diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 094bdbe..57d7cd9 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -1110,8 +1110,6 @@
       "neteq/tools/audio_sink.h",
       "neteq/tools/constant_pcm_packet_source.cc",
       "neteq/tools/constant_pcm_packet_source.h",
-      "neteq/tools/encode_neteq_input.cc",
-      "neteq/tools/encode_neteq_input.h",
       "neteq/tools/fake_decode_from_file.cc",
       "neteq/tools/fake_decode_from_file.h",
       "neteq/tools/input_audio_file.cc",
diff --git a/webrtc/modules/audio_coding/neteq/neteq.gypi b/webrtc/modules/audio_coding/neteq/neteq.gypi
index f07cee4..d9c152e 100644
--- a/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -231,8 +231,6 @@
             'tools/audio_sink.cc',
             'tools/constant_pcm_packet_source.cc',
             'tools/constant_pcm_packet_source.h',
-            'tools/encode_neteq_input.cc',
-            'tools/encode_neteq_input.h',
             'tools/fake_decode_from_file.cc',
             'tools/fake_decode_from_file.h',
             'tools/input_audio_file.cc',
diff --git a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
deleted file mode 100644
index 5468216..0000000
--- a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h"
-
-#include <utility>
-
-#include "webrtc/base/checks.h"
-
-namespace webrtc {
-namespace test {
-
-EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<InputAudioFile> input,
-                                   std::unique_ptr<AudioEncoder> encoder,
-                                   int64_t input_duration_ms)
-    : input_(std::move(input)),
-      encoder_(std::move(encoder)),
-      input_duration_ms_(input_duration_ms) {
-  CreatePacket();
-}
-
-rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
-  RTC_DCHECK(packet_data_);
-  return rtc::Optional<int64_t>(static_cast<int64_t>(packet_data_->time_ms));
-}
-
-rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
-  return rtc::Optional<int64_t>(next_output_event_ms_);
-}
-
-std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
-  RTC_DCHECK(packet_data_);
-  // Grab the packet to return...
-  std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_);
-  // ... and line up the next packet for future use.
-  CreatePacket();
-
-  return packet_to_return;
-}
-
-void EncodeNetEqInput::AdvanceOutputEvent() {
-  next_output_event_ms_ += kOutputPeriodMs;
-}
-
-rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
-  RTC_DCHECK(packet_data_);
-  return rtc::Optional<RTPHeader>(packet_data_->header.header);
-}
-
-void EncodeNetEqInput::CreatePacket() {
-  // Create a new PacketData object.
-  RTC_DCHECK(!packet_data_);
-  packet_data_.reset(new NetEqInput::PacketData);
-  RTC_DCHECK_EQ(packet_data_->payload.size(), 0u);
-
-  // Loop until we get a packet.
-  AudioEncoder::EncodedInfo info;
-  RTC_DCHECK(!info.send_even_if_empty);
-  int num_blocks = 0;
-  while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
-    const size_t num_samples = rtc::CheckedDivExact(
-        static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
-    std::unique_ptr<int16_t[]> audio(new int16_t[num_samples]);
-    RTC_CHECK(input_->Read(num_samples, audio.get()));
-
-    info = encoder_->Encode(
-        rtp_timestamp_, rtc::ArrayView<const int16_t>(audio.get(), num_samples),
-        &packet_data_->payload);
-
-    rtp_timestamp_ +=
-        num_samples * encoder_->RtpTimestampRateHz() / encoder_->SampleRateHz();
-    ++num_blocks;
-  }
-  packet_data_->header.header.timestamp = info.encoded_timestamp;
-  packet_data_->header.header.payloadType = info.payload_type;
-  packet_data_->header.header.sequenceNumber = sequence_number_++;
-  packet_data_->time_ms = next_packet_time_ms_;
-  next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
-}
-
-}  // namespace test
-}  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
deleted file mode 100644
index ab28fd9..0000000
--- a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- *  Copyright (c) 2016 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
-
-#include <memory>
-
-#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
-#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
-#include "webrtc/modules/include/module_common_types.h"
-
-namespace webrtc {
-namespace test {
-
-// This class provides a NetEqInput that takes audio from an input file and
-// encodes it using a given audio encoder.
-class EncodeNetEqInput : public NetEqInput {
- public:
-  // The source will end after the given input duration.
-  EncodeNetEqInput(std::unique_ptr<InputAudioFile> input,
-                   std::unique_ptr<AudioEncoder> encoder,
-                   int64_t input_duration_ms);
-
-  rtc::Optional<int64_t> NextPacketTime() const override;
-
-  rtc::Optional<int64_t> NextOutputEventTime() const override;
-
-  std::unique_ptr<PacketData> PopPacket() override;
-
-  void AdvanceOutputEvent() override;
-
-  bool ended() const override {
-    return next_output_event_ms_ <= input_duration_ms_;
-  }
-
-  rtc::Optional<RTPHeader> NextHeader() const override;
-
- private:
-  static constexpr int64_t kOutputPeriodMs = 10;
-
-  void CreatePacket();
-
-  std::unique_ptr<InputAudioFile> input_;
-  std::unique_ptr<AudioEncoder> encoder_;
-  std::unique_ptr<PacketData> packet_data_;
-  int32_t rtp_timestamp_ = 0;
-  int16_t sequence_number_ = 0;
-  int64_t next_packet_time_ms_ = 0;
-  int64_t next_output_event_ms_ = 0;
-  const int64_t input_duration_ms_;
-};
-
-}  // namespace test
-}  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
index be08a79..8abec63 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
@@ -65,9 +65,7 @@
   // time).
   virtual void AdvanceOutputEvent() = 0;
 
-  // Returns true if the source has come to an end. An implementation must
-  // eventually return true from this method, or the test will end up in an
-  // infinite loop.
+  // Returns true if the source has come to an end.
   virtual bool ended() const = 0;
 
   // Returns the RTP header for the next packet, i.e., the packet that will be
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 34ed84b..c531947 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -184,19 +184,6 @@
   ]
 }
 
-webrtc_fuzzer_test("neteq_rtp_fuzzer") {
-  sources = [
-    "neteq_rtp_fuzzer.cc",
-  ]
-  deps = [
-    "../../modules/audio_coding:neteq",
-    "../../modules/audio_coding:neteq_unittest_tools",
-    "../../modules/audio_coding:pcm16b",
-    "../../modules/rtp_rtcp",
-    "../../test:test_support",
-  ]
-}
-
 # TODO(katrielc) Enable in Chromium when CL 2022833002 lands.
 # Although the dependency on media compiles in standalone, it is
 # flagged by gn check, so breaks when rolled into Chromium.
diff --git a/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc b/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
deleted file mode 100644
index 8d09691..0000000
--- a/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- *  Copyright (c) 2016 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 <memory>
-
-#include "webrtc/base/array_view.h"
-#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
-#include "webrtc/modules/audio_coding/neteq/tools/audio_checksum.h"
-#include "webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h"
-#include "webrtc/modules/audio_coding/neteq/tools/neteq_test.h"
-#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-namespace test {
-namespace {
-constexpr int kPayloadType = 95;
-
-class FuzzRtpInput : public NetEqInput {
- public:
-  explicit FuzzRtpInput(rtc::ArrayView<const uint8_t> data) : data_(data) {
-    std::unique_ptr<InputAudioFile> audio_input(
-        new InputAudioFile(ResourcePath("audio_coding/testfile32kHz", "pcm")));
-    AudioEncoderPcm16B::Config config;
-    config.payload_type = kPayloadType;
-    config.sample_rate_hz = 32000;
-    std::unique_ptr<AudioEncoder> encoder(new AudioEncoderPcm16B(config));
-    input_.reset(new EncodeNetEqInput(std::move(audio_input),
-                                      std::move(encoder),
-                                      std::numeric_limits<int64_t>::max()));
-    packet_ = input_->PopPacket();
-    FuzzHeader();
-  }
-
-  rtc::Optional<int64_t> NextPacketTime() const override {
-    return rtc::Optional<int64_t>(packet_->time_ms);
-  }
-
-  rtc::Optional<int64_t> NextOutputEventTime() const override {
-    return input_->NextOutputEventTime();
-  }
-
-  std::unique_ptr<PacketData> PopPacket() override {
-    RTC_DCHECK(packet_);
-    std::unique_ptr<PacketData> packet_to_return = std::move(packet_);
-    packet_ = input_->PopPacket();
-    FuzzHeader();
-    return packet_to_return;
-  }
-
-  void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); }
-
-  bool ended() const override { return ended_; }
-
-  rtc::Optional<RTPHeader> NextHeader() const override {
-    RTC_DCHECK(packet_);
-    return rtc::Optional<RTPHeader>(packet_->header.header);
-  }
-
- private:
-  void FuzzHeader() {
-    constexpr size_t kNumBytesToFuzz = 11;
-    if (data_ix_ + kNumBytesToFuzz > data_.size()) {
-      ended_ = true;
-      return;
-    }
-    RTC_DCHECK(packet_);
-    const size_t start_ix = data_ix_;
-    packet_->header.header.payloadType =
-        ByteReader<uint8_t>::ReadLittleEndian(&data_[data_ix_]);
-    packet_->header.header.payloadType &= 0x7F;
-    data_ix_ += sizeof(uint8_t);
-    packet_->header.header.sequenceNumber =
-        ByteReader<uint16_t>::ReadLittleEndian(&data_[data_ix_]);
-    data_ix_ += sizeof(uint16_t);
-    packet_->header.header.timestamp =
-        ByteReader<uint32_t>::ReadLittleEndian(&data_[data_ix_]);
-    data_ix_ += sizeof(uint32_t);
-    packet_->header.header.ssrc =
-        ByteReader<uint32_t>::ReadLittleEndian(&data_[data_ix_]);
-    data_ix_ += sizeof(uint32_t);
-    RTC_CHECK_EQ(data_ix_ - start_ix, kNumBytesToFuzz);
-  }
-
-  bool ended_ = false;
-  rtc::ArrayView<const uint8_t> data_;
-  size_t data_ix_ = 0;
-  std::unique_ptr<EncodeNetEqInput> input_;
-  std::unique_ptr<PacketData> packet_;
-};
-}  // namespace
-
-void FuzzOneInputTest(const uint8_t* data, size_t size) {
-  std::unique_ptr<FuzzRtpInput> input(
-      new FuzzRtpInput(rtc::ArrayView<const uint8_t>(data, size)));
-  std::unique_ptr<AudioChecksum> output(new AudioChecksum);
-  NetEqTestErrorCallback dummy_callback;  // Does nothing with error callbacks.
-  NetEq::Config config;
-  NetEqTest::DecoderMap codecs;
-  codecs[0] = std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu");
-  codecs[8] = std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma");
-  codecs[102] = std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc");
-  codecs[103] = std::make_pair(NetEqDecoder::kDecoderISAC, "isac");
-  codecs[104] = std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb");
-  codecs[111] = std::make_pair(NetEqDecoder::kDecoderOpus, "opus");
-  codecs[93] = std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
-  codecs[94] = std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
-  codecs[96] =
-      std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
-  codecs[9] = std::make_pair(NetEqDecoder::kDecoderG722, "g722");
-  codecs[106] = std::make_pair(NetEqDecoder::kDecoderAVT, "avt");
-  codecs[117] = std::make_pair(NetEqDecoder::kDecoderRED, "red");
-  codecs[13] = std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb");
-  codecs[98] = std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb");
-  codecs[99] = std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32");
-  codecs[100] = std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48");
-  // This is the payload type that will be used for encoding.
-  codecs[kPayloadType] =
-      std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
-  NetEqTest::ExtDecoderMap ext_codecs;
-
-  NetEqTest test(config, codecs, ext_codecs, std::move(input),
-                 std::move(output), &dummy_callback);
-  test.Run();
-}
-
-}  // namespace test
-
-void FuzzOneInput(const uint8_t* data, size_t size) {
-  test::FuzzOneInputTest(data, size);
-}
-
-}  // namespace webrtc