mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-04 23:45:14 +00:00
102 lines
4.2 KiB
C++
102 lines
4.2 KiB
C++
/*
|
|
* 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
|
|
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
|
|
|
|
#include "webrtc/base/constructormagic.h"
|
|
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
|
|
#include "webrtc/typedefs.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) {}
|
|
|
|
virtual ~SyncBuffer() {}
|
|
|
|
// Returns the number of samples yet to play out form 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);
|
|
|
|
// 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.
|
|
size_t GetNextAudioInterleaved(size_t requested_len, int16_t* 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.
|
|
|
|
DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
|