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platform-external-webrtc/webrtc/modules/audio_coding/neteq4/neteq_unittest.cc
henrik.lundin@webrtc.org d94659dc27 Initial upload of NetEq4 
This is the first public upload of the new NetEq, version 4.

It has been through extensive internal review during the course of
the project.

TEST=trybots

Review URL: https://webrtc-codereview.appspot.com/1073005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-29 12:09:21 +00:00

695 lines
24 KiB
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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* This file includes unit tests for NetEQ.
*/
#include "webrtc/modules/audio_coding/neteq4/interface/neteq.h"
#include <stdlib.h>
#include <string.h> // memset
#include <string>
#include <vector>
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq4/test/NETEQTEST_RTPpacket.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RefFiles {
public:
RefFiles(const std::string& input_file, const std::string& output_file);
~RefFiles();
template<class T> void ProcessReference(const T& test_results);
template<typename T, size_t n> void ProcessReference(
const T (&test_results)[n],
size_t length);
template<typename T, size_t n> void WriteToFile(
const T (&test_results)[n],
size_t length);
template<typename T, size_t n> void ReadFromFileAndCompare(
const T (&test_results)[n],
size_t length);
void WriteToFile(const 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);
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));
// Load iLBC.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
// Load iSAC.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
// Load iSAC SWB.
ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
// 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_));
}
// RecOut
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.
}
TEST_F(NetEqDecodingTest, TestBitExactness) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/neteq_universal.rtp";
const std::string kInputRefFile =
webrtc::test::ResourcePath("neteq_universal_ref", "pcm");
DecodeAndCompare(kInputRtpFile, kInputRefFile);
}
TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
"resources/neteq_universal.rtp";
const std::string kNetworkStatRefFile =
webrtc::test::ResourcePath("neteq_network_stats", "dat");
const std::string kRtcpStatRefFile =
webrtc::test::ResourcePath("neteq_rtcp_stats", "dat");
DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
}
// TODO(hlundin): Re-enable test once the statistics interface is up and again.
TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
// Use fax mode to avoid time-scaling. This is to simplify the testing of
// packet waiting times in the packet buffer.
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, 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, 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, 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, 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, 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, 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]);
}
}
} // namespace
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