chromium/third_party/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.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 MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
#define MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_

#include <memory>

#include "absl/types/optional.h"
#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 { … };

}  // namespace test
}  // namespace webrtc
#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_