mp3lib: drop internal mp3lib tree

Delete mp3lib which has been the default mp3 decoder until now. In
addition to being an unnecessary embedded library it now fails to
compile correctly with the new gcc-4.6, producing noise.

After the deletion the default decoder priority for mp3 will be first
libmpg123 (a newer version of the code that mp3lib was based on) if
available, then ffmp3float which should be available in all normal
compiles. I think that some tweaking may be required as these decoder
alternatives get wider testing, but any problems should be solvable
and there should be no need for mp3lib.

1 files changed, 0 insertions, 253 deletions

diff --git a/mp3lib/decod386.c b/mp3lib/decod386.c
deleted file mode 100644
index 82657aedc3..0000000000
--- a/mp3lib/decod386.c
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Modified for use with MPlayer, for details see the changelog at
- * http://svn.mplayerhq.hu/mplayer/trunk/
- * $Id$
- */
-
-/*
- * Mpeg Layer-1,2,3 audio decoder
- * ------------------------------
- * copyright (c) 1995,1996,1997 by Michael Hipp, All rights reserved.
- * See also 'README'
- *
- * 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 "config.h"
-
-#if 0
- /* old WRITE_SAMPLE */
-   /* is portable */
-#define WRITE_SAMPLE(samples,sum,clip) {                        \
-  if( (sum) > 32767.0) { *(samples) = 0x7fff; (clip)++; }       \
-  else if( (sum) < -32768.0) { *(samples) = -0x8000; (clip)++; }\
-  else { *(samples) = sum;  }                                   \
-}
-#else
- /* new WRITE_SAMPLE */
-
-/*
- * should be the same as the "old WRITE_SAMPLE" macro above, but uses
- * some tricks to avoid double->int conversions and floating point compares.
- *
- * Here's how it works:
- * ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) is
- * 0x0010000080000000LL in hex.  It computes 0x0010000080000000LL + sum
- * as a double IEEE fp value and extracts the low-order 32-bits from the
- * IEEE fp representation stored in memory.  The 2^56 bit in the constant
- * is intended to force the bits of "sum" into the least significant bits
- * of the double mantissa.  After an integer substraction of 0x80000000
- * we have the original double value "sum" converted to an 32-bit int value.
- *
- * (Is that really faster than the clean and simple old version of the macro?)
- */
-
-/*
- * On a SPARC cpu, we fetch the low-order 32-bit from the second 32-bit
- * word of the double fp value stored in memory.  On an x86 cpu, we fetch it
- * from the first 32-bit word.
- * I'm not sure if the HAVE_BIGENDIAN feature test covers all possible memory
- * layouts of double floating point values an all cpu architectures.  If
- * it doesn't work for you, just enable the "old WRITE_SAMPLE" macro.
- */
-#if HAVE_BIGENDIAN
-#define MANTISSA_OFFSET 1
-#else
-#define MANTISSA_OFFSET 0
-#endif
-
-   /* sizeof(int) == 4 */
-#define WRITE_SAMPLE(samples,sum,clip) { \
-  union { double dtemp; int itemp[2]; } u; int v; \
-  u.dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
-  v = u.itemp[MANTISSA_OFFSET] - 0x80000000; \
-  if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
-  else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
-  else { *(samples) = v; } \
-}
-#endif
-
-
-/*
-#define WRITE_SAMPLE(samples,sum,clip) { \
-  double dtemp; int v;                    \
-  dtemp = ((((65536.0 * 65536.0 * 16)+(65536.0 * 0.5))* 65536.0)) + (sum);\
-  v = ((*(int *)&dtemp) - 0x80000000); \
-  if( v > 32767) { *(samples) = 0x7fff; (clip)++; } \
-  else if( v < -32768) { *(samples) = -0x8000; (clip)++; } \
-  else { *(samples) = v; } \
-}
-*/
-
-static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt);
-
-static int synth_1to1_mono2stereo(real *bandPtr,unsigned char *samples,int *pnt)
-{
-  int i,ret;
-
-  ret = synth_1to1(bandPtr,0,samples,pnt);
-  samples = samples + *pnt - 128;
-
-  for(i=0;i<32;i++) {
-    ((short *)samples)[1] = ((short *)samples)[0];
-    samples+=4;
-  }
-
-  return ret;
-}
-
-static synth_func_t synth_func;
-
-#if HAVE_ALTIVEC
-#define dct64_base(a,b,c) if(gCpuCaps.hasAltiVec) dct64_altivec(a,b,c); else dct64(a,b,c)
-#else /* HAVE_ALTIVEC */
-#define dct64_base(a,b,c) dct64(a,b,c)
-#endif /* HAVE_ALTIVEC */
-
-static int synth_1to1(real *bandPtr,int channel,unsigned char *out,int *pnt)
-{
-  static real buffs[2][2][0x110];
-  static const int step = 2;
-  static int bo = 1;
-  short *samples = (short *) (out + *pnt);
-  real *b0,(*buf)[0x110];
-  int clip = 0;
-  int bo1;
-
-  *pnt += 128;
-
-/* optimized for x86 */
-#if ARCH_X86
-  if ( synth_func )
-   {
-//    printf("Calling %p, bandPtr=%p channel=%d samples=%p\n",synth_func,bandPtr,channel,samples);
-    // FIXME: synth_func() may destroy EBP, don't rely on stack contents!!!
-    return (*synth_func)( bandPtr,channel,samples);
-   }
-#endif
-  if(!channel) {     /* channel=0 */
-    bo--;
-    bo &= 0xf;
-    buf = buffs[0];
-  }
-  else {
-    samples++;
-    buf = buffs[1];
-  }
-
-  if(bo & 0x1) {
-    b0 = buf[0];
-    bo1 = bo;
-    dct64_base(buf[1]+((bo+1)&0xf),buf[0]+bo,bandPtr);
-  }
-  else {
-    b0 = buf[1];
-    bo1 = bo+1;
-    dct64_base(buf[0]+bo,buf[1]+bo+1,bandPtr);
-  }
-
-  {
-    register int j;
-    real *window = mp3lib_decwin + 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);
-    }
-  }
-
-  return clip;
-
-}
-
-#ifdef CONFIG_FAKE_MONO
-static int synth_1to1_l(real *bandPtr,int channel,unsigned char *out,int *pnt)
-{
-  int i,ret;
-
-  ret = synth_1to1(bandPtr,channel,out,pnt);
-  out = out + *pnt - 128;
-
-  for(i=0;i<32;i++) {
-    ((short *)out)[1] = ((short *)out)[0];
-    out+=4;
-  }
-
-  return ret;
-}
-
-static int synth_1to1_r(real *bandPtr,int channel,unsigned char *out,int *pnt)
-{
-  int i,ret;
-
-  ret = synth_1to1(bandPtr,channel,out,pnt);
-  out = out + *pnt - 128;
-
-  for(i=0;i<32;i++) {
-    ((short *)out)[0] = ((short *)out)[1];
-    out+=4;
-  }
-
-  return ret;
-}
-#endif