legacy_encoded_audio_frame.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_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_

#define MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_

#include <stddef.h>

#include <stdint.h>

#include <vector>

#include "absl/types/optional.h"

#include "api/array_view.h"

#include "api/audio_codecs/audio_decoder.h"

#include "rtc_base/buffer.h"

namespace webrtc {

class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {

 public:

  LegacyEncodedAudioFrame(AudioDecoder* decoder, rtc::Buffer&& payload);

  ~LegacyEncodedAudioFrame() override;

  static std::vector<AudioDecoder::ParseResult> SplitBySamples(

      AudioDecoder* decoder,

      rtc::Buffer&& payload,

      uint32_t timestamp,

      size_t bytes_per_ms,

      uint32_t timestamps_per_ms);

  size_t Duration() const override;

  absl::optional<DecodeResult> Decode(

      rtc::ArrayView<int16_t> decoded) const override;

  // For testing:

  const rtc::Buffer& payload() const { return payload_; }

 private:

  AudioDecoder* const decoder_;

  const rtc::Buffer payload_;

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_CODECS_LEGACY_ENCODED_AUDIO_FRAME_H_