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4f90677527d8577d310ef718d3bba089bc7b36b9
platform-external-webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
minyue 4f90677527 Making NetEq bitexactness test independent on reference files. 
NetEq bitexactness test depended on reference files which differs from platform to platform. This makes it very hard to update Neteq.

New method maintains the ability to save output into files. But it verifies the checksum only. With this, when bitexactness test fails, we can still check closely to the output file if need, but the test becomes much easier to modify.

BUG=

Review-Url: https://codereview.webrtc.org/1928923002
Cr-Commit-Position: refs/heads/master@{#12567}
2016-04-29 18:05:18 +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.
*/
#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
#include <math.h>
#include <stdlib.h>
#include <string.h> // memset
#include <algorithm>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/sha1digest.h"
#include "webrtc/base/stringencode.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/pcm16b.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/typedefs.h"
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
#else
#include "webrtc/audio_coding/neteq/neteq_unittest.pb.h"
#endif
#endif
DEFINE_bool(gen_ref, false, "Generate reference files.");
namespace {
const std::string& PlatformChecksum(const std::string& checksum_general,
const std::string& checksum_android,
const std::string& checksum_win_32,
const std::string& checksum_win_64) {
#ifdef WEBRTC_ANDROID
return checksum_android;
#elif WEBRTC_WIN
#ifdef WEBRTC_ARCH_64_BITS
return checksum_win_64;
#else
return checksum_win_32;
#endif // WEBRTC_ARCH_64_BITS
#else
return checksum_general;
#endif // WEBRTC_WIN
}
bool IsAllZero(const int16_t* buf, size_t buf_length) {
bool all_zero = true;
for (size_t n = 0; n < buf_length && all_zero; ++n)
all_zero = buf[n] == 0;
return all_zero;
}
bool IsAllNonZero(const int16_t* buf, size_t buf_length) {
bool all_non_zero = true;
for (size_t n = 0; n < buf_length && all_non_zero; ++n)
all_non_zero = buf[n] != 0;
return all_non_zero;
}
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
stats->set_packet_loss_rate(stats_raw.packet_loss_rate);
stats->set_packet_discard_rate(stats_raw.packet_discard_rate);
stats->set_expand_rate(stats_raw.expand_rate);
stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
stats->set_preemptive_rate(stats_raw.preemptive_rate);
stats->set_accelerate_rate(stats_raw.accelerate_rate);
stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
stats->set_clockdrift_ppm(stats_raw.clockdrift_ppm);
stats->set_added_zero_samples(stats_raw.added_zero_samples);
stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
}
void Convert(const webrtc::RtcpStatistics& stats_raw,
webrtc::neteq_unittest::RtcpStatistics* stats) {
stats->set_fraction_lost(stats_raw.fraction_lost);
stats->set_cumulative_lost(stats_raw.cumulative_lost);
stats->set_extended_max_sequence_number(
stats_raw.extended_max_sequence_number);
stats->set_jitter(stats_raw.jitter);
}
void AddMessage(FILE* file, rtc::MessageDigest* digest,
const std::string& message) {
int32_t size = message.length();
if (file)
ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
digest->Update(&size, sizeof(size));
if (file)
ASSERT_EQ(static_cast<size_t>(size),
fwrite(message.data(), sizeof(char), size, file));
digest->Update(message.data(), sizeof(char) * size);
}
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
} // namespace
namespace webrtc {
class ResultSink {
public:
explicit ResultSink(const std::string& output_file);
~ResultSink();
template<typename T, size_t n> void AddResult(
const T (&test_results)[n],
size_t length);
void AddResult(const NetEqNetworkStatistics& stats);
void AddResult(const RtcpStatistics& stats);
void VerifyChecksum(const std::string& ref_check_sum);
private:
FILE* output_fp_;
std::unique_ptr<rtc::MessageDigest> digest_;
};
ResultSink::ResultSink(const std::string &output_file)
: output_fp_(nullptr),
digest_(new rtc::Sha1Digest()) {
if (!output_file.empty()) {
output_fp_ = fopen(output_file.c_str(), "wb");
EXPECT_TRUE(output_fp_ != NULL);
}
}
ResultSink::~ResultSink() {
if (output_fp_)
fclose(output_fp_);
}
template<typename T, size_t n>
void ResultSink::AddResult(const T (&test_results)[n], size_t length) {
if (output_fp_) {
ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_));
}
digest_->Update(&test_results, sizeof(T) * length);
}
void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
neteq_unittest::NetEqNetworkStatistics stats;
Convert(stats_raw, &stats);
std::string stats_string;
ASSERT_TRUE(stats.SerializeToString(&stats_string));
AddMessage(output_fp_, digest_.get(), stats_string);
#else
FAIL() << "Writing to reference file requires Proto Buffer.";
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
}
void ResultSink::AddResult(const RtcpStatistics& stats_raw) {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
neteq_unittest::RtcpStatistics stats;
Convert(stats_raw, &stats);
std::string stats_string;
ASSERT_TRUE(stats.SerializeToString(&stats_string));
AddMessage(output_fp_, digest_.get(), stats_string);
#else
FAIL() << "Writing to reference file requires Proto Buffer.";
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
}
void ResultSink::VerifyChecksum(const std::string& checksum) {
std::vector<char> buffer;
buffer.resize(digest_->Size());
digest_->Finish(&buffer[0], buffer.size());
const std::string result = rtc::hex_encode(&buffer[0], digest_->Size());
EXPECT_EQ(checksum, result);
}
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 size_t kBlockSize8kHz = kTimeStepMs * 8;
static const size_t kBlockSize16kHz = kTimeStepMs * 16;
static const size_t kBlockSize32kHz = kTimeStepMs * 32;
static const size_t kBlockSize48kHz = kTimeStepMs * 48;
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();
void DecodeAndCompare(const std::string& rtp_file,
const std::string& output_checksum,
const std::string& network_stats_checksum,
const std::string& rtcp_stats_checksum,
bool gen_ref);
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();
rtc::Optional<uint32_t> PlayoutTimestamp();
NetEq* neteq_;
NetEq::Config config_;
std::unique_ptr<test::RtpFileSource> rtp_source_;
std::unique_ptr<test::Packet> packet_;
unsigned int sim_clock_;
AudioFrame out_frame_;
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 size_t NetEqDecodingTest::kBlockSize8kHz;
const size_t NetEqDecodingTest::kBlockSize16kHz;
const size_t NetEqDecodingTest::kBlockSize32kHz;
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;
}
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(NetEqDecoder::kDecoderPCMu, "pcmu", 0));
// Load PCMa.
ASSERT_EQ(0,
neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCMa, "pcma", 8));
#ifdef WEBRTC_CODEC_ILBC
// Load iLBC.
ASSERT_EQ(
0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderILBC, "ilbc", 102));
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
// Load iSAC.
ASSERT_EQ(
0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac", 103));
#endif
#ifdef WEBRTC_CODEC_ISAC
// Load iSAC SWB.
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISACswb,
"isac-swb", 104));
#endif
#ifdef WEBRTC_CODEC_OPUS
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderOpus,
"opus", 111));
#endif
// Load PCM16B nb.
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16B,
"pcm16-nb", 93));
// Load PCM16B wb.
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bwb,
"pcm16-wb", 94));
// Load PCM16B swb32.
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderPCM16Bswb32kHz,
"pcm16-swb32", 95));
// Load CNG 8 kHz.
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
"cng-nb", 13));
// Load CNG 16 kHz.
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
"cng-wb", 98));
}
void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
}
void NetEqDecodingTest::Process() {
// 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);
#ifndef WEBRTC_CODEC_ISAC
// Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
if (rtp_header.header.payloadType != 104)
#endif
ASSERT_EQ(0, neteq_->InsertPacket(
rtp_header,
rtc::ArrayView<const uint8_t>(
packet_->payload(), packet_->payload_length_bytes()),
static_cast<uint32_t>(packet_->time_ms() *
(output_sample_rate_ / 1000))));
}
// Get next packet.
packet_.reset(rtp_source_->NextPacket());
}
// Get audio from NetEq.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
(out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
(out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
(out_frame_.samples_per_channel_ == kBlockSize48kHz));
output_sample_rate_ = out_frame_.sample_rate_hz_;
EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
// Increase time.
sim_clock_ += kTimeStepMs;
}
void NetEqDecodingTest::DecodeAndCompare(
const std::string& rtp_file,
const std::string& output_checksum,
const std::string& network_stats_checksum,
const std::string& rtcp_stats_checksum,
bool gen_ref) {
OpenInputFile(rtp_file);
std::string ref_out_file =
gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : "";
ResultSink output(ref_out_file);
std::string stat_out_file =
gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
ResultSink network_stats(stat_out_file);
std::string rtcp_out_file =
gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : "";
ResultSink rtcp_stats(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.
ASSERT_NO_FATAL_FAILURE(Process());
ASSERT_NO_FATAL_FAILURE(output.AddResult(
out_frame_.data_, out_frame_.samples_per_channel_));
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
// Process NetworkStatistics.
NetEqNetworkStatistics current_network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
ASSERT_NO_FATAL_FAILURE(network_stats.AddResult(current_network_stats));
// Compare with CurrentDelay, which should be identical.
EXPECT_EQ(current_network_stats.current_buffer_size_ms,
neteq_->CurrentDelayMs());
// Process RTCPstat.
RtcpStatistics current_rtcp_stats;
neteq_->GetRtcpStatistics(&current_rtcp_stats);
ASSERT_NO_FATAL_FAILURE(rtcp_stats.AddResult(current_rtcp_stats));
}
}
SCOPED_TRACE("Check output audio.");
output.VerifyChecksum(output_checksum);
SCOPED_TRACE("Check network stats.");
network_stats.VerifyChecksum(network_stats_checksum);
SCOPED_TRACE("Check rtcp stats.");
rtcp_stats.VerifyChecksum(rtcp_stats_checksum);
}
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.
}
// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) && \
!defined(WEBRTC_ARCH_ARM64) && !defined(UNDEFINED_SANITIZER)
#define MAYBE_TestBitExactness TestBitExactness
#else
#define MAYBE_TestBitExactness DISABLED_TestBitExactness
#endif
TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
const std::string output_checksum = PlatformChecksum(
"f587883b7c371ee8d87dbf1b0f07525af7d959b8",
"a349bd71dba548029b05d1d2a6dc7caafab9a856",
"f587883b7c371ee8d87dbf1b0f07525af7d959b8",
"08266b198e7686b3cd9330813e0d2cd72fc8fdc2");
const std::string network_stats_checksum = PlatformChecksum(
"2cf380a05ee07080bd72471e8ec7777a39644ec9",
"2853ab577fe571adfc7b18f77bbe58f1253d2019",
"2cf380a05ee07080bd72471e8ec7777a39644ec9",
"2cf380a05ee07080bd72471e8ec7777a39644ec9");
const std::string rtcp_stats_checksum = PlatformChecksum(
"b8880bf9fed2487efbddcb8d94b9937a29ae521d",
"f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
"b8880bf9fed2487efbddcb8d94b9937a29ae521d",
"b8880bf9fed2487efbddcb8d94b9937a29ae521d");
DecodeAndCompare(input_rtp_file,
output_checksum,
network_stats_checksum,
rtcp_stats_checksum,
FLAGS_gen_ref);
}
// Disabled for UBSan: https://bugs.chromium.org/p/webrtc/issues/detail?id=5820
#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS) && !defined(UNDEFINED_SANITIZER)
#define MAYBE_TestOpusBitExactness TestOpusBitExactness
#else
#define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness
#endif
TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string output_checksum = PlatformChecksum(
"c23004d91ffbe5e7a1f24620fc89b58c0426040f",
"c23004d91ffbe5e7a1f24620fc89b58c0426040f",
"c23004d91ffbe5e7a1f24620fc89b58c0426040f",
"c23004d91ffbe5e7a1f24620fc89b58c0426040f");
const std::string network_stats_checksum = PlatformChecksum(
"dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
"dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
"dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb",
"dc2d9f584efb0111ebcd71a2c86f1fb09cd8c2bb");
const std::string rtcp_stats_checksum = PlatformChecksum(
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
"e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
DecodeAndCompare(input_rtp_file,
output_checksum,
network_stats_checksum,
rtcp_stats_checksum,
FLAGS_gen_ref);
}
// 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) {
const uint8_t payload[kPayloadBytes] = {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, payload, 0));
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
NetEqNetworkStatistics stats;
EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
// 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. Thus, we are calculating the statistics for a series from 10
// to 300, in steps of 10 ms.
EXPECT_EQ(155, stats.mean_waiting_time_ms);
EXPECT_EQ(155, stats.median_waiting_time_ms);
EXPECT_EQ(10, stats.min_waiting_time_ms);
EXPECT_EQ(300, stats.max_waiting_time_ms);
// Check statistics again and make sure it's been reset.
EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
EXPECT_EQ(-1, stats.mean_waiting_time_ms);
EXPECT_EQ(-1, stats.median_waiting_time_ms);
EXPECT_EQ(-1, stats.min_waiting_time_ms);
EXPECT_EQ(-1, stats.max_waiting_time_ms);
}
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, 0));
++frame_index;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
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, 0));
++frame_index;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
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;
// 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, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
int32_t delay_before = timestamp - *playout_timestamp;
// 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,
rtc::ArrayView<const uint8_t>(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(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_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(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_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(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_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,
rtc::ArrayView<const uint8_t>(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 (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
// 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, 0));
++seq_no;
timestamp += kSamples;
next_input_time_ms += kFrameSizeMs * drift_factor;
}
// Pull out data once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
// 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);
playout_timestamp = PlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
int32_t delay_after = timestamp - *playout_timestamp;
// 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, 0));
EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define MAYBE_DecoderError DecoderError
#else
#define MAYBE_DecoderError DISABLED_DecoderError
#endif
TEST_F(NetEqDecodingTest, MAYBE_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, 0));
// 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 < AudioFrame::kMaxDataSizeSamples; ++i) {
out_frame_.data_[i] = 1;
}
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_));
// Verify that there is a decoder error to check.
EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
enum NetEqDecoderError {
ISAC_LENGTH_MISMATCH = 6730,
ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH = 6640
};
#if defined(WEBRTC_CODEC_ISAC)
EXPECT_EQ(ISAC_LENGTH_MISMATCH, neteq_->LastDecoderError());
#elif defined(WEBRTC_CODEC_ISACFX)
EXPECT_EQ(ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH, neteq_->LastDecoderError());
#endif
// 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_frame_.data_[i]);
}
for (size_t i = kExpectedOutputLength; i < AudioFrame::kMaxDataSizeSamples;
++i) {
std::ostringstream ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(1, out_frame_.data_[i]);
}
}
TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
// 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 < AudioFrame::kMaxDataSizeSamples; ++i) {
out_frame_.data_[i] = 1;
}
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_));
// 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_frame_.data_[i]);
}
// Verify that the sample rate did not change from the initial configuration.
EXPECT_EQ(config_.sample_rate_hz, neteq_->last_output_sample_rate_hz());
}
class NetEqBgnTest : public NetEqDecodingTest {
protected:
virtual void TestCondition(double sum_squared_noise,
bool should_be_faded) = 0;
void CheckBgn(int sampling_rate_hz) {
size_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.
}
AudioFrame output;
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.
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;
uint32_t receive_timestamp = 0;
for (int n = 0; n < 10; ++n) { // Insert few packets and get audio.
auto block = input.GetNextBlock();
ASSERT_EQ(expected_samples_per_channel, block.size());
size_t enc_len_bytes =
WebRtcPcm16b_Encode(block.data(), block.size(), payload);
ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
payload, enc_len_bytes),
receive_timestamp));
output.Reset();
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(1u, output.num_channels_);
ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Next packet.
rtp_info.header.timestamp += expected_samples_per_channel;
rtp_info.header.sequenceNumber++;
receive_timestamp += expected_samples_per_channel;
}
output.Reset();
// 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(&output));
ASSERT_EQ(1u, output.num_channels_);
ASSERT_EQ(expected_samples_per_channel, output.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) {
output.Reset();
memset(output.data_, 1, sizeof(output.data_)); // Set to non-zero.
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(1u, output.num_channels_);
ASSERT_EQ(expected_samples_per_channel, output.samples_per_channel_);
if (output.speech_type_ == AudioFrame::kPLCCNG) {
plc_to_cng = true;
double sum_squared = 0;
for (size_t k = 0;
k < output.num_channels_ * output.samples_per_channel_; ++k)
sum_squared += output.data_[k] * output.data_[k];
TestCondition(sum_squared, n > kFadingThreshold);
} else {
EXPECT_EQ(AudioFrame::kPLC, output.speech_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);
}
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
#define MAYBE_SyncPacketInsert SyncPacketInsert
#else
#define MAYBE_SyncPacketInsert DISABLED_SyncPacketInsert
#endif
TEST_F(NetEqDecodingTest, MAYBE_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(NetEqDecoder::kDecoderPCM16Bwb,
"pcm16-wb", kPcm16WbPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGnb,
"cng-nb", kCngNbPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGwb,
"cng-wb", kCngWbPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb32kHz,
"cng-swb32", kCngSwb32PayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderCNGswb48kHz,
"cng-swb48", kCngSwb48PayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderAVT, "avt",
kAvtPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderRED, "red",
kRedPayloadType));
ASSERT_EQ(0, neteq_->RegisterPayloadType(NetEqDecoder::kDecoderISAC, "isac",
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, 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];
AudioFrame output;
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.
uint32_t receive_timestamp = 0;
for (int n = 0; n < 100; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.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(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
if (n > algorithmic_frame_delay) {
EXPECT_TRUE(IsAllZero(
output.data_, output.samples_per_channel_ * output.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, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output));
if (n >= algorithmic_frame_delay + 1) {
// Expect that this frame contain samples from regular RTP.
EXPECT_TRUE(IsAllNonZero(
output.data_, output.samples_per_channel_ * output.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];
AudioFrame output;
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.
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, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.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, 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(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
EXPECT_TRUE(IsAllNonZero(
output.data_, output.samples_per_channel_ * output.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;
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, 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.
AudioFrame output;
ASSERT_EQ(0, neteq_->GetAudio(&output));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
// Expect delay (in samples) to be less than 2 packets.
rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_LE(timestamp - *playout_timestamp,
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.
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, 0));
++seq_no;
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
// Verify speech output.
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_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, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
// Pull audio once and make sure CNG is played.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_FALSE(PlayoutTimestamp()); // Returns empty value during CNG.
EXPECT_EQ(timestamp - algorithmic_delay_samples,
out_frame_.timestamp_ + out_frame_.samples_per_channel_);
// 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, rtc::ArrayView<const uint8_t>(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(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_FALSE(PlayoutTimestamp()); // Returns empty value during CNG.
EXPECT_EQ(timestamp - algorithmic_delay_samples,
out_frame_.timestamp_ + out_frame_.samples_per_channel_);
}
// Insert speech again.
++seq_no;
timestamp += kCngPeriodSamples;
PopulateRtpInfo(seq_no, timestamp, &rtp_info);
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull audio once and verify that the output is speech again.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
rtc::Optional<uint32_t> playout_timestamp = PlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
*playout_timestamp);
}
rtc::Optional<uint32_t> NetEqDecodingTest::PlayoutTimestamp() {
return neteq_->GetPlayoutTimestamp();
}
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, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
++seq_no;
timestamp += kCngPeriodSamples;
// Pull audio once and make sure CNG is played.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_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, 0));
++seq_no;
timestamp += kSamples;
// Pull audio once.
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
// Verify speech output.
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
}
} // namespace webrtc
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