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1e8c93c953be8ac0772bb6ddbde9cf5f398f8246
platform-external-webrtc/webrtc/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
henrik.lundin@webrtc.org 6f6ba6edee Fix issues with sequence number wrap-around in jitter statistics 
Wrap-arounds in sequence numbers (and in timestamps) were not always
treated correctly. This is fixed by introducing two helper functions
for correct comparisons, and by casting to the right word size.

Also added a new member variable to AutomodeInst_t. The new member keeps
track of when the first packet has been registered in the automode code.
This was previously done implicitly (and not very good) using the fact
that the lastSeqNo and lastTimestamp members were initialized to zero.

Two new unit tests were added as part of this CL.
NetEqDecodingTest.SequenceNumberWrap was failing before the fixes were
made; now it is ok.

BUG=2654
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5150 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:17:29 +00:00

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29 KiB
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/*
* 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 "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq.h"
#include <stdlib.h>
#include <string.h> // memset
#include <set>
#include <sstream>
#include <string>
#include <vector>
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
#include "webrtc/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h"
#include "webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
#include "webrtc/modules/audio_coding/neteq4/tools/input_audio_file.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RefFiles {
public:
RefFiles(const std::string& input_file, const std::string& output_file);
~RefFiles();
template<class T> void ProcessReference(const T& test_results);
template<typename T, size_t n> void ProcessReference(
const T (&test_results)[n],
size_t length);
template<typename T, size_t n> void WriteToFile(
const T (&test_results)[n],
size_t length);
template<typename T, size_t n> void ReadFromFileAndCompare(
const T (&test_results)[n],
size_t length);
void WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats);
void ReadFromFileAndCompare(const WebRtcNetEQ_NetworkStatistics& stats);
void WriteToFile(const WebRtcNetEQ_RTCPStat& stats);
void ReadFromFileAndCompare(const WebRtcNetEQ_RTCPStat& stats);
FILE* input_fp_;
FILE* output_fp_;
};
RefFiles::RefFiles(const std::string &input_file,
const std::string &output_file)
: input_fp_(NULL),
output_fp_(NULL) {
if (!input_file.empty()) {
input_fp_ = fopen(input_file.c_str(), "rb");
EXPECT_TRUE(input_fp_ != NULL);
}
if (!output_file.empty()) {
output_fp_ = fopen(output_file.c_str(), "wb");
EXPECT_TRUE(output_fp_ != NULL);
}
}
RefFiles::~RefFiles() {
if (input_fp_) {
EXPECT_EQ(EOF, fgetc(input_fp_)); // Make sure that we reached the end.
fclose(input_fp_);
}
if (output_fp_) fclose(output_fp_);
}
template<class T>
void RefFiles::ProcessReference(const T& test_results) {
WriteToFile(test_results);
ReadFromFileAndCompare(test_results);
}
template<typename T, size_t n>
void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) {
WriteToFile(test_results, length);
ReadFromFileAndCompare(test_results, length);
}
template<typename T, size_t n>
void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) {
if (output_fp_) {
ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_));
}
}
template<typename T, size_t n>
void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
size_t length) {
if (input_fp_) {
// Read from ref file.
T* ref = new T[length];
ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_));
// Compare
ASSERT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length));
delete [] ref;
}
}
void RefFiles::WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats) {
if (output_fp_) {
ASSERT_EQ(1u, fwrite(&stats, sizeof(WebRtcNetEQ_NetworkStatistics), 1,
output_fp_));
}
}
void RefFiles::ReadFromFileAndCompare(
const WebRtcNetEQ_NetworkStatistics& stats) {
if (input_fp_) {
// Read from ref file.
size_t stat_size = sizeof(WebRtcNetEQ_NetworkStatistics);
WebRtcNetEQ_NetworkStatistics ref_stats;
ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
// Compare
EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size));
}
}
void RefFiles::WriteToFile(const WebRtcNetEQ_RTCPStat& stats) {
if (output_fp_) {
ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.cum_lost), sizeof(stats.cum_lost), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.ext_max), sizeof(stats.ext_max), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
output_fp_));
}
}
void RefFiles::ReadFromFileAndCompare(
const WebRtcNetEQ_RTCPStat& stats) {
if (input_fp_) {
// Read from ref file.
WebRtcNetEQ_RTCPStat ref_stats;
ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
sizeof(ref_stats.fraction_lost), 1, input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.cum_lost), sizeof(ref_stats.cum_lost), 1,
input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.ext_max), sizeof(ref_stats.ext_max), 1,
input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
input_fp_));
// Compare
EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
EXPECT_EQ(ref_stats.cum_lost, stats.cum_lost);
EXPECT_EQ(ref_stats.ext_max, stats.ext_max);
EXPECT_EQ(ref_stats.jitter, stats.jitter);
}
}
class NetEqDecodingTest : public ::testing::Test {
protected:
// NetEQ must be polled for data once every 10 ms. Thus, neither of the
// constants below can be changed.
static const int kTimeStepMs = 10;
static const int kBlockSize8kHz = kTimeStepMs * 8;
static const int kBlockSize16kHz = kTimeStepMs * 16;
static const int kBlockSize32kHz = kTimeStepMs * 32;
static const int kMaxBlockSize = kBlockSize32kHz;
NetEqDecodingTest();
virtual void SetUp();
virtual void TearDown();
void SelectDecoders(WebRtcNetEQDecoder* used_codec);
void LoadDecoders();
void OpenInputFile(const std::string &rtp_file);
void Process(NETEQTEST_RTPpacket* rtp_ptr, int16_t* out_len);
void DecodeAndCompare(const std::string &rtp_file,
const std::string &ref_file);
void DecodeAndCheckStats(const std::string &rtp_file,
const std::string &stat_ref_file,
const std::string &rtcp_ref_file);
static void PopulateRtpInfo(int frame_index,
int timestamp,
WebRtcNetEQ_RTPInfo* rtp_info);
static void PopulateCng(int frame_index,
int timestamp,
WebRtcNetEQ_RTPInfo* rtp_info,
uint8_t* payload,
int* payload_len);
void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers);
NETEQTEST_NetEQClass* neteq_inst_;
std::vector<NETEQTEST_Decoder*> dec_;
FILE* rtp_fp_;
unsigned int sim_clock_;
int16_t out_data_[kMaxBlockSize];
};
NetEqDecodingTest::NetEqDecodingTest()
: neteq_inst_(NULL),
rtp_fp_(NULL),
sim_clock_(0) {
memset(out_data_, 0, sizeof(out_data_));
}
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];
}
if (rtp_fp_)
fclose(rtp_fp_);
}
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++ = kDecoderISACfb;
dec_.push_back(new decoder_iSACFB(105));
*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, 8000));
*used_codec++ = kDecoderCNG;
dec_.push_back(new decoder_CNG(98, 16000));
}
void NetEqDecodingTest::LoadDecoders() {
for (size_t i = 0; i < dec_.size(); ++i) {
ASSERT_EQ(0, dec_[i]->loadToNetEQ(*neteq_inst_));
}
}
void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
rtp_fp_ = fopen(rtp_file.c_str(), "rb");
ASSERT_TRUE(rtp_fp_ != NULL);
ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp_));
}
void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int16_t* out_len) {
// 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
*out_len = neteq_inst_->recOut(out_data_);
ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
(*out_len == kBlockSize16kHz) ||
(*out_len == kBlockSize32kHz));
// Increase time.
sim_clock_ += kTimeStepMs;
}
void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
const std::string &ref_file) {
OpenInputFile(rtp_file);
std::string ref_out_file = "";
if (ref_file.empty()) {
ref_out_file = webrtc::test::OutputPath() + "neteq_out.pcm";
}
RefFiles ref_files(ref_file, ref_out_file);
NETEQTEST_RTPpacket rtp;
ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
int i = 0;
while (rtp.dataLen() >= 0) {
std::ostringstream ss;
ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
int16_t out_len;
ASSERT_NO_FATAL_FAILURE(Process(&rtp, &out_len));
ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, out_len));
}
}
void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file,
const std::string &stat_ref_file,
const std::string &rtcp_ref_file) {
OpenInputFile(rtp_file);
std::string stat_out_file = "";
if (stat_ref_file.empty()) {
stat_out_file = webrtc::test::OutputPath() +
"neteq_network_stats.dat";
}
RefFiles network_stat_files(stat_ref_file, stat_out_file);
std::string rtcp_out_file = "";
if (rtcp_ref_file.empty()) {
rtcp_out_file = webrtc::test::OutputPath() +
"neteq_rtcp_stats.dat";
}
RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);
NETEQTEST_RTPpacket rtp;
ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
while (rtp.dataLen() >= 0) {
int16_t out_len;
Process(&rtp, &out_len);
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
// Process NetworkStatistics.
WebRtcNetEQ_NetworkStatistics network_stats;
ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
&network_stats));
network_stat_files.ProcessReference(network_stats);
// Process RTCPstat.
WebRtcNetEQ_RTCPStat rtcp_stats;
ASSERT_EQ(0, WebRtcNetEQ_GetRTCPStats(neteq_inst_->instance(),
&rtcp_stats));
rtcp_stat_files.ProcessReference(rtcp_stats);
}
}
}
void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
int timestamp,
WebRtcNetEQ_RTPInfo* rtp_info) {
rtp_info->sequenceNumber = frame_index;
rtp_info->timeStamp = timestamp;
rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC.
rtp_info->payloadType = 94; // PCM16b WB codec.
rtp_info->markerBit = 0;
}
void NetEqDecodingTest::PopulateCng(int frame_index,
int timestamp,
WebRtcNetEQ_RTPInfo* rtp_info,
uint8_t* payload,
int* payload_len) {
rtp_info->sequenceNumber = frame_index;
rtp_info->timeStamp = timestamp;
rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC.
rtp_info->payloadType = 98; // WB CNG.
rtp_info->markerBit = 0;
payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
*payload_len = 1; // Only noise level, no spectral parameters.
}
#if (defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)) || defined(WEBRTC_ANDROID)
// Disabled for Windows 64-bit until webrtc:1460 is fixed.
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
#else
#define MAYBE_TestBitExactness TestBitExactness
#endif
TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
// For Visual Studio 2012 and later, we will have to use the generic reference
// file, rather than the windows-specific one.
const std::string kInputRefFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_ref.pcm";
#else
const std::string kInputRefFile =
webrtc::test::ResourcePath("audio_coding/neteq_universal_ref", "pcm");
#endif
DecodeAndCompare(kInputRtpFile, kInputRefFile);
}
TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal.rtp";
#if defined(_MSC_VER) && (_MSC_VER >= 1700)
// For Visual Studio 2012 and later, we will have to use the generic reference
// file, rather than the windows-specific one.
const std::string kNetworkStatRefFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_network_stats.dat";
#else
const std::string kNetworkStatRefFile =
webrtc::test::ResourcePath("audio_coding/neteq_network_stats", "dat");
#endif
const std::string kRtcpStatRefFile =
webrtc::test::ResourcePath("audio_coding/neteq_rtcp_stats", "dat");
DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
}
TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
// Use fax mode to avoid time-scaling. This is to simplify the testing of
// packet waiting times in the packet buffer.
ASSERT_EQ(0,
WebRtcNetEQ_SetPlayoutMode(neteq_inst_->instance(), kPlayoutFax));
// Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
int num_frames = 30;
const int kSamples = 10 * 16;
const int kPayloadBytes = kSamples * 2;
for (int i = 0; i < num_frames; ++i) {
uint16_t payload[kSamples] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
rtp_info.sequenceNumber = i;
rtp_info.timeStamp = i * kSamples;
rtp_info.SSRC = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info,
reinterpret_cast<uint8_t*>(payload),
kPayloadBytes, 0));
}
// Pull out all data.
for (int i = 0; i < num_frames; ++i) {
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
const int kVecLen = 110; // More than kLenWaitingTimes in mcu.h.
int waiting_times[kVecLen];
int len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
kVecLen, waiting_times);
EXPECT_EQ(num_frames, len);
// Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
// spacing (per definition), we expect the delay to increase with 10 ms for
// each packet.
for (int i = 0; i < len; ++i) {
EXPECT_EQ((i + 1) * 10, waiting_times[i]);
}
// Check statistics again and make sure it's been reset.
EXPECT_EQ(0, WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
kVecLen, waiting_times));
// Process > 100 frames, and make sure that that we get statistics
// only for 100 frames. Note the new SSRC, causing NetEQ to reset.
num_frames = 110;
for (int i = 0; i < num_frames; ++i) {
uint16_t payload[kSamples] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
rtp_info.sequenceNumber = i;
rtp_info.timeStamp = i * kSamples;
rtp_info.SSRC = 0x1235; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info,
reinterpret_cast<uint8_t*>(payload),
kPayloadBytes, 0));
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
kVecLen, waiting_times);
EXPECT_EQ(100, len);
}
TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
const int kNumFrames = 3000; // Needed for convergence.
int frame_index = 0;
const int kSamples = 10 * 16;
const int kPayloadBytes = kSamples * 2;
while (frame_index < kNumFrames) {
// Insert one packet each time, except every 10th time where we insert two
// packets at once. This will create a negative clock-drift of approx. 10%.
int num_packets = (frame_index % 10 == 0 ? 2 : 1);
for (int n = 0; n < num_packets; ++n) {
uint8_t payload[kPayloadBytes] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
kPayloadBytes, 0));
++frame_index;
}
// Pull out data once.
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
WebRtcNetEQ_NetworkStatistics network_stats;
ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
&network_stats));
EXPECT_EQ(-103196, network_stats.clockDriftPPM);
}
TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
const int kNumFrames = 5000; // Needed for convergence.
int frame_index = 0;
const int kSamples = 10 * 16;
const int kPayloadBytes = kSamples * 2;
for (int i = 0; i < kNumFrames; ++i) {
// Insert one packet each time, except every 10th time where we don't insert
// any packet. This will create a positive clock-drift of approx. 11%.
int num_packets = (i % 10 == 9 ? 0 : 1);
for (int n = 0; n < num_packets; ++n) {
uint8_t payload[kPayloadBytes] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
kPayloadBytes, 0));
++frame_index;
}
// Pull out data once.
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
WebRtcNetEQ_NetworkStatistics network_stats;
ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
&network_stats));
EXPECT_EQ(110946, network_stats.clockDriftPPM);
}
TEST_F(NetEqDecodingTest, LongCngWithClockDrift) {
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 30;
const int kSamples = kFrameSizeMs * 16;
const int kPayloadBytes = kSamples * 2;
// Apply a clock drift of -25 ms / s (sender faster than receiver).
const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
double next_input_time_ms = 0.0;
double t_ms;
// Insert speech for 5 seconds.
const int kSpeechDurationMs = 5000;
for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
// Each turn in this for loop is 10 ms.
while (next_input_time_ms <= t_ms) {
// Insert one 30 ms speech frame.
uint8_t payload[kPayloadBytes] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
}
// Pull out data once.
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
EXPECT_EQ(kOutputNormal, neteq_inst_->getOutputType());
int32_t delay_before = timestamp - neteq_inst_->getSpeechTimeStamp();
// Insert CNG for 1 minute (= 60000 ms).
const int kCngPeriodMs = 100;
const int kCngPeriodSamples = kCngPeriodMs * 16; // Period in 16 kHz samples.
const int kCngDurationMs = 60000;
for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
// Each turn in this for loop is 10 ms.
while (next_input_time_ms <= t_ms) {
// Insert one CNG frame each 100 ms.
uint8_t payload[kPayloadBytes];
int payload_len;
WebRtcNetEQ_RTPInfo rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
payload_len, 0));
++seq_no;
timestamp += kCngPeriodSamples;
next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor;
}
// Pull out data once.
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
EXPECT_EQ(kOutputCNG, neteq_inst_->getOutputType());
// Insert speech again until output type is speech.
while (neteq_inst_->getOutputType() != kOutputNormal) {
// Each turn in this for loop is 10 ms.
while (next_input_time_ms <= t_ms) {
// Insert one 30 ms speech frame.
uint8_t payload[kPayloadBytes] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
}
// Pull out data once.
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
// Increase clock.
t_ms += 10;
}
int32_t delay_after = timestamp - neteq_inst_->getSpeechTimeStamp();
// Compare delay before and after, and make sure it differs less than 20 ms.
EXPECT_LE(delay_after, delay_before + 20 * 16);
EXPECT_GE(delay_after, delay_before - 20 * 16);
}
TEST_F(NetEqDecodingTest, NoInputDataStereo) {
void *ms_info;
ms_info = malloc(WebRtcNetEQ_GetMasterSlaveInfoSize());
neteq_inst_->setMaster();
// Slave instance without decoders (because it is easier).
WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1];
usedCodec[0] = kDecoderPCMu;
NETEQTEST_NetEQClass* slave_inst =
new NETEQTEST_NetEQClass(usedCodec, 1, 8000, kTCPLargeJitter);
ASSERT_TRUE(slave_inst);
NETEQTEST_Decoder* dec = new decoder_PCMU(0);
ASSERT_TRUE(dec != NULL);
dec->loadToNetEQ(*slave_inst);
slave_inst->setSlave();
// Pull out data.
const int kNumFrames = 100;
for (int i = 0; i < kNumFrames; ++i) {
ASSERT_TRUE(kBlockSize8kHz == neteq_inst_->recOut(out_data_, ms_info));
ASSERT_TRUE(kBlockSize8kHz == slave_inst->recOut(out_data_, ms_info));
}
delete dec;
delete slave_inst;
free(ms_info);
}
TEST_F(NetEqDecodingTest, TestExtraDelay) {
static const int kNumFrames = 120000; // Needed for convergence.
int frame_index = 0;
static const int kFrameSizeSamples = 30 * 16;
static const int kPayloadBytes = kFrameSizeSamples * 2;
test::InputAudioFile input_file(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
int16_t input[kFrameSizeSamples];
// Buffers of NetEq cannot accommodate larger delays for PCM16.
static const int kExtraDelayMs = 3200;
ASSERT_EQ(0, WebRtcNetEQ_SetExtraDelay(neteq_inst_->instance(),
kExtraDelayMs));
for (int i = 0; i < kNumFrames; ++i) {
ASSERT_TRUE(input_file.Read(kFrameSizeSamples, input));
WebRtcNetEQ_RTPInfo rtp_info;
PopulateRtpInfo(frame_index, frame_index * kFrameSizeSamples, &rtp_info);
uint8_t* payload = reinterpret_cast<uint8_t*>(input);
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
kPayloadBytes, 0));
++frame_index;
// Pull out data.
for (int j = 0; j < 3; ++j) {
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
}
if (i % 100 == 0) {
WebRtcNetEQ_NetworkStatistics network_stats;
ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
&network_stats));
const int expected_lower_limit =
std::min(i * 0.083 - 210, 0.9 * network_stats.preferredBufferSize);
EXPECT_GE(network_stats.currentBufferSize, expected_lower_limit);
const int expected_upper_limit =
std::min(i * 0.083 + 255, 1.2 * network_stats.preferredBufferSize);
EXPECT_LE(network_stats.currentBufferSize, expected_upper_limit);
}
}
}
void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers) {
uint16_t seq_no = start_seq_no;
uint32_t timestamp = start_timestamp;
const int kFrameSizeMs = 30;
const int kSamples = kFrameSizeMs * 16;
const int kPayloadBytes = kSamples * 2;
double next_input_time_ms = 0.0;
// Insert speech for 1 second.
const int kSpeechDurationMs = 1000;
for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
// Each turn in this for loop is 10 ms.
while (next_input_time_ms <= t_ms) {
// Insert one 30 ms speech frame.
uint8_t payload[kPayloadBytes] = {0};
WebRtcNetEQ_RTPInfo rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
// This sequence number was not in the set to drop. Insert it.
ASSERT_EQ(0,
WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
&rtp_info,
payload,
kPayloadBytes, 0));
}
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs);
WebRtcNetEQ_NetworkStatistics network_stats;
ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
&network_stats));
// Expect preferred and actual buffer size to be no more than 2 frames.
EXPECT_LE(network_stats.preferredBufferSize, kFrameSizeMs * 2);
EXPECT_LE(network_stats.currentBufferSize, kFrameSizeMs * 2);
}
// Pull out data once.
ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
// Expect delay (in samples) to be less than 2 packets.
EXPECT_LE(timestamp - neteq_inst_->getSpeechTimeStamp(),
static_cast<uint32_t>(kSamples * 2));
}
}
TEST_F(NetEqDecodingTest, SequenceNumberWrap) {
// Start with a sequence number that will soon wrap.
std::set<uint16_t> drop_seq_numbers; // Don't drop any packets.
WrapTest(0xFFFF - 5, 0, drop_seq_numbers);
}
TEST_F(NetEqDecodingTest, SequenceNumberWrapAndDrop) {
// Start with a sequence number that will soon wrap.
std::set<uint16_t> drop_seq_numbers;
drop_seq_numbers.insert(0xFFFF);
drop_seq_numbers.insert(0x0);
WrapTest(0xFFFF - 5, 0, drop_seq_numbers);
}
TEST_F(NetEqDecodingTest, TimestampWrap) {
// Start with a timestamp that will soon wrap.
std::set<uint16_t> drop_seq_numbers;
WrapTest(0, 0xFFFFFFFF - 1000, drop_seq_numbers);
}
TEST_F(NetEqDecodingTest, TimestampAndSequenceNumberWrap) {
// Start with a timestamp and a sequence number that will wrap at the same
// time.
std::set<uint16_t> drop_seq_numbers;
WrapTest(0xFFFF - 2, 0xFFFFFFFF - 1000, drop_seq_numbers);
}
} // namespace
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