/*
* 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 "api/neteq/neteq.h"
#include <math.h>
#include <stdlib.h>
#include <string.h> // memset
#include <algorithm>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "absl/flags/flag.h"
#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/test/neteq_decoding_test.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/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"
ABSL_FLAG(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
}
} // namespace
#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("6ae9f643dc3e5f3452d28a772eef7e00e74158bc",
"f4374430e870d66268c1b8e22fb700eb072d567e", "not used",
"6ae9f643dc3e5f3452d28a772eef7e00e74158bc",
"8d73c98645917cdeaaa01c20cf095ccc5a10b2b5");
const std::string network_stats_checksum =
PlatformChecksum("8e50f528f245b7957db20ab406a72d81be60f5f4",
"4260b22ea6d2723b2d573e50d2c1476680c7fa4c", "not used",
"8e50f528f245b7957db20ab406a72d81be60f5f4",
"8e50f528f245b7957db20ab406a72d81be60f5f4");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_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");
const std::string maybe_sse =
"554ad4133934e3920f97575579a46f674683d77c"
"|de316e2bfb15192edb820fe5fb579d11ff5a524b";
const std::string output_checksum = PlatformChecksum(
maybe_sse, "459c356a0ef245ddff381f7d82d205d426ef2002",
"625055e5eb0e6de2c9d170b4494eadc5afab08c8", maybe_sse, maybe_sse);
const std::string network_stats_checksum =
PlatformChecksum("ec29e047b019a86ec06e2c40643143dc1975c69f",
"0c24649824eb7147d4891b0767e86e732dd6ecc8",
"10f3e0b66c6947f78d60301454f2841033a6fcc0",
"ec29e047b019a86ec06e2c40643143dc1975c69f",
"ec29e047b019a86ec06e2c40643143dc1975c69f");
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_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 =
"df5d1d3019bf3764829b84f4fb315721f4adde29"
"|5935d2fad14a69a8b61dbc8e6f2d37c8c0814925";
const std::string output_checksum = PlatformChecksum(
maybe_sse, "551df04e8f45cd99eff28503edf0cf92974898ac",
"709a3f0f380393d3a67bace10e2265b90a6ebbeb", maybe_sse, maybe_sse);
const std::string network_stats_checksum =
"80f5283ac71b27596204210152927666c1732de4";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(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_.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));
}
// 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, 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 = 40;
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 = 60;
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 = 40;
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 = 40;
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));
}
#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));
// 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)));
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);
}
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);
}
TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
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));
++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)));
// 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)));
// 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));
// 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, 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)));
++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));
++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));
}
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)));
}
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();
}
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));
EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
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));
EXPECT_EQ(0, neteq2_->InsertPacket(rtp_info, payload));
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));
// 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));
constexpr uint32_t kRtpTimestamp2 = kRtpTimestamp1 + kPayloadSamples;
PopulateRtpInfo(1, kRtpTimestamp2, &rtp_info);
EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
// 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);
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;
int expected_target_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);
}
// 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_target_delay += neteq_->TargetDelayMs() * 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,
rtc::checked_cast<int>(stats.jitter_buffer_delay_ms));
EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
EXPECT_EQ(expected_target_delay,
rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
}
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};
int expected_target_delay = neteq_->TargetDelayMs() * kSamples;
neteq_->InsertPacket(rtp_info, payload);
bool muted;
neteq_->GetAudio(&out_frame_, &muted);
rtp_info.sequenceNumber += 1;
rtp_info.timestamp += kSamples;
neteq_->InsertPacket(rtp_info, payload);
rtp_info.sequenceNumber += 1;
rtp_info.timestamp += kSamples;
neteq_->InsertPacket(rtp_info, payload);
expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples;
// We have two packets in the buffer and kAccelerate operation will
// extract 20 ms of data.
neteq_->GetAudio(&out_frame_, &muted, NetEq::Operation::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);
EXPECT_EQ(expected_target_delay,
rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
}
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,
/*text_log=*/nullptr, /*neteq_factory=*/nullptr,
/*input=*/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 {
// Helper classes and data types and functions for NetEqOutputDelayTest.
class VectorAudioSink : public AudioSink {
public:
// Does not take ownership of the vector.
VectorAudioSink(std::vector<int16_t>* output_vector) : v_(output_vector) {}
virtual ~VectorAudioSink() = default;
bool WriteArray(const int16_t* audio, size_t num_samples) override {
v_->reserve(v_->size() + num_samples);
for (size_t i = 0; i < num_samples; ++i) {
v_->push_back(audio[i]);
}
return true;
}
private:
std::vector<int16_t>* const v_;
};
struct TestResult {
NetEqLifetimeStatistics lifetime_stats;
NetEqNetworkStatistics network_stats;
absl::optional<uint32_t> playout_timestamp;
int target_delay_ms;
int filtered_current_delay_ms;
int sample_rate_hz;
};
// This class is used as callback object to NetEqTest to collect some stats
// at the end of the simulation.
class SimEndStatsCollector : public NetEqSimulationEndedCallback {
public:
SimEndStatsCollector(TestResult& result) : result_(result) {}
void SimulationEnded(int64_t /*simulation_time_ms*/, NetEq* neteq) override {
result_.playout_timestamp = neteq->GetPlayoutTimestamp();
result_.target_delay_ms = neteq->TargetDelayMs();
result_.filtered_current_delay_ms = neteq->FilteredCurrentDelayMs();
result_.sample_rate_hz = neteq->last_output_sample_rate_hz();
}
private:
TestResult& result_;
};
TestResult DelayLineNetEqTest(int delay_ms,
std::vector<int16_t>* output_vector) {
NetEq::Config config;
config.for_test_no_time_stretching = true;
config.extra_output_delay_ms = delay_ms;
auto codecs = NetEqTest::StandardDecoderMap();
NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
{1, kRtpExtensionAudioLevel},
{3, kRtpExtensionAbsoluteSendTime},
{5, kRtpExtensionTransportSequenceNumber},
{7, kRtpExtensionVideoContentType},
{8, kRtpExtensionVideoTiming}};
std::unique_ptr<NetEqInput> input = std::make_unique<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), 10000));
std::unique_ptr<AudioSink> output =
std::make_unique<VectorAudioSink>(output_vector);
TestResult result;
SimEndStatsCollector stats_collector(result);
NetEqTest::Callbacks callbacks;
callbacks.simulation_ended_callback = &stats_collector;
NetEqTest test(config, CreateBuiltinAudioDecoderFactory(), codecs,
/*text_log=*/nullptr, /*neteq_factory=*/nullptr,
/*input=*/std::move(input_time_limit), std::move(output),
callbacks);
test.Run();
result.lifetime_stats = test.LifetimeStats();
result.network_stats = test.SimulationStats();
return result;
}
} // namespace
// Tests the extra output delay functionality of NetEq.
TEST(NetEqOutputDelayTest, RunTest) {
std::vector<int16_t> output;
const auto result_no_delay = DelayLineNetEqTest(0, &output);
std::vector<int16_t> output_delayed;
constexpr int kDelayMs = 100;
const auto result_delay = DelayLineNetEqTest(kDelayMs, &output_delayed);
// Verify that the loss concealment remains unchanged. The point of the delay
// is to not affect the jitter buffering behavior.
// First verify that there are concealments in the test.
EXPECT_GT(result_no_delay.lifetime_stats.concealed_samples, 0u);
// And that not all of the output is concealment.
EXPECT_GT(result_no_delay.lifetime_stats.total_samples_received,
result_no_delay.lifetime_stats.concealed_samples);
// Now verify that they remain unchanged by the delay.
EXPECT_EQ(result_no_delay.lifetime_stats.concealed_samples,
result_delay.lifetime_stats.concealed_samples);
// Accelerate and pre-emptive expand should also be unchanged.
EXPECT_EQ(result_no_delay.lifetime_stats.inserted_samples_for_deceleration,
result_delay.lifetime_stats.inserted_samples_for_deceleration);
EXPECT_EQ(result_no_delay.lifetime_stats.removed_samples_for_acceleration,
result_delay.lifetime_stats.removed_samples_for_acceleration);
// Verify that delay stats are increased with the delay chain.
EXPECT_EQ(
result_no_delay.lifetime_stats.jitter_buffer_delay_ms +
kDelayMs * result_no_delay.lifetime_stats.jitter_buffer_emitted_count,
result_delay.lifetime_stats.jitter_buffer_delay_ms);
EXPECT_EQ(
result_no_delay.lifetime_stats.jitter_buffer_target_delay_ms +
kDelayMs * result_no_delay.lifetime_stats.jitter_buffer_emitted_count,
result_delay.lifetime_stats.jitter_buffer_target_delay_ms);
EXPECT_EQ(result_no_delay.network_stats.current_buffer_size_ms + kDelayMs,
result_delay.network_stats.current_buffer_size_ms);
EXPECT_EQ(result_no_delay.network_stats.preferred_buffer_size_ms + kDelayMs,
result_delay.network_stats.preferred_buffer_size_ms);
EXPECT_EQ(result_no_delay.network_stats.mean_waiting_time_ms + kDelayMs,
result_delay.network_stats.mean_waiting_time_ms);
EXPECT_EQ(result_no_delay.network_stats.median_waiting_time_ms + kDelayMs,
result_delay.network_stats.median_waiting_time_ms);
EXPECT_EQ(result_no_delay.network_stats.min_waiting_time_ms + kDelayMs,
result_delay.network_stats.min_waiting_time_ms);
EXPECT_EQ(result_no_delay.network_stats.max_waiting_time_ms + kDelayMs,
result_delay.network_stats.max_waiting_time_ms);
ASSERT_TRUE(result_no_delay.playout_timestamp);
ASSERT_TRUE(result_delay.playout_timestamp);
EXPECT_EQ(*result_no_delay.playout_timestamp -
static_cast<uint32_t>(
kDelayMs *
rtc::CheckedDivExact(result_no_delay.sample_rate_hz, 1000)),
*result_delay.playout_timestamp);
EXPECT_EQ(result_no_delay.target_delay_ms + kDelayMs,
result_delay.target_delay_ms);
EXPECT_EQ(result_no_delay.filtered_current_delay_ms + kDelayMs,
result_delay.filtered_current_delay_ms);
// Verify expected delay in decoded signal. The test vector uses 8 kHz sample
// rate, so the delay will be 8 times the delay in ms.
constexpr size_t kExpectedDelaySamples = kDelayMs * 8;
for (size_t i = 0;
i < output.size() && i + kExpectedDelaySamples < output_delayed.size();
++i) {
EXPECT_EQ(output[i], output_delayed[i + kExpectedDelaySamples]);
}
}
// Tests the extra output delay functionality of NetEq when configured via
// field trial.
TEST(NetEqOutputDelayTest, RunTestWithFieldTrial) {
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqExtraDelay/Enabled-50/");
constexpr int kExpectedDelayMs = 50;
std::vector<int16_t> output;
const auto result = DelayLineNetEqTest(0, &output);
// The base delay values are taken from the resuts of the non-delayed case in
// NetEqOutputDelayTest.RunTest above.
EXPECT_EQ(10 + kExpectedDelayMs, result.target_delay_ms);
EXPECT_EQ(24 + kExpectedDelayMs, result.filtered_current_delay_ms);
}
// Set a non-multiple-of-10 value in the field trial, and verify that we don't
// crash, and that the result is rounded down.
TEST(NetEqOutputDelayTest, RunTestWithFieldTrialOddValue) {
test::ScopedFieldTrials field_trial(
"WebRTC-Audio-NetEqExtraDelay/Enabled-103/");
constexpr int kRoundedDelayMs = 100;
std::vector<int16_t> output;
const auto result = DelayLineNetEqTest(0, &output);
// The base delay values are taken from the resuts of the non-delayed case in
// NetEqOutputDelayTest.RunTest above.
EXPECT_EQ(10 + kRoundedDelayMs, result.target_delay_ms);
EXPECT_EQ(24 + kRoundedDelayMs, result.filtered_current_delay_ms);
}
} // namespace test
} // namespace webrtc