text: /*
   *  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 MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
  #define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
  
  #include <memory>
  
  #include BOSS_ABSEILCPP_U_absl__types__optional_h //original-code:"absl/types/optional.h"
  #include BOSS_WEBRTC_U_api__array_view_h //original-code:"api/array_view.h"
  #include BOSS_WEBRTC_U_api__audio_codecs__audio_decoder_h //original-code:"api/audio_codecs/audio_decoder.h"
  #include "modules/audio_coding/neteq/tools/input_audio_file.h"
  
  namespace webrtc {
  namespace test {
  
  // Provides an AudioDecoder implementation that delivers audio data from a file.
  // The "encoded" input should contain information about what RTP timestamp the
  // encoding represents, and how many samples the decoder should produce for that
  // encoding. A helper method PrepareEncoded is provided to prepare such
  // encodings. If packets are missing, as determined from the timestamps, the
  // file reading will skip forward to match the loss.
  class FakeDecodeFromFile : public AudioDecoder {
   public:
    FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,
                       int sample_rate_hz,
                       bool stereo)
        : input_(std::move(input)),
          sample_rate_hz_(sample_rate_hz),
          stereo_(stereo) {}
  
    ~FakeDecodeFromFile() = default;
  
    void Reset() override {}
  
    int SampleRateHz() const override { return sample_rate_hz_; }
  
    size_t Channels() const override { return stereo_ ? 2 : 1; }
  
    int DecodeInternal(const uint8_t* encoded,
                       size_t encoded_len,
                       int sample_rate_hz,
                       int16_t* decoded,
                       SpeechType* speech_type) override;
  
    int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
  
    // Helper method. Writes |timestamp|, |samples| and
    // |original_payload_size_bytes| to |encoded| in a format that the
    // FakeDecodeFromFile decoder will understand. |encoded| must be at least 12
    // bytes long.
    static void PrepareEncoded(uint32_t timestamp,
                               size_t samples,
                               size_t original_payload_size_bytes,
                               rtc::ArrayView<uint8_t> encoded);
  
   private:
    std::unique_ptr<InputAudioFile> input_;
    absl::optional<uint32_t> next_timestamp_from_input_;
    const int sample_rate_hz_;
    const bool stereo_;
    size_t last_decoded_length_ = 0;
    bool cng_mode_ = false;
  };
  
  }  // namespace test
  }  // namespace webrtc
  #endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
  