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, 381 deletions

diff --git a/libavresample/audio_data.c b/libavresample/audio_data.c
deleted file mode 100644
index b54ead841a..0000000000
--- a/libavresample/audio_data.c
+++ /dev/null
@@ -1,381 +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
- */
-
-#include <stdint.h>
-#include <string.h>
-
-#include "libavutil/mem.h"
-#include "audio_data.h"
-
-static const AVClass audio_data_class = {
-    .class_name = "AudioData",
-    .item_name  = av_default_item_name,
-    .version    = LIBAVUTIL_VERSION_INT,
-};
-
-/*
- * Calculate alignment for data pointers.
- */
-static void calc_ptr_alignment(AudioData *a)
-{
-    int p;
-    int min_align = 128;
-
-    for (p = 0; p < a->planes; p++) {
-        int cur_align = 128;
-        while ((intptr_t)a->data[p] % cur_align)
-            cur_align >>= 1;
-        if (cur_align < min_align)
-            min_align = cur_align;
-    }
-    a->ptr_align = min_align;
-}
-
-int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels)
-{
-    if (channels == 1)
-        return 1;
-    else
-        return av_sample_fmt_is_planar(sample_fmt);
-}
-
-int ff_audio_data_set_channels(AudioData *a, int channels)
-{
-    if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS ||
-        channels > a->allocated_channels)
-        return AVERROR(EINVAL);
-
-    a->channels  = channels;
-    a->planes    = a->is_planar ? channels : 1;
-
-    calc_ptr_alignment(a);
-
-    return 0;
-}
-
-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)
-{
-    int p;
-
-    memset(a, 0, sizeof(*a));
-    a->class = &audio_data_class;
-
-    if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) {
-        av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels);
-        return AVERROR(EINVAL);
-    }
-
-    a->sample_size = av_get_bytes_per_sample(sample_fmt);
-    if (!a->sample_size) {
-        av_log(a, AV_LOG_ERROR, "invalid sample format\n");
-        return AVERROR(EINVAL);
-    }
-    a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels);
-    a->planes    = a->is_planar ? channels : 1;
-    a->stride    = a->sample_size * (a->is_planar ? 1 : channels);
-
-    for (p = 0; p < (a->is_planar ? channels : 1); p++) {
-        if (!src[p]) {
-            av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p);
-            return AVERROR(EINVAL);
-        }
-        a->data[p] = src[p];
-    }
-    a->allocated_samples  = nb_samples * !read_only;
-    a->nb_samples         = nb_samples;
-    a->sample_fmt         = sample_fmt;
-    a->channels           = channels;
-    a->allocated_channels = channels;
-    a->read_only          = read_only;
-    a->allow_realloc      = 0;
-    a->name               = name ? name : "{no name}";
-
-    calc_ptr_alignment(a);
-    a->samples_align = plane_size / a->stride;
-
-    return 0;
-}
-
-AudioData *ff_audio_data_alloc(int channels, int nb_samples,
-                               enum AVSampleFormat sample_fmt, const char *name)
-{
-    AudioData *a;
-    int ret;
-
-    if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS)
-        return NULL;
-
-    a = av_mallocz(sizeof(*a));
-    if (!a)
-        return NULL;
-
-    a->sample_size = av_get_bytes_per_sample(sample_fmt);
-    if (!a->sample_size) {
-        av_free(a);
-        return NULL;
-    }
-    a->is_planar = ff_sample_fmt_is_planar(sample_fmt, channels);
-    a->planes    = a->is_planar ? channels : 1;
-    a->stride    = a->sample_size * (a->is_planar ? 1 : channels);
-
-    a->class              = &audio_data_class;
-    a->sample_fmt         = sample_fmt;
-    a->channels           = channels;
-    a->allocated_channels = channels;
-    a->read_only          = 0;
-    a->allow_realloc      = 1;
-    a->name               = name ? name : "{no name}";
-
-    if (nb_samples > 0) {
-        ret = ff_audio_data_realloc(a, nb_samples);
-        if (ret < 0) {
-            av_free(a);
-            return NULL;
-        }
-        return a;
-    } else {
-        calc_ptr_alignment(a);
-        return a;
-    }
-}
-
-int ff_audio_data_realloc(AudioData *a, int nb_samples)
-{
-    int ret, new_buf_size, plane_size, p;
-
-    /* check if buffer is already large enough */
-    if (a->allocated_samples >= nb_samples)
-        return 0;
-
-    /* validate that the output is not read-only and realloc is allowed */
-    if (a->read_only || !a->allow_realloc)
-        return AVERROR(EINVAL);
-
-    new_buf_size = av_samples_get_buffer_size(&plane_size,
-                                              a->allocated_channels, nb_samples,
-                                              a->sample_fmt, 0);
-    if (new_buf_size < 0)
-        return new_buf_size;
-
-    /* if there is already data in the buffer and the sample format is planar,
-       allocate a new buffer and copy the data, otherwise just realloc the
-       internal buffer and set new data pointers */
-    if (a->nb_samples > 0 && a->is_planar) {
-        uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL };
-
-        ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels,
-                               nb_samples, a->sample_fmt, 0);
-        if (ret < 0)
-            return ret;
-
-        for (p = 0; p < a->planes; p++)
-            memcpy(new_data[p], a->data[p], a->nb_samples * a->stride);
-
-        av_freep(&a->buffer);
-        memcpy(a->data, new_data, sizeof(new_data));
-        a->buffer = a->data[0];
-    } else {
-        av_freep(&a->buffer);
-        a->buffer = av_malloc(new_buf_size);
-        if (!a->buffer)
-            return AVERROR(ENOMEM);
-        ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer,
-                                     a->allocated_channels, nb_samples,
-                                     a->sample_fmt, 0);
-        if (ret < 0)
-            return ret;
-    }
-    a->buffer_size       = new_buf_size;
-    a->allocated_samples = nb_samples;
-
-    calc_ptr_alignment(a);
-    a->samples_align = plane_size / a->stride;
-
-    return 0;
-}
-
-void ff_audio_data_free(AudioData **a)
-{
-    if (!*a)
-        return;
-    av_free((*a)->buffer);
-    av_freep(a);
-}
-
-int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map)
-{
-    int ret, p;
-
-    /* validate input/output compatibility */
-    if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels)
-        return AVERROR(EINVAL);
-
-    if (map && !src->is_planar) {
-        av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n");
-        return AVERROR(EINVAL);
-    }
-
-    /* if the input is empty, just empty the output */
-    if (!src->nb_samples) {
-        dst->nb_samples = 0;
-        return 0;
-    }
-
-    /* reallocate output if necessary */
-    ret = ff_audio_data_realloc(dst, src->nb_samples);
-    if (ret < 0)
-        return ret;
-
-    /* copy data */
-    if (map) {
-        if (map->do_remap) {
-            for (p = 0; p < src->planes; p++) {
-                if (map->channel_map[p] >= 0)
-                    memcpy(dst->data[p], src->data[map->channel_map[p]],
-                           src->nb_samples * src->stride);
-            }
-        }
-        if (map->do_copy || map->do_zero) {
-            for (p = 0; p < src->planes; p++) {
-                if (map->channel_copy[p])
-                    memcpy(dst->data[p], dst->data[map->channel_copy[p]],
-                           src->nb_samples * src->stride);
-                else if (map->channel_zero[p])
-                    av_samples_set_silence(&dst->data[p], 0, src->nb_samples,
-                                           1, dst->sample_fmt);
-            }
-        }
-    } else {
-        for (p = 0; p < src->planes; p++)
-            memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride);
-    }
-
-    dst->nb_samples = src->nb_samples;
-
-    return 0;
-}
-
-int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
-                          int src_offset, int nb_samples)
-{
-    int ret, p, dst_offset2, dst_move_size;
-
-    /* validate input/output compatibility */
-    if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) {
-        av_log(src, AV_LOG_ERROR, "sample format mismatch\n");
-        return AVERROR(EINVAL);
-    }
-
-    /* validate offsets are within the buffer bounds */
-    if (dst_offset < 0 || dst_offset > dst->nb_samples ||
-        src_offset < 0 || src_offset > src->nb_samples) {
-        av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n",
-               src_offset, dst_offset);
-        return AVERROR(EINVAL);
-    }
-
-    /* check offsets and sizes to see if we can just do nothing and return */
-    if (nb_samples > src->nb_samples - src_offset)
-        nb_samples = src->nb_samples - src_offset;
-    if (nb_samples <= 0)
-        return 0;
-
-    /* validate that the output is not read-only */
-    if (dst->read_only) {
-        av_log(dst, AV_LOG_ERROR, "dst is read-only\n");
-        return AVERROR(EINVAL);
-    }
-
-    /* reallocate output if necessary */
-    ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples);
-    if (ret < 0) {
-        av_log(dst, AV_LOG_ERROR, "error reallocating dst\n");
-        return ret;
-    }
-
-    dst_offset2   = dst_offset + nb_samples;
-    dst_move_size = dst->nb_samples - dst_offset;
-
-    for (p = 0; p < src->planes; p++) {
-        if (dst_move_size > 0) {
-            memmove(dst->data[p] + dst_offset2 * dst->stride,
-                    dst->data[p] + dst_offset  * dst->stride,
-                    dst_move_size * dst->stride);
-        }
-        memcpy(dst->data[p] + dst_offset * dst->stride,
-               src->data[p] + src_offset * src->stride,
-               nb_samples * src->stride);
-    }
-    dst->nb_samples += nb_samples;
-
-    return 0;
-}
-
-void ff_audio_data_drain(AudioData *a, int nb_samples)
-{
-    if (a->nb_samples <= nb_samples) {
-        /* drain the whole buffer */
-        a->nb_samples = 0;
-    } else {
-        int p;
-        int move_offset = a->stride * nb_samples;
-        int move_size   = a->stride * (a->nb_samples - nb_samples);
-
-        for (p = 0; p < a->planes; p++)
-            memmove(a->data[p], a->data[p] + move_offset, move_size);
-
-        a->nb_samples -= nb_samples;
-    }
-}
-
-int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
-                              int nb_samples)
-{
-    uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS];
-    int offset_size, p;
-
-    if (offset >= a->nb_samples)
-        return 0;
-    offset_size = offset * a->stride;
-    for (p = 0; p < a->planes; p++)
-        offset_data[p] = a->data[p] + offset_size;
-
-    return av_audio_fifo_write(af, (void **)offset_data, nb_samples);
-}
-
-int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples)
-{
-    int ret;
-
-    if (a->read_only)
-        return AVERROR(EINVAL);
-
-    ret = ff_audio_data_realloc(a, nb_samples);
-    if (ret < 0)
-        return ret;
-
-    ret = av_audio_fifo_read(af, (void **)a->data, nb_samples);
-    if (ret >= 0)
-        a->nb_samples = ret;
-    return ret;
-}