Revert "Merge commit '0517c9e0'" bring the old...

Revert "Merge commit '0517c9e0'" bring the old audio resampling API back

This reverts commit d3edc65d, reversing
changes made to 150de78d.

Conflicts:

	libavcodec/version.h

It seems there are several applications still using it
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>

libavcodec/Makefile

@@ -27,6 +27,8 @@ OBJS = allcodecs.o                                                      \
        parser.o                                                         \
        raw.o                                                            \
        rawdec.o                                                         \
+       resample.o                                                       \
+       resample2.o                                                      \
        utils.o                                                          \
 
 # parts needed for many different codecs

libavcodec/avcodec.h

@@ -4096,6 +4096,103 @@ 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 libswresample 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
  * @{

libavcodec/resample.c

+/*
+ * samplerate conversion for both audio and video
+ * Copyright (c) 2000 Fabrice Bellard
+ *
+ * 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
+ */
+
+/**
+ * @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--;
+    }
+}
+
+/*
+5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
+- Left = front_left + rear_gain * rear_left + center_gain * center
+- Right = front_right + rear_gain * rear_right + center_gain * center
+Where rear_gain is usually around 0.5-1.0 and
+      center_gain is almost always 0.7 (-3 dB)
+*/
+static void surround_to_stereo(short **output, short *input, int channels, int samples)
+{
+    int i;
+    short l, r;
+
+    for (i = 0; i < samples; i++) {
+        int fl,fr,c,rl,rr;
+        fl = input[0];
+        fr = input[1];
+        c = input[2];
+        // lfe = input[3];
+        rl = input[4];
+        rr = input[5];
+
+        l = av_clip_int16(fl + (0.5 * rl) + (0.7 * c));
+        r = av_clip_int16(fr + (0.5 * rr) + (0.7 * c));
+
+        /* output l & r. */
+        *output[0]++ = l;
+        *output[1]++ = r;
+
+        /* increment input. */
+        input += channels;
+    }
+}
+
+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 */
+    }
+}
+
+#define SUPPORT_RESAMPLE(ch1, ch2, ch3, ch4, ch5, ch6, ch7, ch8) \
+    ch8<<7 | ch7<<6 | ch6<<5 | ch5<<4 | ch4<<3 | ch3<<2 | ch2<<1 | ch1<<0
+
+static const uint8_t supported_resampling[MAX_CHANNELS] = {
+    // output ch:    1  2  3  4  5  6  7  8
+    SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 0, 0, 0), // 1 input channel
+    SUPPORT_RESAMPLE(1, 1, 0, 0, 0, 1, 0, 0), // 2 input channels
+    SUPPORT_RESAMPLE(0, 0, 1, 0, 0, 0, 0, 0), // 3 input channels
+    SUPPORT_RESAMPLE(0, 0, 0, 1, 0, 0, 0, 0), // 4 input channels
+    SUPPORT_RESAMPLE(0, 0, 0, 0, 1, 0, 0, 0), // 5 input channels
+    SUPPORT_RESAMPLE(0, 1, 0, 0, 0, 1, 0, 0), // 6 input channels
+    SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 1, 0), // 7 input channels
+    SUPPORT_RESAMPLE(0, 0, 0, 0, 0, 0, 0, 1), // 8 input channels
+};
+
+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 (!(supported_resampling[input_channels-1] & (1<<(output_channels-1)))) {
+        int i;
+        av_log(NULL, AV_LOG_ERROR, "Unsupported audio resampling. Allowed "
+               "output channels for %d input channel%s", input_channels,
+               input_channels > 1 ? "s:" : ":");
+        for (i = 0; i < MAX_CHANNELS; i++)
+            if (supported_resampling[input_channels-1] & (1<<i))
+                av_log(NULL, AV_LOG_ERROR, " %d", i + 1);
+        av_log(NULL, AV_LOG_ERROR, "\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= 2*s->output_channels*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->input_channels == 6 && s->output_channels ==2) {
+        buftmp3[0] = bufout[0];
+        buftmp3[1] = bufout[1];
+        surround_to_stereo(buftmp2, input, s->input_channels, nb_samples);
+    } 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) ||
+               (s->output_channels == 2 && s->input_channels == 6)) {
+        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

libavcodec/resample2.c

+/*
+ * audio resampling
+ * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * audio resampling
+ * @author Michael Niedermayer <michaelni@gmx.at>
+ */
+
+#include "libavutil/avassert.h"
+#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];
+
+    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->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;
+        }
+        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++){
+        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;
+        av_assert2(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;
+    }
+
+    return dst_index;
+}
+
+#endif

libavcodec/version.h

@@ -73,6 +73,9 @@
 #ifndef FF_API_CODEC_ID
 #define FF_API_CODEC_ID          (LIBAVCODEC_VERSION_MAJOR < 56)
 #endif
+#ifndef FF_API_AVCODEC_RESAMPLE
+#define FF_API_AVCODEC_RESAMPLE  (LIBAVCODEC_VERSION_MAJOR < 56)
+#endif
 #ifndef FF_API_MMI
 #define FF_API_MMI               (LIBAVCODEC_VERSION_MAJOR < 55)
 #endif