0,0 → 1,175 |
/* |
* Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
* |
* This file is part of FFmpeg. |
* |
* FFmpeg 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. |
* |
* FFmpeg 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 FFmpeg; 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" |
#include "internal.h" |
|
/** |
* Audio buffer used for intermediate storage between conversion phases. |
*/ |
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 */ |
}; |
|
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 |
*/ |
int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, |
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 |
* @param map channel map, NULL if not remapping |
* @return 0 on success, negative AVERROR value on error |
*/ |
int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map); |
|
/** |
* 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); |
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#endif /* AVRESAMPLE_AUDIO_DATA_H */ |