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ff43c85ef1745ed62048c3b5bf0cb776eea97864
platform-external-webrtc/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
turaj@webrtc.org ff43c85ef1 API add to set background noise mode. 
Background noise mode.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2194005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4835 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-25 00:07:27 +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 "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"
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 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), sizeof(stats.extended_max), 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),
sizeof(ref_stats.extended_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.cumulative_lost, stats.cumulative_lost);
EXPECT_EQ(ref_stats.extended_max, stats.extended_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;
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_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.
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/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, 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/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);
}
// 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, 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());
}
} // namespace
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