view decoders/libmpg123/decode_i386.c @ 590:1c8414cd5839

Introduced new Ogg Tremor decoder, based on the existing Ogg Vorbis decoder.
author Eric Wing <ewing@anscamobile.com>
date Wed, 27 Apr 2011 19:37:16 -0700
parents 7e08477b0fc1
children
line wrap: on
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/*
	decode_i386.c: decode for i386 (really faster?)

	copyright 1995-2006 by the mpg123 project - free software under the terms of the LGPL 2.1
	see COPYING and AUTHORS files in distribution or http://mpg123.org
	initially written by Michael Hipp

	slighlty optimized for machines without autoincrement/decrement.
	The performance is highly compiler dependend. Maybe
	the decode.c version for 'normal' processor may be faster
	even for Intel processors.
*/

#include "mpg123lib_intern.h"

int synth_1to1_8bit_i386(real *bandPtr,int channel, mpg123_handle *fr, int final)
{
  short samples_tmp[64];
  short *tmp1 = samples_tmp + channel;
  int i,ret;

  unsigned char *samples = fr->buffer.data;
  int pnt = fr->buffer.fill;
  fr->buffer.data = (unsigned char*) samples_tmp;
  fr->buffer.fill = 0;
  ret = opt_synth_1to1(fr)(bandPtr, channel, fr , 0);
  fr->buffer.data = samples;

  samples += channel + pnt;
  for(i=0;i<32;i++) {
    *samples = fr->conv16to8[*tmp1>>AUSHIFT];
    samples += 2;
    tmp1 += 2;
  }
  fr->buffer.fill = pnt + (final ? 64 : 0 );

  return ret;
}

int synth_1to1_8bit_mono_i386(real *bandPtr, mpg123_handle *fr)
{
  short samples_tmp[64];
  short *tmp1 = samples_tmp;
  int i,ret;
 
  unsigned char *samples = fr->buffer.data;
  int pnt = fr->buffer.fill;
  fr->buffer.data = (unsigned char*) samples_tmp;
  fr->buffer.fill = 0;
  ret = opt_synth_1to1(fr)(bandPtr, 0, fr, 0);
  fr->buffer.data = samples;

  samples += pnt;
  for(i=0;i<32;i++) {
    *samples++ = fr->conv16to8[*tmp1>>AUSHIFT];
    tmp1+=2;
  }
  fr->buffer.fill = pnt + 32;

  return ret;
}

int synth_1to1_8bit_mono2stereo_i386(real *bandPtr, mpg123_handle *fr)
{
  short samples_tmp[64];
  short *tmp1 = samples_tmp;
  int i,ret;

  unsigned char *samples = fr->buffer.data;
  int pnt = fr->buffer.fill;
  fr->buffer.data = (unsigned char*) samples_tmp;
  fr->buffer.fill = 0;
  ret = opt_synth_1to1(fr)(bandPtr, 0, fr, 0);
  fr->buffer.data = samples;

  samples += pnt;
  for(i=0;i<32;i++) {
    *samples++ = fr->conv16to8[*tmp1>>AUSHIFT];
    *samples++ = fr->conv16to8[*tmp1>>AUSHIFT];
    tmp1 += 2;
  }
  fr->buffer.fill = pnt + 64;

  return ret;
}

int synth_1to1_mono_i386(real *bandPtr, mpg123_handle *fr)
{
  short samples_tmp[64];
  short *tmp1 = samples_tmp;
  int i,ret;

  unsigned char *samples = fr->buffer.data;
  int pnt = fr->buffer.fill;
  fr->buffer.data = (unsigned char*) samples_tmp;
  fr->buffer.fill = 0;
  ret = opt_synth_1to1(fr)(bandPtr, 0, fr, 0);
  fr->buffer.data = samples;

  samples += pnt;
  for(i=0;i<32;i++) {
    *( (short *) samples) = *tmp1;
    samples += 2;
    tmp1 += 2;
  }
  fr->buffer.fill = pnt + 64;

  return ret;
}


int synth_1to1_mono2stereo_i386(real *bandPtr, mpg123_handle *fr)
{
  int i,ret;
  unsigned char *samples = fr->buffer.data;

  ret = opt_synth_1to1(fr)(bandPtr, 0, fr, 1);
  samples += fr->buffer.fill - 128;

  for(i=0;i<32;i++) {
    ((short *)samples)[1] = ((short *)samples)[0];
    samples+=4;
  }

  return ret;
}

/* needed for i386, i486 */
#ifdef OPT_I386_SYNTH
int synth_1to1_i386(real *bandPtr,int channel, mpg123_handle *fr, int final)
{
  static const int step = 2;
  short *samples = (short *) (fr->buffer.data + fr->buffer.fill);

  real *b0, **buf;
  int clip = 0; 
  int bo1;

  if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);

  if(!channel) {
    fr->bo[0]--;
    fr->bo[0] &= 0xf;
    buf = fr->real_buffs[0];
  }
  else {
    samples++;
    buf = fr->real_buffs[1];
  }

  if(fr->bo[0] & 0x1) {
    b0 = buf[0];
    bo1 = fr->bo[0];
    dct64_i386(buf[1]+((fr->bo[0]+1)&0xf),buf[0]+fr->bo[0],bandPtr);
  }
  else {
    b0 = buf[1];
    bo1 = fr->bo[0]+1;
    dct64_i386(buf[0]+fr->bo[0],buf[1]+fr->bo[0]+1,bandPtr);
  }
  
  {
    register int j;
    real *window = opt_decwin(fr) + 16 - bo1;

    for (j=16;j;j--,b0+=0x10,window+=0x20,samples+=step)
    {
      real sum;
      sum  = window[0x0] * b0[0x0];
      sum -= window[0x1] * b0[0x1];
      sum += window[0x2] * b0[0x2];
      sum -= window[0x3] * b0[0x3];
      sum += window[0x4] * b0[0x4];
      sum -= window[0x5] * b0[0x5];
      sum += window[0x6] * b0[0x6];
      sum -= window[0x7] * b0[0x7];
      sum += window[0x8] * b0[0x8];
      sum -= window[0x9] * b0[0x9];
      sum += window[0xA] * b0[0xA];
      sum -= window[0xB] * b0[0xB];
      sum += window[0xC] * b0[0xC];
      sum -= window[0xD] * b0[0xD];
      sum += window[0xE] * b0[0xE];
      sum -= window[0xF] * b0[0xF];

      WRITE_SAMPLE(samples,sum,clip);
    }

    {
      real sum;
      sum  = window[0x0] * b0[0x0];
      sum += window[0x2] * b0[0x2];
      sum += window[0x4] * b0[0x4];
      sum += window[0x6] * b0[0x6];
      sum += window[0x8] * b0[0x8];
      sum += window[0xA] * b0[0xA];
      sum += window[0xC] * b0[0xC];
      sum += window[0xE] * b0[0xE];
      WRITE_SAMPLE(samples,sum,clip);
      b0-=0x10,window-=0x20,samples+=step;
    }
    window += bo1<<1;

    for (j=15;j;j--,b0-=0x10,window-=0x20,samples+=step)
    {
      real sum;
      sum = -window[-0x1] * b0[0x0];
      sum -= window[-0x2] * b0[0x1];
      sum -= window[-0x3] * b0[0x2];
      sum -= window[-0x4] * b0[0x3];
      sum -= window[-0x5] * b0[0x4];
      sum -= window[-0x6] * b0[0x5];
      sum -= window[-0x7] * b0[0x6];
      sum -= window[-0x8] * b0[0x7];
      sum -= window[-0x9] * b0[0x8];
      sum -= window[-0xA] * b0[0x9];
      sum -= window[-0xB] * b0[0xA];
      sum -= window[-0xC] * b0[0xB];
      sum -= window[-0xD] * b0[0xC];
      sum -= window[-0xE] * b0[0xD];
      sum -= window[-0xF] * b0[0xE];
      sum -= window[-0x0] * b0[0xF];

      WRITE_SAMPLE(samples,sum,clip);
    }
  }
  if(final) fr->buffer.fill += 128;

  return clip;
}
#endif

#ifdef OPT_PENTIUM
int synth_1to1_i586(real *bandPtr,int channel, mpg123_handle *fr, int final)
{
	int ret;
	if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);

	/* this is in asm, can be dither or not */
	/* uh, is this return from pointer correct? */ 
	ret = (int) opt_synth_1to1_i586_asm(fr)(bandPtr, channel, fr->buffer.data+fr->buffer.fill, fr->rawbuffs, fr->bo, fr->decwin);
	if(final) fr->buffer.fill += 128;
	return ret;
}
#endif

#ifdef OPT_3DNOW
int synth_1to1_3dnow(real *bandPtr,int channel, mpg123_handle *fr, int final)
{
	int ret;

	if(fr->have_eq_settings) do_equalizer_3dnow(bandPtr,channel,fr->equalizer);

	/* this is in asm, can be dither or not */
	/* uh, is this return from pointer correct? */ 
	ret = (int) synth_1to1_3dnow_asm(bandPtr, channel, fr->buffer.data+fr->buffer.fill, fr->rawbuffs, fr->bo, fr->decwin);
	if(final) fr->buffer.fill += 128;
	return ret;
}
#endif

#ifdef OPT_MMX
/* wrapper for da interface */
int synth_1to1_mmx(real *bandPtr, int channel, mpg123_handle *fr, int final)
{
	if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);

	/* in asm */
	synth_1to1_MMX(bandPtr, channel, (short*) (fr->buffer.data+fr->buffer.fill), (short *) fr->rawbuffs, fr->bo, fr->decwins); 
	if(final) fr->buffer.fill += 128;
	return 0;
}
#endif

#ifdef OPT_SSE
int synth_1to1_sse(real *bandPtr, int channel, mpg123_handle *fr, int final)
{
	if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);

	synth_1to1_sse_asm(bandPtr, channel, (short*) (fr->buffer.data+fr->buffer.fill), (short *) fr->rawbuffs, fr->bo, fr->decwins); 
	if(final) fr->buffer.fill += 128;
	return 0;
}
#endif

#ifdef OPT_3DNOWEXT
int synth_1to1_3dnowext(real *bandPtr, int channel, mpg123_handle *fr, int final)
{
	if(fr->have_eq_settings) do_equalizer(bandPtr,channel,fr->equalizer);

	synth_1to1_3dnowext_asm(bandPtr, channel, (short*) (fr->buffer.data+fr->buffer.fill), (short *) fr->rawbuffs, fr->bo, fr->decwins); 
	if(final) fr->buffer.fill += 128;
	return 0;
}
#endif