af_lavrresample: add new resampling filter to replace the old ones

Remove `af_resample` and `af_lavcresample`. The former is a mess while the
latter uses an API that was long deprecated in libavcodec and is now removed.

`af_lavrresample` rougly has the same features and structure of
`af_lavcresample`.

libswresample fallback by wm4.

1 files changed, 0 insertions, 213 deletions

diff --git a/audio/filter/af_lavcresample.c b/audio/filter/af_lavcresample.c
deleted file mode 100644
index ce777fed31..0000000000
--- a/audio/filter/af_lavcresample.c
+++ /dev/null
@@ -1,213 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include "config.h"
-#include "af.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/rational.h"
-
-// Data for specific instances of this filter
-typedef struct af_resample_s{
-    struct AVResampleContext *avrctx;
-    int16_t *in[AF_NCH];
-    int in_alloc;
-    int index;
-
-    int filter_length;
-    int linear;
-    int phase_shift;
-    double cutoff;
-
-    int ctx_out_rate;
-    int ctx_in_rate;
-    int ctx_filter_size;
-    int ctx_phase_shift;
-    int ctx_linear;
-    double ctx_cutoff;
-}af_resample_t;
-
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
-  af_resample_t* s   = (af_resample_t*)af->setup;
-  struct mp_audio *data= (struct mp_audio*)arg;
-  int out_rate, test_output_res; // helpers for checking input format
-
-  switch(cmd){
-  case AF_CONTROL_REINIT:
-    if((af->data->rate == data->rate) || (af->data->rate == 0))
-        return AF_DETACH;
-
-    af->data->nch    = data->nch;
-    if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
-    af->data->format = AF_FORMAT_S16_NE;
-    af->data->bps    = 2;
-    af->mul = (double)af->data->rate / data->rate;
-    af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
-
-    if (s->ctx_out_rate != af->data->rate || s->ctx_in_rate != data->rate || s->ctx_filter_size != s->filter_length ||
-        s->ctx_phase_shift != s->phase_shift || s->ctx_linear != s->linear || s->ctx_cutoff != s->cutoff) {
-        if(s->avrctx) av_resample_close(s->avrctx);
-        s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);
-        s->ctx_out_rate    = af->data->rate;
-        s->ctx_in_rate     = data->rate;
-        s->ctx_filter_size = s->filter_length;
-        s->ctx_phase_shift = s->phase_shift;
-        s->ctx_linear      = s->linear;
-        s->ctx_cutoff      = s->cutoff;
-    }
-
-    // hack to make af_test_output ignore the samplerate change
-    out_rate = af->data->rate;
-    af->data->rate = data->rate;
-    test_output_res = af_test_output(af, (struct mp_audio*)arg);
-    af->data->rate = out_rate;
-    return test_output_res;
-  case AF_CONTROL_COMMAND_LINE:{
-    s->cutoff= 0.0;
-    sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
-    if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
-    return AF_OK;
-  }
-  case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
-    af->data->rate = *(int*)arg;
-    return AF_OK;
-  }
-  return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance* af)
-{
-    if(af->data)
-        free(af->data->audio);
-    free(af->data);
-    if(af->setup){
-        int i;
-        af_resample_t *s = af->setup;
-        if(s->avrctx) av_resample_close(s->avrctx);
-        for (i=0; i < AF_NCH; i++)
-            free(s->in[i]);
-        free(s);
-    }
-}
-
-// Filter data through filter
-static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
-{
-  af_resample_t *s = af->setup;
-  int i, j, consumed, ret = 0;
-  int16_t *in = (int16_t*)data->audio;
-  int16_t *out;
-  int chans   = data->nch;
-  int in_len  = data->len/(2*chans);
-  int out_len = in_len * af->mul + 10;
-  int16_t tmp[AF_NCH][out_len];
-
-  if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
-      return NULL;
-
-  out= (int16_t*)af->data->audio;
-
-  out_len= min(out_len, af->data->len/(2*chans));
-
-  if(s->in_alloc < in_len + s->index){
-      s->in_alloc= in_len + s->index;
-      for(i=0; i<chans; i++){
-          s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
-      }
-  }
-
-  if(chans==1){
-      memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
-  }else if(chans==2){
-      for(j=0; j<in_len; j++){
-          s->in[0][j + s->index]= *(in++);
-          s->in[1][j + s->index]= *(in++);
-      }
-  }else{
-      for(j=0; j<in_len; j++){
-          for(i=0; i<chans; i++){
-              s->in[i][j + s->index]= *(in++);
-          }
-      }
-  }
-  in_len += s->index;
-
-  for(i=0; i<chans; i++){
-      ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
-  }
-  out_len= ret;
-
-  s->index= in_len - consumed;
-  for(i=0; i<chans; i++){
-      memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
-  }
-
-  if(chans==1){
-      memcpy(out, tmp[0], out_len*sizeof(int16_t));
-  }else if(chans==2){
-      for(j=0; j<out_len; j++){
-          *(out++)= tmp[0][j];
-          *(out++)= tmp[1][j];
-      }
-  }else{
-      for(j=0; j<out_len; j++){
-          for(i=0; i<chans; i++){
-              *(out++)= tmp[i][j];
-          }
-      }
-  }
-
-  data->audio = af->data->audio;
-  data->len   = out_len*chans*2;
-  data->rate  = af->data->rate;
-  return data;
-}
-
-static int af_open(struct af_instance* af){
-  af_resample_t *s = calloc(1,sizeof(af_resample_t));
-  af->control=control;
-  af->uninit=uninit;
-  af->play=play;
-  af->mul=1;
-  af->data=calloc(1,sizeof(struct mp_audio));
-  s->filter_length= 16;
-  s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
-  s->phase_shift= 10;
-//  s->setup = RSMP_INT | FREQ_SLOPPY;
-  af->setup=s;
-  return AF_OK;
-}
-
-struct af_info af_info_lavcresample = {
-  "Sample frequency conversion using libavcodec",
-  "lavcresample",
-  "Michael Niedermayer",
-  "",
-  AF_FLAGS_REENTRANT,
-  af_open
-};