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