Revert of Setting up an RTP input fuzzer for NetEq

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

-/*
- *  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_