session-android/jni/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

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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 "webrtc/modules/audio_coding/neteq/interface/neteq.h"
#include "webrtc/modules/audio_coding/neteq/neteq_impl.h"
#include "gmock/gmock.h"
#include "gtest/gtest.h"
#include "webrtc/modules/audio_coding/neteq/accelerate.h"
#include "webrtc/modules/audio_coding/neteq/expand.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h"
#include "webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h"
#include "webrtc/modules/audio_coding/neteq/preemptive_expand.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
#include "webrtc/modules/audio_coding/neteq/timestamp_scaler.h"
using ::testing::Return;
using ::testing::ReturnNull;
using ::testing::_;
using ::testing::SetArgPointee;
using ::testing::InSequence;
using ::testing::Invoke;
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) {
PacketBuffer::DeleteAllPackets(packet_list);
return PacketBuffer::kOK;
}
class NetEqImplTest : public ::testing::Test {
protected:
NetEqImplTest()
: neteq_(NULL),
config_(),
mock_buffer_level_filter_(NULL),
buffer_level_filter_(NULL),
use_mock_buffer_level_filter_(true),
mock_decoder_database_(NULL),
decoder_database_(NULL),
use_mock_decoder_database_(true),
mock_delay_peak_detector_(NULL),
delay_peak_detector_(NULL),
use_mock_delay_peak_detector_(true),
mock_delay_manager_(NULL),
delay_manager_(NULL),
use_mock_delay_manager_(true),
mock_dtmf_buffer_(NULL),
dtmf_buffer_(NULL),
use_mock_dtmf_buffer_(true),
mock_dtmf_tone_generator_(NULL),
dtmf_tone_generator_(NULL),
use_mock_dtmf_tone_generator_(true),
mock_packet_buffer_(NULL),
packet_buffer_(NULL),
use_mock_packet_buffer_(true),
mock_payload_splitter_(NULL),
payload_splitter_(NULL),
use_mock_payload_splitter_(true),
timestamp_scaler_(NULL) {
config_.sample_rate_hz = 8000;
}
void CreateInstance() {
if (use_mock_buffer_level_filter_) {
mock_buffer_level_filter_ = new MockBufferLevelFilter;
buffer_level_filter_ = mock_buffer_level_filter_;
} else {
buffer_level_filter_ = new BufferLevelFilter;
}
if (use_mock_decoder_database_) {
mock_decoder_database_ = new MockDecoderDatabase;
EXPECT_CALL(*mock_decoder_database_, GetActiveCngDecoder())
.WillOnce(ReturnNull());
decoder_database_ = mock_decoder_database_;
} else {
decoder_database_ = new DecoderDatabase;
}
if (use_mock_delay_peak_detector_) {
mock_delay_peak_detector_ = new MockDelayPeakDetector;
EXPECT_CALL(*mock_delay_peak_detector_, Reset()).Times(1);
delay_peak_detector_ = mock_delay_peak_detector_;
} else {
delay_peak_detector_ = new DelayPeakDetector;
}
if (use_mock_delay_manager_) {
mock_delay_manager_ = new MockDelayManager(config_.max_packets_in_buffer,
delay_peak_detector_);
EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
delay_manager_ = mock_delay_manager_;
} else {
delay_manager_ =
new DelayManager(config_.max_packets_in_buffer, delay_peak_detector_);
}
if (use_mock_dtmf_buffer_) {
mock_dtmf_buffer_ = new MockDtmfBuffer(config_.sample_rate_hz);
dtmf_buffer_ = mock_dtmf_buffer_;
} else {
dtmf_buffer_ = new DtmfBuffer(config_.sample_rate_hz);
}
if (use_mock_dtmf_tone_generator_) {
mock_dtmf_tone_generator_ = new MockDtmfToneGenerator;
dtmf_tone_generator_ = mock_dtmf_tone_generator_;
} else {
dtmf_tone_generator_ = new DtmfToneGenerator;
}
if (use_mock_packet_buffer_) {
mock_packet_buffer_ = new MockPacketBuffer(config_.max_packets_in_buffer);
packet_buffer_ = mock_packet_buffer_;
} else {
packet_buffer_ = new PacketBuffer(config_.max_packets_in_buffer);
}
if (use_mock_payload_splitter_) {
mock_payload_splitter_ = new MockPayloadSplitter;
payload_splitter_ = mock_payload_splitter_;
} else {
payload_splitter_ = new PayloadSplitter;
}
timestamp_scaler_ = new TimestampScaler(*decoder_database_);
AccelerateFactory* accelerate_factory = new AccelerateFactory;
ExpandFactory* expand_factory = new ExpandFactory;
PreemptiveExpandFactory* preemptive_expand_factory =
new PreemptiveExpandFactory;
neteq_ = new NetEqImpl(config_,
buffer_level_filter_,
decoder_database_,
delay_manager_,
delay_peak_detector_,
dtmf_buffer_,
dtmf_tone_generator_,
packet_buffer_,
payload_splitter_,
timestamp_scaler_,
accelerate_factory,
expand_factory,
preemptive_expand_factory);
ASSERT_TRUE(neteq_ != NULL);
}
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);
}
delete neteq_;
}
NetEqImpl* neteq_;
NetEq::Config config_;
MockBufferLevelFilter* mock_buffer_level_filter_;
BufferLevelFilter* buffer_level_filter_;
bool use_mock_buffer_level_filter_;
MockDecoderDatabase* mock_decoder_database_;
DecoderDatabase* decoder_database_;
bool use_mock_decoder_database_;
MockDelayPeakDetector* mock_delay_peak_detector_;
DelayPeakDetector* delay_peak_detector_;
bool use_mock_delay_peak_detector_;
MockDelayManager* mock_delay_manager_;
DelayManager* delay_manager_;
bool use_mock_delay_manager_;
MockDtmfBuffer* mock_dtmf_buffer_;
DtmfBuffer* dtmf_buffer_;
bool use_mock_dtmf_buffer_;
MockDtmfToneGenerator* mock_dtmf_tone_generator_;
DtmfToneGenerator* dtmf_tone_generator_;
bool use_mock_dtmf_tone_generator_;
MockPacketBuffer* mock_packet_buffer_;
PacketBuffer* packet_buffer_;
bool use_mock_packet_buffer_;
MockPayloadSplitter* mock_payload_splitter_;
PayloadSplitter* payload_splitter_;
bool use_mock_payload_splitter_;
TimestampScaler* timestamp_scaler_;
};
// This tests the interface class NetEq.
// TODO(hlundin): Move to separate file?
TEST(NetEq, CreateAndDestroy) {
NetEq::Config config;
NetEq* neteq = NetEq::Create(config);
delete neteq;
}
TEST_F(NetEqImplTest, RegisterPayloadType) {
CreateInstance();
uint8_t rtp_payload_type = 0;
NetEqDecoder codec_type = kDecoderPCMu;
EXPECT_CALL(*mock_decoder_database_,
RegisterPayload(rtp_payload_type, codec_type));
neteq_->RegisterPayloadType(codec_type, rtp_payload_type);
}
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 kFail is returned when database returns kDecoderNotFound.
EXPECT_EQ(NetEq::kFail, neteq_->RemovePayloadType(rtp_payload_type));
}
TEST_F(NetEqImplTest, InsertPacket) {
CreateInstance();
const int 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};
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = kFirstSequenceNumber;
rtp_header.header.timestamp = kFirstTimestamp;
rtp_header.header.ssrc = kSsrc;
// Create a mock decoder object.
MockAudioDecoder mock_decoder;
// 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.
// Expectations for decoder database.
EXPECT_CALL(*mock_decoder_database_, IsRed(kPayloadType))
.WillRepeatedly(Return(false)); // This is not RED.
EXPECT_CALL(*mock_decoder_database_, CheckPayloadTypes(_))
.Times(2)
.WillRepeatedly(Return(DecoderDatabase::kOK)); // Payload type is valid.
EXPECT_CALL(*mock_decoder_database_, IsDtmf(kPayloadType))
.WillRepeatedly(Return(false)); // This is not DTMF.
EXPECT_CALL(*mock_decoder_database_, GetDecoder(kPayloadType))
.Times(3)
.WillRepeatedly(Return(&mock_decoder));
EXPECT_CALL(*mock_decoder_database_, IsComfortNoise(kPayloadType))
.WillRepeatedly(Return(false)); // This is not CNG.
DecoderDatabase::DecoderInfo info;
info.codec_type = kDecoderPCMu;
EXPECT_CALL(*mock_decoder_database_, GetDecoderInfo(kPayloadType))
.WillRepeatedly(Return(&info));
// Expectations for packet buffer.
EXPECT_CALL(*mock_packet_buffer_, NumPacketsInBuffer())
.WillOnce(Return(0)) // First packet.
.WillOnce(Return(1)) // Second packet.
.WillOnce(Return(2)); // Second packet, checking after it was inserted.
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_, NextRtpHeader())
.Times(1)
.WillOnce(Return(&rtp_header.header));
// 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_, LastDecoderType(kDecoderPCMu))
.Times(1);
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_, LastDecoderType(kDecoderPCMu))
.Times(1);
EXPECT_CALL(*mock_delay_manager_, last_pack_cng_or_dtmf())
.WillOnce(Return(0));
EXPECT_CALL(*mock_delay_manager_, SetPacketAudioLength(30))
.WillOnce(Return(0));
}
// Expectations for payload splitter.
EXPECT_CALL(*mock_payload_splitter_, SplitAudio(_, _))
.Times(2)
.WillRepeatedly(Return(PayloadSplitter::kOK));
// Insert first packet.
neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime);
// Insert second packet.
rtp_header.header.timestamp += 160;
rtp_header.header.sequenceNumber += 1;
neteq_->InsertPacket(rtp_header, payload, kPayloadLength,
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};
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = 0x1234;
rtp_header.header.timestamp = 0x12345678;
rtp_header.header.ssrc = 0x87654321;
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
// Insert packets. The buffer should not flush.
for (int i = 1; i <= config_.max_packets_in_buffer; ++i) {
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
rtp_header.header.timestamp += kPayloadLengthSamples;
rtp_header.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, kPayloadLengthBytes, kReceiveTime));
EXPECT_EQ(1, packet_buffer_->NumPacketsInBuffer());
const RTPHeader* test_header = packet_buffer_->NextRtpHeader();
EXPECT_EQ(rtp_header.header.timestamp, test_header->timestamp);
EXPECT_EQ(rtp_header.header.sequenceNumber, test_header->sequenceNumber);
}
// This test verifies that timestamps propagate from the incoming packets
// through to the sync buffer and to the playout timestamp.
TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
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 int kPayloadLengthSamples = 10 * kSampleRateHz / 1000; // 10 ms.
const size_t kPayloadLengthBytes = kPayloadLengthSamples;
uint8_t payload[kPayloadLengthBytes] = {0};
WebRtcRTPHeader rtp_header;
rtp_header.header.payloadType = kPayloadType;
rtp_header.header.sequenceNumber = 0x1234;
rtp_header.header.timestamp = 0x12345678;
rtp_header.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:
explicit CountingSamplesDecoder(enum NetEqDecoder type)
: AudioDecoder(type), next_value_(1) {}
// Produce as many samples as input bytes (|encoded_len|).
virtual int Decode(const uint8_t* encoded,
size_t encoded_len,
int16_t* decoded,
SpeechType* speech_type) {
for (size_t i = 0; i < encoded_len; ++i) {
decoded[i] = next_value_++;
}
*speech_type = kSpeech;
return encoded_len;
}
virtual int Init() {
next_value_ = 1;
return 0;
}
uint16_t next_value() const { return next_value_; }
private:
int16_t next_value_;
} decoder_(kDecoderPCM16B);
EXPECT_EQ(NetEq::kOK,
neteq_->RegisterExternalDecoder(
&decoder_, kDecoderPCM16B, kPayloadType));
// Insert one packet.
EXPECT_EQ(NetEq::kOK,
neteq_->InsertPacket(
rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
// Pull audio once.
const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
int16_t output[kMaxOutputSize];
int samples_per_channel;
int num_channels;
NetEqOutputType type;
EXPECT_EQ(
NetEq::kOK,
neteq_->GetAudio(
kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
ASSERT_EQ(kMaxOutputSize, samples_per_channel);
EXPECT_EQ(1, num_channels);
EXPECT_EQ(kOutputNormal, type);
// Start with a simple check that the fake decoder is behaving as expected.
EXPECT_EQ(kPayloadLengthSamples, 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.
uint32_t timestamp = 0;
EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&timestamp));
EXPECT_EQ(rtp_header.header.timestamp + output[samples_per_channel - 1],
timestamp);
// 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.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[samples_per_channel - 1],
static_cast<int>(sync_buffer->FutureLength()));
}
} // namespace webrtc