view libaf/af_lavcresample.c @ 16805:50fb26acbcba

processing audio is sometimes essential for a/v sync, so 1000l to whoever made rawvideo muxer disable audio!! with this patch, audio is processed but simply thrown away by the muxer. various 'error' conditions in rawvideo muxer are removed to make it work. feel free to re-add them if they can be done without breaking anything, but do not use printf !!!! btw old behavior can be obtained by manually specifying -nosound.

// Copyright (c) 2004 Michael Niedermayer <[e-mail]>
// #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

#ifdef USE_LIBAVCODEC_SO
#include <ffmpeg/avcodec.h>
#include <ffmpeg/rational.h>
#else
#include "avcodec.h"
#include "rational.h"
#endif

#define CHANS 6

int64_t ff_gcd(int64_t a, int64_t b);

// 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;
    double cutoff;
}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;
  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 > CHANS) af->data->nch = CHANS;
    af->data->format = AF_FORMAT_S16_NE;
    af->data->bps    = 2;
    af->mul.n = af->data->rate;
    af->mul.d = data->rate;
    af_frac_cancel(&af->mul);
    af->delay = 500*s->filter_length/(double)min(af->data->rate, data->rate);

    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, 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, (af_data_t*)arg);
    af->data->rate = out_rate;
    return test_output_res;
  case AF_CONTROL_COMMAND_LINE:{
    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 - 1.0/s->filter_length, 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_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;
  
  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)); //FIXME free this maybe ;)
      }
  }

  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 open(af_instance_t* af){
  af_resample_t *s = calloc(1,sizeof(af_resample_t));
  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));
  s->filter_length= 16;
  s->cutoff= max(1.0 - 1.0/s->filter_length, 0.80);
  s->phase_shift= 10;
//  s->setup = RSMP_INT | FREQ_SLOPPY;
  af->setup=s;
  return AF_OK;
}

af_info_t af_info_lavcresample = {
  "Sample frequency conversion using libavcodec",
  "lavcresample",
  "Michael Niedermayer",
  "",
  AF_FLAGS_REENTRANT,
  open
};
#endif