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fe5d36b6fe3867ebd56d7285cd56b9aa57bb1ede
platform-external-webrtc/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
sprang@webrtc.org fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. 
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.

BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-10-28 09:21:07 +00:00

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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/neteq4/interface/neteq.h"
#include <stdlib.h>
#include <string.h> // memset
#include <cmath>
#include <string>
#include <vector>
#include "gflags/gflags.h"
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
#include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/gtest_disable.h"
#include "webrtc/typedefs.h"
DEFINE_bool(gen_ref, false, "Generate reference files.");
namespace webrtc {
static bool IsAllZero(const int16_t* buf, int buf_length) {
bool all_zero = true;
for (int n = 0; n < buf_length && all_zero; ++n)
all_zero = buf[n] == 0;
return all_zero;
}
static bool IsAllNonZero(const int16_t* buf, int buf_length) {
bool all_non_zero = true;
for (int n = 0; n < buf_length && all_non_zero; ++n)
all_non_zero = buf[n] != 0;
return all_non_zero;
}
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 NetEqNetworkStatistics& stats);
void ReadFromFileAndCompare(const NetEqNetworkStatistics& stats);
void WriteToFile(const RtcpStatistics& stats);
void ReadFromFileAndCompare(const RtcpStatistics& 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 NetEqNetworkStatistics& stats) {
if (output_fp_) {
ASSERT_EQ(1u, fwrite(&stats, sizeof(NetEqNetworkStatistics), 1,
output_fp_));
}
}
void RefFiles::ReadFromFileAndCompare(
const NetEqNetworkStatistics& stats) {
if (input_fp_) {
// Read from ref file.
size_t stat_size = sizeof(NetEqNetworkStatistics);
NetEqNetworkStatistics 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 RtcpStatistics& stats) {
if (output_fp_) {
ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.cumulative_lost),
sizeof(stats.cumulative_lost), 1, output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.extended_max_sequence_number),
sizeof(stats.extended_max_sequence_number), 1,
output_fp_));
ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
output_fp_));
}
}
void RefFiles::ReadFromFileAndCompare(
const RtcpStatistics& stats) {
if (input_fp_) {
// Read from ref file.
RtcpStatistics ref_stats;
ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
sizeof(ref_stats.fraction_lost), 1, input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.cumulative_lost),
sizeof(ref_stats.cumulative_lost), 1, input_fp_));
ASSERT_EQ(1u, fread(&(ref_stats.extended_max_sequence_number),
sizeof(ref_stats.extended_max_sequence_number), 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.cumulative_lost, stats.cumulative_lost);
EXPECT_EQ(ref_stats.extended_max_sequence_number,
stats.extended_max_sequence_number);
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;
static const int kInitSampleRateHz = 8000;
NetEqDecodingTest();
virtual void SetUp();
virtual void TearDown();
void SelectDecoders(NetEqDecoder* used_codec);
void LoadDecoders();
void OpenInputFile(const std::string &rtp_file);
void Process(NETEQTEST_RTPpacket* rtp_ptr, int* 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,
WebRtcRTPHeader* rtp_info);
static void PopulateCng(int frame_index,
int timestamp,
WebRtcRTPHeader* rtp_info,
uint8_t* payload,
int* payload_len);
void CheckBgnOff(int sampling_rate, NetEqBackgroundNoiseMode bgn_mode);
NetEq* neteq_;
FILE* rtp_fp_;
unsigned int sim_clock_;
int16_t out_data_[kMaxBlockSize];
int output_sample_rate_;
};
// Allocating the static const so that it can be passed by reference.
const int NetEqDecodingTest::kTimeStepMs;
const int NetEqDecodingTest::kBlockSize8kHz;
const int NetEqDecodingTest::kBlockSize16kHz;
const int NetEqDecodingTest::kBlockSize32kHz;
const int NetEqDecodingTest::kMaxBlockSize;
const int NetEqDecodingTest::kInitSampleRateHz;
NetEqDecodingTest::NetEqDecodingTest()
: neteq_(NULL),
rtp_fp_(NULL),
sim_clock_(0),
output_sample_rate_(kInitSampleRateHz) {
memset(out_data_, 0, sizeof(out_data_));
}
void NetEqDecodingTest::SetUp() {
neteq_ = NetEq::Create(kInitSampleRateHz);
ASSERT_TRUE(neteq_);
LoadDecoders();
}
void NetEqDecodingTest::TearDown() {
delete neteq_;
if (rtp_fp_)
fclose(rtp_fp_);
}
void NetEqDecodingTest::LoadDecoders() {
// Load PCMu.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
// Load PCMa.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
#ifndef WEBRTC_ANDROID
// Load iLBC.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
#endif // WEBRTC_ANDROID
// Load iSAC.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
// Load iSAC SWB.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
// Load iSAC FB.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
// Load PCM16B nb.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16B, 93));
// Load PCM16B wb.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb, 94));
// Load PCM16B swb32.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bswb32kHz, 95));
// Load CNG 8 kHz.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, 13));
// Load CNG 16 kHz.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, 98));
}
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, int* out_len) {
// Check if time to receive.
while ((sim_clock_ >= rtp->time()) &&
(rtp->dataLen() >= 0)) {
if (rtp->dataLen() > 0) {
WebRtcRTPHeader rtpInfo;
rtp->parseHeader(&rtpInfo);
ASSERT_EQ(0, neteq_->InsertPacket(
rtpInfo,
rtp->payload(),
rtp->payloadLen(),
rtp->time() * (output_sample_rate_ / 1000)));
}
// Get next packet.
ASSERT_NE(-1, rtp->readFromFile(rtp_fp_));
}
// Get audio from NetEq.
NetEqOutputType type;
int num_channels;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
&num_channels, &type));
ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
(*out_len == kBlockSize16kHz) ||
(*out_len == kBlockSize32kHz));
output_sample_rate_ = *out_len / 10 * 1000;
// 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_universal_ref.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.
int 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) {
int out_len;
Process(&rtp, &out_len);
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
// Process NetworkStatistics.
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
network_stat_files.ProcessReference(network_stats);
// Process RTCPstat.
RtcpStatistics rtcp_stats;
neteq_->GetRtcpStatistics(&rtcp_stats);
rtcp_stat_files.ProcessReference(rtcp_stats);
}
}
}
void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
int timestamp,
WebRtcRTPHeader* rtp_info) {
rtp_info->header.sequenceNumber = frame_index;
rtp_info->header.timestamp = timestamp;
rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info->header.payloadType = 94; // PCM16b WB codec.
rtp_info->header.markerBit = 0;
}
void NetEqDecodingTest::PopulateCng(int frame_index,
int timestamp,
WebRtcRTPHeader* rtp_info,
uint8_t* payload,
int* payload_len) {
rtp_info->header.sequenceNumber = frame_index;
rtp_info->header.timestamp = timestamp;
rtp_info->header.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info->header.payloadType = 98; // WB CNG.
rtp_info->header.markerBit = 0;
payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
*payload_len = 1; // Only noise level, no spectral parameters.
}
void NetEqDecodingTest::CheckBgnOff(int sampling_rate_hz,
NetEqBackgroundNoiseMode bgn_mode) {
int expected_samples_per_channel = 0;
uint8_t payload_type = 0xFF; // Invalid.
if (sampling_rate_hz == 8000) {
expected_samples_per_channel = kBlockSize8kHz;
payload_type = 93; // PCM 16, 8 kHz.
} else if (sampling_rate_hz == 16000) {
expected_samples_per_channel = kBlockSize16kHz;
payload_type = 94; // PCM 16, 16 kHZ.
} else if (sampling_rate_hz == 32000) {
expected_samples_per_channel = kBlockSize32kHz;
payload_type = 95; // PCM 16, 32 kHz.
} else {
ASSERT_TRUE(false); // Unsupported test case.
}
NetEqOutputType type;
int16_t output[kBlockSize32kHz]; // Maximum size is chosen.
int16_t input[kBlockSize32kHz]; // Maximum size is chosen.
// Payload of 10 ms of PCM16 32 kHz.
uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
// Random payload.
for (int n = 0; n < expected_samples_per_channel; ++n) {
input[n] = (rand() & ((1 << 10) - 1)) - ((1 << 5) - 1);
}
int enc_len_bytes = WebRtcPcm16b_EncodeW16(
input, expected_samples_per_channel, reinterpret_cast<int16_t*>(payload));
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = payload_type;
int number_channels = 0;
int samples_per_channel = 0;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
number_channels = 0;
samples_per_channel = 0;
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_info, payload, enc_len_bytes, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
&number_channels, &type));
ASSERT_EQ(1, number_channels);
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
ASSERT_EQ(kOutputNormal, type);
// Next packet.
rtp_info.header.timestamp += expected_samples_per_channel;
rtp_info.header.sequenceNumber++;
receive_timestamp += expected_samples_per_channel;
}
number_channels = 0;
samples_per_channel = 0;
// Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull one
// frame without checking speech-type. This is the first frame pulled without
// inserting any packet, and might not be labeled as PCL.
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
&number_channels, &type));
ASSERT_EQ(1, number_channels);
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
// To be able to test the fading of background noise we need at lease to pull
// 610 frames.
const int kFadingThreshold = 610;
// Test several CNG-to-PLC packet for the expected behavior. The number 20 is
// arbitrary, but sufficiently large to test enough number of frames.
const int kNumPlcToCngTestFrames = 20;
bool plc_to_cng = false;
for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
number_channels = 0;
samples_per_channel = 0;
memset(output, 1, sizeof(output)); // Set to non-zero.
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize32kHz, output, &samples_per_channel,
&number_channels, &type));
ASSERT_EQ(1, number_channels);
ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
if (type == kOutputPLCtoCNG) {
plc_to_cng = true;
double sum_squared = 0;
for (int k = 0; k < number_channels * samples_per_channel; ++k)
sum_squared += output[k] * output[k];
if (bgn_mode == kBgnOn) {
EXPECT_NE(0, sum_squared);
} else if (bgn_mode == kBgnOff || n > kFadingThreshold) {
EXPECT_EQ(0, sum_squared);
}
} else {
EXPECT_EQ(kOutputPLC, type);
}
}
EXPECT_TRUE(plc_to_cng); // Just to be sure that PLC-to-CNG has occurred.
}
#if defined(_WIN32) && defined(WEBRTC_ARCH_64_BITS)
// Disabled for Windows 64-bit until webrtc:1458 is fixed.
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
#else
#define MAYBE_TestBitExactness TestBitExactness
#endif
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(MAYBE_TestBitExactness)) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_new.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/neteq4_universal_ref.pcm";
#else
const std::string kInputRefFile =
webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
#endif
if (FLAGS_gen_ref) {
DecodeAndCompare(kInputRtpFile, "");
} else {
DecodeAndCompare(kInputRtpFile, kInputRefFile);
}
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestNetworkStatistics)) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_new.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/neteq4_network_stats.dat";
#else
const std::string kNetworkStatRefFile =
webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat");
#endif
const std::string kRtcpStatRefFile =
webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
if (FLAGS_gen_ref) {
DecodeAndCheckStats(kInputRtpFile, "", "");
} else {
DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
}
}
// TODO(hlundin): Re-enable test once the statistics interface is up and again.
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestFrameWaitingTimeStatistics)) {
// Use fax mode to avoid time-scaling. This is to simplify the testing of
// packet waiting times in the packet buffer.
neteq_->SetPlayoutMode(kPlayoutFax);
ASSERT_EQ(kPlayoutFax, neteq_->PlayoutMode());
// Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
size_t num_frames = 30;
const int kSamples = 10 * 16;
const int kPayloadBytes = kSamples * 2;
for (size_t i = 0; i < num_frames; ++i) {
uint16_t payload[kSamples] = {0};
WebRtcRTPHeader rtp_info;
rtp_info.header.sequenceNumber = i;
rtp_info.header.timestamp = i * kSamples;
rtp_info.header.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.header.payloadType = 94; // PCM16b WB codec.
rtp_info.header.markerBit = 0;
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_info,
reinterpret_cast<uint8_t*>(payload),
kPayloadBytes, 0));
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
int out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
std::vector<int> waiting_times;
neteq_->WaitingTimes(&waiting_times);
int len = waiting_times.size();
EXPECT_EQ(num_frames, waiting_times.size());
// 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 (size_t i = 0; i < waiting_times.size(); ++i) {
EXPECT_EQ(static_cast<int>(i + 1) * 10, waiting_times[i]);
}
// Check statistics again and make sure it's been reset.
neteq_->WaitingTimes(&waiting_times);
len = waiting_times.size();
EXPECT_EQ(0, len);
// 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 (size_t i = 0; i < num_frames; ++i) {
uint16_t payload[kSamples] = {0};
WebRtcRTPHeader rtp_info;
rtp_info.header.sequenceNumber = i;
rtp_info.header.timestamp = i * kSamples;
rtp_info.header.ssrc = 0x1235; // Just an arbitrary SSRC.
rtp_info.header.payloadType = 94; // PCM16b WB codec.
rtp_info.header.markerBit = 0;
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_info,
reinterpret_cast<uint8_t*>(payload),
kPayloadBytes, 0));
int out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
neteq_->WaitingTimes(&waiting_times);
EXPECT_EQ(100u, waiting_times.size());
}
TEST_F(NetEqDecodingTest,
DISABLED_ON_ANDROID(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};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++frame_index;
}
// Pull out data once.
int out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
}
TEST_F(NetEqDecodingTest,
DISABLED_ON_ANDROID(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};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++frame_index;
}
// Pull out data once.
int out_len;
int num_channels;
NetEqOutputType type;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
EXPECT_EQ(110946, network_stats.clockdrift_ppm);
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(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;
NetEqOutputType type;
// 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};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
}
// Pull out data once.
int out_len;
int num_channels;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
EXPECT_EQ(kOutputNormal, type);
int32_t delay_before = timestamp - neteq_->PlayoutTimestamp();
// 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;
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
++seq_no;
timestamp += kCngPeriodSamples;
next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor;
}
// Pull out data once.
int out_len;
int num_channels;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
EXPECT_EQ(kOutputCNG, type);
// Insert speech again until output type is speech.
while (type != 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};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
}
// Pull out data once.
int out_len;
int num_channels;
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
// Increase clock.
t_ms += 10;
}
int32_t delay_after = timestamp - neteq_->PlayoutTimestamp();
// 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, DISABLED_ON_ANDROID(UnknownPayloadType)) {
const int kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = 1; // Not registered as a decoder.
EXPECT_EQ(NetEq::kFail,
neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(OversizePacket)) {
// Payload size is greater than packet buffer size
const int kPayloadBytes = NetEq::kMaxBytesInBuffer + 1;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = 103; // iSAC, no packet splitting.
EXPECT_EQ(NetEq::kFail,
neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
EXPECT_EQ(NetEq::kOversizePacket, neteq_->LastError());
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
const int kPayloadBytes = 100;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = 103; // iSAC, but the payload is invalid.
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
for (int i = 0; i < kMaxBlockSize; ++i) {
out_data_[i] = 1;
}
int num_channels;
int samples_per_channel;
EXPECT_EQ(NetEq::kFail,
neteq_->GetAudio(kMaxBlockSize, out_data_,
&samples_per_channel, &num_channels, &type));
// Verify that there is a decoder error to check.
EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
// Code 6730 is an iSAC error code.
EXPECT_EQ(6730, neteq_->LastDecoderError());
// Verify that the first 160 samples are set to 0, and that the remaining
// samples are left unmodified.
static const int kExpectedOutputLength = 160; // 10 ms at 16 kHz sample rate.
for (int i = 0; i < kExpectedOutputLength; ++i) {
std::ostringstream ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(0, out_data_[i]);
}
for (int i = kExpectedOutputLength; i < kMaxBlockSize; ++i) {
std::ostringstream ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(1, out_data_[i]);
}
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(GetAudioBeforeInsertPacket)) {
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
for (int i = 0; i < kMaxBlockSize; ++i) {
out_data_[i] = 1;
}
int num_channels;
int samples_per_channel;
EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
&samples_per_channel,
&num_channels, &type));
// Verify that the first block of samples is set to 0.
static const int kExpectedOutputLength =
kInitSampleRateHz / 100; // 10 ms at initial sample rate.
for (int i = 0; i < kExpectedOutputLength; ++i) {
std::ostringstream ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(0, out_data_[i]);
}
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(BackgroundNoise)) {
neteq_->SetBackgroundNoiseMode(kBgnOn);
CheckBgnOff(8000, kBgnOn);
CheckBgnOff(16000, kBgnOn);
CheckBgnOff(32000, kBgnOn);
EXPECT_EQ(kBgnOn, neteq_->BackgroundNoiseMode());
neteq_->SetBackgroundNoiseMode(kBgnOff);
CheckBgnOff(8000, kBgnOff);
CheckBgnOff(16000, kBgnOff);
CheckBgnOff(32000, kBgnOff);
EXPECT_EQ(kBgnOff, neteq_->BackgroundNoiseMode());
neteq_->SetBackgroundNoiseMode(kBgnFade);
CheckBgnOff(8000, kBgnFade);
CheckBgnOff(16000, kBgnFade);
CheckBgnOff(32000, kBgnFade);
EXPECT_EQ(kBgnFade, neteq_->BackgroundNoiseMode());
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketInsert)) {
WebRtcRTPHeader rtp_info;
uint32_t receive_timestamp = 0;
// For the readability use the following payloads instead of the defaults of
// this test.
uint8_t kPcm16WbPayloadType = 1;
uint8_t kCngNbPayloadType = 2;
uint8_t kCngWbPayloadType = 3;
uint8_t kCngSwb32PayloadType = 4;
uint8_t kCngSwb48PayloadType = 5;
uint8_t kAvtPayloadType = 6;
uint8_t kRedPayloadType = 7;
uint8_t kIsacPayloadType = 9; // Payload type 8 is already registered.
// Register decoders.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb,
kPcm16WbPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, kCngNbPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, kCngWbPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb32kHz,
kCngSwb32PayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGswb48kHz,
kCngSwb48PayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderAVT, kAvtPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderRED, kRedPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, kIsacPayloadType));
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.header.payloadType = kPcm16WbPayloadType;
// The first packet injected cannot be sync-packet.
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
// Payload length of 10 ms PCM16 16 kHz.
const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes] = {0};
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_info, payload, kPayloadBytes, receive_timestamp));
// Next packet. Last packet contained 10 ms audio.
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
// Unacceptable payload types CNG, AVT (DTMF), RED.
rtp_info.header.payloadType = kCngNbPayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
rtp_info.header.payloadType = kCngWbPayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
rtp_info.header.payloadType = kCngSwb32PayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
rtp_info.header.payloadType = kCngSwb48PayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
rtp_info.header.payloadType = kAvtPayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
rtp_info.header.payloadType = kRedPayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
// Change of codec cannot be initiated with a sync packet.
rtp_info.header.payloadType = kIsacPayloadType;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
// Change of SSRC is not allowed with a sync packet.
rtp_info.header.payloadType = kPcm16WbPayloadType;
++rtp_info.header.ssrc;
EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
--rtp_info.header.ssrc;
EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
}
// First insert several noise like packets, then sync-packets. Decoding all
// packets should not produce error, statistics should not show any packet loss
// and sync-packets should decode to zero.
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(SyncPacketDecode)) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
for (int n = 0; n < kPayloadBytes; ++n) {
payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
}
// Insert some packets which decode to noise. We are not interested in
// actual decoded values.
NetEqOutputType output_type;
int num_channels;
int samples_per_channel;
uint32_t receive_timestamp = 0;
int delay_samples = 0;
for (int n = 0; n < 100; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
ASSERT_EQ(1, num_channels);
// Even if there is RTP packet in NetEq's buffer, the first frame pulled
// from NetEq starts with few zero samples. Here we measure this delay.
if (n == 0) {
while(decoded[delay_samples] == 0) delay_samples++;
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
const int kNumSyncPackets = 10;
// Insert sync-packets, the decoded sequence should be all-zero.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
ASSERT_EQ(1, num_channels);
EXPECT_TRUE(IsAllZero(&decoded[delay_samples],
samples_per_channel * num_channels - delay_samples));
delay_samples = 0; // Delay only matters in the first frame.
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
// We insert a regular packet, if sync packet are not correctly buffered then
// network statistics would show some packet loss.
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
// Make sure the last inserted packet is decoded and there are non-zero
// samples.
EXPECT_FALSE(IsAllZero(decoded, samples_per_channel * num_channels));
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
// Expecting a "clean" network.
EXPECT_EQ(0, network_stats.packet_loss_rate);
EXPECT_EQ(0, network_stats.expand_rate);
EXPECT_EQ(0, network_stats.accelerate_rate);
EXPECT_EQ(0, network_stats.preemptive_rate);
}
// Test if the size of the packet buffer reported correctly when containing
// sync packets. Also, test if network packets override sync packets. That is to
// prefer decoding a network packet to a sync packet, if both have same sequence
// number and timestamp.
TEST_F(NetEqDecodingTest,
DISABLED_ON_ANDROID(SyncPacketBufferSizeAndOverridenByNetworkPackets)) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
const int kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
for (int n = 0; n < kPayloadBytes; ++n) {
payload[n] = (rand() & 0xF0) + 1; // Non-zero random sequence.
}
// Insert some packets which decode to noise. We are not interested in
// actual decoded values.
NetEqOutputType output_type;
int num_channels;
int samples_per_channel;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 1; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
ASSERT_EQ(1, num_channels);
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
const int kNumSyncPackets = 10;
WebRtcRTPHeader first_sync_packet_rtp_info;
memcpy(&first_sync_packet_rtp_info, &rtp_info, sizeof(rtp_info));
// Insert sync-packets, but no decoding.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
EXPECT_EQ(kNumSyncPackets * 10, network_stats.current_buffer_size_ms);
// Rewind |rtp_info| to that of the first sync packet.
memcpy(&rtp_info, &first_sync_packet_rtp_info, sizeof(rtp_info));
// Insert.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
// Decode.
for (int n = 0; n < kNumSyncPackets; ++n) {
ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
&samples_per_channel, &num_channels,
&output_type));
ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
ASSERT_EQ(1, num_channels);
EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
}
}
} // namespace
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