Index: cfg-common.h
===================================================================
RCS file: /cvsroot/mplayer/main/cfg-common.h,v
retrieving revision 1.138
diff -u -p -r1.138 cfg-common.h
--- cfg-common.h	19 Jan 2005 18:05:03 -0000	1.138
+++ cfg-common.h	20 Feb 2005 01:17:49 -0000
@@ -139,6 +139,7 @@
 
 #ifdef USE_LIBA52
         {"a52drc", &a52_drc_level, CONF_TYPE_FLOAT, CONF_RANGE, 0, 1, NULL},
+        {"a52float", &a52_float, CONF_TYPE_FLAG, 0, 0, 1, NULL},
 #endif
 
 // ------------------------- codec/vfilter options --------------------
@@ -315,6 +316,7 @@ extern int network_ipv4_only_proxy;
 #endif
 
 extern float a52_drc_level;
+extern int a52_float;
 
 /* defined in libmpdemux: */
 extern int hr_mp3_seek;
Index: libmpcodecs/ad_liba52.c
===================================================================
RCS file: /cvsroot/mplayer/main/libmpcodecs/ad_liba52.c,v
retrieving revision 1.14
diff -u -p -r1.14 ad_liba52.c
--- libmpcodecs/ad_liba52.c	21 Sep 2004 20:34:46 -0000	1.14
+++ libmpcodecs/ad_liba52.c	20 Feb 2005 01:17:49 -0000
@@ -2,6 +2,7 @@
 #include <stdlib.h>
 #include <unistd.h>
 #include <math.h>
+#include <assert.h>
 
 #include "config.h"
 #ifdef USE_LIBA52
@@ -13,17 +14,21 @@
 
 #include "cpudetect.h"
 
+#include "../libaf/af_format.h"
+
 #include "../liba52/a52.h"
 #include "../liba52/mm_accel.h"
 
 static sample_t * a52_samples;
 static a52_state_t a52_state;
 static uint32_t a52_flags=0;
+static uint32_t channel_map;
 
 #define DRC_NO_ACTION      0
 #define DRC_NO_COMPRESSION 1
 #define DRC_CALLBACK       2
 
+int a52_float = 0;
 float a52_drc_level = 1.0;
 static int a52_drc_action = DRC_NO_ACTION;
 
@@ -115,10 +120,29 @@ static int preinit(sh_audio_t *sh)
   /* Dolby AC3 audio: */
   /* however many channels, 2 bytes in a word, 256 samples in a block, 6 blocks in a frame */
   sh->audio_out_minsize=audio_output_channels*2*256*6;
+  if (a52_float) sh->audio_out_minsize *= 2;
   sh->audio_in_minsize=3840;
   return 1;
 }
 
+static int a52_resample_float(float *in, int16_t *out)
+{
+    unsigned long i;
+    float *p = (float*) out;
+    for (i = 0; i != 256; i++) {
+	unsigned long map = channel_map;
+	do {
+	    unsigned long ch = map & 15;
+	    if (ch == 15)
+		*p = 0;
+	    else
+		*p = in[i + ((ch-1)<<8)];
+	    p++;
+	} while ((map >>= 4));
+    }
+    return (int16_t*) p - out;
+}
+
 static int init(sh_audio_t *sh_audio)
 {
   uint32_t a52_accel=0;
@@ -169,6 +193,11 @@ while(sh_audio->channels>0){
 	    case 5: a52_flags=A52_3F2R; break;
 	    case 6: a52_flags=A52_3F2R|A52_LFE; break; /* 5.1*/
   }
+  if (a52_float) {
+      bias = 0;
+      sh_audio->sample_format = AF_FORMAT_FLOAT_NE;
+      sh_audio->samplesize = 4;
+  }
   /* test:*/
   flags=a52_flags|A52_ADJUST_LEVEL;
   mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags before a52_frame: 0x%X\n",flags);
@@ -178,6 +207,36 @@ while(sh_audio->channels>0){
   }
   mp_msg(MSGT_DECAUDIO,MSGL_V,"A52 flags after a52_frame: 0x%X\n",flags);
   /* frame decoded, let's init resampler:*/
+  channel_map = 0;
+  if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE) {
+      if (!(flags & A52_LFE)) {
+	  switch ((flags<<3) | sh_audio->channels) {
+	    case (A52_MONO    << 3) | 1: channel_map = 0x1; break;
+	    case (A52_CHANNEL << 3) | 2:
+	    case (A52_STEREO  << 3) | 2:
+	    case (A52_DOLBY   << 3) | 2: channel_map =    0x21; break;
+	    case (A52_2F1R    << 3) | 3: channel_map =   0x321; break;
+	    case (A52_2F2R    << 3) | 4: channel_map =  0x4321; break;
+	    case (A52_3F      << 3) | 5: channel_map = 0x2ff31; break;
+	    case (A52_3F2R    << 3) | 5: channel_map = 0x25431; break;
+	  }
+      } else if (sh_audio->channels == 6) {
+	  switch (flags & ~A52_LFE) {
+	    case A52_MONO   : channel_map = 0x12ffff; break;
+	    case A52_CHANNEL:
+	    case A52_STEREO :
+	    case A52_DOLBY  : channel_map = 0x1fff32; break;
+	    case A52_3F     : channel_map = 0x13ff42; break;
+	    case A52_2F1R   : channel_map = 0x1f4432; break;
+	    case A52_2F2R   : channel_map = 0x1f5432; break;
+	    case A52_3F2R   : channel_map = 0x136542; break;
+	  }
+      }
+      if (channel_map) {
+	  a52_resample = a52_resample_float;
+	  break;
+      }
+  } else
   if(a52_resample_init(a52_accel,flags,sh_audio->channels)) break;
   --sh_audio->channels; /* try to decrease no. of channels*/
 }
@@ -208,6 +267,8 @@ static int decode_audio(sh_audio_t *sh_a
     sample_t level=1, bias=384;
     int flags=a52_flags|A52_ADJUST_LEVEL;
     int i,len=-1;
+	if (sh_audio->sample_format == AF_FORMAT_FLOAT_NE)
+	    bias = 0;
 	if(!sh_audio->a_in_buffer_len) 
 	    if(a52_fillbuff(sh_audio)<0) return len; /* EOF */
 	sh_audio->a_in_buffer_len=0;
@@ -232,6 +293,7 @@ static int decode_audio(sh_audio_t *sh_a
 	    }
 	    len+=2*a52_resample(a52_samples,(int16_t *)&buf[len]);
 	}
+	assert(len <= maxlen);
   return len;
 }
 #endif
Index: DOCS/man/en/mplayer.1
===================================================================
RCS file: /cvsroot/mplayer/main/DOCS/man/en/mplayer.1,v
retrieving revision 1.876
diff -u -p -r1.876 mplayer.1
--- DOCS/man/en/mplayer.1	19 Feb 2005 14:58:39 -0000	1.876
+++ DOCS/man/en/mplayer.1	20 Feb 2005 01:17:53 -0000
@@ -709,6 +709,12 @@ This option only shows an effect if the 
 compression information.
 .
 .TP
+.B \-a52float
+Decode AC3 using native float sample format.
+This is useful if you use audio filters such as volume to reduce the
+error coming from float to int conversions.
+.
+.TP
 .B \-aid <ID> (also see \-alang)
 Select audio channel (MPEG: 0\-31, AVI/\:OGM: 1\-99, ASF/\:RM: 0\-127,
 VOB(AC3): 128\-159, VOB(LPCM): 160\-191, MPEG-TS 17\-8190).