Revert of Setting up an RTP input fuzzer for NetEq

webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc

Index: webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.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 54682166aa7a7dd1588c53cf9f8119f96fa2e91a..0000000000000000000000000000000000000000
--- 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