[Mplayer-cvslog] CVS: main/libaf af_lavcresample.c, NONE, 1.1 af.c, 1.31, 1.32 Makefile, 1.15, 1.16

[Michael Niedermayer CVS]

CVS change done by Michael Niedermayer CVS

Update of /cvsroot/mplayer/main/libaf
In directory mail:/var2/tmp/cvs-serv3434

Modified Files:
	af.c Makefile 
Added Files:
	af_lavcresample.c 
Log Message:
libavcodec resampling ...
libaf doesnt seem to support planar audio, so we need to convert it :(


--- NEW FILE ---
// Copyright (c) 2004 Michael Niedermayer <michaelni at 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

Index: af.c
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/af.c,v
retrieving revision 1.31
retrieving revision 1.32
diff -u -r1.31 -r1.32
--- af.c	5 Oct 2004 19:36:47 -0000	1.31
+++ af.c	21 Oct 2004 03:32:31 -0000	1.32
@@ -25,6 +25,7 @@
 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 @@
 #endif
    &af_info_volnorm,
    &af_info_extrastereo,
+#ifdef USE_LIBAVCODEC
+   &af_info_lavcresample,
+#endif
    NULL 
 };
 

Index: Makefile
===================================================================
RCS file: /cvsroot/mplayer/main/libaf/Makefile,v
retrieving revision 1.15
retrieving revision 1.16
diff -u -r1.15 -r1.16
--- Makefile	4 Oct 2004 19:11:05 -0000	1.15
+++ Makefile	21 Oct 2004 03:32:31 -0000	1.16
@@ -2,7 +2,7 @@
 
 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)