178 lines
6.9 KiB
C
178 lines
6.9 KiB
C
/*
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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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int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels);
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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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