Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 1008 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
new file mode 100644
index 0000000000..77bd5b5035
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -0,0 +1,1008 @@
+/*
+ *  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