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/*
* 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_frame.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 : 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;
// 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 rtc::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.
virtual 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.
virtual 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).
virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
size_t position);
// Reads `requested_len` samples from each channel and writes them interleaved
// into `output`. The `next_index_` is updated to point to the sample to read
// next time. The AudioFrame `output` is first reset, and the `data_`,
// `num_channels_`, and `samples_per_channel_` fields are updated.
void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
// 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_
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