From c8af852b97447491823ff9b91413e32415e2babf Mon Sep 17 00:00:00 2001 From: Justin Ruggles Date: Fri, 23 Mar 2012 17:42:17 -0400 Subject: Add libavresample This is a new library for audio sample format, channel layout, and sample rate conversion. --- libavresample/audio_data.h | 173 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 173 insertions(+) create mode 100644 libavresample/audio_data.h (limited to 'libavresample/audio_data.h') diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h new file mode 100644 index 0000000000..4609ebc284 --- /dev/null +++ b/libavresample/audio_data.h @@ -0,0 +1,173 @@ +/* + * Copyright (c) 2012 Justin Ruggles + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; 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" + +/** + * Audio buffer used for intermediate storage between conversion phases. + */ +typedef 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 */ +} AudioData; + +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, void **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 + * @return 0 on success, negative AVERROR value on error + */ +int ff_audio_data_copy(AudioData *out, AudioData *in); + +/** + * 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 */ -- cgit v1.2.3