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/*
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 * Copyright (c) 2012 Justin Ruggles 
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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 
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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 */