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/*
* Copyright (c) 2011 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.
*/
/*
* Specifies the interface for the AEC generic buffer.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
// Determines buffer datatype
typedef short bufdata_t;
// Unless otherwise specified, functions return 0 on success and -1 on error
int WebRtcApm_CreateBuffer(void **bufInst, int size);
int WebRtcApm_InitBuffer(void *bufInst);
int WebRtcApm_FreeBuffer(void *bufInst);
// Returns number of samples read
int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size);
// Returns number of samples written
int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size);
// Returns number of samples flushed
int WebRtcApm_FlushBuffer(void *bufInst, int size);
// Returns number of samples stuffed
int WebRtcApm_StuffBuffer(void *bufInst, int size);
// Returns number of samples in buffer
int WebRtcApm_get_buffer_size(const void *bufInst);
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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