lavc: remove disabled FF_API_AVCODEC_RESAMPLE cruft

5 files changed, 0 insertions, 805 deletions

diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 33344fb584..addf5a5ae2 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -27,8 +27,6 @@ OBJS = allcodecs.o                                                      \
        options.o                                                        \
        parser.o                                                         \
        raw.o                                                            \
-       resample.o                                                       \
-       resample2.o                                                      \
        simple_idct.o                                                    \
        utils.o                                                          \
 
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 8e89e1f9cf..975ad9840a 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3702,103 +3702,6 @@ int avcodec_encode_subtitle(AVCodecContext *avctx, uint8_t *buf, int buf_size,
  * @}
  */
 
-#if FF_API_AVCODEC_RESAMPLE
-/**
- * @defgroup lavc_resample Audio resampling
- * @ingroup libavc
- * @deprecated use libavresample instead
- *
- * @{
- */
-struct ReSampleContext;
-struct AVResampleContext;
-
-typedef struct ReSampleContext ReSampleContext;
-
-/**
- *  Initialize audio resampling context.
- *
- * @param output_channels  number of output channels
- * @param input_channels   number of input channels
- * @param output_rate      output sample rate
- * @param input_rate       input sample rate
- * @param sample_fmt_out   requested output sample format
- * @param sample_fmt_in    input sample format
- * @param filter_length    length of each FIR filter in the filterbank relative to the cutoff frequency
- * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
- * @param linear           if 1 then the used FIR filter will be linearly interpolated
-                           between the 2 closest, if 0 the closest will be used
- * @param cutoff           cutoff frequency, 1.0 corresponds to half the output sampling rate
- * @return allocated ReSampleContext, NULL if error occurred
- */
-attribute_deprecated
-ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
-                                        int output_rate, int input_rate,
-                                        enum AVSampleFormat sample_fmt_out,
-                                        enum AVSampleFormat sample_fmt_in,
-                                        int filter_length, int log2_phase_count,
-                                        int linear, double cutoff);
-
-attribute_deprecated
-int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples);
-
-/**
- * Free resample context.
- *
- * @param s a non-NULL pointer to a resample context previously
- *          created with av_audio_resample_init()
- */
-attribute_deprecated
-void audio_resample_close(ReSampleContext *s);
-
-
-/**
- * Initialize an audio resampler.
- * Note, if either rate is not an integer then simply scale both rates up so they are.
- * @param filter_length length of each FIR filter in the filterbank relative to the cutoff freq
- * @param log2_phase_count log2 of the number of entries in the polyphase filterbank
- * @param linear If 1 then the used FIR filter will be linearly interpolated
-                 between the 2 closest, if 0 the closest will be used
- * @param cutoff cutoff frequency, 1.0 corresponds to half the output sampling rate
- */
-attribute_deprecated
-struct AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_length, int log2_phase_count, int linear, double cutoff);
-
-/**
- * Resample an array of samples using a previously configured context.
- * @param src an array of unconsumed samples
- * @param consumed the number of samples of src which have been consumed are returned here
- * @param src_size the number of unconsumed samples available
- * @param dst_size the amount of space in samples available in dst
- * @param update_ctx If this is 0 then the context will not be modified, that way several channels can be resampled with the same context.
- * @return the number of samples written in dst or -1 if an error occurred
- */
-attribute_deprecated
-int av_resample(struct AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx);
-
-
-/**
- * Compensate samplerate/timestamp drift. The compensation is done by changing
- * the resampler parameters, so no audible clicks or similar distortions occur
- * @param compensation_distance distance in output samples over which the compensation should be performed
- * @param sample_delta number of output samples which should be output less
- *
- * example: av_resample_compensate(c, 10, 500)
- * here instead of 510 samples only 500 samples would be output
- *
- * note, due to rounding the actual compensation might be slightly different,
- * especially if the compensation_distance is large and the in_rate used during init is small
- */
-attribute_deprecated
-void av_resample_compensate(struct AVResampleContext *c, int sample_delta, int compensation_distance);
-attribute_deprecated
-void av_resample_close(struct AVResampleContext *c);
-
-/**
- * @}
- */
-#endif
-
 /**
  * @addtogroup lavc_picture
  * @{
diff --git a/libavcodec/resample.c b/libavcodec/resample.c
deleted file mode 100644
index 1b3bb834f3..0000000000
--- a/libavcodec/resample.c
+++ /dev/null
@@ -1,379 +0,0 @@
-/*
- * samplerate conversion for both audio and video
- * Copyright (c) 2000 Fabrice Bellard
- *
- * 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
- */
-
-/**
- * @file
- * samplerate conversion for both audio and video
- */
-
-#include <string.h>
-
-#include "avcodec.h"
-#include "audioconvert.h"
-#include "libavutil/opt.h"
-#include "libavutil/mem.h"
-#include "libavutil/samplefmt.h"
-
-#if FF_API_AVCODEC_RESAMPLE
-
-#define MAX_CHANNELS 8
-
-struct AVResampleContext;
-
-static const char *context_to_name(void *ptr)
-{
-    return "audioresample";
-}
-
-static const AVOption options[] = {{NULL}};
-static const AVClass audioresample_context_class = {
-    "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
-};
-
-struct ReSampleContext {
-    struct AVResampleContext *resample_context;
-    short *temp[MAX_CHANNELS];
-    int temp_len;
-    float ratio;
-    /* channel convert */
-    int input_channels, output_channels, filter_channels;
-    AVAudioConvert *convert_ctx[2];
-    enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
-    unsigned sample_size[2];           ///< size of one sample in sample_fmt
-    short *buffer[2];                  ///< buffers used for conversion to S16
-    unsigned buffer_size[2];           ///< sizes of allocated buffers
-};
-
-/* n1: number of samples */
-static void stereo_to_mono(short *output, short *input, int n1)
-{
-    short *p, *q;
-    int n = n1;
-
-    p = input;
-    q = output;
-    while (n >= 4) {
-        q[0] = (p[0] + p[1]) >> 1;
-        q[1] = (p[2] + p[3]) >> 1;
-        q[2] = (p[4] + p[5]) >> 1;
-        q[3] = (p[6] + p[7]) >> 1;
-        q += 4;
-        p += 8;
-        n -= 4;
-    }
-    while (n > 0) {
-        q[0] = (p[0] + p[1]) >> 1;
-        q++;
-        p += 2;
-        n--;
-    }
-}
-
-/* n1: number of samples */
-static void mono_to_stereo(short *output, short *input, int n1)
-{
-    short *p, *q;
-    int n = n1;
-    int v;
-
-    p = input;
-    q = output;
-    while (n >= 4) {
-        v = p[0]; q[0] = v; q[1] = v;
-        v = p[1]; q[2] = v; q[3] = v;
-        v = p[2]; q[4] = v; q[5] = v;
-        v = p[3]; q[6] = v; q[7] = v;
-        q += 8;
-        p += 4;
-        n -= 4;
-    }
-    while (n > 0) {
-        v = p[0]; q[0] = v; q[1] = v;
-        q += 2;
-        p += 1;
-        n--;
-    }
-}
-
-static void deinterleave(short **output, short *input, int channels, int samples)
-{
-    int i, j;
-
-    for (i = 0; i < samples; i++) {
-        for (j = 0; j < channels; j++) {
-            *output[j]++ = *input++;
-        }
-    }
-}
-
-static void interleave(short *output, short **input, int channels, int samples)
-{
-    int i, j;
-
-    for (i = 0; i < samples; i++) {
-        for (j = 0; j < channels; j++) {
-            *output++ = *input[j]++;
-        }
-    }
-}
-
-static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
-{
-    int i;
-    short l, r;
-
-    for (i = 0; i < n; i++) {
-        l = *input1++;
-        r = *input2++;
-        *output++ = l;                  /* left */
-        *output++ = (l / 2) + (r / 2);  /* center */
-        *output++ = r;                  /* right */
-        *output++ = 0;                  /* left surround */
-        *output++ = 0;                  /* right surroud */
-        *output++ = 0;                  /* low freq */
-    }
-}
-
-ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
-                                        int output_rate, int input_rate,
-                                        enum AVSampleFormat sample_fmt_out,
-                                        enum AVSampleFormat sample_fmt_in,
-                                        int filter_length, int log2_phase_count,
-                                        int linear, double cutoff)
-{
-    ReSampleContext *s;
-
-    if (input_channels > MAX_CHANNELS) {
-        av_log(NULL, AV_LOG_ERROR,
-               "Resampling with input channels greater than %d is unsupported.\n",
-               MAX_CHANNELS);
-        return NULL;
-    }
-    if (output_channels != input_channels &&
-        (input_channels  > 2 ||
-         output_channels > 2 &&
-         !(output_channels == 6 && input_channels == 2))) {
-        av_log(NULL, AV_LOG_ERROR,
-               "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n");
-        return NULL;
-    }
-
-    s = av_mallocz(sizeof(ReSampleContext));
-    if (!s) {
-        av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
-        return NULL;
-    }
-
-    s->ratio = (float)output_rate / (float)input_rate;
-
-    s->input_channels = input_channels;
-    s->output_channels = output_channels;
-
-    s->filter_channels = s->input_channels;
-    if (s->output_channels < s->filter_channels)
-        s->filter_channels = s->output_channels;
-
-    s->sample_fmt[0]  = sample_fmt_in;
-    s->sample_fmt[1]  = sample_fmt_out;
-    s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
-    s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
-
-    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
-        if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
-                                                         s->sample_fmt[0], 1, NULL, 0))) {
-            av_log(s, AV_LOG_ERROR,
-                   "Cannot convert %s sample format to s16 sample format\n",
-                   av_get_sample_fmt_name(s->sample_fmt[0]));
-            av_free(s);
-            return NULL;
-        }
-    }
-
-    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
-        if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
-                                                         AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
-            av_log(s, AV_LOG_ERROR,
-                   "Cannot convert s16 sample format to %s sample format\n",
-                   av_get_sample_fmt_name(s->sample_fmt[1]));
-            av_audio_convert_free(s->convert_ctx[0]);
-            av_free(s);
-            return NULL;
-        }
-    }
-
-    s->resample_context = av_resample_init(output_rate, input_rate,
-                                           filter_length, log2_phase_count,
-                                           linear, cutoff);
-
-    *(const AVClass**)s->resample_context = &audioresample_context_class;
-
-    return s;
-}
-
-/* resample audio. 'nb_samples' is the number of input samples */
-/* XXX: optimize it ! */
-int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
-{
-    int i, nb_samples1;
-    short *bufin[MAX_CHANNELS];
-    short *bufout[MAX_CHANNELS];
-    short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
-    short *output_bak = NULL;
-    int lenout;
-
-    if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
-        /* nothing to do */
-        memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
-        return nb_samples;
-    }
-
-    if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
-        int istride[1] = { s->sample_size[0] };
-        int ostride[1] = { 2 };
-        const void *ibuf[1] = { input };
-        void       *obuf[1];
-        unsigned input_size = nb_samples * s->input_channels * 2;
-
-        if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
-            av_free(s->buffer[0]);
-            s->buffer_size[0] = input_size;
-            s->buffer[0] = av_malloc(s->buffer_size[0]);
-            if (!s->buffer[0]) {
-                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
-                return 0;
-            }
-        }
-
-        obuf[0] = s->buffer[0];
-
-        if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
-                             ibuf, istride, nb_samples * s->input_channels) < 0) {
-            av_log(s->resample_context, AV_LOG_ERROR,
-                   "Audio sample format conversion failed\n");
-            return 0;
-        }
-
-        input = s->buffer[0];
-    }
-
-    lenout = 4 * nb_samples * s->ratio + 16;
-
-    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
-        int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
-                       s->output_channels;
-        output_bak = output;
-
-        if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
-            av_free(s->buffer[1]);
-            s->buffer_size[1] = out_size;
-            s->buffer[1] = av_malloc(s->buffer_size[1]);
-            if (!s->buffer[1]) {
-                av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
-                return 0;
-            }
-        }
-
-        output = s->buffer[1];
-    }
-
-    /* XXX: move those malloc to resample init code */
-    for (i = 0; i < s->filter_channels; i++) {
-        bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
-        memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
-        buftmp2[i] = bufin[i] + s->temp_len;
-        bufout[i] = av_malloc(lenout * sizeof(short));
-    }
-
-    if (s->input_channels == 2 && s->output_channels == 1) {
-        buftmp3[0] = output;
-        stereo_to_mono(buftmp2[0], input, nb_samples);
-    } else if (s->output_channels >= 2 && s->input_channels == 1) {
-        buftmp3[0] = bufout[0];
-        memcpy(buftmp2[0], input, nb_samples * sizeof(short));
-    } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
-        for (i = 0; i < s->input_channels; i++) {
-            buftmp3[i] = bufout[i];
-        }
-        deinterleave(buftmp2, input, s->input_channels, nb_samples);
-    } else {
-        buftmp3[0] = output;
-        memcpy(buftmp2[0], input, nb_samples * sizeof(short));
-    }
-
-    nb_samples += s->temp_len;
-
-    /* resample each channel */
-    nb_samples1 = 0; /* avoid warning */
-    for (i = 0; i < s->filter_channels; i++) {
-        int consumed;
-        int is_last = i + 1 == s->filter_channels;
-
-        nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
-                                  &consumed, nb_samples, lenout, is_last);
-        s->temp_len = nb_samples - consumed;
-        s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
-        memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
-    }
-
-    if (s->output_channels == 2 && s->input_channels == 1) {
-        mono_to_stereo(output, buftmp3[0], nb_samples1);
-    } else if (s->output_channels == 6 && s->input_channels == 2) {
-        ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
-    } else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
-        interleave(output, buftmp3, s->output_channels, nb_samples1);
-    }
-
-    if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
-        int istride[1] = { 2 };
-        int ostride[1] = { s->sample_size[1] };
-        const void *ibuf[1] = { output };
-        void       *obuf[1] = { output_bak };
-
-        if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
-                             ibuf, istride, nb_samples1 * s->output_channels) < 0) {
-            av_log(s->resample_context, AV_LOG_ERROR,
-                   "Audio sample format conversion failed\n");
-            return 0;
-        }
-    }
-
-    for (i = 0; i < s->filter_channels; i++) {
-        av_free(bufin[i]);
-        av_free(bufout[i]);
-    }
-
-    return nb_samples1;
-}
-
-void audio_resample_close(ReSampleContext *s)
-{
-    int i;
-    av_resample_close(s->resample_context);
-    for (i = 0; i < s->filter_channels; i++)
-        av_freep(&s->temp[i]);
-    av_freep(&s->buffer[0]);
-    av_freep(&s->buffer[1]);
-    av_audio_convert_free(s->convert_ctx[0]);
-    av_audio_convert_free(s->convert_ctx[1]);
-    av_free(s);
-}
-
-#endif
diff --git a/libavcodec/resample2.c b/libavcodec/resample2.c
deleted file mode 100644
index 5b6e85514a..0000000000
--- a/libavcodec/resample2.c
+++ /dev/null
@@ -1,324 +0,0 @@
-/*
- * audio resampling
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- *
- * 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
- */
-
-/**
- * @file
- * audio resampling
- * @author Michael Niedermayer <michaelni@gmx.at>
- */
-
-#include "avcodec.h"
-#include "libavutil/common.h"
-
-#if FF_API_AVCODEC_RESAMPLE
-
-#ifndef CONFIG_RESAMPLE_HP
-#define FILTER_SHIFT 15
-
-#define FELEM int16_t
-#define FELEM2 int32_t
-#define FELEML int64_t
-#define FELEM_MAX INT16_MAX
-#define FELEM_MIN INT16_MIN
-#define WINDOW_TYPE 9
-#elif !defined(CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE)
-#define FILTER_SHIFT 30
-
-#define FELEM int32_t
-#define FELEM2 int64_t
-#define FELEML int64_t
-#define FELEM_MAX INT32_MAX
-#define FELEM_MIN INT32_MIN
-#define WINDOW_TYPE 12
-#else
-#define FILTER_SHIFT 0
-
-#define FELEM double
-#define FELEM2 double
-#define FELEML double
-#define WINDOW_TYPE 24
-#endif
-
-
-typedef struct AVResampleContext{
-    const AVClass *av_class;
-    FELEM *filter_bank;
-    int filter_length;
-    int ideal_dst_incr;
-    int dst_incr;
-    int index;
-    int frac;
-    int src_incr;
-    int compensation_distance;
-    int phase_shift;
-    int phase_mask;
-    int linear;
-}AVResampleContext;
-
-/**
- * 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.
- * @param factor resampling factor
- * @param scale wanted sum of coefficients for each filter
- * @param type 0->cubic, 1->blackman nuttall windowed sinc, 2..16->kaiser windowed sinc beta=2..16
- * @return 0 on success, negative on error
- */
-static int build_filter(FELEM *filter, double factor, int tap_count, int phase_count, int scale, int type){
-    int ph, i;
-    double x, y, w;
-    double *tab = av_malloc(tap_count * sizeof(*tab));
-    const int center= (tap_count-1)/2;
-
-    if (!tab)
-        return AVERROR(ENOMEM);
-
-    /* if upsampling, only need to interpolate, no filter */
-    if (factor > 1.0)
-        factor = 1.0;
-
-    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(type){
-            case 0:{
-                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 1:
-                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;
-            default:
-                w = 2.0*x / (factor*tap_count*M_PI);
-                y *= bessel(type*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++) {
-#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
-            filter[ph * tap_count + i] = tab[i] / norm;
-#else
-            filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm), FELEM_MIN, FELEM_MAX);
-#endif
-        }
-    }
-#if 0
-    {
-#define LEN 1024
-        int j,k;
-        double sine[LEN + tap_count];
-        double filtered[LEN];
-        double maxff=-2, minff=2, maxsf=-2, minsf=2;
-        for(i=0; i<LEN; i++){
-            double ss=0, sf=0, ff=0;
-            for(j=0; j<LEN+tap_count; j++)
-                sine[j]= cos(i*j*M_PI/LEN);
-            for(j=0; j<LEN; j++){
-                double sum=0;
-                ph=0;
-                for(k=0; k<tap_count; k++)
-                    sum += filter[ph * tap_count + k] * sine[k+j];
-                filtered[j]= sum / (1<<FILTER_SHIFT);
-                ss+= sine[j + center] * sine[j + center];
-                ff+= filtered[j] * filtered[j];
-                sf+= sine[j + center] * filtered[j];
-            }
-            ss= sqrt(2*ss/LEN);
-            ff= sqrt(2*ff/LEN);
-            sf= 2*sf/LEN;
-            maxff= FFMAX(maxff, ff);
-            minff= FFMIN(minff, ff);
-            maxsf= FFMAX(maxsf, sf);
-            minsf= FFMIN(minsf, sf);
-            if(i%11==0){
-                av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf);
-                minff=minsf= 2;
-                maxff=maxsf= -2;
-            }
-        }
-    }
-#endif
-
-    av_free(tab);
-    return 0;
-}
-
-AVResampleContext *av_resample_init(int out_rate, int in_rate, int filter_size, int phase_shift, int linear, double cutoff){
-    AVResampleContext *c= av_mallocz(sizeof(AVResampleContext));
-    double factor= FFMIN(out_rate * cutoff / in_rate, 1.0);
-    int phase_count= 1<<phase_shift;
-
-    if (!c)
-        return NULL;
-
-    c->phase_shift= phase_shift;
-    c->phase_mask= phase_count-1;
-    c->linear= linear;
-
-    c->filter_length= FFMAX((int)ceil(filter_size/factor), 1);
-    c->filter_bank= av_mallocz(c->filter_length*(phase_count+1)*sizeof(FELEM));
-    if (!c->filter_bank)
-        goto error;
-    if (build_filter(c->filter_bank, factor, c->filter_length, phase_count, 1<<FILTER_SHIFT, WINDOW_TYPE))
-        goto error;
-    memcpy(&c->filter_bank[c->filter_length*phase_count+1], c->filter_bank, (c->filter_length-1)*sizeof(FELEM));
-    c->filter_bank[c->filter_length*phase_count]= c->filter_bank[c->filter_length - 1];
-
-    c->src_incr= out_rate;
-    c->ideal_dst_incr= c->dst_incr= in_rate * phase_count;
-    c->index= -phase_count*((c->filter_length-1)/2);
-
-    return c;
-error:
-    av_free(c->filter_bank);
-    av_free(c);
-    return NULL;
-}
-
-void av_resample_close(AVResampleContext *c){
-    av_freep(&c->filter_bank);
-    av_freep(&c);
-}
-
-void av_resample_compensate(AVResampleContext *c, int sample_delta, int compensation_distance){
-//    sample_delta += (c->ideal_dst_incr - c->dst_incr)*(int64_t)c->compensation_distance / c->ideal_dst_incr;
-    c->compensation_distance= compensation_distance;
-    c->dst_incr = c->ideal_dst_incr - c->ideal_dst_incr * (int64_t)sample_delta / compensation_distance;
-}
-
-int av_resample(AVResampleContext *c, short *dst, short *src, int *consumed, int src_size, int dst_size, int update_ctx){
-    int dst_index, i;
-    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(compensation_distance == 0 && c->filter_length == 1 && c->phase_shift==0){
-        int64_t index2= ((int64_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);
-
-        for(dst_index=0; dst_index < dst_size; dst_index++){
-            dst[dst_index] = src[index2>>32];
-            index2 += incr;
-        }
-        frac += dst_index * dst_incr_frac;
-        index += dst_index * dst_incr;
-        index += frac / c->src_incr;
-        frac %= c->src_incr;
-  }else{
-    for(dst_index=0; dst_index < dst_size; dst_index++){
-        FELEM *filter= c->filter_bank + c->filter_length*(index & c->phase_mask);
-        int sample_index= index >> c->phase_shift;
-        FELEM2 val=0;
-
-        if(sample_index < 0){
-            for(i=0; i<c->filter_length; i++)
-                val += src[FFABS(sample_index + i) % src_size] * filter[i];
-        }else if(sample_index + c->filter_length > src_size){
-            break;
-        }else if(c->linear){
-            FELEM2 v2=0;
-            for(i=0; i<c->filter_length; i++){
-                val += src[sample_index + i] * (FELEM2)filter[i];
-                v2  += src[sample_index + i] * (FELEM2)filter[i + c->filter_length];
-            }
-            val+=(v2-val)*(FELEML)frac / c->src_incr;
-        }else{
-            for(i=0; i<c->filter_length; i++){
-                val += src[sample_index + i] * (FELEM2)filter[i];
-            }
-        }
-
-#ifdef CONFIG_RESAMPLE_AUDIOPHILE_KIDDY_MODE
-        dst[dst_index] = av_clip_int16(lrintf(val));
-#else
-        val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
-        dst[dst_index] = (unsigned)(val + 32768) > 65535 ? (val>>31) ^ 32767 : val;
-#endif
-
-        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;
-        }
-    }
-  }
-    *consumed= FFMAX(index, 0) >> c->phase_shift;
-    if(index>=0) index &= c->phase_mask;
-
-    if(compensation_distance){
-        compensation_distance -= dst_index;
-        assert(compensation_distance > 0);
-    }
-    if(update_ctx){
-        c->frac= frac;
-        c->index= index;
-        c->dst_incr= dst_incr_frac + c->src_incr*dst_incr;
-        c->compensation_distance= compensation_distance;
-    }
-#if 0
-    if(update_ctx && !c->compensation_distance){
-#undef rand
-        av_resample_compensate(c, rand() % (8000*2) - 8000, 8000*2);
-av_log(NULL, AV_LOG_DEBUG, "%d %d %d\n", c->dst_incr, c->ideal_dst_incr, c->compensation_distance);
-    }
-#endif
-
-    return dst_index;
-}
-
-#endif
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 20d30efaa3..82c97ed027 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -49,9 +49,6 @@
 #ifndef FF_API_REQUEST_CHANNELS
 #define FF_API_REQUEST_CHANNELS (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
-#ifndef FF_API_AVCODEC_RESAMPLE
-#define FF_API_AVCODEC_RESAMPLE  (LIBAVCODEC_VERSION_MAJOR < 55)
-#endif
 #ifndef FF_API_LIBMPEG2
 #define FF_API_LIBMPEG2          (LIBAVCODEC_VERSION_MAJOR < 55)
 #endif