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, 446 deletions

diff --git a/libavresample/resample.c b/libavresample/resample.c
deleted file mode 100644
index dc14cc2d2a..0000000000
--- a/libavresample/resample.c
+++ /dev/null
@@ -1,446 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- * 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 "libavutil/common.h"
-#include "libavutil/libm.h"
-#include "libavutil/log.h"
-#include "internal.h"
-#include "resample.h"
-#include "audio_data.h"
-
-
-/* double template */
-#define CONFIG_RESAMPLE_DBL
-#include "resample_template.c"
-#undef CONFIG_RESAMPLE_DBL
-
-/* float template */
-#define CONFIG_RESAMPLE_FLT
-#include "resample_template.c"
-#undef CONFIG_RESAMPLE_FLT
-
-/* s32 template */
-#define CONFIG_RESAMPLE_S32
-#include "resample_template.c"
-#undef CONFIG_RESAMPLE_S32
-
-/* s16 template */
-#include "resample_template.c"
-
-
-/* 0th order modified Bessel function of the first kind. */
-static double bessel(double x)
-{
-    double v     = 1;
-    double lastv = 0;
-    double t     = 1;
-    int i;
-
-    x = x * x / 4;
-    for (i = 1; v != lastv; i++) {
-        lastv = v;
-        t    *= x / (i * i);
-        v    += t;
-    }
-    return v;
-}
-
-/* Build a polyphase filterbank. */
-static int build_filter(ResampleContext *c, double factor)
-{
-    int ph, i;
-    double x, y, w;
-    double *tab;
-    int tap_count    = c->filter_length;
-    int phase_count  = 1 << c->phase_shift;
-    const int center = (tap_count - 1) / 2;
-
-    tab = av_malloc(tap_count * sizeof(*tab));
-    if (!tab)
-        return AVERROR(ENOMEM);
-
-    for (ph = 0; ph < phase_count; ph++) {
-        double norm = 0;
-        for (i = 0; i < tap_count; i++) {
-            x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
-            if (x == 0) y = 1.0;
-            else        y = sin(x) / x;
-            switch (c->filter_type) {
-            case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
-                const float d = -0.5; //first order derivative = -0.5
-                x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
-                if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * (                -x*x + x*x*x);
-                else         y =                           d * (-4 + 8 * x - 5 * x*x + x*x*x);
-                break;
-            }
-            case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
-                w  = 2.0 * x / (factor * tap_count) + M_PI;
-                y *= 0.3635819 - 0.4891775 * cos(    w) +
-                                 0.1365995 * cos(2 * w) -
-                                 0.0106411 * cos(3 * w);
-                break;
-            case AV_RESAMPLE_FILTER_TYPE_KAISER:
-                w  = 2.0 * x / (factor * tap_count * M_PI);
-                y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
-                break;
-            }
-
-            tab[i] = y;
-            norm  += y;
-        }
-        /* normalize so that an uniform color remains the same */
-        for (i = 0; i < tap_count; i++)
-            tab[i] = tab[i] / norm;
-
-        c->set_filter(c->filter_bank, tab, ph, tap_count);
-    }
-
-    av_free(tab);
-    return 0;
-}
-
-ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
-{
-    ResampleContext *c;
-    int out_rate    = avr->out_sample_rate;
-    int in_rate     = avr->in_sample_rate;
-    double factor   = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
-    int phase_count = 1 << avr->phase_shift;
-    int felem_size;
-
-    if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
-        avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
-        avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
-        avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
-        av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
-               "resampling: %s\n",
-               av_get_sample_fmt_name(avr->internal_sample_fmt));
-        return NULL;
-    }
-    c = av_mallocz(sizeof(*c));
-    if (!c)
-        return NULL;
-
-    c->avr           = avr;
-    c->phase_shift   = avr->phase_shift;
-    c->phase_mask    = phase_count - 1;
-    c->linear        = avr->linear_interp;
-    c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
-    c->filter_type   = avr->filter_type;
-    c->kaiser_beta   = avr->kaiser_beta;
-
-    switch (avr->internal_sample_fmt) {
-    case AV_SAMPLE_FMT_DBLP:
-        c->resample_one  = c->linear ? resample_linear_dbl : resample_one_dbl;
-        c->resample_nearest = resample_nearest_dbl;
-        c->set_filter    = set_filter_dbl;
-        break;
-    case AV_SAMPLE_FMT_FLTP:
-        c->resample_one  = c->linear ? resample_linear_flt : resample_one_flt;
-        c->resample_nearest = resample_nearest_flt;
-        c->set_filter    = set_filter_flt;
-        break;
-    case AV_SAMPLE_FMT_S32P:
-        c->resample_one  = c->linear ? resample_linear_s32 : resample_one_s32;
-        c->resample_nearest = resample_nearest_s32;
-        c->set_filter    = set_filter_s32;
-        break;
-    case AV_SAMPLE_FMT_S16P:
-        c->resample_one  = c->linear ? resample_linear_s16 : resample_one_s16;
-        c->resample_nearest = resample_nearest_s16;
-        c->set_filter    = set_filter_s16;
-        break;
-    }
-
-    if (ARCH_AARCH64)
-        ff_audio_resample_init_aarch64(c, avr->internal_sample_fmt);
-    if (ARCH_ARM)
-        ff_audio_resample_init_arm(c, avr->internal_sample_fmt);
-
-    felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
-    c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
-    if (!c->filter_bank)
-        goto error;
-
-    if (build_filter(c, factor) < 0)
-        goto error;
-
-    memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
-           c->filter_bank, (c->filter_length - 1) * felem_size);
-    memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
-           &c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
-
-    c->compensation_distance = 0;
-    if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
-                   in_rate * (int64_t)phase_count, INT32_MAX / 2))
-        goto error;
-    c->ideal_dst_incr = c->dst_incr;
-
-    c->padding_size   = (c->filter_length - 1) / 2;
-    c->initial_padding_filled = 0;
-    c->index = 0;
-    c->frac  = 0;
-
-    /* allocate internal buffer */
-    c->buffer = ff_audio_data_alloc(avr->resample_channels, c->padding_size,
-                                    avr->internal_sample_fmt,
-                                    "resample buffer");
-    if (!c->buffer)
-        goto error;
-    c->buffer->nb_samples      = c->padding_size;
-    c->initial_padding_samples = c->padding_size;
-
-    av_log(avr, AV_LOG_DEBUG, "resample: %s from %d Hz to %d Hz\n",
-           av_get_sample_fmt_name(avr->internal_sample_fmt),
-           avr->in_sample_rate, avr->out_sample_rate);
-
-    return c;
-
-error:
-    ff_audio_data_free(&c->buffer);
-    av_free(c->filter_bank);
-    av_free(c);
-    return NULL;
-}
-
-void ff_audio_resample_free(ResampleContext **c)
-{
-    if (!*c)
-        return;
-    ff_audio_data_free(&(*c)->buffer);
-    av_free((*c)->filter_bank);
-    av_freep(c);
-}
-
-int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
-                                int compensation_distance)
-{
-    ResampleContext *c;
-
-    if (compensation_distance < 0)
-        return AVERROR(EINVAL);
-    if (!compensation_distance && sample_delta)
-        return AVERROR(EINVAL);
-
-    if (!avr->resample_needed) {
-        av_log(avr, AV_LOG_ERROR, "Unable to set resampling compensation\n");
-        return AVERROR(EINVAL);
-    }
-    c = avr->resample;
-    c->compensation_distance = compensation_distance;
-    if (compensation_distance) {
-        c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr *
-                      (int64_t)sample_delta / compensation_distance;
-    } else {
-        c->dst_incr = c->ideal_dst_incr;
-    }
-
-    return 0;
-}
-
-static int resample(ResampleContext *c, void *dst, const void *src,
-                    int *consumed, int src_size, int dst_size, int update_ctx,
-                    int nearest_neighbour)
-{
-    int dst_index;
-    unsigned int index = c->index;
-    int frac          = c->frac;
-    int dst_incr_frac = c->dst_incr % c->src_incr;
-    int dst_incr      = c->dst_incr / c->src_incr;
-    int compensation_distance = c->compensation_distance;
-
-    if (!dst != !src)
-        return AVERROR(EINVAL);
-
-    if (nearest_neighbour) {
-        uint64_t index2 = ((uint64_t)index) << 32;
-        int64_t incr   = (1LL << 32) * c->dst_incr / c->src_incr;
-        dst_size       = FFMIN(dst_size,
-                               (src_size-1-index) * (int64_t)c->src_incr /
-                               c->dst_incr);
-
-        if (dst) {
-            for(dst_index = 0; dst_index < dst_size; dst_index++) {
-                c->resample_nearest(dst, dst_index, src, index2 >> 32);
-                index2 += incr;
-            }
-        } else {
-            dst_index = dst_size;
-        }
-        index += dst_index * dst_incr;
-        index += (frac + dst_index * (int64_t)dst_incr_frac) / c->src_incr;
-        frac   = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
-    } else {
-        for (dst_index = 0; dst_index < dst_size; dst_index++) {
-            int sample_index = index >> c->phase_shift;
-
-            if (sample_index + c->filter_length > src_size)
-                break;
-
-            if (dst)
-                c->resample_one(c, dst, dst_index, src, index, frac);
-
-            frac  += dst_incr_frac;
-            index += dst_incr;
-            if (frac >= c->src_incr) {
-                frac -= c->src_incr;
-                index++;
-            }
-            if (dst_index + 1 == compensation_distance) {
-                compensation_distance = 0;
-                dst_incr_frac = c->ideal_dst_incr % c->src_incr;
-                dst_incr      = c->ideal_dst_incr / c->src_incr;
-            }
-        }
-    }
-    if (consumed)
-        *consumed = index >> c->phase_shift;
-
-    if (update_ctx) {
-        index &= c->phase_mask;
-
-        if (compensation_distance) {
-            compensation_distance -= dst_index;
-            if (compensation_distance <= 0)
-                return AVERROR_BUG;
-        }
-        c->frac     = frac;
-        c->index    = index;
-        c->dst_incr = dst_incr_frac + c->src_incr*dst_incr;
-        c->compensation_distance = compensation_distance;
-    }
-
-    return dst_index;
-}
-
-int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src)
-{
-    int ch, in_samples, in_leftover, consumed = 0, out_samples = 0;
-    int ret = AVERROR(EINVAL);
-    int nearest_neighbour = (c->compensation_distance == 0 &&
-                             c->filter_length == 1 &&
-                             c->phase_shift == 0);
-
-    in_samples  = src ? src->nb_samples : 0;
-    in_leftover = c->buffer->nb_samples;
-
-    /* add input samples to the internal buffer */
-    if (src) {
-        ret = ff_audio_data_combine(c->buffer, in_leftover, src, 0, in_samples);
-        if (ret < 0)
-            return ret;
-    } else if (in_leftover <= c->final_padding_samples) {
-        /* no remaining samples to flush */
-        return 0;
-    }
-
-    if (!c->initial_padding_filled) {
-        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
-        int i;
-
-        if (src && c->buffer->nb_samples < 2 * c->padding_size)
-            return 0;
-
-        for (i = 0; i < c->padding_size; i++)
-            for (ch = 0; ch < c->buffer->channels; ch++) {
-                if (c->buffer->nb_samples > 2 * c->padding_size - i) {
-                    memcpy(c->buffer->data[ch] + bps * i,
-                           c->buffer->data[ch] + bps * (2 * c->padding_size - i), bps);
-                } else {
-                    memset(c->buffer->data[ch] + bps * i, 0, bps);
-                }
-            }
-        c->initial_padding_filled = 1;
-    }
-
-    if (!src && !c->final_padding_filled) {
-        int bps = av_get_bytes_per_sample(c->avr->internal_sample_fmt);
-        int i;
-
-        ret = ff_audio_data_realloc(c->buffer,
-                                    FFMAX(in_samples, in_leftover) +
-                                    c->padding_size);
-        if (ret < 0) {
-            av_log(c->avr, AV_LOG_ERROR, "Error reallocating resampling buffer\n");
-            return AVERROR(ENOMEM);
-        }
-
-        for (i = 0; i < c->padding_size; i++)
-            for (ch = 0; ch < c->buffer->channels; ch++) {
-                if (in_leftover > i) {
-                    memcpy(c->buffer->data[ch] + bps * (in_leftover + i),
-                           c->buffer->data[ch] + bps * (in_leftover - i - 1),
-                           bps);
-                } else {
-                    memset(c->buffer->data[ch] + bps * (in_leftover + i),
-                           0, bps);
-                }
-            }
-        c->buffer->nb_samples   += c->padding_size;
-        c->final_padding_samples = c->padding_size;
-        c->final_padding_filled  = 1;
-    }
-
-
-    /* calculate output size and reallocate output buffer if needed */
-    /* TODO: try to calculate this without the dummy resample() run */
-    if (!dst->read_only && dst->allow_realloc) {
-        out_samples = resample(c, NULL, NULL, NULL, c->buffer->nb_samples,
-                               INT_MAX, 0, nearest_neighbour);
-        ret = ff_audio_data_realloc(dst, out_samples);
-        if (ret < 0) {
-            av_log(c->avr, AV_LOG_ERROR, "error reallocating output\n");
-            return ret;
-        }
-    }
-
-    /* resample each channel plane */
-    for (ch = 0; ch < c->buffer->channels; ch++) {
-        out_samples = resample(c, (void *)dst->data[ch],
-                               (const void *)c->buffer->data[ch], &consumed,
-                               c->buffer->nb_samples, dst->allocated_samples,
-                               ch + 1 == c->buffer->channels, nearest_neighbour);
-    }
-    if (out_samples < 0) {
-        av_log(c->avr, AV_LOG_ERROR, "error during resampling\n");
-        return out_samples;
-    }
-
-    /* drain consumed samples from the internal buffer */
-    ff_audio_data_drain(c->buffer, consumed);
-    c->initial_padding_samples = FFMAX(c->initial_padding_samples - consumed, 0);
-
-    av_log(c->avr, AV_LOG_TRACE, "resampled %d in + %d leftover to %d out + %d leftover\n",
-            in_samples, in_leftover, out_samples, c->buffer->nb_samples);
-
-    dst->nb_samples = out_samples;
-    return 0;
-}
-
-int avresample_get_delay(AVAudioResampleContext *avr)
-{
-    ResampleContext *c = avr->resample;
-
-    if (!avr->resample_needed || !avr->resample)
-        return 0;
-
-    return FFMAX(c->buffer->nb_samples - c->padding_size, 0);
-}