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webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
Henrik Lundin 2a8bd090a3 NetEq: Create an audio interruption metric 
This CL adds a new metric to NetEq, which logs whenever a loss
concealment event has lasted longer than 150 ms (an "interruption").
The number of such events, as well as the sum length of them, is kept
in a SampleCounter, which can be queried at any time.

Any initial PLC at the beginning of a call, before the first packet is
decoded, is ignored.

Unit tests and piping to neteq_rtpplay are included.

Bug: webrtc:10549
Change-Id: I8a224a34254c47c74317617f420f6de997232d88
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132796
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27781}
2019-04-26 09:48:05 +00:00

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/*
* Copyright (c) 2012 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 <memory>
#include "absl/memory/memory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "modules/audio_coding/neteq/accelerate.h"
#include "modules/audio_coding/neteq/expand.h"
#include "modules/audio_coding/neteq/histogram.h"
#include "modules/audio_coding/neteq/include/neteq.h"
#include "modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
#include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "modules/audio_coding/neteq/mock/mock_delay_manager.h"
#include "modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
#include "modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
#include "modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
#include "modules/audio_coding/neteq/mock/mock_packet_buffer.h"
#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
#include "modules/audio_coding/neteq/neteq_impl.h"
#include "modules/audio_coding/neteq/preemptive_expand.h"
#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/timestamp_scaler.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "test/audio_decoder_proxy_factory.h"
#include "test/function_audio_decoder_factory.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/mock_audio_decoder.h"
#include "test/mock_audio_decoder_factory.h"
using ::testing::_;
using ::testing::AtLeast;
using ::testing::DoAll;
using ::testing::InSequence;
using ::testing::Invoke;
using ::testing::IsNull;
using ::testing::Pointee;
using ::testing::Return;
using ::testing::ReturnNull;
using ::testing::SetArgPointee;
using ::testing::SetArrayArgument;
using ::testing::WithArg;
namespace webrtc {
// This function is called when inserting a packet list into the mock packet
// buffer. The purpose is to delete all inserted packets properly, to avoid
// memory leaks in the test.
int DeletePacketsAndReturnOk(PacketList* packet_list) {
packet_list->clear();
return PacketBuffer::kOK;
}
class NetEqImplTest : public ::testing::Test {
protected:
NetEqImplTest() { config_.sample_rate_hz = 8000; }
void CreateInstance(
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
ASSERT_TRUE(decoder_factory);
NetEqImpl::Dependencies deps(config_, decoder_factory);
// Get a local pointer to NetEq's TickTimer object.
tick_timer_ = deps.tick_timer.get();
if (use_mock_buffer_level_filter_) {
std::unique_ptr<MockBufferLevelFilter> mock(new MockBufferLevelFilter);
mock_buffer_level_filter_ = mock.get();
deps.buffer_level_filter = std::move(mock);
}
buffer_level_filter_ = deps.buffer_level_filter.get();
if (use_mock_decoder_database_) {
std::unique_ptr<MockDecoderDatabase> mock(new MockDecoderDatabase);
mock_decoder_database_ = mock.get();
EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
.WillOnce(ReturnNull());
deps.decoder_database = std::move(mock);
}
decoder_database_ = deps.decoder_database.get();
if (use_mock_delay_peak_detector_) {
std::unique_ptr<MockDelayPeakDetector> mock(
new MockDelayPeakDetector(tick_timer_, config_.enable_rtx_handling));
mock_delay_peak_detector_ = mock.get();
EXPECT_CALL(*mock_delay_peak_detector_, Reset()).Times(1);
deps.delay_peak_detector = std::move(mock);
}
delay_peak_detector_ = deps.delay_peak_detector.get();
if (use_mock_delay_manager_) {
std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
config_.max_packets_in_buffer, config_.min_delay_ms, 1020054733,
DelayManager::HistogramMode::INTER_ARRIVAL_TIME,
config_.enable_rtx_handling, delay_peak_detector_, tick_timer_,
deps.stats.get(), absl::make_unique<Histogram>(50, 32745)));
mock_delay_manager_ = mock.get();
EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
deps.delay_manager = std::move(mock);
}
delay_manager_ = deps.delay_manager.get();
if (use_mock_dtmf_buffer_) {
std::unique_ptr<MockDtmfBuffer> mock(
new MockDtmfBuffer(config_.sample_rate_hz));
mock_dtmf_buffer_ = mock.get();
deps.dtmf_buffer = std::move(mock);
}
dtmf_buffer_ = deps.dtmf_buffer.get();
if (use_mock_dtmf_tone_generator_) {
std::unique_ptr<MockDtmfToneGenerator> mock(new MockDtmfToneGenerator);
mock_dtmf_tone_generator_ = mock.get();
deps.dtmf_tone_generator = std::move(mock);
}
dtmf_tone_generator_ = deps.dtmf_tone_generator.get();
if (use_mock_packet_buffer_) {
std::unique_ptr<MockPacketBuffer> mock(
new MockPacketBuffer(config_.max_packets_in_buffer, tick_timer_));
mock_packet_buffer_ = mock.get();
deps.packet_buffer = std::move(mock);
}
packet_buffer_ = deps.packet_buffer.get();
if (use_mock_payload_splitter_) {
std::unique_ptr<MockRedPayloadSplitter> mock(new MockRedPayloadSplitter);
mock_payload_splitter_ = mock.get();
deps.red_payload_splitter = std::move(mock);
}
red_payload_splitter_ = deps.red_payload_splitter.get();
deps.timestamp_scaler = std::unique_ptr<TimestampScaler>(
new TimestampScaler(*deps.decoder_database.get()));
neteq_.reset(new NetEqImpl(config_, std::move(deps)));
ASSERT_TRUE(neteq_ != NULL);
}
void CreateInstance() { CreateInstance(CreateBuiltinAudioDecoderFactory()); }
void UseNoMocks() {
ASSERT_TRUE(neteq_ == NULL) << "Must call UseNoMocks before CreateInstance";
use_mock_buffer_level_filter_ = false;
use_mock_decoder_database_ = false;
use_mock_delay_peak_detector_ = false;
use_mock_delay_manager_ = false;
use_mock_dtmf_buffer_ = false;
use_mock_dtmf_tone_generator_ = false;
use_mock_packet_buffer_ = false;
use_mock_payload_splitter_ = false;
}
virtual ~NetEqImplTest() {
if (use_mock_buffer_level_filter_) {
EXPECT_CALL(*mock_buffer_level_filter_, Die()).Times(1);
}
if (use_mock_decoder_database_) {
EXPECT_CALL(*mock_decoder_database_, Die()).Times(1);
}
if (use_mock_delay_manager_) {
EXPECT_CALL(*mock_delay_manager_, Die()).Times(1);
}
if (use_mock_delay_peak_detector_) {
EXPECT_CALL(*mock_delay_peak_detector_, Die()).Times(1);
}
if (use_mock_dtmf_buffer_) {
EXPECT_CALL(*mock_dtmf_buffer_, Die()).Times(1);
}
if (use_mock_dtmf_tone_generator_) {
EXPECT_CALL(*mock_dtmf_tone_generator_, Die()).Times(1);
}
if (use_mock_packet_buffer_) {
EXPECT_CALL(*mock_packet_buffer_, Die()).Times(1);
}
}
void TestDtmfPacket(int sample_rate_hz) {
const size_t kPayloadLength = 4;
const uint8_t kPayloadType = 110;
const uint32_t kReceiveTime = 17;
const int kSampleRateHz = 16000;
config_.sample_rate_hz = kSampleRateHz;
UseNoMocks();
CreateInstance();
// Event: 2, E bit, Volume: 17, Length: 4336.
uint8_t payload[kPayloadLength] = { 0x02, 0x80 + 0x11, 0x10, 0xF0 };
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_TRUE(neteq_->RegisterPayloadType(
kPayloadType, SdpAudioFormat("telephone-event", sample_rate_hz, 1)));
// Insert first packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize =
static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_FALSE(muted);
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Verify first 64 samples of actual output.
const std::vector<int16_t> kOutput({
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, -1578, -2816, -3460, -3403, -2709, -1594,
-363, 671, 1269, 1328, 908, 202, -513, -964, -955, -431, 504, 1617,
2602, 3164, 3101, 2364, 1073, -511, -2047, -3198, -3721, -3525, -2688,
-1440, -99, 1015, 1663, 1744, 1319, 588, -171, -680, -747, -315, 515,
1512, 2378, 2828, 2674, 1877, 568, -986, -2446, -3482, -3864, -3516,
-2534, -1163 });
ASSERT_GE(kMaxOutputSize, kOutput.size());
EXPECT_TRUE(std::equal(kOutput.begin(), kOutput.end(), output.data()));
}
std::unique_ptr<NetEqImpl> neteq_;
NetEq::Config config_;
TickTimer* tick_timer_ = nullptr;
MockBufferLevelFilter* mock_buffer_level_filter_ = nullptr;
BufferLevelFilter* buffer_level_filter_ = nullptr;
bool use_mock_buffer_level_filter_ = true;
MockDecoderDatabase* mock_decoder_database_ = nullptr;
DecoderDatabase* decoder_database_ = nullptr;
bool use_mock_decoder_database_ = true;
MockDelayPeakDetector* mock_delay_peak_detector_ = nullptr;
DelayPeakDetector* delay_peak_detector_ = nullptr;
bool use_mock_delay_peak_detector_ = true;
MockDelayManager* mock_delay_manager_ = nullptr;
DelayManager* delay_manager_ = nullptr;
bool use_mock_delay_manager_ = true;
MockDtmfBuffer* mock_dtmf_buffer_ = nullptr;
DtmfBuffer* dtmf_buffer_ = nullptr;
bool use_mock_dtmf_buffer_ = true;
MockDtmfToneGenerator* mock_dtmf_tone_generator_ = nullptr;
DtmfToneGenerator* dtmf_tone_generator_ = nullptr;
bool use_mock_dtmf_tone_generator_ = true;
MockPacketBuffer* mock_packet_buffer_ = nullptr;
PacketBuffer* packet_buffer_ = nullptr;
bool use_mock_packet_buffer_ = true;
MockRedPayloadSplitter* mock_payload_splitter_ = nullptr;
RedPayloadSplitter* red_payload_splitter_ = nullptr;
bool use_mock_payload_splitter_ = true;
};
// This tests the interface class NetEq.
// TODO(hlundin): Move to separate file?
TEST(NetEq, CreateAndDestroy) {
NetEq::Config config;
NetEq* neteq = NetEq::Create(config, CreateBuiltinAudioDecoderFactory());
delete neteq;
}
TEST_F(NetEqImplTest, RegisterPayloadType) {
CreateInstance();
constexpr int rtp_payload_type = 0;
const SdpAudioFormat format("pcmu", 8000, 1);
EXPECT_CALL(*mock_decoder_database_,
RegisterPayload(rtp_payload_type, format));
neteq_->RegisterPayloadType(rtp_payload_type, format);
}
TEST_F(NetEqImplTest, RemovePayloadType) {
CreateInstance();
uint8_t rtp_payload_type = 0;
EXPECT_CALL(*mock_decoder_database_, Remove(rtp_payload_type))
.WillOnce(Return(DecoderDatabase::kDecoderNotFound));
// Check that kOK is returned when database returns kDecoderNotFound, because
// removing a payload type that was never registered is not an error.
EXPECT_EQ(NetEq::kOK, neteq_->RemovePayloadType(rtp_payload_type));
}
TEST_F(NetEqImplTest, RemoveAllPayloadTypes) {
CreateInstance();
EXPECT_CALL(*mock_decoder_database_, RemoveAll()).WillOnce(Return());
neteq_->RemoveAllPayloadTypes();
}
TEST_F(NetEqImplTest, InsertPacket) {
CreateInstance();
const size_t kPayloadLength = 100;
const uint8_t kPayloadType = 0;
const uint16_t kFirstSequenceNumber = 0x1234;
const uint32_t kFirstTimestamp = 0x12345678;
const uint32_t kSsrc = 0x87654321;
const uint32_t kFirstReceiveTime = 17;
uint8_t payload[kPayloadLength] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = kFirstSequenceNumber;
rtp_header.timestamp = kFirstTimestamp;
rtp_header.ssrc = kSsrc;
Packet fake_packet;
fake_packet.payload_type = kPayloadType;
fake_packet.sequence_number = kFirstSequenceNumber;
fake_packet.timestamp = kFirstTimestamp;
rtc::scoped_refptr<MockAudioDecoderFactory> mock_decoder_factory(
new rtc::RefCountedObject<MockAudioDecoderFactory>);
EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
.WillOnce(Invoke([&](const SdpAudioFormat& format,
absl::optional<AudioCodecPairId> codec_pair_id,
std::unique_ptr<AudioDecoder>* dec) {
EXPECT_EQ("pcmu", format.name);
std::unique_ptr<MockAudioDecoder> mock_decoder(new MockAudioDecoder);
EXPECT_CALL(*mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(*mock_decoder, SampleRateHz()).WillRepeatedly(Return(8000));
// BWE update function called with first packet.
EXPECT_CALL(*mock_decoder,
IncomingPacket(_, kPayloadLength, kFirstSequenceNumber,
kFirstTimestamp, kFirstReceiveTime));
// BWE update function called with second packet.
EXPECT_CALL(
*mock_decoder,
IncomingPacket(_, kPayloadLength, kFirstSequenceNumber + 1,
kFirstTimestamp + 160, kFirstReceiveTime + 155));
EXPECT_CALL(*mock_decoder, Die()).Times(1); // Called when deleted.
*dec = std::move(mock_decoder);
}));
DecoderDatabase::DecoderInfo info(SdpAudioFormat("pcmu", 8000, 1),
absl::nullopt, mock_decoder_factory);
// Expectations for decoder database.
EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(&info));
// Expectations for packet buffer.
EXPECT_CALL(*mock_packet_buffer_, Empty())
.WillOnce(Return(false)); // Called once after first packet is inserted.
EXPECT_CALL(*mock_packet_buffer_, Flush())
.Times(1);
EXPECT_CALL(*mock_packet_buffer_, InsertPacketList(_, _, _, _, _))
.Times(2)
.WillRepeatedly(DoAll(SetArgPointee<2>(kPayloadType),
WithArg<0>(Invoke(DeletePacketsAndReturnOk))));
// SetArgPointee<2>(kPayloadType) means that the third argument (zero-based
// index) is a pointer, and the variable pointed to is set to kPayloadType.
// Also invoke the function DeletePacketsAndReturnOk to properly delete all
// packets in the list (to avoid memory leaks in the test).
EXPECT_CALL(*mock_packet_buffer_, PeekNextPacket())
.Times(1)
.WillOnce(Return(&fake_packet));
// Expectations for DTMF buffer.
EXPECT_CALL(*mock_dtmf_buffer_, Flush())
.Times(1);
// Expectations for delay manager.
{
// All expectations within this block must be called in this specific order.
InSequence sequence; // Dummy variable.
// Expectations when the first packet is inserted.
EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
.Times(2)
.WillRepeatedly(Return(-1));
EXPECT_CALL(*mock_delay_manager_, set_last_pack_cng_or_dtmf(0))
.Times(1);
EXPECT_CALL(*mock_delay_manager_, ResetPacketIatCount()).Times(1);
// Expectations when the second packet is inserted. Slightly different.
EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
.WillOnce(Return(0));
EXPECT_CALL(*mock_delay_manager_, SetPacketAudioLength(30))
.WillOnce(Return(0));
}
// Insert first packet.
neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime);
// Insert second packet.
rtp_header.timestamp += 160;
rtp_header.sequenceNumber += 1;
neteq_->InsertPacket(rtp_header, payload, kFirstReceiveTime + 155);
}
TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
UseNoMocks();
CreateInstance();
const int kPayloadLengthSamples = 80;
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit.
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
// Insert packets. The buffer should not flush.
for (size_t i = 1; i <= config_.max_packets_in_buffer; ++i) {
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.timestamp += kPayloadLengthSamples;
rtp_header.sequenceNumber += 1;
EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
}
// Insert one more packet and make sure the buffer got flushed. That is, it
// should only hold one single packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
const Packet* test_packet = packet_buffer_->PeekNextPacket();
EXPECT_EQ(rtp_header.timestamp, test_packet->timestamp);
EXPECT_EQ(rtp_header.sequenceNumber, test_packet->sequence_number);
}
TEST_F(NetEqImplTest, TestDtmfPacketAVT) {
TestDtmfPacket(8000);
}
TEST_F(NetEqImplTest, TestDtmfPacketAVT16kHz) {
TestDtmfPacket(16000);
}
TEST_F(NetEqImplTest, TestDtmfPacketAVT32kHz) {
TestDtmfPacket(32000);
}
TEST_F(NetEqImplTest, TestDtmfPacketAVT48kHz) {
TestDtmfPacket(48000);
}
// This test verifies that timestamps propagate from the incoming packets
// through to the sync buffer and to the playout timestamp.
TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
// This is a dummy decoder that produces as many output samples as the input
// has bytes. The output is an increasing series, starting at 1 for the first
// sample, and then increasing by 1 for each sample.
class CountingSamplesDecoder : public AudioDecoder {
public:
CountingSamplesDecoder() : next_value_(1) {}
// Produce as many samples as input bytes (|encoded_len|).
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int /* sample_rate_hz */,
int16_t* decoded,
SpeechType* speech_type) override {
for (size_t i = 0; i < encoded_len; ++i) {
decoded[i] = next_value_++;
}
*speech_type = kSpeech;
return rtc::checked_cast<int>(encoded_len);
}
void Reset() override { next_value_ = 1; }
int SampleRateHz() const override { return kSampleRateHz; }
size_t Channels() const override { return 1; }
uint16_t next_value() const { return next_value_; }
private:
int16_t next_value_;
} decoder_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory =
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&decoder_);
UseNoMocks();
CreateInstance(decoder_factory);
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_FALSE(muted);
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Start with a simple check that the fake decoder is behaving as expected.
EXPECT_EQ(kPayloadLengthSamples,
static_cast<size_t>(decoder_.next_value() - 1));
// The value of the last of the output samples is the same as the number of
// samples played from the decoded packet. Thus, this number + the RTP
// timestamp should match the playout timestamp.
// Wrap the expected value in an absl::optional to compare them as such.
EXPECT_EQ(
absl::optional<uint32_t>(rtp_header.timestamp +
output.data()[output.samples_per_channel_ - 1]),
neteq_->GetPlayoutTimestamp());
// Check the timestamp for the last value in the sync buffer. This should
// be one full frame length ahead of the RTP timestamp.
const SyncBuffer* sync_buffer = neteq_->sync_buffer_for_test();
ASSERT_TRUE(sync_buffer != NULL);
EXPECT_EQ(rtp_header.timestamp + kPayloadLengthSamples,
sync_buffer->end_timestamp());
// Check that the number of samples still to play from the sync buffer add
// up with what was already played out.
EXPECT_EQ(
kPayloadLengthSamples - output.data()[output.samples_per_channel_ - 1],
sync_buffer->FutureLength());
}
TEST_F(NetEqImplTest, ReorderedPacket) {
UseNoMocks();
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
CreateInstance(
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateHz));
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
int16_t dummy_output[kPayloadLengthSamples] = {0};
// The below expectation will make the mock decoder write
// |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
kSampleRateHz, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Insert two more packets. The first one is out of order, and is already too
// old, the second one is the expected next packet.
rtp_header.sequenceNumber -= 1;
rtp_header.timestamp -= kPayloadLengthSamples;
payload[0] = 1;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
payload[0] = 2;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Expect only the second packet to be decoded (the one with "2" as the first
// payload byte).
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
kSampleRateHz, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
// Pull audio once.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Now check the packet buffer, and make sure it is empty, since the
// out-of-order packet should have been discarded.
EXPECT_TRUE(packet_buffer_->Empty());
EXPECT_CALL(mock_decoder, Die());
}
// This test verifies that NetEq can handle the situation where the first
// incoming packet is rejected.
TEST_F(NetEqImplTest, FirstPacketUnknown) {
UseNoMocks();
CreateInstance();
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
// Insert one packet. Note that we have not registered any payload type, so
// this packet will be rejected.
EXPECT_EQ(NetEq::kFail,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
// Register the payload type.
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
// Insert 10 packets.
for (size_t i = 0; i < 10; ++i) {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
// Pull audio repeatedly and make sure we get normal output, that is not PLC.
for (size_t i = 0; i < 3; ++i) {
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
<< "NetEq did not decode the packets as expected.";
}
}
// This test verifies that audio interruption is not logged for the initial
// PLC period before the first packet is deocoded.
// TODO(henrik.lundin) Maybe move this test to neteq_network_stats_unittest.cc.
TEST_F(NetEqImplTest, NoAudioInterruptionLoggedBeforeFirstDecode) {
UseNoMocks();
CreateInstance();
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples * 2;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
// Register the payload type.
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
// Pull audio several times. No packets have been inserted yet.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
for (int i = 0; i < 100; ++i) {
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_NE(AudioFrame::kNormalSpeech, output.speech_type_);
}
// Insert 10 packets.
for (size_t i = 0; i < 10; ++i) {
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
}
// Pull audio repeatedly and make sure we get normal output, that is not PLC.
for (size_t i = 0; i < 3; ++i) {
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_LE(output.samples_per_channel_, kMaxOutputSize);
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_)
<< "NetEq did not decode the packets as expected.";
}
auto lifetime_stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(0u, lifetime_stats.interruption_count);
}
// This test verifies that NetEq can handle comfort noise and enters/quits codec
// internal CNG mode properly.
TEST_F(NetEqImplTest, CodecInternalCng) {
UseNoMocks();
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
CreateInstance(
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateKhz = 48;
const size_t kPayloadLengthSamples =
static_cast<size_t>(20 * kSampleRateKhz); // 20 ms.
const size_t kPayloadLengthBytes = 10;
uint8_t payload[kPayloadLengthBytes] = {0};
int16_t dummy_output[kPayloadLengthSamples] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateKhz * 1000));
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, kPayloadLengthBytes))
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
// Packed duration when asking the decoder for more CNG data (without a new
// packet).
EXPECT_CALL(mock_decoder, PacketDuration(nullptr, 0))
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
// Pointee(x) verifies that first byte of the payload equals x, this makes it
// possible to verify that the correct payload is fed to Decode().
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(0), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(1), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
EXPECT_CALL(mock_decoder,
DecodeInternal(IsNull(), 0, kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
EXPECT_CALL(mock_decoder, DecodeInternal(Pointee(2), kPayloadLengthBytes,
kSampleRateKhz * 1000, _, _))
.WillOnce(DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples))));
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("opus", 48000, 2)));
// Insert one packet (decoder will return speech).
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Insert second packet (decoder will return CNG).
payload[0] = 1;
rtp_header.sequenceNumber++;
rtp_header.timestamp += kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
AudioFrame output;
AudioFrame::SpeechType expected_type[8] = {
AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech,
AudioFrame::kCNG, AudioFrame::kCNG,
AudioFrame::kCNG, AudioFrame::kCNG,
AudioFrame::kNormalSpeech, AudioFrame::kNormalSpeech
};
int expected_timestamp_increment[8] = {
-1, // will not be used.
10 * kSampleRateKhz,
-1, -1, // timestamp will be empty during CNG mode; indicated by -1 here.
-1, -1,
50 * kSampleRateKhz, 10 * kSampleRateKhz
};
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
absl::optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
ASSERT_TRUE(last_timestamp);
// Lambda for verifying the timestamps.
auto verify_timestamp = [&last_timestamp, &expected_timestamp_increment](
absl::optional<uint32_t> ts, size_t i) {
if (expected_timestamp_increment[i] == -1) {
// Expect to get an empty timestamp value during CNG and PLC.
EXPECT_FALSE(ts) << "i = " << i;
} else {
ASSERT_TRUE(ts) << "i = " << i;
EXPECT_EQ(*ts, *last_timestamp + expected_timestamp_increment[i])
<< "i = " << i;
last_timestamp = ts;
}
};
for (size_t i = 1; i < 6; ++i) {
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(expected_type[i - 1], output.speech_type_);
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
SCOPED_TRACE("");
verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
}
// Insert third packet, which leaves a gap from last packet.
payload[0] = 2;
rtp_header.sequenceNumber += 2;
rtp_header.timestamp += 2 * kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
for (size_t i = 6; i < 8; ++i) {
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(expected_type[i - 1], output.speech_type_);
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
SCOPED_TRACE("");
verify_timestamp(neteq_->GetPlayoutTimestamp(), i);
}
// Now check the packet buffer, and make sure it is empty.
EXPECT_TRUE(packet_buffer_->Empty());
EXPECT_CALL(mock_decoder, Die());
}
TEST_F(NetEqImplTest, UnsupportedDecoder) {
UseNoMocks();
::testing::NiceMock<MockAudioDecoder> decoder;
CreateInstance(
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&decoder));
static const size_t kNetEqMaxFrameSize = 5760; // 120 ms @ 48 kHz.
static const size_t kChannels = 2;
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = 1;
uint8_t payload[kPayloadLengthBytes] = {0};
int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
const uint8_t kFirstPayloadValue = 1;
const uint8_t kSecondPayloadValue = 2;
EXPECT_CALL(decoder,
PacketDuration(Pointee(kFirstPayloadValue), kPayloadLengthBytes))
.Times(AtLeast(1))
.WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize + 1)));
EXPECT_CALL(decoder, DecodeInternal(Pointee(kFirstPayloadValue), _, _, _, _))
.Times(0);
EXPECT_CALL(decoder, DecodeInternal(Pointee(kSecondPayloadValue),
kPayloadLengthBytes, kSampleRateHz, _, _))
.Times(1)
.WillOnce(DoAll(
SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples * kChannels),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(static_cast<int>(kPayloadLengthSamples * kChannels))));
EXPECT_CALL(decoder,
PacketDuration(Pointee(kSecondPayloadValue), kPayloadLengthBytes))
.Times(AtLeast(1))
.WillRepeatedly(Return(rtc::checked_cast<int>(kNetEqMaxFrameSize)));
EXPECT_CALL(decoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateHz));
EXPECT_CALL(decoder, Channels())
.WillRepeatedly(Return(kChannels));
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
payload[0] = kFirstPayloadValue; // This will make Decode() fail.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
// Insert another packet.
payload[0] = kSecondPayloadValue; // This will make Decode() successful.
rtp_header.sequenceNumber++;
// The second timestamp needs to be at least 30 ms after the first to make
// the second packet get decoded.
rtp_header.timestamp += 3 * kPayloadLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
AudioFrame output;
bool muted;
// First call to GetAudio will try to decode the "faulty" packet.
// Expect kFail return value.
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
// Output size and number of channels should be correct.
const size_t kExpectedOutputSize = 10 * (kSampleRateHz / 1000) * kChannels;
EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
EXPECT_EQ(kChannels, output.num_channels_);
// Second call to GetAudio will decode the packet that is ok. No errors are
// expected.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kExpectedOutputSize, output.samples_per_channel_ * kChannels);
EXPECT_EQ(kChannels, output.num_channels_);
// Die isn't called through NiceMock (since it's called by the
// MockAudioDecoder constructor), so it needs to be mocked explicitly.
EXPECT_CALL(decoder, Die());
}
// This test inserts packets until the buffer is flushed. After that, it asks
// NetEq for the network statistics. The purpose of the test is to make sure
// that even though the buffer size increment is negative (which it becomes when
// the packet causing a flush is inserted), the packet length stored in the
// decision logic remains valid.
TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
UseNoMocks();
CreateInstance();
const size_t kPayloadLengthSamples = 80;
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit.
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
// Insert packets until the buffer flushes.
for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
rtp_header.timestamp += rtc::checked_cast<uint32_t>(kPayloadLengthSamples);
++rtp_header.sequenceNumber;
}
EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
// Ask for network statistics. This should not crash.
NetEqNetworkStatistics stats;
EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
}
TEST_F(NetEqImplTest, DecodedPayloadTooShort) {
UseNoMocks();
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
CreateInstance(
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const size_t kPayloadLengthSamples =
static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms.
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateHz));
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.WillRepeatedly(Return(rtc::checked_cast<int>(kPayloadLengthSamples)));
int16_t dummy_output[kPayloadLengthSamples] = {0};
// The below expectation will make the mock decoder write
// |kPayloadLengthSamples| - 5 zeros to the output array, and mark it as
// speech. That is, the decoded length is 5 samples shorter than the expected.
EXPECT_CALL(mock_decoder,
DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
.WillOnce(
DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kPayloadLengthSamples - 5),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kPayloadLengthSamples - 5))));
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("L16", 8000, 1)));
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
EXPECT_EQ(5u, neteq_->sync_buffer_for_test()->FutureLength());
// Pull audio once.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
EXPECT_CALL(mock_decoder, Die());
}
// This test checks the behavior of NetEq when audio decoder fails.
TEST_F(NetEqImplTest, DecodingError) {
UseNoMocks();
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
CreateInstance(
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kDecoderErrorCode = -97; // Any negative number.
// We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
const size_t kFrameLengthSamples =
static_cast<size_t>(5 * kSampleRateHz / 1000);
const size_t kPayloadLengthBytes = 1; // This can be arbitrary.
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateHz));
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
EXPECT_CALL(mock_decoder, ErrorCode())
.WillOnce(Return(kDecoderErrorCode));
EXPECT_CALL(mock_decoder, HasDecodePlc())
.WillOnce(Return(false));
int16_t dummy_output[kFrameLengthSamples] = {0};
{
InSequence sequence; // Dummy variable.
// Mock decoder works normally the first time.
EXPECT_CALL(mock_decoder,
DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
.Times(3)
.WillRepeatedly(
DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kFrameLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kFrameLengthSamples))))
.RetiresOnSaturation();
// Then mock decoder fails. A common reason for failure can be buffer being
// too short
EXPECT_CALL(mock_decoder,
DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
.WillOnce(Return(-1))
.RetiresOnSaturation();
// Mock decoder finally returns to normal.
EXPECT_CALL(mock_decoder,
DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
.Times(2)
.WillRepeatedly(
DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kFrameLengthSamples),
SetArgPointee<4>(AudioDecoder::kSpeech),
Return(rtc::checked_cast<int>(kFrameLengthSamples))));
}
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("L16", 8000, 1)));
// Insert packets.
for (int i = 0; i < 6; ++i) {
rtp_header.sequenceNumber += 1;
rtp_header.timestamp += kFrameLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
}
// Pull audio.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
// Pull audio again. Decoder fails.
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
// We are not expecting anything for output.speech_type_, since an error was
// returned.
// Pull audio again, should continue an expansion.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
// Pull audio again, should behave normal.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
EXPECT_CALL(mock_decoder, Die());
}
// This test checks the behavior of NetEq when audio decoder fails during CNG.
TEST_F(NetEqImplTest, DecodingErrorDuringInternalCng) {
UseNoMocks();
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
CreateInstance(
new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&mock_decoder));
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17; // Value doesn't matter for this test.
const int kSampleRateHz = 8000;
const int kDecoderErrorCode = -97; // Any negative number.
// We let decoder return 5 ms each time, and therefore, 2 packets make 10 ms.
const size_t kFrameLengthSamples =
static_cast<size_t>(5 * kSampleRateHz / 1000);
const size_t kPayloadLengthBytes = 1; // This can be arbitrary.
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_CALL(mock_decoder, Reset()).WillRepeatedly(Return());
EXPECT_CALL(mock_decoder, SampleRateHz())
.WillRepeatedly(Return(kSampleRateHz));
EXPECT_CALL(mock_decoder, Channels()).WillRepeatedly(Return(1));
EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
.WillRepeatedly(Return(0));
EXPECT_CALL(mock_decoder, PacketDuration(_, _))
.WillRepeatedly(Return(rtc::checked_cast<int>(kFrameLengthSamples)));
EXPECT_CALL(mock_decoder, ErrorCode())
.WillOnce(Return(kDecoderErrorCode));
int16_t dummy_output[kFrameLengthSamples] = {0};
{
InSequence sequence; // Dummy variable.
// Mock decoder works normally the first 2 times.
EXPECT_CALL(mock_decoder,
DecodeInternal(_, kPayloadLengthBytes, kSampleRateHz, _, _))
.Times(2)
.WillRepeatedly(
DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kFrameLengthSamples),
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kFrameLengthSamples))))
.RetiresOnSaturation();
// Then mock decoder fails. A common reason for failure can be buffer being
// too short
EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
.WillOnce(Return(-1))
.RetiresOnSaturation();
// Mock decoder finally returns to normal.
EXPECT_CALL(mock_decoder, DecodeInternal(nullptr, 0, kSampleRateHz, _, _))
.Times(2)
.WillRepeatedly(
DoAll(SetArrayArgument<3>(dummy_output,
dummy_output + kFrameLengthSamples),
SetArgPointee<4>(AudioDecoder::kComfortNoise),
Return(rtc::checked_cast<int>(kFrameLengthSamples))));
}
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
// Insert 2 packets. This will make netEq into codec internal CNG mode.
for (int i = 0; i < 2; ++i) {
rtp_header.sequenceNumber += 1;
rtp_header.timestamp += kFrameLengthSamples;
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
}
// Pull audio.
const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
// Pull audio again. Decoder fails.
EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
// We are not expecting anything for output.speech_type_, since an error was
// returned.
// Pull audio again, should resume codec CNG.
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(kMaxOutputSize, output.samples_per_channel_);
EXPECT_EQ(1u, output.num_channels_);
EXPECT_EQ(AudioFrame::kCNG, output.speech_type_);
EXPECT_CALL(mock_decoder, Die());
}
// Tests that the return value from last_output_sample_rate_hz() is equal to the
// configured inital sample rate.
TEST_F(NetEqImplTest, InitialLastOutputSampleRate) {
UseNoMocks();
config_.sample_rate_hz = 48000;
CreateInstance();
EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz());
}
TEST_F(NetEqImplTest, TickTimerIncrement) {
UseNoMocks();
CreateInstance();
ASSERT_TRUE(tick_timer_);
EXPECT_EQ(0u, tick_timer_->ticks());
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
EXPECT_EQ(1u, tick_timer_->ticks());
}
TEST_F(NetEqImplTest, SetBaseMinimumDelay) {
UseNoMocks();
use_mock_delay_manager_ = true;
CreateInstance();
EXPECT_CALL(*mock_delay_manager_, SetBaseMinimumDelay(_))
.WillOnce(Return(true))
.WillOnce(Return(false));
const int delay_ms = 200;
EXPECT_EQ(true, neteq_->SetBaseMinimumDelayMs(delay_ms));
EXPECT_EQ(false, neteq_->SetBaseMinimumDelayMs(delay_ms));
}
TEST_F(NetEqImplTest, GetBaseMinimumDelayMs) {
UseNoMocks();
use_mock_delay_manager_ = true;
CreateInstance();
const int delay_ms = 200;
EXPECT_CALL(*mock_delay_manager_, GetBaseMinimumDelay())
.WillOnce(Return(delay_ms));
EXPECT_EQ(delay_ms, neteq_->GetBaseMinimumDelayMs());
}
TEST_F(NetEqImplTest, TargetDelayMs) {
UseNoMocks();
use_mock_delay_manager_ = true;
CreateInstance();
// Let the dummy target delay be 17 packets.
constexpr int kTargetLevelPacketsQ8 = 17 << 8;
EXPECT_CALL(*mock_delay_manager_, TargetLevel())
.WillOnce(Return(kTargetLevelPacketsQ8));
// Default packet size before any packet has been decoded is 30 ms, so we are
// expecting 17 * 30 = 510 ms target delay.
EXPECT_EQ(17 * 30, neteq_->TargetDelayMs());
}
TEST_F(NetEqImplTest, InsertEmptyPacket) {
UseNoMocks();
use_mock_delay_manager_ = true;
CreateInstance();
RTPHeader rtp_header;
rtp_header.payloadType = 17;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_CALL(*mock_delay_manager_, RegisterEmptyPacket());
neteq_->InsertEmptyPacket(rtp_header);
}
TEST_F(NetEqImplTest, EnableRtxHandling) {
UseNoMocks();
use_mock_delay_manager_ = true;
config_.enable_rtx_handling = true;
CreateInstance();
EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
.Times(1)
.WillOnce(DoAll(SetArgPointee<0>(0), SetArgPointee<1>(0)));
const int kPayloadLengthSamples = 80;
const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples; // PCM 16-bit.
const uint8_t kPayloadType = 17; // Just an arbitrary number.
const uint32_t kReceiveTime = 17;
uint8_t payload[kPayloadLengthBytes] = {0};
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = 0x1234;
rtp_header.timestamp = 0x12345678;
rtp_header.ssrc = 0x87654321;
EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
SdpAudioFormat("l16", 8000, 1)));
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
AudioFrame output;
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
// Insert second packet that was sent before the first packet.
rtp_header.sequenceNumber -= 1;
rtp_header.timestamp -= kPayloadLengthSamples;
EXPECT_CALL(*mock_delay_manager_,
Update(rtp_header.sequenceNumber, rtp_header.timestamp, _));
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
}
class Decoder120ms : public AudioDecoder {
public:
Decoder120ms(int sample_rate_hz, SpeechType speech_type)
: sample_rate_hz_(sample_rate_hz),
next_value_(1),
speech_type_(speech_type) {}
int DecodeInternal(const uint8_t* encoded,
size_t encoded_len,
int sample_rate_hz,
int16_t* decoded,
SpeechType* speech_type) override {
EXPECT_EQ(sample_rate_hz_, sample_rate_hz);
size_t decoded_len =
rtc::CheckedDivExact(sample_rate_hz, 1000) * 120 * Channels();
for (size_t i = 0; i < decoded_len; ++i) {
decoded[i] = next_value_++;
}
*speech_type = speech_type_;
return rtc::checked_cast<int>(decoded_len);
}
void Reset() override { next_value_ = 1; }
int SampleRateHz() const override { return sample_rate_hz_; }
size_t Channels() const override { return 2; }
private:
int sample_rate_hz_;
int16_t next_value_;
SpeechType speech_type_;
};
class NetEqImplTest120ms : public NetEqImplTest {
protected:
NetEqImplTest120ms() : NetEqImplTest() {}
virtual ~NetEqImplTest120ms() {}
void CreateInstanceNoMocks() {
UseNoMocks();
CreateInstance(decoder_factory_);
EXPECT_TRUE(neteq_->RegisterPayloadType(
kPayloadType, SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})));
}
void CreateInstanceWithDelayManagerMock() {
UseNoMocks();
use_mock_delay_manager_ = true;
CreateInstance(decoder_factory_);
EXPECT_TRUE(neteq_->RegisterPayloadType(
kPayloadType, SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}})));
}
uint32_t timestamp_diff_between_packets() const {
return rtc::CheckedDivExact(kSamplingFreq_, 1000u) * 120;
}
uint32_t first_timestamp() const { return 10u; }
void GetFirstPacket() {
bool muted;
for (int i = 0; i < 12; i++) {
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_FALSE(muted);
}
}
void InsertPacket(uint32_t timestamp) {
RTPHeader rtp_header;
rtp_header.payloadType = kPayloadType;
rtp_header.sequenceNumber = sequence_number_;
rtp_header.timestamp = timestamp;
rtp_header.ssrc = 15;
const size_t kPayloadLengthBytes = 1; // This can be arbitrary.
uint8_t payload[kPayloadLengthBytes] = {0};
EXPECT_EQ(NetEq::kOK, neteq_->InsertPacket(rtp_header, payload, 10));
sequence_number_++;
}
void Register120msCodec(AudioDecoder::SpeechType speech_type) {
const uint32_t sampling_freq = kSamplingFreq_;
decoder_factory_ =
new rtc::RefCountedObject<test::FunctionAudioDecoderFactory>(
[sampling_freq, speech_type]() {
std::unique_ptr<AudioDecoder> decoder =
absl::make_unique<Decoder120ms>(sampling_freq, speech_type);
RTC_CHECK_EQ(2, decoder->Channels());
return decoder;
});
}
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
AudioFrame output_;
const uint32_t kPayloadType = 17;
const uint32_t kSamplingFreq_ = 48000;
uint16_t sequence_number_ = 1;
};
TEST_F(NetEqImplTest120ms, CodecInternalCng) {
Register120msCodec(AudioDecoder::kComfortNoise);
CreateInstanceNoMocks();
InsertPacket(first_timestamp());
GetFirstPacket();
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kCodecInternalCng, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Normal) {
Register120msCodec(AudioDecoder::kSpeech);
CreateInstanceNoMocks();
InsertPacket(first_timestamp());
GetFirstPacket();
EXPECT_EQ(kNormal, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Merge) {
Register120msCodec(AudioDecoder::kSpeech);
CreateInstanceWithDelayManagerMock();
InsertPacket(first_timestamp());
GetFirstPacket();
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
InsertPacket(first_timestamp() + 2 * timestamp_diff_between_packets());
// Delay manager reports a target level which should cause a Merge.
EXPECT_CALL(*mock_delay_manager_, TargetLevel()).WillOnce(Return(-10));
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kMerge, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Expand) {
Register120msCodec(AudioDecoder::kSpeech);
CreateInstanceNoMocks();
InsertPacket(first_timestamp());
GetFirstPacket();
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kExpand, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, FastAccelerate) {
Register120msCodec(AudioDecoder::kSpeech);
CreateInstanceWithDelayManagerMock();
InsertPacket(first_timestamp());
GetFirstPacket();
InsertPacket(first_timestamp() + timestamp_diff_between_packets());
// Delay manager report buffer limit which should cause a FastAccelerate.
EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
.Times(1)
.WillOnce(DoAll(SetArgPointee<0>(0), SetArgPointee<1>(0)));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kFastAccelerate, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, PreemptiveExpand) {
Register120msCodec(AudioDecoder::kSpeech);
CreateInstanceWithDelayManagerMock();
InsertPacket(first_timestamp());
GetFirstPacket();
InsertPacket(first_timestamp() + timestamp_diff_between_packets());
// Delay manager report buffer limit which should cause a PreemptiveExpand.
EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
.Times(1)
.WillOnce(DoAll(SetArgPointee<0>(100), SetArgPointee<1>(100)));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kPreemptiveExpand, neteq_->last_operation_for_test());
}
TEST_F(NetEqImplTest120ms, Accelerate) {
Register120msCodec(AudioDecoder::kSpeech);
CreateInstanceWithDelayManagerMock();
InsertPacket(first_timestamp());
GetFirstPacket();
InsertPacket(first_timestamp() + timestamp_diff_between_packets());
// Delay manager report buffer limit which should cause a Accelerate.
EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
.Times(1)
.WillOnce(DoAll(SetArgPointee<0>(1), SetArgPointee<1>(2)));
bool muted;
EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
}
}// namespace webrtc
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