libavcodec resampling ...

libaf doesnt seem to support planar audio, so we need to convert it :(


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@13714 b3059339-0415-0410-9bf9-f77b7e298cf2

libaf/Makefile

@@ -2,7 +2,7 @@ include ../config.mak
 
 LIBNAME = libaf.a
 
-SRCS=af.c af_mp.c af_dummy.c af_delay.c af_channels.c af_format.c af_resample.c window.c filter.c af_volume.c af_equalizer.c af_tools.c af_comp.c af_gate.c af_pan.c af_surround.c af_sub.c af_export.c af_volnorm.c af_extrastereo.c
+SRCS=af.c af_mp.c af_dummy.c af_delay.c af_channels.c af_format.c af_resample.c window.c filter.c af_volume.c af_equalizer.c af_tools.c af_comp.c af_gate.c af_pan.c af_surround.c af_sub.c af_export.c af_volnorm.c af_extrastereo.c af_lavcresample.c
 
 OBJS=$(SRCS:.c=.o)
 

libaf/af.c

@@ -25,6 +25,7 @@ extern af_info_t af_info_sub;
 extern af_info_t af_info_export;
 extern af_info_t af_info_volnorm;
 extern af_info_t af_info_extrastereo;
+extern af_info_t af_info_lavcresample;
 
 static af_info_t* filter_list[]={ 
    &af_info_dummy,
@@ -44,6 +45,9 @@ static af_info_t* filter_list[]={
 #endif
    &af_info_volnorm,
    &af_info_extrastereo,
+#ifdef USE_LIBAVCODEC
+   &af_info_lavcresample,
+#endif
    NULL 
 };
 

libaf/af_lavcresample.c

@@ -0,0 +1,155 @@
+// Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
+// #inlcude <GPL_v2.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <inttypes.h>
+
+#include "../config.h"
+#include "af.h"
+
+#ifdef USE_LIBAVCODEC
+
+#include "../libavcodec/avcodec.h"
+#include "../libavcodec/rational.h"
+
+#define CHANS 6
+
+// Data for specific instances of this filter
+typedef struct af_resample_s{
+    struct AVResampleContext *avrctx;
+    int16_t *in[CHANS];
+    int in_alloc;
+    int index;
+    
+    int filter_length;
+    int linear;
+    int phase_shift;
+}af_resample_t;
+
+
+// Initialization and runtime control
+static int control(struct af_instance_s* af, int cmd, void* arg)
+{
+  af_resample_t* s   = (af_resample_t*)af->setup; 
+  af_data_t *data= (af_data_t*)arg;
+
+  switch(cmd){
+  case AF_CONTROL_REINIT:
+    if((af->data->rate == data->rate) || (af->data->rate == 0))
+        return AF_DETACH;
+
+    if(data->format != (AF_FORMAT_SI | AF_FORMAT_NE) || data->nch > CHANS)
+       return AF_ERROR;
+
+    af->data->nch    = data->nch;
+    af->data->format = AF_FORMAT_SI | AF_FORMAT_NE;
+    af->data->bps    = 2;
+    af->mul.n = af->data->rate;
+    af->mul.d = data->rate;
+    af->delay = 500*s->filter_length/(double)min(af->mul.n, af->mul.d);
+
+    if(s->avrctx) av_resample_close(s->avrctx);
+    s->avrctx= av_resample_init(af->mul.n, /*in_rate*/af->mul.d, s->filter_length, s->phase_shift, s->linear);
+
+    return AF_OK;
+  case AF_CONTROL_COMMAND_LINE:{
+    sscanf((char*)arg,"%d:%d:%d:%d", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift);
+    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_s* af)
+{
+    if(af->data)
+        free(af->data);
+    if(af->setup){
+        af_resample_t *s = af->setup;
+        if(s->avrctx) av_resample_close(s->avrctx);
+        free(s);
+    }
+}
+
+// Filter data through filter
+static af_data_t* play(struct af_instance_s* af, af_data_t* data)
+{    
+  af_resample_t *s = af->setup;
+  int i, j, consumed, ret;
+  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.n) / af->mul.d + 10;
+  int16_t tmp[CHANS][out_len];
+    
+  if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
+      return NULL;
+  
+  out= (int16_t*)af->data->audio;
+
+  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)); //FIXME free this maybe ;)
+      }
+  }
+
+  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));
+  }
+
+  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 open(af_instance_t* af){
+  af->control=control;
+  af->uninit=uninit;
+  af->play=play;
+  af->mul.n=1;
+  af->mul.d=1;
+  af->data=calloc(1,sizeof(af_data_t));
+  af->setup=calloc(1,sizeof(af_resample_t));
+  ((af_resample_t*)af->setup)->filter_length= 16;
+  ((af_resample_t*)af->setup)->phase_shift= 10;
+//  ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
+  return AF_OK;
+}
+
+af_info_t af_info_lavcresample = {
+  "Sample frequency conversion using libavcodec",
+  "lavcresample",
+  "Michael Niedermayer",
+  "",
+  AF_FLAGS_REENTRANT,
+  open
+};
+#endif