/* * 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" int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels); /** * 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 * const *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); #endif /* AVRESAMPLE_AUDIO_DATA_H */