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