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libavresample
audio_data.h
Go to the documentation of this file.
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/*
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* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#ifndef AVRESAMPLE_AUDIO_DATA_H
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#define AVRESAMPLE_AUDIO_DATA_H
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#include <stdint.h>
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#include "
libavutil/audio_fifo.h
"
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#include "
libavutil/log.h
"
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#include "
libavutil/samplefmt.h
"
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#include "
avresample.h
"
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#include "
internal.h
"
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/**
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* Audio buffer used for intermediate storage between conversion phases.
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*/
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struct
AudioData
{
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const
AVClass
*
class
;
/**< AVClass for logging */
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uint8_t
*
data
[
AVRESAMPLE_MAX_CHANNELS
];
/**< data plane pointers */
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uint8_t
*
buffer
;
/**< data buffer */
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unsigned
int
buffer_size
;
/**< allocated buffer size */
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int
allocated_samples
;
/**< number of samples the buffer can hold */
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int
nb_samples
;
/**< current number of samples */
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enum
AVSampleFormat
sample_fmt
;
/**< sample format */
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int
channels
;
/**< channel count */
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int
allocated_channels
;
/**< allocated channel count */
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int
is_planar
;
/**< sample format is planar */
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int
planes
;
/**< number of data planes */
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int
sample_size
;
/**< bytes per sample */
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int
stride
;
/**< sample byte offset within a plane */
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int
read_only
;
/**< data is read-only */
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int
allow_realloc
;
/**< realloc is allowed */
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int
ptr_align
;
/**< minimum data pointer alignment */
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int
samples_align
;
/**< allocated samples alignment */
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const
char
*
name
;
/**< name for debug logging */
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};
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int
ff_audio_data_set_channels
(
AudioData
*
a
,
int
channels);
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/**
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* Initialize AudioData using a given source.
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*
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* This does not allocate an internal buffer. It only sets the data pointers
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* and audio parameters.
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*
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* @param a AudioData struct
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* @param src source data pointers
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* @param plane_size plane size, in bytes.
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* This can be 0 if unknown, but that will lead to
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* optimized functions not being used in many cases,
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* which could slow down some conversions.
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* @param channels channel count
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* @param nb_samples number of samples in the source data
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* @param sample_fmt sample format
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* @param read_only indicates if buffer is read only or read/write
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* @param name name for debug logging (can be NULL)
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* @return 0 on success, negative AVERROR value on error
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*/
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int
ff_audio_data_init
(
AudioData
*
a
,
uint8_t
**
src
,
int
plane_size,
int
channels,
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int
nb_samples,
enum
AVSampleFormat
sample_fmt,
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int
read_only,
const
char
*
name
);
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/**
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* Allocate AudioData.
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*
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* This allocates an internal buffer and sets audio parameters.
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*
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* @param channels channel count
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* @param nb_samples number of samples to allocate space for
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* @param sample_fmt sample format
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* @param name name for debug logging (can be NULL)
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* @return newly allocated AudioData struct, or NULL on error
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*/
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AudioData
*
ff_audio_data_alloc
(
int
channels,
int
nb_samples,
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enum
AVSampleFormat
sample_fmt,
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const
char
*
name
);
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/**
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* Reallocate AudioData.
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*
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* The AudioData must have been previously allocated with ff_audio_data_alloc().
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*
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* @param a AudioData struct
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* @param nb_samples number of samples to allocate space for
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* @return 0 on success, negative AVERROR value on error
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*/
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int
ff_audio_data_realloc
(
AudioData
*
a
,
int
nb_samples);
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/**
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* Free AudioData.
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*
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* The AudioData must have been previously allocated with ff_audio_data_alloc().
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*
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* @param a AudioData struct
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*/
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void
ff_audio_data_free
(
AudioData
**
a
);
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/**
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* Copy data from one AudioData to another.
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*
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* @param out output AudioData
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* @param in input AudioData
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* @param map channel map, NULL if not remapping
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* @return 0 on success, negative AVERROR value on error
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*/
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int
ff_audio_data_copy
(
AudioData
*
out
,
AudioData
*
in
,
ChannelMapInfo
*map);
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/**
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* Append data from one AudioData to the end of another.
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*
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* @param dst destination AudioData
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* @param dst_offset offset, in samples, to start writing, relative to the
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* start of dst
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* @param src source AudioData
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* @param src_offset offset, in samples, to start copying, relative to the
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* start of the src
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* @param nb_samples number of samples to copy
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* @return 0 on success, negative AVERROR value on error
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*/
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int
ff_audio_data_combine
(
AudioData
*dst,
int
dst_offset,
AudioData
*
src
,
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int
src_offset,
int
nb_samples);
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/**
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* Drain samples from the start of the AudioData.
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*
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* Remaining samples are shifted to the start of the AudioData.
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*
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* @param a AudioData struct
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* @param nb_samples number of samples to drain
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*/
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void
ff_audio_data_drain
(
AudioData
*
a
,
int
nb_samples);
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/**
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* Add samples in AudioData to an AVAudioFifo.
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*
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* @param af Audio FIFO Buffer
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* @param a AudioData struct
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* @param offset number of samples to skip from the start of the data
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* @param nb_samples number of samples to add to the FIFO
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* @return number of samples actually added to the FIFO, or
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* negative AVERROR code on error
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*/
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int
ff_audio_data_add_to_fifo
(
AVAudioFifo
*af,
AudioData
*
a
,
int
offset
,
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int
nb_samples);
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/**
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* Read samples from an AVAudioFifo to AudioData.
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*
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* @param af Audio FIFO Buffer
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* @param a AudioData struct
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* @param nb_samples number of samples to read from the FIFO
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* @return number of samples actually read from the FIFO, or
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* negative AVERROR code on error
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*/
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int
ff_audio_data_read_from_fifo
(
AVAudioFifo
*af,
AudioData
*
a
,
int
nb_samples);
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#endif
/* AVRESAMPLE_AUDIO_DATA_H */
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