tor-browser

sync_buffer.h (4839B)

---

/*
*  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_