modules/audio_coding/neteq/sync_buffer.h

/*
 *  Copyright (c) 2012 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_SYNC_BUFFER_H_
#define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_

#include <stddef.h>
#include <stdint.h>

#include <vector>

#include "api/audio/audio_view.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/audio_vector.h"
#include "rtc_base/buffer.h"

namespace webrtc {

class SyncBuffer final : public AudioMultiVector {
 public:
  SyncBuffer(size_t channels, size_t length)
      : AudioMultiVector(channels, length),
        next_index_(length),
        end_timestamp_(0),
        dtmf_index_(0) {}

  SyncBuffer(const SyncBuffer&) = delete;
  SyncBuffer& operator=(const SyncBuffer&) = delete;

  ~SyncBuffer() override = default;

  // Returns the number of samples yet to play out from the buffer.
  size_t FutureLength() const;

  // Adds the contents of `append_this` to the back of the SyncBuffer. Removes
  // the same number of samples from the beginning of the SyncBuffer, to
  // maintain a constant buffer size. The `next_index_` is updated to reflect
  // the move of the beginning of "future" data.
  void PushBack(const AudioMultiVector& append_this) override;

  // Like PushBack, but reads the samples channel-interleaved from the input.
  void PushBackInterleaved(const BufferT<int16_t>& append_this);

  // Adds `length` zeros to the beginning of each channel. Removes
  // the same number of samples from the end of the SyncBuffer, to
  // maintain a constant buffer size. The `next_index_` is updated to reflect
  // the move of the beginning of "future" data.
  // Note that this operation may delete future samples that are waiting to
  // be played.
  void PushFrontZeros(size_t length);

  // Inserts `length` zeros into each channel at index `position`. The size of
  // the SyncBuffer is kept constant, which means that the last `length`
  // elements in each channel will be purged.
  void InsertZerosAtIndex(size_t length, size_t position);

  // Overwrites each channel in this SyncBuffer with values taken from
  // `insert_this`. The values are taken from the beginning of `insert_this` and
  // are inserted starting at `position`. `length` values are written into each
  // channel. The size of the SyncBuffer is kept constant. That is, if `length`
  // and `position` are selected such that the new data would extend beyond the
  // end of the current SyncBuffer, the buffer is not extended.
  // The `next_index_` is not updated.
  void ReplaceAtIndex(const AudioMultiVector& insert_this,
                      size_t length,
                      size_t position);

  // Same as the above method, but where all of `insert_this` is written (with
  // the same constraints as above, that the SyncBuffer is not extended).
  void ReplaceAtIndex(const AudioMultiVector& insert_this, size_t position);

  // If enough data is available, reads `audio.samples_per_channel()` samples
  // from each channel into `audio` and return true.
  // If not enough data is available, the function returns false and no data
  // will have been read.
  //
  // When successful, the internal `next_index_` position is updated to point
  // to the samples to read next time around.
  //
  // Note: `audio` must be configured to have the same number of channels as
  // `Channels()` and expected to meet the size limitations of AudioFrame.
  bool GetNextAudioInterleaved(InterleavedView<int16_t> audio);

  // Adds `increment` to `end_timestamp_`.
  void IncreaseEndTimestamp(uint32_t increment);

  // Flushes the buffer. The buffer will contain only zeros after the flush, and
  // `next_index_` will point to the end, like when the buffer was first
  // created.
  void Flush();

  const AudioVector& Channel(size_t n) const { return *channels_[n]; }
  AudioVector& Channel(size_t n) { return *channels_[n]; }

  // Accessors and mutators.
  size_t next_index() const { return next_index_; }
  void set_next_index(size_t value);
  uint32_t end_timestamp() const { return end_timestamp_; }
  void set_end_timestamp(uint32_t value) { end_timestamp_ = value; }
  size_t dtmf_index() const { return dtmf_index_; }
  void set_dtmf_index(size_t value);

 private:
  size_t next_index_;
  uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
  size_t dtmf_index_;       // Index to the first non-DTMF sample in the buffer.
};

}  // namespace webrtc
#endif  // MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_