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gtest EXPECT_EQ

时间: 2023-11-17 14:57:20 浏览: 394
gtest 的 EXPECT_EQ 是 Google Test 提供的一个断言宏,用于比较两个值是否相等。这个宏的使用方式如下: ```cpp EXPECT_EQ(expected_value, actual_value); ``` 其中,expected_value 是期望的值,actual_value 是实际的值。当两个值相等时,断言通过,测试继续进行;当两个值不相等时,断言失败,测试停止并报告失败信息。 示例代码如下: ```cpp #include <gtest/gtest.h> TEST(MyTest, ValueComparison) { int expected = 5; int actual = 5; EXPECT_EQ(expected, actual); // 断言期望值和实际值相等 } int main(int argc, char** argv) { testing::InitGoogleTest(&argc, argv); return RUN_ALL_TESTS(); } ``` 在上面的示例中,我们定义了一个测试用例 `MyTest`,其中进行了一个值比较的断言。我们期望 `expected` 和 `actual` 的值相等,因此使用了 `EXPECT_EQ` 断言来进行判断。 如果 `expected` 和 `actual` 的值相等,测试通过;如果不相等,测试失败,并输出错误信息。 请注意,当使用 `EXPECT_EQ` 断言时,如果断言失败,测试仍然会继续执行,因此可以在一个测试用例中包含多个断言来检查不同的条件。
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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. 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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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