Google Git
Sign in
webrtc / src / branch-heads/43 / . / webrtc / modules / audio_coding / neteq / neteq_unittest.cc
blob: b3d6f25f8910e778c522ce188a8d51a1b90b5cc5 [file] [log] [blame]
/*
* 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/neteq.h"
#include <math.h>
#include <stdlib.h>
#include <string.h> // memset
#include <algorithm>
#include <set>
#include <string>
#include <vector>
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.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) {
// TODO(minyue): Update resource/audio_coding/neteq_network_stats.dat and
// resource/audio_coding/neteq_network_stats_win32.dat.
struct NetEqNetworkStatisticsOld {
uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
// jitter; 0 otherwise.
uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
uint16_t packet_discard_rate; // Late loss rate in Q14.
uint16_t expand_rate; // Fraction (of original stream) of synthesized
// audio inserted through expansion (in Q14).
uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
// expansion (in Q14).
uint16_t accelerate_rate; // Fraction of data removed through acceleration
// (in Q14).
int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
// (positive or negative).
int added_zero_samples; // Number of zero samples added in "off" mode.
};
if (input_fp_) {
// Read from ref file.
size_t stat_size = sizeof(NetEqNetworkStatisticsOld);
NetEqNetworkStatisticsOld ref_stats;
ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
// Compare
ASSERT_EQ(stats.current_buffer_size_ms, ref_stats.current_buffer_size_ms);
ASSERT_EQ(stats.preferred_buffer_size_ms,
ref_stats.preferred_buffer_size_ms);
ASSERT_EQ(stats.jitter_peaks_found, ref_stats.jitter_peaks_found);
ASSERT_EQ(stats.packet_loss_rate, ref_stats.packet_loss_rate);
ASSERT_EQ(stats.packet_discard_rate, ref_stats.packet_discard_rate);
ASSERT_EQ(stats.expand_rate, ref_stats.expand_rate);
ASSERT_EQ(stats.preemptive_rate, ref_stats.preemptive_rate);
ASSERT_EQ(stats.accelerate_rate, ref_stats.accelerate_rate);
ASSERT_EQ(stats.clockdrift_ppm, ref_stats.clockdrift_ppm);
ASSERT_EQ(stats.added_zero_samples, ref_stats.added_zero_samples);
ASSERT_EQ(stats.secondary_decoded_rate, 0);
ASSERT_LE(stats.speech_expand_rate, ref_stats.expand_rate);
}
}
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
ASSERT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
ASSERT_EQ(ref_stats.cumulative_lost, stats.cumulative_lost);
ASSERT_EQ(ref_stats.extended_max_sequence_number,
stats.extended_max_sequence_number);
ASSERT_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 size_t 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(int* out_len);
void DecodeAndCompare(const std::string& rtp_file,
const std::string& ref_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,
size_t* payload_len);
void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers,
bool expect_seq_no_wrap, bool expect_timestamp_wrap);
void LongCngWithClockDrift(double drift_factor,
double network_freeze_ms,
bool pull_audio_during_freeze,
int delay_tolerance_ms,
int max_time_to_speech_ms);
void DuplicateCng();
uint32_t PlayoutTimestamp();
NetEq* neteq_;
NetEq::Config config_;
rtc::scoped_ptr<test::RtpFileSource> rtp_source_;
rtc::scoped_ptr<test::Packet> packet_;
unsigned int sim_clock_;
int16_t out_data_[kMaxBlockSize];
int output_sample_rate_;
int algorithmic_delay_ms_;
};
// 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 size_t NetEqDecodingTest::kMaxBlockSize;
const int NetEqDecodingTest::kInitSampleRateHz;
NetEqDecodingTest::NetEqDecodingTest()
: neteq_(NULL),
config_(),
sim_clock_(0),
output_sample_rate_(kInitSampleRateHz),
algorithmic_delay_ms_(0) {
config_.sample_rate_hz = kInitSampleRateHz;
memset(out_data_, 0, sizeof(out_data_));
}
void NetEqDecodingTest::SetUp() {
neteq_ = NetEq::Create(config_);
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;
ASSERT_TRUE(neteq_);
LoadDecoders();
}
void NetEqDecodingTest::TearDown() {
delete neteq_;
}
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));
#ifndef WEBRTC_ANDROID
// Load iSAC SWB.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
// Load iSAC FB.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
#endif // WEBRTC_ANDROID
// 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_source_.reset(test::RtpFileSource::Create(rtp_file));
}
void NetEqDecodingTest::Process(int* out_len) {
// Check if time to receive.
while (packet_ && sim_clock_ >= packet_->time_ms()) {
if (packet_->payload_length_bytes() > 0) {
WebRtcRTPHeader rtp_header;
packet_->ConvertHeader(&rtp_header);
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_header, packet_->payload(),
packet_->payload_length_bytes(),
packet_->time_ms() * (output_sample_rate_ / 1000)));
}
// Get next packet.
packet_.reset(rtp_source_->NextPacket());
}
// 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,
const std::string& stat_ref_file,
const std::string& rtcp_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);
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);
packet_.reset(rtp_source_->NextPacket());
int i = 0;
while (packet_) {
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 = 0;
ASSERT_NO_FATAL_FAILURE(Process(&out_len));
ASSERT_NO_FATAL_FAILURE(ref_files.ProcessReference(out_data_, 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));
ASSERT_NO_FATAL_FAILURE(
network_stat_files.ProcessReference(network_stats));
// Process RTCPstat.
RtcpStatistics rtcp_stats;
neteq_->GetRtcpStatistics(&rtcp_stats);
ASSERT_NO_FATAL_FAILURE(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,
size_t* 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.
}
TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(TestBitExactness)) {
const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq_universal_new.rtp";
// Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
// are identical. The latter could have been removed, but if clients still
// have a copy of the file, the test will fail.
const std::string input_ref_file =
webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
#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 network_stat_ref_file = webrtc::test::ProjectRootPath() +
"resources/audio_coding/neteq4_network_stats.dat";
#else
const std::string network_stat_ref_file =
webrtc::test::ResourcePath("audio_coding/neteq4_network_stats", "dat");
#endif
const std::string rtcp_stat_ref_file =
webrtc::test::ResourcePath("audio_coding/neteq4_rtcp_stats", "dat");
if (FLAGS_gen_ref) {
DecodeAndCompare(input_rtp_file, "", "", "");
} else {
DecodeAndCompare(input_rtp_file,
input_ref_file,
network_stat_ref_file,
rtcp_stat_ref_file);
}
}
// Use fax mode to avoid time-scaling. This is to simplify the testing of
// packet waiting times in the packet buffer.
class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
protected:
NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
config_.playout_mode = kPlayoutFax;
}
};
TEST_F(NetEqDecodingTestFaxMode, TestFrameWaitingTimeStatistics) {
// Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
size_t num_frames = 30;
const size_t kSamples = 10 * 16;
const size_t 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);
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);
int 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, TestAverageInterArrivalTimeNegative) {
const int kNumFrames = 3000; // Needed for convergence.
int frame_index = 0;
const size_t kSamples = 10 * 16;
const size_t 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, TestAverageInterArrivalTimePositive) {
const int kNumFrames = 5000; // Needed for convergence.
int frame_index = 0;
const size_t kSamples = 10 * 16;
const size_t 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);
}
void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
double network_freeze_ms,
bool pull_audio_during_freeze,
int delay_tolerance_ms,
int max_time_to_speech_ms) {
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 30;
const size_t kSamples = kFrameSizeMs * 16;
const size_t kPayloadBytes = kSamples * 2;
double next_input_time_ms = 0.0;
double t_ms;
int out_len;
int num_channels;
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) * drift_factor;
}
// Pull out data once.
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 - 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];
size_t 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) * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
EXPECT_EQ(kOutputCNG, type);
if (network_freeze_ms > 0) {
// First keep pulling audio for |network_freeze_ms| without inserting
// any data, then insert CNG data corresponding to |network_freeze_ms|
// without pulling any output audio.
const double loop_end_time = t_ms + network_freeze_ms;
for (; t_ms < loop_end_time; t_ms += 10) {
// Pull out data once.
ASSERT_EQ(0,
neteq_->GetAudio(
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
}
bool pull_once = pull_audio_during_freeze;
// If |pull_once| is true, GetAudio will be called once half-way through
// the network recovery period.
double pull_time_ms = (t_ms + next_input_time_ms) / 2;
while (next_input_time_ms <= t_ms) {
if (pull_once && next_input_time_ms >= pull_time_ms) {
pull_once = false;
// Pull out data once.
ASSERT_EQ(
0,
neteq_->GetAudio(
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
t_ms += 10;
}
// Insert one CNG frame each 100 ms.
uint8_t payload[kPayloadBytes];
size_t 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 += kCngPeriodMs * drift_factor;
}
}
// Insert speech again until output type is speech.
double speech_restart_time_ms = t_ms;
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 += kFrameSizeMs * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
// Increase clock.
t_ms += 10;
}
// Check that the speech starts again within reasonable time.
double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
int32_t delay_after = timestamp - PlayoutTimestamp();
// Compare delay before and after, and make sure it differs less than 20 ms.
EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
}
TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) {
// Apply a clock drift of -25 ms / s (sender faster than receiver).
const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
const double kNetworkFreezeTimeMs = 0.0;
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
LongCngWithClockDrift(kDriftFactor,
kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery,
kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDrift) {
// Apply a clock drift of +25 ms / s (sender slower than receiver).
const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
const double kNetworkFreezeTimeMs = 0.0;
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
LongCngWithClockDrift(kDriftFactor,
kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery,
kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDriftNetworkFreeze) {
// Apply a clock drift of -25 ms / s (sender faster than receiver).
const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
const double kNetworkFreezeTimeMs = 5000.0;
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 50;
const int kMaxTimeToSpeechMs = 200;
LongCngWithClockDrift(kDriftFactor,
kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery,
kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreeze) {
// Apply a clock drift of +25 ms / s (sender slower than receiver).
const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
const double kNetworkFreezeTimeMs = 5000.0;
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
LongCngWithClockDrift(kDriftFactor,
kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery,
kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreezeExtraPull) {
// Apply a clock drift of +25 ms / s (sender slower than receiver).
const double kDriftFactor = 1000.0 / (1000.0 - 25.0);
const double kNetworkFreezeTimeMs = 5000.0;
const bool kGetAudioDuringFreezeRecovery = true;
const int kDelayToleranceMs = 20;
const int kMaxTimeToSpeechMs = 100;
LongCngWithClockDrift(kDriftFactor,
kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery,
kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
TEST_F(NetEqDecodingTest, LongCngWithoutClockDrift) {
const double kDriftFactor = 1.0; // No drift.
const double kNetworkFreezeTimeMs = 0.0;
const bool kGetAudioDuringFreezeRecovery = false;
const int kDelayToleranceMs = 10;
const int kMaxTimeToSpeechMs = 50;
LongCngWithClockDrift(kDriftFactor,
kNetworkFreezeTimeMs,
kGetAudioDuringFreezeRecovery,
kDelayToleranceMs,
kMaxTimeToSpeechMs);
}
TEST_F(NetEqDecodingTest, UnknownPayloadType) {
const size_t 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(DecoderError)) {
const size_t 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 (size_t 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 (size_t 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, GetAudioBeforeInsertPacket) {
NetEqOutputType type;
// Set all of |out_data_| to 1, and verify that it was set to 0 by the call
// to GetAudio.
for (size_t 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]);
}
}
class NetEqBgnTest : public NetEqDecodingTest {
protected:
virtual void TestCondition(double sum_squared_noise,
bool should_be_faded) = 0;
void CheckBgn(int sampling_rate_hz) {
int16_t 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.
test::AudioLoop input;
// We are using the same 32 kHz input file for all tests, regardless of
// |sampling_rate_hz|. The output may sound weird, but the test is still
// valid.
ASSERT_TRUE(input.Init(
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
10 * sampling_rate_hz, // Max 10 seconds loop length.
static_cast<size_t>(expected_samples_per_channel)));
// Payload of 10 ms of PCM16 32 kHz.
uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
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.
int16_t enc_len_bytes = WebRtcPcm16b_Encode(
input.GetNextBlock(), expected_samples_per_channel, payload);
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
number_channels = 0;
samples_per_channel = 0;
ASSERT_EQ(0,
neteq_->InsertPacket(rtp_info, payload,
static_cast<size_t>(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 PLC.
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 611 frames.
const int kFadingThreshold = 611;
// 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];
TestCondition(sum_squared, n > kFadingThreshold);
} else {
EXPECT_EQ(kOutputPLC, type);
}
}
EXPECT_TRUE(plc_to_cng); // Just to be sure that PLC-to-CNG has occurred.
}
};
class NetEqBgnTestOn : public NetEqBgnTest {
protected:
NetEqBgnTestOn() : NetEqBgnTest() {
config_.background_noise_mode = NetEq::kBgnOn;
}
void TestCondition(double sum_squared_noise, bool /*should_be_faded*/) {
EXPECT_NE(0, sum_squared_noise);
}
};
class NetEqBgnTestOff : public NetEqBgnTest {
protected:
NetEqBgnTestOff() : NetEqBgnTest() {
config_.background_noise_mode = NetEq::kBgnOff;
}
void TestCondition(double sum_squared_noise, bool /*should_be_faded*/) {
EXPECT_EQ(0, sum_squared_noise);
}
};
class NetEqBgnTestFade : public NetEqBgnTest {
protected:
NetEqBgnTestFade() : NetEqBgnTest() {
config_.background_noise_mode = NetEq::kBgnFade;
}
void TestCondition(double sum_squared_noise, bool should_be_faded) {
if (should_be_faded)
EXPECT_EQ(0, sum_squared_noise);
}
};
TEST_F(NetEqBgnTestOn, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
TEST_F(NetEqBgnTestOff, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
TEST_F(NetEqBgnTestFade, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
TEST_F(NetEqDecodingTest, 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 size_t 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.
// TODO(turajs) we will have a better test if we have a referece NetEq, and
// when Sync packets are inserted in "test" NetEq we insert all-zero payload
// in reference NetEq and compare the output of those two.
TEST_F(NetEqDecodingTest, SyncPacketDecode) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
for (size_t 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 < 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);
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
const int kNumSyncPackets = 10;
// Make sure sufficient number of sync packets are inserted that we can
// conduct a test.
ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay);
// 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);
if (n > algorithmic_frame_delay) {
EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
// We insert regular packets, if sync packet are not correctly buffered then
// network statistics would show some packet loss.
for (int n = 0; n <= algorithmic_frame_delay + 10; ++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));
if (n >= algorithmic_frame_delay + 1) {
// Expect that this frame contain samples from regular RTP.
EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
}
rtp_info.header.sequenceNumber++;
rtp_info.header.timestamp += kBlockSize16kHz;
receive_timestamp += kBlockSize16kHz;
}
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_LE(network_stats.preemptive_rate, 150);
}
// 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, SyncPacketBufferSizeAndOverridenByNetworkPackets) {
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
uint8_t payload[kPayloadBytes];
int16_t decoded[kBlockSize16kHz];
for (size_t 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 algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
for (int n = 0; n < algorithmic_frame_delay; ++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 + algorithmic_delay_ms_,
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));
}
}
void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers,
bool expect_seq_no_wrap,
bool expect_timestamp_wrap) {
uint16_t seq_no = start_seq_no;
uint32_t timestamp = start_timestamp;
const int kBlocksPerFrame = 3; // Number of 10 ms blocks per frame.
const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
const size_t kPayloadBytes = kSamples * sizeof(int16_t);
double next_input_time_ms = 0.0;
int16_t decoded[kBlockSize16kHz];
int num_channels;
int samples_per_channel;
NetEqOutputType output_type;
uint32_t receive_timestamp = 0;
// Insert speech for 2 seconds.
const int kSpeechDurationMs = 2000;
int packets_inserted = 0;
uint16_t last_seq_no;
uint32_t last_timestamp;
bool timestamp_wrapped = false;
bool seq_no_wrapped = false;
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};
WebRtcRTPHeader 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,
neteq_->InsertPacket(rtp_info, payload, kPayloadBytes,
receive_timestamp));
++packets_inserted;
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
// Due to internal NetEq logic, preferred buffer-size is about 4 times the
// packet size for first few packets. Therefore we refrain from checking
// the criteria.
if (packets_inserted > 4) {
// Expect preferred and actual buffer size to be no more than 2 frames.
EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 +
algorithmic_delay_ms_);
}
last_seq_no = seq_no;
last_timestamp = timestamp;
++seq_no;
timestamp += kSamples;
receive_timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs);
seq_no_wrapped |= seq_no < last_seq_no;
timestamp_wrapped |= timestamp < last_timestamp;
}
// Pull out data once.
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 delay (in samples) to be less than 2 packets.
EXPECT_LE(timestamp - PlayoutTimestamp(),
static_cast<uint32_t>(kSamples * 2));
}
// Make sure we have actually tested wrap-around.
ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
}
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 - 10, 0, drop_seq_numbers, true, false);
}
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 - 10, 0, drop_seq_numbers, true, false);
}
TEST_F(NetEqDecodingTest, TimestampWrap) {
// Start with a timestamp that will soon wrap.
std::set<uint16_t> drop_seq_numbers;
WrapTest(0, 0xFFFFFFFF - 3000, drop_seq_numbers, false, true);
}
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 - 10, 0xFFFFFFFF - 5000, drop_seq_numbers, true, true);
}
void NetEqDecodingTest::DuplicateCng() {
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 10;
const int kSampleRateKhz = 16;
const int kSamples = kFrameSizeMs * kSampleRateKhz;
const size_t kPayloadBytes = kSamples * 2;
const int algorithmic_delay_samples = std::max(
algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
// Insert three speech packets. Three are needed to get the frame length
// correct.
int out_len;
int num_channels;
NetEqOutputType type;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
for (int i = 0; i < 3; ++i) {
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0,
neteq_->GetAudio(
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
// Verify speech output.
EXPECT_EQ(kOutputNormal, type);
// Insert same CNG packet twice.
const int kCngPeriodMs = 100;
const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
size_t payload_len;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
// This is the first time this CNG packet is inserted.
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
// Pull audio once and make sure CNG is played.
ASSERT_EQ(0,
neteq_->GetAudio(
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(timestamp - algorithmic_delay_samples, PlayoutTimestamp());
// Insert the same CNG packet again. Note that at this point it is old, since
// we have already decoded the first copy of it.
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
// Pull audio until we have played |kCngPeriodMs| of CNG. Start at 10 ms since
// we have already pulled out CNG once.
for (int cng_time_ms = 10; cng_time_ms < kCngPeriodMs; cng_time_ms += 10) {
ASSERT_EQ(0,
neteq_->GetAudio(
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputCNG, type);
EXPECT_EQ(timestamp - algorithmic_delay_samples,
PlayoutTimestamp());
}
// Insert speech again.
++seq_no;
timestamp += kCngPeriodSamples;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
// Pull audio once and verify that the output is speech again.
ASSERT_EQ(0,
neteq_->GetAudio(
kMaxBlockSize, out_data_, &out_len, &num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
EXPECT_EQ(kOutputNormal, type);
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
PlayoutTimestamp());
}
uint32_t NetEqDecodingTest::PlayoutTimestamp() {
uint32_t playout_timestamp = 0;
EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&playout_timestamp));
return playout_timestamp;
}
TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
TEST_F(NetEqDecodingTest, CngFirst) {
uint16_t seq_no = 0;
uint32_t timestamp = 0;
const int kFrameSizeMs = 10;
const int kSampleRateKhz = 16;
const int kSamples = kFrameSizeMs * kSampleRateKhz;
const int kPayloadBytes = kSamples * 2;
const int kCngPeriodMs = 100;
const int kCngPeriodSamples = kCngPeriodMs * kSampleRateKhz;
size_t payload_len;
uint8_t payload[kPayloadBytes] = {0};
WebRtcRTPHeader rtp_info;
PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
ASSERT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_info, payload, payload_len, 0));
++seq_no;
timestamp += kCngPeriodSamples;
// Pull audio once and make sure CNG is played.
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);
EXPECT_EQ(kOutputCNG, type);
// Insert some speech packets.
for (int i = 0; i < 3; ++i) {
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
++seq_no;
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
&num_channels, &type));
ASSERT_EQ(kBlockSize16kHz, out_len);
}
// Verify speech output.
EXPECT_EQ(kOutputNormal, type);
}
} // namespace webrtc