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webrtc / src / 46ba49c622d305dff756d4d228d9ec953a2aea4e / . / webrtc / modules / audio_coding / neteq / neteq_unittest.cc
blob: 24a8774ae4797b40ec1e388be5fac5e1d4069877 [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 "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
void LoadDecoders(webrtc::NetEq* neteq) {
// Load PCMu.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderPCMu,
"pcmu", 0));
// Load PCMa.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderPCMa,
"pcma", 8));
#ifdef WEBRTC_CODEC_ILBC
// Load iLBC.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderILBC,
"ilbc", 102));
#endif
#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
// Load iSAC.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderISAC,
"isac", 103));
#endif
#ifdef WEBRTC_CODEC_ISAC
// Load iSAC SWB.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderISACswb,
"isac-swb", 104));
#endif
#ifdef WEBRTC_CODEC_OPUS
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderOpus,
"opus", 111));
#endif
// Load PCM16B nb.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderPCM16B,
"pcm16-nb", 93));
// Load PCM16B wb.
ASSERT_EQ(0, neteq->RegisterPayloadType(
webrtc::NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb", 94));
// Load PCM16B swb32.
ASSERT_EQ(
0, neteq->RegisterPayloadType(
webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32", 95));
// Load CNG 8 kHz.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderCNGnb,
"cng-nb", 13));
// Load CNG 16 kHz.
ASSERT_EQ(0, neteq->RegisterPayloadType(webrtc::NetEqDecoder::kDecoderCNGwb,
"cng-wb", 98));
}
} // 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 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(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) {
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_ = 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,
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_ = 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.
}
#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)
#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(
"472ebe1126f41fdb6b5c63c87f625a52e7604e49",
"d2a6b6ff54b340cf9f961c7f07768d86b3761073",
"472ebe1126f41fdb6b5c63c87f625a52e7604e49",
"f9749813dbc3fb59dae761de518fec65b8407c5b");
const std::string network_stats_checksum = PlatformChecksum(
"2cf380a05ee07080bd72471e8ec7777a39644ec9",
"01be67dc4c3b8e74743a45cbd8684c0535dec9ad",
"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);
}
#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \
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");
const std::string output_checksum = PlatformChecksum(
"19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
"19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
"19ad24b4a1eb7a9620e6da09f98c49aa5792ade4",
"19ad24b4a1eb7a9620e6da09f98c49aa5792ade4");
const std::string network_stats_checksum = PlatformChecksum(
"6eab76efbde753d4dde38983445ca16b4ce59b39",
"6eab76efbde753d4dde38983445ca16b4ce59b39",
"6eab76efbde753d4dde38983445ca16b4ce59b39",
"6eab76efbde753d4dde38983445ca16b4ce59b39");
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) {
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};
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.
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(-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.
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(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;
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};
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_, &muted));
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_, &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;
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_, &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 = 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;
}
bool muted;
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&out_frame_, &muted));
ASSERT_FALSE(muted);
// 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;
}
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.
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;
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.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, &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();
memset(output.data_, 1, sizeof(output.data_)); // Set to non-zero.
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;
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;
bool muted;
for (int n = 0; n < 100; ++n) {
ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
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, &muted));
ASSERT_FALSE(muted);
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, &muted));
ASSERT_FALSE(muted);
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;
bool muted;
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, &muted));
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, &muted));
ASSERT_FALSE(muted);
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;
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.
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;
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(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_, &muted));
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_, &muted));
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.
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.
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_);
}
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};
WebRtcRTPHeader rtp_info;
PopulateRtpInfo(0, rtp_timestamp, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 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();
// Verify that output audio is not written during muted mode. Other parameters
// should be correct, though.
AudioFrame new_frame;
for (auto& d : new_frame.data_) {
d = 17;
}
bool muted;
EXPECT_EQ(0, neteq_->GetAudio(&new_frame, &muted));
EXPECT_TRUE(muted);
for (auto d : new_frame.data_) {
EXPECT_EQ(17, d);
}
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();
}
// 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_);
}
class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
public:
NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
void SetUp() override {
NetEqDecodingTest::SetUp();
config2_ = config_;
}
void CreateSecondInstance() {
neteq2_.reset(NetEq::Create(config2_));
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])) != 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};
WebRtcRTPHeader 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) {
std::ostringstream 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";
std::ostringstream 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);
}
} // namespace webrtc