/* * 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 "testing/gmock/include/gmock/gmock.h" #include "testing/gtest/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(×tamp)); 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(sync_buffer->FutureLength())); } } // namespace webrtc