fake_decode_from_file.h

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/*

 *  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 <optional>

#include "api/array_view.h"

#include "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;

  std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,

                                        uint32_t timestamp) override;

  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;

  // Reads `samples` from the input file and writes the results to

  // `destination`. Location in file is determined by `timestamp`.

  void ReadFromFile(uint32_t timestamp, size_t samples, int16_t* destination);

  // 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_;

  std::optional<uint32_t> next_timestamp_from_input_;

  const int sample_rate_hz_;

  const bool stereo_;

};

}  // namespace test

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_