompare(const char* rtp_file,
const char* ref_file) {
NETEQTEST_RTPpacket rtp;
FILE* rtp_fp = fopen(rtp_file, "rb");
ASSERT_TRUE(rtp_fp != NULL);
ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp));
ASSERT_GT(rtp.readFromFile(rtp_fp), 0);
FILE* ref_fp = fopen(ref_file, "rb");
ASSERT_TRUE(ref_fp != NULL);
unsigned int sim_clock = 0;
const int kTimeStep = 10;
while (rtp.dataLen() >= 0) {
// Check if time to receive.
while ((sim_clock >= rtp.time()) &&
(rtp.dataLen() >= 0)) {
if (rtp.dataLen() > 0) {
ASSERT_EQ(0, neteq_inst_->recIn(rtp));
}
// Get next packet.
ASSERT_NE(-1, rtp.readFromFile(rtp_fp));
}
// RecOut
WebRtc_Word16 out_data[10 * 32]; // 10 ms at 32 kHz
WebRtc_Word16 out_len = neteq_inst_->recOut(out_data);
ASSERT_TRUE((out_len == 80) || (out_len == 160) || (out_len == 320));
// Read from ref file
WebRtc_Word16 ref_data[10 * 32]; // 10 ms at 32 kHz
if (static_cast<size_t>(out_len) !=
fread(ref_data, sizeof(WebRtc_Word16), out_len, ref_fp)) {
break;
}
// Compare
EXPECT_EQ(0, memcmp(out_data, ref_data, sizeof(WebRtc_Word16) * out_len));
// Increase time
sim_clock += kTimeStep;
}
ASSERT_NE(0, feof(ref_fp)); // Make sure that we reached the end.
fclose(rtp_fp);
fclose(ref_fp);
}
TEST_F(NetEqDecodingTest, TestBitExactness) {
DecodeAndCompare("test/data/audio_coding/universal.rtp",
"test/data/audio_coding/universal_ref.pcm");
}
} // namespace
int main(int argc, char** argv) {
::testing::InitGoogleTest(&argc, argv);
return RUN_ALL_TESTS();
}

33
src/modules/audio_coding/NetEQ/main/source/neteq_tests.gyp Normal file
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@ -0,0 +1,33 @@
# 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.
{
'includes': [
'../../../../../common_settings.gypi',
],
'targets': [
{
'target_name': 'neteq_unittest',
'type': 'executable',
'dependencies': [
'neteq.gyp:NetEq',
'neteq.gyp:NetEqTestTools',
'../../../../../../testing/gtest.gyp:gtest',
],
'sources': [
'neteq_api_unittest.cc',
],
},
],
}
# Local Variables:
# tab-width:2
# indent-tabs-mode:nil
# End:
# vim: set expandtab tabstop=2 shiftwidth=2:

2
src/modules/audio_coding/NetEQ/main/test/NETEQTEST_NetEQClass.cc
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@ -210,7 +210,7 @@ void NETEQTEST_NetEQClass::printError(NETEQTEST_RTPpacket &rtp)
printError();
// print extra info from packet
printf("\tRTP: TS=%lu, SN=%u, PT=%u, M=%i, len=%i\n",
printf("\tRTP: TS=%u, SN=%u, PT=%u, M=%i, len=%i\n",
rtp.timeStamp(), rtp.sequenceNumber(), rtp.payloadType(),
rtp.markerBit(), rtp.payloadLen());

78
src/modules/audio_coding/NetEQ/main/test/NETEQTEST_RTPpacket.cc
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@ -10,6 +10,7 @@
#include "NETEQTEST_RTPpacket.h"
#include <assert.h>
#include <string.h>
#ifdef WIN32
@ -72,7 +73,7 @@ NETEQTEST_RTPpacket & NETEQTEST_RTPpacket::operator = (const NETEQTEST_RTPpacket
// deallocate datagram memory if allocated
if(_datagram)
{
delete[] _datagram;
delete [] _datagram;
}
// do shallow copy
@ -113,7 +114,7 @@ NETEQTEST_RTPpacket::~NETEQTEST_RTPpacket()
{
if(_datagram)
{
delete _datagram;
delete [] _datagram;
}
}
@ -121,7 +122,7 @@ NETEQTEST_RTPpacket::~NETEQTEST_RTPpacket()
void NETEQTEST_RTPpacket::reset()
{
if(_datagram) {
delete _datagram;
delete [] _datagram;
}
_datagram = NULL;
_memSize = 0;
@ -134,6 +135,37 @@ void NETEQTEST_RTPpacket::reset()
}
int NETEQTEST_RTPpacket::skipFileHeader(FILE *fp)
{
if (!fp) {
return -1;
}
const int kFirstLineLength = 40;
char firstline[kFirstLineLength];
fgets(firstline, kFirstLineLength, fp);
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)
{
@ -201,15 +233,15 @@ int NETEQTEST_RTPpacket::readFromFile(FILE *fp)
}
int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, int length)
int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, size_t length)
{
if(!fp)
if (!fp)
{
return(-1);
return -1;
}
// check buffer size
if (_datagram && _memSize < length)
if (_datagram && _memSize < static_cast<int>(length))
{
reset();
}
@ -220,10 +252,10 @@ int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, int length)
_memSize = length;
}
if (fread((unsigned short *) _datagram,1,length,fp) != length)
if (fread(_datagram, 1, length, fp) != length)
{
reset();
return(-1);
return -1;
}
_datagramLen = length;
@ -232,53 +264,54 @@ int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, int length)
if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
{
// discard this payload
return(readFromFile(fp));
return readFromFile(fp);
}
return(length);
return length;
}
int NETEQTEST_RTPpacket::writeToFile(FILE *fp)
{
if(!fp)
if (!fp)
{
return(-1);
return -1;
}
WebRtc_UWord16 length, plen;
WebRtc_UWord16 length, plen;
WebRtc_UWord32 offset;
// length including RTPplay header
length = htons(_datagramLen + HDR_SIZE);
if (fwrite(&length, 2, 1, fp) != 1)
{
return(-1);
return -1;
}
// payload length
plen = htons(_datagramLen);
if (fwrite(&plen, 2, 1, fp) != 1)
{
return(-1);
return -1;
}
// offset (=receive time)
offset = htonl(_receiveTime);
if (fwrite(&offset, 4, 1, fp) != 1)
{
return(-1);
return -1;
}
// write packet data
if (fwrite((unsigned short *) _datagram, 1, _datagramLen, fp) != _datagramLen)
if (fwrite(_datagram, 1, _datagramLen, fp) !=
static_cast<size_t>(_datagramLen))
{
return(-1);
return -1;
}
return(_datagramLen + HDR_SIZE); // total number of bytes written
return _datagramLen + HDR_SIZE; // total number of bytes written
}
@ -634,6 +667,11 @@ int NETEQTEST_RTPpacket::splitStereo(NETEQTEST_RTPpacket& slaveRtp, enum stereoM
splitStereoFrame(slaveRtp);
break;
}
case stereoModeMono:
{
assert(false);
return -1;
}
}
return 0;

3
src/modules/audio_coding/NetEQ/main/test/NETEQTEST_RTPpacket.h
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@ -32,8 +32,9 @@ public:
bool operator !() const { return (dataLen() < 0); };
~NETEQTEST_RTPpacket();
void reset();
static int skipFileHeader(FILE *fp);
int readFromFile(FILE *fp);
int readFixedFromFile(FILE *fp, int len);
int readFixedFromFile(FILE *fp, size_t len);
int writeToFile(FILE *fp);
void blockPT(WebRtc_UWord8 pt);
//WebRtc_Word16 payloadType();

54
src/modules/audio_coding/NetEQ/main/test/NetEqRTPplay.cc
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@ -150,7 +150,6 @@ int main(int argc, char* argv[])
std::vector<NETEQTEST_NetEQClass *> NetEQvector;
NETEQTEST_RTPpacket rtp;
char version[20];
char firstline[FIRSTLINELEN];
NETEQTEST_RTPpacket slaveRtp;
//bool switchMS = false;
@ -359,40 +358,11 @@ int main(int argc, char* argv[])
/* read RTP file header */
if (!rtpOnly)
{
fgets(firstline, FIRSTLINELEN, in_file);
if(strncmp(firstline,"#!rtpplay",9) == 0) {
if(strncmp(firstline,"#!rtpplay1.0",12) != 0){
printf("ERROR: wrong rtpplay version, must be 1.0\n");
exit(0);
}
if (NETEQTEST_RTPpacket::skipFileHeader(in_file) != 0)
{
fprintf(stderr, "Wrong format in RTP file.\n");
return -1;
}
else if (strncmp(firstline,"#!RTPencode",11) == 0) {
if(strncmp(firstline,"#!RTPencode1.0",14) != 0){
printf("ERROR: wrong RTPencode version, must be 1.0\n");
exit(0);
}
}
else {
printf("ERROR: wrong file format of input file\n");
exit(0);
}
WebRtc_UWord32 start_sec;
WebRtc_UWord32 start_usec;
WebRtc_UWord32 source;
WebRtc_UWord16 port;
WebRtc_UWord16 padding;
fread(&start_sec, 4, 1, in_file);
start_sec=ntohl(start_sec);
fread(&start_usec, 4, 1, in_file);
start_usec=ntohl(start_usec);
fread(&source, 4, 1, in_file);
source=ntohl(source);
fread(&port, 2, 1, in_file);
port=ntohs(port);
fread(&padding, 2, 1, in_file);
padding=ntohs(padding);
}
/* check payload type for first speech packet */
@ -683,14 +653,14 @@ int main(int argc, char* argv[])
WebRtcNetEQ_GetJitterStatistics(NetEQvector[0]->instance(), &jitterStats);
printf("\nPost-call statistics:\n");
printf(" Call duration ms : %lu\n", simClock-start_clock);
printf(" Expand (voice) ms : %lu \t(%.2f%%)\n", jitterStats.interpolatedVoiceMs, (float) 100.0 * jitterStats.interpolatedVoiceMs/(simClock-start_clock));
printf(" Expand (silence) ms : %lu \t(%.2f%%)\n", jitterStats.interpolatedSilentMs, (float) 100.0 * jitterStats.interpolatedSilentMs/(simClock-start_clock));
printf(" Accelerate ms : %lu \t(%.2f%%)\n", jitterStats.accelerateMs, (float) 100.0 * jitterStats.accelerateMs/(simClock-start_clock));
printf(" Flushed ms : %lu \t(%.2f%%)\n", jitterStats.flushedMs, (float) 100.0 * jitterStats.flushedMs/(simClock-start_clock));
printf(" JB avg size ms : %lu\n", jitterStats.jbAvgSize);
printf(" JB max size ms : %lu\n", jitterStats.jbMaxSize);
printf(" Max inter-arrival ms: %lu\n", jitterStats.longestIATms);
printf(" Call duration ms : %u\n", simClock-start_clock);
printf(" Expand (voice) ms : %u \t(%.2f%%)\n", jitterStats.interpolatedVoiceMs, (float) 100.0 * jitterStats.interpolatedVoiceMs/(simClock-start_clock));
printf(" Expand (silence) ms : %u \t(%.2f%%)\n", jitterStats.interpolatedSilentMs, (float) 100.0 * jitterStats.interpolatedSilentMs/(simClock-start_clock));
printf(" Accelerate ms : %u \t(%.2f%%)\n", jitterStats.accelerateMs, (float) 100.0 * jitterStats.accelerateMs/(simClock-start_clock));
printf(" Flushed ms : %u \t(%.2f%%)\n", jitterStats.flushedMs, (float) 100.0 * jitterStats.flushedMs/(simClock-start_clock));
printf(" JB avg size ms : %u\n", jitterStats.jbAvgSize);
printf(" JB max size ms : %u\n", jitterStats.jbMaxSize);
printf(" Max inter-arrival ms: %u\n", jitterStats.longestIATms);
printf("\nComplexity estimates (including sub-components):\n");
printf(" RecIn complexity : %.2f MCPS\n", NetEQvector[0]->getRecInTime() / ((float) 1000*(simClock-start_clock)));

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test/data/audio_coding/universal.rtp Normal file
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test/data/audio_coding/universal_ref.pcm Normal file
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