libavresample: Remove deprecated library

Deprecated in c29038f3041a4080342b2e333c1967d136749c0f.
The resample filter based upon this library has been removed as well.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: James Almer <jamrial@gmail.com>

1 files changed, 0 insertions, 178 deletions

diff --git a/libavresample/audio_data.h b/libavresample/audio_data.h
deleted file mode 100644
index 1280307a95..0000000000
--- a/libavresample/audio_data.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * 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 */