7270a6bcc2
This CL includes a larger structural change in how we handle buffers in AEC. We now perform FFT at once and move within blocks to compensate for system delays. TEST=audioproc_unittest(float and fix), voe_auto_test Review URL: http://webrtc-codereview.appspot.com/335012 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1299 4adac7df-926f-26a2-2b94-8c16560cd09d
83 lines
3.5 KiB
C
83 lines
3.5 KiB
C
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
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// otherwise specified, functions return 0 on success and -1 on error.
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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#include <stddef.h> // size_t
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// Determines buffer datatype
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typedef short bufdata_t; // TODO(bjornv): Remove together with the below.
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// TODO(bjornv): Remove WebRtcApm_CreateBuffer, WebRtcApm_InitBuffer and
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// WebRtcApm_FreeBuffer when replaced in APM.
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// Rename WebRtcApm_FreeBuffer to WebRtc_FreeBuffer() and replace.
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// Replace WebRtcApm_FlushBuffer and WebRtcApm_StuffBuffer with
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// WebRtc_MoveReadPtr and Read/Write-Buffer with its new versions.
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int WebRtcApm_CreateBuffer(void **bufInst, int size);
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int WebRtcApm_InitBuffer(void *bufInst);
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int WebRtcApm_FreeBuffer(void *bufInst);
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// Returns number of samples read
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int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size);
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// Returns number of samples written
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int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size);
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// Returns number of samples flushed
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int WebRtcApm_FlushBuffer(void *bufInst, int size);
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// Returns number of samples stuffed
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int WebRtcApm_StuffBuffer(void *bufInst, int size);
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// Returns number of samples in buffer
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int WebRtcApm_get_buffer_size(const void *bufInst);
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// TODO(bjornv): Below are the new functions, to replace the older ones above.
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int WebRtc_CreateBuffer(void** handle,
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size_t element_count,
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size_t element_size);
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int WebRtc_InitBuffer(void* handle);
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int WebRtc_FreeBuffer(void* handle);
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// Reads data from the buffer. The |data_ptr| will point to the address where
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// it is located. If all |element_count| data are feasible to read without
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// buffer wrap around |data_ptr| will point to the location in the buffer.
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// Otherwise, the data will be copied to |data| (memory allocation done by the
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// user) and |data_ptr| points to the address of |data|. |data_ptr| is only
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// guaranteed to be valid until the next call to WebRtc_WriteBuffer().
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// Returns number of elements read.
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size_t WebRtc_ReadBuffer(void* handle,
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void** data_ptr,
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void* data,
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size_t element_count);
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// Writes |data| to buffer and returns the number of elements written.
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size_t WebRtc_WriteBuffer(void* handle, const void* data, size_t element_count);
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// Moves the buffer read position and returns the number of elements moved.
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// Positive |element_count| moves the read position towards the write position,
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// that is, flushing the buffer. Negative |element_count| moves the read
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// position away from the the write position, that is, stuffing the buffer.
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// Returns number of elements moved.
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int WebRtc_MoveReadPtr(void* handle, int element_count);
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// Returns number of available elements to read.
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size_t WebRtc_available_read(const void* handle);
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// Returns number of available elements for write.
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size_t WebRtc_available_write(const void* handle);
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
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