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webrtc / src / 413ccc49ecfe459f9c0db96169668571d1485e9b / . / modules / audio_coding / neteq / neteq_unittest.cc
blob: 6c67ca8ea0be1740a33342c26e5a6e439cecfa55 [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.
*/
#include "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 "api/audio/audio_frame.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
#include "modules/audio_coding/neteq/tools/audio_loop.h"
#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "modules/include/module_common_types_public.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/arch.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
// This must come after test/gtest.h
#include "rtc_base/flags.h" // NOLINT(build/include)
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
#else
#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
#endif
WEBRTC_DEFINE_bool(gen_ref, false, "Generate reference files.");
namespace webrtc {
namespace {
const std::string& PlatformChecksum(const std::string& checksum_general,
const std::string& checksum_android_32,
const std::string& checksum_android_64,
const std::string& checksum_win_32,
const std::string& checksum_win_64) {
#if defined(WEBRTC_ANDROID)
#ifdef WEBRTC_ARCH_64_BITS
return checksum_android_64;
#else
return checksum_android_32;
#endif // WEBRTC_ARCH_64_BITS
#elif defined(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
}
#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_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_secondary_discarded_rate(stats_raw.secondary_discarded_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.packets_lost);
stats->set_extended_max_sequence_number(
stats_raw.extended_highest_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
void LoadDecoders(webrtc::NetEq* neteq) {
ASSERT_EQ(true,
neteq->RegisterPayloadType(0, SdpAudioFormat("pcmu", 8000, 1)));
ASSERT_EQ(true,
neteq->RegisterPayloadType(8, SdpAudioFormat("pcma", 8000, 1)));
#ifdef WEBRTC_CODEC_ILBC
ASSERT_EQ(true,
neteq->RegisterPayloadType(102, SdpAudioFormat("ilbc", 8000, 1)));
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
ASSERT_EQ(true,
neteq->RegisterPayloadType(103, SdpAudioFormat("isac", 16000, 1)));
#endif
#ifdef WEBRTC_CODEC_ISAC
ASSERT_EQ(true,
neteq->RegisterPayloadType(104, SdpAudioFormat("isac", 32000, 1)));
#endif
#ifdef WEBRTC_CODEC_OPUS
ASSERT_EQ(true,
neteq->RegisterPayloadType(
111, SdpAudioFormat("opus", 48000, 2, {{"stereo", "0"}})));
#endif
ASSERT_EQ(true,
neteq->RegisterPayloadType(93, SdpAudioFormat("L16", 8000, 1)));
ASSERT_EQ(true,
neteq->RegisterPayloadType(94, SdpAudioFormat("L16", 16000, 1)));
ASSERT_EQ(true,
neteq->RegisterPayloadType(95, SdpAudioFormat("L16", 32000, 1)));
ASSERT_EQ(true,
neteq->RegisterPayloadType(13, SdpAudioFormat("cn", 8000, 1)));
ASSERT_EQ(true,
neteq->RegisterPayloadType(98, SdpAudioFormat("cn", 16000, 1)));
}
} // namespace
class ResultSink {
public:
explicit ResultSink(const std::string& output_file);
~ResultSink();
template <typename T>
void AddResult(const T* test_results, 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_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
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>
void ResultSink::AddResult(const T* test_results, 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());
if (checksum.size() == result.size()) {
EXPECT_EQ(checksum, result);
} else {
// Check result is one the '|'-separated checksums.
EXPECT_NE(checksum.find(result), std::string::npos)
<< result << " should be one of these:\n"
<< checksum;
}
}
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 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,
bool gen_ref);
static void PopulateRtpInfo(int frame_index,
int timestamp,
RTPHeader* rtp_info);
static void PopulateCng(int frame_index,
int timestamp,
RTPHeader* 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();
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_, CreateBuiltinAudioDecoderFactory());
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;
ASSERT_TRUE(neteq_);
LoadDecoders(neteq_);
}
void NetEqDecodingTest::TearDown() {
delete neteq_;
}
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) {
#ifndef WEBRTC_CODEC_ISAC
// Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
if (packet_->header().payloadType != 104)
#endif
ASSERT_EQ(0,
neteq_->InsertPacket(
packet_->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_ = rtp_source_->NextPacket();
}
// Get audio from NetEq.
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_FALSE(muted);
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,
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);
packet_ = rtp_source_->NextPacket();
int i = 0;
uint64_t last_concealed_samples = 0;
uint64_t last_total_samples_received = 0;
while (packet_) {
rtc::StringBuilder 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));
// Verify that liftime stats and network stats report similar loss
// concealment rates.
auto lifetime_stats = neteq_->GetLifetimeStatistics();
const uint64_t delta_concealed_samples =
lifetime_stats.concealed_samples - last_concealed_samples;
last_concealed_samples = lifetime_stats.concealed_samples;
const uint64_t delta_total_samples_received =
lifetime_stats.total_samples_received - last_total_samples_received;
last_total_samples_received = lifetime_stats.total_samples_received;
// The tolerance is 1% but expressed in Q14.
EXPECT_NEAR(
(delta_concealed_samples << 14) / delta_total_samples_received,
current_network_stats.expand_rate, (2 << 14) / 100.0);
}
}
SCOPED_TRACE("Check output audio.");
output.VerifyChecksum(output_checksum);
SCOPED_TRACE("Check network stats.");
network_stats.VerifyChecksum(network_stats_checksum);
}
void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
int timestamp,
RTPHeader* rtp_info) {
rtp_info->sequenceNumber = frame_index;
rtp_info->timestamp = timestamp;
rtp_info->ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info->payloadType = 94; // PCM16b WB codec.
rtp_info->markerBit = 0;
}
void NetEqDecodingTest::PopulateCng(int frame_index,
int timestamp,
RTPHeader* rtp_info,
uint8_t* payload,
size_t* payload_len) {
rtp_info->sequenceNumber = frame_index;
rtp_info->timestamp = timestamp;
rtp_info->ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info->payloadType = 98; // WB CNG.
rtp_info->markerBit = 0;
payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
*payload_len = 1; // Only noise level, no spectral parameters.
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
defined(WEBRTC_CODEC_ILBC) && !defined(WEBRTC_ARCH_ARM64)
#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("9652cee1d6771a9cbfda821ae1bbdb41b0dd4dee",
"54a7e32f163663c0af35bf70bf45cefc24ad62ef", "not used",
"9652cee1d6771a9cbfda821ae1bbdb41b0dd4dee",
"79496b0a1ef0a3824f3ee04789748a461bed643f");
const std::string network_stats_checksum =
PlatformChecksum("c59b1f9f282b6d8733cdff975e3c150ca4a47d51",
"bca95e565996a4ffd6e2ac15736e08843bdca93b", "not used",
"c59b1f9f282b6d8733cdff975e3c150ca4a47d51",
"c59b1f9f282b6d8733cdff975e3c150ca4a47d51");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS)
#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");
// Checksum depends on libopus being compiled with or without SSE.
const std::string maybe_sse =
"14a63b3c7b925c82296be4bafc71bec85f2915c2|"
"2c05677daa968d6c68b92adf4affb7cd9bb4d363";
const std::string output_checksum = PlatformChecksum(
maybe_sse, "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
"5876e52dda90d5ca433c3726555b907b97c86374", maybe_sse, maybe_sse);
const std::string network_stats_checksum =
PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
"fa935a91abc7291db47428a2d7c5361b98713a92",
"42106aa5267300f709f63737707ef07afd9dac61",
"adb3272498e436d1c019cbfd71610e9510c54497",
"adb3272498e436d1c019cbfd71610e9510c54497");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_gen_ref);
}
#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
defined(WEBRTC_CODEC_OPUS)
#define MAYBE_TestOpusDtxBitExactness TestOpusDtxBitExactness
#else
#define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
#endif
TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
const std::string input_rtp_file =
webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");
const std::string maybe_sse =
"713af6c92881f5aab1285765ee6680da9d1c06ce|"
"2ac10c4e79aeedd0df2863b079da5848b40f00b5";
const std::string output_checksum = PlatformChecksum(
maybe_sse, "3ec991b96872123f1554c03c543ca5d518431e46",
"da9f9a2d94e0c2d67342fad4965d7b91cda50b25", maybe_sse, maybe_sse);
const std::string network_stats_checksum =
"bab58dc587d956f326056d7340c96eb9d2d3cc21";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
FLAG_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_.for_test_no_time_stretching = true;
}
void TestJitterBufferDelay(bool apply_packet_loss);
};
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};
RTPHeader rtp_info;
rtp_info.sequenceNumber = rtc::checked_cast<uint16_t>(i);
rtp_info.timestamp = rtc::checked_cast<uint32_t>(i * kSamples);
rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
}
// Pull out all data.
for (size_t i = 0; i < num_frames; ++i) {
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
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};
RTPHeader 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.
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
EXPECT_EQ(-103192, 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};
RTPHeader 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.
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
NetEqNetworkStatistics network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
EXPECT_EQ(110953, 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;
bool muted;
// 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};
RTPHeader 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_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
}
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
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;
RTPHeader 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_, &muted));
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_, &muted));
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_, &muted));
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;
RTPHeader 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};
RTPHeader 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_, &muted));
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 = neteq_->GetPlayoutTimestamp();
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};
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.payloadType = 1; // Not registered as a decoder.
EXPECT_EQ(NetEq::kFail, neteq_->InsertPacket(rtp_info, payload, 0));
}
#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};
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.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.
int16_t* out_frame_data = out_frame_.mutable_data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
out_frame_data[i] = 1;
}
bool muted;
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_FALSE(muted);
// Verify that the first 160 samples are set to 0.
static const int kExpectedOutputLength = 160; // 10 ms at 16 kHz sample rate.
const int16_t* const_out_frame_data = out_frame_.data();
for (int i = 0; i < kExpectedOutputLength; ++i) {
rtc::StringBuilder ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(0, const_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.
int16_t* out_frame_data = out_frame_.mutable_data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; ++i) {
out_frame_data[i] = 1;
}
bool muted;
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_FALSE(muted);
// Verify that the first block of samples is set to 0.
static const int kExpectedOutputLength =
kInitSampleRateHz / 100; // 10 ms at initial sample rate.
const int16_t* const_out_frame_data = out_frame_.data();
for (int i = 0; i < kExpectedOutputLength; ++i) {
rtc::StringBuilder ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the parameter values on failure.
EXPECT_EQ(0, const_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:
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)];
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
rtp_info.payloadType = payload_type;
uint32_t receive_timestamp = 0;
bool muted;
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, &muted));
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.timestamp +=
rtc::checked_cast<uint32_t>(expected_samples_per_channel);
rtp_info.sequenceNumber++;
receive_timestamp +=
rtc::checked_cast<uint32_t>(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, &muted));
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();
// Set to non-zero.
memset(output.mutable_data(), 1, AudioFrame::kMaxDataSizeBytes);
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
ASSERT_FALSE(muted);
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;
const int16_t* output_data = output.data();
for (size_t k = 0;
k < output.num_channels_ * output.samples_per_channel_; ++k)
sum_squared += output_data[k] * output_data[k];
EXPECT_EQ(0, sum_squared);
} else {
EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
}
}
EXPECT_TRUE(plc_to_cng); // Just to be sure that PLC-to-CNG has occurred.
}
};
TEST_F(NetEqBgnTest, RunTest) {
CheckBgn(8000);
CheckBgn(16000);
CheckBgn(32000);
}
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};
RTPHeader 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;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
ASSERT_EQ(1u, output.num_channels_);
// Expect delay (in samples) to be less than 2 packets.
absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
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};
RTPHeader rtp_info;
bool muted;
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_, &muted));
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_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_FALSE(
neteq_->GetPlayoutTimestamp()); // 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_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
EXPECT_FALSE(
neteq_->GetPlayoutTimestamp()); // 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_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(playout_timestamp);
EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
*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};
RTPHeader 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.
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
// Insert some speech packets.
const uint32_t first_speech_timestamp = timestamp;
int timeout_counter = 0;
do {
ASSERT_LT(timeout_counter++, 20) << "Test timed out";
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_, &muted));
ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
} while (!IsNewerTimestamp(out_frame_.timestamp_, first_speech_timestamp));
// Verify speech output.
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
}
class NetEqDecodingTestWithMutedState : public NetEqDecodingTest {
public:
NetEqDecodingTestWithMutedState() : NetEqDecodingTest() {
config_.enable_muted_state = true;
}
protected:
static constexpr size_t kSamples = 10 * 16;
static constexpr size_t kPayloadBytes = kSamples * 2;
void InsertPacket(uint32_t rtp_timestamp) {
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
}
void InsertCngPacket(uint32_t rtp_timestamp) {
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
size_t payload_len;
PopulateCng(0, rtp_timestamp, &rtp_info, payload, &payload_len);
EXPECT_EQ(
NetEq::kOK,
neteq_->InsertPacket(
rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
}
bool GetAudioReturnMuted() {
bool muted;
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
return muted;
}
void GetAudioUntilMuted() {
while (!GetAudioReturnMuted()) {
ASSERT_LT(counter_++, 1000) << "Test timed out";
}
}
void GetAudioUntilNormal() {
bool muted = false;
while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_LT(counter_++, 1000) << "Test timed out";
}
EXPECT_FALSE(muted);
}
int counter_ = 0;
};
// Verifies that NetEq goes in and out of muted state as expected.
TEST_F(NetEqDecodingTestWithMutedState, MutedState) {
// Insert one speech packet.
InsertPacket(0);
// Pull out audio once and expect it not to be muted.
EXPECT_FALSE(GetAudioReturnMuted());
// Pull data until faded out.
GetAudioUntilMuted();
EXPECT_TRUE(out_frame_.muted());
// Verify that output audio is not written during muted mode. Other parameters
// should be correct, though.
AudioFrame new_frame;
int16_t* frame_data = new_frame.mutable_data();
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
frame_data[i] = 17;
}
bool muted;
EXPECT_EQ(0, neteq_->GetAudio(&new_frame, &muted));
EXPECT_TRUE(muted);
EXPECT_TRUE(out_frame_.muted());
for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
EXPECT_EQ(17, frame_data[i]);
}
EXPECT_EQ(out_frame_.timestamp_ + out_frame_.samples_per_channel_,
new_frame.timestamp_);
EXPECT_EQ(out_frame_.samples_per_channel_, new_frame.samples_per_channel_);
EXPECT_EQ(out_frame_.sample_rate_hz_, new_frame.sample_rate_hz_);
EXPECT_EQ(out_frame_.num_channels_, new_frame.num_channels_);
EXPECT_EQ(out_frame_.speech_type_, new_frame.speech_type_);
EXPECT_EQ(out_frame_.vad_activity_, new_frame.vad_activity_);
// Insert new data. Timestamp is corrected for the time elapsed since the last
// packet. Verify that normal operation resumes.
InsertPacket(kSamples * counter_);
GetAudioUntilNormal();
EXPECT_FALSE(out_frame_.muted());
NetEqNetworkStatistics stats;
EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
// NetEqNetworkStatistics::expand_rate tells the fraction of samples that were
// concealment samples, in Q14 (16384 = 100%) .The vast majority should be
// concealment samples in this test.
EXPECT_GT(stats.expand_rate, 14000);
// And, it should be greater than the speech_expand_rate.
EXPECT_GT(stats.expand_rate, stats.speech_expand_rate);
}
// Verifies that NetEq goes out of muted state when given a delayed packet.
TEST_F(NetEqDecodingTestWithMutedState, MutedStateDelayedPacket) {
// Insert one speech packet.
InsertPacket(0);
// Pull out audio once and expect it not to be muted.
EXPECT_FALSE(GetAudioReturnMuted());
// Pull data until faded out.
GetAudioUntilMuted();
// Insert new data. Timestamp is only corrected for the half of the time
// elapsed since the last packet. That is, the new packet is delayed. Verify
// that normal operation resumes.
InsertPacket(kSamples * counter_ / 2);
GetAudioUntilNormal();
}
// Verifies that NetEq goes out of muted state when given a future packet.
TEST_F(NetEqDecodingTestWithMutedState, MutedStateFuturePacket) {
// Insert one speech packet.
InsertPacket(0);
// Pull out audio once and expect it not to be muted.
EXPECT_FALSE(GetAudioReturnMuted());
// Pull data until faded out.
GetAudioUntilMuted();
// Insert new data. Timestamp is over-corrected for the time elapsed since the
// last packet. That is, the new packet is too early. Verify that normal
// operation resumes.
InsertPacket(kSamples * counter_ * 2);
GetAudioUntilNormal();
}
// Verifies that NetEq goes out of muted state when given an old packet.
TEST_F(NetEqDecodingTestWithMutedState, MutedStateOldPacket) {
// Insert one speech packet.
InsertPacket(0);
// Pull out audio once and expect it not to be muted.
EXPECT_FALSE(GetAudioReturnMuted());
// Pull data until faded out.
GetAudioUntilMuted();
EXPECT_NE(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
// Insert packet which is older than the first packet.
InsertPacket(kSamples * (counter_ - 1000));
EXPECT_FALSE(GetAudioReturnMuted());
EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
}
// Verifies that NetEq doesn't enter muted state when CNG mode is active and the
// packet stream is suspended for a long time.
TEST_F(NetEqDecodingTestWithMutedState, DoNotMuteExtendedCngWithoutPackets) {
// Insert one CNG packet.
InsertCngPacket(0);
// Pull 10 seconds of audio (10 ms audio generated per lap).
for (int i = 0; i < 1000; ++i) {
bool muted;
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_FALSE(muted);
}
EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
}
// Verifies that NetEq goes back to normal after a long CNG period with the
// packet stream suspended.
TEST_F(NetEqDecodingTestWithMutedState, RecoverAfterExtendedCngWithoutPackets) {
// Insert one CNG packet.
InsertCngPacket(0);
// Pull 10 seconds of audio (10 ms audio generated per lap).
for (int i = 0; i < 1000; ++i) {
bool muted;
EXPECT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
}
// Insert new data. Timestamp is corrected for the time elapsed since the last
// packet. Verify that normal operation resumes.
InsertPacket(kSamples * counter_);
GetAudioUntilNormal();
}
class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
public:
NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
void SetUp() override {
NetEqDecodingTest::SetUp();
config2_ = config_;
}
void CreateSecondInstance() {
neteq2_.reset(NetEq::Create(config2_, CreateBuiltinAudioDecoderFactory()));
ASSERT_TRUE(neteq2_);
LoadDecoders(neteq2_.get());
}
protected:
std::unique_ptr<NetEq> neteq2_;
NetEq::Config config2_;
};
namespace {
::testing::AssertionResult AudioFramesEqualExceptData(const AudioFrame& a,
const AudioFrame& b) {
if (a.timestamp_ != b.timestamp_)
return ::testing::AssertionFailure() << "timestamp_ diff (" << a.timestamp_
<< " != " << b.timestamp_ << ")";
if (a.sample_rate_hz_ != b.sample_rate_hz_)
return ::testing::AssertionFailure()
<< "sample_rate_hz_ diff (" << a.sample_rate_hz_
<< " != " << b.sample_rate_hz_ << ")";
if (a.samples_per_channel_ != b.samples_per_channel_)
return ::testing::AssertionFailure()
<< "samples_per_channel_ diff (" << a.samples_per_channel_
<< " != " << b.samples_per_channel_ << ")";
if (a.num_channels_ != b.num_channels_)
return ::testing::AssertionFailure()
<< "num_channels_ diff (" << a.num_channels_
<< " != " << b.num_channels_ << ")";
if (a.speech_type_ != b.speech_type_)
return ::testing::AssertionFailure()
<< "speech_type_ diff (" << a.speech_type_
<< " != " << b.speech_type_ << ")";
if (a.vad_activity_ != b.vad_activity_)
return ::testing::AssertionFailure()
<< "vad_activity_ diff (" << a.vad_activity_
<< " != " << b.vad_activity_ << ")";
return ::testing::AssertionSuccess();
}
::testing::AssertionResult AudioFramesEqual(const AudioFrame& a,
const AudioFrame& b) {
::testing::AssertionResult res = AudioFramesEqualExceptData(a, b);
if (!res)
return res;
if (memcmp(a.data(), b.data(),
a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) !=
0) {
return ::testing::AssertionFailure() << "data_ diff";
}
return ::testing::AssertionSuccess();
}
} // namespace
TEST_F(NetEqDecodingTestTwoInstances, CompareMutedStateOnOff) {
ASSERT_FALSE(config_.enable_muted_state);
config2_.enable_muted_state = true;
CreateSecondInstance();
// Insert one speech packet into both NetEqs.
const size_t kSamples = 10 * 16;
const size_t kPayloadBytes = kSamples * 2;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
PopulateRtpInfo(0, 0, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
AudioFrame out_frame1, out_frame2;
bool muted;
for (int i = 0; i < 1000; ++i) {
rtc::StringBuilder ss;
ss << "i = " << i;
SCOPED_TRACE(ss.str()); // Print out the loop iterator on failure.
EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
EXPECT_FALSE(muted);
EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
if (muted) {
EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
} else {
EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
}
}
EXPECT_TRUE(muted);
// Insert new data. Timestamp is corrected for the time elapsed since the last
// packet.
PopulateRtpInfo(0, kSamples * 1000, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload, 0));
int counter = 0;
while (out_frame1.speech_type_ != AudioFrame::kNormalSpeech) {
ASSERT_LT(counter++, 1000) << "Test timed out";
rtc::StringBuilder ss;
ss << "counter = " << counter;
SCOPED_TRACE(ss.str()); // Print out the loop iterator on failure.
EXPECT_EQ(0, neteq_->GetAudio(&out_frame1, &muted));
EXPECT_FALSE(muted);
EXPECT_EQ(0, neteq2_->GetAudio(&out_frame2, &muted));
if (muted) {
EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2));
} else {
EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2));
}
}
EXPECT_FALSE(muted);
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsEmpty) {
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsOneDecoded) {
// Insert one packet with PCM16b WB data (this is what PopulateRtpInfo does by
// default). Make the length 10 ms.
constexpr size_t kPayloadSamples = 16 * 10;
constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
constexpr uint32_t kRtpTimestamp = 0x1234;
PopulateRtpInfo(0, kRtpTimestamp, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp}),
neteq_->LastDecodedTimestamps());
// Nothing decoded on the second call.
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_TRUE(neteq_->LastDecodedTimestamps().empty());
}
TEST_F(NetEqDecodingTest, LastDecodedTimestampsTwoDecoded) {
// Insert two packets with PCM16b WB data (this is what PopulateRtpInfo does
// by default). Make the length 5 ms so that NetEq must decode them both in
// the same GetAudio call.
constexpr size_t kPayloadSamples = 16 * 5;
constexpr size_t kPayloadBytes = 2 * kPayloadSamples;
uint8_t payload[kPayloadBytes] = {0};
RTPHeader rtp_info;
constexpr uint32_t kRtpTimestamp1 = 0x1234;
PopulateRtpInfo(0, kRtpTimestamp1, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + kPayloadSamples;
PopulateRtpInfo(1, kRtpTimestamp2, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
// Pull out data once.
AudioFrame output;
bool muted;
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(std::vector<uint32_t>({kRtpTimestamp1, kRtpTimestamp2}),
neteq_->LastDecodedTimestamps());
}
TEST_F(NetEqDecodingTest, TestConcealmentEvents) {
const int kNumConcealmentEvents = 19;
const size_t kSamples = 10 * 16;
const size_t kPayloadBytes = kSamples * 2;
int seq_no = 0;
RTPHeader rtp_info;
rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
const uint8_t payload[kPayloadBytes] = {0};
bool muted;
for (int i = 0; i < kNumConcealmentEvents; i++) {
// Insert some packets of 10 ms size.
for (int j = 0; j < 10; j++) {
rtp_info.sequenceNumber = seq_no++;
rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
neteq_->InsertPacket(rtp_info, payload, 0);
neteq_->GetAudio(&out_frame_, &muted);
}
// Lose a number of packets.
int num_lost = 1 + i;
for (int j = 0; j < num_lost; j++) {
seq_no++;
neteq_->GetAudio(&out_frame_, &muted);
}
}
// Check number of concealment events.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(kNumConcealmentEvents, static_cast<int>(stats.concealment_events));
}
// Test that the jitter buffer delay stat is computed correctly.
void NetEqDecodingTestFaxMode::TestJitterBufferDelay(bool apply_packet_loss) {
const int kNumPackets = 10;
const int kDelayInNumPackets = 2;
const int kPacketLenMs = 10; // All packets are of 10 ms size.
const size_t kSamples = kPacketLenMs * 16;
const size_t kPayloadBytes = kSamples * 2;
RTPHeader rtp_info;
rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
const uint8_t payload[kPayloadBytes] = {0};
bool muted;
int packets_sent = 0;
int packets_received = 0;
int expected_delay = 0;
uint64_t expected_emitted_count = 0;
while (packets_received < kNumPackets) {
// Insert packet.
if (packets_sent < kNumPackets) {
rtp_info.sequenceNumber = packets_sent++;
rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
neteq_->InsertPacket(rtp_info, payload, 0);
}
// Get packet.
if (packets_sent > kDelayInNumPackets) {
neteq_->GetAudio(&out_frame_, &muted);
packets_received++;
// The delay reported by the jitter buffer never exceeds
// the number of samples previously fetched with GetAudio
// (hence the min()).
int packets_delay = std::min(packets_received, kDelayInNumPackets + 1);
// The increase of the expected delay is the product of
// the current delay of the jitter buffer in ms * the
// number of samples that are sent for play out.
int current_delay_ms = packets_delay * kPacketLenMs;
expected_delay += current_delay_ms * kSamples;
expected_emitted_count += kSamples;
}
}
if (apply_packet_loss) {
// Extra call to GetAudio to cause concealment.
neteq_->GetAudio(&out_frame_, &muted);
}
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(expected_delay, static_cast<int>(stats.jitter_buffer_delay_ms));
EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
TestJitterBufferDelay(false);
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
TestJitterBufferDelay(true);
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithAcceleration) {
const int kPacketLenMs = 10; // All packets are of 10 ms size.
const size_t kSamples = kPacketLenMs * 16;
const size_t kPayloadBytes = kSamples * 2;
RTPHeader rtp_info;
rtp_info.ssrc = 0x1234; // Just an arbitrary SSRC.
rtp_info.payloadType = 94; // PCM16b WB codec.
rtp_info.markerBit = 0;
const uint8_t payload[kPayloadBytes] = {0};
neteq_->InsertPacket(rtp_info, payload, 0);
bool muted;
neteq_->GetAudio(&out_frame_, &muted);
rtp_info.sequenceNumber += 1;
rtp_info.timestamp += kSamples;
neteq_->InsertPacket(rtp_info, payload, 0);
rtp_info.sequenceNumber += 1;
rtp_info.timestamp += kSamples;
neteq_->InsertPacket(rtp_info, payload, 0);
// We have two packets in the buffer and kAccelerate operation will
// extract 20 ms of data.
neteq_->GetAudio(&out_frame_, &muted, Operations::kAccelerate);
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(10 * kSamples * 3, stats.jitter_buffer_delay_ms);
EXPECT_EQ(kSamples * 3, stats.jitter_buffer_emitted_count);
}
namespace test {
TEST(NetEqNoTimeStretchingMode, RunTest) {
NetEq::Config config;
config.for_test_no_time_stretching = true;
auto codecs = NetEqTest::StandardDecoderMap();
NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
{1, kRtpExtensionAudioLevel},
{3, kRtpExtensionAbsoluteSendTime},
{5, kRtpExtensionTransportSequenceNumber},
{7, kRtpExtensionVideoContentType},
{8, kRtpExtensionVideoTiming}};
std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
rtp_ext_map, absl::nullopt /*No SSRC filter*/));
std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
new TimeLimitedNetEqInput(std::move(input), 20000));
std::unique_ptr<AudioSink> output(new VoidAudioSink);
NetEqTest::Callbacks callbacks;
NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs, nullptr,
std::move(input_time_limit), std::move(output), callbacks);
test.Run();
const auto stats = test.SimulationStats();
EXPECT_EQ(0, stats.accelerate_rate);
EXPECT_EQ(0, stats.preemptive_rate);
}
} // namespace test
} // namespace webrtc