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TEST_F(类, 测试名),这个gtest定义是做什么

时间: 2024-04-10 13:30:20 浏览: 303
TEST_F(类, 测试名) 是 Google Test (gtest) 中的一个宏定义,用于定义一个测试用例。在 gtest 中,测试用例是由一个测试类和一个或多个测试函数组成的。 这个宏定义的作用是创建一个测试用例,并指定测试用例所属的测试类和测试函数的名称。其中,'类' 是测试类的名称,'测试名' 是测试函数的名称。 使用 TEST_F 宏定义的测试用例可以在测试类的作用域内使用测试固件(fixture),即可以在测试函数中共享一些初始化和清理的代码。测试固件可以帮助我们在每个测试函数执行前后执行一些共同的操作,以减少重复代码。 总结起来,TEST_F(类, 测试名) 宏定义用于创建一个测试用例,指定测试类和测试函数的名称,并且支持测试固件的使用。
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gtest TEST_F 和 TEST

gtest是Google Test框架的一部分,它是一个用于编写单元测试的C++库。`TEST_F`和`TEST`是gtest中两个基本的测试宏,用于定义测试用例。 `TEST_F(classname, test_name)`是针对特定测试 fixture (测试套件) 的测试函数,其中`classname`是你自定义的测试fixture类名,`test_name`是测试用例的具体名称。测试fixture通常包含了你需要在测试中复用的数据或其他设置,这样可以避免在每个测试中都重复初始化资源。 例如: ```cpp class MyTest : public ::testing::Test { public: void SetUp() override { /* 初始化操作 */ } void TearDown() override { /* 清理操作 */ } private: // 测试fixture成员变量 }; TEST_F(MyTest, TestAddition) { EXPECT_EQ(3, addNumbers(1, 2)); } ``` 这里`MyTest`是fixture,`TestAddition`是具体的测试用例。 相比之下,`TEST(testcase_name, test_case)`是一个独立的测试用例,它不属于任何特定的fixture,`testcase_name`代表测试集名,`test_case`是单个测试的名称。这适用于那些不需要复用设置的简单测试: ```cpp TEST(MyTestSuite, AdditionTest) { EXPECT_EQ(3, addNumbers(1, 2)); } ``` 在这里,`MyTestSuite`是测试集名,`AdditionTest`是测试案例。

问题解决 ../../base/security/crypto_framework/test/unittest/src/crypto_md_sm3_test.cpp:82:1: error: redefinition of 'CryptoMd_SM3Test_CryptoFrameworkMd_SM3_CreateTest001___LINE__' [OHOS ERROR] HWTEST_F(CryptoMd_SM3Test, CryptoFrameworkMd_SM3_CreateTest001, TestSize.Level0) [OHOS ERROR] ^ [OHOS ERROR] ../../third_party/googletest/googletest/include/gtest/hwext/gtest-ext.h:25:10: note: expanded from macro 'HWTEST_F' [OHOS ERROR] bool GTEST_TEST_UNIQUE_ID_(test_case_name, test_name, __LINE__) = testing::ext::TestDefManager::instance()->regist(#test_case_name, #test_name, test_flags, testing::ext::Fixtured);\ [OHOS ERROR] ^ [OHOS ERROR] ../../third_party/googletest/googletest/include/gtest/hwext/gtest-ext.h:14:5: note: expanded from macro 'GTEST_TEST_UNIQUE_ID_' [OHOS ERROR] test_case_name##_##test_name##_##file_line [OHOS ERROR] ^

这个错误提示是在编译crypto_md_sm3_test.cpp文件时发生的,它指出在第82行重新定义了一个名为 'CryptoMd_SM3Test_CryptoFrameworkMd_SM3_CreateTest001___LINE__' 的符号,这可能是由于在同一文件中多次定义了相同的测试用例或测试函数所导致的。 要解决这个问题,您需要检查crypto_md_sm3_test.cpp文件,确保没有重复定义相同名称的测试用例或测试函数。如果确实存在这样的重复定义,您可以删除其中一个或将它们重命名为不同的名称以避免冲突。如果您确认没有重复定义,那么可能是其他文件中定义了相同的符号,您需要检查是否存在这种情况,并将其解决。
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/* * 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_rtp_dump_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/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 { #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \ defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ defined(WEBRTC_CODEC_ILBC) #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 = "dee7a10ab92526876a70a85bc48a4906901af3df"; const std::string network_stats_checksum = "911dbf5fd97f48d25b8f0967286eb73c9d6f6158"; DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum, absl::GetFlag(FLAGS_gen_ref)); } #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \ defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && defined(WEBRTC_CODEC_OPUS) #define MAYBE_TestOpusBitExactness TestOpusBitExactness #else #define MAYBE_TestOpusBitExactness DISABLED_TestOpusBitExactness #endif TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) { const std::string input_rtp_file = webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp"); const std::string output_checksum = "fec6827bb9ee0b21770bbbb4a3a6f8823bf537dc|" "3610cc7be4b3407b9c273b1299ab7f8f47cca96b"; const std::string network_stats_checksum = "3d043e47e5f4bb81d37e7bce8c44bf802965c853|" "076662525572dba753b11578330bd491923f7f5e"; DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum, absl::GetFlag(FLAGS_gen_ref)); } #if defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && \ 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 output_checksum = "b3c4899eab5378ef5e54f2302948872149f6ad5e|" "589e975ec31ea13f302457fea1425be9380ffb96"; const std::string network_stats_checksum = "dc8447b9fee1a21fd5d1f4045d62b982a3fb0215"; 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; 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++; } 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_); } ++seq_no; timestamp += kCngPeriodSamples; uint32_t first_speech_timestamp = timestamp; // Insert speech again. 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 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(first_speech_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 a few packets which are older than the first packet. for (int i = 0; i < 5; ++i) { InsertPacket(kSamples * (i - 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. for (int i = 0; i < 5; ++i) { PopulateRtpInfo(0, kSamples * 1000 + kSamples * i, &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, 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, nullptr, 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(); std::map<int, RTPExtensionType> rtp_ext_map = { {1, kRtpExtensionAudioLevel}, {3, kRtpExtensionAbsoluteSendTime}, {5, kRtpExtensionTransportSequenceNumber}, {7, kRtpExtensionVideoContentType}, {8, kRtpExtensionVideoTiming}}; std::unique_ptr<NetEqInput> input = CreateNetEqRtpDumpInput( 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 test } // namespace webrtc

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